From mario_fs at mgtech.com  Fri May  1 00:36:35 2015
From: mario_fs at mgtech.com (Mario)
Date: Thu, 30 Apr 2015 13:36:35 -0700
Subject: [Freeswitch-users] Custom Ringback
In-Reply-To: <1430215614081-7596156.post@n2.nabble.com>
References: <1430215614081-7596156.post@n2.nabble.com>
Message-ID: <67F5E6F2-8A91-40AD-BC47-69138E72EE4F@mgtech.com>

I use modify early media a lot for different reasons and it works fine. A couple of things to check:

1. The biggie? does your ITSP support changing early media? One of my ITSPs used to then stopped so I changed ITSPs.

2. See https://wiki.freeswitch.org/wiki/Mod_local_stream <https://wiki.freeswitch.org/wiki/Mod_local_stream>, I added the timeout stuff at the bottom which shows syntax for a file.

3. Did you check the FreeSWITCH log for the call to make sure the early media file was found and used properly?

Mario G

> On Apr 28, 2015, at 3:06 AM, ankitdoshi <ankit_doshi2011 at yahoo.com> wrote:
> 
> Hello friends,
> 
> I am newbie for ringback
> http://https://wiki.freeswitch.org/wiki/Variable_ringback <http://>  
> i need something like whenever i will make call until call answer ring must
> be replace with my custom ring 
> i have tried different solutions for that but didn't get exact solution as i
> need 
> 
> My current dialplan show me like
> 
> <action application="bridge"
> data={ring_back=file_string:///usr/local/freeswitch/sounds/en/us/callie/promofiles/sound_file.wav,ignore_early_media=true}[leg_timeout=30]sofia/gateway/gateway_name/XXXXX"/>
> 
> 
> but its didn't work
> 
> is it right way to make a ringback as dialplan shows ?
> 
> Please let me know if i am doing anything wrong or need to change something
> to make it work
> 
> Thanks in advance for suggestions and help
> 
> 
> 
> 
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Custom-Ringback-tp7596156.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 bote_radio at botecomm.com  Fri May  1 08:09:53 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Fri, 1 May 2015 00:09:53 -0400
Subject: [Freeswitch-users] voicemail + S3
In-Reply-To: <CAO3hyzFtEv2ydxjhWO-fXTDfR80fFEknK4-foMzEdQ+0n2LErQ@mail.gmail.com>
References: <CALJtzxDRev2yQgKN4hnVTjL4A0DqgJyrzUCAeSKv9UpFOpQ7ow@mail.gmail.com>	<CAFiqYum6v2A6QB=H5keDzpWFTiVjJYwr+Dn_YOnPLCsPVHF1_g@mail.gmail.com>	<CALJtzxC3x3KLaq+fGxko6q+aWoQz5a0BVL_tFXZ6MQrbwvjCRA@mail.gmail.com>	<CAFiqYun=ty6fEK0AiJG9jD5zLriy-MvOBEQS+dOC4WP=uZ5Epw@mail.gmail.com>	<CALJtzxDvyj1-B1jC63yHK3KkW0fkC8ggUTPfcoaz1ad6=e-j5A@mail.gmail.com>	<CAO3hyzH7eeE7qBc5t7KADz8nDenw67Kd4vrdY=ch0HUkESFnYw@mail.gmail.com>
	<CAO3hyzFtEv2ydxjhWO-fXTDfR80fFEknK4-foMzEdQ+0n2LErQ@mail.gmail.com>
Message-ID: <002401d083c4$ad5e9530$081bbf90$@botecomm.com>

The default dialplan that ships with the vanilla configuration files for FreeSWITCH contains an example of routing based on time of day and day of week. Perhaps that could also be used to determine voice mail greetings to play, with some additional work.

 

Bote

 

 

From: Chris Tunbridge
Sent: Thursday, 30 April, 2015 13:23
Subject: Re: [Freeswitch-users] voicemail + S3

 

Alejandro the way you would change the voicemail greeting is by using the following action

<action application="set" data="voicemail_greeting_number=1"/>

setting the number to a different one depending on the greeting # you want to play.

You can control the time based stuff by using conditions, see here for more information https://freeswitch.org/confluence/display/FREESWITCH/Time+of+Day+and+Holiday+Routing

 

On Thu, Apr 30, 2015 at 11:19 AM, Chris Tunbridge <blasterjr at gmail.com> wrote:

Alejandro i figured I'd chime in, i am the one who submitted that ticket regarding http_cache+voicemail, the issue is the system doesn't read the http_put status and thinks that the voicemail didn't get stored so it never makes a record.

S3FS works okay for voice mails, it'd be nice is the voice mail module was updated to function with the http_cache module as this would help out with scaling deployments.

 

On Thu, Apr 30, 2015 at 11:11 AM, Alejandro <cdgraff at gmail.com> wrote:

hi Steven, 

 

thanks a lot for the time to gide me, during mean time will use s3fs (i'm not big fan, but i has this into other services into production and is a good alternative, and can pass the issue of use http_cache module to upload.)

 

just one more question... i like to give the user the option to has custom Voicemail Grettings recorded by some profesional presenter, and rotate in base to the hour of the day automatically.

 

Something like, from 9am to 5pm, play gretting_1.mp3 and from 5pm to 9am grettings_2.mp3

 

Can teel me if that be possible and guide me to some documentation or link to investigate more about this?

 

THanks again, i'm working with freeswitch just a few hours, and is really amazing, I come from Asterisk and feel more flexible and powerfull freeswitch.


Regards from Argentina

Alejandro 

 

2015-04-30 13:35 GMT-03:00 Steven Ayre <steveayre at gmail.com>:

 

between description and comments
http_cache.conf, s3 profile

voicemail.conf, see the storage-dir param towards the bottom

 

But as I say it's not yet functional - the comments show it was failing as mod_voicemail checks if the file exists which mod_http_cache seems not to support yet and I don't see any mention of FS-7280 in the commit log since the ticket was created.

 

 

On 30 April 2015 at 14:31, Alejandro <cdgraff at gmail.com> wrote:

Thanks Steven, to point me to the attach, I previously read the ticket, but don't see the attached file.

 

Regards, 

Alejandro

 

2015-04-30 10:23 GMT-03:00 Steven Ayre <steveayre at gmail.com>:

 

There's a sample config attached to https://freeswitch.org/jira/browse/FS-7280, but from that ticket I'm not sure that it is actually functioning yet.

 

On 30 April 2015 at 13:24, Alejandro <cdgraff at gmail.com> wrote:

Hi all, 

 

I read that if I combine, mod_voicemail with mod_http_cache I can save my voicemail message directly into S3 storage.

 

Any are using this implementation? can share with me some config file to understand how i need setup this?

 

Thanks in advance

Alejandro

 

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

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

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

 


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

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

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

 


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

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

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

 


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

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

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

 


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

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

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

 

 

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From tculjaga at gmail.com  Fri May  1 15:29:30 2015
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 1 May 2015 13:29:30 +0200
Subject: [Freeswitch-users] T.30 faxing - V.27ter
In-Reply-To: <CAJoKKdUZq-6nJ+XPXCMXLbPaxNz3bfkZMw=_cAs7xT0FzZn63A@mail.gmail.com>
References: <CAJoKKdXF9v6BKsiJQrOov9AD4aktFT4wMA+xHH6XuMnJAYu=Dg@mail.gmail.com>
	<CAEJMVkBAvp9Vxv3-cZgVKdxVi1m7H_03GDTL9iVpM9fUiSQqXg@mail.gmail.com>
	<CAJoKKdUZq-6nJ+XPXCMXLbPaxNz3bfkZMw=_cAs7xT0FzZn63A@mail.gmail.com>
Message-ID: <CAJoKKdX+L9GjLasaecL5L1SrJt7UbVqitLk8Syng8mZTAyL9tw@mail.gmail.com>

does anyone have a clue ?

On Wed, Apr 22, 2015 at 11:53 AM, Tihomir Culjaga <tculjaga at gmail.com>
wrote:

> you are right!
>
> trouble making fax machine is: Samsung Xpress M2875FD
>
> On Wed, Apr 22, 2015 at 10:40 AM, Brian West <brian at freeswitch.org> wrote:
>
>> JIRA FS-7453:  It's helpful to reference it in your post so others can
>> comment on it too, keeping all relevant info in one place.
>>
>> Thanks,
>> /b
>>
>>
>> On Wednesday, April 22, 2015, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> i got an issue with faxing between a fax machine and FreeSWITCH T.30
>>> protocol.
>>>
>>> I'm able to send and get faxes everywhere except to this machine.
>>> The fax machine uses V.27ter protocol at 4800 and no ECM.
>>> faxing from a different fax machine works perfectly but not when i send
>>> it from FS.
>>>
>>> FS logs are here: https://pastebin.freeswitch.org/24139
>>>
>>> I extracted audio
>>>  and tried to decode it with fax_decode ( from spends/tests ) and it
>>> fails to decode as well.
>>> Also, i tried to decode the audio between two fax machines when the fax
>>> went ok and it fax_decode fails on this one as well.
>>>
>>> fax_decode logs:
>>> https://pastebin.freeswitch.org/24140
>>>
>>>
>>> so maybe we have an issue with V.27ter perhaps.
>>>
>>> can anyone look say a bit more on this issue ?
>>>
>>> P.S.: FreeSWITCH Version 1.4.18+git~20150312T185523Z~4eed221b69~32bit
>>> (git 4eed221 2015-03-12 18:55:23Z 32bit)
>>>
>>>
>>> regards,
>>> Tihomir.
>>>
>>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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  Fri May  1 18:00:41 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 01 May 2015 14:00:41 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <55438709554d6_2fbe90f330703c@resque-worker.18.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From brian at freeswitch.org  Fri May  1 18:08:26 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 1 May 2015 09:08:26 -0500
Subject: [Freeswitch-users] Join us live today on the VUC
	(http://www.vuc.me) @ noon EST
Message-ID: <CAEJMVkDCpXj7aUrgPDenGCRRvRhUU49adeFpqEvnWmC61GcttQ@mail.gmail.com>

More Information at:

https://www.youtube.com/watch?v=6mLik6RzeZk

Be sure to follow us on twitter @freeswitch and @cluecon

Thanks,

-- 

*Brian West*
brian at freeswitch.org


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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 phenix at vfemail.net  Fri May  1 20:09:29 2015
From: phenix at vfemail.net (Tanguy)
Date: Fri, 01 May 2015 18:09:29 +0200
Subject: [Freeswitch-users] sip_to_user and destination number
Message-ID: <5543A539.3000704@vfemail.net>

Hello,

My provider did not send correct DID number in the INVITE packet but i 
can use "To" argument

INVITE 
sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429 at 92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429 
SIP/2.0.
Call-ID: 25016-VB-188fd96b-526e3dbd4 at sip.ovh.fr.
Contact: <sip:10.7.1.60:5060>.
Content-Type: application/sdp.
CSeq: 403749831 INVITE.
From: "0967212xxx" 
<sip:0967212xxx at sip.ovh.fr;user=phone>;tag=25016-VE-188fd96c-18bb43586.
Max-Forwards: 27.
Record-Route: <sip:91.121.129.20:5060;lr>.
*To: <sip:0557590xxx at 10.7.1.60;user=phone>.*
Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-WGZO-1fe73949-2df58378.

Using asterisk i can bypass the issue using something like exten => 
s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) but i am 
unable to do the same under freeswitch.

My trunk configuration seems correct, as you can see i used  
auto_to_user, but the destination number remains 0033972480xxx when i 
call 0557590xxx.

2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635 Processing 
0967212xxx <0967212xxx>->0033972480xxx in context public


<include>
<gateway name="0a96c3d3-0b0e-4864-b9ec-759fa4422429">
<param name="username" value="0033972480xxx"/>
<param name="password" value="xxxxxxx"/>
<param name="proxy" value="sip.ovh.fr"/>
<param name="expire-seconds" value="800"/>
<param name="register" value="true"/>
<param name="retry-seconds" value="30"/>
<param name="extension" value="auto_to_user"/>
<param name="context" value="public"/>
</gateway>
</include>

I tried to edit my inbound dialplan manually, it works using <condition 
field="${sip_to_user}" expression="0557590xxx" > but i prefer a proper 
way to do this because i will also use telcos with normal invite packets

I how i can copy $sip_to_header to destination for this specific trunk ?

Please note that i use fusionpbx.

Best regards, sorry for my bad English



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From jdickson at evolvetsi.com  Fri May  1 20:50:49 2015
From: jdickson at evolvetsi.com (Joseph Dickson)
Date: Fri, 1 May 2015 12:50:49 -0400
Subject: [Freeswitch-users] freeswitch systemd unit file for debian?
Message-ID: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>

Happy Friday!

I'm having trouble using latest release systemd unit file on Debian
Jessie..  It looks like the unit file is the same in master, so I imagine
the issue exists there too..

On my system (fresh Debian 8 install), I get the following failure when
trying to start using the included unit file:

May 01 12:48:22 XXX systemd[9119]: Failed at step CHDIR spawning
/bin/mkdir: No such file or directory

I'm new to systemd, but it looks like the problem is that the
WorkingDirectory is set to /run/freeswitch.  Trouble is that
/run/freeswitch is created in an ExecStartPre statement.  That's the best
explanation I have for the CHDIR failure that systemd is complaining about.

It looks like the only way to get /run/freeswitch created soon enough to be
used as a WorkingDirectory is the tmpfiles.d mechanism that systemd has.
Am I on the right track, or am I missing an obvious solution?

Thanks!

Joseph Dickson
jdickson at evolvetsi.com
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From vipkilla at gmail.com  Fri May  1 21:40:00 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 1 May 2015 13:40:00 -0400
Subject: [Freeswitch-users] freeswitch systemd unit file for debian?
In-Reply-To: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>
References: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>
Message-ID: <CAC-LwPPtPBmZpYeRpBZqg22KYuDKJDbRhZnhvg9S_HNXtA1oGA@mail.gmail.com>

Hello,
I modified the path variables in the systemd init file.
My file looks like this:

;;;;; Author: Travis Cross <tc at traviscross.com>

[Unit]
Description=freeswitch
After=syslog.target network.target local-fs.target

[Service]
; service
Type=forking
PIDFile=/usr/local/freeswitch/run/freeswitch.pid
PermissionsStartOnly=true
ExecStartPre=/bin/mkdir -p /usr/local/freeswitch/run
ExecStartPre=/bin/chown freeswitch:freeswitch /usr/local/freeswitch/run
ExecStart=/usr/bin/freeswitch -ncwait -nonat
TimeoutSec=45s
Restart=always
; exec
WorkingDirectory=/usr/local/freeswitch/run
User=freeswitch
Group=freeswitch
LimitCORE=infinity
LimitNOFILE=100000
LimitNPROC=60000
;LimitSTACK=240
LimitRTPRIO=infinity
LimitRTTIME=7000000
IOSchedulingClass=realtime
IOSchedulingPriority=2
CPUSchedulingPolicy=rr
CPUSchedulingPriority=89
UMask=0007

[Install]
WantedBy=multi-user.target


On Fri, May 1, 2015 at 12:50 PM, Joseph Dickson <jdickson at evolvetsi.com>
wrote:

> Happy Friday!
>
> I'm having trouble using latest release systemd unit file on Debian
> Jessie..  It looks like the unit file is the same in master, so I imagine
> the issue exists there too..
>
> On my system (fresh Debian 8 install), I get the following failure when
> trying to start using the included unit file:
>
> May 01 12:48:22 XXX systemd[9119]: Failed at step CHDIR spawning
> /bin/mkdir: No such file or directory
>
> I'm new to systemd, but it looks like the problem is that the
> WorkingDirectory is set to /run/freeswitch.  Trouble is that
> /run/freeswitch is created in an ExecStartPre statement.  That's the best
> explanation I have for the CHDIR failure that systemd is complaining about.
>
> It looks like the only way to get /run/freeswitch created soon enough to
> be used as a WorkingDirectory is the tmpfiles.d mechanism that systemd
> has.  Am I on the right track, or am I missing an obvious solution?
>
> Thanks!
>
> Joseph Dickson
> jdickson at evolvetsi.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 mike at jerris.com  Fri May  1 21:56:45 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 1 May 2015 13:56:45 -0400
Subject: [Freeswitch-users] freeswitch systemd unit file for debian?
In-Reply-To: <CAC-LwPPtPBmZpYeRpBZqg22KYuDKJDbRhZnhvg9S_HNXtA1oGA@mail.gmail.com>
References: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>
	<CAC-LwPPtPBmZpYeRpBZqg22KYuDKJDbRhZnhvg9S_HNXtA1oGA@mail.gmail.com>
Message-ID: <A706245D-22A6-48FB-9361-B3317A7F30AA@jerris.com>

If you have a working one, please get a pull request to us so we can review the changes.

Thanks
Mike

> On May 1, 2015, at 1:40 PM, Vik Killa <vipkilla at gmail.com> wrote:
> 
> Hello,
> I modified the path variables in the systemd init file. 
> My file looks like this:
> 
> ;;;;; Author: Travis Cross <tc at traviscross.com <mailto:tc at traviscross.com>>
> 
> [Unit]
> Description=freeswitch
> After=syslog.target network.target local-fs.target
> 
> [Service]
> ; service
> Type=forking
> PIDFile=/usr/local/freeswitch/run/freeswitch.pid
> PermissionsStartOnly=true
> ExecStartPre=/bin/mkdir -p /usr/local/freeswitch/run
> ExecStartPre=/bin/chown freeswitch:freeswitch /usr/local/freeswitch/run
> ExecStart=/usr/bin/freeswitch -ncwait -nonat
> TimeoutSec=45s
> Restart=always
> ; exec
> WorkingDirectory=/usr/local/freeswitch/run
> User=freeswitch
> Group=freeswitch
> LimitCORE=infinity
> LimitNOFILE=100000
> LimitNPROC=60000
> ;LimitSTACK=240
> LimitRTPRIO=infinity
> LimitRTTIME=7000000
> IOSchedulingClass=realtime
> IOSchedulingPriority=2
> CPUSchedulingPolicy=rr
> CPUSchedulingPriority=89
> UMask=0007
> 
> [Install]
> WantedBy=multi-user.target
> 
> 
> On Fri, May 1, 2015 at 12:50 PM, Joseph Dickson <jdickson at evolvetsi.com <mailto:jdickson at evolvetsi.com>> wrote:
> Happy Friday!
> 
> I'm having trouble using latest release systemd unit file on Debian Jessie..  It looks like the unit file is the same in master, so I imagine the issue exists there too..
> 
> On my system (fresh Debian 8 install), I get the following failure when trying to start using the included unit file:
> 
> May 01 12:48:22 XXX systemd[9119]: Failed at step CHDIR spawning /bin/mkdir: No such file or directory
> 
> I'm new to systemd, but it looks like the problem is that the WorkingDirectory is set to /run/freeswitch.  Trouble is that /run/freeswitch is created in an ExecStartPre statement.  That's the best explanation I have for the CHDIR failure that systemd is complaining about.
> 
> It looks like the only way to get /run/freeswitch created soon enough to be used as a WorkingDirectory is the tmpfiles.d mechanism that systemd has.  Am I on the right track, or am I missing an obvious solution?
> 
> Thanks!
> 
> Joseph Dickson
> jdickson at evolvetsi.com <mailto:jdickson at evolvetsi.com>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> 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 ssinyagin at gmail.com  Fri May  1 22:04:46 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 1 May 2015 20:04:46 +0200
Subject: [Freeswitch-users] sip_to_user and destination number
In-Reply-To: <5543A539.3000704@vfemail.net>
References: <5543A539.3000704@vfemail.net>
Message-ID: <CACOYK=hhrQ-dJoF_vpYKkceh1M=iHcc_AhYj2Rb1+4_LCNCgFw@mail.gmail.com>

Remove the extension parameter and see if it helps.
On May 1, 2015 6:11 PM, "Tanguy" <phenix at vfemail.net> wrote:

>  Hello,
>
> My provider did not send correct DID number in the INVITE packet but i can
> use "To" argument
>
> INVITE
> sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429 at 92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429
> SIP/2.0.
> Call-ID: 25016-VB-188fd96b-526e3dbd4 at sip.ovh.fr.
> Contact: <sip:10.7.1.60:5060>.
> Content-Type: application/sdp.
> CSeq: 403749831 INVITE.
> From: "0967212xxx" <sip:0967212xxx at sip.ovh.fr;user=phone>
> <sip:0967212xxx at sip.ovh.fr;user=phone>;tag=25016-VE-188fd96c-18bb43586.
> Max-Forwards: 27.
> Record-Route: <sip:91.121.129.20:5060;lr>.
> *To: <sip:0557590xxx at 10.7.1.60;user=phone>
> <sip:0557590xxx at 10.7.1.60;user=phone>.*
> Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-WGZO-1fe73949-2df58378.
>
> Using asterisk i can bypass the issue using something like  exten =>
> s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) but i am
> unable to do the same under freeswitch.
>
> My trunk configuration seems correct, as you can see i used  auto_to_user,
> but the destination number remains 0033972480xxx when i call 0557590xxx.
>
> 2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635 Processing
> 0967212xxx <0967212xxx>->0033972480xxx in context public
>
>
> <include>
>     <gateway name="0a96c3d3-0b0e-4864-b9ec-759fa4422429">
>       <param name="username" value="0033972480xxx"/>
>       <param name="password" value="xxxxxxx"/>
>       <param name="proxy" value="sip.ovh.fr"/>
>       <param name="expire-seconds" value="800"/>
>       <param name="register" value="true"/>
>       <param name="retry-seconds" value="30"/>
>       <param name="extension" value="auto_to_user"/>
>       <param name="context" value="public"/>
>     </gateway>
> </include>
>
> I tried to edit my inbound dialplan manually, it works using  <condition
> field="${sip_to_user}" expression="0557590xxx" > but i prefer a proper way
> to do this because i will also use telcos with normal invite packets
>
> I how i can copy $sip_to_header to destination for this specific trunk ?
>
> Please note that i use fusionpbx.
>
> Best regards, sorry for my bad English
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jdickson at evolvetsi.com  Fri May  1 22:07:22 2015
From: jdickson at evolvetsi.com (Joseph Dickson)
Date: Fri, 1 May 2015 14:07:22 -0400
Subject: [Freeswitch-users] freeswitch systemd unit file for debian?
In-Reply-To: <A706245D-22A6-48FB-9361-B3317A7F30AA@jerris.com>
References: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>
	<CAC-LwPPtPBmZpYeRpBZqg22KYuDKJDbRhZnhvg9S_HNXtA1oGA@mail.gmail.com>
	<A706245D-22A6-48FB-9361-B3317A7F30AA@jerris.com>
Message-ID: <CAOHpEota04EeGXSE9_x2ohq2S4m9GbCMAQpZkUbWVYEvSFY8YQ@mail.gmail.com>

I've got one that works now after discovering that RuntimeDirectory= will
create a directory for you under /run..  I'll try to get you the pull
request next week

On Fri, May 1, 2015 at 1:56 PM, Michael Jerris <mike at jerris.com> wrote:

> If you have a working one, please get a pull request to us so we can
> review the changes.
>
> Thanks
> Mike
>
> On May 1, 2015, at 1:40 PM, Vik Killa <vipkilla at gmail.com> wrote:
>
> Hello,
> I modified the path variables in the systemd init file.
> My file looks like this:
>
> ;;;;; Author: Travis Cross <tc at traviscross.com>
>
> [Unit]
> Description=freeswitch
> After=syslog.target network.target local-fs.target
>
> [Service]
> ; service
> Type=forking
> PIDFile=/usr/local/freeswitch/run/freeswitch.pid
> PermissionsStartOnly=true
> ExecStartPre=/bin/mkdir -p /usr/local/freeswitch/run
> ExecStartPre=/bin/chown freeswitch:freeswitch /usr/local/freeswitch/run
> ExecStart=/usr/bin/freeswitch -ncwait -nonat
> TimeoutSec=45s
> Restart=always
> ; exec
> WorkingDirectory=/usr/local/freeswitch/run
> User=freeswitch
> Group=freeswitch
> LimitCORE=infinity
> LimitNOFILE=100000
> LimitNPROC=60000
> ;LimitSTACK=240
> LimitRTPRIO=infinity
> LimitRTTIME=7000000
> IOSchedulingClass=realtime
> IOSchedulingPriority=2
> CPUSchedulingPolicy=rr
> CPUSchedulingPriority=89
> UMask=0007
>
> [Install]
> WantedBy=multi-user.target
>
>
> On Fri, May 1, 2015 at 12:50 PM, Joseph Dickson <jdickson at evolvetsi.com>
> wrote:
>
>> Happy Friday!
>>
>> I'm having trouble using latest release systemd unit file on Debian
>> Jessie..  It looks like the unit file is the same in master, so I imagine
>> the issue exists there too..
>>
>> On my system (fresh Debian 8 install), I get the following failure when
>> trying to start using the included unit file:
>>
>> May 01 12:48:22 XXX systemd[9119]: Failed at step CHDIR spawning
>> /bin/mkdir: No such file or directory
>>
>> I'm new to systemd, but it looks like the problem is that the
>> WorkingDirectory is set to /run/freeswitch.  Trouble is that
>> /run/freeswitch is created in an ExecStartPre statement.  That's the best
>> explanation I have for the CHDIR failure that systemd is complaining about.
>>
>> It looks like the only way to get /run/freeswitch created soon enough to
>> be used as a WorkingDirectory is the tmpfiles.d mechanism that systemd
>> has.  Am I on the right track, or am I missing an obvious solution?
>>
>> Thanks!
>>
>> Joseph Dickson
>> jdickson at evolvetsi.com
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mario_fs at mgtech.com  Fri May  1 22:19:20 2015
From: mario_fs at mgtech.com (Mario)
Date: Fri, 1 May 2015 11:19:20 -0700
Subject: [Freeswitch-users] voicemail + S3
In-Reply-To: <002401d083c4$ad5e9530$081bbf90$@botecomm.com>
References: <CALJtzxDRev2yQgKN4hnVTjL4A0DqgJyrzUCAeSKv9UpFOpQ7ow@mail.gmail.com>
	<CAFiqYum6v2A6QB=H5keDzpWFTiVjJYwr+Dn_YOnPLCsPVHF1_g@mail.gmail.com>
	<CALJtzxC3x3KLaq+fGxko6q+aWoQz5a0BVL_tFXZ6MQrbwvjCRA@mail.gmail.com>
	<CAFiqYun=ty6fEK0AiJG9jD5zLriy-MvOBEQS+dOC4WP=uZ5Epw@mail.gmail.com>
	<CALJtzxDvyj1-B1jC63yHK3KkW0fkC8ggUTPfcoaz1ad6=e-j5A@mail.gmail.com>
	<CAO3hyzH7eeE7qBc5t7KADz8nDenw67Kd4vrdY=ch0HUkESFnYw@mail.gmail.com>
	<CAO3hyzFtEv2ydxjhWO-fXTDfR80fFEknK4-foMzEdQ+0n2LErQ@mail.gmail.com>
	<002401d083c4$ad5e9530$081bbf90$@botecomm.com>
Message-ID: <105DEDA5-21FC-47A3-98D1-5E28541789B0@mgtech.com>

LUA would give you the most power/flexibility.
Mario G

> On Apr 30, 2015, at 9:09 PM, Bote Man <bote_radio at botecomm.com> wrote:
> 
> The default dialplan that ships with the vanilla configuration files for FreeSWITCH contains an example of routing based on time of day and day of week. Perhaps that could also be used to determine voice mail greetings to play, with some additional work.
>  
> Bote
>  
>  
> From: Chris Tunbridge
> Sent: Thursday, 30 April, 2015 13:23
> Subject: Re: [Freeswitch-users] voicemail + S3
>  
> Alejandro the way you would change the voicemail greeting is by using the following action
> 
> <action application="set" data="voicemail_greeting_number=1"/>
> 
> setting the number to a different one depending on the greeting # you want to play.
> 
> You can control the time based stuff by using conditions, see here for more information https://freeswitch.org/confluence/display/FREESWITCH/Time+of+Day+and+Holiday+Routing <https://freeswitch.org/confluence/display/FREESWITCH/Time+of+Day+and+Holiday+Routing>
>  
> On Thu, Apr 30, 2015 at 11:19 AM, Chris Tunbridge <blasterjr at gmail.com <mailto:blasterjr at gmail.com>> wrote:
> Alejandro i figured I'd chime in, i am the one who submitted that ticket regarding http_cache+voicemail, the issue is the system doesn't read the http_put status and thinks that the voicemail didn't get stored so it never makes a record.
> 
> S3FS works okay for voice mails, it'd be nice is the voice mail module was updated to function with the http_cache module as this would help out with scaling deployments.
>  
> On Thu, Apr 30, 2015 at 11:11 AM, Alejandro <cdgraff at gmail.com <mailto:cdgraff at gmail.com>> wrote:
> hi Steven, 
>  
> thanks a lot for the time to gide me, during mean time will use s3fs (i'm not big fan, but i has this into other services into production and is a good alternative, and can pass the issue of use http_cache module to upload.)
>  
> just one more question... i like to give the user the option to has custom Voicemail Grettings recorded by some profesional presenter, and rotate in base to the hour of the day automatically.
>  
> Something like, from 9am to 5pm, play gretting_1.mp3 and from 5pm to 9am grettings_2.mp3
>  
> Can teel me if that be possible and guide me to some documentation or link to investigate more about this?
>  
> THanks again, i'm working with freeswitch just a few hours, and is really amazing, I come from Asterisk and feel more flexible and powerfull freeswitch.
> 
> Regards from Argentina
> Alejandro 
>  
> 2015-04-30 13:35 GMT-03:00 Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>>:
>>  
>> between description and comments
>> http_cache.conf, s3 profile
>> voicemail.conf, see the storage-dir param towards the bottom
>>  
>> But as I say it's not yet functional - the comments show it was failing as mod_voicemail checks if the file exists which mod_http_cache seems not to support yet and I don't see any mention of FS-7280 in the commit log since the ticket was created.
>>  
>>  
>> On 30 April 2015 at 14:31, Alejandro <cdgraff at gmail.com <mailto:cdgraff at gmail.com>> wrote:
>> Thanks Steven, to point me to the attach, I previously read the ticket, but don't see the attached file.
>>  
>> Regards, 
>> Alejandro
>>  
>> 2015-04-30 10:23 GMT-03:00 Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>>:
>>>  
>>> There's a sample config attached to https://freeswitch.org/jira/browse/FS-7280 <https://freeswitch.org/jira/browse/FS-7280>, but from that ticket I'm not sure that it is actually functioning yet.
>>>  
>>> On 30 April 2015 at 13:24, Alejandro <cdgraff at gmail.com <mailto:cdgraff at gmail.com>> wrote:
>>>> Hi all, 
>>>>  
>>>> I read that if I combine, mod_voicemail with mod_http_cache I can save my voicemail message directly into S3 storage.
>>>>  
>>>> Any are using this implementation? can share with me some config file to understand how i need setup this?
>>>>  
>>>> Thanks in advance
>>>> Alejandro
>>>>  
>>>> _________________________________________________________________________
>>>> 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 <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 phenix at vfemail.net  Sat May  2 00:03:04 2015
From: phenix at vfemail.net (Tanguy)
Date: Fri, 01 May 2015 22:03:04 +0200
Subject: [Freeswitch-users] sip_to_user and destination number
In-Reply-To: <CACOYK=hhrQ-dJoF_vpYKkceh1M=iHcc_AhYj2Rb1+4_LCNCgFw@mail.gmail.com>
References: <5543A539.3000704@vfemail.net>
	<CACOYK=hhrQ-dJoF_vpYKkceh1M=iHcc_AhYj2Rb1+4_LCNCgFw@mail.gmail.com>
Message-ID: <5543DBF8.30804@vfemail.net>

Hello

With or without the extension parameter, it's exactly the same.

Thanks

On 01/05/2015 20:04, Stanislav Sinyagin wrote:
>
> Remove the extension parameter and see if it helps.
>
> On May 1, 2015 6:11 PM, "Tanguy" <phenix at vfemail.net 
> <mailto:phenix at vfemail.net>> wrote:
>
>     Hello,
>
>     My provider did not send correct DID number in the INVITE packet
>     but i can use "To" argument
>
>     INVITE
>     sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429 at 92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429
>     <mailto:sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429 at 92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429>
>     SIP/2.0.
>     Call-ID: 25016-VB-188fd96b-526e3dbd4 at sip.ovh.fr
>     <mailto:25016-VB-188fd96b-526e3dbd4 at sip.ovh.fr>.
>     Contact: <sip:10.7.1.60:5060 <http://10.7.1.60:5060>>.
>     Content-Type: application/sdp.
>     CSeq: 403749831 INVITE.
>     From: "0967212xxx" <sip:0967212xxx at sip.ovh.fr;user=phone>
>     <mailto:sip:0967212xxx at sip.ovh.fr;user=phone>;tag=25016-VE-188fd96c-18bb43586.
>     Max-Forwards: 27.
>     Record-Route: <sip:91.121.129.20:5060;lr>.
>     *To: <sip:0557590xxx at 10.7.1.60;user=phone>
>     <mailto:sip:0557590xxx at 10.7.1.60;user=phone>.*
>     Via: SIP/2.0/UDP
>     91.121.129.20:5060;branch=z9hG4bK-WGZO-1fe73949-2df58378.
>
>     Using asterisk i can bypass the issue using something like exten
>     => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) but
>     i am unable to do the same under freeswitch.
>
>     My trunk configuration seems correct, as you can see i used 
>     auto_to_user, but the destination number remains 0033972480xxx
>     when i call 0557590xxx.
>
>     2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635
>     Processing 0967212xxx <0967212xxx>->0033972480xxx in context public
>
>
>     <include>
>     <gateway name="0a96c3d3-0b0e-4864-b9ec-759fa4422429">
>     <param name="username" value="0033972480xxx"/>
>     <param name="password" value="xxxxxxx"/>
>     <param name="proxy" value="sip.ovh.fr <http://sip.ovh.fr>"/>
>     <param name="expire-seconds" value="800"/>
>     <param name="register" value="true"/>
>     <param name="retry-seconds" value="30"/>
>     <param name="extension" value="auto_to_user"/>
>     <param name="context" value="public"/>
>     </gateway>
>     </include>
>
>     I tried to edit my inbound dialplan manually, it works using
>     <condition field="${sip_to_user}" expression="0557590xxx" > but i
>     prefer a proper way to do this because i will also use telcos with
>     normal invite packets
>
>     I how i can copy $sip_to_header to destination for this specific
>     trunk ?
>
>     Please note that i use fusionpbx.
>
>     Best regards, sorry for my bad English
>
>

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From blasterjr at gmail.com  Sat May  2 00:11:16 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Fri, 1 May 2015 14:11:16 -0600
Subject: [Freeswitch-users] sip_to_user and destination number
In-Reply-To: <5543DBF8.30804@vfemail.net>
References: <5543A539.3000704@vfemail.net>
	<CACOYK=hhrQ-dJoF_vpYKkceh1M=iHcc_AhYj2Rb1+4_LCNCgFw@mail.gmail.com>
	<5543DBF8.30804@vfemail.net>
Message-ID: <CAO3hyzGW3BdSwXV4_2bnGQfhyzXLGFJojGxNV_+Haz6XyAvY7Q@mail.gmail.com>

Tanguy,

setting the auto_to_user should set the destination number field to the
correct one according to the wiki

"Note: *extension* parameter influence the contents of channel variable
*Caller-Destination-Number* and *destination_number*. If it is blank,
*Caller-Destination-Number* will always be set to gateway's username. If it
has a value, *Caller-Destination-Number* will always be set to this value.
If it has value *auto_to_user*, *Caller-Destination-Number* will be
populated with value *${sip_to_user}* which means the real dialed number in
case of an inbound call."

from: https://wiki.freeswitch.org/wiki/Sofia.conf.xml

where is this config that you listed currently located? (full file path
please)

On Fri, May 1, 2015 at 2:03 PM, Tanguy <phenix at vfemail.net> wrote:

>  Hello
>
> With or without the extension parameter, it's exactly the same.
>
> Thanks
>
>
> On 01/05/2015 20:04, Stanislav Sinyagin wrote:
>
> Remove the extension parameter and see if it helps.
> On May 1, 2015 6:11 PM, "Tanguy" <phenix at vfemail.net> wrote:
>
>>  Hello,
>>
>> My provider did not send correct DID number in the INVITE packet but i
>> can use "To" argument
>>
>> INVITE
>> sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429 at 92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429
>> SIP/2.0.
>> Call-ID: 25016-VB-188fd96b-526e3dbd4 at sip.ovh.fr.
>> Contact: <sip:10.7.1.60:5060>.
>> Content-Type: application/sdp.
>> CSeq: 403749831 INVITE.
>> From: "0967212xxx" <sip:0967212xxx at sip.ovh.fr;user=phone>
>> <sip:0967212xxx at sip.ovh.fr;user=phone>;tag=25016-VE-188fd96c-18bb43586.
>> Max-Forwards: 27.
>> Record-Route: <sip:91.121.129.20:5060;lr>.
>> *To: <sip:0557590xxx at 10.7.1.60;user=phone>
>> <sip:0557590xxx at 10.7.1.60;user=phone>.*
>> Via: SIP/2.0/UDP 91.121.129.20:5060
>> ;branch=z9hG4bK-WGZO-1fe73949-2df58378.
>>
>> Using asterisk i can bypass the issue using something like  exten =>
>> s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) but i am
>> unable to do the same under freeswitch.
>>
>> My trunk configuration seems correct, as you can see i used
>> auto_to_user, but the destination number remains 0033972480xxx when i
>> call 0557590xxx.
>>
>> 2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635 Processing
>> 0967212xxx <0967212xxx>->0033972480xxx in context public
>>
>>
>> <include>
>>     <gateway name="0a96c3d3-0b0e-4864-b9ec-759fa4422429">
>>       <param name="username" value="0033972480xxx"/>
>>       <param name="password" value="xxxxxxx"/>
>>       <param name="proxy" value="sip.ovh.fr"/>
>>       <param name="expire-seconds" value="800"/>
>>       <param name="register" value="true"/>
>>       <param name="retry-seconds" value="30"/>
>>       <param name="extension" value="auto_to_user"/>
>>       <param name="context" value="public"/>
>>     </gateway>
>> </include>
>>
>> I tried to edit my inbound dialplan manually, it works using  <condition
>> field="${sip_to_user}" expression="0557590xxx" > but i prefer a proper way
>> to do this because i will also use telcos with normal invite packets
>>
>> I how i can copy $sip_to_header to destination for this specific trunk ?
>>
>> Please note that i use fusionpbx.
>>
>> Best regards, sorry for my bad English
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From phenix at vfemail.net  Sat May  2 00:38:03 2015
From: phenix at vfemail.net (Tanguy)
Date: Fri, 01 May 2015 22:38:03 +0200
Subject: [Freeswitch-users] sip_to_user and destination number
In-Reply-To: <CAO3hyzGW3BdSwXV4_2bnGQfhyzXLGFJojGxNV_+Haz6XyAvY7Q@mail.gmail.com>
References: <5543A539.3000704@vfemail.net>
	<CACOYK=hhrQ-dJoF_vpYKkceh1M=iHcc_AhYj2Rb1+4_LCNCgFw@mail.gmail.com>
	<5543DBF8.30804@vfemail.net>
	<CAO3hyzGW3BdSwXV4_2bnGQfhyzXLGFJojGxNV_+Haz6XyAvY7Q@mail.gmail.com>
Message-ID: <5543E42B.5020000@vfemail.net>

Hello Chris

The file ( generated by fusionpbx ) is located in 
/usr/local/freeswitch/conf/sip_profiles/external/v_0a96c3d3-0b0e-4864-b9ec-759fa4422429.xml, 
i also tried to force a number in this field but it did not change anything.

Thanks



On 01/05/2015 22:11, Chris Tunbridge wrote:
> Tanguy,
>
> setting the auto_to_user should set the destination number field to 
> the correct one according to the wiki
>
> "Note: /extension/ parameter influence the contents of channel 
> variable /Caller-Destination-Number/ and /destination_number/. If it 
> is blank, /Caller-Destination-Number/ will always be set to gateway's 
> username. If it has a value, /Caller-Destination-Number/ will always 
> be set to this value. If it has value /auto_to_user/, 
> /Caller-Destination-Number/ will be populated with value 
> /${sip_to_user}/ which means the real dialed number in case of an 
> inbound call."
>
> from: https://wiki.freeswitch.org/wiki/Sofia.conf.xml
>
> where is this config that you listed currently located? (full file 
> path please)
>

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From s.safarov at gmail.com  Sat May  2 10:17:12 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 2 May 2015 09:17:12 +0300
Subject: [Freeswitch-users] sip_to_user and destination number
In-Reply-To: <5543A539.3000704@vfemail.net>
References: <5543A539.3000704@vfemail.net>
Message-ID: <CAHtxdDfigUd681ECVwZJDOa0J9ZyjHvc0b+t1r4xdvXzj9+iCA@mail.gmail.com>

Try
1) link gateway to "internal" profile;
2) create dialplan with name "provider_inbound_calls" and add required
extensions;
3) create user "provider_gw1" in directory with attribute cidr="10.7.1.60/32"
(value from you example), with random value in param "password",
and "provider_inbound_calls"  value in variable "user_context"
After it you can make inbound call from.
If provider has several gateways, add user record in directory for each
gateway.

Sergey


On Fri, May 1, 2015 at 7:09 PM, Tanguy <phenix at vfemail.net> wrote:

>  Hello,
>
> My provider did not send correct DID number in the INVITE packet but i can
> use "To" argument
>
> INVITE
> sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429 at 92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429
> SIP/2.0.
> Call-ID: 25016-VB-188fd96b-526e3dbd4 at sip.ovh.fr.
> Contact: <sip:10.7.1.60:5060>.
> Content-Type: application/sdp.
> CSeq: 403749831 INVITE.
> From: "0967212xxx" <sip:0967212xxx at sip.ovh.fr;user=phone>
> <sip:0967212xxx at sip.ovh.fr;user=phone>;tag=25016-VE-188fd96c-18bb43586.
> Max-Forwards: 27.
> Record-Route: <sip:91.121.129.20:5060;lr>.
> *To: <sip:0557590xxx at 10.7.1.60;user=phone>
> <sip:0557590xxx at 10.7.1.60;user=phone>.*
> Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-WGZO-1fe73949-2df58378.
>
> Using asterisk i can bypass the issue using something like  exten =>
> s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) but i am
> unable to do the same under freeswitch.
>
> My trunk configuration seems correct, as you can see i used  auto_to_user,
> but the destination number remains 0033972480xxx when i call 0557590xxx.
>
> 2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635 Processing
> 0967212xxx <0967212xxx>->0033972480xxx in context public
>
>
> <include>
>     <gateway name="0a96c3d3-0b0e-4864-b9ec-759fa4422429">
>       <param name="username" value="0033972480xxx"/>
>       <param name="password" value="xxxxxxx"/>
>       <param name="proxy" value="sip.ovh.fr"/>
>       <param name="expire-seconds" value="800"/>
>       <param name="register" value="true"/>
>       <param name="retry-seconds" value="30"/>
>       <param name="extension" value="auto_to_user"/>
>       <param name="context" value="public"/>
>     </gateway>
> </include>
>
> I tried to edit my inbound dialplan manually, it works using  <condition
> field="${sip_to_user}" expression="0557590xxx" > but i prefer a proper way
> to do this because i will also use telcos with normal invite packets
>
> I how i can copy $sip_to_header to destination for this specific trunk ?
>
> Please note that i use fusionpbx.
>
> Best regards, sorry for my bad English
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gavin.henry at gmail.com  Sat May  2 15:52:34 2015
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sat, 2 May 2015 12:52:34 +0100
Subject: [Freeswitch-users] FS, SS7 and Dialogic IMG2020 in the UK
Message-ID: <CAA8_NKB_L+vc-KkB=BthxMT8eKwH4ym9dksr3Pk3rwx4hYCvUQ@mail.gmail.com>

Hi all,

We're looking to get a pair of Dialogic IMG2020 for some BT Ofcom
regulated SS7 interconnects:

http://www.dialogic.com/en/products/gateways/img/img2020.aspx

Any issues known with FS behind these?

Thanks,

Gavin.

-- 
http://www.surevoip.co.uk


From phenix at vfemail.net  Sat May  2 20:52:03 2015
From: phenix at vfemail.net (Tanguy)
Date: Sat, 02 May 2015 18:52:03 +0200
Subject: [Freeswitch-users] sip_to_user and destination number
In-Reply-To: <CAHtxdDfigUd681ECVwZJDOa0J9ZyjHvc0b+t1r4xdvXzj9+iCA@mail.gmail.com>
References: <5543A539.3000704@vfemail.net>
	<CAHtxdDfigUd681ECVwZJDOa0J9ZyjHvc0b+t1r4xdvXzj9+iCA@mail.gmail.com>
Message-ID: <554500B3.2080905@vfemail.net>

Hello

I'm not sure to understand why i should use a so strange trick. Using 
"internal" profile for external trunk can't be a security leak ? If my 
provider change something about the 10.7.6.60 gateway, the trunk will no 
longer work until i update my configuration ?

Nevertheless i tried to apply your advice, so i moved by gateway in 
internal profile and i created a new user

/usr/local/freeswitch/conf/dialplan/provider_inbound_calls/user.xml

<include>
<user id="provider_gw1"  cidr="10.7.1.60/32" >
<params>
<param name="password" value="xxxxxx"/>
</params>
</user>
<variables>
<variable name="user_context" value="provider_inbound_calls"/>
</variables>

</include>

Unfortunately it did not work, but i have the good DID number in 
[0557590xxx at 10.7.1.60]

2015-05-02 18:30:15.940354 [DEBUG] sofia.c:9015 IP 91.121.129.20 
Rejected by acl "domains". Falling back to Digest auth.
2015-05-02 18:30:15.940354 [WARNING] sofia_reg.c:2827 Can't find user 
[anonymous at 10.7.1.60] from 91.121.129.20
You must define a domain called '10.7.1.60' in your directory and add a 
user with the id="anonymous" attribute
and you must configure your device to use the proper domain in it's 
authentication credentials.
2015-05-02 18:30:15.940354 [WARNING] sofia_reg.c:1687 SIP auth failure 
(INVITE) on sofia profile 'internal' for *[0557590xxx at 10.7.1.60] *from 
ip 91.121.129.20
2015-05-02 18:30:15.940354 [DEBUG] switch_core_session.c:1061 Send 
signal sofia/internal/0967212xxx at sip.ovh.fr [BREAK]
2015-05-02 18:30:15.940354 [NOTICE] sofia.c:2063 Hangup 
sofia/internal/0967212xxx at sip.ovh.fr [CS_NEW] [CALL_REJECTED]



Just for fun i created a user anonymous in domain 10.7.1.60 and reloaded 
with reloadxml but i still have the same message.

/usr/local/freeswitch/conf/dialplan/10.7.1.60/user.xml

<include>
<user id="anonymous"  cidr="10.7.1.60/32" >
<params>
<param name="password" value="xxxxx"/>
</params>
</user>
<variables>
<variable name="user_context" value="10.7.1.60"/>
</variables>
</include>




Thanks


On 02/05/2015 08:17, Sergey Safarov wrote:
> Try
> 1) link gateway to "internal" profile;
> 2) create dialplan with name "provider_inbound_calls" and add required 
> extensions;
> 3) create user "provider_gw1" in directory with 
> attribute cidr="10.7.1.60/32 <http://10.7.1.60/32>" (value from you 
> example), with random value in param "password", 
> and "provider_inbound_calls"  value in variable "user_context"
> After it you can make inbound call from.
> If provider has several gateways, add user record in directory for 
> each gateway.
>
> Sergey
>
>

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From rossbcan at gmail.com  Sat May  2 22:08:12 2015
From: rossbcan at gmail.com (Bill Ross)
Date: Sat, 2 May 2015 14:08:12 -0400
Subject: [Freeswitch-users] MondoTalk trunk configuration
Message-ID: <09ac01d08502$f44bc6e0$dce354a0$@gmail.com>

Hi;

 

I have two freeswitch installations each with a free mondotalk number as
gateway. Attempting to get them to communicate extension to extension via
these gateways.

 

The originating extension sends a proper invite (destination # is 8757469):

INVITE sip:8757469 at 192.168.1.1 SIP/2.0

 

Freeswitch, in forwarding invite through mondotalk gateway sends this:

INVITE sip:@sip99.mondotalk.com SIP/2.0

 

Which lacks the destination # and is rejected. Clearly, I have messed up in
my mondotalk dialplan.

 

Any hints as to what I am missing?

 

Thanks;

Bill

 

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From s.safarov at gmail.com  Sat May  2 22:55:35 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 2 May 2015 21:55:35 +0300
Subject: [Freeswitch-users] sip_to_user and destination number
In-Reply-To: <554500B3.2080905@vfemail.net>
References: <5543A539.3000704@vfemail.net>
	<CAHtxdDfigUd681ECVwZJDOa0J9ZyjHvc0b+t1r4xdvXzj9+iCA@mail.gmail.com>
	<554500B3.2080905@vfemail.net>
Message-ID: <CAHtxdDcnQ2z33HqL0KtuzETpj5aCpMotzD_vZyvHQyU9p_=kEg@mail.gmail.com>

In log you can see invite arrived from IP address 91.121.129.20
Change IP 10.7.1.60/32 to 91.121.129.20/32

I'm not sure to understand why i should use a so strange trick. Using
"internal" profile for external trunk can't be a security leak ?
It is not problem. In this case profile used for authenticate call by user
"provider_gw1". Real call processing executed in dialplan
"provider_inbound_calls". In this dialplan you can allow one DID number and
block all other.

If my provider change something about the 10.7.6.60 gateway, the trunk will
no longer work until i update my configuration ?
If provider can configure username and password for DID call authentication
you can remove "cird" attribute.
If username and password can't be configured, then say me how much DID
number provider assigned for you.

Sergey



On Sat, May 2, 2015 at 7:52 PM, Tanguy <phenix at vfemail.net> wrote:

>  Hello
>
> I'm not sure to understand why i should use a so strange trick. Using
> "internal" profile for external trunk can't be a security leak ? If my
> provider change something about the 10.7.6.60 gateway, the trunk will no
> longer work until i update my configuration ?
>
> Nevertheless i tried to apply your advice, so i moved by gateway in
> internal profile and i created a new user
>
> /usr/local/freeswitch/conf/dialplan/provider_inbound_calls/user.xml
>
> <include>
>   <user id="provider_gw1"  cidr="10.7.1.60/32" >
>     <params>
>       <param name="password" value="xxxxxx"/>
>     </params>
>   </user>
>   <variables>
>      <variable name="user_context" value="provider_inbound_calls"/>
>   </variables>
>
> </include>
>
> Unfortunately it did not work, but i have the good DID number in [
> 0557590xxx at 10.7.1.60]
>
> 2015-05-02 18:30:15.940354 [DEBUG] sofia.c:9015 IP 91.121.129.20 Rejected
> by acl "domains". Falling back to Digest auth.
> 2015-05-02 18:30:15.940354 [WARNING] sofia_reg.c:2827 Can't find user [
> anonymous at 10.7.1.60] from 91.121.129.20
> You must define a domain called '10.7.1.60' in your directory and add a
> user with the id="anonymous" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
> 2015-05-02 18:30:15.940354 [WARNING] sofia_reg.c:1687 SIP auth failure
> (INVITE) on sofia profile 'internal' for *[0557590xxx at 10.7.1.60
> <0557590xxx at 10.7.1.60>] *from ip 91.121.129.20
> 2015-05-02 18:30:15.940354 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/0967212xxx at sip.ovh.fr [BREAK]
> 2015-05-02 18:30:15.940354 [NOTICE] sofia.c:2063 Hangup
> sofia/internal/0967212xxx at sip.ovh.fr [CS_NEW] [CALL_REJECTED]
>
>
>
> Just for fun i created a user anonymous in domain 10.7.1.60 and reloaded
> with reloadxml but i still have the same message.
>
> /usr/local/freeswitch/conf/dialplan/10.7.1.60/user.xml
>
> <include>
>   <user id="anonymous"  cidr="10.7.1.60/32" >
>     <params>
>       <param name="password" value="xxxxx"/>
>     </params>
>   </user>
>   <variables>
>      <variable name="user_context" value="10.7.1.60"/>
>   </variables>
> </include>
>
>
>
>
> Thanks
>
>
> On 02/05/2015 08:17, Sergey Safarov wrote:
>
> Try
> 1) link gateway to "internal" profile;
> 2) create dialplan with name "provider_inbound_calls" and add required
> extensions;
> 3) create user "provider_gw1" in directory with attribute cidr="
> 10.7.1.60/32" (value from you example), with random value in param
> "password", and "provider_inbound_calls"  value in variable "user_context"
> After it you can make inbound call from.
> If provider has several gateways, add user record in directory for each
> gateway.
>
>  Sergey
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May  2 23:19:24 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Sat, 2 May 2015 12:19:24 -0700
Subject: [Freeswitch-users] core dumped. PANIC: unprotected error in call to
 Lua API (multiple Lua VMs detected)
Message-ID: <CAGVOjVUO3+Dz66u1+q+hgMYr0g6r6gpL3CTNQA9t=yhd0+Ke3w@mail.gmail.com>

Hi, users.

I have installed the freeswitch(FreeSWITCH Version 1.4.18~64bit ( 64bit) on
fresh installation of  ubuntu(14.04). When i try to load mod_vlc i see this
error and core dumped.


PANIC: unprotected error in call to Lua API (multiple Lua VMs detected)

Your help would be much appreciated.
Thanks
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From kris at kriskinc.com  Sat May  2 23:38:50 2015
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Sat, 2 May 2015 15:38:50 -0400
Subject: [Freeswitch-users] FS, SS7 and Dialogic IMG2020 in the UK
In-Reply-To: <CAA8_NKB_L+vc-KkB=BthxMT8eKwH4ym9dksr3Pk3rwx4hYCvUQ@mail.gmail.com>
References: <CAA8_NKB_L+vc-KkB=BthxMT8eKwH4ym9dksr3Pk3rwx4hYCvUQ@mail.gmail.com>
Message-ID: <CAKDfjgd=b=HHNUZkvwG1+DC3NB9DNu2Hi9=CoOLGx2Z-bp9wMg@mail.gmail.com>

Hi Gavin,

  The only potential things that jump out to me:

- Support for/requiring PRACK. This is often suggested/strongly
recommended/required with SIP/SS7 interop. FS has a long standing bug with
100rel.

- Same for SIP-I/SIP-T. FreeSWITCH can pass these through with multipart
but it can't currently parse or use them in any effective way. With that
said I think I've seen people use LUA and other crazy stuff to parse
specific data they absolutely needed...

  Other than that if these can be configured for fairly
plain/straightforward SIP there shouldn't be any significant issues that I
can see.

On Saturday, May 2, 2015, Gavin Henry <gavin.henry at gmail.com> wrote:

> Hi all,
>
> We're looking to get a pair of Dialogic IMG2020 for some BT Ofcom
> regulated SS7 interconnects:
>
> http://www.dialogic.com/en/products/gateways/img/img2020.aspx
>
> Any issues known with FS behind these?
>
> Thanks,
>
> Gavin.
>
> --
> http://www.surevoip.co.uk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
Sent from mobile device
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From abalashov at evaristesys.com  Sat May  2 23:41:32 2015
From: abalashov at evaristesys.com (Alex Balashov)
Date: Sat, 02 May 2015 15:41:32 -0400
Subject: [Freeswitch-users] FS, SS7 and Dialogic IMG2020 in the UK
In-Reply-To: <CAKDfjgd=b=HHNUZkvwG1+DC3NB9DNu2Hi9=CoOLGx2Z-bp9wMg@mail.gmail.com>
References: <CAA8_NKB_L+vc-KkB=BthxMT8eKwH4ym9dksr3Pk3rwx4hYCvUQ@mail.gmail.com>
	<CAKDfjgd=b=HHNUZkvwG1+DC3NB9DNu2Hi9=CoOLGx2Z-bp9wMg@mail.gmail.com>
Message-ID: <20150502194132.5419088.23543.54831@evaristesys.com>

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From bote_radio at botecomm.com  Sat May  2 23:57:35 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Sat, 2 May 2015 15:57:35 -0400
Subject: [Freeswitch-users] MondoTalk trunk configuration
In-Reply-To: <09ac01d08502$f44bc6e0$dce354a0$@gmail.com>
References: <09ac01d08502$f44bc6e0$dce354a0$@gmail.com>
Message-ID: <006901d08512$3c8be110$b5a3a330$@botecomm.com>

It seems that each carrier has its own weirdness to overcome.

 

To start with you should turn on sip tracing to see what they are sending
you and if some other field contains the destination number. In the
FreeSWITCH console type:

Sofia global siptrace on

 

And enjoy the data that spews forth as you place each test call J

 

That's how I figured out how CallCentric was sending calls to my FS
installation so I had to modify my inbound dialplan to look for a different
field than the vanilla configuration looks for, but that's to be expected.

 

It's also possible, or even likely, that Mondotalk is expecting you to send
the destination number in a different field, but I have no good suggestions
on that. 

 

Bote

 

 

From: Bill Ross
Sent: Saturday, 02 May, 2015 14:08
Subject: [Freeswitch-users] MondoTalk trunk configuration

 

Hi;

 

I have two freeswitch installations each with a free mondotalk number as
gateway. Attempting to get them to communicate extension to extension via
these gateways.

 

The originating extension sends a proper invite (destination # is 8757469):

INVITE sip:8757469 at 192.168.1.1 SIP/2.0

 

Freeswitch, in forwarding invite through mondotalk gateway sends this:

INVITE sip:@sip99.mondotalk.com SIP/2.0

 

Which lacks the destination # and is rejected. Clearly, I have messed up in
my mondotalk dialplan.

 

Any hints as to what I am missing?

 

Thanks;

Bill

 

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From kris at kriskinc.com  Sun May  3 02:00:55 2015
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Sat, 2 May 2015 18:00:55 -0400
Subject: [Freeswitch-users] FS, SS7 and Dialogic IMG2020 in the UK
In-Reply-To: <20150502194132.5419088.23543.54831@evaristesys.com>
References: <CAA8_NKB_L+vc-KkB=BthxMT8eKwH4ym9dksr3Pk3rwx4hYCvUQ@mail.gmail.com>
	<CAKDfjgd=b=HHNUZkvwG1+DC3NB9DNu2Hi9=CoOLGx2Z-bp9wMg@mail.gmail.com>
	<20150502194132.5419088.23543.54831@evaristesys.com>
Message-ID: <CAKDfjgcCd2fjnBfzTSCHBG2DkKQN25oC=Zsv9LKQx5dSBuyJJg@mail.gmail.com>

Hi Alex,

  I'm on my phone but Mike/Brian can probably elaborate.

  From what I recall it was somewhat sporadic but reproducible.

On Saturday, May 2, 2015, Alex Balashov <abalashov at evaristesys.com> wrote:

> Kristian,
>
> What is the nature of the 100rel bug?
>
>  --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> Sent from my BlackBerry.
>   *From: *Kristian Kielhofner
> *Sent: *Saturday, May 2, 2015 15:40
> *To: *FreeSWITCH Users Help
> *Reply To: *FreeSWITCH Users Help
> *Subject: *Re: [Freeswitch-users] FS, SS7 and Dialogic IMG2020 in the UK
>
> Hi Gavin,
>
>   The only potential things that jump out to me:
>
> - Support for/requiring PRACK. This is often suggested/strongly
> recommended/required with SIP/SS7 interop. FS has a long standing bug with
> 100rel.
>
> - Same for SIP-I/SIP-T. FreeSWITCH can pass these through with multipart
> but it can't currently parse or use them in any effective way. With that
> said I think I've seen people use LUA and other crazy stuff to parse
> specific data they absolutely needed...
>
>   Other than that if these can be configured for fairly
> plain/straightforward SIP there shouldn't be any significant issues that I
> can see.
>
> On Saturday, May 2, 2015, Gavin Henry <gavin.henry at gmail.com
> <javascript:_e(%7B%7D,'cvml','gavin.henry at gmail.com');>> wrote:
>
>> Hi all,
>>
>> We're looking to get a pair of Dialogic IMG2020 for some BT Ofcom
>> regulated SS7 interconnects:
>>
>> http://www.dialogic.com/en/products/gateways/img/img2020.aspx
>>
>> Any issues known with FS behind these?
>>
>> Thanks,
>>
>> Gavin.
>>
>> --
>> http://www.surevoip.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
>>
>
>
> --
> Sent from mobile device
>
>

-- 
Sent from mobile device
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From cdgraff at gmail.com  Sun May  3 05:25:21 2015
From: cdgraff at gmail.com (Alejandro)
Date: Sat, 2 May 2015 22:25:21 -0300
Subject: [Freeswitch-users] Fail at save voicemail message
Message-ID: <CALJtzxB4ezUNadt=dU3e6-2euTiGsad6a6kbP9xMXb1cWvMcSA@mail.gmail.com>

Hi all,

I has an issue with my voicemail setup. I'm running this:

FXO -> FreeSwitch -> Voicemail

When I record the message all is ok, just that this fail to save.

I see this error:
2015-05-02 21:18:05.539885 [DEBUG] switch_core_session.c:2893
sofia/internal/6303XXXX at fs-01.domain.net skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)

My Dialplan be:

<!--
https://wiki.freeswitch.org/wiki/Voicemail#Another_Dialplan_to_send_unanswered_calls_to_voicemail
-->
<extension name="public_did">
  <condition field="destination_number" expression="^(4862XXXX)$">
    <action application="answer"/>
      <action application="playback" data="/usr/storage/voicemail/default/
domain.com/1000/msg_06c6288e-ee11-11e4-b862-1dea27edc68b.wav"/>
      <!-- Next line removes default greeting, we already played one above
-->
      <action application="set" data="skip_greeting=true"/>
      <action application="set" data="skip_instructions=true"/>
      <action application="voicemail" data="default ${domain_name}
${dialed_extension}"/>
  </condition>
</extension>

Some idea, what I'm doing wrong?

Thanks in advance.
Alejandro
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From bote_radio at botecomm.com  Sun May  3 07:00:57 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Sat, 2 May 2015 23:00:57 -0400
Subject: [Freeswitch-users] Fail at save voicemail message
In-Reply-To: <CALJtzxB4ezUNadt=dU3e6-2euTiGsad6a6kbP9xMXb1cWvMcSA@mail.gmail.com>
References: <CALJtzxB4ezUNadt=dU3e6-2euTiGsad6a6kbP9xMXb1cWvMcSA@mail.gmail.com>
Message-ID: <00b101d0854d$615068a0$23f139e0$@botecomm.com>

This is only a guess, but you might check that the FXO is providing proper supervision signal to FreeSWITCH. Analog telephone lines can cause problems in FXO setups because FreeSWITCH does not know when the remote caller has hung up, or in your case FS thinks the caller has already hung up so there is no voicemail to record.

 

These settings would be in your FXO gateway, not in FreeSWITCH typically.


Hope this helps.

 

Bote

 

 

From: Alejandro
Sent: Saturday, 02 May, 2015 21:25
Subject: [Freeswitch-users] Fail at save voicemail message

 

Hi all, 

 

I has an issue with my voicemail setup. I'm running this:

 

FXO -> FreeSwitch -> Voicemail

 

When I record the message all is ok, just that this fail to save.

 

I see this error:

2015-05-02 21:18:05.539885 [DEBUG] switch_core_session.c:2893 sofia/internal/6303XXXX at fs-01.domain.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)

 

My Dialplan be:

 

<!-- https://wiki.freeswitch.org/wiki/Voicemail#Another_Dialplan_to_send_unanswered_calls_to_voicemail -->

<extension name="public_did">

  <condition field="destination_number" expression="^(4862XXXX)$">

    <action application="answer"/>

      <action application="playback" data="/usr/storage/voicemail/default/domain.com/1000/msg_06c6288e-ee11-11e4-b862-1dea27edc68b.wav"/>

      <!-- Next line removes default greeting, we already played one above -->

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

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

      <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>

  </condition>

</extension>

 

Some idea, what I'm doing wrong?

 

Thanks in advance.

Alejandro

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From phenix at vfemail.net  Sun May  3 11:35:35 2015
From: phenix at vfemail.net (Tanguy)
Date: Sun, 03 May 2015 09:35:35 +0200
Subject: [Freeswitch-users] sip_to_user and destination number
In-Reply-To: <CAHtxdDcnQ2z33HqL0KtuzETpj5aCpMotzD_vZyvHQyU9p_=kEg@mail.gmail.com>
References: <5543A539.3000704@vfemail.net>
	<CAHtxdDfigUd681ECVwZJDOa0J9ZyjHvc0b+t1r4xdvXzj9+iCA@mail.gmail.com>
	<554500B3.2080905@vfemail.net>
	<CAHtxdDcnQ2z33HqL0KtuzETpj5aCpMotzD_vZyvHQyU9p_=kEg@mail.gmail.com>
Message-ID: <5545CFC7.7070505@vfemail.net>

Hello Sergey

I tried to change the cidr attribute with my gateway public IP but this 
change nothing. My provider did not provide a different SIP accound for 
each DID number. I actually have 2 DID numbers on this test platform but 
if i migrate my production servers from asterisk to freeswitch i will 
manage about 50 DID numbers for this provider and probably more in the 
future.

I also tried to fetch a minimal configuration from freeswitch source (  
$somewhere/freeswitch.git/conf/minimal/ ) to replace the default 
configuration provided by the fusionpbx installer, then i just copied my 
original trunk configuration in 
/usr/local/freeswitch/conf/sip_profiles/external

I did not set a dialplan for testing everything but it's seems to work 
better: when i call 0557590xxx i got the following message

2015-05-03 09:14:54.587286 [INFO] mod_dialplan_xml.c:635 Processing 
0967212xxx <0967212xxx>->*0557590xxx* in context public

In comparison with

2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635 Processing 
0967212xxx <0967212xxx>->*0033972480xxx* in context public

There is probably something bad in the default configuration provided by 
fusionpbx witch prevent auto_to_user to work properly. I will compare 
configuration files from minimal freeswitch sample configuration and 
from freeswitch and try to determine where is the problem. This may 
deserve a but report :-)

Thanks


On 02/05/2015 20:55, Sergey Safarov wrote:
> In log you can see invite arrived from IP address 91.121.129.20
> Change IP 10.7.1.60/32 <http://10.7.1.60/32> to 91.121.129.20/32 
> <http://91.121.129.20/32>
>
> I'm not sure to understand why i should use a so strange trick. Using 
> "internal" profile for external trunk can't be a security leak ?
> It is not problem. In this case profile used for authenticate call by 
> user "provider_gw1". Real call processing executed in dialplan 
> "provider_inbound_calls". In this dialplan you can allow one DID 
> number and block all other.
>
> If my provider change something about the 10.7.6.60 gateway, the trunk 
> will no longer work until i update my configuration ?
> If provider can configure username and password for DID call 
> authentication you can remove "cird" attribute.
> If username and password can't be configured, then say me how much DID 
> number provider assigned for you.
>
> Sergey
>

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From phenix at vfemail.net  Sun May  3 20:25:02 2015
From: phenix at vfemail.net (Tanguy)
Date: Sun, 03 May 2015 18:25:02 +0200
Subject: [Freeswitch-users] sip_to_user and destination number
In-Reply-To: <5545CFC7.7070505@vfemail.net>
References: <5543A539.3000704@vfemail.net>
	<CAHtxdDfigUd681ECVwZJDOa0J9ZyjHvc0b+t1r4xdvXzj9+iCA@mail.gmail.com>
	<554500B3.2080905@vfemail.net>
	<CAHtxdDcnQ2z33HqL0KtuzETpj5aCpMotzD_vZyvHQyU9p_=kEg@mail.gmail.com>
	<5545CFC7.7070505@vfemail.net>
Message-ID: <55464BDE.40506@vfemail.net>

Hello

It solved my problem. I restored my yesterday backup and i notice that 
my inbound calls are now working. Very strange

I tried to remove the "extension" parameter, so inbound calls should not 
working but they are still working, and i noticed this message in the 
CLI when i updated my configuration using the fusionpbx GUI

2015-05-03 18:00:27.048335 [WARNING] sofia.c:3310 Ignoring duplicate 
gateway '0a96c3d3-0b0e-4864-b9ec-759fa4422429'

The gateway is not reloaded, all modifications where ignored but if i 
reload my external profile manually with *sofia profile external restart 
* i don't have this kind of message and my gateway is properly reloaded.

It seems to be a bug in fusionpbx, i will make a bug report

Regards





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From s.safarov at gmail.com  Sun May  3 20:54:12 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 3 May 2015 19:54:12 +0300
Subject: [Freeswitch-users] sip_to_user and destination number
In-Reply-To: <55464BDE.40506@vfemail.net>
References: <5543A539.3000704@vfemail.net>
	<CAHtxdDfigUd681ECVwZJDOa0J9ZyjHvc0b+t1r4xdvXzj9+iCA@mail.gmail.com>
	<554500B3.2080905@vfemail.net>
	<CAHtxdDcnQ2z33HqL0KtuzETpj5aCpMotzD_vZyvHQyU9p_=kEg@mail.gmail.com>
	<5545CFC7.7070505@vfemail.net> <55464BDE.40506@vfemail.net>
Message-ID: <CAHtxdDeZsA8LbKWTwbaYr4dGSj4MsQ_pNaV=LA-YLxaEMaO-CA@mail.gmail.com>

Hi Tanguy
I has tested case where:
1) gateway linked to public context;
2) gateway has parameter "<param name="context" value="did_processing"/>";
3) FS configuration has following dialplan.

<context name="did_processing">
      <extension name="info" continue="true">
         <condition>
            <action application="info" />
         </condition>
      </extension>

      <extension name="did_processing">
         <condition field="${sip_to_user}"
expression="^11_my_did_number_22$" break="on-true">
            <action application="transfer" data="sales XML defaul" />
         </condition>
         <condition field="${sip_to_user}"
expression="^33_my_did_number_44$" break="on-true">
            <action application="transfer" data="support XML defaul" />
         </condition>
         <condition>
            <action application="hangup" data="INCOMING_CALL_BARRED"/>
         </condition>
      </extension>
</context>

It it working correctly.




On Sun, May 3, 2015 at 7:25 PM, Tanguy <phenix at vfemail.net> wrote:

>  Hello
>
> It solved my problem. I restored my yesterday backup and i notice that my
> inbound calls are now working. Very strange
>
> I tried to remove the "extension" parameter, so inbound calls should not
> working but they are still working, and i noticed this message in the CLI
> when i updated my configuration using the fusionpbx GUI
>
> 2015-05-03 18:00:27.048335 [WARNING] sofia.c:3310 Ignoring duplicate
> gateway '0a96c3d3-0b0e-4864-b9ec-759fa4422429'
>
> The gateway is not reloaded, all modifications where ignored but if i
> reload my external profile manually with *sofia profile external restart *
> i don't have this kind of message and my gateway is properly reloaded.
>
> It seems to be a bug in fusionpbx, i will make a bug report
>
> 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 phenix at vfemail.net  Sun May  3 21:39:06 2015
From: phenix at vfemail.net (Tanguy)
Date: Sun, 03 May 2015 19:39:06 +0200
Subject: [Freeswitch-users] sip_to_user and destination number
In-Reply-To: <CAHtxdDeZsA8LbKWTwbaYr4dGSj4MsQ_pNaV=LA-YLxaEMaO-CA@mail.gmail.com>
References: <5543A539.3000704@vfemail.net>
	<CAHtxdDfigUd681ECVwZJDOa0J9ZyjHvc0b+t1r4xdvXzj9+iCA@mail.gmail.com>
	<554500B3.2080905@vfemail.net>
	<CAHtxdDcnQ2z33HqL0KtuzETpj5aCpMotzD_vZyvHQyU9p_=kEg@mail.gmail.com>
	<5545CFC7.7070505@vfemail.net> <55464BDE.40506@vfemail.net>
	<CAHtxdDeZsA8LbKWTwbaYr4dGSj4MsQ_pNaV=LA-YLxaEMaO-CA@mail.gmail.com>
Message-ID: <55465D3A.6070802@vfemail.net>

Hi Sergey

I also tried like you ( note the initial post ) using directly a 
condition field with with ${sip_to_user} variable instead of destination 
and it worked even with the bad sip profile but it's better to avoid 
specific dialplan when using a GUI, and i was not sure of the To: field 
with my provider who send correct INVITE.

Regards

On 03/05/2015 18:54, Sergey Safarov wrote:
> Hi Tanguy
> I has tested case where:
> 1) gateway linked to public context;
> 2) gateway has parameter "<param name="context" value="did_processing"/>";
> 3) FS configuration has following dialplan.
>
> <context name="did_processing">
> <extension name="info" continue="true">
> <condition>
> <action application="info" />
> </condition>
> </extension>
>
> <extension name="did_processing">
> <condition field="${sip_to_user}" expression="^11_my_did_number_22$" 
> break="on-true">
> <action application="transfer" data="sales XML defaul" />
> </condition>
> <condition field="${sip_to_user}" expression="^33_my_did_number_44$" 
> break="on-true">
> <action application="transfer" data="support XML defaul" />
> </condition>
> <condition>
> <action application="hangup" data="INCOMING_CALL_BARRED"/>
> </condition>
> </extension>
> </context>
>
> It it working correctly.
>
>



From covici at ccs.covici.com  Mon May  4 08:14:02 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 04 May 2015 00:14:02 -0400
Subject: [Freeswitch-users] problems using systemd freeswitch unit
Message-ID: <6927.1430712842@ccs.covici.com>

Hi.  In the freeswitch tree
freeswitch-new/debian/freeswitch-systemd.freeswitch.service was there
and I tried to get it to work in my systemd.  The problem seems to be
that systemd is getting very confused when fs backgrounds the process.
I got things to work by adding -nf and changing the type from forking to
simple, so fs never backgrounds.

My question is, has anyone gotten the original systemd unit to work and
if so, what did you do?  The only things I changed was the directories
and I added some ExecStartPre commands that I needed.

Thanks in advance for any ideas.

-- 
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 kheimerl at cs.berkeley.edu  Mon May  4 08:48:55 2015
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Sun, 3 May 2015 21:48:55 -0700
Subject: [Freeswitch-users] Simulate incoming SIP Message
Message-ID: <CACyT-3=AVrm3NkFVbCLVVbZPB_JE9jt87NA0gCxeAbdgFpL5Uw@mail.gmail.com>

Hello FreeSWITCH Users,

I am trying to simulate an incoming SIP message in order to test my
chatplan programmatically. Basically, I want FS to send itself a SIP
message, which will then be routed through the chatplan. I've been able to
generate the message using ESL and events, however FS refuses to send it
noting that it is "Not sending message to ourselves!"

That's exactly what I want it to do. I've used this functionality before
for calls, as generating a call that connects back to FS (and hooking it to
echo) is a great way to test the dialplan. Is there no way to get FS to
send itself a message?

Thanks!
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From yadenis at seznam.cz  Mon May  4 11:16:23 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Mon, 4 May 2015 09:16:23 +0200
Subject: [Freeswitch-users] Help with FSV record.
In-Reply-To: <8D8CF81D-30E4-4AF3-BDD1-358F9492342D@jerris.com>
References: <1430394389391-7596159.post@n2.nabble.com>
	<8D8CF81D-30E4-4AF3-BDD1-358F9492342D@jerris.com>
Message-ID: <02732363.20150504091623@seznam.cz>

Hello,

Thanks for the answer. When can we expect the release?

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

?tvrtek 30. dubna 2015, 14:34:47, napsal jste:


Recording video was barely a feature in 1.4.  Full support for this and everything you are trying to do is coming in 1.6.  Stay tuned for beta announcements on that coming soon.  You can try out the feature branch if you like, more info is available at: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video .  We are still working on docs so please be patient as we gather all of that information.


On Apr 30, 2015, at 7:46 AM, dex_Cz <yadenis at seznam.cz> wrote:

Hi guys!
I have a question. Somebody tried to record in the FSV file when i bridge
call?
I hame a simple context from example. But it is not working.
<include>
<context name="centos">
<extension name="from-centos">
<condition field="destination_number" expression="^(onops)$">
<action application="record_fsv"
data="/usr/local/freeswitch/recordings/testrecord.fsv"/>
<action application="sleep" data="100"/>
<action application="bridge" data="user/1004 at 172.16.0.2"/>
<action application="hangup"/>
</condition>
</extension>
</context>
</include>

I tried to change the context
<include>
<context name="centos">
 <extension name="from-centos">
  <condition field="destination_number" expression="^(onops)$">
   <action application="export" data="execute_on_answer=uuid_record ${uuid}
start /usr/local/freeswitch/recordings/testrecord.fsv"/> 
   <action application="bridge"
data="{ignore_early_media=true}user/1004 at 172.16.0.2"/>
   <action application="hangup"/>
  </condition>
 </extension>
</context>
</include>

It also did not help. In the end I try to enter these commands from the
console
uuid_setvar <uuid> enable_file_write_buffering false (its says ok)
and
uuid_record <uuid> start /usr/local/freeswitch/recordings/testrecord.fsv

But i have a error 

2015-04-30 12:27:16.635463 [ERR] mod_fsv.c:702 You are asking to write 342
bytes of data which is not supported. Please set
enable_file_write_buffering=false to use .fsv format
5e9ce2b0-ef23-11e4-b181-d1f037dabce9 2015-04-30 12:27:16.635463 [ERR]
switch_ivr_async.c:1155 Error writing
/usr/local/freeswitch/recordings/testrecord.fsv

Anyone can help me, What I'm doing wrong?

 
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From ssinyagin at gmail.com  Mon May  4 11:38:59 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 4 May 2015 09:38:59 +0200
Subject: [Freeswitch-users] problems using systemd freeswitch unit
In-Reply-To: <6927.1430712842@ccs.covici.com>
References: <6927.1430712842@ccs.covici.com>
Message-ID: <CACOYK=gRs4=5_ym3BRz3q6HTYe5Jt-FDMZLh3yTqKL0eCqvj=g@mail.gmail.com>

see this thread from few days ago:
http://lists.freeswitch.org/pipermail/freeswitch-users/2015-May/112841.html



On Mon, May 4, 2015 at 6:14 AM,  <covici at ccs.covici.com> wrote:
> Hi.  In the freeswitch tree
> freeswitch-new/debian/freeswitch-systemd.freeswitch.service was there
> and I tried to get it to work in my systemd.  The problem seems to be
> that systemd is getting very confused when fs backgrounds the process.
> I got things to work by adding -nf and changing the type from forking to
> simple, so fs never backgrounds.
>
> My question is, has anyone gotten the original systemd unit to work and
> if so, what did you do?  The only things I changed was the directories
> and I added some ExecStartPre commands that I needed.
>
> Thanks in advance for any ideas.
>
> --
> 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 boros at vmtele.com  Mon May  4 12:35:15 2015
From: boros at vmtele.com (=?windows-1252?Q?Tom=E1=9A_Boros?=)
Date: Mon, 04 May 2015 10:35:15 +0200
Subject: [Freeswitch-users] FS, SS7 and Dialogic IMG2020 in the UK
In-Reply-To: <CAKDfjgcCd2fjnBfzTSCHBG2DkKQN25oC=Zsv9LKQx5dSBuyJJg@mail.gmail.com>
References: <CAA8_NKB_L+vc-KkB=BthxMT8eKwH4ym9dksr3Pk3rwx4hYCvUQ@mail.gmail.com>	<CAKDfjgd=b=HHNUZkvwG1+DC3NB9DNu2Hi9=CoOLGx2Z-bp9wMg@mail.gmail.com>	<20150502194132.5419088.23543.54831@evaristesys.com>
	<CAKDfjgcCd2fjnBfzTSCHBG2DkKQN25oC=Zsv9LKQx5dSBuyJJg@mail.gmail.com>
Message-ID: <55472F43.3090503@vmtele.com>

Hi,

we have just installed a freeswitch for SIP interconnection with 100rel 
enabled. Currently it is running fine, but you have mentioned, that 
there is a bug. Should I be worried about segfaulting or any issues?

btw, we use also dialogic with FS behind it with no 100 rel enabled and 
it is working fine. Its an IMG 1010

Thanks,
Thomas

On 03.05.2015 00:00, Kristian Kielhofner wrote:
> Hi Alex,
>
>   I'm on my phone but Mike/Brian can probably elaborate.
>
>   From what I recall it was somewhat sporadic but reproducible.
>
> On Saturday, May 2, 2015, Alex Balashov <abalashov at evaristesys.com 
> <mailto:abalashov at evaristesys.com>> wrote:
>
>     Kristian,
>
>     What is the nature of the 100rel bug?
>
>     --
>     Alex Balashov | Principal | Evariste Systems LLC
>     303 Perimeter Center North, Suite 300
>     Atlanta, GA 30346
>     United States
>
>     Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
>     Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
>     Sent from my BlackBerry.
>     *From: *Kristian Kielhofner
>     *Sent: *Saturday, May 2, 2015 15:40
>     *To: *FreeSWITCH Users Help
>     *Reply To: *FreeSWITCH Users Help
>     *Subject: *Re: [Freeswitch-users] FS, SS7 and Dialogic IMG2020 in
>     the UK
>
>
>     Hi Gavin,
>
>       The only potential things that jump out to me:
>
>     - Support for/requiring PRACK. This is often suggested/strongly
>     recommended/required with SIP/SS7 interop. FS has a long standing
>     bug with 100rel.
>
>     - Same for SIP-I/SIP-T. FreeSWITCH can pass these through with
>     multipart but it can't currently parse or use them in any
>     effective way. With that said I think I've seen people use LUA and
>     other crazy stuff to parse specific data they absolutely needed...
>
>       Other than that if these can be configured for fairly
>     plain/straightforward SIP there shouldn't be any significant
>     issues that I can see.
>
>     On Saturday, May 2, 2015, Gavin Henry <gavin.henry at gmail.com
>     <javascript:_e(%7B%7D,'cvml','gavin.henry at gmail.com');>> wrote:
>
>         Hi all,
>
>         We're looking to get a pair of Dialogic IMG2020 for some BT Ofcom
>         regulated SS7 interconnects:
>
>         http://www.dialogic.com/en/products/gateways/img/img2020.aspx
>
>         Any issues known with FS behind these?
>
>         Thanks,
>
>         Gavin.
>
>         --
>         http://www.surevoip.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
>
>
>
>     -- 
>     Sent from mobile device
>
>
>
> -- 
> Sent from mobile device
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From covici at ccs.covici.com  Mon May  4 12:52:04 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 04 May 2015 04:52:04 -0400
Subject: [Freeswitch-users] problems using systemd freeswitch unit
In-Reply-To: <CACOYK=gRs4=5_ym3BRz3q6HTYe5Jt-FDMZLh3yTqKL0eCqvj=g@mail.gmail.com>
References: <6927.1430712842@ccs.covici.com>
	<CACOYK=gRs4=5_ym3BRz3q6HTYe5Jt-FDMZLh3yTqKL0eCqvj=g@mail.gmail.com>
Message-ID: <22745.1430729524@ccs.covici.com>

But that was not my problem at all, it was in the ExecStart of
freeswitch  that I had the problem, and the solution was not to allow it
to background, but I was wondering if anyone had gotten things to work
without  adding the -nf parameter.
I also had to change it to Type=simple instead of forking.  I also have
my run directory as /var/run, so that problem is solved that way.


Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> see this thread from few days ago:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-May/112841.html
> 
> 
> 
> On Mon, May 4, 2015 at 6:14 AM,  <covici at ccs.covici.com> wrote:
> > Hi.  In the freeswitch tree
> > freeswitch-new/debian/freeswitch-systemd.freeswitch.service was there
> > and I tried to get it to work in my systemd.  The problem seems to be
> > that systemd is getting very confused when fs backgrounds the process.
> > I got things to work by adding -nf and changing the type from forking to
> > simple, so fs never backgrounds.
> >
> > My question is, has anyone gotten the original systemd unit to work and
> > if so, what did you do?  The only things I changed was the directories
> > and I added some ExecStartPre commands that I needed.
> >
> > Thanks in advance for any ideas.
> >
> > --
> > 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
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 s.safarov at gmail.com  Mon May  4 13:48:33 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 4 May 2015 12:48:33 +0300
Subject: [Freeswitch-users] Simulate incoming SIP Message
In-Reply-To: <CACyT-3=AVrm3NkFVbCLVVbZPB_JE9jt87NA0gCxeAbdgFpL5Uw@mail.gmail.com>
References: <CACyT-3=AVrm3NkFVbCLVVbZPB_JE9jt87NA0gCxeAbdgFpL5Uw@mail.gmail.com>
Message-ID: <CAHtxdDfX7Kv8uVs-h6xqUKK4WqVtBB--LpJ4isBQok77G3Ckrw@mail.gmail.com>

Why you not use sipp?

On Mon, May 4, 2015 at 7:48 AM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
wrote:

> Hello FreeSWITCH Users,
>
> I am trying to simulate an incoming SIP message in order to test my
> chatplan programmatically. Basically, I want FS to send itself a SIP
> message, which will then be routed through the chatplan. I've been able to
> generate the message using ESL and events, however FS refuses to send it
> noting that it is "Not sending message to ourselves!"
>
> That's exactly what I want it to do. I've used this functionality before
> for calls, as generating a call that connects back to FS (and hooking it to
> echo) is a great way to test the dialplan. Is there no way to get FS to
> send itself a message?
>
> 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 flokrrr at gmail.com  Mon May  4 16:02:05 2015
From: flokrrr at gmail.com (Florent Krieg)
Date: Mon, 4 May 2015 14:02:05 +0200
Subject: [Freeswitch-users] FS, SS7 and Dialogic IMG2020 in the UK
In-Reply-To: <55472F43.3090503@vmtele.com>
References: <CAA8_NKB_L+vc-KkB=BthxMT8eKwH4ym9dksr3Pk3rwx4hYCvUQ@mail.gmail.com>
	<CAKDfjgd=b=HHNUZkvwG1+DC3NB9DNu2Hi9=CoOLGx2Z-bp9wMg@mail.gmail.com>
	<20150502194132.5419088.23543.54831@evaristesys.com>
	<CAKDfjgcCd2fjnBfzTSCHBG2DkKQN25oC=Zsv9LKQx5dSBuyJJg@mail.gmail.com>
	<55472F43.3090503@vmtele.com>
Message-ID: <CAABKsuRt1TXGRE7dSdYmt-j7PTbO6U-YoQAY2zR2x6kNDQ5PWQ@mail.gmail.com>

Same here, FS boxes receiving and sending SIP trafic from/to IMG1010
gateways.
It has been running for two years without any issue.

Florent

2015-05-04 10:35 GMT+02:00 Tom?? Boros <boros at vmtele.com>:

>  Hi,
>
> we have just installed a freeswitch for SIP interconnection with 100rel
> enabled. Currently it is running fine, but you have mentioned, that there
> is a bug. Should I be worried about segfaulting or any issues?
>
> btw, we use also dialogic with FS behind it with no 100 rel enabled and it
> is working fine. Its an IMG 1010
>
> Thanks,
> Thomas
>
>
> On 03.05.2015 00:00, Kristian Kielhofner wrote:
>
> Hi Alex,
>
>    I'm on my phone but Mike/Brian can probably elaborate.
>
>    From what I recall it was somewhat sporadic but reproducible.
>
> On Saturday, May 2, 2015, Alex Balashov <abalashov at evaristesys.com> wrote:
>
>>  Kristian,
>>
>>  What is the nature of the 100rel bug?
>>
>>  --
>> Alex Balashov | Principal | Evariste Systems LLC
>> 303 Perimeter Center North, Suite 300
>> Atlanta, GA 30346
>> United States
>>
>> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>
>> Sent from my BlackBerry.
>>    *From: *Kristian Kielhofner
>> *Sent: *Saturday, May 2, 2015 15:40
>> *To: *FreeSWITCH Users Help
>> *Reply To: *FreeSWITCH Users Help
>> *Subject: *Re: [Freeswitch-users] FS, SS7 and Dialogic IMG2020 in the UK
>>
>> Hi Gavin,
>>
>>    The only potential things that jump out to me:
>>
>>  - Support for/requiring PRACK. This is often suggested/strongly
>> recommended/required with SIP/SS7 interop. FS has a long standing bug with
>> 100rel.
>>
>>  - Same for SIP-I/SIP-T. FreeSWITCH can pass these through with
>> multipart but it can't currently parse or use them in any effective way.
>> With that said I think I've seen people use LUA and other crazy stuff to
>> parse specific data they absolutely needed...
>>
>>    Other than that if these can be configured for fairly
>> plain/straightforward SIP there shouldn't be any significant issues that I
>> can see.
>>
>> On Saturday, May 2, 2015, Gavin Henry <gavin.henry at gmail.com> wrote:
>>
>>> Hi all,
>>>
>>> We're looking to get a pair of Dialogic IMG2020 for some BT Ofcom
>>> regulated SS7 interconnects:
>>>
>>> http://www.dialogic.com/en/products/gateways/img/img2020.aspx
>>>
>>> Any issues known with FS behind these?
>>>
>>> Thanks,
>>>
>>> Gavin.
>>>
>>> --
>>> http://www.surevoip.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
>>>
>>
>>
>> --
>> Sent from mobile device
>>
>>
>
> --
> Sent from mobile device
>
>
> _________________________________________________________________________
> 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 kamil.nigmatullin at gmail.com  Mon May  4 16:51:58 2015
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Mon, 4 May 2015 18:51:58 +0600
Subject: [Freeswitch-users] 407 Proxy authentication
In-Reply-To: <CACOYK=iJSbHYS8WUZGeM5J=0_NTsX9i+k_Z5PoVdyovdY8F9qA@mail.gmail.com>
References: <CAG6TCM8hOWEG89DC+96ehBWGpwU6nXiT2q8EfVXa1XK5gZc-fA@mail.gmail.com>
	<AEDB707B-FA8D-412A-9E99-B5B8DA7D66CD@jerris.com>
	<CAG6TCM9O20r7u1G5Mq-_7m+Vgsz7BBEPSJ+7r5TvWRSPNQ8ntg@mail.gmail.com>
	<CAM33Ch4HUjjHK2DmXDwu3tVn7YcrQ2yoJRFcN6feXi2w59NWUQ@mail.gmail.com>
	<CAG6TCM8oM_eRcDP1hMA9Z9_65bCkC+sB0sVbCKqgt-8ac4ZHdQ@mail.gmail.com>
	<CACOYK=iJSbHYS8WUZGeM5J=0_NTsX9i+k_Z5PoVdyovdY8F9qA@mail.gmail.com>
Message-ID: <CA+XtiecnBq9aWDxyd-Oi2tMDE70HFNr1zuVP650Df2-bX5gfGQ@mail.gmail.com>

Do you use port 5060 or 5080 (I think you use standard config)?

2015-04-30 14:23 GMT+06:00 Stanislav Sinyagin <ssinyagin at gmail.com>:

> I think it's the proper time you would start reading the book :-)
> https://www.packtpub.com/networking-and-servers/freeswitch-12
>
> or pay someone for training
>
>
> On Thu, Apr 30, 2015 at 8:48 AM, bhavik patel
> <bhavikpatel14388 at gmail.com> wrote:
> > Hi,
> >
> > Thanks for reply,I tried but still same.
> > Any other configuration missing.
> >
> > On Wed, Apr 29, 2015 at 10:59 PM, Varghese Paul <
> varghesepaul87 at gmail.com>
> > wrote:
> >>
> >> Hi ,
> >>
> >> You can set this variable in your inbound sip profile.
> >>
> >> <param name="auth-calls" value="true"/>
> >>
> >>
> >> Regards
> >>
> >> Varghese Paul
> >>
> >>
> >> On Wed, Apr 29, 2015 at 12:50 AM, bhavik patel
> >> <bhavikpatel14388 at gmail.com> wrote:
> >>>
> >>> If you know any specific sip profile parameter then please suggest me.
> >>>
> >>>
> >>> On Tue, Apr 28, 2015 at 7:23 PM, Michael Jerris <mike at jerris.com>
> wrote:
> >>>>
> >>>> Check the profile settings for auth and acl.  The default configs
> >>>> include profiles for anonymous and auth.
> >>>>
> >>>> > On Apr 28, 2015, at 9:26 AM, bhavik patel <
> bhavikpatel14388 at gmail.com>
> >>>> > wrote:
> >>>> >
> >>>> > Hi,
> >>>> >
> >>>> > We need 407 Proxy authentication require sip dialog while calling
> >>>> > between sip to sip extension. Currently when we are calling, we
> never got
> >>>> > that sip dialog.
> >>>> >
> >>>> > I am sure there must be some configuration to achieve this.
> >>>> >
> >>>> > Can anyone please help us to make it working?
> >>>>
> >>>>
> >>>>
> >>>>
> _________________________________________________________________________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://confluence.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >>>
> >>> --
> >>> Thanks,
> >>> Bhavik Patel
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Thanks,
> > Bhavik Patel
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kamil Nigmatullin
Tel: 77272323748
mob: 7 (707) 2517003
Skype: kamil.nigmatullin
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From bhavikpatel14388 at gmail.com  Mon May  4 17:18:55 2015
From: bhavikpatel14388 at gmail.com (bhavik patel)
Date: Mon, 4 May 2015 18:48:55 +0530
Subject: [Freeswitch-users] 407 Proxy authentication
In-Reply-To: <CA+XtiecnBq9aWDxyd-Oi2tMDE70HFNr1zuVP650Df2-bX5gfGQ@mail.gmail.com>
References: <CAG6TCM8hOWEG89DC+96ehBWGpwU6nXiT2q8EfVXa1XK5gZc-fA@mail.gmail.com>
	<AEDB707B-FA8D-412A-9E99-B5B8DA7D66CD@jerris.com>
	<CAG6TCM9O20r7u1G5Mq-_7m+Vgsz7BBEPSJ+7r5TvWRSPNQ8ntg@mail.gmail.com>
	<CAM33Ch4HUjjHK2DmXDwu3tVn7YcrQ2yoJRFcN6feXi2w59NWUQ@mail.gmail.com>
	<CAG6TCM8oM_eRcDP1hMA9Z9_65bCkC+sB0sVbCKqgt-8ac4ZHdQ@mail.gmail.com>
	<CACOYK=iJSbHYS8WUZGeM5J=0_NTsX9i+k_Z5PoVdyovdY8F9qA@mail.gmail.com>
	<CA+XtiecnBq9aWDxyd-Oi2tMDE70HFNr1zuVP650Df2-bX5gfGQ@mail.gmail.com>
Message-ID: <CAG6TCM8x5m2bQAY71vA9FdW0jacnpEEKo2DupGVJxJyPFnVS8Q@mail.gmail.com>

yes that's default configuration and port is 5060

On Mon, May 4, 2015 at 6:21 PM, Kamil Nigmatullin <
kamil.nigmatullin at gmail.com> wrote:

> Do you use port 5060 or 5080 (I think you use standard config)?
>
> 2015-04-30 14:23 GMT+06:00 Stanislav Sinyagin <ssinyagin at gmail.com>:
>
>> I think it's the proper time you would start reading the book :-)
>> https://www.packtpub.com/networking-and-servers/freeswitch-12
>>
>> or pay someone for training
>>
>>
>> On Thu, Apr 30, 2015 at 8:48 AM, bhavik patel
>> <bhavikpatel14388 at gmail.com> wrote:
>> > Hi,
>> >
>> > Thanks for reply,I tried but still same.
>> > Any other configuration missing.
>> >
>> > On Wed, Apr 29, 2015 at 10:59 PM, Varghese Paul <
>> varghesepaul87 at gmail.com>
>> > wrote:
>> >>
>> >> Hi ,
>> >>
>> >> You can set this variable in your inbound sip profile.
>> >>
>> >> <param name="auth-calls" value="true"/>
>> >>
>> >>
>> >> Regards
>> >>
>> >> Varghese Paul
>> >>
>> >>
>> >> On Wed, Apr 29, 2015 at 12:50 AM, bhavik patel
>> >> <bhavikpatel14388 at gmail.com> wrote:
>> >>>
>> >>> If you know any specific sip profile parameter then please suggest me.
>> >>>
>> >>>
>> >>> On Tue, Apr 28, 2015 at 7:23 PM, Michael Jerris <mike at jerris.com>
>> wrote:
>> >>>>
>> >>>> Check the profile settings for auth and acl.  The default configs
>> >>>> include profiles for anonymous and auth.
>> >>>>
>> >>>> > On Apr 28, 2015, at 9:26 AM, bhavik patel <
>> bhavikpatel14388 at gmail.com>
>> >>>> > wrote:
>> >>>> >
>> >>>> > Hi,
>> >>>> >
>> >>>> > We need 407 Proxy authentication require sip dialog while calling
>> >>>> > between sip to sip extension. Currently when we are calling, we
>> never got
>> >>>> > that sip dialog.
>> >>>> >
>> >>>> > I am sure there must be some configuration to achieve this.
>> >>>> >
>> >>>> > Can anyone please help us to make it working?
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> _________________________________________________________________________
>> >>>> Professional FreeSWITCH Consulting Services:
>> >>>> consulting at freeswitch.org
>> >>>> http://www.freeswitchsolutions.com
>> >>>>
>> >>>> Official FreeSWITCH Sites
>> >>>> http://www.freeswitch.org
>> >>>> http://confluence.freeswitch.org
>> >>>> http://www.cluecon.com
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Thanks,
>> >>> Bhavik Patel
>> >>>
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://confluence.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> >
>> > --
>> > Thanks,
>> > Bhavik Patel
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Kamil Nigmatullin
> Tel: 77272323748
> mob: 7 (707) 2517003
> Skype: kamil.nigmatullin
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks,
Bhavik Patel
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From faycal.noushi at gmail.com  Sun May  3 02:20:55 2015
From: faycal.noushi at gmail.com (=?UTF-8?Q?fay=C3=A7al_noushi?=)
Date: Sat, 2 May 2015 23:20:55 +0100
Subject: [Freeswitch-users] FS, SS7 and Dialogic IMG2020 in the UK
In-Reply-To: <CAKDfjgcCd2fjnBfzTSCHBG2DkKQN25oC=Zsv9LKQx5dSBuyJJg@mail.gmail.com>
References: <CAA8_NKB_L+vc-KkB=BthxMT8eKwH4ym9dksr3Pk3rwx4hYCvUQ@mail.gmail.com>
	<CAKDfjgd=b=HHNUZkvwG1+DC3NB9DNu2Hi9=CoOLGx2Z-bp9wMg@mail.gmail.com>
	<20150502194132.5419088.23543.54831@evaristesys.com>
	<CAKDfjgcCd2fjnBfzTSCHBG2DkKQN25oC=Zsv9LKQx5dSBuyJJg@mail.gmail.com>
Message-ID: <CAEUm5BQUUiJsVVZDLc-ZamT+Fj2yJ_zvPmoF7Cx+kLHxzDyK1Q@mail.gmail.com>

Hi,

  I've had to use FS in a SIP-I context. The first issue was that FS
truncates SIP messages once it finds 0x00... The dirty workaround was
creating a binary to hex representation converter module in Kamailio.

  We also had some problems with multipart invites/responses. Especially
multipart responses with SDP in it as well as ISUP. Here are the bugs :

https://freeswitch.org/jira/browse/FS-6875
https://freeswitch.org/jira/browse/FS-6877

On Sat, May 2, 2015 at 11:00 PM, Kristian Kielhofner <kris at kriskinc.com>
wrote:

> Hi Alex,
>
>   I'm on my phone but Mike/Brian can probably elaborate.
>
>   From what I recall it was somewhat sporadic but reproducible.
>
>
> On Saturday, May 2, 2015, Alex Balashov <abalashov at evaristesys.com> wrote:
>
>> Kristian,
>>
>> What is the nature of the 100rel bug?
>>
>>  --
>> Alex Balashov | Principal | Evariste Systems LLC
>> 303 Perimeter Center North, Suite 300
>> Atlanta, GA 30346
>> United States
>>
>> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>
>> Sent from my BlackBerry.
>>   *From: *Kristian Kielhofner
>> *Sent: *Saturday, May 2, 2015 15:40
>> *To: *FreeSWITCH Users Help
>> *Reply To: *FreeSWITCH Users Help
>> *Subject: *Re: [Freeswitch-users] FS, SS7 and Dialogic IMG2020 in the UK
>>
>> Hi Gavin,
>>
>>   The only potential things that jump out to me:
>>
>> - Support for/requiring PRACK. This is often suggested/strongly
>> recommended/required with SIP/SS7 interop. FS has a long standing bug with
>> 100rel.
>>
>> - Same for SIP-I/SIP-T. FreeSWITCH can pass these through with multipart
>> but it can't currently parse or use them in any effective way. With that
>> said I think I've seen people use LUA and other crazy stuff to parse
>> specific data they absolutely needed...
>>
>>   Other than that if these can be configured for fairly
>> plain/straightforward SIP there shouldn't be any significant issues that I
>> can see.
>>
>> On Saturday, May 2, 2015, Gavin Henry <gavin.henry at gmail.com> wrote:
>>
>>> Hi all,
>>>
>>> We're looking to get a pair of Dialogic IMG2020 for some BT Ofcom
>>> regulated SS7 interconnects:
>>>
>>> http://www.dialogic.com/en/products/gateways/img/img2020.aspx
>>>
>>> Any issues known with FS behind these?
>>>
>>> Thanks,
>>>
>>> Gavin.
>>>
>>> --
>>> http://www.surevoip.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
>>>
>>
>>
>> --
>> Sent from mobile device
>>
>>
>
> --
> Sent from mobile device
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
NOUSHI Fay?al
GSM : +212 661 56 03 37
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From roman at dissauer.net  Mon May  4 19:08:52 2015
From: roman at dissauer.net (Roman Dissauer)
Date: Mon, 4 May 2015 17:08:52 +0200
Subject: [Freeswitch-users] restrict gateway to domain
Message-ID: <8C22FA96-C15C-4D62-A97A-139D3E2196AA@dissauer.net>

Hey!

I?m having troubles restricting a gateway to a single domain in a multi-tenant freeswitch setup.

It is clear for me that if the gateway is defined in sip_profiles everyone on the switch can bridge to "sofia/gateway/<gateway id>/<number>?. So I defined the gateway in directory/<domain>/<gateway>.xml. Now the gateway is wrapped in <domain name="<domain>"> tag, but users from other domains can still bridge to "sofia/gateway/<gateway id>/<number>?.

What do I miss here?

Thanks!
Roman Dissauer

From ssinyagin at gmail.com  Mon May  4 19:18:42 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 4 May 2015 17:18:42 +0200
Subject: [Freeswitch-users] restrict gateway to domain
In-Reply-To: <8C22FA96-C15C-4D62-A97A-139D3E2196AA@dissauer.net>
References: <8C22FA96-C15C-4D62-A97A-139D3E2196AA@dissauer.net>
Message-ID: <CACOYK=gnP0dY94VppHEofcK8eSFsi=HRhYyuYsCbk3pvyGgVvA@mail.gmail.com>

hi Roman,

gateways are global anyway. It's up to the  upper layers (XML
dialplan, ESL appication) to take care about security and access
permissions.



On Mon, May 4, 2015 at 5:08 PM, Roman Dissauer <roman at dissauer.net> wrote:
> Hey!
>
> I?m having troubles restricting a gateway to a single domain in a multi-tenant freeswitch setup.
>
> It is clear for me that if the gateway is defined in sip_profiles everyone on the switch can bridge to "sofia/gateway/<gateway id>/<number>?. So I defined the gateway in directory/<domain>/<gateway>.xml. Now the gateway is wrapped in <domain name="<domain>"> tag, but users from other domains can still bridge to "sofia/gateway/<gateway id>/<number>?.
>
> What do I miss here?
>
> Thanks!
> Roman Dissauer
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 roman at dissauer.net  Mon May  4 19:27:11 2015
From: roman at dissauer.net (Roman Dissauer)
Date: Mon, 4 May 2015 17:27:11 +0200
Subject: [Freeswitch-users] restrict gateway to domain
In-Reply-To: <CACOYK=gnP0dY94VppHEofcK8eSFsi=HRhYyuYsCbk3pvyGgVvA@mail.gmail.com>
References: <8C22FA96-C15C-4D62-A97A-139D3E2196AA@dissauer.net>
	<CACOYK=gnP0dY94VppHEofcK8eSFsi=HRhYyuYsCbk3pvyGgVvA@mail.gmail.com>
Message-ID: <9083A583-3395-4322-AF9B-64AC952E58DF@dissauer.net>

That makes sense, so I don?t have to investigate further in this topic :)

Thanks!

> Am 04.05.2015 um 17:18 schrieb Stanislav Sinyagin <ssinyagin at gmail.com>:
> 
> hi Roman,
> 
> gateways are global anyway. It's up to the  upper layers (XML
> dialplan, ESL appication) to take care about security and access
> permissions.
> 
> 
> 
> On Mon, May 4, 2015 at 5:08 PM, Roman Dissauer <roman at dissauer.net> wrote:
>> Hey!
>> 
>> I?m having troubles restricting a gateway to a single domain in a multi-tenant freeswitch setup.
>> 
>> It is clear for me that if the gateway is defined in sip_profiles everyone on the switch can bridge to "sofia/gateway/<gateway id>/<number>?. So I defined the gateway in directory/<domain>/<gateway>.xml. Now the gateway is wrapped in <domain name="<domain>"> tag, but users from other domains can still bridge to "sofia/gateway/<gateway id>/<number>?.
>> 
>> What do I miss here?
>> 
>> Thanks!
>> Roman Dissauer
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 May  4 19:30:22 2015
From: bote_radio at botecomm.com (Bote Man)
Date: Mon, 4 May 2015 11:30:22 -0400
Subject: [Freeswitch-users] restrict gateway to domain
In-Reply-To: <CACOYK=gnP0dY94VppHEofcK8eSFsi=HRhYyuYsCbk3pvyGgVvA@mail.gmail.com>
References: <8C22FA96-C15C-4D62-A97A-139D3E2196AA@dissauer.net>
	<CACOYK=gnP0dY94VppHEofcK8eSFsi=HRhYyuYsCbk3pvyGgVvA@mail.gmail.com>
Message-ID: <007501d0867f$3e41c290$bac547b0$@botecomm.com>

Agreed. The system administrator creates the dialplans, so that is where routing to gateways is controlled.

Bote


> -----Original Message-----
> From: Stanislav Sinyagin
> Sent: Monday, 04 May, 2015 11:19
> Subject: Re: [Freeswitch-users] restrict gateway to domain
> 
> hi Roman,
> 
> gateways are global anyway. It's up to the  upper layers (XML
> dialplan, ESL appication) to take care about security and access
> permissions.
> 
> 
> 
> On Mon, May 4, 2015 at 5:08 PM, Roman Dissauer <roman at dissauer.net>
> wrote:
> > Hey!
> >
> > I?m having troubles restricting a gateway to a single domain in a multi-
> tenant freeswitch setup.
> >
> > It is clear for me that if the gateway is defined in sip_profiles everyone on
> the switch can bridge to "sofia/gateway/<gateway id>/<number>?. So I
> defined the gateway in directory/<domain>/<gateway>.xml. Now the
> gateway is wrapped in <domain name="<domain>"> tag, but users from
> other domains can still bridge to "sofia/gateway/<gateway id>/<number>?.
> >
> > What do I miss here?
> >
> > Thanks!
> > Roman Dissauer
> >



From mike at jerris.com  Mon May  4 20:38:49 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 May 2015 12:38:49 -0400
Subject: [Freeswitch-users] core dumped. PANIC: unprotected error in
	call to Lua API (multiple Lua VMs detected)
In-Reply-To: <CAGVOjVUO3+Dz66u1+q+hgMYr0g6r6gpL3CTNQA9t=yhd0+Ke3w@mail.gmail.com>
References: <CAGVOjVUO3+Dz66u1+q+hgMYr0g6r6gpL3CTNQA9t=yhd0+Ke3w@mail.gmail.com>
Message-ID: <C6C383EF-F4BC-4E19-BCEF-7A34E988BAB7@jerris.com>

https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA <https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA>

> On May 2, 2015, at 3:19 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
> 
> Hi, users.
> 
> I have installed the freeswitch(FreeSWITCH Version 1.4.18~64bit ( 64bit) on fresh installation of  ubuntu(14.04). When i try to load mod_vlc i see this error and core dumped.
> 
> 
> PANIC: unprotected error in call to Lua API (multiple Lua VMs detected)
> 
> Your help would be much appreciated.
> Thanks

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From mike at jerris.com  Mon May  4 20:40:59 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 May 2015 12:40:59 -0400
Subject: [Freeswitch-users] Help with FSV record.
In-Reply-To: <02732363.20150504091623@seznam.cz>
References: <1430394389391-7596159.post@n2.nabble.com>
	<8D8CF81D-30E4-4AF3-BDD1-358F9492342D@jerris.com>
	<02732363.20150504091623@seznam.cz>
Message-ID: <1698FF51-3959-45E0-95BC-BAD0CA26FD3E@jerris.com>

we are aiming at merging into master sometime in the next few weeks and a final release this summer.

> On May 4, 2015, at 3:16 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> 
> Hello,
> 
> Thanks for the answer. When can we expect the release?
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel. 775-415-382
> 
> ?tvrtek 30. dubna 2015, 14:34:47, napsal jste:
> 
> 
> Recording video was barely a feature in 1.4.  Full support for this and everything you are trying to do is coming in 1.6.  Stay tuned for beta announcements on that coming soon.  You can try out the feature branch if you like, more info is available at: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video <https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video> .  We are still working on docs so please be patient as we gather all of that information.
> 
> 
> On Apr 30, 2015, at 7:46 AM, dex_Cz <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
> 
> Hi guys!
> I have a question. Somebody tried to record in the FSV file when i bridge
> call?
> I hame a simple context from example. But it is not working.
> <include>
> <context name="centos">
> <extension name="from-centos">
> <condition field="destination_number" expression="^(onops)$">
> <action application="record_fsv"
> data="/usr/local/freeswitch/recordings/testrecord.fsv"/>
> <action application="sleep" data="100"/>
> <action application="bridge" data="user/1004 at 172.16.0.2 <mailto:user/1004 at 172.16.0.2>"/>
> <action application="hangup"/>
> </condition>
> </extension>
> </context>
> </include>
> 
> I tried to change the context
> <include>
> <context name="centos">
> <extension name="from-centos">
>  <condition field="destination_number" expression="^(onops)$">
>   <action application="export" data="execute_on_answer=uuid_record ${uuid}
> start /usr/local/freeswitch/recordings/testrecord.fsv"/> 
>   <action application="bridge"
> data="{ignore_early_media=true}user/1004 at 172.16.0.2 <mailto:user/1004 at 172.16.0.2>"/>
>   <action application="hangup"/>
>  </condition>
> </extension>
> </context>
> </include>
> 
> It also did not help. In the end I try to enter these commands from the
> console
> uuid_setvar <uuid> enable_file_write_buffering false (its says ok)
> and
> uuid_record <uuid> start /usr/local/freeswitch/recordings/testrecord.fsv
> 
> But i have a error 
> 
> 2015-04-30 12:27:16.635463 [ERR] mod_fsv.c:702 You are asking to write 342
> bytes of data which is not supported. Please set
> enable_file_write_buffering=false to use .fsv format
> 5e9ce2b0-ef23-11e4-b181-d1f037dabce9 2015-04-30 12:27:16.635463 [ERR]
> switch_ivr_async.c:1155 Error writing
> /usr/local/freeswitch/recordings/testrecord.fsv
> 
> Anyone can help me, What I'm doing wrong?

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From mike at jerris.com  Mon May  4 20:44:27 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 May 2015 12:44:27 -0400
Subject: [Freeswitch-users] Help with FSV record.
In-Reply-To: <1698FF51-3959-45E0-95BC-BAD0CA26FD3E@jerris.com>
References: <1430394389391-7596159.post@n2.nabble.com>
	<8D8CF81D-30E4-4AF3-BDD1-358F9492342D@jerris.com>
	<02732363.20150504091623@seznam.cz>
	<1698FF51-3959-45E0-95BC-BAD0CA26FD3E@jerris.com>
Message-ID: <3F10A55D-AF20-4E35-8CF7-E2E69CD04221@jerris.com>

But please don't wait for that.  Please put this in your lab and try it out and give us some feedback.  We need the community feedback to help get this out to you.


> On May 4, 2015, at 12:40 PM, Michael Jerris <mike at jerris.com> wrote:
> 
> we are aiming at merging into master sometime in the next few weeks and a final release this summer.
> 
>> On May 4, 2015, at 3:16 AM, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
>> 
>> Hello,
>> 
>> Thanks for the answer. When can we expect the release?
>> 
>> -- 
>> S pozdravem,
>> Ing.Denis Jakovlev                           
>> mob.tel. 775-415-382
>> 
>> ?tvrtek 30. dubna 2015, 14:34:47, napsal jste:
>> 
>> 
>> Recording video was barely a feature in 1.4.  Full support for this and everything you are trying to do is coming in 1.6.  Stay tuned for beta announcements on that coming soon.  You can try out the feature branch if you like, more info is available at: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video <https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video> .  We are still working on docs so please be patient as we gather all of that information.
>> 
>> 
>> On Apr 30, 2015, at 7:46 AM, dex_Cz <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
>> 
>> Hi guys!
>> I have a question. Somebody tried to record in the FSV file when i bridge
>> call?
>> I hame a simple context from example. But it is not working.
>> <include>
>> <context name="centos">
>> <extension name="from-centos">
>> <condition field="destination_number" expression="^(onops)$">
>> <action application="record_fsv"
>> data="/usr/local/freeswitch/recordings/testrecord.fsv"/>
>> <action application="sleep" data="100"/>
>> <action application="bridge" data="user/1004 at 172.16.0.2 <mailto:user/1004 at 172.16.0.2>"/>
>> <action application="hangup"/>
>> </condition>
>> </extension>
>> </context>
>> </include>
>> 
>> I tried to change the context
>> <include>
>> <context name="centos">
>> <extension name="from-centos">
>>  <condition field="destination_number" expression="^(onops)$">
>>   <action application="export" data="execute_on_answer=uuid_record ${uuid}
>> start /usr/local/freeswitch/recordings/testrecord.fsv"/> 
>>   <action application="bridge"
>> data="{ignore_early_media=true}user/1004 at 172.16.0.2 <mailto:user/1004 at 172.16.0.2>"/>
>>   <action application="hangup"/>
>>  </condition>
>> </extension>
>> </context>
>> </include>
>> 
>> It also did not help. In the end I try to enter these commands from the
>> console
>> uuid_setvar <uuid> enable_file_write_buffering false (its says ok)
>> and
>> uuid_record <uuid> start /usr/local/freeswitch/recordings/testrecord.fsv
>> 
>> But i have a error 
>> 
>> 2015-04-30 12:27:16.635463 [ERR] mod_fsv.c:702 You are asking to write 342
>> bytes of data which is not supported. Please set
>> enable_file_write_buffering=false to use .fsv format
>> 5e9ce2b0-ef23-11e4-b181-d1f037dabce9 2015-04-30 12:27:16.635463 [ERR]
>> switch_ivr_async.c:1155 Error writing
>> /usr/local/freeswitch/recordings/testrecord.fsv
>> 
>> Anyone can help me, What I'm doing wrong?
> 

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From Sharath.Kumar at meZocliq.com  Mon May  4 22:07:40 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Mon, 4 May 2015 18:07:40 +0000
Subject: [Freeswitch-users] DTMF relay with INFO across bridge
Message-ID: <CY1PR0501MB10986151A3C7FF38ED9A806384D20@CY1PR0501MB1098.namprd05.prod.outlook.com>

Hi,

I looked around in the mailing list and don't see any conclusive solution to this problem.

I have a setup like this below.

Webrtc client -----FS----sonus gateway.

The Webrtc client is sending INFO as DTMF. The FS currently bridges the call to the gateway and in the outbound leg uses rfc2833.

Problem: Some calls suffer from missing DTMF and repeated duplicate DTMF.

Solution: Use INFO all through to sonus gateway ?

How do I achieve this ?
In the internal profile if I change the "dtmf-type" to "info". Nothing really happens, it still sends rfc2833 dtmf.    I believe it is because of "liberal-dtmf" is "true".[accept any but always offer rfc2833] But if I change "liberal-dtmf" to "false". I get "channel is not configured to use info dtmf".

Is this a bug ? Is there any way around it ?

Any input much appreciated.
Thank you,
Sharath
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From yadenis at seznam.cz  Mon May  4 23:26:18 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Mon, 4 May 2015 21:26:18 +0200
Subject: [Freeswitch-users] Help with FSV record.
In-Reply-To: <3F10A55D-AF20-4E35-8CF7-E2E69CD04221@jerris.com>
References: <1430394389391-7596159.post@n2.nabble.com>
	<8D8CF81D-30E4-4AF3-BDD1-358F9492342D@jerris.com>
	<02732363.20150504091623@seznam.cz>
	<1698FF51-3959-45E0-95BC-BAD0CA26FD3E@jerris.com>
	<3F10A55D-AF20-4E35-8CF7-E2E69CD04221@jerris.com>
Message-ID: <DEA257FA-71A6-4F9A-BEBF-F2192CCC370A@seznam.cz>

Hello.

Thank you for your response. Of course I'll wait master witch I can put on centos, and of course I'll let you know how it will work. 

As for the current version 1.5.15b. I really would not be able to record video at a bridge calls?

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

> On 4. 5. 2015, at 18:44, Michael Jerris <mike at jerris.com> wrote:
> 
> But please don't wait for that.  Please put this in your lab and try it out and give us some feedback.  We need the community feedback to help get this out to you.
> 
> 
>> On May 4, 2015, at 12:40 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> 
>> we are aiming at merging into master sometime in the next few weeks and a final release this summer.
>> 
>>> On May 4, 2015, at 3:16 AM, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
>>> 
>>> Hello,
>>> 
>>> Thanks for the answer. When can we expect the release?
>>> 
>>> -- 
>>> S pozdravem,
>>> Ing.Denis Jakovlev                           
>>> mob.tel. 775-415-382
>>> 
>>> ?tvrtek 30. dubna 2015, 14:34:47, napsal jste:
>>> 
>>> 
>>> Recording video was barely a feature in 1.4.  Full support for this and everything you are trying to do is coming in 1.6.  Stay tuned for beta announcements on that coming soon.  You can try out the feature branch if you like, more info is available at: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video <https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video> .  We are still working on docs so please be patient as we gather all of that information.
>>> 
>>> 
>>> On Apr 30, 2015, at 7:46 AM, dex_Cz <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
>>> 
>>> Hi guys!
>>> I have a question. Somebody tried to record in the FSV file when i bridge
>>> call?
>>> I hame a simple context from example. But it is not working.
>>> <include>
>>> <context name="centos">
>>> <extension name="from-centos">
>>> <condition field="destination_number" expression="^(onops)$">
>>> <action application="record_fsv"
>>> data="/usr/local/freeswitch/recordings/testrecord.fsv"/>
>>> <action application="sleep" data="100"/>
>>> <action application="bridge" data="user/1004 at 172.16.0.2 <mailto:user/1004 at 172.16.0.2>"/>
>>> <action application="hangup"/>
>>> </condition>
>>> </extension>
>>> </context>
>>> </include>
>>> 
>>> I tried to change the context
>>> <include>
>>> <context name="centos">
>>> <extension name="from-centos">
>>>  <condition field="destination_number" expression="^(onops)$">
>>>   <action application="export" data="execute_on_answer=uuid_record ${uuid}
>>> start /usr/local/freeswitch/recordings/testrecord.fsv"/> 
>>>   <action application="bridge"
>>> data="{ignore_early_media=true}user/1004 at 172.16.0.2 <mailto:user/1004 at 172.16.0.2>"/>
>>>   <action application="hangup"/>
>>>  </condition>
>>> </extension>
>>> </context>
>>> </include>
>>> 
>>> It also did not help. In the end I try to enter these commands from the
>>> console
>>> uuid_setvar <uuid> enable_file_write_buffering false (its says ok)
>>> and
>>> uuid_record <uuid> start /usr/local/freeswitch/recordings/testrecord.fsv
>>> 
>>> But i have a error 
>>> 
>>> 2015-04-30 12:27:16.635463 [ERR] mod_fsv.c:702 You are asking to write 342
>>> bytes of data which is not supported. Please set
>>> enable_file_write_buffering=false to use .fsv format
>>> 5e9ce2b0-ef23-11e4-b181-d1f037dabce9 2015-04-30 12:27:16.635463 [ERR]
>>> switch_ivr_async.c:1155 Error writing
>>> /usr/local/freeswitch/recordings/testrecord.fsv
>>> 
>>> Anyone can help me, What I'm doing wrong?
>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ssinyagin at gmail.com  Tue May  5 01:15:40 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 4 May 2015 23:15:40 +0200
Subject: [Freeswitch-users] Help with FSV record.
In-Reply-To: <8D8CF81D-30E4-4AF3-BDD1-358F9492342D@jerris.com>
References: <1430394389391-7596159.post@n2.nabble.com>
	<8D8CF81D-30E4-4AF3-BDD1-358F9492342D@jerris.com>
Message-ID: <CACOYK=jaQKkZEm5mFbm-11kh68iExNQU1UkQLhDFg=3F3FYZ2A@mail.gmail.com>

is there a plan to re-implement "echo" application to work on waveform
level (or picture frame level) instead of copying the RTP packets?  It
would be then handy for audio and video testing.

At the moment echo is not compatible with SILK and Opus codecs because
they don't tolerate simple RTP duplication.


On Thu, Apr 30, 2015 at 2:34 PM, Michael Jerris <mike at jerris.com> wrote:
> Recording video was barely a feature in 1.4.  Full support for this and
> everything you are trying to do is coming in 1.6.  Stay tuned for beta
> announcements on that coming soon.  You can try out the feature branch if
> you like, more info is available at:
> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video .
> We are still working on docs so please be patient as we gather all of that
> information.
>
>
> On Apr 30, 2015, at 7:46 AM, dex_Cz <yadenis at seznam.cz> wrote:
>
> Hi guys!
> I have a question. Somebody tried to record in the FSV file when i bridge
> call?
> I hame a simple context from example. But it is not working.
> <include>
> <context name="centos">
> <extension name="from-centos">
> <condition field="destination_number" expression="^(onops)$">
> <action application="record_fsv"
> data="/usr/local/freeswitch/recordings/testrecord.fsv"/>
> <action application="sleep" data="100"/>
> <action application="bridge" data="user/1004 at 172.16.0.2"/>
> <action application="hangup"/>
> </condition>
> </extension>
> </context>
> </include>
>
> I tried to change the context
> <include>
> <context name="centos">
>  <extension name="from-centos">
>   <condition field="destination_number" expression="^(onops)$">
>    <action application="export" data="execute_on_answer=uuid_record ${uuid}
> start /usr/local/freeswitch/recordings/testrecord.fsv"/>
>    <action application="bridge"
> data="{ignore_early_media=true}user/1004 at 172.16.0.2"/>
>    <action application="hangup"/>
>   </condition>
>  </extension>
> </context>
> </include>
>
> It also did not help. In the end I try to enter these commands from the
> console
> uuid_setvar <uuid> enable_file_write_buffering false (its says ok)
> and
> uuid_record <uuid> start /usr/local/freeswitch/recordings/testrecord.fsv
>
> But i have a error
>
> 2015-04-30 12:27:16.635463 [ERR] mod_fsv.c:702 You are asking to write 342
> bytes of data which is not supported. Please set
> enable_file_write_buffering=false to use .fsv format
> 5e9ce2b0-ef23-11e4-b181-d1f037dabce9 2015-04-30 12:27:16.635463 [ERR]
> switch_ivr_async.c:1155 Error writing
> /usr/local/freeswitch/recordings/testrecord.fsv
>
> Anyone can help me, What I'm doing wrong?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Sharath.Kumar at meZocliq.com  Tue May  5 01:35:09 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Mon, 4 May 2015 21:35:09 +0000
Subject: [Freeswitch-users] DTMF relay with INFO across bridge
In-Reply-To: <CY1PR0501MB10986151A3C7FF38ED9A806384D20@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CY1PR0501MB10986151A3C7FF38ED9A806384D20@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <1430775308568.9063@meZocliq.com>

I believe I found a small bug.

In mod_sofia.c


8589       if (dtmf.digit) {
8590         if (tech_pvt->mparams.dtmf_type == DTMF_INFO ||

I believe it should instead be checking what is set in the profile. if (tech_pvt->profile->dtmf_type? == DTMF_INFO?).  Otherwise, in order for dtmf INFO to be processed by the channel, we need to set liberal-dtmf=true.

Of course my original problem of sending INFO outbound is still not resolved.
Please confirm.

thank you,
?Sharath


From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Sharath Kumar <Sharath.Kumar at meZocliq.com>

Sent: Monday, May 4, 2015 2:07 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] DTMF relay with INFO across bridge

Hi,

I looked around in the mailing list and don't see any conclusive solution to this problem.

I have a setup like this below.

Webrtc client -----FS----sonus gateway.

The Webrtc client is sending INFO as DTMF. The FS currently bridges the call to the gateway and in the outbound leg uses rfc2833.

Problem: Some calls suffer from missing DTMF and repeated duplicate DTMF.

Solution: Use INFO all through to sonus gateway ?

How do I achieve this ?
In the internal profile if I change the "dtmf-type" to "info". Nothing really happens, it still sends rfc2833 dtmf.    I believe it is because of "liberal-dtmf" is "true".[accept any but always offer rfc2833] But if I change "liberal-dtmf" to "false". I get "channel is not configured to use info dtmf".

Is this a bug ? Is there any way around it ?

Any input much appreciated.
Thank you,
Sharath
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From brian at freeswitch.org  Tue May  5 01:39:57 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 May 2015 16:39:57 -0500
Subject: [Freeswitch-users] DTMF relay with INFO across bridge
In-Reply-To: <CY1PR0501MB10986151A3C7FF38ED9A806384D20@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CY1PR0501MB10986151A3C7FF38ED9A806384D20@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <CAEJMVkAbyCW92jvi+C89S7mkRMZ7D5TVQD+ojZRw9fzY6ByA8A@mail.gmail.com>

https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-dtmf_type

I suspect you only need to set the dtmf_type inside the {} on the bridge
line if you wish to send INFO, You do know we have a bunch of RTP bug flags
that can be tweaked to make Sonus work correctly.

https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues



On Mon, May 4, 2015 at 1:07 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
wrote:

>  Hi,
>
>
>
> I looked around in the mailing list and don?t see any conclusive solution
> to this problem.
>
>
>
> I have a setup like this below.
>
>
>
> Webrtc client -----FS----sonus gateway.
>
>
>
> The Webrtc client is sending INFO as DTMF. The FS currently bridges the
> call to the gateway and in the outbound leg uses rfc2833.
>
>
>
> Problem: Some calls suffer from missing DTMF and repeated duplicate DTMF.
>
>
>
> Solution: Use INFO all through to sonus gateway ?
>
>
>
> How do I achieve this ?
>
> In the internal profile if I change the ?dtmf-type? to ?info?. Nothing
> really happens, it still sends rfc2833 dtmf.    I believe it is because of
> ?liberal-dtmf? is ?true?.[accept any but always offer rfc2833] But if I
> change ?liberal-dtmf? to ?false?. I get ?channel is not configured to use
> info dtmf?.
>
>
>
> Is this a bug ? Is there any way around it ?
>
>
>
> Any input much appreciated.
>
> Thank you,
>
> Sharath
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From mike at jerris.com  Tue May  5 01:48:32 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 May 2015 17:48:32 -0400
Subject: [Freeswitch-users] Help with FSV record.
In-Reply-To: <CACOYK=jaQKkZEm5mFbm-11kh68iExNQU1UkQLhDFg=3F3FYZ2A@mail.gmail.com>
References: <1430394389391-7596159.post@n2.nabble.com>
	<8D8CF81D-30E4-4AF3-BDD1-358F9492342D@jerris.com>
	<CACOYK=jaQKkZEm5mFbm-11kh68iExNQU1UkQLhDFg=3F3FYZ2A@mail.gmail.com>
Message-ID: <43BD9579-D137-42F8-9EDB-0849C8B29CF3@jerris.com>

It exists in the fs-video2 branch

echo_decode_video and echo_decode_audio channel vars.

> On May 4, 2015, at 5:15 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> 
> is there a plan to re-implement "echo" application to work on waveform
> level (or picture frame level) instead of copying the RTP packets?  It
> would be then handy for audio and video testing.
> 
> At the moment echo is not compatible with SILK and Opus codecs because
> they don't tolerate simple RTP duplication.
> 
> 
> On Thu, Apr 30, 2015 at 2:34 PM, Michael Jerris <mike at jerris.com> wrote:
>> Recording video was barely a feature in 1.4.  Full support for this and
>> everything you are trying to do is coming in 1.6.  Stay tuned for beta
>> announcements on that coming soon.  You can try out the feature branch if
>> you like, more info is available at:
>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video .
>> We are still working on docs so please be patient as we gather all of that
>> information.
>> 
>> 
>> On Apr 30, 2015, at 7:46 AM, dex_Cz <yadenis at seznam.cz> wrote:
>> 
>> Hi guys!
>> I have a question. Somebody tried to record in the FSV file when i bridge
>> call?
>> I hame a simple context from example. But it is not working.
>> <include>
>> <context name="centos">
>> <extension name="from-centos">
>> <condition field="destination_number" expression="^(onops)$">
>> <action application="record_fsv"
>> data="/usr/local/freeswitch/recordings/testrecord.fsv"/>
>> <action application="sleep" data="100"/>
>> <action application="bridge" data="user/1004 at 172.16.0.2"/>
>> <action application="hangup"/>
>> </condition>
>> </extension>
>> </context>
>> </include>
>> 
>> I tried to change the context
>> <include>
>> <context name="centos">
>> <extension name="from-centos">
>>  <condition field="destination_number" expression="^(onops)$">
>>   <action application="export" data="execute_on_answer=uuid_record ${uuid}
>> start /usr/local/freeswitch/recordings/testrecord.fsv"/>
>>   <action application="bridge"
>> data="{ignore_early_media=true}user/1004 at 172.16.0.2"/>
>>   <action application="hangup"/>
>>  </condition>
>> </extension>
>> </context>
>> </include>
>> 
>> It also did not help. In the end I try to enter these commands from the
>> console
>> uuid_setvar <uuid> enable_file_write_buffering false (its says ok)
>> and
>> uuid_record <uuid> start /usr/local/freeswitch/recordings/testrecord.fsv
>> 
>> But i have a error
>> 
>> 2015-04-30 12:27:16.635463 [ERR] mod_fsv.c:702 You are asking to write 342
>> bytes of data which is not supported. Please set
>> enable_file_write_buffering=false to use .fsv format
>> 5e9ce2b0-ef23-11e4-b181-d1f037dabce9 2015-04-30 12:27:16.635463 [ERR]
>> switch_ivr_async.c:1155 Error writing
>> /usr/local/freeswitch/recordings/testrecord.fsv
>> 
>> Anyone can help me, What I'm doing wrong?



From ssinyagin at gmail.com  Tue May  5 01:55:56 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 4 May 2015 23:55:56 +0200
Subject: [Freeswitch-users] Help with FSV record.
In-Reply-To: <43BD9579-D137-42F8-9EDB-0849C8B29CF3@jerris.com>
References: <1430394389391-7596159.post@n2.nabble.com>
	<8D8CF81D-30E4-4AF3-BDD1-358F9492342D@jerris.com>
	<CACOYK=jaQKkZEm5mFbm-11kh68iExNQU1UkQLhDFg=3F3FYZ2A@mail.gmail.com>
	<43BD9579-D137-42F8-9EDB-0849C8B29CF3@jerris.com>
Message-ID: <CACOYK=hobD5v=Cky+uXA4XGeLxQ0FGRikd6u-hD2kOLC7VkS7A@mail.gmail.com>

Cool, thanks, will give it a try.
On May 4, 2015 11:49 PM, "Michael Jerris" <mike at jerris.com> wrote:

> It exists in the fs-video2 branch
>
> echo_decode_video and echo_decode_audio channel vars.
>
> > On May 4, 2015, at 5:15 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
> >
> > is there a plan to re-implement "echo" application to work on waveform
> > level (or picture frame level) instead of copying the RTP packets?  It
> > would be then handy for audio and video testing.
> >
> > At the moment echo is not compatible with SILK and Opus codecs because
> > they don't tolerate simple RTP duplication.
> >
> >
> > On Thu, Apr 30, 2015 at 2:34 PM, Michael Jerris <mike at jerris.com> wrote:
> >> Recording video was barely a feature in 1.4.  Full support for this and
> >> everything you are trying to do is coming in 1.6.  Stay tuned for beta
> >> announcements on that coming soon.  You can try out the feature branch
> if
> >> you like, more info is available at:
> >>
> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
> .
> >> We are still working on docs so please be patient as we gather all of
> that
> >> information.
> >>
> >>
> >> On Apr 30, 2015, at 7:46 AM, dex_Cz <yadenis at seznam.cz> wrote:
> >>
> >> Hi guys!
> >> I have a question. Somebody tried to record in the FSV file when i
> bridge
> >> call?
> >> I hame a simple context from example. But it is not working.
> >> <include>
> >> <context name="centos">
> >> <extension name="from-centos">
> >> <condition field="destination_number" expression="^(onops)$">
> >> <action application="record_fsv"
> >> data="/usr/local/freeswitch/recordings/testrecord.fsv"/>
> >> <action application="sleep" data="100"/>
> >> <action application="bridge" data="user/1004 at 172.16.0.2"/>
> >> <action application="hangup"/>
> >> </condition>
> >> </extension>
> >> </context>
> >> </include>
> >>
> >> I tried to change the context
> >> <include>
> >> <context name="centos">
> >> <extension name="from-centos">
> >>  <condition field="destination_number" expression="^(onops)$">
> >>   <action application="export" data="execute_on_answer=uuid_record
> ${uuid}
> >> start /usr/local/freeswitch/recordings/testrecord.fsv"/>
> >>   <action application="bridge"
> >> data="{ignore_early_media=true}user/1004 at 172.16.0.2"/>
> >>   <action application="hangup"/>
> >>  </condition>
> >> </extension>
> >> </context>
> >> </include>
> >>
> >> It also did not help. In the end I try to enter these commands from the
> >> console
> >> uuid_setvar <uuid> enable_file_write_buffering false (its says ok)
> >> and
> >> uuid_record <uuid> start /usr/local/freeswitch/recordings/testrecord.fsv
> >>
> >> But i have a error
> >>
> >> 2015-04-30 12:27:16.635463 [ERR] mod_fsv.c:702 You are asking to write
> 342
> >> bytes of data which is not supported. Please set
> >> enable_file_write_buffering=false to use .fsv format
> >> 5e9ce2b0-ef23-11e4-b181-d1f037dabce9 2015-04-30 12:27:16.635463 [ERR]
> >> switch_ivr_async.c:1155 Error writing
> >> /usr/local/freeswitch/recordings/testrecord.fsv
> >>
> >> Anyone can help me, What I'm doing wrong?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May  5 02:39:25 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 04 May 2015 22:39:25 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) April
	25th-May 1st
Message-ID: <5547f51d84299_65b345533065484@resque-worker.6.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1Ia2Ngj
FreeSWITCH Week in Review (Master Branch) April 25th-May 1st
Hello, again. This passed week in the FreeSWITCH master branch we had 14 commits.

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

The following bugs were squashed:

FS-7472 [mod_sofia] Fix for a bug where the rtp-digit-delay profile param was being ignored

FS-7488 [mod_managed] Fixed a build error

FS-7490 [mod_rayo] Fixed a bug with the format of mod_rayo generated regex not working with newer libpcre

FS-7491 [mod_graylog2] Send timestamp with millisecond precision instead of microsecond as required by GELF

FS-7466 Fixed a bug causing audio issues by repeated log lines printing when rtp_manual_rtp_bugs is set to ALWAYS_AUTO_ADJUST

FS-7496 [mod_http_cache] Fixed an issue with the URL args being included in the cache file name and causing problems opening the files later

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From krice at freeswitch.org  Tue May  5 02:54:40 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 04 May 2015 22:54:40 +0000
Subject: [Freeswitch-users] FreeSWITCH and the VUC
Message-ID: <5547f8b0b947b_7549c51318868e4@resque-worker-high.0.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1KHXdS4
FreeSWITCH and the VUC
Go check out all the new, awesome features going into FreeSWITCH! The core developers made an appearance on the VUC call this past Friday to talk about and demonstrate some of the fantastic, new features in FreeSWITCH.

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From kamil.nigmatullin at gmail.com  Tue May  5 08:22:45 2015
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Tue, 5 May 2015 10:22:45 +0600
Subject: [Freeswitch-users] 407 Proxy authentication
In-Reply-To: <CAG6TCM8x5m2bQAY71vA9FdW0jacnpEEKo2DupGVJxJyPFnVS8Q@mail.gmail.com>
References: <CAG6TCM8hOWEG89DC+96ehBWGpwU6nXiT2q8EfVXa1XK5gZc-fA@mail.gmail.com>
	<AEDB707B-FA8D-412A-9E99-B5B8DA7D66CD@jerris.com>
	<CAG6TCM9O20r7u1G5Mq-_7m+Vgsz7BBEPSJ+7r5TvWRSPNQ8ntg@mail.gmail.com>
	<CAM33Ch4HUjjHK2DmXDwu3tVn7YcrQ2yoJRFcN6feXi2w59NWUQ@mail.gmail.com>
	<CAG6TCM8oM_eRcDP1hMA9Z9_65bCkC+sB0sVbCKqgt-8ac4ZHdQ@mail.gmail.com>
	<CACOYK=iJSbHYS8WUZGeM5J=0_NTsX9i+k_Z5PoVdyovdY8F9qA@mail.gmail.com>
	<CA+XtiecnBq9aWDxyd-Oi2tMDE70HFNr1zuVP650Df2-bX5gfGQ@mail.gmail.com>
	<CAG6TCM8x5m2bQAY71vA9FdW0jacnpEEKo2DupGVJxJyPFnVS8Q@mail.gmail.com>
Message-ID: <CA+XtiefsRcYdmtYO1zOn6YerjO+RH3La6=aROzacDxJb4egPJw@mail.gmail.com>

Port 5060 will require digest authentication. That's absolutely normal. If
you need to call like to gateway without gijest auth use 5080 port with
gateways




2015-05-04 19:18 GMT+06:00 bhavik patel <bhavikpatel14388 at gmail.com>:

> yes that's default configuration and port is 5060
>
> On Mon, May 4, 2015 at 6:21 PM, Kamil Nigmatullin <
> kamil.nigmatullin at gmail.com> wrote:
>
>> Do you use port 5060 or 5080 (I think you use standard config)?
>>
>> 2015-04-30 14:23 GMT+06:00 Stanislav Sinyagin <ssinyagin at gmail.com>:
>>
>>> I think it's the proper time you would start reading the book :-)
>>> https://www.packtpub.com/networking-and-servers/freeswitch-12
>>>
>>> or pay someone for training
>>>
>>>
>>> On Thu, Apr 30, 2015 at 8:48 AM, bhavik patel
>>> <bhavikpatel14388 at gmail.com> wrote:
>>> > Hi,
>>> >
>>> > Thanks for reply,I tried but still same.
>>> > Any other configuration missing.
>>> >
>>> > On Wed, Apr 29, 2015 at 10:59 PM, Varghese Paul <
>>> varghesepaul87 at gmail.com>
>>> > wrote:
>>> >>
>>> >> Hi ,
>>> >>
>>> >> You can set this variable in your inbound sip profile.
>>> >>
>>> >> <param name="auth-calls" value="true"/>
>>> >>
>>> >>
>>> >> Regards
>>> >>
>>> >> Varghese Paul
>>> >>
>>> >>
>>> >> On Wed, Apr 29, 2015 at 12:50 AM, bhavik patel
>>> >> <bhavikpatel14388 at gmail.com> wrote:
>>> >>>
>>> >>> If you know any specific sip profile parameter then please suggest
>>> me.
>>> >>>
>>> >>>
>>> >>> On Tue, Apr 28, 2015 at 7:23 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>> >>>>
>>> >>>> Check the profile settings for auth and acl.  The default configs
>>> >>>> include profiles for anonymous and auth.
>>> >>>>
>>> >>>> > On Apr 28, 2015, at 9:26 AM, bhavik patel <
>>> bhavikpatel14388 at gmail.com>
>>> >>>> > wrote:
>>> >>>> >
>>> >>>> > Hi,
>>> >>>> >
>>> >>>> > We need 407 Proxy authentication require sip dialog while calling
>>> >>>> > between sip to sip extension. Currently when we are calling, we
>>> never got
>>> >>>> > that sip dialog.
>>> >>>> >
>>> >>>> > I am sure there must be some configuration to achieve this.
>>> >>>> >
>>> >>>> > Can anyone please help us to make it working?
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> _________________________________________________________________________
>>> >>>> Professional FreeSWITCH Consulting Services:
>>> >>>> consulting at freeswitch.org
>>> >>>> http://www.freeswitchsolutions.com
>>> >>>>
>>> >>>> Official FreeSWITCH Sites
>>> >>>> http://www.freeswitch.org
>>> >>>> http://confluence.freeswitch.org
>>> >>>> http://www.cluecon.com
>>> >>>>
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>
>>> >>>
>>> >>>
>>> >>>
>>> >>> --
>>> >>> Thanks,
>>> >>> Bhavik Patel
>>> >>>
>>> >>>
>>> >>>
>>> _________________________________________________________________________
>>> >>> Professional FreeSWITCH Consulting Services:
>>> >>> consulting at freeswitch.org
>>> >>> http://www.freeswitchsolutions.com
>>> >>>
>>> >>> Official FreeSWITCH Sites
>>> >>> http://www.freeswitch.org
>>> >>> http://confluence.freeswitch.org
>>> >>> http://www.cluecon.com
>>> >>>
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://confluence.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> >
>>> > --
>>> > Thanks,
>>> > Bhavik Patel
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Kamil Nigmatullin
>> Tel: 77272323748
>> mob: 7 (707) 2517003
>> Skype: kamil.nigmatullin
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Thanks,
> Bhavik Patel
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kamil Nigmatullin
Tel: 77272323748
mob: 7 (707) 2517003
Skype: kamil.nigmatullin
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From kamil.nigmatullin at gmail.com  Tue May  5 08:47:32 2015
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Tue, 5 May 2015 10:47:32 +0600
Subject: [Freeswitch-users] 407 Proxy authentication
In-Reply-To: <CA+XtiefsRcYdmtYO1zOn6YerjO+RH3La6=aROzacDxJb4egPJw@mail.gmail.com>
References: <CAG6TCM8hOWEG89DC+96ehBWGpwU6nXiT2q8EfVXa1XK5gZc-fA@mail.gmail.com>
	<AEDB707B-FA8D-412A-9E99-B5B8DA7D66CD@jerris.com>
	<CAG6TCM9O20r7u1G5Mq-_7m+Vgsz7BBEPSJ+7r5TvWRSPNQ8ntg@mail.gmail.com>
	<CAM33Ch4HUjjHK2DmXDwu3tVn7YcrQ2yoJRFcN6feXi2w59NWUQ@mail.gmail.com>
	<CAG6TCM8oM_eRcDP1hMA9Z9_65bCkC+sB0sVbCKqgt-8ac4ZHdQ@mail.gmail.com>
	<CACOYK=iJSbHYS8WUZGeM5J=0_NTsX9i+k_Z5PoVdyovdY8F9qA@mail.gmail.com>
	<CA+XtiecnBq9aWDxyd-Oi2tMDE70HFNr1zuVP650Df2-bX5gfGQ@mail.gmail.com>
	<CAG6TCM8x5m2bQAY71vA9FdW0jacnpEEKo2DupGVJxJyPFnVS8Q@mail.gmail.com>
	<CA+XtiefsRcYdmtYO1zOn6YerjO+RH3La6=aROzacDxJb4egPJw@mail.gmail.com>
Message-ID: <CA+XtiedeG9htDWVhx-x=qu5qHxGhJfOUC9ac_unv7QDgm_LHsg@mail.gmail.com>

Sorry, wrong answer. Could you show sip trace?

2015-05-05 10:22 GMT+06:00 Kamil Nigmatullin <kamil.nigmatullin at gmail.com>:

> Port 5060 will require digest authentication. That's absolutely normal. If
> you need to call like to gateway without gijest auth use 5080 port with
> gateways
>
>
>
>
> 2015-05-04 19:18 GMT+06:00 bhavik patel <bhavikpatel14388 at gmail.com>:
>
>> yes that's default configuration and port is 5060
>>
>> On Mon, May 4, 2015 at 6:21 PM, Kamil Nigmatullin <
>> kamil.nigmatullin at gmail.com> wrote:
>>
>>> Do you use port 5060 or 5080 (I think you use standard config)?
>>>
>>> 2015-04-30 14:23 GMT+06:00 Stanislav Sinyagin <ssinyagin at gmail.com>:
>>>
>>>> I think it's the proper time you would start reading the book :-)
>>>> https://www.packtpub.com/networking-and-servers/freeswitch-12
>>>>
>>>> or pay someone for training
>>>>
>>>>
>>>> On Thu, Apr 30, 2015 at 8:48 AM, bhavik patel
>>>> <bhavikpatel14388 at gmail.com> wrote:
>>>> > Hi,
>>>> >
>>>> > Thanks for reply,I tried but still same.
>>>> > Any other configuration missing.
>>>> >
>>>> > On Wed, Apr 29, 2015 at 10:59 PM, Varghese Paul <
>>>> varghesepaul87 at gmail.com>
>>>> > wrote:
>>>> >>
>>>> >> Hi ,
>>>> >>
>>>> >> You can set this variable in your inbound sip profile.
>>>> >>
>>>> >> <param name="auth-calls" value="true"/>
>>>> >>
>>>> >>
>>>> >> Regards
>>>> >>
>>>> >> Varghese Paul
>>>> >>
>>>> >>
>>>> >> On Wed, Apr 29, 2015 at 12:50 AM, bhavik patel
>>>> >> <bhavikpatel14388 at gmail.com> wrote:
>>>> >>>
>>>> >>> If you know any specific sip profile parameter then please suggest
>>>> me.
>>>> >>>
>>>> >>>
>>>> >>> On Tue, Apr 28, 2015 at 7:23 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>> >>>>
>>>> >>>> Check the profile settings for auth and acl.  The default configs
>>>> >>>> include profiles for anonymous and auth.
>>>> >>>>
>>>> >>>> > On Apr 28, 2015, at 9:26 AM, bhavik patel <
>>>> bhavikpatel14388 at gmail.com>
>>>> >>>> > wrote:
>>>> >>>> >
>>>> >>>> > Hi,
>>>> >>>> >
>>>> >>>> > We need 407 Proxy authentication require sip dialog while calling
>>>> >>>> > between sip to sip extension. Currently when we are calling, we
>>>> never got
>>>> >>>> > that sip dialog.
>>>> >>>> >
>>>> >>>> > I am sure there must be some configuration to achieve this.
>>>> >>>> >
>>>> >>>> > Can anyone please help us to make it working?
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> _________________________________________________________________________
>>>> >>>> Professional FreeSWITCH Consulting Services:
>>>> >>>> consulting at freeswitch.org
>>>> >>>> http://www.freeswitchsolutions.com
>>>> >>>>
>>>> >>>> Official FreeSWITCH Sites
>>>> >>>> http://www.freeswitch.org
>>>> >>>> http://confluence.freeswitch.org
>>>> >>>> http://www.cluecon.com
>>>> >>>>
>>>> >>>> FreeSWITCH-users mailing list
>>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>>> http://www.freeswitch.org
>>>> >>>
>>>> >>>
>>>> >>>
>>>> >>>
>>>> >>> --
>>>> >>> Thanks,
>>>> >>> Bhavik Patel
>>>> >>>
>>>> >>>
>>>> >>>
>>>> _________________________________________________________________________
>>>> >>> Professional FreeSWITCH Consulting Services:
>>>> >>> consulting at freeswitch.org
>>>> >>> http://www.freeswitchsolutions.com
>>>> >>>
>>>> >>> Official FreeSWITCH Sites
>>>> >>> http://www.freeswitch.org
>>>> >>> http://confluence.freeswitch.org
>>>> >>> http://www.cluecon.com
>>>> >>>
>>>> >>> FreeSWITCH-users mailing list
>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>> http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> _________________________________________________________________________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> consulting at freeswitch.org
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://confluence.freeswitch.org
>>>> >> http://www.cluecon.com
>>>> >>
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> >
>>>> >
>>>> > --
>>>> > Thanks,
>>>> > Bhavik Patel
>>>> >
>>>> >
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://confluence.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Kamil Nigmatullin
>>> Tel: 77272323748
>>> mob: 7 (707) 2517003
>>> Skype: kamil.nigmatullin
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Thanks,
>> Bhavik Patel
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Kamil Nigmatullin
> Tel: 77272323748
> mob: 7 (707) 2517003
> Skype: kamil.nigmatullin
>



-- 
Kamil Nigmatullin
Tel: 77272323748
mob: 7 (707) 2517003
Skype: kamil.nigmatullin
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From danny.gershman at gmail.com  Tue May  5 10:04:18 2015
From: danny.gershman at gmail.com (radius314)
Date: Mon, 4 May 2015 23:04:18 -0700 (MST)
Subject: [Freeswitch-users] Receive chat
In-Reply-To: <CAA2gtHBghZ5so8Shoaxy5cu161F5y9OuBT_PZmHhUW2RVoYk-w@mail.gmail.com>
References: <CAA2gtHBghZ5so8Shoaxy5cu161F5y9OuBT_PZmHhUW2RVoYk-w@mail.gmail.com>
Message-ID: <1430805858583-7596160.post@n2.nabble.com>

You could you the event socket to receive the messages.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Receive-chat-tp7489501p7596160.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From vishal.sharma at knowlarity.com  Tue May  5 14:42:09 2015
From: vishal.sharma at knowlarity.com (Vishal Sharma)
Date: Tue, 5 May 2015 16:12:09 +0530
Subject: [Freeswitch-users] legB not Disconnected
Message-ID: <CAEaAqfga43Br2NwZB2cQVYQV1wF+HjnnSNB1i6Ep1eCwQJ21sg@mail.gmail.com>

Hi All,
I am using ESL to control the call flow.
My initial call flow is as following

1.      INVITE---> FS
2.      180<--------|
3.      200<--------|
4.      ------------>ACK
5.     *************|-------------->INVITE
6.     *************|<---------------180
7.     BYE------->|
8.     Retransmissions of BYE
9.     *************|<----------------200 OK
10.   200(bye)<---|



Now Leg B remains hanging... Shouldn't FS have killed it since legA is dead.

On Application I get Channel Hangup on 200 bye. leg B call is generated
using bridge command with hangup_on_answer = TRUE.


Regards,
Vishal Sharma
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From Sharath.Kumar at meZocliq.com  Tue May  5 18:37:37 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Tue, 5 May 2015 14:37:37 +0000
Subject: [Freeswitch-users] DTMF relay with INFO across bridge
In-Reply-To: <CAEJMVkAbyCW92jvi+C89S7mkRMZ7D5TVQD+ojZRw9fzY6ByA8A@mail.gmail.com>
References: <CY1PR0501MB10986151A3C7FF38ED9A806384D20@CY1PR0501MB1098.namprd05.prod.outlook.com>,
	<CAEJMVkAbyCW92jvi+C89S7mkRMZ7D5TVQD+ojZRw9fzY6ByA8A@mail.gmail.com>
Message-ID: <1430836657047.2067@meZocliq.com>

Brian,


Thanks for responding.

1] I had already set it in the channel variable for the bridge line. Freeswitch *always* seems to convert between INFO --> rfc2833. In the internal and external profiles. I have removed "liberal-dtmf" and also set "dtmf-type" to "info". And the bridge line contains "dtmf_type" as info. Am I missing something here?


logs - 1fc06e8a-f334-11e4-ade6-413a2da7e1fc Dialplan: sofia/internal/nXXX.sXXX at voip.XXX.com Action bridge({sip_cid_type=pid,dtmf_type=info,origination_privacy=screen,ignore_early_media=true}sofia/gateway/tata/XXXXXXX)?


Tested on latest master and on 1.5.13b. Same thing.


2] The bug from what I tested.

- if "liberal-dtmf" is set to false and even though "dtmf-type" is set to "info" in the profile, then I see "channel is not configured to use info dtmf"  .

>From the latest master. In mod_sofia.c

8589       if (dtmf.digit) {
8590         if (tech_pvt->mparams.dtmf_type == DTMF_INFO ||

I believe it should instead be checking what is set in the profile. if (tech_pvt->profile->dtmf_type? == DTMF_INFO?).

3] Sonus RTP fixes - Yes I have applied them. It probably works most of the time. But I occasionally get complains from my customers regarding - missing dtmf and repeated dtmfs. I am hoping moving to INFO will fix the problem for good ? Any thoughts ?


thank you very much!

Sharath

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Brian West <brian at freeswitch.org>
Sent: Monday, May 4, 2015 5:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF relay with INFO across bridge

https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-dtmf_type

I suspect you only need to set the dtmf_type inside the {} on the bridge line if you wish to send INFO, You do know we have a bunch of RTP bug flags that can be tweaked to make Sonus work correctly.

https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues



On Mon, May 4, 2015 at 1:07 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
Hi,

I looked around in the mailing list and don't see any conclusive solution to this problem.

I have a setup like this below.

Webrtc client -----FS----sonus gateway.

The Webrtc client is sending INFO as DTMF. The FS currently bridges the call to the gateway and in the outbound leg uses rfc2833.

Problem: Some calls suffer from missing DTMF and repeated duplicate DTMF.

Solution: Use INFO all through to sonus gateway ?

How do I achieve this ?
In the internal profile if I change the "dtmf-type" to "info". Nothing really happens, it still sends rfc2833 dtmf.    I believe it is because of "liberal-dtmf" is "true".[accept any but always offer rfc2833] But if I change "liberal-dtmf" to "false". I get "channel is not configured to use info dtmf".

Is this a bug ? Is there any way around it ?

Any input much appreciated.
Thank you,
Sharath

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

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

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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch<https://www.reddit.com/r/freeswitch>

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
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From brian at freeswitch.org  Tue May  5 18:42:18 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 May 2015 09:42:18 -0500
Subject: [Freeswitch-users] DTMF relay with INFO across bridge
In-Reply-To: <1430836657047.2067@meZocliq.com>
References: <CY1PR0501MB10986151A3C7FF38ED9A806384D20@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAEJMVkAbyCW92jvi+C89S7mkRMZ7D5TVQD+ojZRw9fzY6ByA8A@mail.gmail.com>
	<1430836657047.2067@meZocliq.com>
Message-ID: <CAEJMVkAKa887iY7qyqs4FMYaQk5zCCODkd2kUY9abCnae+8WMw@mail.gmail.com>

If you think you've found a bug please file a JIRA, We can review it.  As
for INFO dtmf, you do realize there is no way to guarantee order of
digits.  So you could have them arrive out of order and there is nothing
that can be done.  Its best to get your upstream to fix their broken
implementations.  What RTP bug options did you try?

On Tue, May 5, 2015 at 9:37 AM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
wrote:

>  Brian,
>
>
>  Thanks for responding.
>
> 1] I had already set it in the channel variable for the bridge line.
> Freeswitch *always* seems to convert between INFO --> rfc2833. In the
> internal and external profiles. I have removed "liberal-dtmf" and also set
> "dtmf-type" to "info". And the bridge line contains "dtmf_type" as info. Am
> I missing something here?
>
>
>  logs - 1fc06e8a-f334-11e4-ade6-413a2da7e1fc Dialplan: sofia/internal/
> nXXX.sXXX at voip.XXX.com Action bridge({sip_cid_type=pid,*dtmf_type=info*
> ,origination_privacy=screen,ignore_early_media=true}sofia/gateway/tata/XXXXXXX)
> ?
>
>
>  Tested on latest master and on 1.5.13b. Same thing.
>
>
>  2] The bug from what I tested.
>
> - if "liberal-dtmf" is set to false and even though "dtmf-type" is set to
> "info" in the profile, then I see "channel is not configured to use info
> dtmf?  .
>  >From the latest master. In mod_sofia.c
>
>  8589       if (dtmf.digit) {
>  8590         if (tech_pvt->mparams.dtmf_type == DTMF_INFO ||
>
>  I believe it should instead be checking what is set in the profile. if
> (tech_pvt->profile->dtmf_type? == DTMF_INFO?).
>
>  3] Sonus RTP fixes - Yes I have applied them. It probably works most of
> the time. But I occasionally get complains from my customers regarding -
> missing dtmf and repeated dtmfs. I am hoping moving to INFO will fix the
> problem for good ? Any thoughts ?
>
>
> thank you very much!
>
> Sharath
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org <
> freeswitch-users-bounces at lists.freeswitch.org> on behalf of Brian West <
> brian at freeswitch.org>
> *Sent:* Monday, May 4, 2015 5:39 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] DTMF relay with INFO across bridge
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-dtmf_type
>
>  I suspect you only need to set the dtmf_type inside the {} on the bridge
> line if you wish to send INFO, You do know we have a bunch of RTP bug flags
> that can be tweaked to make Sonus work correctly.
>
>  https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues
>
>
>
> On Mon, May 4, 2015 at 1:07 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
> wrote:
>
>>  Hi,
>>
>>
>>
>> I looked around in the mailing list and don?t see any conclusive solution
>> to this problem.
>>
>>
>>
>> I have a setup like this below.
>>
>>
>>
>> Webrtc client -----FS----sonus gateway.
>>
>>
>>
>> The Webrtc client is sending INFO as DTMF. The FS currently bridges the
>> call to the gateway and in the outbound leg uses rfc2833.
>>
>>
>>
>> Problem: Some calls suffer from missing DTMF and repeated duplicate DTMF.
>>
>>
>>
>> Solution: Use INFO all through to sonus gateway ?
>>
>>
>>
>> How do I achieve this ?
>>
>> In the internal profile if I change the ?dtmf-type? to ?info?. Nothing
>> really happens, it still sends rfc2833 dtmf.    I believe it is because of
>> ?liberal-dtmf? is ?true?.[accept any but always offer rfc2833] But if I
>> change ?liberal-dtmf? to ?false?. I get ?channel is not configured to use
>> info dtmf?.
>>
>>
>>
>> Is this a bug ? Is there any way around it ?
>>
>>
>>
>> Any input much appreciated.
>>
>> Thank you,
>>
>> Sharath
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>



-- 

*Brian West*
brian at freeswitch.org


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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
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From mike at jerris.com  Tue May  5 18:42:28 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 5 May 2015 10:42:28 -0400
Subject: [Freeswitch-users] legB not Disconnected
In-Reply-To: <CAEaAqfga43Br2NwZB2cQVYQV1wF+HjnnSNB1i6Ep1eCwQJ21sg@mail.gmail.com>
References: <CAEaAqfga43Br2NwZB2cQVYQV1wF+HjnnSNB1i6Ep1eCwQJ21sg@mail.gmail.com>
Message-ID: <2863E8B0-D30A-49DC-89E8-63887BAC2C37@jerris.com>

hangup_after_bridge=true?  I'm not sure there is a hangup_after_answer or any reason to use one.

> On May 5, 2015, at 6:42 AM, Vishal Sharma <vishal.sharma at knowlarity.com> wrote:
> 
> Hi All,
> I am using ESL to control the call flow.
> My initial call flow is as following
> 
> 1.      INVITE---> FS
> 2.      180<--------|
> 3.      200<--------|
> 4.      ------------>ACK
> 5.     *************|-------------->INVITE
> 6.     *************|<---------------180
> 7.     BYE------->|
> 8.     Retransmissions of BYE
> 9.     *************|<----------------200 OK
> 10.   200(bye)<---|
> 
> 
> 
> Now Leg B remains hanging... Shouldn't FS have killed it since legA is dead.
> 
> On Application I get Channel Hangup on 200 bye. leg B call is generated using bridge command with hangup_on_answer = TRUE.
> 
> 
> Regards,
> Vishal Sharma



From sertys at gmail.com  Tue May  5 18:57:52 2015
From: sertys at gmail.com (Daniel Ivanov)
Date: Tue, 5 May 2015 17:57:52 +0300
Subject: [Freeswitch-users] Simulate incoming SIP Message
In-Reply-To: <CAHtxdDfX7Kv8uVs-h6xqUKK4WqVtBB--LpJ4isBQok77G3Ckrw@mail.gmail.com>
References: <CACyT-3=AVrm3NkFVbCLVVbZPB_JE9jt87NA0gCxeAbdgFpL5Uw@mail.gmail.com>
	<CAHtxdDfX7Kv8uVs-h6xqUKK4WqVtBB--LpJ4isBQok77G3Ckrw@mail.gmail.com>
Message-ID: <CAMeZp60XNtmj7_x7wWFr9rP+ewgfht9y0Fb-JeHmeZEQkAK1Ew@mail.gmail.com>

You got it wrong. "Not sending a message to ourselves" means "not sending a
message to the same extension it came from". Just use the fs_cli chat
command and provide a fake sender to get your tests done.
>From wiki :

chat sip|1000 at 172.20.0.1|1001 at 172.20.0.1|Hello chat from freeswitch



On Mon, May 4, 2015 at 12:48 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Why you not use sipp?
>
> On Mon, May 4, 2015 at 7:48 AM, Kurtis Heimerl <kheimerl at cs.berkeley.edu>
> wrote:
>
>> Hello FreeSWITCH Users,
>>
>> I am trying to simulate an incoming SIP message in order to test my
>> chatplan programmatically. Basically, I want FS to send itself a SIP
>> message, which will then be routed through the chatplan. I've been able to
>> generate the message using ESL and events, however FS refuses to send it
>> noting that it is "Not sending message to ourselves!"
>>
>> That's exactly what I want it to do. I've used this functionality before
>> for calls, as generating a call that connects back to FS (and hooking it to
>> echo) is a great way to test the dialplan. Is there no way to get FS to
>> send itself a message?
>>
>> Thanks!
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vishal.sharma at knowlarity.com  Tue May  5 19:04:37 2015
From: vishal.sharma at knowlarity.com (Vishal Sharma)
Date: Tue, 5 May 2015 20:34:37 +0530
Subject: [Freeswitch-users] legB not Disconnected
In-Reply-To: <2863E8B0-D30A-49DC-89E8-63887BAC2C37@jerris.com>
References: <CAEaAqfga43Br2NwZB2cQVYQV1wF+HjnnSNB1i6Ep1eCwQJ21sg@mail.gmail.com>
	<2863E8B0-D30A-49DC-89E8-63887BAC2C37@jerris.com>
Message-ID: <CAEaAqfgAv1PLW8cx0nuaz6MWOoHd4e3sk72GS5Z+cxsw8RLdZw@mail.gmail.com>

Yes it is hangup_after_bridge... it is used because if legb hangs up ... I
want call to be disconnected. ..
On 5 May 2015 20:13, "Michael Jerris" <mike at jerris.com> wrote:

> hangup_after_bridge=true?  I'm not sure there is a hangup_after_answer or
> any reason to use one.
>
> > On May 5, 2015, at 6:42 AM, Vishal Sharma <vishal.sharma at knowlarity.com>
> wrote:
> >
> > Hi All,
> > I am using ESL to control the call flow.
> > My initial call flow is as following
> >
> > 1.      INVITE---> FS
> > 2.      180<--------|
> > 3.      200<--------|
> > 4.      ------------>ACK
> > 5.     *************|-------------->INVITE
> > 6.     *************|<---------------180
> > 7.     BYE------->|
> > 8.     Retransmissions of BYE
> > 9.     *************|<----------------200 OK
> > 10.   200(bye)<---|
> >
> >
> >
> > Now Leg B remains hanging... Shouldn't FS have killed it since legA is
> dead.
> >
> > On Application I get Channel Hangup on 200 bye. leg B call is generated
> using bridge command with hangup_on_answer = TRUE.
> >
> >
> > Regards,
> > Vishal Sharma
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Sharath.Kumar at meZocliq.com  Tue May  5 19:31:06 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Tue, 5 May 2015 15:31:06 +0000
Subject: [Freeswitch-users] DTMF relay with INFO across bridge
In-Reply-To: <CAEJMVkAKa887iY7qyqs4FMYaQk5zCCODkd2kUY9abCnae+8WMw@mail.gmail.com>
References: <CY1PR0501MB10986151A3C7FF38ED9A806384D20@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAEJMVkAbyCW92jvi+C89S7mkRMZ7D5TVQD+ojZRw9fzY6ByA8A@mail.gmail.com>
	<1430836657047.2067@meZocliq.com>,
	<CAEJMVkAKa887iY7qyqs4FMYaQk5zCCODkd2kUY9abCnae+8WMw@mail.gmail.com>
Message-ID: <1430839866075.22898@meZocliq.com>

Brian,

Okay I will file it on JIRA.

Thanks for the input. I tried the following options, that was suggested on the FS website.


<X-PRE-PROCESS cmd="set" data="send_silence_when_idle=400"/>

<param name="rtp-rewrite-timestamps" value="true"/>


Is there something else I am missing ?


thank you,

Sharath

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Brian West <brian at freeswitch.org>
Sent: Tuesday, May 5, 2015 10:42 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF relay with INFO across bridge

If you think you've found a bug please file a JIRA, We can review it.  As for INFO dtmf, you do realize there is no way to guarantee order of digits.  So you could have them arrive out of order and there is nothing that can be done.  Its best to get your upstream to fix their broken implementations.  What RTP bug options did you try?

On Tue, May 5, 2015 at 9:37 AM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:

Brian,


Thanks for responding.

1] I had already set it in the channel variable for the bridge line. Freeswitch *always* seems to convert between INFO --> rfc2833. In the internal and external profiles. I have removed "liberal-dtmf" and also set "dtmf-type" to "info". And the bridge line contains "dtmf_type" as info. Am I missing something here?


logs - 1fc06e8a-f334-11e4-ade6-413a2da7e1fc Dialplan: sofia/internal/nXXX.sXXX at voip.XXX.com<mailto:nXXX.sXXX at voip.XXX.com> Action bridge({sip_cid_type=pid,dtmf_type=info,origination_privacy=screen,ignore_early_media=true}sofia/gateway/tata/XXXXXXX)?


Tested on latest master and on 1.5.13b. Same thing.


2] The bug from what I tested.

- if "liberal-dtmf" is set to false and even though "dtmf-type" is set to "info" in the profile, then I see "channel is not configured to use info dtmf?  .

>From the latest master. In mod_sofia.c

8589       if (dtmf.digit) {
8590         if (tech_pvt->mparams.dtmf_type == DTMF_INFO ||

I believe it should instead be checking what is set in the profile. if (tech_pvt->profile->dtmf_type? == DTMF_INFO?).

3] Sonus RTP fixes - Yes I have applied them. It probably works most of the time. But I occasionally get complains from my customers regarding - missing dtmf and repeated dtmfs. I am hoping moving to INFO will fix the problem for good ? Any thoughts ?


thank you very much!

Sharath

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>>
Sent: Monday, May 4, 2015 5:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF relay with INFO across bridge

https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-dtmf_type

I suspect you only need to set the dtmf_type inside the {} on the bridge line if you wish to send INFO, You do know we have a bunch of RTP bug flags that can be tweaked to make Sonus work correctly.

https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues



On Mon, May 4, 2015 at 1:07 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
Hi,

I looked around in the mailing list and don?t see any conclusive solution to this problem.

I have a setup like this below.

Webrtc client -----FS----sonus gateway.

The Webrtc client is sending INFO as DTMF. The FS currently bridges the call to the gateway and in the outbound leg uses rfc2833.

Problem: Some calls suffer from missing DTMF and repeated duplicate DTMF.

Solution: Use INFO all through to sonus gateway ?

How do I achieve this ?
In the internal profile if I change the ?dtmf-type? to ?info?. Nothing really happens, it still sends rfc2833 dtmf.    I believe it is because of ?liberal-dtmf? is ?true?.[accept any but always offer rfc2833] But if I change ?liberal-dtmf? to ?false?. I get ?channel is not configured to use info dtmf?.

Is this a bug ? Is there any way around it ?

Any input much appreciated.
Thank you,
Sharath

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

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

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--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch<https://www.reddit.com/r/freeswitch>

T:+19184209001<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

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

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

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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch<https://www.reddit.com/r/freeswitch>

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From s.safarov at gmail.com  Tue May  5 20:19:42 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 5 May 2015 19:19:42 +0300
Subject: [Freeswitch-users] legB not Disconnected
In-Reply-To: <CAEaAqfga43Br2NwZB2cQVYQV1wF+HjnnSNB1i6Ep1eCwQJ21sg@mail.gmail.com>
References: <CAEaAqfga43Br2NwZB2cQVYQV1wF+HjnnSNB1i6Ep1eCwQJ21sg@mail.gmail.com>
Message-ID: <CAHtxdDcz10tnN1whDHnuog7Ua+wD2sg2RBjG+d4aJf+83Sr5wQ@mail.gmail.com>

Similar case https://freeswitch.org/jira/browse/FS-7399


On Tue, May 5, 2015 at 1:42 PM, Vishal Sharma <vishal.sharma at knowlarity.com>
wrote:

> Hi All,
> I am using ESL to control the call flow.
> My initial call flow is as following
>
> 1.      INVITE---> FS
> 2.      180<--------|
> 3.      200<--------|
> 4.      ------------>ACK
> 5.     *************|-------------->INVITE
> 6.     *************|<---------------180
> 7.     BYE------->|
> 8.     Retransmissions of BYE
> 9.     *************|<----------------200 OK
> 10.   200(bye)<---|
>
>
>
> Now Leg B remains hanging... Shouldn't FS have killed it since legA is
> dead.
>
> On Application I get Channel Hangup on 200 bye. leg B call is generated
> using bridge command with hangup_on_answer = TRUE.
>
>
> Regards,
> Vishal Sharma
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May  5 20:41:08 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 5 May 2015 11:41:08 -0500
Subject: [Freeswitch-users] DTMF relay with INFO across bridge
In-Reply-To: <1430839866075.22898@meZocliq.com>
References: <CY1PR0501MB10986151A3C7FF38ED9A806384D20@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAEJMVkAbyCW92jvi+C89S7mkRMZ7D5TVQD+ojZRw9fzY6ByA8A@mail.gmail.com>
	<1430836657047.2067@meZocliq.com>
	<CAEJMVkAKa887iY7qyqs4FMYaQk5zCCODkd2kUY9abCnae+8WMw@mail.gmail.com>
	<1430839866075.22898@meZocliq.com>
Message-ID: <CAEJMVkCE3DZxs+rqSNXUTkEsu=6gbVRw3+WB_F2wD+feZ0LMcA@mail.gmail.com>

https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues

https://wiki.freeswitch.org/wiki/RTP_Issues

Looks like some of the data wasn't moved possibly.

On Tue, May 5, 2015 at 10:31 AM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
wrote:

>  Brian,
>
> Okay I will file it on JIRA.
>
> Thanks for the input. I tried the following options, that was suggested
> on the FS website.
>
>
>  <X-PRE-PROCESS cmd="set" data="send_silence_when_idle=400"/>
>
> <param name="rtp-rewrite-timestamps" value="true"/>
>
>
>  Is there something else I am missing ?
>
>
>  thank you,
>
> Sharath
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org <
> freeswitch-users-bounces at lists.freeswitch.org> on behalf of Brian West <
> brian at freeswitch.org>
> *Sent:* Tuesday, May 5, 2015 10:42 AM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] DTMF relay with INFO across bridge
>
>  If you think you've found a bug please file a JIRA, We can review it.
> As for INFO dtmf, you do realize there is no way to guarantee order of
> digits.  So you could have them arrive out of order and there is nothing
> that can be done.  Its best to get your upstream to fix their broken
> implementations.  What RTP bug options did you try?
>
> On Tue, May 5, 2015 at 9:37 AM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
> wrote:
>
>>  Brian,
>>
>>
>>  Thanks for responding.
>>
>> 1] I had already set it in the channel variable for the bridge line.
>> Freeswitch *always* seems to convert between INFO --> rfc2833. In the
>> internal and external profiles. I have removed "liberal-dtmf" and also set
>> "dtmf-type" to "info". And the bridge line contains "dtmf_type" as info. Am
>> I missing something here?
>>
>>
>>  logs - 1fc06e8a-f334-11e4-ade6-413a2da7e1fc Dialplan: sofia/internal/
>> nXXX.sXXX at voip.XXX.com Action bridge({sip_cid_type=pid,*dtmf_type=info*
>> ,origination_privacy=screen,ignore_early_media=true}sofia/gateway/tata/XXXXXXX)
>> ?
>>
>>
>>  Tested on latest master and on 1.5.13b. Same thing.
>>
>>
>>  2] The bug from what I tested.
>>
>> - if "liberal-dtmf" is set to false and even though "dtmf-type" is set to
>> "info" in the profile, then I see "channel is not configured to use info
>> dtmf?  .
>>  >From the latest master. In mod_sofia.c
>>
>>  8589       if (dtmf.digit) {
>> 8590         if (tech_pvt->mparams.dtmf_type == DTMF_INFO ||
>>
>>  I believe it should instead be checking what is set in the profile. if
>> (tech_pvt->profile->dtmf_type? == DTMF_INFO?).
>>
>>  3] Sonus RTP fixes - Yes I have applied them. It probably works most of
>> the time. But I occasionally get complains from my customers regarding -
>> missing dtmf and repeated dtmfs. I am hoping moving to INFO will fix the
>> problem for good ? Any thoughts ?
>>
>>
>> thank you very much!
>>
>> Sharath
>>  ------------------------------
>> *From:* freeswitch-users-bounces at lists.freeswitch.org <
>> freeswitch-users-bounces at lists.freeswitch.org> on behalf of Brian West <
>> brian at freeswitch.org>
>> *Sent:* Monday, May 4, 2015 5:39 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] DTMF relay with INFO across bridge
>>
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-dtmf_type
>>
>>  I suspect you only need to set the dtmf_type inside the {} on the
>> bridge line if you wish to send INFO, You do know we have a bunch of RTP
>> bug flags that can be tweaked to make Sonus work correctly.
>>
>>  https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues
>>
>>
>>
>> On Mon, May 4, 2015 at 1:07 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com
>> > wrote:
>>
>>>  Hi,
>>>
>>>
>>>
>>> I looked around in the mailing list and don?t see any conclusive
>>> solution to this problem.
>>>
>>>
>>>
>>> I have a setup like this below.
>>>
>>>
>>>
>>> Webrtc client -----FS----sonus gateway.
>>>
>>>
>>>
>>> The Webrtc client is sending INFO as DTMF. The FS currently bridges the
>>> call to the gateway and in the outbound leg uses rfc2833.
>>>
>>>
>>>
>>> Problem: Some calls suffer from missing DTMF and repeated duplicate DTMF.
>>>
>>>
>>>
>>> Solution: Use INFO all through to sonus gateway ?
>>>
>>>
>>>
>>> How do I achieve this ?
>>>
>>> In the internal profile if I change the ?dtmf-type? to ?info?. Nothing
>>> really happens, it still sends rfc2833 dtmf.    I believe it is because of
>>> ?liberal-dtmf? is ?true?.[accept any but always offer rfc2833] But if I
>>> change ?liberal-dtmf? to ?false?. I get ?channel is not configured to use
>>> info dtmf?.
>>>
>>>
>>>
>>> Is this a bug ? Is there any way around it ?
>>>
>>>
>>>
>>> Any input much appreciated.
>>>
>>> Thank you,
>>>
>>> Sharath
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>>
>
>
>
>  --
>
> *Brian West*
> brian at freeswitch.org
>
>
>  *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>



-- 

*Brian West*
brian at freeswitch.org


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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 Sharath.Kumar at meZocliq.com  Tue May  5 22:35:51 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Tue, 5 May 2015 18:35:51 +0000
Subject: [Freeswitch-users] DTMF relay with INFO across bridge
In-Reply-To: <CAEJMVkCE3DZxs+rqSNXUTkEsu=6gbVRw3+WB_F2wD+feZ0LMcA@mail.gmail.com>
References: <CY1PR0501MB10986151A3C7FF38ED9A806384D20@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAEJMVkAbyCW92jvi+C89S7mkRMZ7D5TVQD+ojZRw9fzY6ByA8A@mail.gmail.com>
	<1430836657047.2067@meZocliq.com>
	<CAEJMVkAKa887iY7qyqs4FMYaQk5zCCODkd2kUY9abCnae+8WMw@mail.gmail.com>
	<1430839866075.22898@meZocliq.com>,
	<CAEJMVkCE3DZxs+rqSNXUTkEsu=6gbVRw3+WB_F2wD+feZ0LMcA@mail.gmail.com>
Message-ID: <1430850950511.84767@meZocliq.com>

Opened this https://freeswitch.org/jira/browse/FS-7532?


Also, for the Sonus specific DTMF ones. I thought those settings were automatically applied when Sonus is detected. I see in my logs indicating "sonus" was detected. So, I am not sure at all whether adding them will make any difference.


thanks

Sharath


________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Brian West <brian at freeswitch.org>
Sent: Tuesday, May 5, 2015 12:41 PM
To: FreeSWITCH Users Help; FreeSWITCH Docs Team
Subject: Re: [Freeswitch-users] DTMF relay with INFO across bridge

https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues

https://wiki.freeswitch.org/wiki/RTP_Issues

Looks like some of the data wasn't moved possibly.

On Tue, May 5, 2015 at 10:31 AM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:

Brian,

Okay I will file it on JIRA.

Thanks for the input. I tried the following options, that was suggested on the FS website.


<X-PRE-PROCESS cmd="set" data="send_silence_when_idle=400"/>

<param name="rtp-rewrite-timestamps" value="true"/>


Is there something else I am missing ?


thank you,

Sharath

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>>
Sent: Tuesday, May 5, 2015 10:42 AM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF relay with INFO across bridge

If you think you've found a bug please file a JIRA, We can review it.  As for INFO dtmf, you do realize there is no way to guarantee order of digits.  So you could have them arrive out of order and there is nothing that can be done.  Its best to get your upstream to fix their broken implementations.  What RTP bug options did you try?

On Tue, May 5, 2015 at 9:37 AM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:

Brian,


Thanks for responding.

1] I had already set it in the channel variable for the bridge line. Freeswitch *always* seems to convert between INFO --> rfc2833. In the internal and external profiles. I have removed "liberal-dtmf" and also set "dtmf-type" to "info". And the bridge line contains "dtmf_type" as info. Am I missing something here?


logs - 1fc06e8a-f334-11e4-ade6-413a2da7e1fc Dialplan: sofia/internal/nXXX.sXXX at voip.XXX.com<mailto:nXXX.sXXX at voip.XXX.com> Action bridge({sip_cid_type=pid,dtmf_type=info,origination_privacy=screen,ignore_early_media=true}sofia/gateway/tata/XXXXXXX)?


Tested on latest master and on 1.5.13b. Same thing.


2] The bug from what I tested.

- if "liberal-dtmf" is set to false and even though "dtmf-type" is set to "info" in the profile, then I see "channel is not configured to use info dtmf?  .

>From the latest master. In mod_sofia.c

8589       if (dtmf.digit) {
8590         if (tech_pvt->mparams.dtmf_type == DTMF_INFO ||

I believe it should instead be checking what is set in the profile. if (tech_pvt->profile->dtmf_type? == DTMF_INFO?).

3] Sonus RTP fixes - Yes I have applied them. It probably works most of the time. But I occasionally get complains from my customers regarding - missing dtmf and repeated dtmfs. I am hoping moving to INFO will fix the problem for good ? Any thoughts ?


thank you very much!

Sharath

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>>
Sent: Monday, May 4, 2015 5:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF relay with INFO across bridge

https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables#ChannelVariables-dtmf_type

I suspect you only need to set the dtmf_type inside the {} on the bridge line if you wish to send INFO, You do know we have a bunch of RTP bug flags that can be tweaked to make Sonus work correctly.

https://freeswitch.org/confluence/display/FREESWITCH/RTP+Issues



On Mon, May 4, 2015 at 1:07 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
Hi,

I looked around in the mailing list and don?t see any conclusive solution to this problem.

I have a setup like this below.

Webrtc client -----FS----sonus gateway.

The Webrtc client is sending INFO as DTMF. The FS currently bridges the call to the gateway and in the outbound leg uses rfc2833.

Problem: Some calls suffer from missing DTMF and repeated duplicate DTMF.

Solution: Use INFO all through to sonus gateway ?

How do I achieve this ?
In the internal profile if I change the ?dtmf-type? to ?info?. Nothing really happens, it still sends rfc2833 dtmf.    I believe it is because of ?liberal-dtmf? is ?true?.[accept any but always offer rfc2833] But if I change ?liberal-dtmf? to ?false?. I get ?channel is not configured to use info dtmf?.

Is this a bug ? Is there any way around it ?

Any input much appreciated.
Thank you,
Sharath

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

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

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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch<https://www.reddit.com/r/freeswitch>

T:+19184209001<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

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

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

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

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

ClueCon 2015 Call for Speakers<https://www.cluecon.com/call-for-speakers/> | Register<https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch<https://www.reddit.com/r/freeswitch>

T:+19184209001<tel:%2B19184209001> | F:+19184209002<tel:%2B19184209002> | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

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

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

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



--

Brian West
brian at freeswitch.org<mailto:brian at freeswitch.org>

[http://billing.freeswitch.org/templates/default/img/whmcslogo.png]

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

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From fernando at softov.com.br  Wed May  6 01:24:58 2015
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Tue, 5 May 2015 17:24:58 -0400
Subject: [Freeswitch-users] CISCO + FreeSwitch = nobody@domain
Message-ID: <CAKy26iNdXTD4SEN5sHa4cn6ENWrFHzRfd0GxnKu7kSoLLNu3dw@mail.gmail.com>

Hi, i'm trying to configure a CISCO 1700 IOS, 12.4 + FXO

Is there my conf in CISCO
********************
voice-port 0/0
 connection plar opx 1002
 description 1002
!
dial-peer voice 1002 voip
 description INCOMING MATCHING
 preference 1
 destination-pattern .T
 session protocol sipv2
 session target sip-server
 session transport udp
 incoming called-number .
 dtmf-relay rtp-nte
 no vad
!
sip-ua
 credentials username 1002 password 1415000915102F7F757967 realm 192.168.142.14
 authentication username 1002 password 7 08235E4C100D0043435A58
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 timers trying 1000
 registrar ipv4:192.168.142.14:5061 expires 3600
 sip-server ipv4:192.168.142.14:5061
!
********************

I have a profile running in 192.168.142.14:5061
And trying to use the user 1002 to make (authenticated) calls

This register fine, but when i'm trying to make calls i receive this

**** sofia/profile_1001/nobody at 192.168.142.14

I there a way/conf to receive 1002 at ip? like:

**** sofia/profile_1001/1002 at 192.168.142.14

Or i only receive nobody, and need to create a extension/user nobody?

   ------------------------------------------------------------------------
   ------------------------------------------------------------------------
   ------------------------------------------------------------------------
freeswitch at internal>
recv 1062 bytes from udp/[192.168.140.139]:64907 at 16:55:21.082728:
   ------------------------------------------------------------------------
   INVITE sip:1002 at 192.168.142.14:5061 SIP/2.0
   Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK743303
   From: <sip:192.168.142.14>;tag=2E582AC-1000
   To: <sip:1002 at 192.168.142.14>
   Date: Fri, 01 Mar 2002 13:38:56 GMT
   Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
   Supported: 100rel,timer,resource-priority,replaces
   Min-SE:  1800
   Cisco-Guid: 2203265591-743444950-2210067198-2261915992
   User-Agent: Cisco-SIPGateway/IOS-12.x
   Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
   CSeq: 101 INVITE
   Max-Forwards: 70
   Timestamp: 1014989936
   Contact: <sip:192.168.140.139:5060>
   Expires: 180
   Allow-Events: telephone-event
   Content-Type: application/sdp
   Content-Disposition: session;handling=required
   Content-Length: 278

   v=0
   o=CiscoSystemsSIP-GW-UserAgent 9597 428 IN IP4 192.168.140.139
   s=SIP Call
   c=IN IP4 192.168.140.139
   t=0 0
   m=audio 18060 RTP/AVP 18 101
   c=IN IP4 192.168.140.139
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
send 315 bytes to udp/[192.168.140.139]:5060 at 16:55:21.082952:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK743303
   From: <sip:192.168.142.14>;tag=2E582AC-1000
   To: <sip:1002 at 192.168.142.14>
   Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
   CSeq: 101 INVITE
   Timestamp: 1014989936 0.000085
   User-Agent: BrByte Agent
   Content-Length: 0

   ------------------------------------------------------------------------
2015-05-05 16:55:21.075398 [NOTICE] switch_channel.c:1075 New Channel
sofia/profile_1001/nobody at 192.168.142.14
[b3054ea9-60f3-e411-a31b-e840f23bc0a0]
2015-05-05 16:55:21.075398 [DEBUG] switch_core_session.c:1061 Send
signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
2015-05-05 16:55:21.075398 [DEBUG] switch_core_state_machine.c:472
(sofia/profile_1001/nobody at 192.168.142.14) Running State Change CS_NEW
2015-05-05 16:55:21.075398 [DEBUG] switch_core_session.c:1061 Send
signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
2015-05-05 16:55:21.075398 [DEBUG] sofia.c:8848
sofia/profile_1001/nobody at 192.168.142.14 receiving invite from
192.168.140.139:64907 version: 1.5.15b git d55c4a0 2015-04-07
23:11:34Z 64bit
2015-05-05 16:55:21.075398 [DEBUG] sofia.c:9015 IP 192.168.140.139
Rejected by acl "domains". Falling back to Digest auth.
2015-05-05 16:55:21.075398 [DEBUG] switch_core_state_machine.c:491
(sofia/profile_1001/nobody at 192.168.142.14) State NEW
send 788 bytes to udp/[192.168.140.139]:5060 at 16:55:21.084838:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK743303
   From: <sip:192.168.142.14>;tag=2E582AC-1000
   To: <sip:1002 at 192.168.142.14>;tag=gj3vFZ76pevSg
   Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
   CSeq: 101 INVITE
   User-Agent: BrByte Agent
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event,
dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="192.168.142.14",
nonce="4e0d4ea9-60f3-e411-a31b-e840f23bc0a0", algorithm=MD5,
qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
2015-05-05 16:55:21.075398 [DEBUG] switch_core_session.c:1061 Send
signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
2015-05-05 16:55:21.075398 [DEBUG] sofia.c:2065 detaching session
b3054ea9-60f3-e411-a31b-e840f23bc0a0
recv 380 bytes from udp/[192.168.140.139]:64907 at 16:55:21.110969:
   ------------------------------------------------------------------------
   ACK sip:1002 at 192.168.142.14:5061 SIP/2.0
   Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK743303
   From: <sip:192.168.142.14>;tag=2E582AC-1000
   To: <sip:1002 at 192.168.142.14>;tag=gj3vFZ76pevSg
   Date: Fri, 01 Mar 2002 13:38:56 GMT
   Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
   Max-Forwards: 70
   CSeq: 101 ACK
   Allow-Events: telephone-event
   Content-Length: 0

   ------------------------------------------------------------------------
recv 1310 bytes from udp/[192.168.140.139]:64907 at 16:55:21.113190:
   ------------------------------------------------------------------------
   INVITE sip:1002 at 192.168.142.14:5061 SIP/2.0
   Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK74418FA
   From: <sip:192.168.142.14>;tag=2E582AC-1000
   To: <sip:1002 at 192.168.142.14>
   Date: Fri, 01 Mar 2002 13:38:56 GMT
   Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
   Supported: 100rel,timer,resource-priority,replaces
   Min-SE:  1800
   Cisco-Guid: 2203265591-743444950-2210067198-2261915992
   User-Agent: Cisco-SIPGateway/IOS-12.x
   Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
   CSeq: 102 INVITE
   Max-Forwards: 70
   Timestamp: 1014989936
   Contact: <sip:192.168.140.139:5060>
   Expires: 180
   Allow-Events: telephone-event
   Proxy-Authorization: Digest
username="1002",realm="192.168.142.14",uri="sip:1002 at 192.168.142.14:5061",response="7d6773297fbfb70d44136a2232d62ec5",nonce="4e0d4ea9-60f3-e411-a31b-e840f23bc0a0",cnonce="66998CBE",qop="auth",algorithm=MD5,nc=00000001
   Content-Type: application/sdp
   Content-Disposition: session;handling=required
   Content-Length: 278

   v=0
   o=CiscoSystemsSIP-GW-UserAgent 9597 428 IN IP4 192.168.140.139
   s=SIP Call
   c=IN IP4 192.168.140.139
   t=0 0
   m=audio 18060 RTP/AVP 18 101
   c=IN IP4 192.168.140.139
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   ------------------------------------------------------------------------
send 316 bytes to udp/[192.168.140.139]:5060 at 16:55:21.113382:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK74418FA
   From: <sip:192.168.142.14>;tag=2E582AC-1000
   To: <sip:1002 at 192.168.142.14>
   Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
   CSeq: 102 INVITE
   Timestamp: 1014989936 0.000070
   User-Agent: BrByte Agent
   Content-Length: 0

   ------------------------------------------------------------------------
2015-05-05 16:55:21.106397 [DEBUG] sofia.c:2173 Re-attaching to
session b3054ea9-60f3-e411-a31b-e840f23bc0a0
2015-05-05 16:55:21.106397 [DEBUG] switch_core_session.c:1061 Send
signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
2015-05-05 16:55:21.106397 [DEBUG] switch_core_session.c:1061 Send
signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
2015-05-05 16:55:21.126156 [DEBUG] sofia.c:8848
sofia/profile_1001/nobody at 192.168.142.14 receiving invite from
192.168.140.139:64907 version: 1.5.15b git d55c4a0 2015-04-07
23:11:34Z 64bit
2015-05-05 16:55:21.126156 [DEBUG] sofia.c:9015 IP 192.168.140.139
Rejected by acl "domains". Falling back to Digest auth.
2015-05-05 16:55:21.126156 [DEBUG] sofia.c:6627 Channel
sofia/profile_1001/nobody at 192.168.142.14 entering state
[received][100]
2015-05-05 16:55:21.126156 [DEBUG] sofia.c:6637 Remote SDP:
v=0
o=CiscoSystemsSIP-GW-UserAgent 9597 428 IN IP4 192.168.140.139
s=SIP Call
c=IN IP4 192.168.140.139
t=0 0
m=audio 18060 RTP/AVP 18 101
c=IN IP4 192.168.140.139
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3221 Activate
Buggy RFC2833 Mode!
2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3666 Audio
Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]
2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3666 Audio
Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]
2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3666 Audio
Codec Compare [G729:18:8000:20:8000:1]/[GSM:3:8000:20:13200:1]
2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3666 Audio
Codec Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1]
2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3666 Audio
Codec Compare [G729:18:8000:20:8000:1]/[opus:116:48000:20:0:1]
2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3582 Set
telephone-event payload to 101
2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3929 Set 2833
dtmf send/recv payload to 101
2015-05-05 16:55:21.126156 [NOTICE] sofia.c:6981 Hangup
sofia/profile_1001/nobody at 192.168.142.14 [CS_NEW]
[INCOMPATIBLE_DESTINATION]
2015-05-05 16:55:21.126156 [DEBUG] switch_channel.c:3242 Send signal
sofia/profile_1001/nobody at 192.168.142.14 [KILL]
2015-05-05 16:55:21.126156 [DEBUG] switch_core_session.c:1396 Send
signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:472
(sofia/profile_1001/nobody at 192.168.142.14) Running State Change
CS_HANGUP
2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:735
(sofia/profile_1001/nobody at 192.168.142.14) Callstate Change DOWN ->
HANGUP
2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:737
(sofia/profile_1001/nobody at 192.168.142.14) State HANGUP
2015-05-05 16:55:21.126156 [DEBUG] mod_sofia.c:413 Channel
sofia/profile_1001/nobody at 192.168.142.14 hanging up, cause:
INCOMPATIBLE_DESTINATION
2015-05-05 16:55:21.126156 [DEBUG] mod_sofia.c:549 Responding to
INVITE with: 488
2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:60
sofia/profile_1001/nobody at 192.168.142.14 Standard HANGUP, cause:
INCOMPATIBLE_DESTINATION
2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:737
(sofia/profile_1001/nobody at 192.168.142.14) State HANGUP going to sleep
2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:504
(sofia/profile_1001/nobody at 192.168.142.14) State Change CS_HANGUP ->
CS_REPORTING
2015-05-05 16:55:21.126156 [DEBUG] switch_core_session.c:1396 Send
signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:472
(sofia/profile_1001/nobody at 192.168.142.14) Running State Change
CS_REPORTING
send 816 bytes to udp/[192.168.140.139]:5060 at 16:55:21.132211:
   ------------------------------------------------------------------------
   SIP/2.0 488 Not Acceptable Here
   Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK74418FA
   Max-Forwards: 70
   From: <sip:192.168.142.14>;tag=2E582AC-1000
   To: <sip:1002 at 192.168.142.14>;tag=HUvNHtramQjcc
   Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
   CSeq: 102 INVITE
   User-Agent: BrByte Agent
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event,
dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer
   Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
   Content-Length: 0
2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:823
(sofia/profile_1001/nobody at 192.168.142.14) State REPORTING
   Remote-Party-ID: "1002"
<sip:1002 at 192.168.142.14>;party=calling;privacy=off;screen=no

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

-- 
Luiz Fernando Softov
http://www.softov.com.br
fernando at softov.com.br


From italorossib at gmail.com  Wed May  6 06:11:13 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 5 May 2015 23:11:13 -0300
Subject: [Freeswitch-users] CISCO + FreeSwitch = nobody@domain
In-Reply-To: <CAKy26iNdXTD4SEN5sHa4cn6ENWrFHzRfd0GxnKu7kSoLLNu3dw@mail.gmail.com>
References: <CAKy26iNdXTD4SEN5sHa4cn6ENWrFHzRfd0GxnKu7kSoLLNu3dw@mail.gmail.com>
Message-ID: <CAJR7f2G81B-jEwuKf0dyU8fu1prPM7CV6d-mjbTLNCzEfo7YEQ@mail.gmail.com>

Cisco is not forwarding the caller-id to your box, I don't know what
exactly command you need to enable this on cisco, probably on your
dial-peer.

But,  you're getting 488 in this call, Cisco is offering only g729 and you
don't have it enabled in your inbound codes prefs (check vars.xml)
Em 05/05/2015 18:26, "Luiz Fernando Softov" <fernando at softov.com.br>
escreveu:

> Hi, i'm trying to configure a CISCO 1700 IOS, 12.4 + FXO
>
> Is there my conf in CISCO
> ********************
> voice-port 0/0
>  connection plar opx 1002
>  description 1002
> !
> dial-peer voice 1002 voip
>  description INCOMING MATCHING
>  preference 1
>  destination-pattern .T
>  session protocol sipv2
>  session target sip-server
>  session transport udp
>  incoming called-number .
>  dtmf-relay rtp-nte
>  no vad
> !
> sip-ua
>  credentials username 1002 password 1415000915102F7F757967 realm
> 192.168.142.14
>  authentication username 1002 password 7 08235E4C100D0043435A58
>  retry invite 3
>  retry response 3
>  retry bye 3
>  retry cancel 3
>  timers trying 1000
>  registrar ipv4:192.168.142.14:5061 expires 3600
>  sip-server ipv4:192.168.142.14:5061
> !
> ********************
>
> I have a profile running in 192.168.142.14:5061
> And trying to use the user 1002 to make (authenticated) calls
>
> This register fine, but when i'm trying to make calls i receive this
>
> **** sofia/profile_1001/nobody at 192.168.142.14
>
> I there a way/conf to receive 1002 at ip? like:
>
> **** sofia/profile_1001/1002 at 192.168.142.14
>
> Or i only receive nobody, and need to create a extension/user nobody?
>
>    ------------------------------------------------------------------------
>    ------------------------------------------------------------------------
>    ------------------------------------------------------------------------
> freeswitch at internal>
> recv 1062 bytes from udp/[192.168.140.139]:64907 at 16:55:21.082728:
>    ------------------------------------------------------------------------
>    INVITE sip:1002 at 192.168.142.14:5061 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK743303
>    From: <sip:192.168.142.14>;tag=2E582AC-1000
>    To: <sip:1002 at 192.168.142.14>
>    Date: Fri, 01 Mar 2002 13:38:56 GMT
>    Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
>    Supported: 100rel,timer,resource-priority,replaces
>    Min-SE:  1800
>    Cisco-Guid: 2203265591-743444950-2210067198-2261915992
>    User-Agent: Cisco-SIPGateway/IOS-12.x
>    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY, INFO, REGISTER
>    CSeq: 101 INVITE
>    Max-Forwards: 70
>    Timestamp: 1014989936
>    Contact: <sip:192.168.140.139:5060>
>    Expires: 180
>    Allow-Events: telephone-event
>    Content-Type: application/sdp
>    Content-Disposition: session;handling=required
>    Content-Length: 278
>
>    v=0
>    o=CiscoSystemsSIP-GW-UserAgent 9597 428 IN IP4 192.168.140.139
>    s=SIP Call
>    c=IN IP4 192.168.140.139
>    t=0 0
>    m=audio 18060 RTP/AVP 18 101
>    c=IN IP4 192.168.140.139
>    a=rtpmap:18 G729/8000
>    a=fmtp:18 annexb=no
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
> send 315 bytes to udp/[192.168.140.139]:5060 at 16:55:21.082952:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK743303
>    From: <sip:192.168.142.14>;tag=2E582AC-1000
>    To: <sip:1002 at 192.168.142.14>
>    Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
>    CSeq: 101 INVITE
>    Timestamp: 1014989936 0.000085
>    User-Agent: BrByte Agent
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2015-05-05 16:55:21.075398 [NOTICE] switch_channel.c:1075 New Channel
> sofia/profile_1001/nobody at 192.168.142.14
> [b3054ea9-60f3-e411-a31b-e840f23bc0a0]
> 2015-05-05 16:55:21.075398 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
> 2015-05-05 16:55:21.075398 [DEBUG] switch_core_state_machine.c:472
> (sofia/profile_1001/nobody at 192.168.142.14) Running State Change CS_NEW
> 2015-05-05 16:55:21.075398 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
> 2015-05-05 16:55:21.075398 [DEBUG] sofia.c:8848
> sofia/profile_1001/nobody at 192.168.142.14 receiving invite from
> 192.168.140.139:64907 version: 1.5.15b git d55c4a0 2015-04-07
> 23:11:34Z 64bit
> 2015-05-05 16:55:21.075398 [DEBUG] sofia.c:9015 IP 192.168.140.139
> Rejected by acl "domains". Falling back to Digest auth.
> 2015-05-05 16:55:21.075398 [DEBUG] switch_core_state_machine.c:491
> (sofia/profile_1001/nobody at 192.168.142.14) State NEW
> send 788 bytes to udp/[192.168.140.139]:5060 at 16:55:21.084838:
>    ------------------------------------------------------------------------
>    SIP/2.0 407 Proxy Authentication Required
>    Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK743303
>    From: <sip:192.168.142.14>;tag=2E582AC-1000
>    To: <sip:1002 at 192.168.142.14>;tag=gj3vFZ76pevSg
>    Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
>    CSeq: 101 INVITE
>    User-Agent: BrByte Agent
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, presence, as-feature-event,
> dialog, line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer
>    Proxy-Authenticate: Digest realm="192.168.142.14",
> nonce="4e0d4ea9-60f3-e411-a31b-e840f23bc0a0", algorithm=MD5,
> qop="auth"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2015-05-05 16:55:21.075398 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
> 2015-05-05 16:55:21.075398 [DEBUG] sofia.c:2065 detaching session
> b3054ea9-60f3-e411-a31b-e840f23bc0a0
> recv 380 bytes from udp/[192.168.140.139]:64907 at 16:55:21.110969:
>    ------------------------------------------------------------------------
>    ACK sip:1002 at 192.168.142.14:5061 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK743303
>    From: <sip:192.168.142.14>;tag=2E582AC-1000
>    To: <sip:1002 at 192.168.142.14>;tag=gj3vFZ76pevSg
>    Date: Fri, 01 Mar 2002 13:38:56 GMT
>    Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
>    Max-Forwards: 70
>    CSeq: 101 ACK
>    Allow-Events: telephone-event
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 1310 bytes from udp/[192.168.140.139]:64907 at 16:55:21.113190:
>    ------------------------------------------------------------------------
>    INVITE sip:1002 at 192.168.142.14:5061 SIP/2.0
>    Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK74418FA
>    From: <sip:192.168.142.14>;tag=2E582AC-1000
>    To: <sip:1002 at 192.168.142.14>
>    Date: Fri, 01 Mar 2002 13:38:56 GMT
>    Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
>    Supported: 100rel,timer,resource-priority,replaces
>    Min-SE:  1800
>    Cisco-Guid: 2203265591-743444950-2210067198-2261915992
>    User-Agent: Cisco-SIPGateway/IOS-12.x
>    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY, INFO, REGISTER
>    CSeq: 102 INVITE
>    Max-Forwards: 70
>    Timestamp: 1014989936
>    Contact: <sip:192.168.140.139:5060>
>    Expires: 180
>    Allow-Events: telephone-event
>    Proxy-Authorization: Digest
> username="1002",realm="192.168.142.14",uri="sip:1002 at 192.168.142.14:5061
> ",response="7d6773297fbfb70d44136a2232d62ec5",nonce="4e0d4ea9-60f3-e411-a31b-e840f23bc0a0",cnonce="66998CBE",qop="auth",algorithm=MD5,nc=00000001
>    Content-Type: application/sdp
>    Content-Disposition: session;handling=required
>    Content-Length: 278
>
>    v=0
>    o=CiscoSystemsSIP-GW-UserAgent 9597 428 IN IP4 192.168.140.139
>    s=SIP Call
>    c=IN IP4 192.168.140.139
>    t=0 0
>    m=audio 18060 RTP/AVP 18 101
>    c=IN IP4 192.168.140.139
>    a=rtpmap:18 G729/8000
>    a=fmtp:18 annexb=no
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    ------------------------------------------------------------------------
> send 316 bytes to udp/[192.168.140.139]:5060 at 16:55:21.113382:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK74418FA
>    From: <sip:192.168.142.14>;tag=2E582AC-1000
>    To: <sip:1002 at 192.168.142.14>
>    Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
>    CSeq: 102 INVITE
>    Timestamp: 1014989936 0.000070
>    User-Agent: BrByte Agent
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2015-05-05 16:55:21.106397 [DEBUG] sofia.c:2173 Re-attaching to
> session b3054ea9-60f3-e411-a31b-e840f23bc0a0
> 2015-05-05 16:55:21.106397 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
> 2015-05-05 16:55:21.106397 [DEBUG] switch_core_session.c:1061 Send
> signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
> 2015-05-05 16:55:21.126156 [DEBUG] sofia.c:8848
> sofia/profile_1001/nobody at 192.168.142.14 receiving invite from
> 192.168.140.139:64907 version: 1.5.15b git d55c4a0 2015-04-07
> 23:11:34Z 64bit
> 2015-05-05 16:55:21.126156 [DEBUG] sofia.c:9015 IP 192.168.140.139
> Rejected by acl "domains". Falling back to Digest auth.
> 2015-05-05 16:55:21.126156 [DEBUG] sofia.c:6627 Channel
> sofia/profile_1001/nobody at 192.168.142.14 entering state
> [received][100]
> 2015-05-05 16:55:21.126156 [DEBUG] sofia.c:6637 Remote SDP:
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 9597 428 IN IP4 192.168.140.139
> s=SIP Call
> c=IN IP4 192.168.140.139
> t=0 0
> m=audio 18060 RTP/AVP 18 101
> c=IN IP4 192.168.140.139
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3221 Activate
> Buggy RFC2833 Mode!
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3666 Audio
> Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3666 Audio
> Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3666 Audio
> Codec Compare [G729:18:8000:20:8000:1]/[GSM:3:8000:20:13200:1]
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3666 Audio
> Codec Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1]
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3666 Audio
> Codec Compare [G729:18:8000:20:8000:1]/[opus:116:48000:20:0:1]
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3582 Set
> telephone-event payload to 101
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_media.c:3929 Set 2833
> dtmf send/recv payload to 101
> 2015-05-05 16:55:21.126156 [NOTICE] sofia.c:6981 Hangup
> sofia/profile_1001/nobody at 192.168.142.14 [CS_NEW]
> [INCOMPATIBLE_DESTINATION]
> 2015-05-05 16:55:21.126156 [DEBUG] switch_channel.c:3242 Send signal
> sofia/profile_1001/nobody at 192.168.142.14 [KILL]
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:472
> (sofia/profile_1001/nobody at 192.168.142.14) Running State Change
> CS_HANGUP
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:735
> (sofia/profile_1001/nobody at 192.168.142.14) Callstate Change DOWN ->
> HANGUP
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:737
> (sofia/profile_1001/nobody at 192.168.142.14) State HANGUP
> 2015-05-05 16:55:21.126156 [DEBUG] mod_sofia.c:413 Channel
> sofia/profile_1001/nobody at 192.168.142.14 hanging up, cause:
> INCOMPATIBLE_DESTINATION
> 2015-05-05 16:55:21.126156 [DEBUG] mod_sofia.c:549 Responding to
> INVITE with: 488
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:60
> sofia/profile_1001/nobody at 192.168.142.14 Standard HANGUP, cause:
> INCOMPATIBLE_DESTINATION
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:737
> (sofia/profile_1001/nobody at 192.168.142.14) State HANGUP going to sleep
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:504
> (sofia/profile_1001/nobody at 192.168.142.14) State Change CS_HANGUP ->
> CS_REPORTING
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_session.c:1396 Send
> signal sofia/profile_1001/nobody at 192.168.142.14 [BREAK]
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:472
> (sofia/profile_1001/nobody at 192.168.142.14) Running State Change
> CS_REPORTING
> send 816 bytes to udp/[192.168.140.139]:5060 at 16:55:21.132211:
>    ------------------------------------------------------------------------
>    SIP/2.0 488 Not Acceptable Here
>    Via: SIP/2.0/UDP 192.168.140.139:5060;branch=z9hG4bK74418FA
>    Max-Forwards: 70
>    From: <sip:192.168.142.14>;tag=2E582AC-1000
>    To: <sip:1002 at 192.168.142.14>;tag=HUvNHtramQjcc
>    Call-ID: 83580E71-2C5011D6-83BFF2FE-86D21958 at 192.168.140.139
>    CSeq: 102 INVITE
>    User-Agent: BrByte Agent
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, presence, as-feature-event,
> dialog, line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer
>    Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
>    Content-Length: 0
> 2015-05-05 16:55:21.126156 [DEBUG] switch_core_state_machine.c:823
> (sofia/profile_1001/nobody at 192.168.142.14) State REPORTING
>    Remote-Party-ID: "1002"
> <sip:1002 at 192.168.142.14>;party=calling;privacy=off;screen=no
>
>    ------------------------------------------------------------------------
>
> --
> Luiz Fernando Softov
> http://www.softov.com.br
> fernando at softov.com.br
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bob.mccarthy at experient.com  Wed May  6 09:01:08 2015
From: bob.mccarthy at experient.com (Bob McCarthy)
Date: Tue, 5 May 2015 23:01:08 -0600
Subject: [Freeswitch-users] Polycom VVX Shared Call Appearance
In-Reply-To: <CAEwTsAgZgtgYwac4z=Nt7GGUeiT7kYrWBN9K5gYA325GnGnCUA@mail.gmail.com>
References: <040c01d08163$c3f78b70$4be6a250$@experient.com>
	<CAEwTsAgZgtgYwac4z=Nt7GGUeiT7kYrWBN9K5gYA325GnGnCUA@mail.gmail.com>
Message-ID: <007801d087b9$ac3ea2f0$04bbe8d0$@experient.com>

I got the sip traces but I am not sure what I am looking at/for.  What is puzzling is that the barge works but the on hold line grab does not.

 

It?s most likely some setting in the polycom VVX.  Anybody with a trained eye ?

 

Bob

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, April 28, 2015 6:55 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Polycom VVX Shared Call Appearance

 

we have had that working for a long time.  Can you compare sip traces on working vs non working to see what changed?

On Monday, April 27, 2015, Bob McCarthy <bob.mccarthy at experient.com <mailto:bob.mccarthy at experient.com> > wrote:

Trying to upgrade our Polycom Phones from SoundPoint 670?s to VVX 600?s.  

We have had Shared Call appearance working for the last couple of years on the 670?s.  We have it mostly working on the VVX except for the on hold pick-off.

 

The pick-off enters the dial-plan which then fails. We never needed to dial plan this for the 670?s. Using the same registrations / IP addresses i.e. one for one swap out.

 

Anyone have VVX?s working with SCA ?

 

 

Bob

 

 

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From vishal.sharma at knowlarity.com  Wed May  6 09:29:24 2015
From: vishal.sharma at knowlarity.com (Vishal Sharma)
Date: Wed, 6 May 2015 10:59:24 +0530
Subject: [Freeswitch-users] legB not Disconnected
In-Reply-To: <CAHtxdDcz10tnN1whDHnuog7Ua+wD2sg2RBjG+d4aJf+83Sr5wQ@mail.gmail.com>
References: <CAEaAqfga43Br2NwZB2cQVYQV1wF+HjnnSNB1i6Ep1eCwQJ21sg@mail.gmail.com>
	<CAHtxdDcz10tnN1whDHnuog7Ua+wD2sg2RBjG+d4aJf+83Sr5wQ@mail.gmail.com>
Message-ID: <CAEaAqfg5P=2f76ELOLL87M0W54Cp7HL6LRMoDauU=F1Qfps5jQ@mail.gmail.com>

It's a different case, I am not using TCP,

On Tue, May 5, 2015 at 9:49 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Similar case https://freeswitch.org/jira/browse/FS-7399
>
>
> On Tue, May 5, 2015 at 1:42 PM, Vishal Sharma <
> vishal.sharma at knowlarity.com> wrote:
>
>> Hi All,
>> I am using ESL to control the call flow.
>> My initial call flow is as following
>>
>> 1.      INVITE---> FS
>> 2.      180<--------|
>> 3.      200<--------|
>> 4.      ------------>ACK
>> 5.     *************|-------------->INVITE
>> 6.     *************|<---------------180
>> 7.     BYE------->|
>> 8.     Retransmissions of BYE
>> 9.     *************|<----------------200 OK
>> 10.   200(bye)<---|
>>
>>
>>
>> Now Leg B remains hanging... Shouldn't FS have killed it since legA is
>> dead.
>>
>> On Application I get Channel Hangup on 200 bye. leg B call is generated
>> using bridge command with hangup_on_answer = TRUE.
>>
>>
>> Regards,
>> Vishal Sharma
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Chris.Young at enghouse.com  Wed May  6 14:59:57 2015
From: Chris.Young at enghouse.com (Chris Young)
Date: Wed, 6 May 2015 10:59:57 +0000
Subject: [Freeswitch-users] UNENCRYPTED_SRTCP
Message-ID: <72741d4c65204a9badeaa21a2a269bd9@UK-MAIL-001.edge.local>

Hi all,

When configuring FreeSWITCH to use SRTP, is it possible to have the UNENCRYPTED_SRTCP parameter appended to the end of the crypto suite when the RTP/SAVP profile is in use? So, far I have an entry similar to the following in the SDP:

                a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:o+6sFNUpD01VfzI3XXH2/0rGQJJyauPiNgBAvgVl

but the gateway we are connecting to seems to require UNENCRYPTED_SRTCP to be present as well e.g.

                a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:o+6sFNUpD01VfzI3XXH2/0rGQJJyauPiNgBAvgVl UNENCRYPTED_SRTCP

Many thanks,
Chris

Chris Young
Software Engineer
[cid:image7482a0.PNG at dc00f514.4fb88205]
t:  +44 118 943 9249
e:  chris.young at enghouse.com<mailto:chris.young at enghouse.com>
w: www.enghouseinteractive.co.uk<http://www.enghouseinteractive.co.uk>
[cid:image6c1ba6.PNG at b8963a35.449174f5]

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From yadenis at seznam.cz  Wed May  6 15:47:51 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 6 May 2015 13:47:51 +0200
Subject: [Freeswitch-users]  Video record problem
Message-ID: <1414775449.20150506134751@seznam.cz>

Hi all,

I'm trying to make a video recording of the conference calls (sound works without problems)

Easy extension

<extension name="conference_set_auto_outcall">
     <condition field="destination_number" expression="^(12345)$">
       <action application="answer"/>
       <action application="execute_extension" data="RECORD"/>
       <action application="set" data="conference_auto_outcall_timeout=3"/>
       <action application="set" data="conference_auto_outcall_flags=none"/>
       <action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/>
       <action application="set" data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/>
       <action application="set" data="conference_auto_outcall_profile=default"/>
       <action application="conference_set_auto_outcall" data="{ignore_early_media=true}user/1005@$${domain}"/>
       <action application="conference" data="$1 at simple"/>
     </condition>
</extension>

Plus extension for recording

<extension name="video record that call">
      <condition field="destination_number" expression="^RECORD$" break="on-true">
            <action application="set" data="res=${bgapi uuid_setvar ${uuid} enable_file_write_buffering false}"/>
            <action application="set" data="res=${bgapi uuid_record ${uuid} start /usr/local/freeswitch/recordings/Video_${uuid}.fsv}"/>
      </condition>
</extension>
  
But I always have an error in the log.

2015-05-06 13:35:37.601687 [ERR] mod_fsv.c:702 You are asking to write 1024 bytes of data which is not supported. Please set enable_file_write_buffering=false to use .fsv format
fe2f13e2-f3e3-11e4-91db-33c69ed9247a 2015-05-06 13:35:37.601687 [ERR] switch_ivr_async.c:1155 Error writing /usr/local/freeswitch/recordings/Video_fe2f13e2-f3e3-11e4-91db-33c69ed9247a.fsv
2015-05-06 13:35:37.721700 [ERR] mod_fsv.c:702 You are asking to write 1024 bytes of data which is not supported. Please set enable_file_write_buffering=false to use .fsv format
fe2f13e2-f3e3-11e4-91db-33c69ed9247a 2015-05-06 13:35:37.721700 [ERR] switch_ivr_async.c:1155 Error writing /usr/local/freeswitch/recordings/Video_fe2f13e2-f3e3-11e4-91db-33c69ed9247a.fsv
2015-05-06 13:35:37.861731 [ERR] mod_fsv.c:702 You are asking to write 1024 bytes of data which is not supported. Please set enable_file_write_buffering=false to use .fsv format
fe2f13e2-f3e3-11e4-91db-33c69ed9247a 2015-05-06 13:35:37.861731 [ERR] switch_ivr_async.c:1155 Error writing /usr/local/freeswitch/recordings/Video_fe2f13e2-f3e3-11e4-91db-33c69ed9247a.fsv
2015-05-06 13:35:37.981710 [ERR] mod_fsv.c:702 You are asking to write 1024 bytes of data which is not supported. Please set enable_file_write_buffering=false to use .fsv format
fe2f13e2-f3e3-11e4-91db-33c69ed9247a 2015-05-06 13:35:37.981710 [ERR] switch_ivr_async.c:1155 Error writing /usr/local/freeswitch/recordings/Video_fe2f13e2-f3e3-11e4-91db-33c69ed9247a.fsv
2015-05-06 13:35:38.101712 [ERR] mod_fsv.c:702 You are asking to write 1024 bytes of data which is not supported. Please set enable_file_write_buffering=false to use .fsv format 


Although this option, I of course turn. 

Hence the question. Is there a possibility to include this option globally? Or how it can be solved differently? What am I doing wrong?

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From Sharath.Kumar at meZocliq.com  Wed May  6 18:26:21 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Wed, 6 May 2015 14:26:21 +0000
Subject: [Freeswitch-users] DTMF INFO for outbound bridge support
Message-ID: <CY1PR0501MB1098436E325C2693B13910A084D00@CY1PR0501MB1098.namprd05.prod.outlook.com>

>From https://freeswitch.org/jira/browse/FS-7532

Yossi Neimann says -  "As we aren't bridging from FreeSWITCH to anything that needs SIP INFO, we don't make use of that. It's strictly used in our scenario for inbound calls. All of the outbound systems we communicate with support rfc2833."

So, can I assume that "Freeswitch" DOES NOT support sending SIP INFO for dtmf on outbound legs that are bridged ? Please can someone else confirm. If this is so I believe it should be documented. And also I am not sure what is the purpose of dtmf_type in the {} bridge line.

https://freeswitch.org/confluence/display/FREESWITCH/Variables

Thank you!
Sharath


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From k.pabijanskas at gmail.com  Wed May  6 16:49:57 2015
From: k.pabijanskas at gmail.com (Karolis Pabijanskas)
Date: Wed, 6 May 2015 13:49:57 +0100
Subject: [Freeswitch-users] mod_callcenter: wrap_up_time
Message-ID: <CANYNSSL9s1puLh=qNa=6N2qbVhM9gKi+ttebxGnhNq0jqPta3g@mail.gmail.com>

Hi All,

I am using a mod_callcenter for this particular set-up.

I need the agents to have a wrap_up_time set to, say, 300 seconds, which
works great. In some cases, however, the agent needs to have a code he can
dial so he resets his wrap_up_time (say, he receives a call to the wrong
number).

It does seem that if I call:

callcenter_config agent set state AGENT Waiting

It sets the call center to available and the wrap-up time is cancelled, but
the wrap-up never kicks in again!

Is there a way to reset the agent wrap_up_time via the API (which I can
then call via the dialplan) so that the agent is available for a new call
immediately, but that the wrap-up time kicks in afterwards as normal?

Many thanks for your help!

Karolis
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From ali.jibran44 at gmail.com  Wed May  6 18:58:26 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Wed, 6 May 2015 19:58:26 +0500
Subject: [Freeswitch-users] mod_callcenter: wrap_up_time
In-Reply-To: <CANYNSSL9s1puLh=qNa=6N2qbVhM9gKi+ttebxGnhNq0jqPta3g@mail.gmail.com>
References: <CANYNSSL9s1puLh=qNa=6N2qbVhM9gKi+ttebxGnhNq0jqPta3g@mail.gmail.com>
Message-ID: <CAP9PqkT1S9stcsNS7cJ_WXwt=eB40v+k=mdK+GzLH0fsW7YpLw@mail.gmail.com>

Maybe try

callcenter_config agent set
[key(contact|status|state|type|max_no_answer|wrap_up_time|ready_time|reject_delay_time|busy_delay_time)]
[agent name] [value]


On Wed, May 6, 2015 at 5:49 PM, Karolis Pabijanskas <k.pabijanskas at gmail.com
> wrote:

> Hi All,
>
> I am using a mod_callcenter for this particular set-up.
>
> I need the agents to have a wrap_up_time set to, say, 300 seconds, which
> works great. In some cases, however, the agent needs to have a code he can
> dial so he resets his wrap_up_time (say, he receives a call to the wrong
> number).
>
> It does seem that if I call:
>
> callcenter_config agent set state AGENT Waiting
>
> It sets the call center to available and the wrap-up time is cancelled,
> but the wrap-up never kicks in again!
>
> Is there a way to reset the agent wrap_up_time via the API (which I can
> then call via the dialplan) so that the agent is available for a new call
> immediately, but that the wrap-up time kicks in afterwards as normal?
>
> Many thanks for your help!
>
> Karolis
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mishehu at freeswitch.org  Wed May  6 21:35:36 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Wed, 06 May 2015 12:35:36 -0500
Subject: [Freeswitch-users] DTMF INFO for outbound bridge support
In-Reply-To: <CY1PR0501MB1098436E325C2693B13910A084D00@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CY1PR0501MB1098436E325C2693B13910A084D00@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <554A50E8.5020602@freeswitch.org>

I'm only speaking about the specific usage that we make of FreeSWITCH.  
FreeSWITCH *should* be able to use SIP INFO if configured for such 
regardless of channel direction (inbound or outbound).

Once you have a Jira open, please keep the bulk of correspondence in the 
Jira.  It makes it difficult to track the conversation if the 
conversation is happening in 3 places at once.  :-)

-Yossi

On 05/06/2015 09:26 AM, Sharath Kumar wrote:
>
> From https://freeswitch.org/jira/browse/FS-7532
>
> Yossi Neimann says -  ?As we aren't bridging from FreeSWITCH to 
> anything that needs SIP INFO, we don't make use of that. It's strictly 
> used in our scenario for inbound calls. *All of the outbound systems 
> we communicate with support rfc2833.?*
>
> So, can I assume that ?Freeswitch? DOES NOT support sending SIP INFO 
> for dtmf on outbound legs that are bridged ? Please can someone else 
> confirm. If this is so I believe it should be documented. And also I 
> am not sure what is the purpose of dtmf_type in the {} bridge line.
>
> https://freeswitch.org/confluence/display/FREESWITCH/Variables
>
> Thank you!
>
> Sharath
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From Sharath.Kumar at meZocliq.com  Wed May  6 22:02:18 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Wed, 6 May 2015 18:02:18 +0000
Subject: [Freeswitch-users] DTMF INFO for outbound bridge support
In-Reply-To: <554A50E8.5020602@freeswitch.org>
References: <CY1PR0501MB1098436E325C2693B13910A084D00@CY1PR0501MB1098.namprd05.prod.outlook.com>,
	<554A50E8.5020602@freeswitch.org>
Message-ID: <1430935336907.63457@meZocliq.com>

Okay sir no problem.

________________________________
From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org> on behalf of I put the Who? in Mishehu <mishehu at freeswitch.org>
Sent: Wednesday, May 6, 2015 1:35 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] DTMF INFO for outbound bridge support

I'm only speaking about the specific usage that we make of FreeSWITCH.  FreeSWITCH *should* be able to use SIP INFO if configured for such regardless of channel direction (inbound or outbound).

Once you have a Jira open, please keep the bulk of correspondence in the Jira.  It makes it difficult to track the conversation if the conversation is happening in 3 places at once.  :-)

-Yossi

On 05/06/2015 09:26 AM, Sharath Kumar wrote:
>From https://freeswitch.org/jira/browse/FS-7532

Yossi Neimann says -  "As we aren't bridging from FreeSWITCH to anything that needs SIP INFO, we don't make use of that. It's strictly used in our scenario for inbound calls. All of the outbound systems we communicate with support rfc2833."

So, can I assume that "Freeswitch" DOES NOT support sending SIP INFO for dtmf on outbound legs that are bridged ? Please can someone else confirm. If this is so I believe it should be documented. And also I am not sure what is the purpose of dtmf_type in the {} bridge line.

https://freeswitch.org/confluence/display/FREESWITCH/Variables

Thank you!
Sharath





_________________________________________________________________________
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  Wed May  6 22:09:29 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 06 May 2015 18:09:29 +0000
Subject: [Freeswitch-users] ClueCon Weekly May 6th, 2015!
Message-ID: <554a58d994056_2dc0d7d3187418@resque-worker-high.3.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1KMlVAW
ClueCon Weekly May 6th, 2015!
Check out the weekly conference call to see the latest news!

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From cdgraff at gmail.com  Wed May  6 22:50:39 2015
From: cdgraff at gmail.com (Alejandro)
Date: Wed, 6 May 2015 15:50:39 -0300
Subject: [Freeswitch-users] Json API for Voicemail?
Message-ID: <CALJtzxDy-3-A0L62KmmhSzqyv=oP68ZKTNKxbkUeq1FTsp8qjQ@mail.gmail.com>

Hi Guys,

Is there some module to list the Voicemail messages in Json or XML.

I has enabled RPC module and see the voicemail and listen into the native
website... but I like to create custom view and list the message true some
API.

Some advice? or link to share?

Thanks in advance!
Alejandro
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From royj at yandex.ru  Thu May  7 11:06:45 2015
From: royj at yandex.ru (royj at yandex.ru)
Date: Thu, 07 May 2015 10:06:45 +0300
Subject: [Freeswitch-users] Outbound ESL, typical load average
In-Reply-To: <212791428325270@web23m.yandex.ru>
References: <212791428325270@web23m.yandex.ru>
Message-ID: <10947341430982405@web14g.yandex.ru>

just for record
we have changed application logic, we use now sync mode instead of async and observe some decrease of load average

06.04.2015, 16:04, "royj at yandex.ru" <royj at yandex.ru>:
> Hi all.
> Is there anyone here, who uses outbound ESL and have 200 and more total channels on FreeSWITCH instance.
> There is https://freeswitch.org/jira/browse/FS-7400 . Seems like FreeSWITCH is OK, at least developers did not found something abnormal with it.
> If somebody here, who uses similar configuration, what is your typical load average. Any comments are appreciated.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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.medina at cibersys.com  Thu May  7 17:18:01 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Thu, 7 May 2015 08:48:01 -0430
Subject: [Freeswitch-users] Error on SQL - data malformed
Message-ID: <CACXsapPzmB_DcAVpLGYj=RPXys4FCGgDOgNZtWNJfDWyUG32yg@mail.gmail.com>

Hi guys!

Im testing Master from yesterday. Im seeing this on the console

freeswitch at srv-ovh-devel-in-000001> show registrations
2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:108 SQL ERR [database
disk image is malformed]
2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:223 SQL ERR [database
disk image is malformed]
2015-05-07 13:16:03.047907 [CRIT] switch_core_sqldb.c:508 Failure to
connect to CORE_DB core!
2015-05-07 13:16:03.047907 [ERR] switch_console.c:256 Database Error
2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:108 SQL ERR [database
disk image is malformed]
2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:223 SQL ERR [database
disk image is malformed]

-ERR Database error!

2015-05-07 13:16:03.047907 [CRIT] switch_core_sqldb.c:508 Failure to
connect to CORE_DB core!
freeswitch at srv-ovh-devel-in-000001>
freeswitch at srv-ovh-devel-in-000001> show registrations
2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:108 SQL ERR [database
disk image is malformed]
2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:223 SQL ERR [database
disk image is malformed]
2015-05-07 13:16:18.047877 [CRIT] switch_core_sqldb.c:508 Failure to
connect to CORE_DB core!
2015-05-07 13:16:18.047877 [ERR] switch_console.c:256 Database Error
2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:108 SQL ERR [database
disk image is malformed]
2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:223 SQL ERR [database
disk image is malformed]
2015-05-07 13:16:18.047877 [CRIT] switch_core_sqldb.c:508 Failure to
connect to CORE_DB core!

-ERR Database error!




-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From italorossib at gmail.com  Thu May  7 18:52:50 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 7 May 2015 11:52:50 -0300
Subject: [Freeswitch-users] Error on SQL - data malformed
In-Reply-To: <CACXsapPzmB_DcAVpLGYj=RPXys4FCGgDOgNZtWNJfDWyUG32yg@mail.gmail.com>
References: <CACXsapPzmB_DcAVpLGYj=RPXys4FCGgDOgNZtWNJfDWyUG32yg@mail.gmail.com>
Message-ID: <CAJR7f2Fn1dcwEo39xmkdH5kMdaOEWmkV+WrxTd_mfRPHuJG4Yg@mail.gmail.com>

Your core.db is corrupt, it could be caused by a power failure.

To solve this you can try to repair the database (search for sqlite repair
malformed) or you can just remove the old db and let FreeSWITCH recreate a
new one for you.

rm /usr/local/freeswitch/db/core.db

On Thu, May 7, 2015 at 10:18 AM, Victor Medina <victor.medina at cibersys.com>
wrote:

> Hi guys!
>
> Im testing Master from yesterday. Im seeing this on the console
>
> freeswitch at srv-ovh-devel-in-000001> show registrations
> 2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:108 SQL ERR [database
> disk image is malformed]
> 2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:223 SQL ERR [database
> disk image is malformed]
> 2015-05-07 13:16:03.047907 [CRIT] switch_core_sqldb.c:508 Failure to
> connect to CORE_DB core!
> 2015-05-07 13:16:03.047907 [ERR] switch_console.c:256 Database Error
> 2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:108 SQL ERR [database
> disk image is malformed]
> 2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:223 SQL ERR [database
> disk image is malformed]
>
> -ERR Database error!
>
> 2015-05-07 13:16:03.047907 [CRIT] switch_core_sqldb.c:508 Failure to
> connect to CORE_DB core!
> freeswitch at srv-ovh-devel-in-000001>
> freeswitch at srv-ovh-devel-in-000001> show registrations
> 2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:108 SQL ERR [database
> disk image is malformed]
> 2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:223 SQL ERR [database
> disk image is malformed]
> 2015-05-07 13:16:18.047877 [CRIT] switch_core_sqldb.c:508 Failure to
> connect to CORE_DB core!
> 2015-05-07 13:16:18.047877 [ERR] switch_console.c:256 Database Error
> 2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:108 SQL ERR [database
> disk image is malformed]
> 2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:223 SQL ERR [database
> disk image is malformed]
> 2015-05-07 13:16:18.047877 [CRIT] switch_core_sqldb.c:508 Failure to
> connect to CORE_DB core!
>
> -ERR Database error!
>
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From victor.medina at cibersys.com  Thu May  7 20:19:43 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Thu, 7 May 2015 11:49:43 -0430
Subject: [Freeswitch-users] Error on SQL - data malformed
In-Reply-To: <CAJR7f2Fn1dcwEo39xmkdH5kMdaOEWmkV+WrxTd_mfRPHuJG4Yg@mail.gmail.com>
References: <CACXsapPzmB_DcAVpLGYj=RPXys4FCGgDOgNZtWNJfDWyUG32yg@mail.gmail.com>
	<CAJR7f2Fn1dcwEo39xmkdH5kMdaOEWmkV+WrxTd_mfRPHuJG4Yg@mail.gmail.com>
Message-ID: <CACXsapNStHkjwR+LYwmrjiZs+OLjAsNTJOD-3B4Phd0JzfuaMA@mail.gmail.com>

Italo!

Thanks for the replay.

I just deleted the db/ files.It was just a test. A vm. No power failures.
Running on vmware.

2015-05-07 10:22 GMT-04:30 ?talo Rossi <italorossib at gmail.com>:

> Your core.db is corrupt, it could be caused by a power failure.
>
> To solve this you can try to repair the database (search for sqlite repair
> malformed) or you can just remove the old db and let FreeSWITCH recreate a
> new one for you.
>
> rm /usr/local/freeswitch/db/core.db
>
> On Thu, May 7, 2015 at 10:18 AM, Victor Medina <victor.medina at cibersys.com
> > wrote:
>
>> Hi guys!
>>
>> Im testing Master from yesterday. Im seeing this on the console
>>
>> freeswitch at srv-ovh-devel-in-000001> show registrations
>> 2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:108 SQL ERR [database
>> disk image is malformed]
>> 2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:223 SQL ERR [database
>> disk image is malformed]
>> 2015-05-07 13:16:03.047907 [CRIT] switch_core_sqldb.c:508 Failure to
>> connect to CORE_DB core!
>> 2015-05-07 13:16:03.047907 [ERR] switch_console.c:256 Database Error
>> 2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:108 SQL ERR [database
>> disk image is malformed]
>> 2015-05-07 13:16:03.047907 [ERR] switch_core_db.c:223 SQL ERR [database
>> disk image is malformed]
>>
>> -ERR Database error!
>>
>> 2015-05-07 13:16:03.047907 [CRIT] switch_core_sqldb.c:508 Failure to
>> connect to CORE_DB core!
>> freeswitch at srv-ovh-devel-in-000001>
>> freeswitch at srv-ovh-devel-in-000001> show registrations
>> 2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:108 SQL ERR [database
>> disk image is malformed]
>> 2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:223 SQL ERR [database
>> disk image is malformed]
>> 2015-05-07 13:16:18.047877 [CRIT] switch_core_sqldb.c:508 Failure to
>> connect to CORE_DB core!
>> 2015-05-07 13:16:18.047877 [ERR] switch_console.c:256 Database Error
>> 2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:108 SQL ERR [database
>> disk image is malformed]
>> 2015-05-07 13:16:18.047877 [ERR] switch_core_db.c:223 SQL ERR [database
>> disk image is malformed]
>> 2015-05-07 13:16:18.047877 [CRIT] switch_core_sqldb.c:508 Failure to
>> connect to CORE_DB core!
>>
>> -ERR Database error!
>>
>>
>>
>>
>> --
>>
>>
>>
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561
>> BB #79A8AFA2
>> @VMCibersys
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From victor.medina at cibersys.com  Fri May  8 00:56:37 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Thu, 7 May 2015 16:26:37 -0430
Subject: [Freeswitch-users] Testing WebRTC with latest master Git 1.5
Message-ID: <CACXsapNC95A86iG4ojhiQ6Wc_yje0VFb40LcDCsvZn7Kye1xtw@mail.gmail.com>

Hi guys!

I?ve been tying to test Freeswitch 1.5 (latest git) for WebRTC, but so far
I have been very unsuccessful.

I have a server, connected directly to internet, NO NAT on server side.


FS ----> INTERNET <--- NAT ---- CLIENTS

My vars.conf includes this:


  <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/>
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxxx.xxx.xxx.xxx"/>

  <!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/>
  <X-PRE-PROCESS cmd="set"
data="internal_ssl_dir=/opt/CloudVoice-vPBX/fs-20150506/certs/"/>

  <!-- External SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
  <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>


On my external profile I have this relevants lines...

 <param name="aggressive-nat-detection" value="true"/>
 <param name="apply-inbound-acl" value="domains"/>
 <param name="local-network-acl" value="localnet.auto"/>
 <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
 <param name="ext-sip-ip" value="$${external_sip_ip}"/>
 <param name="ws-binding"  value=":5066"/>
 <param name="wss-binding" value=":7443"/>


When doing some testing....

Calling to echo test

 2015-05-07 20:51:14.124955 [NOTICE] switch_channel.c:1075 New Channel
sofia/internal/1007 at webrtc.cibersys.com
[39b777d1-6e61-49ec-bfd0-6f7ca0bd0caf]
2015-05-07 20:51:14.384990 [INFO] mod_dialplan_xml.c:635 Processing test
<1007>->9196 in context default
2015-05-07 20:51:14.384990 [WARNING] switch_core_media.c:2791 NO candidate
ACL defined, Defaulting to wan.auto
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio
Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio
Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2953 setting remote
audio ice addr to 201.210.31.83:57630 based on candidate
2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2978 setting remote
rtcp audio addr to 201.210.31.83:57632 based on candidate
2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5345 Activating Audio
ICE
2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTP audio
ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5388 Activating RTCP
PORT 57632
2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5398 Activating RTCP
ICE
2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTCP audio
ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
2015-05-07 20:51:14.384990 [INFO] switch_rtp.c:3103 Activate RTP/RTCP audio
DTLS client
2015-05-07 20:51:14.384990 [NOTICE] sofia_media.c:92 Pre-Answer
sofia/internal/1007 at webrtc.cibersys.com!
2015-05-07 20:51:14.384990 [NOTICE] mod_dptools.c:1292 Channel
[sofia/internal/1007 at webrtc.cibersys.com] has been answered
2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2924 Changing audio DTLS
state from HANDSHAKE to SETUP
2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2832 audio Fingerprint
Verified.
2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3384 Activating Audio Secure
RTP SEND
2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3362 Activating Audio Secure
RTP RECV
2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2872 Changing audio DTLS
state from SETUP to READY
2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating audio
RTCP PORT 57632
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio
Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio
Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2953 setting remote
audio ice addr to 201.210.31.83:57630 based on candidate
2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2978 setting remote
rtcp audio addr to 201.210.31.83:57632 based on candidate
2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:2995 RE-Activating
audio ICE
2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTP audio
ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating audio
RTCP PORT 57632
2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3050 Activating audio
RTCP ICE
2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTCP audio
ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:5030 RE-SETTING video
DTLS
2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:878 sofia/internal/
1007 at webrtc.cibersys.com got stun binding response 487 Role Conflict
2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:889 Changing role to
CONTROLLED

No audio

When calling another ext... no audio en the webrtc side.

Can somebody help me by pointing out the right direction?

Ive been using FF and Chrome with sipML5



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From gmaruzz at gmail.com  Fri May  8 01:05:07 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 7 May 2015 23:05:07 +0200
Subject: [Freeswitch-users] Testing WebRTC with latest master Git 1.5
In-Reply-To: <CACXsapNC95A86iG4ojhiQ6Wc_yje0VFb40LcDCsvZn7Kye1xtw@mail.gmail.com>
References: <CACXsapNC95A86iG4ojhiQ6Wc_yje0VFb40LcDCsvZn7Kye1xtw@mail.gmail.com>
Message-ID: <CALXCt0p_GVoH-+4ouM-OBqCQdXVwbjOW=j=zBHJpSB_iLShz9A@mail.gmail.com>

Start by scratch, from a fresh debian Jessie install.

Then follow exactly, without changes, step by step, what is in the
"freeswitch 1.6" confluence page.

When you have it all working as in that page (contains instruction on
testing) only then you can go further.

Happy testing,

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 7, 2015 10:58 PM, "Victor Medina" <victor.medina at cibersys.com> wrote:

> Hi guys!
>
> I?ve been tying to test Freeswitch 1.5 (latest git) for WebRTC, but so far
> I have been very unsuccessful.
>
> I have a server, connected directly to internet, NO NAT on server side.
>
>
> FS ----> INTERNET <--- NAT ---- CLIENTS
>
> My vars.conf includes this:
>
>
>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/>
>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/>
>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxxx.xxx.xxx.xxx"/>
>
>   <!-- Internal SIP Profile -->
>   <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
>   <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
>   <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
>   <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/>
>   <X-PRE-PROCESS cmd="set"
> data="internal_ssl_dir=/opt/CloudVoice-vPBX/fs-20150506/certs/"/>
>
>   <!-- External SIP Profile -->
>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>
>
> On my external profile I have this relevants lines...
>
>  <param name="aggressive-nat-detection" value="true"/>
>  <param name="apply-inbound-acl" value="domains"/>
>  <param name="local-network-acl" value="localnet.auto"/>
>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>  <param name="ws-binding"  value=":5066"/>
>  <param name="wss-binding" value=":7443"/>
>
>
> When doing some testing....
>
> Calling to echo test
>
>  2015-05-07 20:51:14.124955 [NOTICE] switch_channel.c:1075 New Channel
> sofia/internal/1007 at webrtc.cibersys.com
> [39b777d1-6e61-49ec-bfd0-6f7ca0bd0caf]
> 2015-05-07 20:51:14.384990 [INFO] mod_dialplan_xml.c:635 Processing test
> <1007>->9196 in context default
> 2015-05-07 20:51:14.384990 [WARNING] switch_core_media.c:2791 NO candidate
> ACL defined, Defaulting to wan.auto
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio
> Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio
> Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2953 setting
> remote audio ice addr to 201.210.31.83:57630 based on candidate
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2978 setting
> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5345 Activating
> Audio ICE
> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTP audio
> ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5388 Activating RTCP
> PORT 57632
> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5398 Activating RTCP
> ICE
> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTCP
> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
> 2015-05-07 20:51:14.384990 [INFO] switch_rtp.c:3103 Activate RTP/RTCP
> audio DTLS client
> 2015-05-07 20:51:14.384990 [NOTICE] sofia_media.c:92 Pre-Answer
> sofia/internal/1007 at webrtc.cibersys.com!
> 2015-05-07 20:51:14.384990 [NOTICE] mod_dptools.c:1292 Channel
> [sofia/internal/1007 at webrtc.cibersys.com] has been answered
> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2924 Changing audio DTLS
> state from HANDSHAKE to SETUP
> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2832 audio Fingerprint
> Verified.
> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3384 Activating Audio
> Secure RTP SEND
> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3362 Activating Audio
> Secure RTP RECV
> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2872 Changing audio DTLS
> state from SETUP to READY
> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
> audio RTCP PORT 57632
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio
> Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio
> Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2953 setting
> remote audio ice addr to 201.210.31.83:57630 based on candidate
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2978 setting
> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:2995 RE-Activating
> audio ICE
> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTP audio
> ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
> audio RTCP PORT 57632
> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3050 Activating
> audio RTCP ICE
> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTCP
> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:5030 RE-SETTING
> video DTLS
> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:878 sofia/internal/
> 1007 at webrtc.cibersys.com got stun binding response 487 Role Conflict
> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:889 Changing role to
> CONTROLLED
>
> No audio
>
> When calling another ext... no audio en the webrtc side.
>
> Can somebody help me by pointing out the right direction?
>
> Ive been using FF and Chrome with sipML5
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vittico at gmail.com  Fri May  8 01:17:22 2015
From: vittico at gmail.com (Victor Medina)
Date: Thu, 7 May 2015 16:47:22 -0430
Subject: [Freeswitch-users] Testing WebRTC with latest master Git 1.5
In-Reply-To: <CALXCt0p_GVoH-+4ouM-OBqCQdXVwbjOW=j=zBHJpSB_iLShz9A@mail.gmail.com>
References: <CACXsapNC95A86iG4ojhiQ6Wc_yje0VFb40LcDCsvZn7Kye1xtw@mail.gmail.com>
	<CALXCt0p_GVoH-+4ouM-OBqCQdXVwbjOW=j=zBHJpSB_iLShz9A@mail.gmail.com>
Message-ID: <CAEA-bfGcR67WpBVyoZMD3hEkA2ShfhgsEUoEAYBvriLEG=qJkg@mail.gmail.com>

Hi Giovanni.

Thanks for the replay.

You mean this confluence page?
https://freeswitch.org/confluence/display/FREESWITCH/WebRTC


Sin mas a que hacer referencia,

Victor Medina

On Thu, May 7, 2015 at 4:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Start by scratch, from a fresh debian Jessie install.
>
> Then follow exactly, without changes, step by step, what is in the
> "freeswitch 1.6" confluence page.
>
> When you have it all working as in that page (contains instruction on
> testing) only then you can go further.
>
> Happy testing,
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On May 7, 2015 10:58 PM, "Victor Medina" <victor.medina at cibersys.com>
> wrote:
>
>> Hi guys!
>>
>> I?ve been tying to test Freeswitch 1.5 (latest git) for WebRTC, but so
>> far I have been very unsuccessful.
>>
>> I have a server, connected directly to internet, NO NAT on server side.
>>
>>
>> FS ----> INTERNET <--- NAT ---- CLIENTS
>>
>> My vars.conf includes this:
>>
>>
>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/>
>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/>
>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxxx.xxx.xxx.xxx"/>
>>
>>   <!-- Internal SIP Profile -->
>>   <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
>>   <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
>>   <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
>>   <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/>
>>   <X-PRE-PROCESS cmd="set"
>> data="internal_ssl_dir=/opt/CloudVoice-vPBX/fs-20150506/certs/"/>
>>
>>   <!-- External SIP Profile -->
>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>
>>
>> On my external profile I have this relevants lines...
>>
>>  <param name="aggressive-nat-detection" value="true"/>
>>  <param name="apply-inbound-acl" value="domains"/>
>>  <param name="local-network-acl" value="localnet.auto"/>
>>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>>  <param name="ws-binding"  value=":5066"/>
>>  <param name="wss-binding" value=":7443"/>
>>
>>
>> When doing some testing....
>>
>> Calling to echo test
>>
>>  2015-05-07 20:51:14.124955 [NOTICE] switch_channel.c:1075 New Channel
>> sofia/internal/1007 at webrtc.cibersys.com
>> [39b777d1-6e61-49ec-bfd0-6f7ca0bd0caf]
>> 2015-05-07 20:51:14.384990 [INFO] mod_dialplan_xml.c:635 Processing test
>> <1007>->9196 in context default
>> 2015-05-07 20:51:14.384990 [WARNING] switch_core_media.c:2791 NO
>> candidate ACL defined, Defaulting to wan.auto
>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio
>> Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio
>> Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2953 setting
>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2978 setting
>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5345 Activating
>> Audio ICE
>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTP
>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5388 Activating
>> RTCP PORT 57632
>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5398 Activating
>> RTCP ICE
>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTCP
>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>> 2015-05-07 20:51:14.384990 [INFO] switch_rtp.c:3103 Activate RTP/RTCP
>> audio DTLS client
>> 2015-05-07 20:51:14.384990 [NOTICE] sofia_media.c:92 Pre-Answer
>> sofia/internal/1007 at webrtc.cibersys.com!
>> 2015-05-07 20:51:14.384990 [NOTICE] mod_dptools.c:1292 Channel
>> [sofia/internal/1007 at webrtc.cibersys.com] has been answered
>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2924 Changing audio DTLS
>> state from HANDSHAKE to SETUP
>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2832 audio Fingerprint
>> Verified.
>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3384 Activating Audio
>> Secure RTP SEND
>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3362 Activating Audio
>> Secure RTP RECV
>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2872 Changing audio DTLS
>> state from SETUP to READY
>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>> audio RTCP PORT 57632
>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio
>> Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio
>> Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2953 setting
>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2978 setting
>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:2995 RE-Activating
>> audio ICE
>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTP
>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>> audio RTCP PORT 57632
>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3050 Activating
>> audio RTCP ICE
>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTCP
>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:5030 RE-SETTING
>> video DTLS
>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:878 sofia/internal/
>> 1007 at webrtc.cibersys.com got stun binding response 487 Role Conflict
>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:889 Changing role to
>> CONTROLLED
>>
>> No audio
>>
>> When calling another ext... no audio en the webrtc side.
>>
>> Can somebody help me by pointing out the right direction?
>>
>> Ive been using FF and Chrome with sipML5
>>
>>
>>
>> --
>>
>>
>>
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561
>> BB #79A8AFA2
>> @VMCibersys
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May  8 01:29:45 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 7 May 2015 23:29:45 +0200
Subject: [Freeswitch-users] Testing WebRTC with latest master Git 1.5
In-Reply-To: <CAEA-bfGcR67WpBVyoZMD3hEkA2ShfhgsEUoEAYBvriLEG=qJkg@mail.gmail.com>
References: <CACXsapNC95A86iG4ojhiQ6Wc_yje0VFb40LcDCsvZn7Kye1xtw@mail.gmail.com>
	<CALXCt0p_GVoH-+4ouM-OBqCQdXVwbjOW=j=zBHJpSB_iLShz9A@mail.gmail.com>
	<CAEA-bfGcR67WpBVyoZMD3hEkA2ShfhgsEUoEAYBvriLEG=qJkg@mail.gmail.com>
Message-ID: <CALXCt0pzkypo9HfX6aBgwhDOyVnWNG2nOJYC3KmiSu-veTB3BQ@mail.gmail.com>

No.
This one:
https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/7144556

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 7, 2015 11:18 PM, "Victor Medina" <vittico at gmail.com> wrote:

> Hi Giovanni.
>
> Thanks for the replay.
>
> You mean this confluence page?
> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>
>
> Sin mas a que hacer referencia,
>
> Victor Medina
>
> On Thu, May 7, 2015 at 4:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Start by scratch, from a fresh debian Jessie install.
>>
>> Then follow exactly, without changes, step by step, what is in the
>> "freeswitch 1.6" confluence page.
>>
>> When you have it all working as in that page (contains instruction on
>> testing) only then you can go further.
>>
>> Happy testing,
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On May 7, 2015 10:58 PM, "Victor Medina" <victor.medina at cibersys.com>
>> wrote:
>>
>>> Hi guys!
>>>
>>> I?ve been tying to test Freeswitch 1.5 (latest git) for WebRTC, but so
>>> far I have been very unsuccessful.
>>>
>>> I have a server, connected directly to internet, NO NAT on server side.
>>>
>>>
>>> FS ----> INTERNET <--- NAT ---- CLIENTS
>>>
>>> My vars.conf includes this:
>>>
>>>
>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/>
>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/>
>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxxx.xxx.xxx.xxx"/>
>>>
>>>   <!-- Internal SIP Profile -->
>>>   <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
>>>   <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
>>>   <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
>>>   <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/>
>>>   <X-PRE-PROCESS cmd="set"
>>> data="internal_ssl_dir=/opt/CloudVoice-vPBX/fs-20150506/certs/"/>
>>>
>>>   <!-- External SIP Profile -->
>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>
>>>
>>> On my external profile I have this relevants lines...
>>>
>>>  <param name="aggressive-nat-detection" value="true"/>
>>>  <param name="apply-inbound-acl" value="domains"/>
>>>  <param name="local-network-acl" value="localnet.auto"/>
>>>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>>>  <param name="ws-binding"  value=":5066"/>
>>>  <param name="wss-binding" value=":7443"/>
>>>
>>>
>>> When doing some testing....
>>>
>>> Calling to echo test
>>>
>>>  2015-05-07 20:51:14.124955 [NOTICE] switch_channel.c:1075 New Channel
>>> sofia/internal/1007 at webrtc.cibersys.com
>>> [39b777d1-6e61-49ec-bfd0-6f7ca0bd0caf]
>>> 2015-05-07 20:51:14.384990 [INFO] mod_dialplan_xml.c:635 Processing test
>>> <1007>->9196 in context default
>>> 2015-05-07 20:51:14.384990 [WARNING] switch_core_media.c:2791 NO
>>> candidate ACL defined, Defaulting to wan.auto
>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose
>>> audio Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose
>>> audio Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2953 setting
>>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2978 setting
>>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5345 Activating
>>> Audio ICE
>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTP
>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5388 Activating
>>> RTCP PORT 57632
>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5398 Activating
>>> RTCP ICE
>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTCP
>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>>> 2015-05-07 20:51:14.384990 [INFO] switch_rtp.c:3103 Activate RTP/RTCP
>>> audio DTLS client
>>> 2015-05-07 20:51:14.384990 [NOTICE] sofia_media.c:92 Pre-Answer
>>> sofia/internal/1007 at webrtc.cibersys.com!
>>> 2015-05-07 20:51:14.384990 [NOTICE] mod_dptools.c:1292 Channel
>>> [sofia/internal/1007 at webrtc.cibersys.com] has been answered
>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2924 Changing audio DTLS
>>> state from HANDSHAKE to SETUP
>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2832 audio Fingerprint
>>> Verified.
>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3384 Activating Audio
>>> Secure RTP SEND
>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3362 Activating Audio
>>> Secure RTP RECV
>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2872 Changing audio DTLS
>>> state from SETUP to READY
>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>>> audio RTCP PORT 57632
>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose
>>> audio Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose
>>> audio Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2953 setting
>>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2978 setting
>>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:2995 RE-Activating
>>> audio ICE
>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTP
>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>>> audio RTCP PORT 57632
>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3050 Activating
>>> audio RTCP ICE
>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTCP
>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:5030 RE-SETTING
>>> video DTLS
>>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:878 sofia/internal/
>>> 1007 at webrtc.cibersys.com got stun binding response 487 Role Conflict
>>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:889 Changing role to
>>> CONTROLLED
>>>
>>> No audio
>>>
>>> When calling another ext... no audio en the webrtc side.
>>>
>>> Can somebody help me by pointing out the right direction?
>>>
>>> Ive been using FF and Chrome with sipML5
>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>> V?ctor E. Medina M.
>>> Platform Architect / Chief Infrastructure
>>> +58424 291 4561
>>> BB #79A8AFA2
>>> @VMCibersys
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vittico at gmail.com  Fri May  8 01:59:53 2015
From: vittico at gmail.com (Victor Medina)
Date: Thu, 7 May 2015 17:29:53 -0430
Subject: [Freeswitch-users] Testing WebRTC with latest master Git 1.5
In-Reply-To: <CALXCt0pzkypo9HfX6aBgwhDOyVnWNG2nOJYC3KmiSu-veTB3BQ@mail.gmail.com>
References: <CACXsapNC95A86iG4ojhiQ6Wc_yje0VFb40LcDCsvZn7Kye1xtw@mail.gmail.com>
	<CALXCt0p_GVoH-+4ouM-OBqCQdXVwbjOW=j=zBHJpSB_iLShz9A@mail.gmail.com>
	<CAEA-bfGcR67WpBVyoZMD3hEkA2ShfhgsEUoEAYBvriLEG=qJkg@mail.gmail.com>
	<CALXCt0pzkypo9HfX6aBgwhDOyVnWNG2nOJYC3KmiSu-veTB3BQ@mail.gmail.com>
Message-ID: <CAEA-bfEZM7jQ-1fX-mXdJNxiy72F45O9WctEjUmcUtO-rd7ZGg@mail.gmail.com>

OK, very impressive....

But... I, for the moment ONLY want to have a _basic_ webrtc softhone
connected to FS. As I undestand this, is for Video Conferencing right?

Is this https://freeswitch.org/confluence/display/FREESWITCH/WebRTC already
deprecated? Not working anymore?


Sin mas a que hacer referencia,

Victor Medina

On Thu, May 7, 2015 at 4:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> No.
> This one:
> https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/7144556
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On May 7, 2015 11:18 PM, "Victor Medina" <vittico at gmail.com> wrote:
>
>> Hi Giovanni.
>>
>> Thanks for the replay.
>>
>> You mean this confluence page?
>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>>
>>
>> Sin mas a que hacer referencia,
>>
>> Victor Medina
>>
>> On Thu, May 7, 2015 at 4:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> Start by scratch, from a fresh debian Jessie install.
>>>
>>> Then follow exactly, without changes, step by step, what is in the
>>> "freeswitch 1.6" confluence page.
>>>
>>> When you have it all working as in that page (contains instruction on
>>> testing) only then you can go further.
>>>
>>> Happy testing,
>>>
>>> sent from my mobile,
>>> Giovanni Maruzzelli
>>> cell: +39 347 266 56 18
>>> On May 7, 2015 10:58 PM, "Victor Medina" <victor.medina at cibersys.com>
>>> wrote:
>>>
>>>> Hi guys!
>>>>
>>>> I?ve been tying to test Freeswitch 1.5 (latest git) for WebRTC, but so
>>>> far I have been very unsuccessful.
>>>>
>>>> I have a server, connected directly to internet, NO NAT on server side.
>>>>
>>>>
>>>> FS ----> INTERNET <--- NAT ---- CLIENTS
>>>>
>>>> My vars.conf includes this:
>>>>
>>>>
>>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/>
>>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/>
>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxxx.xxx.xxx.xxx"/>
>>>>
>>>>   <!-- Internal SIP Profile -->
>>>>   <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
>>>>   <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
>>>>   <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
>>>>   <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/>
>>>>   <X-PRE-PROCESS cmd="set"
>>>> data="internal_ssl_dir=/opt/CloudVoice-vPBX/fs-20150506/certs/"/>
>>>>
>>>>   <!-- External SIP Profile -->
>>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
>>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
>>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>>
>>>>
>>>> On my external profile I have this relevants lines...
>>>>
>>>>  <param name="aggressive-nat-detection" value="true"/>
>>>>  <param name="apply-inbound-acl" value="domains"/>
>>>>  <param name="local-network-acl" value="localnet.auto"/>
>>>>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>>>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>>>>  <param name="ws-binding"  value=":5066"/>
>>>>  <param name="wss-binding" value=":7443"/>
>>>>
>>>>
>>>> When doing some testing....
>>>>
>>>> Calling to echo test
>>>>
>>>>  2015-05-07 20:51:14.124955 [NOTICE] switch_channel.c:1075 New Channel
>>>> sofia/internal/1007 at webrtc.cibersys.com
>>>> [39b777d1-6e61-49ec-bfd0-6f7ca0bd0caf]
>>>> 2015-05-07 20:51:14.384990 [INFO] mod_dialplan_xml.c:635 Processing
>>>> test <1007>->9196 in context default
>>>> 2015-05-07 20:51:14.384990 [WARNING] switch_core_media.c:2791 NO
>>>> candidate ACL defined, Defaulting to wan.auto
>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose
>>>> audio Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose
>>>> audio Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2953 setting
>>>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2978 setting
>>>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5345 Activating
>>>> Audio ICE
>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTP
>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5388 Activating
>>>> RTCP PORT 57632
>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5398 Activating
>>>> RTCP ICE
>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTCP
>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>>>> 2015-05-07 20:51:14.384990 [INFO] switch_rtp.c:3103 Activate RTP/RTCP
>>>> audio DTLS client
>>>> 2015-05-07 20:51:14.384990 [NOTICE] sofia_media.c:92 Pre-Answer
>>>> sofia/internal/1007 at webrtc.cibersys.com!
>>>> 2015-05-07 20:51:14.384990 [NOTICE] mod_dptools.c:1292 Channel
>>>> [sofia/internal/1007 at webrtc.cibersys.com] has been answered
>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2924 Changing audio DTLS
>>>> state from HANDSHAKE to SETUP
>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2832 audio Fingerprint
>>>> Verified.
>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3384 Activating Audio
>>>> Secure RTP SEND
>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3362 Activating Audio
>>>> Secure RTP RECV
>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2872 Changing audio DTLS
>>>> state from SETUP to READY
>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>>>> audio RTCP PORT 57632
>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose
>>>> audio Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose
>>>> audio Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2953 setting
>>>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2978 setting
>>>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:2995
>>>> RE-Activating audio ICE
>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTP
>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>>>> audio RTCP PORT 57632
>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3050 Activating
>>>> audio RTCP ICE
>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTCP
>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:5030 RE-SETTING
>>>> video DTLS
>>>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:878 sofia/internal/
>>>> 1007 at webrtc.cibersys.com got stun binding response 487 Role Conflict
>>>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:889 Changing role to
>>>> CONTROLLED
>>>>
>>>> No audio
>>>>
>>>> When calling another ext... no audio en the webrtc side.
>>>>
>>>> Can somebody help me by pointing out the right direction?
>>>>
>>>> Ive been using FF and Chrome with sipML5
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>>
>>>>
>>>> V?ctor E. Medina M.
>>>> Platform Architect / Chief Infrastructure
>>>> +58424 291 4561
>>>> BB #79A8AFA2
>>>> @VMCibersys
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri May  8 02:39:05 2015
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 May 2015 17:39:05 -0500
Subject: [Freeswitch-users] Testing WebRTC with latest master Git 1.5
In-Reply-To: <CAEA-bfEZM7jQ-1fX-mXdJNxiy72F45O9WctEjUmcUtO-rd7ZGg@mail.gmail.com>
References: <CACXsapNC95A86iG4ojhiQ6Wc_yje0VFb40LcDCsvZn7Kye1xtw@mail.gmail.com>
	<CALXCt0p_GVoH-+4ouM-OBqCQdXVwbjOW=j=zBHJpSB_iLShz9A@mail.gmail.com>
	<CAEA-bfGcR67WpBVyoZMD3hEkA2ShfhgsEUoEAYBvriLEG=qJkg@mail.gmail.com>
	<CALXCt0pzkypo9HfX6aBgwhDOyVnWNG2nOJYC3KmiSu-veTB3BQ@mail.gmail.com>
	<CAEA-bfEZM7jQ-1fX-mXdJNxiy72F45O9WctEjUmcUtO-rd7ZGg@mail.gmail.com>
Message-ID: <CAEJMVkC_+4=4GuxoqiVU+Xq7RJVDK+xjp7ET3e4BGES8AnjmMw@mail.gmail.com>

It does both.  The other should work too, but there are probably some
things you're hitting we are unaware of.

On Thu, May 7, 2015 at 4:59 PM, Victor Medina <vittico at gmail.com> wrote:

> OK, very impressive....
>
> But... I, for the moment ONLY want to have a _basic_ webrtc softhone
> connected to FS. As I undestand this, is for Video Conferencing right?
>
> Is this https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
> already deprecated? Not working anymore?
>
>
> Sin mas a que hacer referencia,
>
> Victor Medina
>
> On Thu, May 7, 2015 at 4:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> No.
>> This one:
>> https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/7144556
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On May 7, 2015 11:18 PM, "Victor Medina" <vittico at gmail.com> wrote:
>>
>>> Hi Giovanni.
>>>
>>> Thanks for the replay.
>>>
>>> You mean this confluence page?
>>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>>>
>>>
>>> Sin mas a que hacer referencia,
>>>
>>> Victor Medina
>>>
>>> On Thu, May 7, 2015 at 4:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> Start by scratch, from a fresh debian Jessie install.
>>>>
>>>> Then follow exactly, without changes, step by step, what is in the
>>>> "freeswitch 1.6" confluence page.
>>>>
>>>> When you have it all working as in that page (contains instruction on
>>>> testing) only then you can go further.
>>>>
>>>> Happy testing,
>>>>
>>>> sent from my mobile,
>>>> Giovanni Maruzzelli
>>>> cell: +39 347 266 56 18
>>>> On May 7, 2015 10:58 PM, "Victor Medina" <victor.medina at cibersys.com>
>>>> wrote:
>>>>
>>>>> Hi guys!
>>>>>
>>>>> I?ve been tying to test Freeswitch 1.5 (latest git) for WebRTC, but so
>>>>> far I have been very unsuccessful.
>>>>>
>>>>> I have a server, connected directly to internet, NO NAT on server side.
>>>>>
>>>>>
>>>>> FS ----> INTERNET <--- NAT ---- CLIENTS
>>>>>
>>>>> My vars.conf includes this:
>>>>>
>>>>>
>>>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxxx.xxx.xxx.xxx"/>
>>>>>
>>>>>   <!-- Internal SIP Profile -->
>>>>>   <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/>
>>>>>   <X-PRE-PROCESS cmd="set"
>>>>> data="internal_ssl_dir=/opt/CloudVoice-vPBX/fs-20150506/certs/"/>
>>>>>
>>>>>   <!-- External SIP Profile -->
>>>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>>>
>>>>>
>>>>> On my external profile I have this relevants lines...
>>>>>
>>>>>  <param name="aggressive-nat-detection" value="true"/>
>>>>>  <param name="apply-inbound-acl" value="domains"/>
>>>>>  <param name="local-network-acl" value="localnet.auto"/>
>>>>>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>>>>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>>>>>  <param name="ws-binding"  value=":5066"/>
>>>>>  <param name="wss-binding" value=":7443"/>
>>>>>
>>>>>
>>>>> When doing some testing....
>>>>>
>>>>> Calling to echo test
>>>>>
>>>>>  2015-05-07 20:51:14.124955 [NOTICE] switch_channel.c:1075 New Channel
>>>>> sofia/internal/1007 at webrtc.cibersys.com
>>>>> [39b777d1-6e61-49ec-bfd0-6f7ca0bd0caf]
>>>>> 2015-05-07 20:51:14.384990 [INFO] mod_dialplan_xml.c:635 Processing
>>>>> test <1007>->9196 in context default
>>>>> 2015-05-07 20:51:14.384990 [WARNING] switch_core_media.c:2791 NO
>>>>> candidate ACL defined, Defaulting to wan.auto
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose
>>>>> audio Candidate cid: 1 proto: UDP type: srflx addr:
>>>>> 201.210.31.83:57630
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose
>>>>> audio Candidate cid: 2 proto: UDP type: srflx addr:
>>>>> 201.210.31.83:57632
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2953 setting
>>>>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2978 setting
>>>>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5345 Activating
>>>>> Audio ICE
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTP
>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5388 Activating
>>>>> RTCP PORT 57632
>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5398 Activating
>>>>> RTCP ICE
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTCP
>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_rtp.c:3103 Activate RTP/RTCP
>>>>> audio DTLS client
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] sofia_media.c:92 Pre-Answer
>>>>> sofia/internal/1007 at webrtc.cibersys.com!
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] mod_dptools.c:1292 Channel
>>>>> [sofia/internal/1007 at webrtc.cibersys.com] has been answered
>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2924 Changing audio
>>>>> DTLS state from HANDSHAKE to SETUP
>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2832 audio Fingerprint
>>>>> Verified.
>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3384 Activating Audio
>>>>> Secure RTP SEND
>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3362 Activating Audio
>>>>> Secure RTP RECV
>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2872 Changing audio
>>>>> DTLS state from SETUP to READY
>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>>>>> audio RTCP PORT 57632
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose
>>>>> audio Candidate cid: 1 proto: UDP type: srflx addr:
>>>>> 201.210.31.83:57630
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose
>>>>> audio Candidate cid: 2 proto: UDP type: srflx addr:
>>>>> 201.210.31.83:57632
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2953 setting
>>>>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2978 setting
>>>>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:2995
>>>>> RE-Activating audio ICE
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTP
>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>>>>> audio RTCP PORT 57632
>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3050 Activating
>>>>> audio RTCP ICE
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTCP
>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:5030 RE-SETTING
>>>>> video DTLS
>>>>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:878 sofia/internal/
>>>>> 1007 at webrtc.cibersys.com got stun binding response 487 Role Conflict
>>>>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:889 Changing role to
>>>>> CONTROLLED
>>>>>
>>>>> No audio
>>>>>
>>>>> When calling another ext... no audio en the webrtc side.
>>>>>
>>>>> Can somebody help me by pointing out the right direction?
>>>>>
>>>>> Ive been using FF and Chrome with sipML5
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>>
>>>>>
>>>>> V?ctor E. Medina M.
>>>>> Platform Architect / Chief Infrastructure
>>>>> +58424 291 4561
>>>>> BB #79A8AFA2
>>>>> @VMCibersys
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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*
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From gmaruzz at gmail.com  Fri May  8 02:49:22 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 8 May 2015 00:49:22 +0200
Subject: [Freeswitch-users] Testing WebRTC with latest master Git 1.5
In-Reply-To: <CAEA-bfEZM7jQ-1fX-mXdJNxiy72F45O9WctEjUmcUtO-rd7ZGg@mail.gmail.com>
References: <CACXsapNC95A86iG4ojhiQ6Wc_yje0VFb40LcDCsvZn7Kye1xtw@mail.gmail.com>
	<CALXCt0p_GVoH-+4ouM-OBqCQdXVwbjOW=j=zBHJpSB_iLShz9A@mail.gmail.com>
	<CAEA-bfGcR67WpBVyoZMD3hEkA2ShfhgsEUoEAYBvriLEG=qJkg@mail.gmail.com>
	<CALXCt0pzkypo9HfX6aBgwhDOyVnWNG2nOJYC3KmiSu-veTB3BQ@mail.gmail.com>
	<CAEA-bfEZM7jQ-1fX-mXdJNxiy72F45O9WctEjUmcUtO-rd7ZGg@mail.gmail.com>
Message-ID: <CALXCt0oB+hCot0sqrXhV_yFzu5Fnk8NPKXwgj49=QDmvr3r04Q@mail.gmail.com>

Go with the one I told you. It has webrrc too, and the docs are completely
updated, and we all are using and testing that one.
Eg: is the path of less resistance ;)
Then you can use only what part you want (webrtc), but at least you sure if
you follow the steps it will work without fiddling.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 8, 2015 12:01 AM, "Victor Medina" <vittico at gmail.com> wrote:

> OK, very impressive....
>
> But... I, for the moment ONLY want to have a _basic_ webrtc softhone
> connected to FS. As I undestand this, is for Video Conferencing right?
>
> Is this https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
> already deprecated? Not working anymore?
>
>
> Sin mas a que hacer referencia,
>
> Victor Medina
>
> On Thu, May 7, 2015 at 4:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> No.
>> This one:
>> https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/7144556
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On May 7, 2015 11:18 PM, "Victor Medina" <vittico at gmail.com> wrote:
>>
>>> Hi Giovanni.
>>>
>>> Thanks for the replay.
>>>
>>> You mean this confluence page?
>>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>>>
>>>
>>> Sin mas a que hacer referencia,
>>>
>>> Victor Medina
>>>
>>> On Thu, May 7, 2015 at 4:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> Start by scratch, from a fresh debian Jessie install.
>>>>
>>>> Then follow exactly, without changes, step by step, what is in the
>>>> "freeswitch 1.6" confluence page.
>>>>
>>>> When you have it all working as in that page (contains instruction on
>>>> testing) only then you can go further.
>>>>
>>>> Happy testing,
>>>>
>>>> sent from my mobile,
>>>> Giovanni Maruzzelli
>>>> cell: +39 347 266 56 18
>>>> On May 7, 2015 10:58 PM, "Victor Medina" <victor.medina at cibersys.com>
>>>> wrote:
>>>>
>>>>> Hi guys!
>>>>>
>>>>> I?ve been tying to test Freeswitch 1.5 (latest git) for WebRTC, but so
>>>>> far I have been very unsuccessful.
>>>>>
>>>>> I have a server, connected directly to internet, NO NAT on server side.
>>>>>
>>>>>
>>>>> FS ----> INTERNET <--- NAT ---- CLIENTS
>>>>>
>>>>> My vars.conf includes this:
>>>>>
>>>>>
>>>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxxx.xxx.xxx.xxx"/>
>>>>>
>>>>>   <!-- Internal SIP Profile -->
>>>>>   <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/>
>>>>>   <X-PRE-PROCESS cmd="set"
>>>>> data="internal_ssl_dir=/opt/CloudVoice-vPBX/fs-20150506/certs/"/>
>>>>>
>>>>>   <!-- External SIP Profile -->
>>>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>>>
>>>>>
>>>>> On my external profile I have this relevants lines...
>>>>>
>>>>>  <param name="aggressive-nat-detection" value="true"/>
>>>>>  <param name="apply-inbound-acl" value="domains"/>
>>>>>  <param name="local-network-acl" value="localnet.auto"/>
>>>>>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>>>>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>>>>>  <param name="ws-binding"  value=":5066"/>
>>>>>  <param name="wss-binding" value=":7443"/>
>>>>>
>>>>>
>>>>> When doing some testing....
>>>>>
>>>>> Calling to echo test
>>>>>
>>>>>  2015-05-07 20:51:14.124955 [NOTICE] switch_channel.c:1075 New Channel
>>>>> sofia/internal/1007 at webrtc.cibersys.com
>>>>> [39b777d1-6e61-49ec-bfd0-6f7ca0bd0caf]
>>>>> 2015-05-07 20:51:14.384990 [INFO] mod_dialplan_xml.c:635 Processing
>>>>> test <1007>->9196 in context default
>>>>> 2015-05-07 20:51:14.384990 [WARNING] switch_core_media.c:2791 NO
>>>>> candidate ACL defined, Defaulting to wan.auto
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose
>>>>> audio Candidate cid: 1 proto: UDP type: srflx addr:
>>>>> 201.210.31.83:57630
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose
>>>>> audio Candidate cid: 2 proto: UDP type: srflx addr:
>>>>> 201.210.31.83:57632
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2953 setting
>>>>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2978 setting
>>>>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5345 Activating
>>>>> Audio ICE
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTP
>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5388 Activating
>>>>> RTCP PORT 57632
>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5398 Activating
>>>>> RTCP ICE
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTCP
>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_rtp.c:3103 Activate RTP/RTCP
>>>>> audio DTLS client
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] sofia_media.c:92 Pre-Answer
>>>>> sofia/internal/1007 at webrtc.cibersys.com!
>>>>> 2015-05-07 20:51:14.384990 [NOTICE] mod_dptools.c:1292 Channel
>>>>> [sofia/internal/1007 at webrtc.cibersys.com] has been answered
>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2924 Changing audio
>>>>> DTLS state from HANDSHAKE to SETUP
>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2832 audio Fingerprint
>>>>> Verified.
>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3384 Activating Audio
>>>>> Secure RTP SEND
>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3362 Activating Audio
>>>>> Secure RTP RECV
>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2872 Changing audio
>>>>> DTLS state from SETUP to READY
>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>>>>> audio RTCP PORT 57632
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose
>>>>> audio Candidate cid: 1 proto: UDP type: srflx addr:
>>>>> 201.210.31.83:57630
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose
>>>>> audio Candidate cid: 2 proto: UDP type: srflx addr:
>>>>> 201.210.31.83:57632
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2953 setting
>>>>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2978 setting
>>>>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:2995
>>>>> RE-Activating audio ICE
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTP
>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>>>>> audio RTCP PORT 57632
>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3050 Activating
>>>>> audio RTCP ICE
>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTCP
>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:5030 RE-SETTING
>>>>> video DTLS
>>>>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:878 sofia/internal/
>>>>> 1007 at webrtc.cibersys.com got stun binding response 487 Role Conflict
>>>>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:889 Changing role to
>>>>> CONTROLLED
>>>>>
>>>>> No audio
>>>>>
>>>>> When calling another ext... no audio en the webrtc side.
>>>>>
>>>>> Can somebody help me by pointing out the right direction?
>>>>>
>>>>> Ive been using FF and Chrome with sipML5
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>>
>>>>>
>>>>> V?ctor E. Medina M.
>>>>> Platform Architect / Chief Infrastructure
>>>>> +58424 291 4561
>>>>> BB #79A8AFA2
>>>>> @VMCibersys
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From adam.ben.ayoun1 at gmail.com  Fri May  8 12:27:42 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Fri, 8 May 2015 11:27:42 +0300
Subject: [Freeswitch-users] WebRTC slow connection time
Message-ID: <CADNqFCky4iMXF-5vukKcqmT8=Q+anVHdtn+0zueV+4yJ3GUC6g@mail.gmail.com>

Hi guys,

I am using Freeswitch from master (month old). I tried several WebRTC
clients (my own test app on Android and sipml5) and on certain WiFi's I am
getting connection time up to 35 seconds when using STUN and TURN servers.
Also, when I am using TURN and STUN, Freeswitch chooses the TURN candidate
although as far as connectivity, when I am not using STUN and TURN I am
connecting successfully after ~2 seconds (so no real need for TURN).
Attaching the SIP trace from Freeswitch:

nua.c:575 nua_set_params() nua: nua_set_params: entering
nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
nua.c:575 nua_set_params() nua: nua_set_params: entering
nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
nua.c:575 nua_set_params() nua: nua_set_params: entering
nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
nua.c:575 nua_set_params() nua: nua_set_params: entering
nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
nua.c:575 nua_set_params() nua: nua_set_params: entering
nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
called
nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fe18c001930, ...)
called
nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
called
nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fe184001930, ...)
called
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fe188001930, ...)
called
nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
0x7fe17c089c50 from (ws/82.166.84.247:53645) has 2598 bytes, veclen = 1
recv 2598 bytes from ws/[82.166.84.247]:53645 at 08:21:30.468866:
   ------------------------------------------------------------------------
   INVITE sip:991234 at aaaa SIP/2.0
   Via: SIP/2.0/WS
df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport
   From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
   To: <sip:991234 at aaaa>
   Contact: "asdasda"<sip:1000 at df7jal23ls0d.invalid
;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
   Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
   CSeq: 28092 INVITE
   Content-Type: application/sdp
   Content-Length: 2039
   Max-Forwards: 70
   User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
   Organization: Doubango Telecom

   v=0
   o=- 3843427479443760600 2 IN IP4 127.0.0.1
   s=Doubango Telecom - chrome
   t=0 0
   a=group:BUNDLE audio
   a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
   m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
   c=IN IP4 130.211.78.35
   a=rtcp:58209 IN IP4 130.211.78.35
   a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
generation 0
   a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
generation 0
   a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx
raddr 10.0.0.10 rport 63888 generation 0
   a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx
raddr 10.0.0.10 rport 63888 generation 0
   a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype
active generation 0
   a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype
active generation 0
   a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay
raddr 82.166.84.247 rport 53792 generation 0
   a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay
raddr 82.166.84.247 rport 53792 generation 0
   a=ice-ufrag:EDo5kr308/TXhitG
   a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
   a=ice-options:google-ice
   a=fingerprint:sha-256
2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
   a=setup:actpass
   a=mid:audio
   a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
   a=sendrecv
   a=rtcp-mux
   a=rtpmap:111 opus/48000/2
   a=fmtp:111 minptime=10; useinbandfec=1
   a=rtpmap:103 ISAC/16000
   a=rtpmap:104 ISAC/32000
   a=rtpmap:9 G722/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:106 CN/32000
   a=rtpmap:105 CN/16000
   a=rtpmap:13 CN/8000
   a=rtpmap:126 telephone-event/8000
   a=maxptime:60
   a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
   a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
b8082781-4098-4ac5-8c96-2848d2e9e7e4
   a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
   a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
   ------------------------------------------------------------------------
tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
0x7fe17c089c50 (2598 bytes) from ws/82.166.84.247:53645/sip next=(nil)
nta.c:2880 agent_recv_request() nta: received INVITE sip:991234 at aaaa
SIP/2.0 (CSeq 28092)
nta.c:3174 agent_check_request_via() nta: Via check: received=82.166.84.247
nta.c:3085 agent_recv_request() nta: INVITE (28092) going to a default leg
nta.c:1350 set_timeout() nta: timer set to 2000 ms
nua_server.c:102 nua_stack_process_request() nua:
nua_stack_process_request: entering
nua_stack.c:899 nh_create() nua: nh_create: entering
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fe17c001930, 0x7fe17c001130,
0x7fe17c0cf210) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
called
nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fe17c04a2e0)
soa.c:1302 soa_init_offer_answer()
soa_init_offer_answer(static::0x7fe17c0bf4d0) called
soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fe17c0bf4d0,
(nil), 0x7fe17c0be55f, 2039) called
nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fe17c0cf210): adding session
usage
tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
82.166.84.247:53645
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
0x7fe17c0d4d90 0x7fe17c0190a0 140 (140)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
0x7fe17c0d4d90 0x7fe17c0be3ab 86 (86)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
0x7fe17c0d4d90 0x7fe17c01912c 101 (101)
tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 394 bytes of 394 to
ws/82.166.84.247:53645
tport.c:3492 tport_send_msg() tport_vsend returned 394
send 394 bytes to ws/[82.166.84.247]:53645 at 08:21:30.469343:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/WS
df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
   From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
   To: <sip:991234 at aaaa>
   Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
   CSeq: 28092 INVITE
   User-Agent:
FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
   Content-Length: 0

   ------------------------------------------------------------------------
tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (28092)
nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_invite 100
Trying
nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
state changed: init -> received, received offer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fe17c0bf4d0,
[0x7fe1aa7b35d8], [0x7fe1aa7b35e0], [(nil)]) called
nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 100
Trying
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2015-05-08 08:21:30.467711 [NOTICE] switch_channel.c:1075 New Channel
sofia/internal/1000 at xxx.xxx.xxx.xxx [3ac3c622-f55b-11e4-a447-7d37723461ed]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal
sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal
sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_NEW
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8848
sofia/internal/1000 at xxx.xxx.xxx.xxx receiving invite from
82.166.84.247:53645 version: 1.5.15b git b01352c 2015-04-23 04:33:08Z 64bit
2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8960 IP 82.166.84.247 Approved
by acl "domains[]". Access Granted.
nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil)
2015-05-08 08:21:30.467711 [DEBUG] sofia.c:10113 Setting NAT mode based on
via received
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [received][100]
2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6637 Remote SDP:
v=0
o=- 3843427479443760600 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 130.211.78.35
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=rtcp:58209 IN IP4 130.211.78.35
a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host generation
0
a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host generation
0
a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx raddr
10.0.0.10 rport 63888 generation 0
a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx raddr
10.0.0.10 rport 63888 generation 0
a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype active
generation 0
a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype active
generation 0
a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay raddr
82.166.84.247 rport 53792 generation 0
a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay raddr
82.166.84.247 rport 53792 generation 0
a=ice-ufrag:EDo5kr308/TXhitG
a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
a=ice-options:google-ice
a=fingerprint:sha-256
2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=rtcp-mux
a=maxptime:60
a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
b8082781-4098-4ac5-8c96-2848d2e9e7e4
a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4

2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6903
(sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_NEW -> CS_INIT
2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal
sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:491
(sofia/internal/1000 at xxx.xxx.xxx.xxx) State NEW
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_INIT
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512
(sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT
2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:87
sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA INIT
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:40
sofia/internal/1000 at xxx.xxx.xxx.xxx Standard INIT
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:48
(sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_INIT -> CS_ROUTING
2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal
sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512
(sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT going to sleep
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_ROUTING
2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:2204
(sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change DOWN -> RINGING
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528
(sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING
2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:123
sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA ROUTING
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:166
sofia/internal/1000 at xxx.xxx.xxx.xxx Standard ROUTING
2015-05-08 08:21:30.467711 [INFO] mod_dialplan_xml.c:635 Processing asdasda
<1000>->991234 in context public
Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
[public->cdquality_conferences_with_api] continue=false
Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (FAIL)
[cdquality_conferences_with_api] destination_number(991234) =~
/^(75\d{4,36})$/ break=on-false
Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
[public->test_conferences] continue=false
Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (PASS)
[test_conferences] destination_number(991234) =~ /^(99\d{4,36})$/
break=on-false
Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action answer()
Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action
conference(991234-${domain_name}@test)
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:216
(sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_ROUTING -> CS_EXECUTE
2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal
sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528
(sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING going to sleep
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_EXECUTE
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:535
(sofia/internal/1000 at xxx.xxx.xxx.xxx) State EXECUTE
2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:178
sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA EXECUTE
2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:258
sofia/internal/1000 at xxx.xxx.xxx.xxx Standard EXECUTE
EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx answer()
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [opus:111:48000:60:0:1]/[opus:116:48000:20:0:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
Sticking with opus at 48000h@20i
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
Compare [opus:116:48000:20:0:1] ++++ is saved as a match
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [opus:111:48000:60:0:1]/[PCMU:0:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [opus:111:48000:60:0:1]/[PCMA:8:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [ISAC:103:16000:30:32000:1]/[opus:116:48000:20:0:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [ISAC:103:16000:30:32000:1]/[PCMU:0:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [ISAC:103:16000:30:32000:1]/[PCMA:8:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [ISAC:104:32000:30:32000:1]/[opus:116:48000:20:0:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [ISAC:104:32000:30:32000:1]/[PCMU:0:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [ISAC:104:32000:30:32000:1]/[PCMA:8:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [G722:9:8000:60:64000:1]/[opus:116:48000:20:0:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [G722:9:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [G722:9:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [PCMU:0:8000:60:64000:1]/[opus:116:48000:20:0:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [PCMU:0:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
Sticking with PCMU at 8000h@20i
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [PCMU:0:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [PCMA:8:8000:60:64000:1]/[opus:116:48000:20:0:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [PCMA:8:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [PCMA:8:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
Sticking with PCMA at 8000h@20i
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [CN:105:16000:60:0:1]/[opus:116:48000:20:0:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [CN:105:16000:60:0:1]/[PCMU:0:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [CN:105:16000:60:0:1]/[PCMA:8:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [CN:13:8000:60:0:1]/[opus:116:48000:20:0:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [CN:13:8000:60:0:1]/[PCMU:0:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
Compare [CN:13:8000:60:0:1]/[PCMA:8:8000:20:64000:1]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3588 Set
telephone-event payload to 126
2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set bitrate
to local settings [-1000bps]
2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set bitrate
to local settings [-1000bps]
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2507 Set Codec
sofia/internal/1000 at xxx.xxx.xxx.xxx opus/48000 20 ms 960 samples 0 bits 1
channels
2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:111
sofia/internal/1000 at xxx.xxx.xxx.xxx Original read codec set to opus:116
2015-05-08 08:21:30.467711 [WARNING] switch_core_media.c:2791 NO candidate
ACL defined, Defaulting to wan.auto
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose audio
Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose audio
Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2953 setting remote
audio ice addr to 82.166.84.247:63888 based on candidate
2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2978 setting remote
rtcp audio addr to 82.166.84.247:63888 based on candidate
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3935 Set 2833 dtmf
send/recv payload to 126
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5171 AUDIO RTP
[sofia/internal/1000 at xxx.xxx.xxx.xxx] 172.30.0.219 port 19864 ->
82.166.84.247 port 63888 codec: 111 ms: 20
2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3559 Starting timer [soft]
960 bytes per 20ms
2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5345 Activating Audio
ICE
2015-05-08 08:21:30.467711 [NOTICE] switch_rtp.c:4009 Activating RTP audio
ICE: EDo5kr308/TXhitG:v9QogVGvZ8jGwYIi 82.166.84.247:63888
2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5388 Activating RTCP
PORT 63888
2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3909 RTCP send rate is:
10000 and packet rate is: 20000 Remote Port: 63888
2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:2349 Setting RTCP remote
addr to 82.166.84.247:63888
2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5396 Skipping RTCP
ICE (Same as RTP)
2015-05-08 08:21:30.467711 [INFO] switch_rtp.c:3101 Activate RTP/RTCP audio
DTLS client
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5469 Set 2833 dtmf
send payload to 126
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5475 Set 2833 dtmf
receive payload to 126
2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5503 Set comfort
noise payload to 106
2015-05-08 08:21:30.467711 [NOTICE] sofia_media.c:92 Pre-Answer
sofia/internal/1000 at xxx.xxx.xxx.xxx!
2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3419
(sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change RINGING -> EARLY
2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:780 Local SDP
sofia/internal/1000 at xxx.xxx.xxx.xxx:
v=0
o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
s=FreeSWITCH
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1; minptime=10
a=rtpmap:126 telephone-event/8000
a=rtpmap:106 CN/8000
a=ptime:20
a=sendrecv
a=fingerprint:sha-256
83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
a=rtcp-mux
a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
a=ice-ufrag:v9QogVGvZ8jGwYIi
a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
generation 0

nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:529 nua_signal() nua(0x7fe17c0cf210): sent signal r_respond
2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send signal
sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
nua_stack.c:573 nua_stack_signal() nua(0x7fe17c0cf210): recv signal
r_respond 200 OK
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
called
soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fe17c0bf4d0,
(nil), 0x7fe190041d31, -1) called
soa.c:890 soa_set_capability_sdp()
soa_set_capability_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe190041d31, -1)
called
2015-05-08 08:21:30.467711 [NOTICE] mod_dptools.c:1292 Channel
[sofia/internal/1000 at xxx.xxx.xxx.xxx] has been answered
nua_session.c:2320 nua_invite_server_respond() nua:
nua_invite_server_respond: entering
soa.c:1515 soa_generate_answer()
soa_generate_answer(static::0x7fe17c0bf4d0) called
soa_static.c:1146 offer_answer_step()
soa_static_offer_answer_action(0x7fe17c0bf4d0, soa_generate_answer): called
soa_static.c:1187 offer_answer_step() soa_static(0x7fe17c0bf4d0,
soa_generate_answer): generating local description
soa_static.c:1228 offer_answer_step() soa_static(0x7fe17c0bf4d0,
soa_generate_answer): upgrade with remote description
soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fe1aa7b1a30,
0x7fe17c026ac0, ""): called
soa_static.c:1444 offer_answer_step() soa_static(0x7fe17c0bf4d0,
soa_generate_answer): storing local description
soa.c:1730 soa_activate() soa_activate(static::0x7fe17c0bf4d0, (nil)) called
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe17c0bf4d0,
[(nil)], [0x7fe1aa7b3b58], [0x7fe1aa7b3b54]) called
tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
82.166.84.247:53645
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
0x7fe17c0d4d90 0x7fe17c0c5560 136 (136)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
0x7fe17c0d4d90 0x7fe17c0be3ab 63 (63)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
0x7fe17c0d4d90 0x7fe17c0c55e8 41 (41)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
0x7fe17c0d4d90 0x7fe17c0c5611 631 (631)
tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
0x7fe17c0d4d90 0x7fe17c0d3020 864 (864)
tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 1802 bytes of 1802
to ws/82.166.84.247:53645
tport.c:3492 tport_send_msg() tport_vsend returned 1802
send 1802 bytes to ws/[82.166.84.247]:53645 at 08:21:30.479618:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/WS
df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
   From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
   To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
   Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
   CSeq: 28092 INVITE
   Contact: <sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp>
   User-Agent:
FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
   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: 864
   Remote-Party-ID: "991234" <sip:991234 at aaaa
>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
   s=FreeSWITCH
   c=IN IP4 xxx.xxx.xxx.xxx
   t=0 0
   a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
   m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
   a=rtpmap:111 opus/48000/2
   a=fmtp:111 useinbandfec=1; minptime=10
   a=rtpmap:126 telephone-event/8000
   a=rtpmap:106 CN/8000
   a=ptime:20
   a=fingerprint:sha-256
83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
   a=rtcp-mux
   a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
   a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
   a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
   a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
   a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
   a=ice-ufrag:v9QogVGvZ8jGwYIi
   a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
   a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
generation 0
   ------------------------------------------------------------------------
tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (28092)
nta.c:1348 set_timeout() nta: timer shortened to 500 ms
2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3711
(sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change EARLY -> ACTIVE
nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
state changed: received -> completed, sent answer
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe17c0bf4d0,
[0x7fe1aa7b3c48], [0x7fe1aa7b3c50], [(nil)]) called
soa.c:616 soa_get_params() soa_get_params(static::0x7fe17c0bf4d0, ...)
called
nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 200 OK
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal
sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [completed][200]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx
conference(991234-172.30.0.219 at test)
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10364 using channel
sound prefix: /usr/local/freeswitch/sounds/en/us/callie
2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:8991 Raw Codec
Activation Success L16 at 48000hz 1 channel 20ms
2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:9037 Raw Codec
Activation Success L16 at 16000hz 1 channel 20ms
2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:221
sofia/internal/1000 at xxx.xxx.xxx.xxx Push codec L16:100
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '0' to 'mute'
2015-05-08 08:21:30.467711 [INFO] switch_ivr_async.c:212 Digit parser
mod_conference: Setting realm to 'conf'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding 0/conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010b8e0
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '*' to 'deaf mute'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding */conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010b910
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '9' to 'energy up'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding 9/conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010b940
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '8' to 'energy equ'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding 8/conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010b970
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '7' to 'energy dn'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding 7/conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010b9a0
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '3' to 'vol talk up'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding 3/conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010b9d0
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '2' to 'vol talk zero'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding 2/conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010ba00
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '1' to 'vol talk dn'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding 1/conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010ba30
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '6' to 'vol listen up'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding 6/conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010ba60
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '5' to 'vol listen zero'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding 5/conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010ba90
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '4' to 'vol listen dn'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding 4/conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010bac0
2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
sofia/internal/1000 at xxx.xxx.xxx.xxx binding '#' to 'hangup'
2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
mod_conference: binding #/conf/0 callback: 0x7fe1a92764e0 data:
0x7fe19010baf0
2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send signal
sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:4765 Setup timer soft
success interval: 20  samples: 960
2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:3043 Setup timer
success interval: 30  samples: 480
2015-05-08 08:21:30.507713 [DEBUG] mod_local_stream.c:498 Opening Stream
[moh/16000] 16000hz
2015-05-08 08:21:30.507713 [NOTICE] switch_core_io.c:1261 Activating write
resampler
tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
0x7fe17c0c6ab0 from (ws/82.166.84.247:53645) has 550 bytes, veclen = 1
recv 550 bytes from ws/[82.166.84.247]:53645 at 08:21:30.662998:
   ------------------------------------------------------------------------
   ACK sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0
   Via: SIP/2.0/WS
df7jal23ls0d.invalid;branch=z9hG4bK30nT8FwJ3EqSVIbdgFvT;rport
   From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
   To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
   Contact: "asdasda"<sip:1000 at df7jal23ls0d.invalid
;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
   Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
   CSeq: 28092 ACK
   Content-Length: 0
   Max-Forwards: 70
   User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
   Organization: Doubango Telecom

   ------------------------------------------------------------------------
tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
0x7fe17c0c6ab0 (550 bytes) from ws/82.166.84.247:53645/sip next=(nil)
nta.c:2880 agent_recv_request() nta: received ACK
sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp
SIP/2.0 (CSeq 28092)
nta.c:3174 agent_check_request_via() nta: Via check: received=82.166.84.247
nta.c:3019 agent_recv_request() nta: ACK (28092) is going to INVITE (28092)
nua_session.c:2569 process_ack_or_cancel() nua: process_ack_or_cancel:
entering
soa.c:1214 soa_clear_remote_sdp()
soa_clear_remote_sdp(static::0x7fe17c0bf4d0) called
nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_ack 200 OK
nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
state changed: completed -> ready
nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 200 OK
nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_active 200
Call active
nta.c:5744 incoming_free() nta: incoming_free(0x7fe17c024090)
tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal
sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal
sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal
sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-08 08:21:30.667707 [DEBUG] sofia.c:6627 Channel
sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [ready][200]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nta.c:1289 agent_timer() nta: timer not set
2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2924 Changing audio DTLS
state from HANDSHAKE to SETUP
2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2832 audio Fingerprint
Verified.
2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3374 Activating Audio Secure
RTP SEND
2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3352 Activating Audio Secure
RTP RECV
2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2872 Changing audio DTLS
state from SETUP to READY
2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure Type:
srtp:dtls:AES_CM_128_HMAC_SHA1_80
2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure Type:
srtp:dtls:AES_CM_128_HMAC_SHA1_80
2015-05-08 08:22:02.087708 [DEBUG] switch_rtp.c:1937 rtcp_stats_init:
ssrc[-195423717] base_seq[29507]

Notice that the INVITE was received at 08:21:30 while DTLS was READY
at 2015-05-08 08:22:02 which means that it took 32 seconds to voice. Again,
if I am not using TURN/STUN, the whole process is pretty quick (2 seconds).

Also, notice:
2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209

Which means that the media is relayed via the TURN server (TURN server IP
is 130.211.78.35)...

Any idea?

Thanks,
Adam
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From notify.sina at gmail.com  Fri May  8 13:21:51 2015
From: notify.sina at gmail.com (Sina Owolabi)
Date: Fri, 8 May 2015 10:21:51 +0100
Subject: [Freeswitch-users] Help Needed Generating Calls with Web API and
	Python
Message-ID: <CACK8u8JunNJtM_UEj8fbEioqydiqAY=WzM4E7UMGcQfArbCtUQ@mail.gmail.com>

Hi!

I am sorely in need of guidance on how to do this. I have looked at
the wiki page for web api with python and I can run any of the generic
server commands ("show", etc).

I've tried to generate an outbound call through a gateway, but the
server just silently accepts my commands and does nothing. I also see
a corresponding log in freeswitch_http.log that I have just POSTed
something but that's it.

Please can someone guide me on how to originate this properly? I am
gunning for getting this to dial a number and play something.

My script so far:

import xmlrpc.client

host = 'localhost'
username = 'apicaller'
password = 'password'
port = '8080'

server = xmlrpc.client.ServerProxy("http://%s:%s@%s:%s" % (username,
password, host, port))
#print(server.freeswitch.api("show","status"))
server.freeswitch.api('bgapi originate',
'{origination_caller_id_name="Mr
Whiskers",origination_caller_id_number="012345678",ignore_early_media=true}sofia/gateway/agateway/anumber')

Thanks!


From adam.ben.ayoun1 at gmail.com  Fri May  8 15:36:48 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Fri, 8 May 2015 14:36:48 +0300
Subject: [Freeswitch-users] Trickle ICE with SIP
Message-ID: <CADNqFCmOfu_Nq56N9rc6H2DD=9OG7=MTqB1D4dT+kJ9GFcaBSA@mail.gmail.com>

Hi,

We are currently using Freeswitch for voice conferencing, we use SIP for
signalling. Call setup times are really slow at times (3sec-30sec), can we
somehow do Trickle ICE with Freeswitch?

Thanks,
Adam
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From jason.holden at start.ca  Fri May  8 16:58:08 2015
From: jason.holden at start.ca (Jason Holden)
Date: Fri, 8 May 2015 08:58:08 -0400
Subject: [Freeswitch-users] need SIP IP / RTP recommendation
Message-ID: <1DD529F81979496BADBE1B52EB5D334B@bob>

Currently I have a client who is using multiple WAN IP addresses.

On route fail over FS still tries to use the original IP details from the
external profile and not the secondary circuit details from the secondary
external profile.

What have others done to address this to ensure failover works propperly?

 

 

 

 

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From mishehu at freeswitch.org  Fri May  8 17:33:40 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Fri, 08 May 2015 08:33:40 -0500
Subject: [Freeswitch-users] Help Needed Generating Calls with Web API
 and Python
In-Reply-To: <CACK8u8JunNJtM_UEj8fbEioqydiqAY=WzM4E7UMGcQfArbCtUQ@mail.gmail.com>
References: <CACK8u8JunNJtM_UEj8fbEioqydiqAY=WzM4E7UMGcQfArbCtUQ@mail.gmail.com>
Message-ID: <554CBB34.2070805@freeswitch.org>

Have you attempted to run the command, minus the "bgapi" part directly 
from fs_cli or the FreeSWITCH console?  I would recommend doing this, 
and enabling debug output (f8).  It will be a bit painfully verbose, but 
it should reveal some information.  Based upon the information you 
provided, you're not actually giving any handling commands to this new 
channel that you are originating.  It needs to bridge to something or 
run some sort of dialplan application otherwise it vanishes as quickly 
as it was created.

-Yossi

On 05/08/2015 04:21 AM, Sina Owolabi wrote:
> Hi!
>
> I am sorely in need of guidance on how to do this. I have looked at
> the wiki page for web api with python and I can run any of the generic
> server commands ("show", etc).
>
> I've tried to generate an outbound call through a gateway, but the
> server just silently accepts my commands and does nothing. I also see
> a corresponding log in freeswitch_http.log that I have just POSTed
> something but that's it.
>
> Please can someone guide me on how to originate this properly? I am
> gunning for getting this to dial a number and play something.
>
> My script so far:
>
> import xmlrpc.client
>
> host = 'localhost'
> username = 'apicaller'
> password = 'password'
> port = '8080'
>
> server = xmlrpc.client.ServerProxy("http://%s:%s@%s:%s" % (username,
> password, host, port))
> #print(server.freeswitch.api("show","status"))
> server.freeswitch.api('bgapi originate',
> '{origination_caller_id_name="Mr
> Whiskers",origination_caller_id_number="012345678",ignore_early_media=true}sofia/gateway/agateway/anumber')
>
> Thanks!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From krice at freeswitch.org  Fri May  8 18:00:41 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 08 May 2015 14:00:41 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <554cc189cf92c_c7f5cc3328487af@resque-worker-high.4.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From michel.brabants at gmail.com  Fri May  8 18:34:01 2015
From: michel.brabants at gmail.com (Michel Brabants)
Date: Fri, 8 May 2015 16:34:01 +0200
Subject: [Freeswitch-users] info about sofia sip dns cache and graylisting?
Message-ID: <CAHt+dNGgTiN9eNJXFdig03t5o2FEtPPZt6ksAcPMNN_RvWk=Ng@mail.gmail.com>

Hello all,

we have a problem with a 1.4-version of Freeswitch that FS goes to 100%
cpu. The problem has been traced to a dns-cache-problem. The dns-cache
contains srv-records, which should have 2 a-records, but they have 100's of
a-records, which are mostly the same (some with a different priority). This
causes FS do to a lot of compares (to sort the records) and it goes to 100%
cpu in the end.
The priority almost never is the original priority, so I was thinking that
the cache is maybe growing because of graylisting of entries, causing the
priority to increase and the ttl to be reset.

The code is however very difficult to follow as it is all event-based and
part of big C-structs in sofia-sip. A bug in the gdb-debugger causing
static variables to be almost unreadable makes life not much easier. The
graylisting policy, as well as the dns cache policy is not really
documented as far as I can find, so any info about this would be great.

Anyway, my current thought is that the dns-cache is maybe growing, becuase
the graylisting-code sets the priority to a higher value and resets the ttl
to a certain value (no idea which one, but the default is 10 minutes and
the maximum a day). The srv-record-compare-function however checks also
using priority and weight and returns a negative response if they are
different. I would think that fs maybe then readds the original srv-records
(with priority 0) when it requeries the dns. Those entries will also
increase in priority and ttl when they are added, causing the cache to keep
growing ...
Nscd is also being used for dns-caching, which doesn't help, but I'm not
sure if that is the problem as sofia_dig seems to return valid values
(except maybe ttl).


 I would think that ttl is never touched and that the cached is cleared
when the ttl expires, but I have no idea when the dns-cache is cleared in
FS...

I know this is maybe a lot of text, but any info on this topic (sofia dns
cache) is welcome.

Thanks,

Michel
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From brian at freeswitch.org  Fri May  8 18:38:15 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 May 2015 09:38:15 -0500
Subject: [Freeswitch-users] info about sofia sip dns cache and
	graylisting?
In-Reply-To: <CAHt+dNGgTiN9eNJXFdig03t5o2FEtPPZt6ksAcPMNN_RvWk=Ng@mail.gmail.com>
References: <CAHt+dNGgTiN9eNJXFdig03t5o2FEtPPZt6ksAcPMNN_RvWk=Ng@mail.gmail.com>
Message-ID: <CAEJMVkA=Gw5XHtvv89o4r1X95sjGFwbcyLFAJNixDAuqJtLrYA@mail.gmail.com>

can you get some logs, sofia loglevel all 9, and file a JIRA and attach the
logs and detailed info to the JIRA please.

Also verify which rev of FreeSWITCH you're running when filing the JIRA.

On Fri, May 8, 2015 at 9:34 AM, Michel Brabants <michel.brabants at gmail.com>
wrote:

> Hello all,
>
> we have a problem with a 1.4-version of Freeswitch that FS goes to 100%
> cpu. The problem has been traced to a dns-cache-problem. The dns-cache
> contains srv-records, which should have 2 a-records, but they have 100's of
> a-records, which are mostly the same (some with a different priority). This
> causes FS do to a lot of compares (to sort the records) and it goes to 100%
> cpu in the end.
> The priority almost never is the original priority, so I was thinking that
> the cache is maybe growing because of graylisting of entries, causing the
> priority to increase and the ttl to be reset.
>
> The code is however very difficult to follow as it is all event-based and
> part of big C-structs in sofia-sip. A bug in the gdb-debugger causing
> static variables to be almost unreadable makes life not much easier. The
> graylisting policy, as well as the dns cache policy is not really
> documented as far as I can find, so any info about this would be great.
>
> Anyway, my current thought is that the dns-cache is maybe growing, becuase
> the graylisting-code sets the priority to a higher value and resets the ttl
> to a certain value (no idea which one, but the default is 10 minutes and
> the maximum a day). The srv-record-compare-function however checks also
> using priority and weight and returns a negative response if they are
> different. I would think that fs maybe then readds the original srv-records
> (with priority 0) when it requeries the dns. Those entries will also
> increase in priority and ttl when they are added, causing the cache to keep
> growing ...
> Nscd is also being used for dns-caching, which doesn't help, but I'm not
> sure if that is the problem as sofia_dig seems to return valid values
> (except maybe ttl).
>
>
>  I would think that ttl is never touched and that the cached is cleared
> when the ttl expires, but I have no idea when the dns-cache is cleared in
> FS...
>
> I know this is maybe a lot of text, but any info on this topic (sofia dns
> cache) is welcome.
>
> Thanks,
>
> Michel
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 danb.lists at gmail.com  Fri May  8 19:29:41 2015
From: danb.lists at gmail.com (DanB)
Date: Fri, 08 May 2015 17:29:41 +0200
Subject: [Freeswitch-users] IvrMenu - setting channel variables before
	calling exec-app/transfer
Message-ID: <554CD665.5040905@gmail.com>

Hey Guys,

I was wondering if my scenario would be possible:

I have a menu for my ivr which should execute a transfer to a XML dialplan.
Eg:
<entry digits="2" action="menu-exec-app" param="transfer 1001 XML 
default" />

I would need to set some channel variables before hitting extension 1001 
in dialplan (before the transfer). Would inline dialplan do here or 
anyone able to propose a better solution?

Ta for any kind of tip!

DanB


From mike at jerris.com  Fri May  8 19:34:23 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 May 2015 11:34:23 -0400
Subject: [Freeswitch-users] WebRTC slow connection time
In-Reply-To: <CADNqFCky4iMXF-5vukKcqmT8=Q+anVHdtn+0zueV+4yJ3GUC6g@mail.gmail.com>
References: <CADNqFCky4iMXF-5vukKcqmT8=Q+anVHdtn+0zueV+4yJ3GUC6g@mail.gmail.com>
Message-ID: <E6A2DD46-EA8B-400A-A58D-AA9547DDA682@jerris.com>

If you have freeswitch on a public address, why would you ever use TURN?

> On May 8, 2015, at 4:27 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:
> 
> Hi guys,
> 
> I am using Freeswitch from master (month old). I tried several WebRTC clients (my own test app on Android and sipml5) and on certain WiFi's I am getting connection time up to 35 seconds when using STUN and TURN servers. Also, when I am using TURN and STUN, Freeswitch chooses the TURN candidate although as far as connectivity, when I am not using STUN and TURN I am connecting successfully after ~2 seconds (so no real need for TURN). Attaching the SIP trace from Freeswitch:
> 
> nua.c:575 nua_set_params() nua: nua_set_params: entering
> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
> nua.c:575 nua_set_params() nua: nua_set_params: entering
> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
> nua.c:575 nua_set_params() nua: nua_set_params: entering
> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
> nua.c:575 nua_set_params() nua: nua_set_params: entering
> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> nua.c:575 nua_set_params() nua: nua_set_params: entering
> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...) called
> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe18c001930, ...) called
> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...) called
> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe184001930, ...) called
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe188001930, ...) called
> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg 0x7fe17c089c50 from (ws/82.166.84.247:53645 <http://82.166.84.247:53645/>) has 2598 bytes, veclen = 1
> recv 2598 bytes from ws/[82.166.84.247]:53645 at 08:21:30.468866:
>    ------------------------------------------------------------------------
>    INVITE sip:991234 at aaaa SIP/2.0
>    Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport
>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>    To: <sip:991234 at aaaa>
>    Contact: "asdasda"<sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>    CSeq: 28092 INVITE
>    Content-Type: application/sdp
>    Content-Length: 2039
>    Max-Forwards: 70
>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>    Organization: Doubango Telecom
> 
>    v=0
>    o=- 3843427479443760600 2 IN IP4 127.0.0.1
>    s=Doubango Telecom - chrome
>    t=0 0
>    a=group:BUNDLE audio
>    a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>    m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>    c=IN IP4 130.211.78.35
>    a=rtcp:58209 IN IP4 130.211.78.35
>    a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host generation 0
>    a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host generation 0
>    a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx raddr 10.0.0.10 rport 63888 generation 0
>    a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx raddr 10.0.0.10 rport 63888 generation 0
>    a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype active generation 0
>    a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype active generation 0
>    a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay raddr 82.166.84.247 rport 53792 generation 0
>    a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay raddr 82.166.84.247 rport 53792 generation 0
>    a=ice-ufrag:EDo5kr308/TXhitG
>    a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>    a=ice-options:google-ice
>    a=fingerprint:sha-256 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>    a=setup:actpass
>    a=mid:audio
>    a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>    a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
>    a=sendrecv
>    a=rtcp-mux
>    a=rtpmap:111 opus/48000/2
>    a=fmtp:111 minptime=10; useinbandfec=1
>    a=rtpmap:103 ISAC/16000
>    a=rtpmap:104 ISAC/32000
>    a=rtpmap:9 G722/8000
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:106 CN/32000
>    a=rtpmap:105 CN/16000
>    a=rtpmap:13 CN/8000
>    a=rtpmap:126 telephone-event/8000
>    a=maxptime:60
>    a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>    a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm b8082781-4098-4ac5-8c96-2848d2e9e7e4
>    a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>    a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>    ------------------------------------------------------------------------
> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg 0x7fe17c089c50 (2598 bytes) from ws/82.166.84.247:53645/sip <http://82.166.84.247:53645/sip> next=(nil)
> nta.c:2880 agent_recv_request() nta: received INVITE sip:991234 at aaaa SIP/2.0 (CSeq 28092)
> nta.c:3174 agent_check_request_via() nta: Via check: received=82.166.84.247
> nta.c:3085 agent_recv_request() nta: INVITE (28092) going to a default leg
> nta.c:1350 set_timeout() nta: timer set to 2000 ms
> nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
> nua_stack.c:899 nh_create() nua: nh_create: entering
> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:280 soa_clone() soa_clone(static::0x7fe17c001930, 0x7fe17c001130, 0x7fe17c0cf210) called
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...) called
> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fe17c04a2e0)
> soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7fe17c0bf4d0) called
> soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe17c0be55f, 2039) called
> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fe17c0cf210): adding session usage
> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/82.166.84.247:53645 <http://82.166.84.247:53645/>
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x7fe17c0d4d90 0x7fe17c0190a0 140 (140)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x7fe17c0d4d90 0x7fe17c0be3ab 86 (86)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x7fe17c0d4d90 0x7fe17c01912c 101 (101)
> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 394 bytes of 394 to ws/82.166.84.247:53645 <http://82.166.84.247:53645/>
> tport.c:3492 tport_send_msg() tport_vsend returned 394
> send 394 bytes to ws/[82.166.84.247]:53645 at 08:21:30.469343:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>    To: <sip:991234 at aaaa>
>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>    CSeq: 28092 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
> nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (28092)
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_invite 100 Trying
> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call state changed: init -> received, received offer
> soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fe17c0bf4d0, [0x7fe1aa7b35d8], [0x7fe1aa7b35e0], [(nil)]) called
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 100 Trying
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
> 2015-05-08 08:21:30.467711 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/1000 at xxx.xxx.xxx.xxx [3ac3c622-f55b-11e4-a447-7d37723461ed]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_NEW
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8848 sofia/internal/1000 at xxx.xxx.xxx.xxx receiving invite from 82.166.84.247:53645 <http://82.166.84.247:53645/> version: 1.5.15b git b01352c 2015-04-23 04:33:08Z 64bit
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8960 IP 82.166.84.247 Approved by acl "domains[]". Access Granted.
> nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
> nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil)
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:10113 Setting NAT mode based on via received
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [received][100]
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6637 Remote SDP:
> v=0
> o=- 3843427479443760600 2 IN IP4 127.0.0.1
> s=Doubango Telecom - chrome
> t=0 0
> a=group:BUNDLE audio
> a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
> m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
> c=IN IP4 130.211.78.35
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10; useinbandfec=1
> a=rtpmap:103 ISAC/16000
> a=rtpmap:104 ISAC/32000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=rtcp:58209 IN IP4 130.211.78.35
> a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host generation 0
> a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host generation 0
> a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx raddr 10.0.0.10 rport 63888 generation 0
> a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx raddr 10.0.0.10 rport 63888 generation 0
> a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype active generation 0
> a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype active generation 0
> a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay raddr 82.166.84.247 rport 53792 generation 0
> a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay raddr 82.166.84.247 rport 53792 generation 0
> a=ice-ufrag:EDo5kr308/TXhitG
> a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
> a=ice-options:google-ice
> a=fingerprint:sha-256 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
> a=setup:actpass
> a=mid:audio
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
> a=rtcp-mux
> a=maxptime:60
> a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
> a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm b8082781-4098-4ac5-8c96-2848d2e9e7e4
> a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
> a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
> 
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6903 (sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_NEW -> CS_INIT
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1000 at xxx.xxx.xxx.xxx) State NEW
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_INIT
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT
> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:87 sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA INIT
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1000 at xxx.xxx.xxx.xxx Standard INIT
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_INIT -> CS_ROUTING
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT going to sleep
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_ROUTING
> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:2204 (sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change DOWN -> RINGING
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING
> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:123 sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA ROUTING
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1000 at xxx.xxx.xxx.xxx Standard ROUTING
> 2015-05-08 08:21:30.467711 [INFO] mod_dialplan_xml.c:635 Processing asdasda <1000>->991234 in context public
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing [public->cdquality_conferences_with_api] continue=false
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (FAIL) [cdquality_conferences_with_api] destination_number(991234) =~ /^(75\d{4,36})$/ break=on-false
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing [public->test_conferences] continue=false
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (PASS) [test_conferences] destination_number(991234) =~ /^(99\d{4,36})$/ break=on-false
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action answer()
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action conference(991234-${domain_name}@test)
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_ROUTING -> CS_EXECUTE
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING going to sleep
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_EXECUTE
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1000 at xxx.xxx.xxx.xxx) State EXECUTE
> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:178 sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA EXECUTE
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1000 at xxx.xxx.xxx.xxx Standard EXECUTE
> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx answer()
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [opus:111:48000:60:0:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG! Sticking with opus at 48000h@20i
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [opus:111:48000:60:0:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [opus:111:48000:60:0:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [ISAC:103:16000:30:32000:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [ISAC:103:16000:30:32000:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [ISAC:103:16000:30:32000:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [ISAC:104:32000:30:32000:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [ISAC:104:32000:30:32000:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [ISAC:104:32000:30:32000:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [G722:9:8000:60:64000:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [G722:9:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [G722:9:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [PCMU:0:8000:60:64000:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [PCMU:0:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG! Sticking with PCMU at 8000h@20i
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [PCMU:0:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [PCMA:8:8000:60:64000:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [PCMA:8:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [PCMA:8:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG! Sticking with PCMA at 8000h@20i
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [CN:105:16000:60:0:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [CN:105:16000:60:0:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [CN:105:16000:60:0:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [CN:13:8000:60:0:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [CN:13:8000:60:0:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec Compare [CN:13:8000:60:0:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3588 Set telephone-event payload to 126
> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set bitrate to local settings [-1000bps]
> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set bitrate to local settings [-1000bps]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2507 Set Codec sofia/internal/1000 at xxx.xxx.xxx.xxx opus/48000 20 ms 960 samples 0 bits 1 channels
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:111 sofia/internal/1000 at xxx.xxx.xxx.xxx Original read codec set to opus:116
> 2015-05-08 08:21:30.467711 [WARNING] switch_core_media.c:2791 NO candidate ACL defined, Defaulting to wan.auto
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888 <http://10.0.0.10:63888/>
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888 <http://10.0.0.10:63888/>
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888 <http://10.0.0.10:63888/>
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888 <http://10.0.0.10:63888/>
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888 <http://82.166.84.247:63888/>
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose audio Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888 <http://82.166.84.247:63888/>
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888 <http://82.166.84.247:63888/>
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose audio Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888 <http://82.166.84.247:63888/>
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209 <http://130.211.78.35:58209/>
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209 <http://130.211.78.35:58209/>
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209 <http://130.211.78.35:58209/>
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209 <http://130.211.78.35:58209/>
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2953 setting remote audio ice addr to 82.166.84.247:63888 <http://82.166.84.247:63888/> based on candidate
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2978 setting remote rtcp audio addr to 82.166.84.247:63888 <http://82.166.84.247:63888/> based on candidate
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3935 Set 2833 dtmf send/recv payload to 126
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5171 AUDIO RTP [sofia/internal/1000 at xxx.xxx.xxx.xxx] 172.30.0.219 port 19864 -> 82.166.84.247 port 63888 codec: 111 ms: 20
> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3559 Starting timer [soft] 960 bytes per 20ms
> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5345 Activating Audio ICE
> 2015-05-08 08:21:30.467711 [NOTICE] switch_rtp.c:4009 Activating RTP audio ICE: EDo5kr308/TXhitG:v9QogVGvZ8jGwYIi 82.166.84.247:63888 <http://82.166.84.247:63888/>
> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5388 Activating RTCP PORT 63888
> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3909 RTCP send rate is: 10000 and packet rate is: 20000 Remote Port: 63888
> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:2349 Setting RTCP remote addr to 82.166.84.247:63888 <http://82.166.84.247:63888/>
> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5396 Skipping RTCP ICE (Same as RTP)
> 2015-05-08 08:21:30.467711 [INFO] switch_rtp.c:3101 Activate RTP/RTCP audio DTLS client
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5469 Set 2833 dtmf send payload to 126
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5475 Set 2833 dtmf receive payload to 126
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5503 Set comfort noise payload to 106
> 2015-05-08 08:21:30.467711 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000 at xxx.xxx.xxx.xxx!
> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3419 (sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change RINGING -> EARLY
> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1000 at xxx.xxx.xxx.xxx:
> v=0
> o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
> s=FreeSWITCH
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
> m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 useinbandfec=1; minptime=10
> a=rtpmap:126 telephone-event/8000
> a=rtpmap:106 CN/8000
> a=ptime:20
> a=sendrecv
> a=fingerprint:sha-256 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
> a=rtcp-mux
> a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
> a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
> a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
> a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
> a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
> a=ice-ufrag:v9QogVGvZ8jGwYIi
> a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
> a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host generation 0
> 
> nua.c:879 nua_respond() nua: nua_respond: entering
> nua_stack.c:529 nua_signal() nua(0x7fe17c0cf210): sent signal r_respond
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua_stack.c:573 nua_stack_signal() nua(0x7fe17c0cf210): recv signal r_respond 200 OK
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...) called
> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe190041d31, -1) called
> soa.c:890 soa_set_capability_sdp() soa_set_capability_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe190041d31, -1) called
> 2015-05-08 08:21:30.467711 [NOTICE] mod_dptools.c:1292 Channel [sofia/internal/1000 at xxx.xxx.xxx.xxx] has been answered
> nua_session.c:2320 nua_invite_server_respond() nua: nua_invite_server_respond: entering
> soa.c:1515 soa_generate_answer() soa_generate_answer(static::0x7fe17c0bf4d0) called
> soa_static.c:1146 offer_answer_step() soa_static_offer_answer_action(0x7fe17c0bf4d0, soa_generate_answer): called
> soa_static.c:1187 offer_answer_step() soa_static(0x7fe17c0bf4d0, soa_generate_answer): generating local description
> soa_static.c:1228 offer_answer_step() soa_static(0x7fe17c0bf4d0, soa_generate_answer): upgrade with remote description
> soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fe1aa7b1a30, 0x7fe17c026ac0, ""): called
> soa_static.c:1444 offer_answer_step() soa_static(0x7fe17c0bf4d0, soa_generate_answer): storing local description
> soa.c:1730 soa_activate() soa_activate(static::0x7fe17c0bf4d0, (nil)) called
> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe17c0bf4d0, [(nil)], [0x7fe1aa7b3b58], [0x7fe1aa7b3b54]) called
> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/82.166.84.247:53645 <http://82.166.84.247:53645/>
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x7fe17c0d4d90 0x7fe17c0c5560 136 (136)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x7fe17c0d4d90 0x7fe17c0be3ab 63 (63)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x7fe17c0d4d90 0x7fe17c0c55e8 41 (41)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x7fe17c0d4d90 0x7fe17c0c5611 631 (631)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec 0x7fe17c0d4d90 0x7fe17c0d3020 864 (864)
> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 1802 bytes of 1802 to ws/82.166.84.247:53645 <http://82.166.84.247:53645/>
> tport.c:3492 tport_send_msg() tport_vsend returned 1802
> send 1802 bytes to ws/[82.166.84.247]:53645 at 08:21:30.479618:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>    CSeq: 28092 INVITE
>    Contact: <sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>    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: 864
>    Remote-Party-ID: "991234" <sip:991234 at aaaa>;party=calling;privacy=off;screen=no
> 
>    v=0
>    o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>    s=FreeSWITCH
>    c=IN IP4 xxx.xxx.xxx.xxx
>    t=0 0
>    a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>    m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>    a=rtpmap:111 opus/48000/2
>    a=fmtp:111 useinbandfec=1; minptime=10
>    a=rtpmap:126 telephone-event/8000
>    a=rtpmap:106 CN/8000
>    a=ptime:20
>    a=fingerprint:sha-256 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>    a=rtcp-mux
>    a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>    a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>    a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>    a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>    a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>    a=ice-ufrag:v9QogVGvZ8jGwYIi
>    a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>    a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host generation 0
>    ------------------------------------------------------------------------
> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
> nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (28092)
> nta.c:1348 set_timeout() nta: timer shortened to 500 ms
> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3711 (sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change EARLY -> ACTIVE
> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call state changed: received -> completed, sent answer
> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe17c0bf4d0, [0x7fe1aa7b3c48], [0x7fe1aa7b3c50], [(nil)]) called
> soa.c:616 soa_get_params() soa_get_params(static::0x7fe17c0bf4d0, ...) called
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 200 OK
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [completed][200]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx conference(991234-172.30.0.219 at test)
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10364 using channel sound prefix: /usr/local/freeswitch/sounds/en/us/callie
> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:8991 Raw Codec Activation Success L16 at 48000hz 1 channel 20ms
> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:9037 Raw Codec Activation Success L16 at 16000hz 1 channel 20ms
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:221 sofia/internal/1000 at xxx.xxx.xxx.xxx Push codec L16:100
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '0' to 'mute'
> 2015-05-08 08:21:30.467711 [INFO] switch_ivr_async.c:212 Digit parser mod_conference: Setting realm to 'conf'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding 0/conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010b8e0
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '*' to 'deaf mute'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding */conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010b910
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '9' to 'energy up'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding 9/conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010b940
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '8' to 'energy equ'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding 8/conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010b970
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '7' to 'energy dn'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding 7/conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010b9a0
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '3' to 'vol talk up'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding 3/conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010b9d0
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '2' to 'vol talk zero'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding 2/conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010ba00
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '1' to 'vol talk dn'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding 1/conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010ba30
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '6' to 'vol listen up'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding 6/conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010ba60
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '5' to 'vol listen zero'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding 5/conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010ba90
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '4' to 'vol listen dn'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding 4/conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010bac0
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967 sofia/internal/1000 at xxx.xxx.xxx.xxx binding '#' to 'hangup'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser mod_conference: binding #/conf/0 callback: 0x7fe1a92764e0 data: 0x7fe19010baf0
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:4765 Setup timer soft success interval: 20  samples: 960
> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:3043 Setup timer success interval: 30  samples: 480
> 2015-05-08 08:21:30.507713 [DEBUG] mod_local_stream.c:498 Opening Stream [moh/16000] 16000hz
> 2015-05-08 08:21:30.507713 [NOTICE] switch_core_io.c:1261 Activating write resampler
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg 0x7fe17c0c6ab0 from (ws/82.166.84.247:53645 <http://82.166.84.247:53645/>) has 550 bytes, veclen = 1
> recv 550 bytes from ws/[82.166.84.247]:53645 at 08:21:30.662998:
>    ------------------------------------------------------------------------
>    ACK sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0
>    Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK30nT8FwJ3EqSVIbdgFvT;rport
>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>    Contact: "asdasda"<sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>    CSeq: 28092 ACK
>    Content-Length: 0
>    Max-Forwards: 70
>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>    Organization: Doubango Telecom
> 
>    ------------------------------------------------------------------------
> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg 0x7fe17c0c6ab0 (550 bytes) from ws/82.166.84.247:53645/sip <http://82.166.84.247:53645/sip> next=(nil)
> nta.c:2880 agent_recv_request() nta: received ACK sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0 (CSeq 28092)
> nta.c:3174 agent_check_request_via() nta: Via check: received=82.166.84.247
> nta.c:3019 agent_recv_request() nta: ACK (28092) is going to INVITE (28092)
> nua_session.c:2569 process_ack_or_cancel() nua: process_ack_or_cancel: entering
> soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0x7fe17c0bf4d0) called
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_ack 200 OK
> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call state changed: completed -> ready
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 200 OK
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_active 200 Call active
> nta.c:5744 incoming_free() nta: incoming_free(0x7fe17c024090)
> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-05-08 08:21:30.667707 [DEBUG] sofia.c:6627 Channel sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [ready][200]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nta.c:1289 agent_timer() nta: timer not set
> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing stun/rtp/dtls port from 82.166.84.247:63888 <http://82.166.84.247:63888/> to 130.211.78.35:58209 <http://130.211.78.35:58209/>
> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2924 Changing audio DTLS state from HANDSHAKE to SETUP
> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2832 audio Fingerprint Verified.
> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3374 Activating Audio Secure RTP SEND
> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3352 Activating Audio Secure RTP RECV
> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2872 Changing audio DTLS state from SETUP to READY
> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80
> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80
> 2015-05-08 08:22:02.087708 [DEBUG] switch_rtp.c:1937 rtcp_stats_init: ssrc[-195423717] base_seq[29507]
> 
> Notice that the INVITE was received at 08:21:30 while DTLS was READY at 2015-05-08 08:22:02 which means that it took 32 seconds to voice. Again, if I am not using TURN/STUN, the whole process is pretty quick (2 seconds).
> 
> Also, notice:
> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing stun/rtp/dtls port from 82.166.84.247:63888 <http://82.166.84.247:63888/> to 130.211.78.35:58209 <http://130.211.78.35:58209/>
> 
> Which means that the media is relayed via the TURN server (TURN server IP is 130.211.78.35)...
> 
> Any idea?
> 
> Thanks,
> Adam
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From covici at ccs.covici.com  Fri May  8 19:35:10 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 08 May 2015 11:35:10 -0400
Subject: [Freeswitch-users] IvrMenu - setting channel variables before
	calling exec-app/transfer
In-Reply-To: <554CD665.5040905@gmail.com>
References: <554CD665.5040905@gmail.com>
Message-ID: <12061.1431099310@ccs.covici.com>

Transfer to another extension which in turn transfers to the one you
want.

DanB <danb.lists at gmail.com> wrote:

> Hey Guys,
> 
> I was wondering if my scenario would be possible:
> 
> I have a menu for my ivr which should execute a transfer to a XML dialplan.
> Eg:
> <entry digits="2" action="menu-exec-app" param="transfer 1001 XML 
> default" />
> 
> I would need to set some channel variables before hitting extension 1001 
> in dialplan (before the transfer). Would inline dialplan do here or 
> anyone able to propose a better solution?
> 
> Ta for any kind of tip!
> 
> DanB
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mike at jerris.com  Fri May  8 19:35:08 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 May 2015 11:35:08 -0400
Subject: [Freeswitch-users] Trickle ICE with SIP
In-Reply-To: <CADNqFCmOfu_Nq56N9rc6H2DD=9OG7=MTqB1D4dT+kJ9GFcaBSA@mail.gmail.com>
References: <CADNqFCmOfu_Nq56N9rc6H2DD=9OG7=MTqB1D4dT+kJ9GFcaBSA@mail.gmail.com>
Message-ID: <C56F753C-D2E5-4072-BDC2-43BCE6E367AB@jerris.com>

From your other post you said thats only when using turn?  

> On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:
> 
> Hi,
> 
> We are currently using Freeswitch for voice conferencing, we use SIP for signalling. Call setup times are really slow at times (3sec-30sec), can we somehow do Trickle ICE with Freeswitch?
> 
> Thanks,
> Adam 



From mike at jerris.com  Fri May  8 19:37:03 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 May 2015 11:37:03 -0400
Subject: [Freeswitch-users] IvrMenu - setting channel variables before
	calling exec-app/transfer
In-Reply-To: <554CD665.5040905@gmail.com>
References: <554CD665.5040905@gmail.com>
Message-ID: <12999F81-B4F6-4E74-8D61-70C11C866089@jerris.com>

Can you set the variables before you hit the menu?

> On May 8, 2015, at 11:29 AM, DanB <danb.lists at gmail.com> wrote:
> 
> Hey Guys,
> 
> I was wondering if my scenario would be possible:
> 
> I have a menu for my ivr which should execute a transfer to a XML dialplan.
> Eg:
> <entry digits="2" action="menu-exec-app" param="transfer 1001 XML 
> default" />
> 
> I would need to set some channel variables before hitting extension 1001 
> in dialplan (before the transfer). Would inline dialplan do here or 
> anyone able to propose a better solution?
> 
> Ta for any kind of tip!
> 
> DanB



From adam.ben.ayoun1 at gmail.com  Fri May  8 19:40:12 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Fri, 8 May 2015 18:40:12 +0300
Subject: [Freeswitch-users] WebRTC slow connection time
In-Reply-To: <E6A2DD46-EA8B-400A-A58D-AA9547DDA682@jerris.com>
References: <CADNqFCky4iMXF-5vukKcqmT8=Q+anVHdtn+0zueV+4yJ3GUC6g@mail.gmail.com>
	<E6A2DD46-EA8B-400A-A58D-AA9547DDA682@jerris.com>
Message-ID: <CADNqFC=ba=eD8DCAicLLNx6_=xw0JRFYSdM+UDo2D35OGLR_+g@mail.gmail.com>

I am using TURN as a TCP fallback (when UDP is blocked).

On 8 May 2015 at 18:34, Michael Jerris <mike at jerris.com> wrote:

> If you have freeswitch on a public address, why would you ever use TURN?
>
> On May 8, 2015, at 4:27 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
> wrote:
>
> Hi guys,
>
> I am using Freeswitch from master (month old). I tried several WebRTC
> clients (my own test app on Android and sipml5) and on certain WiFi's I am
> getting connection time up to 35 seconds when using STUN and TURN servers.
> Also, when I am using TURN and STUN, Freeswitch chooses the TURN candidate
> although as far as connectivity, when I am not using STUN and TURN I am
> connecting successfully after ~2 seconds (so no real need for TURN).
> Attaching the SIP trace from Freeswitch:
>
> nua.c:575 nua_set_params() nua: nua_set_params: entering
> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
> nua.c:575 nua_set_params() nua: nua_set_params: entering
> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
> nua.c:575 nua_set_params() nua: nua_set_params: entering
> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
> nua.c:575 nua_set_params() nua: nua_set_params: entering
> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> nua.c:575 nua_set_params() nua: nua_set_params: entering
> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
> called
> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe18c001930, ...)
> called
> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
> called
> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe184001930, ...)
> called
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe188001930, ...)
> called
> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
> 0x7fe17c089c50 from (ws/82.166.84.247:53645) has 2598 bytes, veclen = 1
> recv 2598 bytes from ws/[82.166.84.247]:53645 at 08:21:30.468866:
>    ------------------------------------------------------------------------
>    INVITE sip:991234 at aaaa SIP/2.0
>    Via: SIP/2.0/WS
> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport
>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>    To: <sip:991234 at aaaa>
>    Contact: "asdasda"<
> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
> >;+g.oma.sip-im;language="en,fr"
>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>    CSeq: 28092 INVITE
>    Content-Type: application/sdp
>    Content-Length: 2039
>    Max-Forwards: 70
>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>    Organization: Doubango Telecom
>
>    v=0
>    o=- 3843427479443760600 2 IN IP4 127.0.0.1
>    s=Doubango Telecom - chrome
>    t=0 0
>    a=group:BUNDLE audio
>    a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>    m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>    c=IN IP4 130.211.78.35
>    a=rtcp:58209 IN IP4 130.211.78.35
>    a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
> generation 0
>    a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
> generation 0
>    a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx
> raddr 10.0.0.10 rport 63888 generation 0
>    a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx
> raddr 10.0.0.10 rport 63888 generation 0
>    a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype
> active generation 0
>    a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype
> active generation 0
>    a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay
> raddr 82.166.84.247 rport 53792 generation 0
>    a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay
> raddr 82.166.84.247 rport 53792 generation 0
>    a=ice-ufrag:EDo5kr308/TXhitG
>    a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>    a=ice-options:google-ice
>    a=fingerprint:sha-256
> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>    a=setup:actpass
>    a=mid:audio
>    a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>    a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>    a=sendrecv
>    a=rtcp-mux
>    a=rtpmap:111 opus/48000/2
>    a=fmtp:111 minptime=10; useinbandfec=1
>    a=rtpmap:103 ISAC/16000
>    a=rtpmap:104 ISAC/32000
>    a=rtpmap:9 G722/8000
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:106 CN/32000
>    a=rtpmap:105 CN/16000
>    a=rtpmap:13 CN/8000
>    a=rtpmap:126 telephone-event/8000
>    a=maxptime:60
>    a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>    a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
> b8082781-4098-4ac5-8c96-2848d2e9e7e4
>    a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>    a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>    ------------------------------------------------------------------------
> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
> 0x7fe17c089c50 (2598 bytes) from ws/82.166.84.247:53645/sip next=(nil)
> nta.c:2880 agent_recv_request() nta: received INVITE sip:991234 at aaaa
> SIP/2.0 (CSeq 28092)
> nta.c:3174 agent_check_request_via() nta: Via check: received=82.166.84.247
> nta.c:3085 agent_recv_request() nta: INVITE (28092) going to a default leg
> nta.c:1350 set_timeout() nta: timer set to 2000 ms
> nua_server.c:102 nua_stack_process_request() nua:
> nua_stack_process_request: entering
> nua_stack.c:899 nh_create() nua: nh_create: entering
> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:280 soa_clone() soa_clone(static::0x7fe17c001930, 0x7fe17c001130,
> 0x7fe17c0cf210) called
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
> called
> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fe17c04a2e0)
> soa.c:1302 soa_init_offer_answer()
> soa_init_offer_answer(static::0x7fe17c0bf4d0) called
> soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fe17c0bf4d0,
> (nil), 0x7fe17c0be55f, 2039) called
> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fe17c0cf210): adding
> session usage
> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
> 82.166.84.247:53645
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
> 0x7fe17c0d4d90 0x7fe17c0190a0 140 (140)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
> 0x7fe17c0d4d90 0x7fe17c0be3ab 86 (86)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
> 0x7fe17c0d4d90 0x7fe17c01912c 101 (101)
> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 394 bytes of 394
> to ws/82.166.84.247:53645
> tport.c:3492 tport_send_msg() tport_vsend returned 394
> send 394 bytes to ws/[82.166.84.247]:53645 at 08:21:30.469343:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/WS
> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>    To: <sip:991234 at aaaa>
>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>    CSeq: 28092 INVITE
>    User-Agent:
> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
> nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (28092)
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_invite 100
> Trying
> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
> state changed: init -> received, received offer
> soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fe17c0bf4d0,
> [0x7fe1aa7b35d8], [0x7fe1aa7b35e0], [(nil)]) called
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 100
> Trying
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
> 2015-05-08 08:21:30.467711 [NOTICE] switch_channel.c:1075 New Channel
> sofia/internal/1000 at xxx.xxx.xxx.xxx [3ac3c622-f55b-11e4-a447-7d37723461ed]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_NEW
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8848
> sofia/internal/1000 at xxx.xxx.xxx.xxx receiving invite from
> 82.166.84.247:53645 version: 1.5.15b git b01352c 2015-04-23 04:33:08Z
> 64bit
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8960 IP 82.166.84.247 Approved
> by acl "domains[]". Access Granted.
> nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
> nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil)
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:10113 Setting NAT mode based on
> via received
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [received][100]
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6637 Remote SDP:
> v=0
> o=- 3843427479443760600 2 IN IP4 127.0.0.1
> s=Doubango Telecom - chrome
> t=0 0
> a=group:BUNDLE audio
> a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
> m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
> c=IN IP4 130.211.78.35
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10; useinbandfec=1
> a=rtpmap:103 ISAC/16000
> a=rtpmap:104 ISAC/32000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=rtcp:58209 IN IP4 130.211.78.35
> a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
> generation 0
> a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
> generation 0
> a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx
> raddr 10.0.0.10 rport 63888 generation 0
> a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx
> raddr 10.0.0.10 rport 63888 generation 0
> a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype
> active generation 0
> a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype
> active generation 0
> a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay raddr
> 82.166.84.247 rport 53792 generation 0
> a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay raddr
> 82.166.84.247 rport 53792 generation 0
> a=ice-ufrag:EDo5kr308/TXhitG
> a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
> a=ice-options:google-ice
> a=fingerprint:sha-256
> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
> a=setup:actpass
> a=mid:audio
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=rtcp-mux
> a=maxptime:60
> a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
> a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
> b8082781-4098-4ac5-8c96-2848d2e9e7e4
> a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
> a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6903 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_NEW -> CS_INIT
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:491 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) State NEW
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_INIT
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT
> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:87
> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA INIT
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:40
> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard INIT
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:48 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_INIT -> CS_ROUTING
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT going to sleep
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_ROUTING
> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:2204 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change DOWN -> RINGING
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING
> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:123
> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA ROUTING
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:166
> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard ROUTING
> 2015-05-08 08:21:30.467711 [INFO] mod_dialplan_xml.c:635 Processing
> asdasda <1000>->991234 in context public
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
> [public->cdquality_conferences_with_api] continue=false
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (FAIL)
> [cdquality_conferences_with_api] destination_number(991234) =~
> /^(75\d{4,36})$/ break=on-false
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
> [public->test_conferences] continue=false
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (PASS)
> [test_conferences] destination_number(991234) =~ /^(99\d{4,36})$/
> break=on-false
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action answer()
> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action
> conference(991234-${domain_name}@test)
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:216 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_ROUTING -> CS_EXECUTE
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING going to sleep
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_EXECUTE
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:535 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) State EXECUTE
> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:178
> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA EXECUTE
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:258
> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard EXECUTE
> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx answer()
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [opus:111:48000:60:0:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
> Sticking with opus at 48000h@20i
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
> Compare [opus:116:48000:20:0:1] ++++ is saved as a match
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [opus:111:48000:60:0:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [opus:111:48000:60:0:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [ISAC:103:16000:30:32000:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [ISAC:103:16000:30:32000:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [ISAC:103:16000:30:32000:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [ISAC:104:32000:30:32000:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [ISAC:104:32000:30:32000:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [ISAC:104:32000:30:32000:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [G722:9:8000:60:64000:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [G722:9:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [G722:9:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [PCMU:0:8000:60:64000:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [PCMU:0:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
> Sticking with PCMU at 8000h@20i
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [PCMU:0:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [PCMA:8:8000:60:64000:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [PCMA:8:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [PCMA:8:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
> Sticking with PCMA at 8000h@20i
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [CN:105:16000:60:0:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [CN:105:16000:60:0:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [CN:105:16000:60:0:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [CN:13:8000:60:0:1]/[opus:116:48000:20:0:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [CN:13:8000:60:0:1]/[PCMU:0:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
> Compare [CN:13:8000:60:0:1]/[PCMA:8:8000:20:64000:1]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3588 Set
> telephone-event payload to 126
> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set bitrate
> to local settings [-1000bps]
> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set bitrate
> to local settings [-1000bps]
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2507 Set Codec
> sofia/internal/1000 at xxx.xxx.xxx.xxx opus/48000 20 ms 960 samples 0 bits 1
> channels
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:111
> sofia/internal/1000 at xxx.xxx.xxx.xxx Original read codec set to opus:116
> 2015-05-08 08:21:30.467711 [WARNING] switch_core_media.c:2791 NO candidate
> ACL defined, Defaulting to wan.auto
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
> Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
> Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
> Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose audio
> Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
> Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose audio
> Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
> Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
> Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2953 setting
> remote audio ice addr to 82.166.84.247:63888 based on candidate
> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2978 setting
> remote rtcp audio addr to 82.166.84.247:63888 based on candidate
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3935 Set 2833 dtmf
> send/recv payload to 126
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5171 AUDIO RTP [
> sofia/internal/1000 at xxx.xxx.xxx.xxx] 172.30.0.219 port 19864 ->
> 82.166.84.247 port 63888 codec: 111 ms: 20
> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3559 Starting timer [soft]
> 960 bytes per 20ms
> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5345 Activating
> Audio ICE
> 2015-05-08 08:21:30.467711 [NOTICE] switch_rtp.c:4009 Activating RTP audio
> ICE: EDo5kr308/TXhitG:v9QogVGvZ8jGwYIi 82.166.84.247:63888
> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5388 Activating RTCP
> PORT 63888
> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3909 RTCP send rate is:
> 10000 and packet rate is: 20000 Remote Port: 63888
> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:2349 Setting RTCP remote
> addr to 82.166.84.247:63888
> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5396 Skipping RTCP
> ICE (Same as RTP)
> 2015-05-08 08:21:30.467711 [INFO] switch_rtp.c:3101 Activate RTP/RTCP
> audio DTLS client
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5469 Set 2833 dtmf
> send payload to 126
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5475 Set 2833 dtmf
> receive payload to 126
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5503 Set comfort
> noise payload to 106
> 2015-05-08 08:21:30.467711 [NOTICE] sofia_media.c:92 Pre-Answer
> sofia/internal/1000 at xxx.xxx.xxx.xxx!
> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3419 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change RINGING -> EARLY
> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:780 Local SDP
> sofia/internal/1000 at xxx.xxx.xxx.xxx:
> v=0
> o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
> s=FreeSWITCH
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
> m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 useinbandfec=1; minptime=10
> a=rtpmap:126 telephone-event/8000
> a=rtpmap:106 CN/8000
> a=ptime:20
> a=sendrecv
> a=fingerprint:sha-256
> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
> a=rtcp-mux
> a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
> a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
> a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
> a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
> a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
> a=ice-ufrag:v9QogVGvZ8jGwYIi
> a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
> a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
> generation 0
>
> nua.c:879 nua_respond() nua: nua_respond: entering
> nua_stack.c:529 nua_signal() nua(0x7fe17c0cf210): sent signal r_respond
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send signal
> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua_stack.c:573 nua_stack_signal() nua(0x7fe17c0cf210): recv signal
> r_respond 200 OK
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
> called
> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fe17c0bf4d0,
> (nil), 0x7fe190041d31, -1) called
> soa.c:890 soa_set_capability_sdp()
> soa_set_capability_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe190041d31, -1)
> called
> 2015-05-08 08:21:30.467711 [NOTICE] mod_dptools.c:1292 Channel [
> sofia/internal/1000 at xxx.xxx.xxx.xxx] has been answered
> nua_session.c:2320 nua_invite_server_respond() nua:
> nua_invite_server_respond: entering
> soa.c:1515 soa_generate_answer()
> soa_generate_answer(static::0x7fe17c0bf4d0) called
> soa_static.c:1146 offer_answer_step()
> soa_static_offer_answer_action(0x7fe17c0bf4d0, soa_generate_answer): called
> soa_static.c:1187 offer_answer_step() soa_static(0x7fe17c0bf4d0,
> soa_generate_answer): generating local description
> soa_static.c:1228 offer_answer_step() soa_static(0x7fe17c0bf4d0,
> soa_generate_answer): upgrade with remote description
> soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fe1aa7b1a30,
> 0x7fe17c026ac0, ""): called
> soa_static.c:1444 offer_answer_step() soa_static(0x7fe17c0bf4d0,
> soa_generate_answer): storing local description
> soa.c:1730 soa_activate() soa_activate(static::0x7fe17c0bf4d0, (nil))
> called
> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe17c0bf4d0,
> [(nil)], [0x7fe1aa7b3b58], [0x7fe1aa7b3b54]) called
> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
> 82.166.84.247:53645
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
> 0x7fe17c0d4d90 0x7fe17c0c5560 136 (136)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
> 0x7fe17c0d4d90 0x7fe17c0be3ab 63 (63)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
> 0x7fe17c0d4d90 0x7fe17c0c55e8 41 (41)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
> 0x7fe17c0d4d90 0x7fe17c0c5611 631 (631)
> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
> 0x7fe17c0d4d90 0x7fe17c0d3020 864 (864)
> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 1802 bytes of 1802
> to ws/82.166.84.247:53645
> tport.c:3492 tport_send_msg() tport_vsend returned 1802
> send 1802 bytes to ws/[82.166.84.247]:53645 at 08:21:30.479618:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/WS
> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>    CSeq: 28092 INVITE
>    Contact: <sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>    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: 864
>    Remote-Party-ID: "991234" <sip:991234 at aaaa
> >;party=calling;privacy=off;screen=no
>
>    v=0
>    o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>    s=FreeSWITCH
>    c=IN IP4 xxx.xxx.xxx.xxx
>    t=0 0
>    a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>    m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>    a=rtpmap:111 opus/48000/2
>    a=fmtp:111 useinbandfec=1; minptime=10
>    a=rtpmap:126 telephone-event/8000
>    a=rtpmap:106 CN/8000
>    a=ptime:20
>    a=fingerprint:sha-256
> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>    a=rtcp-mux
>    a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>    a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>    a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>    a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>    a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>    a=ice-ufrag:v9QogVGvZ8jGwYIi
>    a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>    a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
> generation 0
>    ------------------------------------------------------------------------
> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
> nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (28092)
> nta.c:1348 set_timeout() nta: timer shortened to 500 ms
> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3711 (
> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change EARLY -> ACTIVE
> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
> state changed: received -> completed, sent answer
> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe17c0bf4d0,
> [0x7fe1aa7b3c48], [0x7fe1aa7b3c50], [(nil)]) called
> soa.c:616 soa_get_params() soa_get_params(static::0x7fe17c0bf4d0, ...)
> called
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 200 OK
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [completed][200]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx
> conference(991234-172.30.0.219 at test)
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10364 using channel
> sound prefix: /usr/local/freeswitch/sounds/en/us/callie
> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:8991 Raw Codec
> Activation Success L16 at 48000hz 1 channel 20ms
> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:9037 Raw Codec
> Activation Success L16 at 16000hz 1 channel 20ms
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:221
> sofia/internal/1000 at xxx.xxx.xxx.xxx Push codec L16:100
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '0' to 'mute'
> 2015-05-08 08:21:30.467711 [INFO] switch_ivr_async.c:212 Digit parser
> mod_conference: Setting realm to 'conf'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding 0/conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010b8e0
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '*' to 'deaf mute'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding */conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010b910
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '9' to 'energy up'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding 9/conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010b940
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '8' to 'energy equ'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding 8/conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010b970
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '7' to 'energy dn'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding 7/conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010b9a0
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '3' to 'vol talk up'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding 3/conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010b9d0
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '2' to 'vol talk zero'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding 2/conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010ba00
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '1' to 'vol talk dn'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding 1/conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010ba30
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '6' to 'vol listen up'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding 6/conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010ba60
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '5' to 'vol listen zero'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding 5/conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010ba90
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '4' to 'vol listen dn'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding 4/conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010bac0
> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '#' to 'hangup'
> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
> mod_conference: binding #/conf/0 callback: 0x7fe1a92764e0 data:
> 0x7fe19010baf0
> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send signal
> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:4765 Setup timer soft
> success interval: 20  samples: 960
> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:3043 Setup timer
> success interval: 30  samples: 480
> 2015-05-08 08:21:30.507713 [DEBUG] mod_local_stream.c:498 Opening Stream
> [moh/16000] 16000hz
> 2015-05-08 08:21:30.507713 [NOTICE] switch_core_io.c:1261 Activating write
> resampler
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
> 0x7fe17c0c6ab0 from (ws/82.166.84.247:53645) has 550 bytes, veclen = 1
> recv 550 bytes from ws/[82.166.84.247]:53645 at 08:21:30.662998:
>    ------------------------------------------------------------------------
>    ACK sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0
>    Via: SIP/2.0/WS
> df7jal23ls0d.invalid;branch=z9hG4bK30nT8FwJ3EqSVIbdgFvT;rport
>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>    Contact: "asdasda"<
> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
> >;+g.oma.sip-im;language="en,fr"
>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>    CSeq: 28092 ACK
>    Content-Length: 0
>    Max-Forwards: 70
>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>    Organization: Doubango Telecom
>
>    ------------------------------------------------------------------------
> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
> 0x7fe17c0c6ab0 (550 bytes) from ws/82.166.84.247:53645/sip next=(nil)
> nta.c:2880 agent_recv_request() nta: received ACK
> sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0 (CSeq 28092)
> nta.c:3174 agent_check_request_via() nta: Via check: received=82.166.84.247
> nta.c:3019 agent_recv_request() nta: ACK (28092) is going to INVITE (28092)
> nua_session.c:2569 process_ack_or_cancel() nua: process_ack_or_cancel:
> entering
> soa.c:1214 soa_clear_remote_sdp()
> soa_clear_remote_sdp(static::0x7fe17c0bf4d0) called
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_ack 200 OK
> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
> state changed: completed -> ready
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 200 OK
> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_active 200
> Call active
> nta.c:5744 incoming_free() nta: incoming_free(0x7fe17c024090)
> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset timer
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-05-08 08:21:30.667707 [DEBUG] sofia.c:6627 Channel
> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [ready][200]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nta.c:1289 agent_timer() nta: timer not set
> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2924 Changing audio DTLS
> state from HANDSHAKE to SETUP
> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2832 audio Fingerprint
> Verified.
> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3374 Activating Audio
> Secure RTP SEND
> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3352 Activating Audio
> Secure RTP RECV
> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2872 Changing audio DTLS
> state from SETUP to READY
> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure Type:
> srtp:dtls:AES_CM_128_HMAC_SHA1_80
> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure Type:
> srtp:dtls:AES_CM_128_HMAC_SHA1_80
> 2015-05-08 08:22:02.087708 [DEBUG] switch_rtp.c:1937 rtcp_stats_init:
> ssrc[-195423717] base_seq[29507]
>
> Notice that the INVITE was received at 08:21:30 while DTLS was READY
> at 2015-05-08 08:22:02 which means that it took 32 seconds to voice. Again,
> if I am not using TURN/STUN, the whole process is pretty quick (2 seconds).
>
> Also, notice:
> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
>
> Which means that the media is relayed via the TURN server (TURN server IP
> is 130.211.78.35)...
>
> Any idea?
>
> Thanks,
> Adam
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From adam.ben.ayoun1 at gmail.com  Fri May  8 19:43:38 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Fri, 8 May 2015 18:43:38 +0300
Subject: [Freeswitch-users] Trickle ICE with SIP
In-Reply-To: <C56F753C-D2E5-4072-BDC2-43BCE6E367AB@jerris.com>
References: <CADNqFCmOfu_Nq56N9rc6H2DD=9OG7=MTqB1D4dT+kJ9GFcaBSA@mail.gmail.com>
	<C56F753C-D2E5-4072-BDC2-43BCE6E367AB@jerris.com>
Message-ID: <CADNqFCkDy2OtzeTqF_mDzW4hA75qambrEGxs5JpwL9QwYseh9g@mail.gmail.com>

Well, it can get up to 30 seconds when using TURN, in other cases it's
usually 2-4 seconds.

On 8 May 2015 at 18:35, Michael Jerris <mike at jerris.com> wrote:

> >From your other post you said thats only when using turn?
>
> > On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
> wrote:
> >
> > Hi,
> >
> > We are currently using Freeswitch for voice conferencing, we use SIP for
> signalling. Call setup times are really slow at times (3sec-30sec), can we
> somehow do Trickle ICE with Freeswitch?
> >
> > Thanks,
> > Adam
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From danb.lists at gmail.com  Fri May  8 19:44:42 2015
From: danb.lists at gmail.com (DanB)
Date: Fri, 08 May 2015 17:44:42 +0200
Subject: [Freeswitch-users] IvrMenu - setting channel variables before
	calling exec-app/transfer
In-Reply-To: <mailman.515.1431099645.10851.freeswitch-users@lists.freeswitch.org>
References: <mailman.515.1431099645.10851.freeswitch-users@lists.freeswitch.org>
Message-ID: <554CD9EA.3060405@gmail.com>

@covici: thanks, I was hoping in a more elegant solution since every transfer will produce a request to http server for dialplan.

@Michael: not before the ivr since the vars I need to set depend on the menu selected (eg: call will be forwarded when he presses key so the caller becomes the previous called).

DanB



From vipkilla at gmail.com  Fri May  8 19:50:22 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 8 May 2015 11:50:22 -0400
Subject: [Freeswitch-users] Json API for Voicemail?
In-Reply-To: <CALJtzxDy-3-A0L62KmmhSzqyv=oP68ZKTNKxbkUeq1FTsp8qjQ@mail.gmail.com>
References: <CALJtzxDy-3-A0L62KmmhSzqyv=oP68ZKTNKxbkUeq1FTsp8qjQ@mail.gmail.com>
Message-ID: <CAC-LwPNBY4_N7J5P=3Y5Sv35o017uXrY9hMKWU8MO1SLeJfOXg@mail.gmail.com>

Hi,
Try this
vm_fsdb_msg_list

Thanks,
V

On Wed, May 6, 2015 at 2:50 PM, Alejandro <cdgraff at gmail.com> wrote:

> Hi Guys,
>
> Is there some module to list the Voicemail messages in Json or XML.
>
> I has enabled RPC module and see the voicemail and listen into the native
> website... but I like to create custom view and list the message true some
> API.
>
> Some advice? or link to share?
>
> Thanks in advance!
> Alejandro
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From adam.ben.ayoun1 at gmail.com  Fri May  8 19:52:35 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Fri, 8 May 2015 18:52:35 +0300
Subject: [Freeswitch-users] WebRTC slow connection time
In-Reply-To: <CADNqFC=ba=eD8DCAicLLNx6_=xw0JRFYSdM+UDo2D35OGLR_+g@mail.gmail.com>
References: <CADNqFCky4iMXF-5vukKcqmT8=Q+anVHdtn+0zueV+4yJ3GUC6g@mail.gmail.com>
	<E6A2DD46-EA8B-400A-A58D-AA9547DDA682@jerris.com>
	<CADNqFC=ba=eD8DCAicLLNx6_=xw0JRFYSdM+UDo2D35OGLR_+g@mail.gmail.com>
Message-ID: <CADNqFC=EyojairQTzah9PNX0Wwog5jj4Tcudi_UvmeCTJbtCzA@mail.gmail.com>

Is there an easier way to deal with clients which can only do TCP on port
443/80 for example?

On 8 May 2015 at 18:40, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:

> I am using TURN as a TCP fallback (when UDP is blocked).
>
> On 8 May 2015 at 18:34, Michael Jerris <mike at jerris.com> wrote:
>
>> If you have freeswitch on a public address, why would you ever use TURN?
>>
>> On May 8, 2015, at 4:27 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>> wrote:
>>
>> Hi guys,
>>
>> I am using Freeswitch from master (month old). I tried several WebRTC
>> clients (my own test app on Android and sipml5) and on certain WiFi's I am
>> getting connection time up to 35 seconds when using STUN and TURN servers.
>> Also, when I am using TURN and STUN, Freeswitch chooses the TURN candidate
>> although as far as connectivity, when I am not using STUN and TURN I am
>> connecting successfully after ~2 seconds (so no real need for TURN).
>> Attaching the SIP trace from Freeswitch:
>>
>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>> entering
>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
>> called
>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>> entering
>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe18c001930, ...)
>> called
>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>> entering
>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
>> called
>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>> entering
>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe184001930, ...)
>> called
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>> entering
>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe188001930, ...)
>> called
>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
>> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
>> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
>> 0x7fe17c089c50 from (ws/82.166.84.247:53645) has 2598 bytes, veclen = 1
>> recv 2598 bytes from ws/[82.166.84.247]:53645 at 08:21:30.468866:
>>
>>  ------------------------------------------------------------------------
>>    INVITE sip:991234 at aaaa SIP/2.0
>>    Via: SIP/2.0/WS
>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport
>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>    To: <sip:991234 at aaaa>
>>    Contact: "asdasda"<
>> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
>> >;+g.oma.sip-im;language="en,fr"
>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>    CSeq: 28092 INVITE
>>    Content-Type: application/sdp
>>    Content-Length: 2039
>>    Max-Forwards: 70
>>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>>    Organization: Doubango Telecom
>>
>>    v=0
>>    o=- 3843427479443760600 2 IN IP4 127.0.0.1
>>    s=Doubango Telecom - chrome
>>    t=0 0
>>    a=group:BUNDLE audio
>>    a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>    m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>    c=IN IP4 130.211.78.35
>>    a=rtcp:58209 IN IP4 130.211.78.35
>>    a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
>> generation 0
>>    a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
>> generation 0
>>    a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx
>> raddr 10.0.0.10 rport 63888 generation 0
>>    a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx
>> raddr 10.0.0.10 rport 63888 generation 0
>>    a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype
>> active generation 0
>>    a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype
>> active generation 0
>>    a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay
>> raddr 82.166.84.247 rport 53792 generation 0
>>    a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay
>> raddr 82.166.84.247 rport 53792 generation 0
>>    a=ice-ufrag:EDo5kr308/TXhitG
>>    a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>>    a=ice-options:google-ice
>>    a=fingerprint:sha-256
>> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>>    a=setup:actpass
>>    a=mid:audio
>>    a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>    a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>    a=sendrecv
>>    a=rtcp-mux
>>    a=rtpmap:111 opus/48000/2
>>    a=fmtp:111 minptime=10; useinbandfec=1
>>    a=rtpmap:103 ISAC/16000
>>    a=rtpmap:104 ISAC/32000
>>    a=rtpmap:9 G722/8000
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:8 PCMA/8000
>>    a=rtpmap:106 CN/32000
>>    a=rtpmap:105 CN/16000
>>    a=rtpmap:13 CN/8000
>>    a=rtpmap:126 telephone-event/8000
>>    a=maxptime:60
>>    a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>>    a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>> b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>    a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>    a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>
>>  ------------------------------------------------------------------------
>> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
>> 0x7fe17c089c50 (2598 bytes) from ws/82.166.84.247:53645/sip next=(nil)
>> nta.c:2880 agent_recv_request() nta: received INVITE sip:991234 at aaaa
>> SIP/2.0 (CSeq 28092)
>> nta.c:3174 agent_check_request_via() nta: Via check:
>> received=82.166.84.247
>> nta.c:3085 agent_recv_request() nta: INVITE (28092) going to a default leg
>> nta.c:1350 set_timeout() nta: timer set to 2000 ms
>> nua_server.c:102 nua_stack_process_request() nua:
>> nua_stack_process_request: entering
>> nua_stack.c:899 nh_create() nua: nh_create: entering
>> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>> entering
>> soa.c:280 soa_clone() soa_clone(static::0x7fe17c001930, 0x7fe17c001130,
>> 0x7fe17c0cf210) called
>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
>> called
>> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fe17c04a2e0)
>> soa.c:1302 soa_init_offer_answer()
>> soa_init_offer_answer(static::0x7fe17c0bf4d0) called
>> soa.c:1171 soa_set_remote_sdp()
>> soa_set_remote_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe17c0be55f, 2039)
>> called
>> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fe17c0cf210): adding
>> session usage
>> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
>> 82.166.84.247:53645
>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>> 0x7fe17c0d4d90 0x7fe17c0190a0 140 (140)
>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>> 0x7fe17c0d4d90 0x7fe17c0be3ab 86 (86)
>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>> 0x7fe17c0d4d90 0x7fe17c01912c 101 (101)
>> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 394 bytes of 394
>> to ws/82.166.84.247:53645
>> tport.c:3492 tport_send_msg() tport_vsend returned 394
>> send 394 bytes to ws/[82.166.84.247]:53645 at 08:21:30.469343:
>>
>>  ------------------------------------------------------------------------
>>    SIP/2.0 100 Trying
>>    Via: SIP/2.0/WS
>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>    To: <sip:991234 at aaaa>
>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>    CSeq: 28092 INVITE
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>>    Content-Length: 0
>>
>>
>>  ------------------------------------------------------------------------
>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>> timer
>> nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (28092)
>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_invite 100
>> Trying
>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
>> state changed: init -> received, received offer
>> soa.c:1098 soa_get_remote_sdp()
>> soa_get_remote_sdp(static::0x7fe17c0bf4d0, [0x7fe1aa7b35d8],
>> [0x7fe1aa7b35e0], [(nil)]) called
>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 100
>> Trying
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>> timer
>> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
>> 2015-05-08 08:21:30.467711 [NOTICE] switch_channel.c:1075 New Channel
>> sofia/internal/1000 at xxx.xxx.xxx.xxx
>> [3ac3c622-f55b-11e4-a447-7d37723461ed]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal
>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal
>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_NEW
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8848
>> sofia/internal/1000 at xxx.xxx.xxx.xxx receiving invite from
>> 82.166.84.247:53645 version: 1.5.15b git b01352c 2015-04-23 04:33:08Z
>> 64bit
>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8960 IP 82.166.84.247 Approved
>> by acl "domains[]". Access Granted.
>> nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
>> nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle
>> (nil)
>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:10113 Setting NAT mode based
>> on via received
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [received][100]
>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6637 Remote SDP:
>> v=0
>> o=- 3843427479443760600 2 IN IP4 127.0.0.1
>> s=Doubango Telecom - chrome
>> t=0 0
>> a=group:BUNDLE audio
>> a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>> m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>> c=IN IP4 130.211.78.35
>> a=rtpmap:111 opus/48000/2
>> a=fmtp:111 minptime=10; useinbandfec=1
>> a=rtpmap:103 ISAC/16000
>> a=rtpmap:104 ISAC/32000
>> a=rtpmap:9 G722/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:106 CN/32000
>> a=rtpmap:105 CN/16000
>> a=rtpmap:13 CN/8000
>> a=rtpmap:126 telephone-event/8000
>> a=rtcp:58209 IN IP4 130.211.78.35
>> a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
>> generation 0
>> a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
>> generation 0
>> a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx
>> raddr 10.0.0.10 rport 63888 generation 0
>> a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx
>> raddr 10.0.0.10 rport 63888 generation 0
>> a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype
>> active generation 0
>> a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype
>> active generation 0
>> a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay raddr
>> 82.166.84.247 rport 53792 generation 0
>> a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay raddr
>> 82.166.84.247 rport 53792 generation 0
>> a=ice-ufrag:EDo5kr308/TXhitG
>> a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>> a=ice-options:google-ice
>> a=fingerprint:sha-256
>> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>> a=setup:actpass
>> a=mid:audio
>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>> a=rtcp-mux
>> a=maxptime:60
>> a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>> a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>> b8082781-4098-4ac5-8c96-2848d2e9e7e4
>> a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>> a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>
>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6903 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_NEW -> CS_INIT
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal
>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:491 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State NEW
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_INIT
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT
>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:87
>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA INIT
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:40
>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard INIT
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:48 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_INIT -> CS_ROUTING
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal
>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT going to sleep
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_ROUTING
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:2204 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change DOWN -> RINGING
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING
>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:123
>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA ROUTING
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:166
>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard ROUTING
>> 2015-05-08 08:21:30.467711 [INFO] mod_dialplan_xml.c:635 Processing
>> asdasda <1000>->991234 in context public
>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
>> [public->cdquality_conferences_with_api] continue=false
>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (FAIL)
>> [cdquality_conferences_with_api] destination_number(991234) =~
>> /^(75\d{4,36})$/ break=on-false
>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
>> [public->test_conferences] continue=false
>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (PASS)
>> [test_conferences] destination_number(991234) =~ /^(99\d{4,36})$/
>> break=on-false
>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action answer()
>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action
>> conference(991234-${domain_name}@test)
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:216 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_ROUTING ->
>> CS_EXECUTE
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send signal
>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING going to sleep
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_EXECUTE
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:535 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State EXECUTE
>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:178
>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA EXECUTE
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:258
>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard EXECUTE
>> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx answer()
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [opus:111:48000:60:0:1]/[opus:116:48000:20:0:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
>> Sticking with opus at 48000h@20i
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
>> Compare [opus:116:48000:20:0:1] ++++ is saved as a match
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [opus:111:48000:60:0:1]/[PCMU:0:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [opus:111:48000:60:0:1]/[PCMA:8:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [ISAC:103:16000:30:32000:1]/[opus:116:48000:20:0:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [ISAC:103:16000:30:32000:1]/[PCMU:0:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [ISAC:103:16000:30:32000:1]/[PCMA:8:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [ISAC:104:32000:30:32000:1]/[opus:116:48000:20:0:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [ISAC:104:32000:30:32000:1]/[PCMU:0:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [ISAC:104:32000:30:32000:1]/[PCMA:8:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [G722:9:8000:60:64000:1]/[opus:116:48000:20:0:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [G722:9:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [G722:9:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [PCMU:0:8000:60:64000:1]/[opus:116:48000:20:0:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [PCMU:0:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
>> Sticking with PCMU at 8000h@20i
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
>> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [PCMU:0:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [PCMA:8:8000:60:64000:1]/[opus:116:48000:20:0:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [PCMA:8:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [PCMA:8:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
>> Sticking with PCMA at 8000h@20i
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [CN:105:16000:60:0:1]/[opus:116:48000:20:0:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [CN:105:16000:60:0:1]/[PCMU:0:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [CN:105:16000:60:0:1]/[PCMA:8:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [CN:13:8000:60:0:1]/[opus:116:48000:20:0:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [CN:13:8000:60:0:1]/[PCMU:0:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>> Compare [CN:13:8000:60:0:1]/[PCMA:8:8000:20:64000:1]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3588 Set
>> telephone-event payload to 126
>> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set
>> bitrate to local settings [-1000bps]
>> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set
>> bitrate to local settings [-1000bps]
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2507 Set Codec
>> sofia/internal/1000 at xxx.xxx.xxx.xxx opus/48000 20 ms 960 samples 0 bits
>> 1 channels
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:111
>> sofia/internal/1000 at xxx.xxx.xxx.xxx Original read codec set to opus:116
>> 2015-05-08 08:21:30.467711 [WARNING] switch_core_media.c:2791 NO
>> candidate ACL defined, Defaulting to wan.auto
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>> Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>> Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>> Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose audio
>> Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>> Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose audio
>> Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>> Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>> Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>> Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2953 setting
>> remote audio ice addr to 82.166.84.247:63888 based on candidate
>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2978 setting
>> remote rtcp audio addr to 82.166.84.247:63888 based on candidate
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3935 Set 2833 dtmf
>> send/recv payload to 126
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5171 AUDIO RTP [
>> sofia/internal/1000 at xxx.xxx.xxx.xxx] 172.30.0.219 port 19864 ->
>> 82.166.84.247 port 63888 codec: 111 ms: 20
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3559 Starting timer
>> [soft] 960 bytes per 20ms
>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5345 Activating
>> Audio ICE
>> 2015-05-08 08:21:30.467711 [NOTICE] switch_rtp.c:4009 Activating RTP
>> audio ICE: EDo5kr308/TXhitG:v9QogVGvZ8jGwYIi 82.166.84.247:63888
>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5388 Activating
>> RTCP PORT 63888
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3909 RTCP send rate is:
>> 10000 and packet rate is: 20000 Remote Port: 63888
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:2349 Setting RTCP remote
>> addr to 82.166.84.247:63888
>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5396 Skipping RTCP
>> ICE (Same as RTP)
>> 2015-05-08 08:21:30.467711 [INFO] switch_rtp.c:3101 Activate RTP/RTCP
>> audio DTLS client
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5469 Set 2833 dtmf
>> send payload to 126
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5475 Set 2833 dtmf
>> receive payload to 126
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5503 Set comfort
>> noise payload to 106
>> 2015-05-08 08:21:30.467711 [NOTICE] sofia_media.c:92 Pre-Answer
>> sofia/internal/1000 at xxx.xxx.xxx.xxx!
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3419 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change RINGING -> EARLY
>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:780 Local SDP
>> sofia/internal/1000 at xxx.xxx.xxx.xxx:
>> v=0
>> o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>> s=FreeSWITCH
>> c=IN IP4 xxx.xxx.xxx.xxx
>> t=0 0
>> a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>> m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>> a=rtpmap:111 opus/48000/2
>> a=fmtp:111 useinbandfec=1; minptime=10
>> a=rtpmap:126 telephone-event/8000
>> a=rtpmap:106 CN/8000
>> a=ptime:20
>> a=sendrecv
>> a=fingerprint:sha-256
>> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>> a=rtcp-mux
>> a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>> a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>> a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>> a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>> a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>> a=ice-ufrag:v9QogVGvZ8jGwYIi
>> a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>> a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
>> generation 0
>>
>> nua.c:879 nua_respond() nua: nua_respond: entering
>> nua_stack.c:529 nua_signal() nua(0x7fe17c0cf210): sent signal r_respond
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send signal
>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>> nua_stack.c:573 nua_stack_signal() nua(0x7fe17c0cf210): recv signal
>> r_respond 200 OK
>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>> entering
>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
>> called
>> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fe17c0bf4d0,
>> (nil), 0x7fe190041d31, -1) called
>> soa.c:890 soa_set_capability_sdp()
>> soa_set_capability_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe190041d31, -1)
>> called
>> 2015-05-08 08:21:30.467711 [NOTICE] mod_dptools.c:1292 Channel [
>> sofia/internal/1000 at xxx.xxx.xxx.xxx] has been answered
>> nua_session.c:2320 nua_invite_server_respond() nua:
>> nua_invite_server_respond: entering
>> soa.c:1515 soa_generate_answer()
>> soa_generate_answer(static::0x7fe17c0bf4d0) called
>> soa_static.c:1146 offer_answer_step()
>> soa_static_offer_answer_action(0x7fe17c0bf4d0, soa_generate_answer): called
>> soa_static.c:1187 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>> soa_generate_answer): generating local description
>> soa_static.c:1228 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>> soa_generate_answer): upgrade with remote description
>> soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fe1aa7b1a30,
>> 0x7fe17c026ac0, ""): called
>> soa_static.c:1444 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>> soa_generate_answer): storing local description
>> soa.c:1730 soa_activate() soa_activate(static::0x7fe17c0bf4d0, (nil))
>> called
>> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe17c0bf4d0,
>> [(nil)], [0x7fe1aa7b3b58], [0x7fe1aa7b3b54]) called
>> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
>> 82.166.84.247:53645
>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>> 0x7fe17c0d4d90 0x7fe17c0c5560 136 (136)
>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>> 0x7fe17c0d4d90 0x7fe17c0be3ab 63 (63)
>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>> 0x7fe17c0d4d90 0x7fe17c0c55e8 41 (41)
>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>> 0x7fe17c0d4d90 0x7fe17c0c5611 631 (631)
>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>> 0x7fe17c0d4d90 0x7fe17c0d3020 864 (864)
>> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 1802 bytes of
>> 1802 to ws/82.166.84.247:53645
>> tport.c:3492 tport_send_msg() tport_vsend returned 1802
>> send 1802 bytes to ws/[82.166.84.247]:53645 at 08:21:30.479618:
>>
>>  ------------------------------------------------------------------------
>>    SIP/2.0 200 OK
>>    Via: SIP/2.0/WS
>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>    CSeq: 28092 INVITE
>>    Contact: <sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp>
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>>    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: 864
>>    Remote-Party-ID: "991234" <sip:991234 at aaaa
>> >;party=calling;privacy=off;screen=no
>>
>>    v=0
>>    o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>>    s=FreeSWITCH
>>    c=IN IP4 xxx.xxx.xxx.xxx
>>    t=0 0
>>    a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>    m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>>    a=rtpmap:111 opus/48000/2
>>    a=fmtp:111 useinbandfec=1; minptime=10
>>    a=rtpmap:126 telephone-event/8000
>>    a=rtpmap:106 CN/8000
>>    a=ptime:20
>>    a=fingerprint:sha-256
>> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>>    a=rtcp-mux
>>    a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>>    a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>>    a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>>    a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>    a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>>    a=ice-ufrag:v9QogVGvZ8jGwYIi
>>    a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>>    a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
>> generation 0
>>
>>  ------------------------------------------------------------------------
>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>> timer
>> nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (28092)
>> nta.c:1348 set_timeout() nta: timer shortened to 500 ms
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3711 (
>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change EARLY -> ACTIVE
>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
>> state changed: received -> completed, sent answer
>> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe17c0bf4d0,
>> [0x7fe1aa7b3c48], [0x7fe1aa7b3c50], [(nil)]) called
>> soa.c:616 soa_get_params() soa_get_params(static::0x7fe17c0bf4d0, ...)
>> called
>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 200
>> OK
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send signal
>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [completed][200]
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx
>> conference(991234-172.30.0.219 at test)
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10364 using channel
>> sound prefix: /usr/local/freeswitch/sounds/en/us/callie
>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:8991 Raw Codec
>> Activation Success L16 at 48000hz 1 channel 20ms
>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:9037 Raw Codec
>> Activation Success L16 at 16000hz 1 channel 20ms
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:221
>> sofia/internal/1000 at xxx.xxx.xxx.xxx Push codec L16:100
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '0' to 'mute'
>> 2015-05-08 08:21:30.467711 [INFO] switch_ivr_async.c:212 Digit parser
>> mod_conference: Setting realm to 'conf'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding 0/conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010b8e0
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '*' to 'deaf mute'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding */conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010b910
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '9' to 'energy up'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding 9/conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010b940
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '8' to 'energy equ'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding 8/conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010b970
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '7' to 'energy dn'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding 7/conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010b9a0
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '3' to 'vol talk up'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding 3/conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010b9d0
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '2' to 'vol talk zero'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding 2/conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010ba00
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '1' to 'vol talk dn'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding 1/conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010ba30
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '6' to 'vol listen up'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding 6/conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010ba60
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '5' to 'vol listen zero'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding 5/conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010ba90
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '4' to 'vol listen dn'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding 4/conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010bac0
>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '#' to 'hangup'
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>> mod_conference: binding #/conf/0 callback: 0x7fe1a92764e0 data:
>> 0x7fe19010baf0
>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send signal
>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:4765 Setup timer soft
>> success interval: 20  samples: 960
>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:3043 Setup timer
>> success interval: 30  samples: 480
>> 2015-05-08 08:21:30.507713 [DEBUG] mod_local_stream.c:498 Opening Stream
>> [moh/16000] 16000hz
>> 2015-05-08 08:21:30.507713 [NOTICE] switch_core_io.c:1261 Activating
>> write resampler
>> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
>> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
>> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
>> 0x7fe17c0c6ab0 from (ws/82.166.84.247:53645) has 550 bytes, veclen = 1
>> recv 550 bytes from ws/[82.166.84.247]:53645 at 08:21:30.662998:
>>
>>  ------------------------------------------------------------------------
>>    ACK sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0
>>    Via: SIP/2.0/WS
>> df7jal23ls0d.invalid;branch=z9hG4bK30nT8FwJ3EqSVIbdgFvT;rport
>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>>    Contact: "asdasda"<
>> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
>> >;+g.oma.sip-im;language="en,fr"
>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>    CSeq: 28092 ACK
>>    Content-Length: 0
>>    Max-Forwards: 70
>>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>>    Organization: Doubango Telecom
>>
>>
>>  ------------------------------------------------------------------------
>> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
>> 0x7fe17c0c6ab0 (550 bytes) from ws/82.166.84.247:53645/sip next=(nil)
>> nta.c:2880 agent_recv_request() nta: received ACK
>> sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0 (CSeq 28092)
>> nta.c:3174 agent_check_request_via() nta: Via check:
>> received=82.166.84.247
>> nta.c:3019 agent_recv_request() nta: ACK (28092) is going to INVITE
>> (28092)
>> nua_session.c:2569 process_ack_or_cancel() nua: process_ack_or_cancel:
>> entering
>> soa.c:1214 soa_clear_remote_sdp()
>> soa_clear_remote_sdp(static::0x7fe17c0bf4d0) called
>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_ack 200 OK
>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
>> state changed: completed -> ready
>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 200
>> OK
>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_active 200
>> Call active
>> nta.c:5744 incoming_free() nta: incoming_free(0x7fe17c024090)
>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>> timer
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal
>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal
>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>> entering
>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send signal
>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> 2015-05-08 08:21:30.667707 [DEBUG] sofia.c:6627 Channel
>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [ready][200]
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>> nta.c:1289 agent_timer() nta: timer not set
>> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
>> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2924 Changing audio DTLS
>> state from HANDSHAKE to SETUP
>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2832 audio Fingerprint
>> Verified.
>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3374 Activating Audio
>> Secure RTP SEND
>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3352 Activating Audio
>> Secure RTP RECV
>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2872 Changing audio DTLS
>> state from SETUP to READY
>> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure Type:
>> srtp:dtls:AES_CM_128_HMAC_SHA1_80
>> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure Type:
>> srtp:dtls:AES_CM_128_HMAC_SHA1_80
>> 2015-05-08 08:22:02.087708 [DEBUG] switch_rtp.c:1937 rtcp_stats_init:
>> ssrc[-195423717] base_seq[29507]
>>
>> Notice that the INVITE was received at 08:21:30 while DTLS was READY
>> at 2015-05-08 08:22:02 which means that it took 32 seconds to voice. Again,
>> if I am not using TURN/STUN, the whole process is pretty quick (2 seconds).
>>
>> Also, notice:
>> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
>> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
>>
>> Which means that the media is relayed via the TURN server (TURN server IP
>> is 130.211.78.35)...
>>
>> Any idea?
>>
>> Thanks,
>> Adam
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From gmaruzz at gmail.com  Fri May  8 19:58:39 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 8 May 2015 17:58:39 +0200
Subject: [Freeswitch-users] WebRTC slow connection time
In-Reply-To: <CADNqFC=EyojairQTzah9PNX0Wwog5jj4Tcudi_UvmeCTJbtCzA@mail.gmail.com>
References: <CADNqFCky4iMXF-5vukKcqmT8=Q+anVHdtn+0zueV+4yJ3GUC6g@mail.gmail.com>
	<E6A2DD46-EA8B-400A-A58D-AA9547DDA682@jerris.com>
	<CADNqFC=ba=eD8DCAicLLNx6_=xw0JRFYSdM+UDo2D35OGLR_+g@mail.gmail.com>
	<CADNqFC=EyojairQTzah9PNX0Wwog5jj4Tcudi_UvmeCTJbtCzA@mail.gmail.com>
Message-ID: <CALXCt0r_CQMFG5Ldjj9K1Q5g9m_t+JN6HShQgbbeXTzN5nkUqA@mail.gmail.com>

Btw, that (accomodating clients that only do tcp 80/443) is the biggest and
almost unique reason of Skype success...

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 8, 2015 5:53 PM, "Adam Ben-Ayoun" <adam.ben.ayoun1 at gmail.com> wrote:

> Is there an easier way to deal with clients which can only do TCP on port
> 443/80 for example?
>
> On 8 May 2015 at 18:40, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:
>
>> I am using TURN as a TCP fallback (when UDP is blocked).
>>
>> On 8 May 2015 at 18:34, Michael Jerris <mike at jerris.com> wrote:
>>
>>> If you have freeswitch on a public address, why would you ever use TURN?
>>>
>>> On May 8, 2015, at 4:27 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>>> wrote:
>>>
>>> Hi guys,
>>>
>>> I am using Freeswitch from master (month old). I tried several WebRTC
>>> clients (my own test app on Android and sipml5) and on certain WiFi's I am
>>> getting connection time up to 35 seconds when using STUN and TURN servers.
>>> Also, when I am using TURN and STUN, Freeswitch chooses the TURN candidate
>>> although as far as connectivity, when I am not using STUN and TURN I am
>>> connecting successfully after ~2 seconds (so no real need for TURN).
>>> Attaching the SIP trace from Freeswitch:
>>>
>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>> entering
>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
>>> called
>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>> entering
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>> entering
>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe18c001930, ...)
>>> called
>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>> entering
>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
>>> called
>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>> entering
>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe184001930, ...)
>>> called
>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>> entering
>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>> entering
>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>> entering
>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe188001930, ...)
>>> called
>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>> entering
>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>> entering
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
>>> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
>>> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
>>> 0x7fe17c089c50 from (ws/82.166.84.247:53645) has 2598 bytes, veclen = 1
>>> recv 2598 bytes from ws/[82.166.84.247]:53645 at 08:21:30.468866:
>>>
>>>  ------------------------------------------------------------------------
>>>    INVITE sip:991234 at aaaa SIP/2.0
>>>    Via: SIP/2.0/WS
>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport
>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>    To: <sip:991234 at aaaa>
>>>    Contact: "asdasda"<
>>> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
>>> >;+g.oma.sip-im;language="en,fr"
>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>    CSeq: 28092 INVITE
>>>    Content-Type: application/sdp
>>>    Content-Length: 2039
>>>    Max-Forwards: 70
>>>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>>>    Organization: Doubango Telecom
>>>
>>>    v=0
>>>    o=- 3843427479443760600 2 IN IP4 127.0.0.1
>>>    s=Doubango Telecom - chrome
>>>    t=0 0
>>>    a=group:BUNDLE audio
>>>    a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>    m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>>    c=IN IP4 130.211.78.35
>>>    a=rtcp:58209 IN IP4 130.211.78.35
>>>    a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
>>> generation 0
>>>    a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
>>> generation 0
>>>    a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx
>>> raddr 10.0.0.10 rport 63888 generation 0
>>>    a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx
>>> raddr 10.0.0.10 rport 63888 generation 0
>>>    a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype
>>> active generation 0
>>>    a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype
>>> active generation 0
>>>    a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay
>>> raddr 82.166.84.247 rport 53792 generation 0
>>>    a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay
>>> raddr 82.166.84.247 rport 53792 generation 0
>>>    a=ice-ufrag:EDo5kr308/TXhitG
>>>    a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>>>    a=ice-options:google-ice
>>>    a=fingerprint:sha-256
>>> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>>>    a=setup:actpass
>>>    a=mid:audio
>>>    a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>>    a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>    a=sendrecv
>>>    a=rtcp-mux
>>>    a=rtpmap:111 opus/48000/2
>>>    a=fmtp:111 minptime=10; useinbandfec=1
>>>    a=rtpmap:103 ISAC/16000
>>>    a=rtpmap:104 ISAC/32000
>>>    a=rtpmap:9 G722/8000
>>>    a=rtpmap:0 PCMU/8000
>>>    a=rtpmap:8 PCMA/8000
>>>    a=rtpmap:106 CN/32000
>>>    a=rtpmap:105 CN/16000
>>>    a=rtpmap:13 CN/8000
>>>    a=rtpmap:126 telephone-event/8000
>>>    a=maxptime:60
>>>    a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>>>    a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>> b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>    a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>    a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>
>>>  ------------------------------------------------------------------------
>>> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
>>> 0x7fe17c089c50 (2598 bytes) from ws/82.166.84.247:53645/sip next=(nil)
>>> nta.c:2880 agent_recv_request() nta: received INVITE sip:991234 at aaaa
>>> SIP/2.0 (CSeq 28092)
>>> nta.c:3174 agent_check_request_via() nta: Via check:
>>> received=82.166.84.247
>>> nta.c:3085 agent_recv_request() nta: INVITE (28092) going to a default
>>> leg
>>> nta.c:1350 set_timeout() nta: timer set to 2000 ms
>>> nua_server.c:102 nua_stack_process_request() nua:
>>> nua_stack_process_request: entering
>>> nua_stack.c:899 nh_create() nua: nh_create: entering
>>> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>> entering
>>> soa.c:280 soa_clone() soa_clone(static::0x7fe17c001930, 0x7fe17c001130,
>>> 0x7fe17c0cf210) called
>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
>>> called
>>> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fe17c04a2e0)
>>> soa.c:1302 soa_init_offer_answer()
>>> soa_init_offer_answer(static::0x7fe17c0bf4d0) called
>>> soa.c:1171 soa_set_remote_sdp()
>>> soa_set_remote_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe17c0be55f, 2039)
>>> called
>>> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fe17c0cf210): adding
>>> session usage
>>> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
>>> 82.166.84.247:53645
>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>> 0x7fe17c0d4d90 0x7fe17c0190a0 140 (140)
>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>> 0x7fe17c0d4d90 0x7fe17c0be3ab 86 (86)
>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>> 0x7fe17c0d4d90 0x7fe17c01912c 101 (101)
>>> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 394 bytes of 394
>>> to ws/82.166.84.247:53645
>>> tport.c:3492 tport_send_msg() tport_vsend returned 394
>>> send 394 bytes to ws/[82.166.84.247]:53645 at 08:21:30.469343:
>>>
>>>  ------------------------------------------------------------------------
>>>    SIP/2.0 100 Trying
>>>    Via: SIP/2.0/WS
>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>    To: <sip:991234 at aaaa>
>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>    CSeq: 28092 INVITE
>>>    User-Agent:
>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>>>    Content-Length: 0
>>>
>>>
>>>  ------------------------------------------------------------------------
>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>> timer
>>> nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (28092)
>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_invite
>>> 100 Trying
>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
>>> state changed: init -> received, received offer
>>> soa.c:1098 soa_get_remote_sdp()
>>> soa_get_remote_sdp(static::0x7fe17c0bf4d0, [0x7fe1aa7b35d8],
>>> [0x7fe1aa7b35e0], [(nil)]) called
>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 100
>>> Trying
>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>> entering
>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>> timer
>>> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_channel.c:1075 New Channel
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx
>>> [3ac3c622-f55b-11e4-a447-7d37723461ed]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>> entering
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_NEW
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8848
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx receiving invite from
>>> 82.166.84.247:53645 version: 1.5.15b git b01352c 2015-04-23 04:33:08Z
>>> 64bit
>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8960 IP 82.166.84.247
>>> Approved by acl "domains[]". Access Granted.
>>> nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
>>> nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle
>>> (nil)
>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:10113 Setting NAT mode based
>>> on via received
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [received][100]
>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6637 Remote SDP:
>>> v=0
>>> o=- 3843427479443760600 2 IN IP4 127.0.0.1
>>> s=Doubango Telecom - chrome
>>> t=0 0
>>> a=group:BUNDLE audio
>>> a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>> m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>> c=IN IP4 130.211.78.35
>>> a=rtpmap:111 opus/48000/2
>>> a=fmtp:111 minptime=10; useinbandfec=1
>>> a=rtpmap:103 ISAC/16000
>>> a=rtpmap:104 ISAC/32000
>>> a=rtpmap:9 G722/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:106 CN/32000
>>> a=rtpmap:105 CN/16000
>>> a=rtpmap:13 CN/8000
>>> a=rtpmap:126 telephone-event/8000
>>> a=rtcp:58209 IN IP4 130.211.78.35
>>> a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
>>> generation 0
>>> a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
>>> generation 0
>>> a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx
>>> raddr 10.0.0.10 rport 63888 generation 0
>>> a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx
>>> raddr 10.0.0.10 rport 63888 generation 0
>>> a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype
>>> active generation 0
>>> a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype
>>> active generation 0
>>> a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay
>>> raddr 82.166.84.247 rport 53792 generation 0
>>> a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay
>>> raddr 82.166.84.247 rport 53792 generation 0
>>> a=ice-ufrag:EDo5kr308/TXhitG
>>> a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>>> a=ice-options:google-ice
>>> a=fingerprint:sha-256
>>> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>>> a=setup:actpass
>>> a=mid:audio
>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>> a=rtcp-mux
>>> a=maxptime:60
>>> a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>>> a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>> b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>> a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>> a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>
>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6903 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_NEW -> CS_INIT
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:491 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State NEW
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_INIT
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT
>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:87
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA INIT
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:40
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard INIT
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:48 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_INIT -> CS_ROUTING
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT going to sleep
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_ROUTING
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:2204 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change DOWN -> RINGING
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING
>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:123
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA ROUTING
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:166
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard ROUTING
>>> 2015-05-08 08:21:30.467711 [INFO] mod_dialplan_xml.c:635 Processing
>>> asdasda <1000>->991234 in context public
>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
>>> [public->cdquality_conferences_with_api] continue=false
>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (FAIL)
>>> [cdquality_conferences_with_api] destination_number(991234) =~
>>> /^(75\d{4,36})$/ break=on-false
>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
>>> [public->test_conferences] continue=false
>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (PASS)
>>> [test_conferences] destination_number(991234) =~ /^(99\d{4,36})$/
>>> break=on-false
>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action answer()
>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action
>>> conference(991234-${domain_name}@test)
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:216 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_ROUTING ->
>>> CS_EXECUTE
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING going to sleep
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_EXECUTE
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:535 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State EXECUTE
>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:178
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA EXECUTE
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:258
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard EXECUTE
>>> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx answer()
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [opus:111:48000:60:0:1]/[opus:116:48000:20:0:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
>>> Sticking with opus at 48000h@20i
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
>>> Compare [opus:116:48000:20:0:1] ++++ is saved as a match
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [opus:111:48000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [opus:111:48000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [ISAC:103:16000:30:32000:1]/[opus:116:48000:20:0:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [ISAC:103:16000:30:32000:1]/[PCMU:0:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [ISAC:103:16000:30:32000:1]/[PCMA:8:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [ISAC:104:32000:30:32000:1]/[opus:116:48000:20:0:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [ISAC:104:32000:30:32000:1]/[PCMU:0:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [ISAC:104:32000:30:32000:1]/[PCMA:8:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [G722:9:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [G722:9:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [G722:9:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [PCMU:0:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [PCMU:0:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
>>> Sticking with PCMU at 8000h@20i
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
>>> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [PCMU:0:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [PCMA:8:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [PCMA:8:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [PCMA:8:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
>>> Sticking with PCMA at 8000h@20i
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
>>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [CN:105:16000:60:0:1]/[opus:116:48000:20:0:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [CN:105:16000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [CN:105:16000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [CN:13:8000:60:0:1]/[opus:116:48000:20:0:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [CN:13:8000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>> Compare [CN:13:8000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3588 Set
>>> telephone-event payload to 126
>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set
>>> bitrate to local settings [-1000bps]
>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set
>>> bitrate to local settings [-1000bps]
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2507 Set Codec
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx opus/48000 20 ms 960 samples 0 bits
>>> 1 channels
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:111
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Original read codec set to opus:116
>>> 2015-05-08 08:21:30.467711 [WARNING] switch_core_media.c:2791 NO
>>> candidate ACL defined, Defaulting to wan.auto
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>> Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>> Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>> Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose
>>> audio Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>> Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose
>>> audio Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>> Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>> Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>>> Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2953 setting
>>> remote audio ice addr to 82.166.84.247:63888 based on candidate
>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2978 setting
>>> remote rtcp audio addr to 82.166.84.247:63888 based on candidate
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3935 Set 2833
>>> dtmf send/recv payload to 126
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5171 AUDIO RTP [
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx] 172.30.0.219 port 19864 ->
>>> 82.166.84.247 port 63888 codec: 111 ms: 20
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3559 Starting timer
>>> [soft] 960 bytes per 20ms
>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5345 Activating
>>> Audio ICE
>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_rtp.c:4009 Activating RTP
>>> audio ICE: EDo5kr308/TXhitG:v9QogVGvZ8jGwYIi 82.166.84.247:63888
>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5388 Activating
>>> RTCP PORT 63888
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3909 RTCP send rate is:
>>> 10000 and packet rate is: 20000 Remote Port: 63888
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:2349 Setting RTCP remote
>>> addr to 82.166.84.247:63888
>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5396 Skipping RTCP
>>> ICE (Same as RTP)
>>> 2015-05-08 08:21:30.467711 [INFO] switch_rtp.c:3101 Activate RTP/RTCP
>>> audio DTLS client
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5469 Set 2833
>>> dtmf send payload to 126
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5475 Set 2833
>>> dtmf receive payload to 126
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5503 Set comfort
>>> noise payload to 106
>>> 2015-05-08 08:21:30.467711 [NOTICE] sofia_media.c:92 Pre-Answer
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx!
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3419 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change RINGING -> EARLY
>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:780 Local SDP
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx:
>>> v=0
>>> o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>>> s=FreeSWITCH
>>> c=IN IP4 xxx.xxx.xxx.xxx
>>> t=0 0
>>> a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>> m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>>> a=rtpmap:111 opus/48000/2
>>> a=fmtp:111 useinbandfec=1; minptime=10
>>> a=rtpmap:126 telephone-event/8000
>>> a=rtpmap:106 CN/8000
>>> a=ptime:20
>>> a=sendrecv
>>> a=fingerprint:sha-256
>>> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>>> a=rtcp-mux
>>> a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>>> a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>>> a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>>> a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>> a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>>> a=ice-ufrag:v9QogVGvZ8jGwYIi
>>> a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>>> a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
>>> generation 0
>>>
>>> nua.c:879 nua_respond() nua: nua_respond: entering
>>> nua_stack.c:529 nua_signal() nua(0x7fe17c0cf210): sent signal r_respond
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send signal
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>> nua_stack.c:573 nua_stack_signal() nua(0x7fe17c0cf210): recv signal
>>> r_respond 200 OK
>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>> entering
>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
>>> called
>>> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fe17c0bf4d0,
>>> (nil), 0x7fe190041d31, -1) called
>>> soa.c:890 soa_set_capability_sdp()
>>> soa_set_capability_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe190041d31, -1)
>>> called
>>> 2015-05-08 08:21:30.467711 [NOTICE] mod_dptools.c:1292 Channel [
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx] has been answered
>>> nua_session.c:2320 nua_invite_server_respond() nua:
>>> nua_invite_server_respond: entering
>>> soa.c:1515 soa_generate_answer()
>>> soa_generate_answer(static::0x7fe17c0bf4d0) called
>>> soa_static.c:1146 offer_answer_step()
>>> soa_static_offer_answer_action(0x7fe17c0bf4d0, soa_generate_answer): called
>>> soa_static.c:1187 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>> soa_generate_answer): generating local description
>>> soa_static.c:1228 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>> soa_generate_answer): upgrade with remote description
>>> soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fe1aa7b1a30,
>>> 0x7fe17c026ac0, ""): called
>>> soa_static.c:1444 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>> soa_generate_answer): storing local description
>>> soa.c:1730 soa_activate() soa_activate(static::0x7fe17c0bf4d0, (nil))
>>> called
>>> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe17c0bf4d0,
>>> [(nil)], [0x7fe1aa7b3b58], [0x7fe1aa7b3b54]) called
>>> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
>>> 82.166.84.247:53645
>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>> 0x7fe17c0d4d90 0x7fe17c0c5560 136 (136)
>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>> 0x7fe17c0d4d90 0x7fe17c0be3ab 63 (63)
>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>> 0x7fe17c0d4d90 0x7fe17c0c55e8 41 (41)
>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>> 0x7fe17c0d4d90 0x7fe17c0c5611 631 (631)
>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>> 0x7fe17c0d4d90 0x7fe17c0d3020 864 (864)
>>> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 1802 bytes of
>>> 1802 to ws/82.166.84.247:53645
>>> tport.c:3492 tport_send_msg() tport_vsend returned 1802
>>> send 1802 bytes to ws/[82.166.84.247]:53645 at 08:21:30.479618:
>>>
>>>  ------------------------------------------------------------------------
>>>    SIP/2.0 200 OK
>>>    Via: SIP/2.0/WS
>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>    CSeq: 28092 INVITE
>>>    Contact: <sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp>
>>>    User-Agent:
>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>>>    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: 864
>>>    Remote-Party-ID: "991234" <sip:991234 at aaaa
>>> >;party=calling;privacy=off;screen=no
>>>
>>>    v=0
>>>    o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>>>    s=FreeSWITCH
>>>    c=IN IP4 xxx.xxx.xxx.xxx
>>>    t=0 0
>>>    a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>    m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>>>    a=rtpmap:111 opus/48000/2
>>>    a=fmtp:111 useinbandfec=1; minptime=10
>>>    a=rtpmap:126 telephone-event/8000
>>>    a=rtpmap:106 CN/8000
>>>    a=ptime:20
>>>    a=fingerprint:sha-256
>>> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>>>    a=rtcp-mux
>>>    a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>>>    a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>>>    a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>>>    a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>    a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>>>    a=ice-ufrag:v9QogVGvZ8jGwYIi
>>>    a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>>>    a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
>>> generation 0
>>>
>>>  ------------------------------------------------------------------------
>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>> timer
>>> nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (28092)
>>> nta.c:1348 set_timeout() nta: timer shortened to 500 ms
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3711 (
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change EARLY -> ACTIVE
>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
>>> state changed: received -> completed, sent answer
>>> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fe17c0bf4d0,
>>> [0x7fe1aa7b3c48], [0x7fe1aa7b3c50], [(nil)]) called
>>> soa.c:616 soa_get_params() soa_get_params(static::0x7fe17c0bf4d0, ...)
>>> called
>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 200
>>> OK
>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>> entering
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [completed][200]
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx
>>> conference(991234-172.30.0.219 at test)
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10364 using channel
>>> sound prefix: /usr/local/freeswitch/sounds/en/us/callie
>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:8991 Raw Codec
>>> Activation Success L16 at 48000hz 1 channel 20ms
>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:9037 Raw Codec
>>> Activation Success L16 at 16000hz 1 channel 20ms
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:221
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Push codec L16:100
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '0' to 'mute'
>>> 2015-05-08 08:21:30.467711 [INFO] switch_ivr_async.c:212 Digit parser
>>> mod_conference: Setting realm to 'conf'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding 0/conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010b8e0
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '*' to 'deaf mute'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding */conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010b910
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '9' to 'energy up'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding 9/conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010b940
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '8' to 'energy equ'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding 8/conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010b970
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '7' to 'energy dn'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding 7/conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010b9a0
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '3' to 'vol talk up'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding 3/conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010b9d0
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '2' to 'vol talk zero'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding 2/conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010ba00
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '1' to 'vol talk dn'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding 1/conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010ba30
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '6' to 'vol listen up'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding 6/conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010ba60
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '5' to 'vol listen zero'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding 5/conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010ba90
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '4' to 'vol listen dn'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding 4/conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010bac0
>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '#' to 'hangup'
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>> mod_conference: binding #/conf/0 callback: 0x7fe1a92764e0 data:
>>> 0x7fe19010baf0
>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send signal
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:4765 Setup timer
>>> soft success interval: 20  samples: 960
>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:3043 Setup timer
>>> success interval: 30  samples: 480
>>> 2015-05-08 08:21:30.507713 [DEBUG] mod_local_stream.c:498 Opening Stream
>>> [moh/16000] 16000hz
>>> 2015-05-08 08:21:30.507713 [NOTICE] switch_core_io.c:1261 Activating
>>> write resampler
>>> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
>>> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
>>> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
>>> 0x7fe17c0c6ab0 from (ws/82.166.84.247:53645) has 550 bytes, veclen = 1
>>> recv 550 bytes from ws/[82.166.84.247]:53645 at 08:21:30.662998:
>>>
>>>  ------------------------------------------------------------------------
>>>    ACK sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0
>>>    Via: SIP/2.0/WS
>>> df7jal23ls0d.invalid;branch=z9hG4bK30nT8FwJ3EqSVIbdgFvT;rport
>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>>>    Contact: "asdasda"<
>>> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
>>> >;+g.oma.sip-im;language="en,fr"
>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>    CSeq: 28092 ACK
>>>    Content-Length: 0
>>>    Max-Forwards: 70
>>>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>>>    Organization: Doubango Telecom
>>>
>>>
>>>  ------------------------------------------------------------------------
>>> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
>>> 0x7fe17c0c6ab0 (550 bytes) from ws/82.166.84.247:53645/sip next=(nil)
>>> nta.c:2880 agent_recv_request() nta: received ACK
>>> sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0 (CSeq 28092)
>>> nta.c:3174 agent_check_request_via() nta: Via check:
>>> received=82.166.84.247
>>> nta.c:3019 agent_recv_request() nta: ACK (28092) is going to INVITE
>>> (28092)
>>> nua_session.c:2569 process_ack_or_cancel() nua: process_ack_or_cancel:
>>> entering
>>> soa.c:1214 soa_clear_remote_sdp()
>>> soa_clear_remote_sdp(static::0x7fe17c0bf4d0) called
>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_ack 200 OK
>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
>>> state changed: completed -> ready
>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state 200
>>> OK
>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_active
>>> 200 Call active
>>> nta.c:5744 incoming_free() nta: incoming_free(0x7fe17c024090)
>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>> timer
>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>> entering
>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>> entering
>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>> entering
>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> 2015-05-08 08:21:30.667707 [DEBUG] sofia.c:6627 Channel
>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [ready][200]
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>> nta.c:1289 agent_timer() nta: timer not set
>>> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
>>> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2924 Changing audio DTLS
>>> state from HANDSHAKE to SETUP
>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2832 audio Fingerprint
>>> Verified.
>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3374 Activating Audio
>>> Secure RTP SEND
>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3352 Activating Audio
>>> Secure RTP RECV
>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2872 Changing audio DTLS
>>> state from SETUP to READY
>>> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure Type:
>>> srtp:dtls:AES_CM_128_HMAC_SHA1_80
>>> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure Type:
>>> srtp:dtls:AES_CM_128_HMAC_SHA1_80
>>> 2015-05-08 08:22:02.087708 [DEBUG] switch_rtp.c:1937 rtcp_stats_init:
>>> ssrc[-195423717] base_seq[29507]
>>>
>>> Notice that the INVITE was received at 08:21:30 while DTLS was READY
>>> at 2015-05-08 08:22:02 which means that it took 32 seconds to voice. Again,
>>> if I am not using TURN/STUN, the whole process is pretty quick (2 seconds).
>>>
>>> Also, notice:
>>> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
>>> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
>>>
>>> Which means that the media is relayed via the TURN server (TURN server
>>> IP is 130.211.78.35)...
>>>
>>> Any idea?
>>>
>>> Thanks,
>>> Adam
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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From victor.medina at cibersys.com  Fri May  8 20:09:23 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Fri, 8 May 2015 11:39:23 -0430
Subject: [Freeswitch-users] WebRTC slow connection time
In-Reply-To: <CALXCt0r_CQMFG5Ldjj9K1Q5g9m_t+JN6HShQgbbeXTzN5nkUqA@mail.gmail.com>
References: <CADNqFCky4iMXF-5vukKcqmT8=Q+anVHdtn+0zueV+4yJ3GUC6g@mail.gmail.com>
	<E6A2DD46-EA8B-400A-A58D-AA9547DDA682@jerris.com>
	<CADNqFC=ba=eD8DCAicLLNx6_=xw0JRFYSdM+UDo2D35OGLR_+g@mail.gmail.com>
	<CADNqFC=EyojairQTzah9PNX0Wwog5jj4Tcudi_UvmeCTJbtCzA@mail.gmail.com>
	<CALXCt0r_CQMFG5Ldjj9K1Q5g9m_t+JN6HShQgbbeXTzN5nkUqA@mail.gmail.com>
Message-ID: <CACXsapPscnF5kZvtPesYKhUg-neiXoyrD=kP3Zk0vakzvtuy1g@mail.gmail.com>

What are the chalenges of putting the internal channel on 80(tcp) and
443(tls)? (besides the obvious things of having to deal with a server
running under a privileged ports)

2015-05-08 11:28 GMT-04:30 Giovanni Maruzzelli <gmaruzz at gmail.com>:

>
> Btw, that (accomodating clients that only do tcp 80/443) is the biggest
> and almost unique reason of Skype success...
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On May 8, 2015 5:53 PM, "Adam Ben-Ayoun" <adam.ben.ayoun1 at gmail.com>
> wrote:
>
>> Is there an easier way to deal with clients which can only do TCP on port
>> 443/80 for example?
>>
>> On 8 May 2015 at 18:40, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:
>>
>>> I am using TURN as a TCP fallback (when UDP is blocked).
>>>
>>> On 8 May 2015 at 18:34, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> If you have freeswitch on a public address, why would you ever use TURN?
>>>>
>>>> On May 8, 2015, at 4:27 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>>>> wrote:
>>>>
>>>> Hi guys,
>>>>
>>>> I am using Freeswitch from master (month old). I tried several WebRTC
>>>> clients (my own test app on Android and sipml5) and on certain WiFi's I am
>>>> getting connection time up to 35 seconds when using STUN and TURN servers.
>>>> Also, when I am using TURN and STUN, Freeswitch chooses the TURN candidate
>>>> although as far as connectivity, when I am not using STUN and TURN I am
>>>> connecting successfully after ~2 seconds (so no real need for TURN).
>>>> Attaching the SIP trace from Freeswitch:
>>>>
>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>> entering
>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
>>>> called
>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>> entering
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>> entering
>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe18c001930, ...)
>>>> called
>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>> entering
>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
>>>> called
>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>> entering
>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe184001930, ...)
>>>> called
>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>> entering
>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>> entering
>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>> entering
>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe188001930, ...)
>>>> called
>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>> entering
>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>> entering
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
>>>> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
>>>> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
>>>> 0x7fe17c089c50 from (ws/82.166.84.247:53645) has 2598 bytes, veclen = 1
>>>> recv 2598 bytes from ws/[82.166.84.247]:53645 at 08:21:30.468866:
>>>>
>>>>  ------------------------------------------------------------------------
>>>>    INVITE sip:991234 at aaaa SIP/2.0
>>>>    Via: SIP/2.0/WS
>>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport
>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>    To: <sip:991234 at aaaa>
>>>>    Contact: "asdasda"<
>>>> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
>>>> >;+g.oma.sip-im;language="en,fr"
>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>    CSeq: 28092 INVITE
>>>>    Content-Type: application/sdp
>>>>    Content-Length: 2039
>>>>    Max-Forwards: 70
>>>>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>>>>    Organization: Doubango Telecom
>>>>
>>>>    v=0
>>>>    o=- 3843427479443760600 2 IN IP4 127.0.0.1
>>>>    s=Doubango Telecom - chrome
>>>>    t=0 0
>>>>    a=group:BUNDLE audio
>>>>    a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>    m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>>>    c=IN IP4 130.211.78.35
>>>>    a=rtcp:58209 IN IP4 130.211.78.35
>>>>    a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
>>>> generation 0
>>>>    a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
>>>> generation 0
>>>>    a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ
>>>> srflx raddr 10.0.0.10 rport 63888 generation 0
>>>>    a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ
>>>> srflx raddr 10.0.0.10 rport 63888 generation 0
>>>>    a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host
>>>> tcptype active generation 0
>>>>    a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host
>>>> tcptype active generation 0
>>>>    a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay
>>>> raddr 82.166.84.247 rport 53792 generation 0
>>>>    a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay
>>>> raddr 82.166.84.247 rport 53792 generation 0
>>>>    a=ice-ufrag:EDo5kr308/TXhitG
>>>>    a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>>>>    a=ice-options:google-ice
>>>>    a=fingerprint:sha-256
>>>> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>>>>    a=setup:actpass
>>>>    a=mid:audio
>>>>    a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>>>    a=extmap:3
>>>> http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>>    a=sendrecv
>>>>    a=rtcp-mux
>>>>    a=rtpmap:111 opus/48000/2
>>>>    a=fmtp:111 minptime=10; useinbandfec=1
>>>>    a=rtpmap:103 ISAC/16000
>>>>    a=rtpmap:104 ISAC/32000
>>>>    a=rtpmap:9 G722/8000
>>>>    a=rtpmap:0 PCMU/8000
>>>>    a=rtpmap:8 PCMA/8000
>>>>    a=rtpmap:106 CN/32000
>>>>    a=rtpmap:105 CN/16000
>>>>    a=rtpmap:13 CN/8000
>>>>    a=rtpmap:126 telephone-event/8000
>>>>    a=maxptime:60
>>>>    a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>>>>    a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>> b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>>    a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>    a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>>
>>>>  ------------------------------------------------------------------------
>>>> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
>>>> 0x7fe17c089c50 (2598 bytes) from ws/82.166.84.247:53645/sip next=(nil)
>>>> nta.c:2880 agent_recv_request() nta: received INVITE sip:991234 at aaaa
>>>> SIP/2.0 (CSeq 28092)
>>>> nta.c:3174 agent_check_request_via() nta: Via check:
>>>> received=82.166.84.247
>>>> nta.c:3085 agent_recv_request() nta: INVITE (28092) going to a default
>>>> leg
>>>> nta.c:1350 set_timeout() nta: timer set to 2000 ms
>>>> nua_server.c:102 nua_stack_process_request() nua:
>>>> nua_stack_process_request: entering
>>>> nua_stack.c:899 nh_create() nua: nh_create: entering
>>>> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>> entering
>>>> soa.c:280 soa_clone() soa_clone(static::0x7fe17c001930, 0x7fe17c001130,
>>>> 0x7fe17c0cf210) called
>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
>>>> called
>>>> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fe17c04a2e0)
>>>> soa.c:1302 soa_init_offer_answer()
>>>> soa_init_offer_answer(static::0x7fe17c0bf4d0) called
>>>> soa.c:1171 soa_set_remote_sdp()
>>>> soa_set_remote_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe17c0be55f, 2039)
>>>> called
>>>> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fe17c0cf210): adding
>>>> session usage
>>>> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
>>>> 82.166.84.247:53645
>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>> 0x7fe17c0d4d90 0x7fe17c0190a0 140 (140)
>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>> 0x7fe17c0d4d90 0x7fe17c0be3ab 86 (86)
>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>> 0x7fe17c0d4d90 0x7fe17c01912c 101 (101)
>>>> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 394 bytes of
>>>> 394 to ws/82.166.84.247:53645
>>>> tport.c:3492 tport_send_msg() tport_vsend returned 394
>>>> send 394 bytes to ws/[82.166.84.247]:53645 at 08:21:30.469343:
>>>>
>>>>  ------------------------------------------------------------------------
>>>>    SIP/2.0 100 Trying
>>>>    Via: SIP/2.0/WS
>>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>    To: <sip:991234 at aaaa>
>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>    CSeq: 28092 INVITE
>>>>    User-Agent:
>>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>>>>    Content-Length: 0
>>>>
>>>>
>>>>  ------------------------------------------------------------------------
>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>> timer
>>>> nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (28092)
>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_invite
>>>> 100 Trying
>>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
>>>> state changed: init -> received, received offer
>>>> soa.c:1098 soa_get_remote_sdp()
>>>> soa_get_remote_sdp(static::0x7fe17c0bf4d0, [0x7fe1aa7b35d8],
>>>> [0x7fe1aa7b35e0], [(nil)]) called
>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state
>>>> 100 Trying
>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>> entering
>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>> timer
>>>> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_channel.c:1075 New Channel
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx
>>>> [3ac3c622-f55b-11e4-a447-7d37723461ed]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>> entering
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_NEW
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8848
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx receiving invite from
>>>> 82.166.84.247:53645 version: 1.5.15b git b01352c 2015-04-23 04:33:08Z
>>>> 64bit
>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8960 IP 82.166.84.247
>>>> Approved by acl "domains[]". Access Granted.
>>>> nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
>>>> nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle
>>>> (nil)
>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:10113 Setting NAT mode based
>>>> on via received
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [received][100]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6637 Remote SDP:
>>>> v=0
>>>> o=- 3843427479443760600 2 IN IP4 127.0.0.1
>>>> s=Doubango Telecom - chrome
>>>> t=0 0
>>>> a=group:BUNDLE audio
>>>> a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>> m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>>> c=IN IP4 130.211.78.35
>>>> a=rtpmap:111 opus/48000/2
>>>> a=fmtp:111 minptime=10; useinbandfec=1
>>>> a=rtpmap:103 ISAC/16000
>>>> a=rtpmap:104 ISAC/32000
>>>> a=rtpmap:9 G722/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:106 CN/32000
>>>> a=rtpmap:105 CN/16000
>>>> a=rtpmap:13 CN/8000
>>>> a=rtpmap:126 telephone-event/8000
>>>> a=rtcp:58209 IN IP4 130.211.78.35
>>>> a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
>>>> generation 0
>>>> a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
>>>> generation 0
>>>> a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx
>>>> raddr 10.0.0.10 rport 63888 generation 0
>>>> a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx
>>>> raddr 10.0.0.10 rport 63888 generation 0
>>>> a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype
>>>> active generation 0
>>>> a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype
>>>> active generation 0
>>>> a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay
>>>> raddr 82.166.84.247 rport 53792 generation 0
>>>> a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay
>>>> raddr 82.166.84.247 rport 53792 generation 0
>>>> a=ice-ufrag:EDo5kr308/TXhitG
>>>> a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>>>> a=ice-options:google-ice
>>>> a=fingerprint:sha-256
>>>> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>>>> a=setup:actpass
>>>> a=mid:audio
>>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>> a=rtcp-mux
>>>> a=maxptime:60
>>>> a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>>>> a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>> b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>> a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>> a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>>
>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6903 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_NEW -> CS_INIT
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:491 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State NEW
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_INIT
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT
>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:87
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA INIT
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:40
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard INIT
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:48 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_INIT -> CS_ROUTING
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT going to sleep
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_ROUTING
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:2204 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change DOWN -> RINGING
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING
>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:123
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA ROUTING
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:166
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard ROUTING
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_dialplan_xml.c:635 Processing
>>>> asdasda <1000>->991234 in context public
>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
>>>> [public->cdquality_conferences_with_api] continue=false
>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (FAIL)
>>>> [cdquality_conferences_with_api] destination_number(991234) =~
>>>> /^(75\d{4,36})$/ break=on-false
>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
>>>> [public->test_conferences] continue=false
>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (PASS)
>>>> [test_conferences] destination_number(991234) =~ /^(99\d{4,36})$/
>>>> break=on-false
>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action answer()
>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action
>>>> conference(991234-${domain_name}@test)
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:216 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_ROUTING ->
>>>> CS_EXECUTE
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING going to sleep
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_EXECUTE
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:535 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State EXECUTE
>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:178
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA EXECUTE
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:258
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard EXECUTE
>>>> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx answer()
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [opus:111:48000:60:0:1]/[opus:116:48000:20:0:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
>>>> Sticking with opus at 48000h@20i
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
>>>> Compare [opus:116:48000:20:0:1] ++++ is saved as a match
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [opus:111:48000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [opus:111:48000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [ISAC:103:16000:30:32000:1]/[opus:116:48000:20:0:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [ISAC:103:16000:30:32000:1]/[PCMU:0:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [ISAC:103:16000:30:32000:1]/[PCMA:8:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [ISAC:104:32000:30:32000:1]/[opus:116:48000:20:0:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [ISAC:104:32000:30:32000:1]/[PCMU:0:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [ISAC:104:32000:30:32000:1]/[PCMA:8:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [G722:9:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [G722:9:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [G722:9:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [PCMU:0:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [PCMU:0:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
>>>> Sticking with PCMU at 8000h@20i
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
>>>> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [PCMU:0:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [PCMA:8:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [PCMA:8:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [PCMA:8:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah HUMBUG!
>>>> Sticking with PCMA at 8000h@20i
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio Codec
>>>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [CN:105:16000:60:0:1]/[opus:116:48000:20:0:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [CN:105:16000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [CN:105:16000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [CN:13:8000:60:0:1]/[opus:116:48000:20:0:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [CN:13:8000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio Codec
>>>> Compare [CN:13:8000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3588 Set
>>>> telephone-event payload to 126
>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set
>>>> bitrate to local settings [-1000bps]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set
>>>> bitrate to local settings [-1000bps]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2507 Set Codec
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx opus/48000 20 ms 960 samples 0
>>>> bits 1 channels
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:111
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Original read codec set to opus:116
>>>> 2015-05-08 08:21:30.467711 [WARNING] switch_core_media.c:2791 NO
>>>> candidate ACL defined, Defaulting to wan.auto
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>> Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>> Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>> Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose
>>>> audio Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>> Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose
>>>> audio Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>> Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>> Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save audio
>>>> Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2953 setting
>>>> remote audio ice addr to 82.166.84.247:63888 based on candidate
>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2978 setting
>>>> remote rtcp audio addr to 82.166.84.247:63888 based on candidate
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3935 Set 2833
>>>> dtmf send/recv payload to 126
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5171 AUDIO RTP [
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx] 172.30.0.219 port 19864 ->
>>>> 82.166.84.247 port 63888 codec: 111 ms: 20
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3559 Starting timer
>>>> [soft] 960 bytes per 20ms
>>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5345 Activating
>>>> Audio ICE
>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_rtp.c:4009 Activating RTP
>>>> audio ICE: EDo5kr308/TXhitG:v9QogVGvZ8jGwYIi 82.166.84.247:63888
>>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5388 Activating
>>>> RTCP PORT 63888
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3909 RTCP send rate is:
>>>> 10000 and packet rate is: 20000 Remote Port: 63888
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:2349 Setting RTCP
>>>> remote addr to 82.166.84.247:63888
>>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5396 Skipping
>>>> RTCP ICE (Same as RTP)
>>>> 2015-05-08 08:21:30.467711 [INFO] switch_rtp.c:3101 Activate RTP/RTCP
>>>> audio DTLS client
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5469 Set 2833
>>>> dtmf send payload to 126
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5475 Set 2833
>>>> dtmf receive payload to 126
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5503 Set comfort
>>>> noise payload to 106
>>>> 2015-05-08 08:21:30.467711 [NOTICE] sofia_media.c:92 Pre-Answer
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx!
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3419 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change RINGING -> EARLY
>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:780 Local SDP
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx:
>>>> v=0
>>>> o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>>>> s=FreeSWITCH
>>>> c=IN IP4 xxx.xxx.xxx.xxx
>>>> t=0 0
>>>> a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>> m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>>>> a=rtpmap:111 opus/48000/2
>>>> a=fmtp:111 useinbandfec=1; minptime=10
>>>> a=rtpmap:126 telephone-event/8000
>>>> a=rtpmap:106 CN/8000
>>>> a=ptime:20
>>>> a=sendrecv
>>>> a=fingerprint:sha-256
>>>> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>>>> a=rtcp-mux
>>>> a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>>>> a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>>>> a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>>>> a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>> a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>>>> a=ice-ufrag:v9QogVGvZ8jGwYIi
>>>> a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>>>> a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
>>>> generation 0
>>>>
>>>> nua.c:879 nua_respond() nua: nua_respond: entering
>>>> nua_stack.c:529 nua_signal() nua(0x7fe17c0cf210): sent signal r_respond
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send
>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>> nua_stack.c:573 nua_stack_signal() nua(0x7fe17c0cf210): recv signal
>>>> r_respond 200 OK
>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>> entering
>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
>>>> called
>>>> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fe17c0bf4d0,
>>>> (nil), 0x7fe190041d31, -1) called
>>>> soa.c:890 soa_set_capability_sdp()
>>>> soa_set_capability_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe190041d31, -1)
>>>> called
>>>> 2015-05-08 08:21:30.467711 [NOTICE] mod_dptools.c:1292 Channel [
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx] has been answered
>>>> nua_session.c:2320 nua_invite_server_respond() nua:
>>>> nua_invite_server_respond: entering
>>>> soa.c:1515 soa_generate_answer()
>>>> soa_generate_answer(static::0x7fe17c0bf4d0) called
>>>> soa_static.c:1146 offer_answer_step()
>>>> soa_static_offer_answer_action(0x7fe17c0bf4d0, soa_generate_answer): called
>>>> soa_static.c:1187 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>>> soa_generate_answer): generating local description
>>>> soa_static.c:1228 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>>> soa_generate_answer): upgrade with remote description
>>>> soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fe1aa7b1a30,
>>>> 0x7fe17c026ac0, ""): called
>>>> soa_static.c:1444 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>>> soa_generate_answer): storing local description
>>>> soa.c:1730 soa_activate() soa_activate(static::0x7fe17c0bf4d0, (nil))
>>>> called
>>>> soa.c:1270 soa_get_local_sdp()
>>>> soa_get_local_sdp(static::0x7fe17c0bf4d0, [(nil)], [0x7fe1aa7b3b58],
>>>> [0x7fe1aa7b3b54]) called
>>>> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
>>>> 82.166.84.247:53645
>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>> 0x7fe17c0d4d90 0x7fe17c0c5560 136 (136)
>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>> 0x7fe17c0d4d90 0x7fe17c0be3ab 63 (63)
>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>> 0x7fe17c0d4d90 0x7fe17c0c55e8 41 (41)
>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>> 0x7fe17c0d4d90 0x7fe17c0c5611 631 (631)
>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>> 0x7fe17c0d4d90 0x7fe17c0d3020 864 (864)
>>>> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 1802 bytes of
>>>> 1802 to ws/82.166.84.247:53645
>>>> tport.c:3492 tport_send_msg() tport_vsend returned 1802
>>>> send 1802 bytes to ws/[82.166.84.247]:53645 at 08:21:30.479618:
>>>>
>>>>  ------------------------------------------------------------------------
>>>>    SIP/2.0 200 OK
>>>>    Via: SIP/2.0/WS
>>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>    CSeq: 28092 INVITE
>>>>    Contact: <sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp>
>>>>    User-Agent:
>>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>>>>    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: 864
>>>>    Remote-Party-ID: "991234" <sip:991234 at aaaa
>>>> >;party=calling;privacy=off;screen=no
>>>>
>>>>    v=0
>>>>    o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>>>>    s=FreeSWITCH
>>>>    c=IN IP4 xxx.xxx.xxx.xxx
>>>>    t=0 0
>>>>    a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>>    m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>>>>    a=rtpmap:111 opus/48000/2
>>>>    a=fmtp:111 useinbandfec=1; minptime=10
>>>>    a=rtpmap:126 telephone-event/8000
>>>>    a=rtpmap:106 CN/8000
>>>>    a=ptime:20
>>>>    a=fingerprint:sha-256
>>>> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>>>>    a=rtcp-mux
>>>>    a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>>>>    a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>>>>    a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>>>>    a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>>    a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>>>>    a=ice-ufrag:v9QogVGvZ8jGwYIi
>>>>    a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>>>>    a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
>>>> generation 0
>>>>
>>>>  ------------------------------------------------------------------------
>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>> timer
>>>> nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (28092)
>>>> nta.c:1348 set_timeout() nta: timer shortened to 500 ms
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3711 (
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change EARLY -> ACTIVE
>>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
>>>> state changed: received -> completed, sent answer
>>>> soa.c:1270 soa_get_local_sdp()
>>>> soa_get_local_sdp(static::0x7fe17c0bf4d0, [0x7fe1aa7b3c48],
>>>> [0x7fe1aa7b3c50], [(nil)]) called
>>>> soa.c:616 soa_get_params() soa_get_params(static::0x7fe17c0bf4d0, ...)
>>>> called
>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state
>>>> 200 OK
>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>> entering
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [completed][200]
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx
>>>> conference(991234-172.30.0.219 at test)
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10364 using channel
>>>> sound prefix: /usr/local/freeswitch/sounds/en/us/callie
>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:8991 Raw Codec
>>>> Activation Success L16 at 48000hz 1 channel 20ms
>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:9037 Raw Codec
>>>> Activation Success L16 at 16000hz 1 channel 20ms
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:221
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Push codec L16:100
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '0' to 'mute'
>>>> 2015-05-08 08:21:30.467711 [INFO] switch_ivr_async.c:212 Digit parser
>>>> mod_conference: Setting realm to 'conf'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding 0/conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010b8e0
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '*' to 'deaf mute'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding */conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010b910
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '9' to 'energy up'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding 9/conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010b940
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '8' to 'energy equ'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding 8/conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010b970
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '7' to 'energy dn'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding 7/conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010b9a0
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '3' to 'vol talk up'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding 3/conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010b9d0
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '2' to 'vol talk zero'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding 2/conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010ba00
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '1' to 'vol talk dn'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding 1/conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010ba30
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '6' to 'vol listen up'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding 6/conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010ba60
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '5' to 'vol listen zero'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding 5/conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010ba90
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '4' to 'vol listen dn'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding 4/conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010bac0
>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '#' to 'hangup'
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>> mod_conference: binding #/conf/0 callback: 0x7fe1a92764e0 data:
>>>> 0x7fe19010baf0
>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send
>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:4765 Setup timer
>>>> soft success interval: 20  samples: 960
>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:3043 Setup timer
>>>> success interval: 30  samples: 480
>>>> 2015-05-08 08:21:30.507713 [DEBUG] mod_local_stream.c:498 Opening
>>>> Stream [moh/16000] 16000hz
>>>> 2015-05-08 08:21:30.507713 [NOTICE] switch_core_io.c:1261 Activating
>>>> write resampler
>>>> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
>>>> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
>>>> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
>>>> 0x7fe17c0c6ab0 from (ws/82.166.84.247:53645) has 550 bytes, veclen = 1
>>>> recv 550 bytes from ws/[82.166.84.247]:53645 at 08:21:30.662998:
>>>>
>>>>  ------------------------------------------------------------------------
>>>>    ACK sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0
>>>>    Via: SIP/2.0/WS
>>>> df7jal23ls0d.invalid;branch=z9hG4bK30nT8FwJ3EqSVIbdgFvT;rport
>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>>>>    Contact: "asdasda"<
>>>> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
>>>> >;+g.oma.sip-im;language="en,fr"
>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>    CSeq: 28092 ACK
>>>>    Content-Length: 0
>>>>    Max-Forwards: 70
>>>>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>>>>    Organization: Doubango Telecom
>>>>
>>>>
>>>>  ------------------------------------------------------------------------
>>>> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
>>>> 0x7fe17c0c6ab0 (550 bytes) from ws/82.166.84.247:53645/sip next=(nil)
>>>> nta.c:2880 agent_recv_request() nta: received ACK
>>>> sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0 (CSeq 28092)
>>>> nta.c:3174 agent_check_request_via() nta: Via check:
>>>> received=82.166.84.247
>>>> nta.c:3019 agent_recv_request() nta: ACK (28092) is going to INVITE
>>>> (28092)
>>>> nua_session.c:2569 process_ack_or_cancel() nua: process_ack_or_cancel:
>>>> entering
>>>> soa.c:1214 soa_clear_remote_sdp()
>>>> soa_clear_remote_sdp(static::0x7fe17c0bf4d0) called
>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_ack 200
>>>> OK
>>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210): call
>>>> state changed: completed -> ready
>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state
>>>> 200 OK
>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_active
>>>> 200 Call active
>>>> nta.c:5744 incoming_free() nta: incoming_free(0x7fe17c024090)
>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>> timer
>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>> entering
>>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>> entering
>>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>> entering
>>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> 2015-05-08 08:21:30.667707 [DEBUG] sofia.c:6627 Channel
>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [ready][200]
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>> nta.c:1289 agent_timer() nta: timer not set
>>>> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
>>>> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2924 Changing audio DTLS
>>>> state from HANDSHAKE to SETUP
>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2832 audio Fingerprint
>>>> Verified.
>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3374 Activating Audio
>>>> Secure RTP SEND
>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3352 Activating Audio
>>>> Secure RTP RECV
>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2872 Changing audio DTLS
>>>> state from SETUP to READY
>>>> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure
>>>> Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80
>>>> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure
>>>> Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80
>>>> 2015-05-08 08:22:02.087708 [DEBUG] switch_rtp.c:1937 rtcp_stats_init:
>>>> ssrc[-195423717] base_seq[29507]
>>>>
>>>> Notice that the INVITE was received at 08:21:30 while DTLS was READY
>>>> at 2015-05-08 08:22:02 which means that it took 32 seconds to voice. Again,
>>>> if I am not using TURN/STUN, the whole process is pretty quick (2 seconds).
>>>>
>>>> Also, notice:
>>>> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
>>>> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
>>>>
>>>> Which means that the media is relayed via the TURN server (TURN server
>>>> IP is 130.211.78.35)...
>>>>
>>>> Any idea?
>>>>
>>>> Thanks,
>>>> Adam
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From victor.medina at cibersys.com  Fri May  8 20:11:28 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Fri, 8 May 2015 11:41:28 -0430
Subject: [Freeswitch-users] Testing WebRTC with latest master Git 1.5
In-Reply-To: <CALXCt0oB+hCot0sqrXhV_yFzu5Fnk8NPKXwgj49=QDmvr3r04Q@mail.gmail.com>
References: <CACXsapNC95A86iG4ojhiQ6Wc_yje0VFb40LcDCsvZn7Kye1xtw@mail.gmail.com>
	<CALXCt0p_GVoH-+4ouM-OBqCQdXVwbjOW=j=zBHJpSB_iLShz9A@mail.gmail.com>
	<CAEA-bfGcR67WpBVyoZMD3hEkA2ShfhgsEUoEAYBvriLEG=qJkg@mail.gmail.com>
	<CALXCt0pzkypo9HfX6aBgwhDOyVnWNG2nOJYC3KmiSu-veTB3BQ@mail.gmail.com>
	<CAEA-bfEZM7jQ-1fX-mXdJNxiy72F45O9WctEjUmcUtO-rd7ZGg@mail.gmail.com>
	<CALXCt0oB+hCot0sqrXhV_yFzu5Fnk8NPKXwgj49=QDmvr3r04Q@mail.gmail.com>
Message-ID: <CACXsapMMkzL4TfyZmVC6tgZrHddtyu6ctn8vWeebTJiUCe+nBg@mail.gmail.com>

Ill give it try. Installing a VM right now.

I would really like to ask something....

Putting Apache in front of FS for web serving... help us resolving some nat
problems?

Should the fws http binding port be without tls and let apache manage all
tls related traffic?

2015-05-07 18:19 GMT-04:30 Giovanni Maruzzelli <gmaruzz at gmail.com>:

> Go with the one I told you. It has webrrc too, and the docs are completely
> updated, and we all are using and testing that one.
> Eg: is the path of less resistance ;)
> Then you can use only what part you want (webrtc), but at least you sure
> if you follow the steps it will work without fiddling.
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On May 8, 2015 12:01 AM, "Victor Medina" <vittico at gmail.com> wrote:
>
>> OK, very impressive....
>>
>> But... I, for the moment ONLY want to have a _basic_ webrtc softhone
>> connected to FS. As I undestand this, is for Video Conferencing right?
>>
>> Is this https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>> already deprecated? Not working anymore?
>>
>>
>> Sin mas a que hacer referencia,
>>
>> Victor Medina
>>
>> On Thu, May 7, 2015 at 4:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> No.
>>> This one:
>>> https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/7144556
>>>
>>> sent from my mobile,
>>> Giovanni Maruzzelli
>>> cell: +39 347 266 56 18
>>> On May 7, 2015 11:18 PM, "Victor Medina" <vittico at gmail.com> wrote:
>>>
>>>> Hi Giovanni.
>>>>
>>>> Thanks for the replay.
>>>>
>>>> You mean this confluence page?
>>>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
>>>>
>>>>
>>>> Sin mas a que hacer referencia,
>>>>
>>>> Victor Medina
>>>>
>>>> On Thu, May 7, 2015 at 4:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>>> wrote:
>>>>
>>>>> Start by scratch, from a fresh debian Jessie install.
>>>>>
>>>>> Then follow exactly, without changes, step by step, what is in the
>>>>> "freeswitch 1.6" confluence page.
>>>>>
>>>>> When you have it all working as in that page (contains instruction on
>>>>> testing) only then you can go further.
>>>>>
>>>>> Happy testing,
>>>>>
>>>>> sent from my mobile,
>>>>> Giovanni Maruzzelli
>>>>> cell: +39 347 266 56 18
>>>>> On May 7, 2015 10:58 PM, "Victor Medina" <victor.medina at cibersys.com>
>>>>> wrote:
>>>>>
>>>>>> Hi guys!
>>>>>>
>>>>>> I?ve been tying to test Freeswitch 1.5 (latest git) for WebRTC, but
>>>>>> so far I have been very unsuccessful.
>>>>>>
>>>>>> I have a server, connected directly to internet, NO NAT on server
>>>>>> side.
>>>>>>
>>>>>>
>>>>>> FS ----> INTERNET <--- NAT ---- CLIENTS
>>>>>>
>>>>>> My vars.conf includes this:
>>>>>>
>>>>>>
>>>>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxxx.xxx.xxx.xxx"/>
>>>>>>
>>>>>>   <!-- Internal SIP Profile -->
>>>>>>   <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/>
>>>>>>   <X-PRE-PROCESS cmd="set"
>>>>>> data="internal_ssl_dir=/opt/CloudVoice-vPBX/fs-20150506/certs/"/>
>>>>>>
>>>>>>   <!-- External SIP Profile -->
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>>>>
>>>>>>
>>>>>> On my external profile I have this relevants lines...
>>>>>>
>>>>>>  <param name="aggressive-nat-detection" value="true"/>
>>>>>>  <param name="apply-inbound-acl" value="domains"/>
>>>>>>  <param name="local-network-acl" value="localnet.auto"/>
>>>>>>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>>>>>>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>>>>>>  <param name="ws-binding"  value=":5066"/>
>>>>>>  <param name="wss-binding" value=":7443"/>
>>>>>>
>>>>>>
>>>>>> When doing some testing....
>>>>>>
>>>>>> Calling to echo test
>>>>>>
>>>>>>  2015-05-07 20:51:14.124955 [NOTICE] switch_channel.c:1075 New
>>>>>> Channel sofia/internal/1007 at webrtc.cibersys.com
>>>>>> [39b777d1-6e61-49ec-bfd0-6f7ca0bd0caf]
>>>>>> 2015-05-07 20:51:14.384990 [INFO] mod_dialplan_xml.c:635 Processing
>>>>>> test <1007>->9196 in context default
>>>>>> 2015-05-07 20:51:14.384990 [WARNING] switch_core_media.c:2791 NO
>>>>>> candidate ACL defined, Defaulting to wan.auto
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose
>>>>>> audio Candidate cid: 1 proto: UDP type: srflx addr:
>>>>>> 201.210.31.83:57630
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose
>>>>>> audio Candidate cid: 2 proto: UDP type: srflx addr:
>>>>>> 201.210.31.83:57632
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2953 setting
>>>>>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2978 setting
>>>>>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5345 Activating
>>>>>> Audio ICE
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTP
>>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5388 Activating
>>>>>> RTCP PORT 57632
>>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5398 Activating
>>>>>> RTCP ICE
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTCP
>>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>>>>>> 2015-05-07 20:51:14.384990 [INFO] switch_rtp.c:3103 Activate RTP/RTCP
>>>>>> audio DTLS client
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] sofia_media.c:92 Pre-Answer
>>>>>> sofia/internal/1007 at webrtc.cibersys.com!
>>>>>> 2015-05-07 20:51:14.384990 [NOTICE] mod_dptools.c:1292 Channel
>>>>>> [sofia/internal/1007 at webrtc.cibersys.com] has been answered
>>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2924 Changing audio
>>>>>> DTLS state from HANDSHAKE to SETUP
>>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2832 audio Fingerprint
>>>>>> Verified.
>>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3384 Activating Audio
>>>>>> Secure RTP SEND
>>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3362 Activating Audio
>>>>>> Secure RTP RECV
>>>>>> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2872 Changing audio
>>>>>> DTLS state from SETUP to READY
>>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>>>>>> audio RTCP PORT 57632
>>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
>>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
>>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
>>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
>>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose
>>>>>> audio Candidate cid: 1 proto: UDP type: srflx addr:
>>>>>> 201.210.31.83:57630
>>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose
>>>>>> audio Candidate cid: 2 proto: UDP type: srflx addr:
>>>>>> 201.210.31.83:57632
>>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2953 setting
>>>>>> remote audio ice addr to 201.210.31.83:57630 based on candidate
>>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2978 setting
>>>>>> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
>>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:2995
>>>>>> RE-Activating audio ICE
>>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTP
>>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
>>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
>>>>>> audio RTCP PORT 57632
>>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3050 Activating
>>>>>> audio RTCP ICE
>>>>>> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTCP
>>>>>> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
>>>>>> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:5030 RE-SETTING
>>>>>> video DTLS
>>>>>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:878 sofia/internal/
>>>>>> 1007 at webrtc.cibersys.com got stun binding response 487 Role Conflict
>>>>>> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:889 Changing role
>>>>>> to CONTROLLED
>>>>>>
>>>>>> No audio
>>>>>>
>>>>>> When calling another ext... no audio en the webrtc side.
>>>>>>
>>>>>> Can somebody help me by pointing out the right direction?
>>>>>>
>>>>>> Ive been using FF and Chrome with sipML5
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>>
>>>>>>
>>>>>> V?ctor E. Medina M.
>>>>>> Platform Architect / Chief Infrastructure
>>>>>> +58424 291 4561
>>>>>> BB #79A8AFA2
>>>>>> @VMCibersys
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From adam.ben.ayoun1 at gmail.com  Fri May  8 20:14:45 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Fri, 8 May 2015 19:14:45 +0300
Subject: [Freeswitch-users] WebRTC slow connection time
In-Reply-To: <CACXsapPscnF5kZvtPesYKhUg-neiXoyrD=kP3Zk0vakzvtuy1g@mail.gmail.com>
References: <CADNqFCky4iMXF-5vukKcqmT8=Q+anVHdtn+0zueV+4yJ3GUC6g@mail.gmail.com>
	<E6A2DD46-EA8B-400A-A58D-AA9547DDA682@jerris.com>
	<CADNqFC=ba=eD8DCAicLLNx6_=xw0JRFYSdM+UDo2D35OGLR_+g@mail.gmail.com>
	<CADNqFC=EyojairQTzah9PNX0Wwog5jj4Tcudi_UvmeCTJbtCzA@mail.gmail.com>
	<CALXCt0r_CQMFG5Ldjj9K1Q5g9m_t+JN6HShQgbbeXTzN5nkUqA@mail.gmail.com>
	<CACXsapPscnF5kZvtPesYKhUg-neiXoyrD=kP3Zk0vakzvtuy1g@mail.gmail.com>
Message-ID: <CADNqFCm+BW9fY2_r-pEUjndR683VZDNu577DiLGY9e_72J7o_g@mail.gmail.com>

I am probably missing something, we do use 80/443 for TCP/TLS for
signalling. My thinking is that media will use UDP which can be restricted.

On 8 May 2015 at 19:09, Victor Medina <victor.medina at cibersys.com> wrote:

> What are the chalenges of putting the internal channel on 80(tcp) and
> 443(tls)? (besides the obvious things of having to deal with a server
> running under a privileged ports)
>
> 2015-05-08 11:28 GMT-04:30 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>
>
>> Btw, that (accomodating clients that only do tcp 80/443) is the biggest
>> and almost unique reason of Skype success...
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On May 8, 2015 5:53 PM, "Adam Ben-Ayoun" <adam.ben.ayoun1 at gmail.com>
>> wrote:
>>
>>> Is there an easier way to deal with clients which can only do TCP on
>>> port 443/80 for example?
>>>
>>> On 8 May 2015 at 18:40, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>>> wrote:
>>>
>>>> I am using TURN as a TCP fallback (when UDP is blocked).
>>>>
>>>> On 8 May 2015 at 18:34, Michael Jerris <mike at jerris.com> wrote:
>>>>
>>>>> If you have freeswitch on a public address, why would you ever use
>>>>> TURN?
>>>>>
>>>>> On May 8, 2015, at 4:27 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>>>>> wrote:
>>>>>
>>>>> Hi guys,
>>>>>
>>>>> I am using Freeswitch from master (month old). I tried several WebRTC
>>>>> clients (my own test app on Android and sipml5) and on certain WiFi's I am
>>>>> getting connection time up to 35 seconds when using STUN and TURN servers.
>>>>> Also, when I am using TURN and STUN, Freeswitch chooses the TURN candidate
>>>>> although as far as connectivity, when I am not using STUN and TURN I am
>>>>> connecting successfully after ~2 seconds (so no real need for TURN).
>>>>> Attaching the SIP trace from Freeswitch:
>>>>>
>>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>> entering
>>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
>>>>> called
>>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>> entering
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>> entering
>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe18c001930, ...)
>>>>> called
>>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>> entering
>>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930, ...)
>>>>> called
>>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>> entering
>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe184001930, ...)
>>>>> called
>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>> entering
>>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>> entering
>>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal r_set_params
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>> entering
>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe188001930, ...)
>>>>> called
>>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200 OK
>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>> entering
>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>> entering
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
>>>>> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
>>>>> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
>>>>> 0x7fe17c089c50 from (ws/82.166.84.247:53645) has 2598 bytes, veclen =
>>>>> 1
>>>>> recv 2598 bytes from ws/[82.166.84.247]:53645 at 08:21:30.468866:
>>>>>
>>>>>  ------------------------------------------------------------------------
>>>>>    INVITE sip:991234 at aaaa SIP/2.0
>>>>>    Via: SIP/2.0/WS
>>>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport
>>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>>    To: <sip:991234 at aaaa>
>>>>>    Contact: "asdasda"<
>>>>> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
>>>>> >;+g.oma.sip-im;language="en,fr"
>>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>>    CSeq: 28092 INVITE
>>>>>    Content-Type: application/sdp
>>>>>    Content-Length: 2039
>>>>>    Max-Forwards: 70
>>>>>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>>>>>    Organization: Doubango Telecom
>>>>>
>>>>>    v=0
>>>>>    o=- 3843427479443760600 2 IN IP4 127.0.0.1
>>>>>    s=Doubango Telecom - chrome
>>>>>    t=0 0
>>>>>    a=group:BUNDLE audio
>>>>>    a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>>    m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>>>>    c=IN IP4 130.211.78.35
>>>>>    a=rtcp:58209 IN IP4 130.211.78.35
>>>>>    a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
>>>>> generation 0
>>>>>    a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
>>>>> generation 0
>>>>>    a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ
>>>>> srflx raddr 10.0.0.10 rport 63888 generation 0
>>>>>    a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ
>>>>> srflx raddr 10.0.0.10 rport 63888 generation 0
>>>>>    a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host
>>>>> tcptype active generation 0
>>>>>    a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host
>>>>> tcptype active generation 0
>>>>>    a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ
>>>>> relay raddr 82.166.84.247 rport 53792 generation 0
>>>>>    a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ
>>>>> relay raddr 82.166.84.247 rport 53792 generation 0
>>>>>    a=ice-ufrag:EDo5kr308/TXhitG
>>>>>    a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>>>>>    a=ice-options:google-ice
>>>>>    a=fingerprint:sha-256
>>>>> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>>>>>    a=setup:actpass
>>>>>    a=mid:audio
>>>>>    a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>>>>    a=extmap:3
>>>>> http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>>>    a=sendrecv
>>>>>    a=rtcp-mux
>>>>>    a=rtpmap:111 opus/48000/2
>>>>>    a=fmtp:111 minptime=10; useinbandfec=1
>>>>>    a=rtpmap:103 ISAC/16000
>>>>>    a=rtpmap:104 ISAC/32000
>>>>>    a=rtpmap:9 G722/8000
>>>>>    a=rtpmap:0 PCMU/8000
>>>>>    a=rtpmap:8 PCMA/8000
>>>>>    a=rtpmap:106 CN/32000
>>>>>    a=rtpmap:105 CN/16000
>>>>>    a=rtpmap:13 CN/8000
>>>>>    a=rtpmap:126 telephone-event/8000
>>>>>    a=maxptime:60
>>>>>    a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>>>>>    a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>> b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>>>    a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>>    a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>>>
>>>>>  ------------------------------------------------------------------------
>>>>> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
>>>>> 0x7fe17c089c50 (2598 bytes) from ws/82.166.84.247:53645/sip next=(nil)
>>>>> nta.c:2880 agent_recv_request() nta: received INVITE sip:991234 at aaaa
>>>>> SIP/2.0 (CSeq 28092)
>>>>> nta.c:3174 agent_check_request_via() nta: Via check:
>>>>> received=82.166.84.247
>>>>> nta.c:3085 agent_recv_request() nta: INVITE (28092) going to a default
>>>>> leg
>>>>> nta.c:1350 set_timeout() nta: timer set to 2000 ms
>>>>> nua_server.c:102 nua_stack_process_request() nua:
>>>>> nua_stack_process_request: entering
>>>>> nua_stack.c:899 nh_create() nua: nh_create: entering
>>>>> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>> entering
>>>>> soa.c:280 soa_clone() soa_clone(static::0x7fe17c001930,
>>>>> 0x7fe17c001130, 0x7fe17c0cf210) called
>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
>>>>> called
>>>>> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fe17c04a2e0)
>>>>> soa.c:1302 soa_init_offer_answer()
>>>>> soa_init_offer_answer(static::0x7fe17c0bf4d0) called
>>>>> soa.c:1171 soa_set_remote_sdp()
>>>>> soa_set_remote_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe17c0be55f, 2039)
>>>>> called
>>>>> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fe17c0cf210): adding
>>>>> session usage
>>>>> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
>>>>> 82.166.84.247:53645
>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>> 0x7fe17c0d4d90 0x7fe17c0190a0 140 (140)
>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>> 0x7fe17c0d4d90 0x7fe17c0be3ab 86 (86)
>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>> 0x7fe17c0d4d90 0x7fe17c01912c 101 (101)
>>>>> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 394 bytes of
>>>>> 394 to ws/82.166.84.247:53645
>>>>> tport.c:3492 tport_send_msg() tport_vsend returned 394
>>>>> send 394 bytes to ws/[82.166.84.247]:53645 at 08:21:30.469343:
>>>>>
>>>>>  ------------------------------------------------------------------------
>>>>>    SIP/2.0 100 Trying
>>>>>    Via: SIP/2.0/WS
>>>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>>    To: <sip:991234 at aaaa>
>>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>>    CSeq: 28092 INVITE
>>>>>    User-Agent:
>>>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>>>>>    Content-Length: 0
>>>>>
>>>>>
>>>>>  ------------------------------------------------------------------------
>>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>>> timer
>>>>> nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (28092)
>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_invite
>>>>> 100 Trying
>>>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210):
>>>>> call state changed: init -> received, received offer
>>>>> soa.c:1098 soa_get_remote_sdp()
>>>>> soa_get_remote_sdp(static::0x7fe17c0bf4d0, [0x7fe1aa7b35d8],
>>>>> [0x7fe1aa7b35e0], [(nil)]) called
>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state
>>>>> 100 Trying
>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>> entering
>>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>>> timer
>>>>> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_channel.c:1075 New Channel
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx
>>>>> [3ac3c622-f55b-11e4-a447-7d37723461ed]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>> entering
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_NEW
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8848
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx receiving invite from
>>>>> 82.166.84.247:53645 version: 1.5.15b git b01352c 2015-04-23 04:33:08Z
>>>>> 64bit
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8960 IP 82.166.84.247
>>>>> Approved by acl "domains[]". Access Granted.
>>>>> nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
>>>>> nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle
>>>>> (nil)
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:10113 Setting NAT mode
>>>>> based on via received
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [received][100]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6637 Remote SDP:
>>>>> v=0
>>>>> o=- 3843427479443760600 2 IN IP4 127.0.0.1
>>>>> s=Doubango Telecom - chrome
>>>>> t=0 0
>>>>> a=group:BUNDLE audio
>>>>> a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>> m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>>>> c=IN IP4 130.211.78.35
>>>>> a=rtpmap:111 opus/48000/2
>>>>> a=fmtp:111 minptime=10; useinbandfec=1
>>>>> a=rtpmap:103 ISAC/16000
>>>>> a=rtpmap:104 ISAC/32000
>>>>> a=rtpmap:9 G722/8000
>>>>> a=rtpmap:0 PCMU/8000
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:106 CN/32000
>>>>> a=rtpmap:105 CN/16000
>>>>> a=rtpmap:13 CN/8000
>>>>> a=rtpmap:126 telephone-event/8000
>>>>> a=rtcp:58209 IN IP4 130.211.78.35
>>>>> a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
>>>>> generation 0
>>>>> a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
>>>>> generation 0
>>>>> a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx
>>>>> raddr 10.0.0.10 rport 63888 generation 0
>>>>> a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx
>>>>> raddr 10.0.0.10 rport 63888 generation 0
>>>>> a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype
>>>>> active generation 0
>>>>> a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype
>>>>> active generation 0
>>>>> a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay
>>>>> raddr 82.166.84.247 rport 53792 generation 0
>>>>> a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay
>>>>> raddr 82.166.84.247 rport 53792 generation 0
>>>>> a=ice-ufrag:EDo5kr308/TXhitG
>>>>> a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>>>>> a=ice-options:google-ice
>>>>> a=fingerprint:sha-256
>>>>> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>>>>> a=setup:actpass
>>>>> a=mid:audio
>>>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>>> a=rtcp-mux
>>>>> a=maxptime:60
>>>>> a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>>>>> a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>> b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>>> a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>> a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>>>
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6903 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_NEW -> CS_INIT
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:491 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State NEW
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_INIT
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:87
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA INIT
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:40
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard INIT
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:48 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_INIT ->
>>>>> CS_ROUTING
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT going to sleep
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_ROUTING
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:2204 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change DOWN -> RINGING
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:123
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA ROUTING
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:166
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard ROUTING
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_dialplan_xml.c:635 Processing
>>>>> asdasda <1000>->991234 in context public
>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
>>>>> [public->cdquality_conferences_with_api] continue=false
>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (FAIL)
>>>>> [cdquality_conferences_with_api] destination_number(991234) =~
>>>>> /^(75\d{4,36})$/ break=on-false
>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
>>>>> [public->test_conferences] continue=false
>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (PASS)
>>>>> [test_conferences] destination_number(991234) =~ /^(99\d{4,36})$/
>>>>> break=on-false
>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action answer()
>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action
>>>>> conference(991234-${domain_name}@test)
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:216 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_ROUTING ->
>>>>> CS_EXECUTE
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING going to sleep
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_EXECUTE
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:535 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State EXECUTE
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:178
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA EXECUTE
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:258
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard EXECUTE
>>>>> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx answer()
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [opus:111:48000:60:0:1]/[opus:116:48000:20:0:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah
>>>>> HUMBUG! Sticking with opus at 48000h@20i
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio
>>>>> Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [opus:111:48000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [opus:111:48000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [ISAC:103:16000:30:32000:1]/[opus:116:48000:20:0:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [ISAC:103:16000:30:32000:1]/[PCMU:0:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [ISAC:103:16000:30:32000:1]/[PCMA:8:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [ISAC:104:32000:30:32000:1]/[opus:116:48000:20:0:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [ISAC:104:32000:30:32000:1]/[PCMU:0:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [ISAC:104:32000:30:32000:1]/[PCMA:8:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [G722:9:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [G722:9:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [G722:9:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [PCMU:0:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [PCMU:0:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah
>>>>> HUMBUG! Sticking with PCMU at 8000h@20i
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio
>>>>> Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [PCMU:0:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [PCMA:8:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [PCMA:8:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [PCMA:8:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah
>>>>> HUMBUG! Sticking with PCMA at 8000h@20i
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio
>>>>> Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [CN:105:16000:60:0:1]/[opus:116:48000:20:0:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [CN:105:16000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [CN:105:16000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [CN:13:8000:60:0:1]/[opus:116:48000:20:0:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [CN:13:8000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>> Codec Compare [CN:13:8000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3588 Set
>>>>> telephone-event payload to 126
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set
>>>>> bitrate to local settings [-1000bps]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set
>>>>> bitrate to local settings [-1000bps]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2507 Set Codec
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx opus/48000 20 ms 960 samples 0
>>>>> bits 1 channels
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:111
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Original read codec set to
>>>>> opus:116
>>>>> 2015-05-08 08:21:30.467711 [WARNING] switch_core_media.c:2791 NO
>>>>> candidate ACL defined, Defaulting to wan.auto
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>> Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>> Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>> Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose
>>>>> audio Candidate cid: 1 proto: udp type: srflx addr:
>>>>> 82.166.84.247:63888
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>> Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose
>>>>> audio Candidate cid: 2 proto: udp type: srflx addr:
>>>>> 82.166.84.247:63888
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>> Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 1 proto: udp type: relay addr:
>>>>> 130.211.78.35:58209
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>> Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save
>>>>> audio Candidate cid: 2 proto: udp type: relay addr:
>>>>> 130.211.78.35:58209
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2953 setting
>>>>> remote audio ice addr to 82.166.84.247:63888 based on candidate
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2978 setting
>>>>> remote rtcp audio addr to 82.166.84.247:63888 based on candidate
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3935 Set 2833
>>>>> dtmf send/recv payload to 126
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5171 AUDIO RTP [
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx] 172.30.0.219 port 19864 ->
>>>>> 82.166.84.247 port 63888 codec: 111 ms: 20
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3559 Starting timer
>>>>> [soft] 960 bytes per 20ms
>>>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5345 Activating
>>>>> Audio ICE
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_rtp.c:4009 Activating RTP
>>>>> audio ICE: EDo5kr308/TXhitG:v9QogVGvZ8jGwYIi 82.166.84.247:63888
>>>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5388 Activating
>>>>> RTCP PORT 63888
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3909 RTCP send rate
>>>>> is: 10000 and packet rate is: 20000 Remote Port: 63888
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:2349 Setting RTCP
>>>>> remote addr to 82.166.84.247:63888
>>>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5396 Skipping
>>>>> RTCP ICE (Same as RTP)
>>>>> 2015-05-08 08:21:30.467711 [INFO] switch_rtp.c:3101 Activate RTP/RTCP
>>>>> audio DTLS client
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5469 Set 2833
>>>>> dtmf send payload to 126
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5475 Set 2833
>>>>> dtmf receive payload to 126
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5503 Set
>>>>> comfort noise payload to 106
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] sofia_media.c:92 Pre-Answer
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx!
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3419 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change RINGING -> EARLY
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:780 Local SDP
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx:
>>>>> v=0
>>>>> o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 xxx.xxx.xxx.xxx
>>>>> t=0 0
>>>>> a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>>> m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>>>>> a=rtpmap:111 opus/48000/2
>>>>> a=fmtp:111 useinbandfec=1; minptime=10
>>>>> a=rtpmap:126 telephone-event/8000
>>>>> a=rtpmap:106 CN/8000
>>>>> a=ptime:20
>>>>> a=sendrecv
>>>>> a=fingerprint:sha-256
>>>>> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>>>>> a=rtcp-mux
>>>>> a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>>>>> a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>>>>> a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>>>>> a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>>> a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>>>>> a=ice-ufrag:v9QogVGvZ8jGwYIi
>>>>> a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>>>>> a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
>>>>> generation 0
>>>>>
>>>>> nua.c:879 nua_respond() nua: nua_respond: entering
>>>>> nua_stack.c:529 nua_signal() nua(0x7fe17c0cf210): sent signal r_respond
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send
>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>> nua_stack.c:573 nua_stack_signal() nua(0x7fe17c0cf210): recv signal
>>>>> r_respond 200 OK
>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>> entering
>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0, ...)
>>>>> called
>>>>> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fe17c0bf4d0,
>>>>> (nil), 0x7fe190041d31, -1) called
>>>>> soa.c:890 soa_set_capability_sdp()
>>>>> soa_set_capability_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe190041d31, -1)
>>>>> called
>>>>> 2015-05-08 08:21:30.467711 [NOTICE] mod_dptools.c:1292 Channel [
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx] has been answered
>>>>> nua_session.c:2320 nua_invite_server_respond() nua:
>>>>> nua_invite_server_respond: entering
>>>>> soa.c:1515 soa_generate_answer()
>>>>> soa_generate_answer(static::0x7fe17c0bf4d0) called
>>>>> soa_static.c:1146 offer_answer_step()
>>>>> soa_static_offer_answer_action(0x7fe17c0bf4d0, soa_generate_answer): called
>>>>> soa_static.c:1187 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>>>> soa_generate_answer): generating local description
>>>>> soa_static.c:1228 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>>>> soa_generate_answer): upgrade with remote description
>>>>> soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fe1aa7b1a30,
>>>>> 0x7fe17c026ac0, ""): called
>>>>> soa_static.c:1444 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>>>> soa_generate_answer): storing local description
>>>>> soa.c:1730 soa_activate() soa_activate(static::0x7fe17c0bf4d0, (nil))
>>>>> called
>>>>> soa.c:1270 soa_get_local_sdp()
>>>>> soa_get_local_sdp(static::0x7fe17c0bf4d0, [(nil)], [0x7fe1aa7b3b58],
>>>>> [0x7fe1aa7b3b54]) called
>>>>> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
>>>>> 82.166.84.247:53645
>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>> 0x7fe17c0d4d90 0x7fe17c0c5560 136 (136)
>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>> 0x7fe17c0d4d90 0x7fe17c0be3ab 63 (63)
>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>> 0x7fe17c0d4d90 0x7fe17c0c55e8 41 (41)
>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>> 0x7fe17c0d4d90 0x7fe17c0c5611 631 (631)
>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>> 0x7fe17c0d4d90 0x7fe17c0d3020 864 (864)
>>>>> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 1802 bytes of
>>>>> 1802 to ws/82.166.84.247:53645
>>>>> tport.c:3492 tport_send_msg() tport_vsend returned 1802
>>>>> send 1802 bytes to ws/[82.166.84.247]:53645 at 08:21:30.479618:
>>>>>
>>>>>  ------------------------------------------------------------------------
>>>>>    SIP/2.0 200 OK
>>>>>    Via: SIP/2.0/WS
>>>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>>    CSeq: 28092 INVITE
>>>>>    Contact: <sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp>
>>>>>    User-Agent:
>>>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>>>>>    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: 864
>>>>>    Remote-Party-ID: "991234" <sip:991234 at aaaa
>>>>> >;party=calling;privacy=off;screen=no
>>>>>
>>>>>    v=0
>>>>>    o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>>>>>    s=FreeSWITCH
>>>>>    c=IN IP4 xxx.xxx.xxx.xxx
>>>>>    t=0 0
>>>>>    a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>>>    m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>>>>>    a=rtpmap:111 opus/48000/2
>>>>>    a=fmtp:111 useinbandfec=1; minptime=10
>>>>>    a=rtpmap:126 telephone-event/8000
>>>>>    a=rtpmap:106 CN/8000
>>>>>    a=ptime:20
>>>>>    a=fingerprint:sha-256
>>>>> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>>>>>    a=rtcp-mux
>>>>>    a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>>>>>    a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>>>>>    a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>>>>>    a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>>>    a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>>>>>    a=ice-ufrag:v9QogVGvZ8jGwYIi
>>>>>    a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>>>>>    a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
>>>>> generation 0
>>>>>
>>>>>  ------------------------------------------------------------------------
>>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>>> timer
>>>>> nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (28092)
>>>>> nta.c:1348 set_timeout() nta: timer shortened to 500 ms
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3711 (
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change EARLY -> ACTIVE
>>>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210):
>>>>> call state changed: received -> completed, sent answer
>>>>> soa.c:1270 soa_get_local_sdp()
>>>>> soa_get_local_sdp(static::0x7fe17c0bf4d0, [0x7fe1aa7b3c48],
>>>>> [0x7fe1aa7b3c50], [(nil)]) called
>>>>> soa.c:616 soa_get_params() soa_get_params(static::0x7fe17c0bf4d0, ...)
>>>>> called
>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state
>>>>> 200 OK
>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>> entering
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [completed][200]
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx
>>>>> conference(991234-172.30.0.219 at test)
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10364 using channel
>>>>> sound prefix: /usr/local/freeswitch/sounds/en/us/callie
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:8991 Raw Codec
>>>>> Activation Success L16 at 48000hz 1 channel 20ms
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:9037 Raw Codec
>>>>> Activation Success L16 at 16000hz 1 channel 20ms
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:221
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Push codec L16:100
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '0' to 'mute'
>>>>> 2015-05-08 08:21:30.467711 [INFO] switch_ivr_async.c:212 Digit parser
>>>>> mod_conference: Setting realm to 'conf'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding 0/conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010b8e0
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '*' to 'deaf mute'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding */conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010b910
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '9' to 'energy up'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding 9/conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010b940
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '8' to 'energy equ'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding 8/conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010b970
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '7' to 'energy dn'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding 7/conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010b9a0
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '3' to 'vol talk up'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding 3/conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010b9d0
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '2' to 'vol talk zero'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding 2/conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010ba00
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '1' to 'vol talk dn'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding 1/conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010ba30
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '6' to 'vol listen up'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding 6/conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010ba60
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '5' to 'vol listen zero'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding 5/conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010ba90
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '4' to 'vol listen dn'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding 4/conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010bac0
>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '#' to 'hangup'
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit parser
>>>>> mod_conference: binding #/conf/0 callback: 0x7fe1a92764e0 data:
>>>>> 0x7fe19010baf0
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send
>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:4765 Setup timer
>>>>> soft success interval: 20  samples: 960
>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:3043 Setup timer
>>>>> success interval: 30  samples: 480
>>>>> 2015-05-08 08:21:30.507713 [DEBUG] mod_local_stream.c:498 Opening
>>>>> Stream [moh/16000] 16000hz
>>>>> 2015-05-08 08:21:30.507713 [NOTICE] switch_core_io.c:1261 Activating
>>>>> write resampler
>>>>> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
>>>>> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
>>>>> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
>>>>> 0x7fe17c0c6ab0 from (ws/82.166.84.247:53645) has 550 bytes, veclen = 1
>>>>> recv 550 bytes from ws/[82.166.84.247]:53645 at 08:21:30.662998:
>>>>>
>>>>>  ------------------------------------------------------------------------
>>>>>    ACK sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0
>>>>>    Via: SIP/2.0/WS
>>>>> df7jal23ls0d.invalid;branch=z9hG4bK30nT8FwJ3EqSVIbdgFvT;rport
>>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>>>>>    Contact: "asdasda"<
>>>>> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
>>>>> >;+g.oma.sip-im;language="en,fr"
>>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>>    CSeq: 28092 ACK
>>>>>    Content-Length: 0
>>>>>    Max-Forwards: 70
>>>>>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>>>>>    Organization: Doubango Telecom
>>>>>
>>>>>
>>>>>  ------------------------------------------------------------------------
>>>>> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
>>>>> 0x7fe17c0c6ab0 (550 bytes) from ws/82.166.84.247:53645/sip next=(nil)
>>>>> nta.c:2880 agent_recv_request() nta: received ACK
>>>>> sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0 (CSeq 28092)
>>>>> nta.c:3174 agent_check_request_via() nta: Via check:
>>>>> received=82.166.84.247
>>>>> nta.c:3019 agent_recv_request() nta: ACK (28092) is going to INVITE
>>>>> (28092)
>>>>> nua_session.c:2569 process_ack_or_cancel() nua: process_ack_or_cancel:
>>>>> entering
>>>>> soa.c:1214 soa_clear_remote_sdp()
>>>>> soa_clear_remote_sdp(static::0x7fe17c0bf4d0) called
>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_ack 200
>>>>> OK
>>>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210):
>>>>> call state changed: completed -> ready
>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state
>>>>> 200 OK
>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_active
>>>>> 200 Call active
>>>>> nta.c:5744 incoming_free() nta: incoming_free(0x7fe17c024090)
>>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>>> timer
>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>> entering
>>>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>> entering
>>>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>> entering
>>>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> 2015-05-08 08:21:30.667707 [DEBUG] sofia.c:6627 Channel
>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [ready][200]
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>> nta.c:1289 agent_timer() nta: timer not set
>>>>> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
>>>>> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
>>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2924 Changing audio
>>>>> DTLS state from HANDSHAKE to SETUP
>>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2832 audio Fingerprint
>>>>> Verified.
>>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3374 Activating Audio
>>>>> Secure RTP SEND
>>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3352 Activating Audio
>>>>> Secure RTP RECV
>>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2872 Changing audio
>>>>> DTLS state from SETUP to READY
>>>>> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure
>>>>> Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80
>>>>> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure
>>>>> Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80
>>>>> 2015-05-08 08:22:02.087708 [DEBUG] switch_rtp.c:1937 rtcp_stats_init:
>>>>> ssrc[-195423717] base_seq[29507]
>>>>>
>>>>> Notice that the INVITE was received at 08:21:30 while DTLS was READY
>>>>> at 2015-05-08 08:22:02 which means that it took 32 seconds to voice. Again,
>>>>> if I am not using TURN/STUN, the whole process is pretty quick (2 seconds).
>>>>>
>>>>> Also, notice:
>>>>> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
>>>>> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
>>>>>
>>>>> Which means that the media is relayed via the TURN server (TURN server
>>>>> IP is 130.211.78.35)...
>>>>>
>>>>> Any idea?
>>>>>
>>>>> Thanks,
>>>>> Adam
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From italorossib at gmail.com  Fri May  8 20:18:10 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Fri, 8 May 2015 13:18:10 -0300
Subject: [Freeswitch-users] IvrMenu - setting channel variables before
 calling exec-app/transfer
In-Reply-To: <554CD9EA.3060405@gmail.com>
References: <mailman.515.1431099645.10851.freeswitch-users@lists.freeswitch.org>
	<554CD9EA.3060405@gmail.com>
Message-ID: <CAJR7f2F_9We8CxOTpLdH3CnAfrqZec6tosGNdmmycX3abjpx1g@mail.gmail.com>

<entry digits="2" action="menu-exec-app" param="set myvar=value" />
<entry digits="2" action="menu-exec-app" param="transfer 1001 XML
default" />

Not tested with set app, but I'm using this already to generate events when
an option is selected
 Em 08/05/2015 12:46, "DanB" <danb.lists at gmail.com> escreveu:

> @covici: thanks, I was hoping in a more elegant solution since every
> transfer will produce a request to http server for dialplan.
>
> @Michael: not before the ivr since the vars I need to set depend on the
> menu selected (eg: call will be forwarded when he presses key so the caller
> becomes the previous called).
>
> DanB
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri May  8 21:00:43 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 May 2015 13:00:43 -0400
Subject: [Freeswitch-users] Trickle ICE with SIP
In-Reply-To: <CADNqFCkDy2OtzeTqF_mDzW4hA75qambrEGxs5JpwL9QwYseh9g@mail.gmail.com>
References: <CADNqFCmOfu_Nq56N9rc6H2DD=9OG7=MTqB1D4dT+kJ9GFcaBSA@mail.gmail.com>
	<C56F753C-D2E5-4072-BDC2-43BCE6E367AB@jerris.com>
	<CADNqFCkDy2OtzeTqF_mDzW4hA75qambrEGxs5JpwL9QwYseh9g@mail.gmail.com>
Message-ID: <C70FE012-59AB-4022-A72A-14C110A03571@jerris.com>

okay.. so to confirm, there is no reason to ever be using turn, so we only are dealing w/ 2-4 seconds.  What results do you get using sip.js and what results do you get using verto.  I know there is some overhead in the browser and there is nothing we can do to get around that issue.

> On May 8, 2015, at 11:43 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:
> 
> Well, it can get up to 30 seconds when using TURN, in other cases it's usually 2-4 seconds.
> 
> On 8 May 2015 at 18:35, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> >From your other post you said thats only when using turn?
> 
> > On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
> >
> > Hi,
> >
> > We are currently using Freeswitch for voice conferencing, we use SIP for signalling. Call setup times are really slow at times (3sec-30sec), can we somehow do Trickle ICE with Freeswitch?
> >
> > Thanks,
> > Adam
> 

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From adam.ben.ayoun1 at gmail.com  Fri May  8 21:04:11 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Fri, 8 May 2015 20:04:11 +0300
Subject: [Freeswitch-users] Trickle ICE with SIP
In-Reply-To: <C70FE012-59AB-4022-A72A-14C110A03571@jerris.com>
References: <CADNqFCmOfu_Nq56N9rc6H2DD=9OG7=MTqB1D4dT+kJ9GFcaBSA@mail.gmail.com>
	<C56F753C-D2E5-4072-BDC2-43BCE6E367AB@jerris.com>
	<CADNqFCkDy2OtzeTqF_mDzW4hA75qambrEGxs5JpwL9QwYseh9g@mail.gmail.com>
	<C70FE012-59AB-4022-A72A-14C110A03571@jerris.com>
Message-ID: <CADNqFCn1giYz0N9TSx4gKvwqid-0dJh7C3jW1O8KDbhfktJ8XA@mail.gmail.com>

Can you explain how can I enable clients with UDP blocked if I don't use
TURN? RTP ports on Freeswitch are UDP.. I haven't tried Verto yet, actually
we are developing a mobile app, in this case it's easy for us to use
something like pjsip for SIP signalling..

On 8 May 2015 at 20:00, Michael Jerris <mike at jerris.com> wrote:

> okay.. so to confirm, there is no reason to ever be using turn, so we only
> are dealing w/ 2-4 seconds.  What results do you get using sip.js and what
> results do you get using verto.  I know there is some overhead in the
> browser and there is nothing we can do to get around that issue.
>
> On May 8, 2015, at 11:43 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
> wrote:
>
> Well, it can get up to 30 seconds when using TURN, in other cases it's
> usually 2-4 seconds.
>
> On 8 May 2015 at 18:35, Michael Jerris <mike at jerris.com> wrote:
>
>> >From your other post you said thats only when using turn?
>>
>> > On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>> wrote:
>> >
>> > Hi,
>> >
>> > We are currently using Freeswitch for voice conferencing, we use SIP
>> for signalling. Call setup times are really slow at times (3sec-30sec), can
>> we somehow do Trickle ICE with Freeswitch?
>> >
>> > Thanks,
>> > Adam
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri May  8 21:06:45 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 May 2015 13:06:45 -0400
Subject: [Freeswitch-users] Trickle ICE with SIP
In-Reply-To: <CADNqFCn1giYz0N9TSx4gKvwqid-0dJh7C3jW1O8KDbhfktJ8XA@mail.gmail.com>
References: <CADNqFCmOfu_Nq56N9rc6H2DD=9OG7=MTqB1D4dT+kJ9GFcaBSA@mail.gmail.com>
	<C56F753C-D2E5-4072-BDC2-43BCE6E367AB@jerris.com>
	<CADNqFCkDy2OtzeTqF_mDzW4hA75qambrEGxs5JpwL9QwYseh9g@mail.gmail.com>
	<C70FE012-59AB-4022-A72A-14C110A03571@jerris.com>
	<CADNqFCn1giYz0N9TSx4gKvwqid-0dJh7C3jW1O8KDbhfktJ8XA@mail.gmail.com>
Message-ID: <4D2BB9C4-50F6-4D72-A022-24D25A952633@jerris.com>

I had not thought of tcp.  

> On May 8, 2015, at 1:04 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:
> 
> Can you explain how can I enable clients with UDP blocked if I don't use TURN? RTP ports on Freeswitch are UDP.. I haven't tried Verto yet, actually we are developing a mobile app, in this case it's easy for us to use something like pjsip for SIP signalling..
> 
> On 8 May 2015 at 20:00, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> okay.. so to confirm, there is no reason to ever be using turn, so we only are dealing w/ 2-4 seconds.  What results do you get using sip.js and what results do you get using verto.  I know there is some overhead in the browser and there is nothing we can do to get around that issue.
> 
>> On May 8, 2015, at 11:43 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
>> 
>> Well, it can get up to 30 seconds when using TURN, in other cases it's usually 2-4 seconds.
>> 
>> On 8 May 2015 at 18:35, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> >From your other post you said thats only when using turn?
>> 
>> > On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
>> >
>> > Hi,
>> >
>> > We are currently using Freeswitch for voice conferencing, we use SIP for signalling. Call setup times are really slow at times (3sec-30sec), can we somehow do Trickle ICE with Freeswitch?
>> >
>> > Thanks,
>> > Adam
>> 
> 
> 
> _________________________________________________________________________
> 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 adam.ben.ayoun1 at gmail.com  Fri May  8 21:19:41 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Fri, 8 May 2015 20:19:41 +0300
Subject: [Freeswitch-users] Trickle ICE with SIP
In-Reply-To: <4D2BB9C4-50F6-4D72-A022-24D25A952633@jerris.com>
References: <CADNqFCmOfu_Nq56N9rc6H2DD=9OG7=MTqB1D4dT+kJ9GFcaBSA@mail.gmail.com>
	<C56F753C-D2E5-4072-BDC2-43BCE6E367AB@jerris.com>
	<CADNqFCkDy2OtzeTqF_mDzW4hA75qambrEGxs5JpwL9QwYseh9g@mail.gmail.com>
	<C70FE012-59AB-4022-A72A-14C110A03571@jerris.com>
	<CADNqFCn1giYz0N9TSx4gKvwqid-0dJh7C3jW1O8KDbhfktJ8XA@mail.gmail.com>
	<4D2BB9C4-50F6-4D72-A022-24D25A952633@jerris.com>
Message-ID: <CADNqFCnR22Ptwb_e8M3qmOgQEjH4CV2j_j=wZx2xcuXGPs5pKQ@mail.gmail.com>

Can you clarify? Do you think there is no reason to use TURN in this case?

On 8 May 2015 at 20:06, Michael Jerris <mike at jerris.com> wrote:

> I had not thought of tcp.
>
> On May 8, 2015, at 1:04 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
> wrote:
>
> Can you explain how can I enable clients with UDP blocked if I don't use
> TURN? RTP ports on Freeswitch are UDP.. I haven't tried Verto yet, actually
> we are developing a mobile app, in this case it's easy for us to use
> something like pjsip for SIP signalling..
>
> On 8 May 2015 at 20:00, Michael Jerris <mike at jerris.com> wrote:
>
>> okay.. so to confirm, there is no reason to ever be using turn, so we
>> only are dealing w/ 2-4 seconds.  What results do you get using sip.js and
>> what results do you get using verto.  I know there is some overhead in the
>> browser and there is nothing we can do to get around that issue.
>>
>> On May 8, 2015, at 11:43 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>> wrote:
>>
>> Well, it can get up to 30 seconds when using TURN, in other cases it's
>> usually 2-4 seconds.
>>
>> On 8 May 2015 at 18:35, Michael Jerris <mike at jerris.com> wrote:
>>
>>> >From your other post you said thats only when using turn?
>>>
>>> > On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>>> wrote:
>>> >
>>> > Hi,
>>> >
>>> > We are currently using Freeswitch for voice conferencing, we use SIP
>>> for signalling. Call setup times are really slow at times (3sec-30sec), can
>>> we somehow do Trickle ICE with Freeswitch?
>>> >
>>> > Thanks,
>>> > Adam
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri May  8 21:36:17 2015
From: brian at freeswitch.org (Brian West)
Date: Fri, 8 May 2015 12:36:17 -0500
Subject: [Freeswitch-users] IvrMenu - setting channel variables before
 calling exec-app/transfer
In-Reply-To: <CAJR7f2F_9We8CxOTpLdH3CnAfrqZec6tosGNdmmycX3abjpx1g@mail.gmail.com>
References: <mailman.515.1431099645.10851.freeswitch-users@lists.freeswitch.org>
	<554CD9EA.3060405@gmail.com>
	<CAJR7f2F_9We8CxOTpLdH3CnAfrqZec6tosGNdmmycX3abjpx1g@mail.gmail.com>
Message-ID: <CAEJMVkAgAjKWu6kz_TOMgaRqfsGsSKKTAyUCQqBmosoTSeBT+A@mail.gmail.com>

What ?talo says should work.

On Fri, May 8, 2015 at 11:18 AM, ?talo Rossi <italorossib at gmail.com> wrote:

>
> <entry digits="2" action="menu-exec-app" param="set myvar=value" />
> <entry digits="2" action="menu-exec-app" param="transfer 1001 XML
> default" />
>
> Not tested with set app, but I'm using this already to generate events
> when an option is selected
>  Em 08/05/2015 12:46, "DanB" <danb.lists at gmail.com> escreveu:
>
>> @covici: thanks, I was hoping in a more elegant solution since every
>> transfer will produce a request to http server for dialplan.
>>
>> @Michael: not before the ivr since the vars I need to set depend on the
>> menu selected (eg: call will be forwarded when he presses key so the caller
>> becomes the previous called).
>>
>> DanB
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
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From mike at jerris.com  Fri May  8 21:51:26 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 May 2015 13:51:26 -0400
Subject: [Freeswitch-users] Trickle ICE with SIP
In-Reply-To: <CADNqFCnR22Ptwb_e8M3qmOgQEjH4CV2j_j=wZx2xcuXGPs5pKQ@mail.gmail.com>
References: <CADNqFCmOfu_Nq56N9rc6H2DD=9OG7=MTqB1D4dT+kJ9GFcaBSA@mail.gmail.com>
	<C56F753C-D2E5-4072-BDC2-43BCE6E367AB@jerris.com>
	<CADNqFCkDy2OtzeTqF_mDzW4hA75qambrEGxs5JpwL9QwYseh9g@mail.gmail.com>
	<C70FE012-59AB-4022-A72A-14C110A03571@jerris.com>
	<CADNqFCn1giYz0N9TSx4gKvwqid-0dJh7C3jW1O8KDbhfktJ8XA@mail.gmail.com>
	<4D2BB9C4-50F6-4D72-A022-24D25A952633@jerris.com>
	<CADNqFCnR22Ptwb_e8M3qmOgQEjH4CV2j_j=wZx2xcuXGPs5pKQ@mail.gmail.com>
Message-ID: <D15E3188-08F7-45A4-82D5-3044BB2E07BB@jerris.com>

The TCP case is the only case, and I had not thought of it.  Not sure it ever really makes sense to do it if we could add TCP \support, but we don't have that at the moment.

> On May 8, 2015, at 1:19 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:
> 
> Can you clarify? Do you think there is no reason to use TURN in this case?
> 
> On 8 May 2015 at 20:06, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> I had not thought of tcp.  
> 
>> On May 8, 2015, at 1:04 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
>> 
>> Can you explain how can I enable clients with UDP blocked if I don't use TURN? RTP ports on Freeswitch are UDP.. I haven't tried Verto yet, actually we are developing a mobile app, in this case it's easy for us to use something like pjsip for SIP signalling..
>> 
>> On 8 May 2015 at 20:00, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> okay.. so to confirm, there is no reason to ever be using turn, so we only are dealing w/ 2-4 seconds.  What results do you get using sip.js and what results do you get using verto.  I know there is some overhead in the browser and there is nothing we can do to get around that issue.
>> 
>>> On May 8, 2015, at 11:43 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
>>> 
>>> Well, it can get up to 30 seconds when using TURN, in other cases it's usually 2-4 seconds.
>>> 
>>> On 8 May 2015 at 18:35, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>> >From your other post you said thats only when using turn?
>>> 
>>> > On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
>>> >
>>> > Hi,
>>> >
>>> > We are currently using Freeswitch for voice conferencing, we use SIP for signalling. Call setup times are really slow at times (3sec-30sec), can we somehow do Trickle ICE with Freeswitch?
>>> >
>>> > Thanks,
>>> > Adam
>>> 
>> 
>> 
>> _________________________________________________________________________
>> 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 kamil.nigmatullin at gmail.com  Fri May  8 22:11:42 2015
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Sat, 9 May 2015 00:11:42 +0600
Subject: [Freeswitch-users] FEC feature in mod_opus
In-Reply-To: <2091369665.2474442.1429001528976.JavaMail.yahoo@mail.yahoo.com>
References: <2091369665.2474442.1429001528976.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <CA+Xtiee7K3mjCQVoxNWfQiunhr8zUsRmaSxLODSKJvAkzyDuWQ@mail.gmail.com>

Hi Dragos,
I tried to use your patch but have no luck with compiling it.

I clonne stable 1.2 and then replace mod_opus.c with your code. Maybe do
you know how to fix it?

This is what I get:
d_opus/mod_opus.c  -fPIC -DPIC -o .libs/mod_opus.o
/usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function
'switch_opus_fmtp_parse':
/usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:155:18: error:
'switch_codec_fmtp_t' has no member named 'stereo'
/usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function
'switch_opus_decode':
/usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:423:3: error:
statement with no effect [-Werror=unused-value]
cc1: all warnings being treated as errors
make[4]: *** [mod_opus.lo] Error 1
make[3]: *** [all] Error 1
make[2]: *** [mod_opus-all] Error 1
make[1]: *** [mod_opus] Error 2
make: *** [mod_opus] Error 2


2015-04-14 14:52 GMT+06:00 Dragos Oancea <droancea at yahoo.com>:

>
> Hi Kamil,
>
> I've done a patch for Opus with FEC , but it has not been merged:
>
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/184/overview
> I was inspired by how they've done it with SILK.
>
>
> Regards,
> Dragos Oancea
>
>
>
> On 2015-04-13 16:49, Kamil Nigmatullin wrote:
> > Hello.
> >
> > Anybody from developers could please clarify if FreeSWITCH uses
> > Forward Error Correction feature in opus codec? We actually are not a
> > specialists in C and cannot fully understand how decode function works
> > in 414 line. According to opus documentation decode function  is
> > called with special flag if previous packet is lost in order to call
> > fec feature. In mod_opus it is hard to say for me as i don't see i or
> > i+1 packets. 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
>
>


-- 
Kamil Nigmatullin
Tel: 77272323748
mob: 7 (707) 2517003
Skype: kamil.nigmatullin
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From danb.lists at gmail.com  Fri May  8 23:07:13 2015
From: danb.lists at gmail.com (DanB)
Date: Fri, 08 May 2015 21:07:13 +0200
Subject: [Freeswitch-users] IvrMenu - setting channel variables before
	calling exec-app/transfer
In-Reply-To: <mailman.534.1431107521.10851.freeswitch-users@lists.freeswitch.org>
References: <mailman.534.1431107521.10851.freeswitch-users@lists.freeswitch.org>
Message-ID: <554D0961.5000404@gmail.com>

@?talo and @Brian, many thanks! This works for me and it is the solution I was after.

Cheers,
DanB

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From mike at jerris.com  Fri May  8 23:26:23 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 May 2015 15:26:23 -0400
Subject: [Freeswitch-users] FEC feature in mod_opus
In-Reply-To: <CA+Xtiee7K3mjCQVoxNWfQiunhr8zUsRmaSxLODSKJvAkzyDuWQ@mail.gmail.com>
References: <2091369665.2474442.1429001528976.JavaMail.yahoo@mail.yahoo.com>
	<CA+Xtiee7K3mjCQVoxNWfQiunhr8zUsRmaSxLODSKJvAkzyDuWQ@mail.gmail.com>
Message-ID: <74103379-A16C-487D-9CEB-ABD039E2205B@jerris.com>

1.2 is quite old, why would you be using it?  Have you tried on top of 1.4?

> On May 8, 2015, at 2:11 PM, Kamil Nigmatullin <kamil.nigmatullin at gmail.com> wrote:
> 
> Hi Dragos, 
> I tried to use your patch but have no luck with compiling it. 
> 
> I clonne stable 1.2 and then replace mod_opus.c with your code. Maybe do you know how to fix it?
> 
> This is what I get:
> d_opus/mod_opus.c  -fPIC -DPIC -o .libs/mod_opus.o
> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function 'switch_opus_fmtp_parse':
> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:155:18: error: 'switch_codec_fmtp_t' has no member named 'stereo'
> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function 'switch_opus_decode':
> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:423:3: error: statement with no effect [-Werror=unused-value]
> cc1: all warnings being treated as errors
> make[4]: *** [mod_opus.lo] Error 1
> make[3]: *** [all] Error 1
> make[2]: *** [mod_opus-all] Error 1
> make[1]: *** [mod_opus] Error 2
> make: *** [mod_opus] Error 2
> 
> 
> 2015-04-14 14:52 GMT+06:00 Dragos Oancea <droancea at yahoo.com <mailto:droancea at yahoo.com>>:
> 
> Hi Kamil,
> 
> I've done a patch for Opus with FEC , but it has not been merged:
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/184/overview <https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/184/overview>
> I was inspired by how they've done it with SILK.
> 
> 
> Regards,
> Dragos Oancea
> 
> 
> 
> On 2015-04-13 16:49, Kamil Nigmatullin wrote:
> > Hello. 
> > 
> > Anybody from developers could please clarify if FreeSWITCH uses
> > Forward Error Correction feature in opus codec? We actually are not a
> > specialists in C and cannot fully understand how decode function works
> > in 414 line. According to opus documentation decode function  is
> > called with special flag if previous packet is lost in order to call
> > fec feature. In mod_opus it is hard to say for me as i don't see i or
> > i+1 packets. Thanks    
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services: 
> > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> > http://www.cluecon.com <http://www.cluecon.com/>
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> 
> -- 
> Kamil Nigmatullin
> Tel: 77272323748
> mob: 7 (707) 2517003
> Skype: kamil.nigmatullin
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May  9 00:05:17 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 8 May 2015 13:05:17 -0700
Subject: [Freeswitch-users] bind_digit_action for lua
Message-ID: <CAGVOjVXXZ9=gvaXxVSu5DoXViO6-Y1QPxhrZyEbuHKbKJA3m6g@mail.gmail.com>

Hi, users.

Below is my default.xml snippet. I am using bind_digit_action for different
inputs bindings and moves the call to that context upon regex match.

<condition>
<action application="bind_digit_action"
data="moderator,~\*,exec:execute_extension,previous XML Previous"/>
        <action application="bind_digit_action"
data="moderator,~\#,exec:execute_extension,next XML Next"/>
        <action application="bind_digit_action"
data="moderator,~^\d$,exec:execute_extension,exten XML Exten"/>
                <action application="digit_action_set_realm"
data="moderator"/>
        <action application="answer"/>
        <action application="playback" data="shout://127.0.0.1/radio"/>
</condition>

Now I want to write my dialplan in lua which I think is more handy than xml.
Upon regex match call to be executed in corresponding functions.

Like, for *
*function next()*
for #,
*function previous()*



I know there is a application *session.setHangupHook(hangup_function_name)*;
Which executes the function(hangup_function_name) upon call hangup.

Is there anything available in lua witch executes functions upon regex
match.
Or any other way for doing this.

Any pointer would be much appreciated.

Thanks.
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From olegstolyar at gmail.com  Sat May  9 00:11:25 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Fri, 8 May 2015 13:11:25 -0700
Subject: [Freeswitch-users] bind_digit_action for lua
In-Reply-To: <CAGVOjVXXZ9=gvaXxVSu5DoXViO6-Y1QPxhrZyEbuHKbKJA3m6g@mail.gmail.com>
References: <CAGVOjVXXZ9=gvaXxVSu5DoXViO6-Y1QPxhrZyEbuHKbKJA3m6g@mail.gmail.com>
Message-ID: <CAC8JMOLH91fz_mQ07wg_gZ9uLCOWJ3qr1yrWz9VTwEWBe2B24A@mail.gmail.com>

I just finished playing with it.  You can just do this:

<action application="bind_digit_action"
data="my_digits,0,exec:lua,your_script.lua,param1 param2 param3"/>



On Fri, May 8, 2015 at 1:05 PM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi, users.
>
> Below is my default.xml snippet. I am using bind_digit_action for
> different inputs bindings and moves the call to that context upon regex
> match.
>
> <condition>
> <action application="bind_digit_action"
> data="moderator,~\*,exec:execute_extension,previous XML Previous"/>
>         <action application="bind_digit_action"
> data="moderator,~\#,exec:execute_extension,next XML Next"/>
>         <action application="bind_digit_action"
> data="moderator,~^\d$,exec:execute_extension,exten XML Exten"/>
>                 <action application="digit_action_set_realm"
> data="moderator"/>
>         <action application="answer"/>
>         <action application="playback" data="shout://127.0.0.1/radio"/>
> </condition>
>
> Now I want to write my dialplan in lua which I think is more handy than
> xml.
> Upon regex match call to be executed in corresponding functions.
>
> Like, for *
> *function next()*
> for #,
> *function previous()*
>
>
>
> I know there is a application
> *session.setHangupHook(hangup_function_name)*;
> Which executes the function(hangup_function_name) upon call hangup.
>
> Is there anything available in lua witch executes functions upon regex
> match.
> Or any other way for doing this.
>
> Any pointer would be much appreciated.
>
> Thanks.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aqsyounas at gmail.com  Sat May  9 00:35:24 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 8 May 2015 13:35:24 -0700
Subject: [Freeswitch-users] bind_digit_action for lua
In-Reply-To: <CAC8JMOLH91fz_mQ07wg_gZ9uLCOWJ3qr1yrWz9VTwEWBe2B24A@mail.gmail.com>
References: <CAGVOjVXXZ9=gvaXxVSu5DoXViO6-Y1QPxhrZyEbuHKbKJA3m6g@mail.gmail.com>
	<CAC8JMOLH91fz_mQ07wg_gZ9uLCOWJ3qr1yrWz9VTwEWBe2B24A@mail.gmail.com>
Message-ID: <CAGVOjVV69HdRqKE2tipETzNqdHGS+=Xh0SA9mjFEC10o+f3Nzg@mail.gmail.com>

Thanks for your reply.

I think i didn't explain my question well. I don't want to execute a
separate lua script upon each regex match.
I want a single script with different functions for corresponding inputs.

<include>
  <context name="default">
   <extension name="radio">
        <condition>
                <action application="answer"/>
                <action application="lua" data="main.lua"/>
  </condition>
  </extension>
</context>


--main.lua
function next()
  logic
end
function previous()
 logic
end


*logic for calling above functions goes here.  *
Is it possible to right above dialplan in a single script with different
fucntions?

Thanks.


On 8 May 2015 at 13:11, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> I just finished playing with it.  You can just do this:
>
> <action application="bind_digit_action"
> data="my_digits,0,exec:lua,your_script.lua,param1 param2 param3"/>
>
>
>
> On Fri, May 8, 2015 at 1:05 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi, users.
>>
>> Below is my default.xml snippet. I am using bind_digit_action for
>> different inputs bindings and moves the call to that context upon regex
>> match.
>>
>> <condition>
>> <action application="bind_digit_action"
>> data="moderator,~\*,exec:execute_extension,previous XML Previous"/>
>>         <action application="bind_digit_action"
>> data="moderator,~\#,exec:execute_extension,next XML Next"/>
>>         <action application="bind_digit_action"
>> data="moderator,~^\d$,exec:execute_extension,exten XML Exten"/>
>>                 <action application="digit_action_set_realm"
>> data="moderator"/>
>>         <action application="answer"/>
>>         <action application="playback" data="shout://127.0.0.1/radio"/>
>> </condition>
>>
>> Now I want to write my dialplan in lua which I think is more handy than
>> xml.
>> Upon regex match call to be executed in corresponding functions.
>>
>> Like, for *
>> *function next()*
>> for #,
>> *function previous()*
>>
>>
>>
>> I know there is a application
>> *session.setHangupHook(hangup_function_name)*;
>> Which executes the function(hangup_function_name) upon call hangup.
>>
>> Is there anything available in lua witch executes functions upon regex
>> match.
>> Or any other way for doing this.
>>
>> Any pointer would be much appreciated.
>>
>> Thanks.
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From olegstolyar at gmail.com  Sat May  9 00:43:41 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Fri, 8 May 2015 13:43:41 -0700
Subject: [Freeswitch-users] bind_digit_action for lua
In-Reply-To: <CAGVOjVV69HdRqKE2tipETzNqdHGS+=Xh0SA9mjFEC10o+f3Nzg@mail.gmail.com>
References: <CAGVOjVXXZ9=gvaXxVSu5DoXViO6-Y1QPxhrZyEbuHKbKJA3m6g@mail.gmail.com>
	<CAC8JMOLH91fz_mQ07wg_gZ9uLCOWJ3qr1yrWz9VTwEWBe2B24A@mail.gmail.com>
	<CAGVOjVV69HdRqKE2tipETzNqdHGS+=Xh0SA9mjFEC10o+f3Nzg@mail.gmail.com>
Message-ID: <CAC8JMO+YwH0mQkfAOPiADqLkitR4Bf6Avg0W59nasA0epeDU-w@mail.gmail.com>

There are several ways.  One is to use a regex to match any digit and
execute a lua script.  Inside the script check the last_matching_digits
session variable.

I have not tried it myself, so not sure how well it will work (regex and
the variable).  If for some reason it does not, you can still use separate
bindings for each digit, call the same lua script from each and pass the
digit in as a parameter like this:

<action application="bind_digit_action" data="my_digits,0,exec:lua,
your_script.lua,0"/>
<action application="bind_digit_action" data="my_digits,1,exec:lua,
your_script.lua,1"/>
<action application="bind_digit_action" data="my_digits,2,exec:lua,
your_script.lua,2"/>


On Fri, May 8, 2015 at 1:35 PM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Thanks for your reply.
>
> I think i didn't explain my question well. I don't want to execute a
> separate lua script upon each regex match.
> I want a single script with different functions for corresponding inputs.
>
> <include>
>   <context name="default">
>    <extension name="radio">
>         <condition>
>                 <action application="answer"/>
>                 <action application="lua" data="main.lua"/>
>   </condition>
>   </extension>
> </context>
>
>
> --main.lua
> function next()
>   logic
> end
> function previous()
>  logic
> end
>
>
> *logic for calling above functions goes here.  *
> Is it possible to right above dialplan in a single script with different
> fucntions?
>
> Thanks.
>
>
> On 8 May 2015 at 13:11, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>> I just finished playing with it.  You can just do this:
>>
>> <action application="bind_digit_action"
>> data="my_digits,0,exec:lua,your_script.lua,param1 param2 param3"/>
>>
>>
>>
>> On Fri, May 8, 2015 at 1:05 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>>
>>> Hi, users.
>>>
>>> Below is my default.xml snippet. I am using bind_digit_action for
>>> different inputs bindings and moves the call to that context upon regex
>>> match.
>>>
>>> <condition>
>>> <action application="bind_digit_action"
>>> data="moderator,~\*,exec:execute_extension,previous XML Previous"/>
>>>         <action application="bind_digit_action"
>>> data="moderator,~\#,exec:execute_extension,next XML Next"/>
>>>         <action application="bind_digit_action"
>>> data="moderator,~^\d$,exec:execute_extension,exten XML Exten"/>
>>>                 <action application="digit_action_set_realm"
>>> data="moderator"/>
>>>         <action application="answer"/>
>>>         <action application="playback" data="shout://127.0.0.1/radio"/>
>>> </condition>
>>>
>>> Now I want to write my dialplan in lua which I think is more handy than
>>> xml.
>>> Upon regex match call to be executed in corresponding functions.
>>>
>>> Like, for *
>>> *function next()*
>>> for #,
>>> *function previous()*
>>>
>>>
>>>
>>> I know there is a application
>>> *session.setHangupHook(hangup_function_name)*;
>>> Which executes the function(hangup_function_name) upon call hangup.
>>>
>>> Is there anything available in lua witch executes functions upon regex
>>> match.
>>> Or any other way for doing this.
>>>
>>> Any pointer would be much appreciated.
>>>
>>> Thanks.
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at gmail.com  Sat May  9 02:12:58 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 9 May 2015 00:12:58 +0200
Subject: [Freeswitch-users] Clock precision and raw audio recording
Message-ID: <CACOYK=i5z_9=ssMBWQfw_e60q8fe2a-8hmMu_8vWEBk-wWtE1A@mail.gmail.com>

I'm observing an effect which needs explanation. Comments from the
core developers will be appreciated. The effect was tested with
versions 1.4.18 and today's master, on 64-bit Debian 7 and Ubuntu
14.04. All test calls were in PCMU or PCMA.

My customer requested me to build a server for call quality assurance
for their telephony system. I installed FreeSWITCH and set the
following in the public dialplan to record the incoming audio:

    <extension name="record">
      <condition field="destination_number" expression="^record_(.+)$">
        <action application="jitterbuffer" data="60:200:20"/>
        <action application="set" data="RECORD_READ_ONLY=true"/>
        <action application="set" data="send_silence_when_idle=400"/>
        <action application="set" data="record_waste_resources=true"/>
        <action application="answer"/>
        <action application="record_session" data="/var/tmp/record_$1"/>
        <action application="playback" data="silence_stream://-1"/>
      </condition>
    </extension>

The first try was with a DigitalOcean (KVM) virtual machine. I started
recording *.wav files, and sometimes there were skipped frames: 1-2
skipped frames in a 2-minute call, one in every 10-15 calls.

Then I changed the configuration to write raw audio files (removed the
.wav extension from the record_session argument). As a result, the
recorded input audio was quite bad: lost frames every few seconds in
every call.

Then I made test calls within the server itself, by originating a call
to its public profile and playing the test WAV audio:

fs_cli  -x 'originate sofia/external/record_03 at 111.222.222.111:5080
&playback(/var/tmp/ITU-T_P_50_BRITISH_ENGLISH.wav)'

The resulting input raw audio was also choppy. The same result was on
another VM on a different physical server at DigitalOcean.

Then I made the same self-call tests on a Xen VM and on a baremetal
ARM server, and there the recorded audio was of perfect quality.

Self-calls with recording into WAV files produced audio of perfect quality.

Setting RECORD_USE_THREAD=false did not change the effect.

Example of choppy received audio, converted from PCMU to WAV for convenience:
http://www.k-open.com/s/record_04-in.wav
The source audio:
http://murmur.voxserv.ch/media/ITU-T_P_50_BRITISH_ENGLISH.wav

So, it looks like the clock that is available at KVM is not precise
enough, but that's not my question.

QUESTION: why is raw recording so much more sensitive to the clock precision?

thanks,
stanislav


From adam.ben.ayoun1 at gmail.com  Sat May  9 03:26:14 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Sat, 9 May 2015 02:26:14 +0300
Subject: [Freeswitch-users] Trickle ICE with SIP
In-Reply-To: <D15E3188-08F7-45A4-82D5-3044BB2E07BB@jerris.com>
References: <CADNqFCmOfu_Nq56N9rc6H2DD=9OG7=MTqB1D4dT+kJ9GFcaBSA@mail.gmail.com>
	<C56F753C-D2E5-4072-BDC2-43BCE6E367AB@jerris.com>
	<CADNqFCkDy2OtzeTqF_mDzW4hA75qambrEGxs5JpwL9QwYseh9g@mail.gmail.com>
	<C70FE012-59AB-4022-A72A-14C110A03571@jerris.com>
	<CADNqFCn1giYz0N9TSx4gKvwqid-0dJh7C3jW1O8KDbhfktJ8XA@mail.gmail.com>
	<4D2BB9C4-50F6-4D72-A022-24D25A952633@jerris.com>
	<CADNqFCnR22Ptwb_e8M3qmOgQEjH4CV2j_j=wZx2xcuXGPs5pKQ@mail.gmail.com>
	<D15E3188-08F7-45A4-82D5-3044BB2E07BB@jerris.com>
Message-ID: <CADNqFCkKLqRQVg11aGof3WAUzVpnMZryiejBB6mWoZTc2RqAOw@mail.gmail.com>

Got it. Any way to do trickle ice with Freeswitch?

On 8 May 2015 at 20:51, Michael Jerris <mike at jerris.com> wrote:

> The TCP case is the only case, and I had not thought of it.  Not sure it
> ever really makes sense to do it if we could add TCP \support, but we don't
> have that at the moment.
>
> On May 8, 2015, at 1:19 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
> wrote:
>
> Can you clarify? Do you think there is no reason to use TURN in this case?
>
> On 8 May 2015 at 20:06, Michael Jerris <mike at jerris.com> wrote:
>
>> I had not thought of tcp.
>>
>> On May 8, 2015, at 1:04 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>> wrote:
>>
>> Can you explain how can I enable clients with UDP blocked if I don't use
>> TURN? RTP ports on Freeswitch are UDP.. I haven't tried Verto yet, actually
>> we are developing a mobile app, in this case it's easy for us to use
>> something like pjsip for SIP signalling..
>>
>> On 8 May 2015 at 20:00, Michael Jerris <mike at jerris.com> wrote:
>>
>>> okay.. so to confirm, there is no reason to ever be using turn, so we
>>> only are dealing w/ 2-4 seconds.  What results do you get using sip.js and
>>> what results do you get using verto.  I know there is some overhead in the
>>> browser and there is nothing we can do to get around that issue.
>>>
>>> On May 8, 2015, at 11:43 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>>> wrote:
>>>
>>> Well, it can get up to 30 seconds when using TURN, in other cases it's
>>> usually 2-4 seconds.
>>>
>>> On 8 May 2015 at 18:35, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> >From your other post you said thats only when using turn?
>>>>
>>>> > On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>>>> wrote:
>>>> >
>>>> > Hi,
>>>> >
>>>> > We are currently using Freeswitch for voice conferencing, we use SIP
>>>> for signalling. Call setup times are really slow at times (3sec-30sec), can
>>>> we somehow do Trickle ICE with Freeswitch?
>>>> >
>>>> > Thanks,
>>>> > Adam
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat May  9 09:07:39 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 09 May 2015 05:07:39 +0000
Subject: [Freeswitch-users] WebRTC slow connection time
In-Reply-To: <CADNqFCm+BW9fY2_r-pEUjndR683VZDNu577DiLGY9e_72J7o_g@mail.gmail.com>
References: <CADNqFCky4iMXF-5vukKcqmT8=Q+anVHdtn+0zueV+4yJ3GUC6g@mail.gmail.com>
	<E6A2DD46-EA8B-400A-A58D-AA9547DDA682@jerris.com>
	<CADNqFC=ba=eD8DCAicLLNx6_=xw0JRFYSdM+UDo2D35OGLR_+g@mail.gmail.com>
	<CADNqFC=EyojairQTzah9PNX0Wwog5jj4Tcudi_UvmeCTJbtCzA@mail.gmail.com>
	<CALXCt0r_CQMFG5Ldjj9K1Q5g9m_t+JN6HShQgbbeXTzN5nkUqA@mail.gmail.com>
	<CACXsapPscnF5kZvtPesYKhUg-neiXoyrD=kP3Zk0vakzvtuy1g@mail.gmail.com>
	<CADNqFCm+BW9fY2_r-pEUjndR683VZDNu577DiLGY9e_72J7o_g@mail.gmail.com>
Message-ID: <CAHtxdDeENstWfXA-i6OUHam_d5dZPrx3r0+RF+FYZBJKV7WeMQ@mail.gmail.com>

Websockets use http as base transport and can be proxed by http proxy.

On Fri, May 8, 2015, 19:15 Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:

> I am probably missing something, we do use 80/443 for TCP/TLS for
> signalling. My thinking is that media will use UDP which can be restricted.
>
> On 8 May 2015 at 19:09, Victor Medina <victor.medina at cibersys.com> wrote:
>
>> What are the chalenges of putting the internal channel on 80(tcp) and
>> 443(tls)? (besides the obvious things of having to deal with a server
>> running under a privileged ports)
>>
>> 2015-05-08 11:28 GMT-04:30 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>>
>>
>>> Btw, that (accomodating clients that only do tcp 80/443) is the biggest
>>> and almost unique reason of Skype success...
>>>
>>> sent from my mobile,
>>> Giovanni Maruzzelli
>>> cell: +39 347 266 56 18
>>> On May 8, 2015 5:53 PM, "Adam Ben-Ayoun" <adam.ben.ayoun1 at gmail.com>
>>> wrote:
>>>
>>>> Is there an easier way to deal with clients which can only do TCP on
>>>> port 443/80 for example?
>>>>
>>>> On 8 May 2015 at 18:40, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>>>> wrote:
>>>>
>>>>> I am using TURN as a TCP fallback (when UDP is blocked).
>>>>>
>>>>> On 8 May 2015 at 18:34, Michael Jerris <mike at jerris.com> wrote:
>>>>>
>>>>>> If you have freeswitch on a public address, why would you ever use
>>>>>> TURN?
>>>>>>
>>>>>> On May 8, 2015, at 4:27 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>> Hi guys,
>>>>>>
>>>>>> I am using Freeswitch from master (month old). I tried several WebRTC
>>>>>> clients (my own test app on Android and sipml5) and on certain WiFi's I am
>>>>>> getting connection time up to 35 seconds when using STUN and TURN servers.
>>>>>> Also, when I am using TURN and STUN, Freeswitch chooses the TURN candidate
>>>>>> although as far as connectivity, when I am not using STUN and TURN I am
>>>>>> connecting successfully after ~2 seconds (so no real need for TURN).
>>>>>> Attaching the SIP trace from Freeswitch:
>>>>>>
>>>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal
>>>>>> r_set_params
>>>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>>> entering
>>>>>> nua.c:575 nua_set_params() nua: nua_set_params: entering
>>>>>> nua_stack.c:529 nua_signal() nua((nil)): sent signal r_set_params
>>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930,
>>>>>> ...) called
>>>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200
>>>>>> OK
>>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>>> entering
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal
>>>>>> r_set_params
>>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>>> entering
>>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe18c001930,
>>>>>> ...) called
>>>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal
>>>>>> r_set_params
>>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>>> entering
>>>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200
>>>>>> OK
>>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c001930,
>>>>>> ...) called
>>>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200
>>>>>> OK
>>>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal
>>>>>> r_set_params
>>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>>> entering
>>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe184001930,
>>>>>> ...) called
>>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>>> entering
>>>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200
>>>>>> OK
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>>> entering
>>>>>> nua_stack.c:569 nua_stack_signal() nua((nil)): recv signal
>>>>>> r_set_params
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>>> entering
>>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe188001930,
>>>>>> ...) called
>>>>>> nua_stack.c:271 nua_stack_event() nua((nil)): event r_set_params 200
>>>>>> OK
>>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>>> entering
>>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>>> entering
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
>>>>>> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
>>>>>> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
>>>>>> 0x7fe17c089c50 from (ws/82.166.84.247:53645) has 2598 bytes, veclen
>>>>>> = 1
>>>>>> recv 2598 bytes from ws/[82.166.84.247]:53645 at 08:21:30.468866:
>>>>>>
>>>>>>  ------------------------------------------------------------------------
>>>>>>    INVITE sip:991234 at aaaa SIP/2.0
>>>>>>    Via: SIP/2.0/WS
>>>>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport
>>>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>>>    To: <sip:991234 at aaaa>
>>>>>>    Contact: "asdasda"<
>>>>>> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
>>>>>> >;+g.oma.sip-im;language="en,fr"
>>>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>>>    CSeq: 28092 INVITE
>>>>>>    Content-Type: application/sdp
>>>>>>    Content-Length: 2039
>>>>>>    Max-Forwards: 70
>>>>>>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>>>>>>    Organization: Doubango Telecom
>>>>>>
>>>>>>    v=0
>>>>>>    o=- 3843427479443760600 2 IN IP4 127.0.0.1
>>>>>>    s=Doubango Telecom - chrome
>>>>>>    t=0 0
>>>>>>    a=group:BUNDLE audio
>>>>>>    a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>>>    m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>>>>>    c=IN IP4 130.211.78.35
>>>>>>    a=rtcp:58209 IN IP4 130.211.78.35
>>>>>>    a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
>>>>>> generation 0
>>>>>>    a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
>>>>>> generation 0
>>>>>>    a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ
>>>>>> srflx raddr 10.0.0.10 rport 63888 generation 0
>>>>>>    a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ
>>>>>> srflx raddr 10.0.0.10 rport 63888 generation 0
>>>>>>    a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host
>>>>>> tcptype active generation 0
>>>>>>    a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host
>>>>>> tcptype active generation 0
>>>>>>    a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ
>>>>>> relay raddr 82.166.84.247 rport 53792 generation 0
>>>>>>    a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ
>>>>>> relay raddr 82.166.84.247 rport 53792 generation 0
>>>>>>    a=ice-ufrag:EDo5kr308/TXhitG
>>>>>>    a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>>>>>>    a=ice-options:google-ice
>>>>>>    a=fingerprint:sha-256
>>>>>> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>>>>>>    a=setup:actpass
>>>>>>    a=mid:audio
>>>>>>    a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>>>>>    a=extmap:3
>>>>>> http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>>>>    a=sendrecv
>>>>>>    a=rtcp-mux
>>>>>>    a=rtpmap:111 opus/48000/2
>>>>>>    a=fmtp:111 minptime=10; useinbandfec=1
>>>>>>    a=rtpmap:103 ISAC/16000
>>>>>>    a=rtpmap:104 ISAC/32000
>>>>>>    a=rtpmap:9 G722/8000
>>>>>>    a=rtpmap:0 PCMU/8000
>>>>>>    a=rtpmap:8 PCMA/8000
>>>>>>    a=rtpmap:106 CN/32000
>>>>>>    a=rtpmap:105 CN/16000
>>>>>>    a=rtpmap:13 CN/8000
>>>>>>    a=rtpmap:126 telephone-event/8000
>>>>>>    a=maxptime:60
>>>>>>    a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>>>>>>    a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>>> b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>>>>    a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>>>    a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>>>>
>>>>>>  ------------------------------------------------------------------------
>>>>>> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
>>>>>> 0x7fe17c089c50 (2598 bytes) from ws/82.166.84.247:53645/sip
>>>>>> next=(nil)
>>>>>> nta.c:2880 agent_recv_request() nta: received INVITE sip:991234 at aaaa
>>>>>> SIP/2.0 (CSeq 28092)
>>>>>> nta.c:3174 agent_check_request_via() nta: Via check:
>>>>>> received=82.166.84.247
>>>>>> nta.c:3085 agent_recv_request() nta: INVITE (28092) going to a
>>>>>> default leg
>>>>>> nta.c:1350 set_timeout() nta: timer set to 2000 ms
>>>>>> nua_server.c:102 nua_stack_process_request() nua:
>>>>>> nua_stack_process_request: entering
>>>>>> nua_stack.c:899 nh_create() nua: nh_create: entering
>>>>>> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
>>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>>> entering
>>>>>> soa.c:280 soa_clone() soa_clone(static::0x7fe17c001930,
>>>>>> 0x7fe17c001130, 0x7fe17c0cf210) called
>>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0,
>>>>>> ...) called
>>>>>> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fe17c04a2e0)
>>>>>> soa.c:1302 soa_init_offer_answer()
>>>>>> soa_init_offer_answer(static::0x7fe17c0bf4d0) called
>>>>>> soa.c:1171 soa_set_remote_sdp()
>>>>>> soa_set_remote_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe17c0be55f, 2039)
>>>>>> called
>>>>>> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fe17c0cf210): adding
>>>>>> session usage
>>>>>> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
>>>>>> 82.166.84.247:53645
>>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>>> 0x7fe17c0d4d90 0x7fe17c0190a0 140 (140)
>>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>>> 0x7fe17c0d4d90 0x7fe17c0be3ab 86 (86)
>>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>>> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
>>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>>> 0x7fe17c0d4d90 0x7fe17c01912c 101 (101)
>>>>>> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 394 bytes of
>>>>>> 394 to ws/82.166.84.247:53645
>>>>>> tport.c:3492 tport_send_msg() tport_vsend returned 394
>>>>>> send 394 bytes to ws/[82.166.84.247]:53645 at 08:21:30.469343:
>>>>>>
>>>>>>  ------------------------------------------------------------------------
>>>>>>    SIP/2.0 100 Trying
>>>>>>    Via: SIP/2.0/WS
>>>>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>>>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>>>    To: <sip:991234 at aaaa>
>>>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>>>    CSeq: 28092 INVITE
>>>>>>    User-Agent:
>>>>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>>>>>>    Content-Length: 0
>>>>>>
>>>>>>
>>>>>>  ------------------------------------------------------------------------
>>>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>>>> timer
>>>>>> nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (28092)
>>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_invite
>>>>>> 100 Trying
>>>>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210):
>>>>>> call state changed: init -> received, received offer
>>>>>> soa.c:1098 soa_get_remote_sdp()
>>>>>> soa_get_remote_sdp(static::0x7fe17c0bf4d0, [0x7fe1aa7b35d8],
>>>>>> [0x7fe1aa7b35e0], [(nil)]) called
>>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state
>>>>>> 100 Trying
>>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>>> entering
>>>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>>>> timer
>>>>>> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_channel.c:1075 New Channel
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx
>>>>>> [3ac3c622-f55b-11e4-a447-7d37723461ed]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>>> entering
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_NEW
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8848
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx receiving invite from
>>>>>> 82.166.84.247:53645 version: 1.5.15b git b01352c 2015-04-23
>>>>>> 04:33:08Z 64bit
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:8960 IP 82.166.84.247
>>>>>> Approved by acl "domains[]". Access Granted.
>>>>>> nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
>>>>>> nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle
>>>>>> (nil)
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:10113 Setting NAT mode
>>>>>> based on via received
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [received][100]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6637 Remote SDP:
>>>>>> v=0
>>>>>> o=- 3843427479443760600 2 IN IP4 127.0.0.1
>>>>>> s=Doubango Telecom - chrome
>>>>>> t=0 0
>>>>>> a=group:BUNDLE audio
>>>>>> a=msid-semantic: WMS QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>>> m=audio 58209 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>>>>> c=IN IP4 130.211.78.35
>>>>>> a=rtpmap:111 opus/48000/2
>>>>>> a=fmtp:111 minptime=10; useinbandfec=1
>>>>>> a=rtpmap:103 ISAC/16000
>>>>>> a=rtpmap:104 ISAC/32000
>>>>>> a=rtpmap:9 G722/8000
>>>>>> a=rtpmap:0 PCMU/8000
>>>>>> a=rtpmap:8 PCMA/8000
>>>>>> a=rtpmap:106 CN/32000
>>>>>> a=rtpmap:105 CN/16000
>>>>>> a=rtpmap:13 CN/8000
>>>>>> a=rtpmap:126 telephone-event/8000
>>>>>> a=rtcp:58209 IN IP4 130.211.78.35
>>>>>> a=candidate:2162125114 1 udp 2122260223 10.0.0.10 63888 typ host
>>>>>> generation 0
>>>>>> a=candidate:2162125114 2 udp 2122260223 10.0.0.10 63888 typ host
>>>>>> generation 0
>>>>>> a=candidate:3260192690 1 udp 1686052607 82.166.84.247 63888 typ srflx
>>>>>> raddr 10.0.0.10 rport 63888 generation 0
>>>>>> a=candidate:3260192690 2 udp 1686052607 82.166.84.247 63888 typ srflx
>>>>>> raddr 10.0.0.10 rport 63888 generation 0
>>>>>> a=candidate:3462174154 1 tcp 1518280447 10.0.0.10 0 typ host tcptype
>>>>>> active generation 0
>>>>>> a=candidate:3462174154 2 tcp 1518280447 10.0.0.10 0 typ host tcptype
>>>>>> active generation 0
>>>>>> a=candidate:3098925784 1 udp 25108223 130.211.78.35 58209 typ relay
>>>>>> raddr 82.166.84.247 rport 53792 generation 0
>>>>>> a=candidate:3098925784 2 udp 25108223 130.211.78.35 58209 typ relay
>>>>>> raddr 82.166.84.247 rport 53792 generation 0
>>>>>> a=ice-ufrag:EDo5kr308/TXhitG
>>>>>> a=ice-pwd:3HWj7s5L/1g2BwIrjJNPnubB
>>>>>> a=ice-options:google-ice
>>>>>> a=fingerprint:sha-256
>>>>>> 2D:92:8B:F0:BD:9C:C3:85:9F:B6:32:4C:B9:73:38:AD:82:1A:D3:02:F5:D8:7B:9E:0E:D4:86:FB:A7:DF:E1:C6
>>>>>> a=setup:actpass
>>>>>> a=mid:audio
>>>>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>>>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>>>> a=rtcp-mux
>>>>>> a=maxptime:60
>>>>>> a=ssrc:4099543579 cname:X6JZwCe/LeIJBOws
>>>>>> a=ssrc:4099543579 msid:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>>> b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>>>> a=ssrc:4099543579 mslabel:QqT0ePCaEFzZ3vHeaMToEsTZan1K2F3O3czm
>>>>>> a=ssrc:4099543579 label:b8082781-4098-4ac5-8c96-2848d2e9e7e4
>>>>>>
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6903 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_NEW -> CS_INIT
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:491 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State NEW
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_INIT
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:87
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA INIT
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:40
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard INIT
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:48 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_INIT ->
>>>>>> CS_ROUTING
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:512 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State INIT going to sleep
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_ROUTING
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:2204 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change DOWN -> RINGING
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:123
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA ROUTING
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:166
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard ROUTING
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_dialplan_xml.c:635 Processing
>>>>>> asdasda <1000>->991234 in context public
>>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
>>>>>> [public->cdquality_conferences_with_api] continue=false
>>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (FAIL)
>>>>>> [cdquality_conferences_with_api] destination_number(991234) =~
>>>>>> /^(75\d{4,36})$/ break=on-false
>>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx parsing
>>>>>> [public->test_conferences] continue=false
>>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Regex (PASS)
>>>>>> [test_conferences] destination_number(991234) =~ /^(99\d{4,36})$/
>>>>>> break=on-false
>>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action answer()
>>>>>> Dialplan: sofia/internal/1000 at xxx.xxx.xxx.xxx Action
>>>>>> conference(991234-${domain_name}@test)
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:216 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State Change CS_ROUTING ->
>>>>>> CS_EXECUTE
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1396 Send
>>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:528 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State ROUTING going to sleep
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:472 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Running State Change CS_EXECUTE
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:535 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) State EXECUTE
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:178
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx SOFIA EXECUTE
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_state_machine.c:258
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Standard EXECUTE
>>>>>> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx answer()
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [opus:111:48000:60:0:1]/[opus:116:48000:20:0:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah
>>>>>> HUMBUG! Sticking with opus at 48000h@20i
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio
>>>>>> Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [opus:111:48000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [opus:111:48000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [ISAC:103:16000:30:32000:1]/[opus:116:48000:20:0:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [ISAC:103:16000:30:32000:1]/[PCMU:0:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [ISAC:103:16000:30:32000:1]/[PCMA:8:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [ISAC:104:32000:30:32000:1]/[opus:116:48000:20:0:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [ISAC:104:32000:30:32000:1]/[PCMU:0:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [ISAC:104:32000:30:32000:1]/[PCMA:8:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [G722:9:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [G722:9:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [G722:9:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [PCMU:0:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [PCMU:0:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah
>>>>>> HUMBUG! Sticking with PCMU at 8000h@20i
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio
>>>>>> Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [PCMU:0:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [PCMA:8:8000:60:64000:1]/[opus:116:48000:20:0:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [PCMA:8:8000:60:64000:1]/[PCMU:0:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [PCMA:8:8000:60:64000:1]/[PCMA:8:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3701 Bah
>>>>>> HUMBUG! Sticking with PCMA at 8000h@20i
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3727 Audio
>>>>>> Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [CN:105:16000:60:0:1]/[opus:116:48000:20:0:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [CN:105:16000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [CN:105:16000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [CN:13:8000:60:0:1]/[opus:116:48000:20:0:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [CN:13:8000:60:0:1]/[PCMU:0:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3672 Audio
>>>>>> Codec Compare [CN:13:8000:60:0:1]/[PCMA:8:8000:20:64000:1]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3588 Set
>>>>>> telephone-event payload to 126
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set
>>>>>> bitrate to local settings [-1000bps]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_opus.c:289 Opus encoder set
>>>>>> bitrate to local settings [-1000bps]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2507 Set Codec
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx opus/48000 20 ms 960 samples 0
>>>>>> bits 1 channels
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:111
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Original read codec set to
>>>>>> opus:116
>>>>>> 2015-05-08 08:21:30.467711 [WARNING] switch_core_media.c:2791 NO
>>>>>> candidate ACL defined, Defaulting to wan.auto
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>>> Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 1 proto: udp type: host addr: 10.0.0.10:63888
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>>> Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 2 proto: udp type: host addr: 10.0.0.10:63888
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>>> Candidate cid: 1 proto: udp type: srflx addr: 82.166.84.247:63888
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose
>>>>>> audio Candidate cid: 1 proto: udp type: srflx addr:
>>>>>> 82.166.84.247:63888
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>>> Candidate cid: 2 proto: udp type: srflx addr: 82.166.84.247:63888
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2824 Choose
>>>>>> audio Candidate cid: 2 proto: udp type: srflx addr:
>>>>>> 82.166.84.247:63888
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>>> Candidate cid: 1 proto: udp type: relay addr: 130.211.78.35:58209
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 1 proto: udp type: relay addr:
>>>>>> 130.211.78.35:58209
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:2815 Checking
>>>>>> Candidate cid: 2 proto: udp type: relay addr: 130.211.78.35:58209
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2829 Save
>>>>>> audio Candidate cid: 2 proto: udp type: relay addr:
>>>>>> 130.211.78.35:58209
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2953 setting
>>>>>> remote audio ice addr to 82.166.84.247:63888 based on candidate
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_core_media.c:2978 setting
>>>>>> remote rtcp audio addr to 82.166.84.247:63888 based on candidate
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:3935 Set 2833
>>>>>> dtmf send/recv payload to 126
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5171 AUDIO RTP
>>>>>> [sofia/internal/1000 at xxx.xxx.xxx.xxx] 172.30.0.219 port 19864 ->
>>>>>> 82.166.84.247 port 63888 codec: 111 ms: 20
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3559 Starting timer
>>>>>> [soft] 960 bytes per 20ms
>>>>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5345 Activating
>>>>>> Audio ICE
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] switch_rtp.c:4009 Activating RTP
>>>>>> audio ICE: EDo5kr308/TXhitG:v9QogVGvZ8jGwYIi 82.166.84.247:63888
>>>>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5388 Activating
>>>>>> RTCP PORT 63888
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:3909 RTCP send rate
>>>>>> is: 10000 and packet rate is: 20000 Remote Port: 63888
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_rtp.c:2349 Setting RTCP
>>>>>> remote addr to 82.166.84.247:63888
>>>>>> 2015-05-08 08:21:30.467711 [INFO] switch_core_media.c:5396 Skipping
>>>>>> RTCP ICE (Same as RTP)
>>>>>> 2015-05-08 08:21:30.467711 [INFO] switch_rtp.c:3101 Activate RTP/RTCP
>>>>>> audio DTLS client
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5469 Set 2833
>>>>>> dtmf send payload to 126
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5475 Set 2833
>>>>>> dtmf receive payload to 126
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_media.c:5503 Set
>>>>>> comfort noise payload to 106
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] sofia_media.c:92 Pre-Answer
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx!
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3419 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change RINGING ->
>>>>>> EARLY
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_sofia.c:780 Local SDP
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx:
>>>>>> v=0
>>>>>> o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>>>>>> s=FreeSWITCH
>>>>>> c=IN IP4 xxx.xxx.xxx.xxx
>>>>>> t=0 0
>>>>>> a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>>>> m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>>>>>> a=rtpmap:111 opus/48000/2
>>>>>> a=fmtp:111 useinbandfec=1; minptime=10
>>>>>> a=rtpmap:126 telephone-event/8000
>>>>>> a=rtpmap:106 CN/8000
>>>>>> a=ptime:20
>>>>>> a=sendrecv
>>>>>> a=fingerprint:sha-256
>>>>>> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>>>>>> a=rtcp-mux
>>>>>> a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>>>>>> a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>>>>>> a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>>>>>> a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>>>> a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>>>>>> a=ice-ufrag:v9QogVGvZ8jGwYIi
>>>>>> a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>>>>>> a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
>>>>>> generation 0
>>>>>>
>>>>>> nua.c:879 nua_respond() nua: nua_respond: entering
>>>>>> nua_stack.c:529 nua_signal() nua(0x7fe17c0cf210): sent signal
>>>>>> r_respond
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send
>>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>>> nua_stack.c:573 nua_stack_signal() nua(0x7fe17c0cf210): recv signal
>>>>>> r_respond 200 OK
>>>>>> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params:
>>>>>> entering
>>>>>> soa.c:403 soa_set_params() soa_set_params(static::0x7fe17c0bf4d0,
>>>>>> ...) called
>>>>>> soa.c:1052 soa_set_user_sdp()
>>>>>> soa_set_user_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe190041d31, -1) called
>>>>>> soa.c:890 soa_set_capability_sdp()
>>>>>> soa_set_capability_sdp(static::0x7fe17c0bf4d0, (nil), 0x7fe190041d31, -1)
>>>>>> called
>>>>>> 2015-05-08 08:21:30.467711 [NOTICE] mod_dptools.c:1292 Channel [
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx] has been answered
>>>>>> nua_session.c:2320 nua_invite_server_respond() nua:
>>>>>> nua_invite_server_respond: entering
>>>>>> soa.c:1515 soa_generate_answer()
>>>>>> soa_generate_answer(static::0x7fe17c0bf4d0) called
>>>>>> soa_static.c:1146 offer_answer_step()
>>>>>> soa_static_offer_answer_action(0x7fe17c0bf4d0, soa_generate_answer): called
>>>>>> soa_static.c:1187 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>>>>> soa_generate_answer): generating local description
>>>>>> soa_static.c:1228 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>>>>> soa_generate_answer): upgrade with remote description
>>>>>> soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fe1aa7b1a30,
>>>>>> 0x7fe17c026ac0, ""): called
>>>>>> soa_static.c:1444 offer_answer_step() soa_static(0x7fe17c0bf4d0,
>>>>>> soa_generate_answer): storing local description
>>>>>> soa.c:1730 soa_activate() soa_activate(static::0x7fe17c0bf4d0, (nil))
>>>>>> called
>>>>>> soa.c:1270 soa_get_local_sdp()
>>>>>> soa_get_local_sdp(static::0x7fe17c0bf4d0, [(nil)], [0x7fe1aa7b3b58],
>>>>>> [0x7fe1aa7b3b54]) called
>>>>>> tport.c:3257 tport_tsend() tport_tsend(0x7fe17c0d4ba0) tpn = WS/
>>>>>> 82.166.84.247:53645
>>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>>> 0x7fe17c0d4d90 0x7fe17c0c5560 136 (136)
>>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>>> 0x7fe17c0d4d90 0x7fe17c0be3ab 63 (63)
>>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>>> 0x7fe17c0d4d90 0x7fe17c0c55e8 41 (41)
>>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>>> 0x7fe17c0d4d90 0x7fe17c0be480 67 (67)
>>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>>> 0x7fe17c0d4d90 0x7fe17c0c5611 631 (631)
>>>>>> tport_type_ws.c:311 tport_send_stream_ws() tport_ws_writevec: vec
>>>>>> 0x7fe17c0d4d90 0x7fe17c0d3020 864 (864)
>>>>>> tport.c:3594 tport_vsend() tport_vsend(0x7fe17c0d4ba0): 1802 bytes of
>>>>>> 1802 to ws/82.166.84.247:53645
>>>>>> tport.c:3492 tport_send_msg() tport_vsend returned 1802
>>>>>> send 1802 bytes to ws/[82.166.84.247]:53645 at 08:21:30.479618:
>>>>>>
>>>>>>  ------------------------------------------------------------------------
>>>>>>    SIP/2.0 200 OK
>>>>>>    Via: SIP/2.0/WS
>>>>>> df7jal23ls0d.invalid;branch=z9hG4bKml28I3BsDokxOyXDaHfSBPW94I6MO4HK;rport=53645;received=82.166.84.247
>>>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>>>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>>>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>>>    CSeq: 28092 INVITE
>>>>>>    Contact: <sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp>
>>>>>>    User-Agent:
>>>>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150423T043308Z~b01352c133~64bit
>>>>>>    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: 864
>>>>>>    Remote-Party-ID: "991234" <sip:991234 at aaaa
>>>>>> >;party=calling;privacy=off;screen=no
>>>>>>
>>>>>>    v=0
>>>>>>    o=FreeSWITCH 1431053426 1431053427 IN IP4 xxx.xxx.xxx.xxx
>>>>>>    s=FreeSWITCH
>>>>>>    c=IN IP4 xxx.xxx.xxx.xxx
>>>>>>    t=0 0
>>>>>>    a=msid-semantic: WMS tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>>>>    m=audio 19864 UDP/TLS/RTP/SAVPF 111 126 106
>>>>>>    a=rtpmap:111 opus/48000/2
>>>>>>    a=fmtp:111 useinbandfec=1; minptime=10
>>>>>>    a=rtpmap:126 telephone-event/8000
>>>>>>    a=rtpmap:106 CN/8000
>>>>>>    a=ptime:20
>>>>>>    a=fingerprint:sha-256
>>>>>> 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA
>>>>>>    a=rtcp-mux
>>>>>>    a=rtcp:19864 IN IP4 xxx.xxx.xxx.xxx
>>>>>>    a=ssrc:3914469578 cname:VRh1s3ZQ5XPj8UCb
>>>>>>    a=ssrc:3914469578 msid:tycfgLW7xb89okfsdKHP3gespdCXRxPb a0
>>>>>>    a=ssrc:3914469578 mslabel:tycfgLW7xb89okfsdKHP3gespdCXRxPb
>>>>>>    a=ssrc:3914469578 label:tycfgLW7xb89okfsdKHP3gespdCXRxPba0
>>>>>>    a=ice-ufrag:v9QogVGvZ8jGwYIi
>>>>>>    a=ice-pwd:FP9YIYEDMjNL4fNpftDlfaH5
>>>>>>    a=candidate:8233794353 1 udp 659136 xxx.xxx.xxx.xxx 19864 typ host
>>>>>> generation 0
>>>>>>
>>>>>>  ------------------------------------------------------------------------
>>>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>>>> timer
>>>>>> nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (28092)
>>>>>> nta.c:1348 set_timeout() nta: timer shortened to 500 ms
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_channel.c:3711 (
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx) Callstate Change EARLY -> ACTIVE
>>>>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210):
>>>>>> call state changed: received -> completed, sent answer
>>>>>> soa.c:1270 soa_get_local_sdp()
>>>>>> soa_get_local_sdp(static::0x7fe17c0bf4d0, [0x7fe1aa7b3c48],
>>>>>> [0x7fe1aa7b3c50], [(nil)]) called
>>>>>> soa.c:616 soa_get_params() soa_get_params(static::0x7fe17c0bf4d0,
>>>>>> ...) called
>>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state
>>>>>> 200 OK
>>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>>> entering
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:1061 Send
>>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] sofia.c:6627 Channel
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [completed][200]
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> EXECUTE sofia/internal/1000 at xxx.xxx.xxx.xxx
>>>>>> conference(991234-172.30.0.219 at test)
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10364 using
>>>>>> channel sound prefix: /usr/local/freeswitch/sounds/en/us/callie
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:8991 Raw Codec
>>>>>> Activation Success L16 at 48000hz 1 channel 20ms
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:9037 Raw Codec
>>>>>> Activation Success L16 at 16000hz 1 channel 20ms
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_codec.c:221
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx Push codec L16:100
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '0' to 'mute'
>>>>>> 2015-05-08 08:21:30.467711 [INFO] switch_ivr_async.c:212 Digit parser
>>>>>> mod_conference: Setting realm to 'conf'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding 0/conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010b8e0
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '*' to 'deaf mute'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding */conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010b910
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '9' to 'energy up'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding 9/conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010b940
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '8' to 'energy equ'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding 8/conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010b970
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '7' to 'energy dn'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding 7/conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010b9a0
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '3' to 'vol talk up'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding 3/conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010b9d0
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '2' to 'vol talk zero'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding 2/conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010ba00
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '1' to 'vol talk dn'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding 1/conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010ba30
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '6' to 'vol listen up'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding 6/conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010ba60
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '5' to 'vol listen zero'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding 5/conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010ba90
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '4' to 'vol listen dn'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding 4/conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010bac0
>>>>>> 2015-05-08 08:21:30.467711 [INFO] mod_conference.c:10967
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx binding '#' to 'hangup'
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_ivr_async.c:321 Digit
>>>>>> parser mod_conference: binding #/conf/0 callback: 0x7fe1a92764e0 data:
>>>>>> 0x7fe19010baf0
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] switch_core_session.c:912 Send
>>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:4765 Setup timer
>>>>>> soft success interval: 20  samples: 960
>>>>>> 2015-05-08 08:21:30.467711 [DEBUG] mod_conference.c:3043 Setup timer
>>>>>> success interval: 30  samples: 480
>>>>>> 2015-05-08 08:21:30.507713 [DEBUG] mod_local_stream.c:498 Opening
>>>>>> Stream [moh/16000] 16000hz
>>>>>> 2015-05-08 08:21:30.507713 [NOTICE] switch_core_io.c:1261 Activating
>>>>>> write resampler
>>>>>> tport.c:2773 tport_wakeup() tport_wakeup(0x7fe17c0d4ba0): events IN
>>>>>> tport.c:2864 tport_recv_event() tport_recv_event(0x7fe17c0d4ba0)
>>>>>> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fe17c0d4ba0) msg
>>>>>> 0x7fe17c0c6ab0 from (ws/82.166.84.247:53645) has 550 bytes, veclen =
>>>>>> 1
>>>>>> recv 550 bytes from ws/[82.166.84.247]:53645 at 08:21:30.662998:
>>>>>>
>>>>>>  ------------------------------------------------------------------------
>>>>>>    ACK sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0
>>>>>>    Via: SIP/2.0/WS
>>>>>> df7jal23ls0d.invalid;branch=z9hG4bK30nT8FwJ3EqSVIbdgFvT;rport
>>>>>>    From: "asdasda"<sip:1000 at xxx.xxx.xxx.xxx>;tag=THvNqRR9fmMzKb6JP9hp
>>>>>>    To: <sip:991234 at aaaa>;tag=73aKc8ZegaUHr
>>>>>>    Contact: "asdasda"<
>>>>>> sip:1000 at df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws
>>>>>> >;+g.oma.sip-im;language="en,fr"
>>>>>>    Call-ID: fa0bf4d8-2037-a467-61fd-4f78e9746fbe
>>>>>>    CSeq: 28092 ACK
>>>>>>    Content-Length: 0
>>>>>>    Max-Forwards: 70
>>>>>>    User-Agent: IM-client/OMA1.0 sipML5-v1.2015.03.18
>>>>>>    Organization: Doubango Telecom
>>>>>>
>>>>>>
>>>>>>  ------------------------------------------------------------------------
>>>>>> tport.c:3023 tport_deliver() tport_deliver(0x7fe17c0d4ba0): msg
>>>>>> 0x7fe17c0c6ab0 (550 bytes) from ws/82.166.84.247:53645/sip next=(nil)
>>>>>> nta.c:2880 agent_recv_request() nta: received ACK
>>>>>> sip:991234 at xxx.xxx.xxx.xxx:5060;transport=udp SIP/2.0 (CSeq 28092)
>>>>>> nta.c:3174 agent_check_request_via() nta: Via check:
>>>>>> received=82.166.84.247
>>>>>> nta.c:3019 agent_recv_request() nta: ACK (28092) is going to INVITE
>>>>>> (28092)
>>>>>> nua_session.c:2569 process_ack_or_cancel() nua:
>>>>>> process_ack_or_cancel: entering
>>>>>> soa.c:1214 soa_clear_remote_sdp()
>>>>>> soa_clear_remote_sdp(static::0x7fe17c0bf4d0) called
>>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_ack
>>>>>> 200 OK
>>>>>> nua_session.c:4139 signal_call_state_change() nua(0x7fe17c0cf210):
>>>>>> call state changed: completed -> ready
>>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_state
>>>>>> 200 OK
>>>>>> nua_stack.c:271 nua_stack_event() nua(0x7fe17c0cf210): event i_active
>>>>>> 200 Call active
>>>>>> nta.c:5744 incoming_free() nta: incoming_free(0x7fe17c024090)
>>>>>> tport.c:2296 tport_set_secondary_timer() tport(0x7fe17c0d4ba0): reset
>>>>>> timer
>>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>>> entering
>>>>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>>> entering
>>>>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>>> nua_stack.c:359 nua_application_event() nua: nua_application_event:
>>>>>> entering
>>>>>> 2015-05-08 08:21:30.647708 [DEBUG] switch_core_session.c:1061 Send
>>>>>> signal sofia/internal/1000 at xxx.xxx.xxx.xxx [BREAK]
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> 2015-05-08 08:21:30.667707 [DEBUG] sofia.c:6627 Channel
>>>>>> sofia/internal/1000 at xxx.xxx.xxx.xxx entering state [ready][200]
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
>>>>>> nta.c:1289 agent_timer() nta: timer not set
>>>>>> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
>>>>>> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
>>>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2924 Changing audio
>>>>>> DTLS state from HANDSHAKE to SETUP
>>>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2832 audio Fingerprint
>>>>>> Verified.
>>>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3374 Activating Audio
>>>>>> Secure RTP SEND
>>>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:3352 Activating Audio
>>>>>> Secure RTP RECV
>>>>>> 2015-05-08 08:22:02.047711 [INFO] switch_rtp.c:2872 Changing audio
>>>>>> DTLS state from SETUP to READY
>>>>>> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure
>>>>>> Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80
>>>>>> 2015-05-08 08:22:02.047711 [DEBUG] switch_core_sqldb.c:2599 Secure
>>>>>> Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80
>>>>>> 2015-05-08 08:22:02.087708 [DEBUG] switch_rtp.c:1937 rtcp_stats_init:
>>>>>> ssrc[-195423717] base_seq[29507]
>>>>>>
>>>>>> Notice that the INVITE was received at 08:21:30 while DTLS was READY
>>>>>> at 2015-05-08 08:22:02 which means that it took 32 seconds to voice. Again,
>>>>>> if I am not using TURN/STUN, the whole process is pretty quick (2 seconds).
>>>>>>
>>>>>> Also, notice:
>>>>>> 2015-05-08 08:21:48.307689 [NOTICE] switch_rtp.c:1133 Auto Changing
>>>>>> stun/rtp/dtls port from 82.166.84.247:63888 to 130.211.78.35:58209
>>>>>>
>>>>>> Which means that the media is relayed via the TURN server (TURN
>>>>>> server IP is 130.211.78.35)...
>>>>>>
>>>>>> Any idea?
>>>>>>
>>>>>> Thanks,
>>>>>> Adam
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>>
>>
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561
>> BB #79A8AFA2
>> @VMCibersys
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jungle.wang at huawei.com  Sat May  9 07:43:52 2015
From: jungle.wang at huawei.com (jungle)
Date: Fri, 8 May 2015 20:43:52 -0700 (MST)
Subject: [Freeswitch-users] How to send SIP re-invite to attendance client
 in FreeSwitch conference ?
Message-ID: <1431143032770-7596162.post@n2.nabble.com>

Hi all,   I have a trouble in FreeSwitch. The following is the issue
scenario:*After client A  join a conference via IVR, then the conference
need to send a SIP-reinvite  and carry some special parameter in SIP Contact
header to tell the conference type for terminal A.*In this case , 
FreeSwitch how to initiate such a re-invite message to the attendance
client?.Any problem-solving ideas recommendation?thanks



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-send-SIP-re-invite-to-attendance-client-in-FreeSwitch-conference-tp7596162.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From kamil.nigmatullin at gmail.com  Sat May  9 12:49:37 2015
From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin)
Date: Sat, 9 May 2015 14:49:37 +0600
Subject: [Freeswitch-users] FEC feature in mod_opus
In-Reply-To: <74103379-A16C-487D-9CEB-ABD039E2205B@jerris.com>
References: <2091369665.2474442.1429001528976.JavaMail.yahoo@mail.yahoo.com>
	<CA+Xtiee7K3mjCQVoxNWfQiunhr8zUsRmaSxLODSKJvAkzyDuWQ@mail.gmail.com>
	<74103379-A16C-487D-9CEB-ABD039E2205B@jerris.com>
Message-ID: <CA+Xtiec6Aj+ZgHq3Snpmv7uwTGzGpd5kJ9yLkG=nAB6h1n3scg@mail.gmail.com>

Yes, it is old but it is stable. A lot of things it does it does better
than any development version of 1.4. Thanks I will try.

2015-05-09 1:26 GMT+06:00 Michael Jerris <mike at jerris.com>:

> 1.2 is quite old, why would you be using it?  Have you tried on top of 1.4?
>
> On May 8, 2015, at 2:11 PM, Kamil Nigmatullin <kamil.nigmatullin at gmail.com>
> wrote:
>
> Hi Dragos,
> I tried to use your patch but have no luck with compiling it.
>
> I clonne stable 1.2 and then replace mod_opus.c with your code. Maybe do
> you know how to fix it?
>
> This is what I get:
> d_opus/mod_opus.c  -fPIC -DPIC -o .libs/mod_opus.o
> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function
> 'switch_opus_fmtp_parse':
> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:155:18: error:
> 'switch_codec_fmtp_t' has no member named 'stereo'
> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function
> 'switch_opus_decode':
> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:423:3: error:
> statement with no effect [-Werror=unused-value]
> cc1: all warnings being treated as errors
> make[4]: *** [mod_opus.lo] Error 1
> make[3]: *** [all] Error 1
> make[2]: *** [mod_opus-all] Error 1
> make[1]: *** [mod_opus] Error 2
> make: *** [mod_opus] Error 2
>
>
> 2015-04-14 14:52 GMT+06:00 Dragos Oancea <droancea at yahoo.com>:
>
>>
>> Hi Kamil,
>>
>> I've done a patch for Opus with FEC , but it has not been merged:
>>
>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/184/overview
>> I was inspired by how they've done it with SILK.
>>
>>
>> Regards,
>> Dragos Oancea
>>
>>
>>
>> On 2015-04-13 16:49, Kamil Nigmatullin wrote:
>> > Hello.
>> >
>> > Anybody from developers could please clarify if FreeSWITCH uses
>> > Forward Error Correction feature in opus codec? We actually are not a
>> > specialists in C and cannot fully understand how decode function works
>> > in 414 line. According to opus documentation decode function  is
>> > called with special flag if previous packet is lost in order to call
>> > fec feature. In mod_opus it is hard to say for me as i don't see i or
>> > i+1 packets. 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
>>
>>
>
>
> --
> Kamil Nigmatullin
> Tel: 77272323748
> mob: 7 (707) 2517003
> Skype: kamil.nigmatullin
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kamil Nigmatullin
Tel: 77272323748
mob: 7 (707) 2517003
Skype: kamil.nigmatullin
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From krice at freeswitch.org  Sat May  9 16:26:13 2015
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 9 May 2015 07:26:13 -0500
Subject: [Freeswitch-users] FEC feature in mod_opus
In-Reply-To: <CA+Xtiec6Aj+ZgHq3Snpmv7uwTGzGpd5kJ9yLkG=nAB6h1n3scg@mail.gmail.com>
References: <2091369665.2474442.1429001528976.JavaMail.yahoo@mail.yahoo.com>
	<CA+Xtiee7K3mjCQVoxNWfQiunhr8zUsRmaSxLODSKJvAkzyDuWQ@mail.gmail.com>
	<74103379-A16C-487D-9CEB-ABD039E2205B@jerris.com>
	<CA+Xtiec6Aj+ZgHq3Snpmv7uwTGzGpd5kJ9yLkG=nAB6h1n3scg@mail.gmail.com>
Message-ID: <52FE48A8-0573-4B59-BC86-B07B43FBE356@freeswitch.org>

If there are regressions in the 1.4 release versions you should make sure there have been jira tickets opened on them.

Sent from my iPhone

> On May 9, 2015, at 3:49 AM, Kamil Nigmatullin <kamil.nigmatullin at gmail.com> wrote:
> 
> Yes, it is old but it is stable. A lot of things it does it does better than any development version of 1.4. Thanks I will try.
> 
> 2015-05-09 1:26 GMT+06:00 Michael Jerris <mike at jerris.com>:
>> 1.2 is quite old, why would you be using it?  Have you tried on top of 1.4?
>> 
>>> On May 8, 2015, at 2:11 PM, Kamil Nigmatullin <kamil.nigmatullin at gmail.com> wrote:
>>> 
>>> Hi Dragos, 
>>> I tried to use your patch but have no luck with compiling it. 
>>> 
>>> I clonne stable 1.2 and then replace mod_opus.c with your code. Maybe do you know how to fix it?
>>> 
>>> This is what I get:
>>> d_opus/mod_opus.c  -fPIC -DPIC -o .libs/mod_opus.o
>>> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function 'switch_opus_fmtp_parse':
>>> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:155:18: error: 'switch_codec_fmtp_t' has no member named 'stereo'
>>> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function 'switch_opus_decode':
>>> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:423:3: error: statement with no effect [-Werror=unused-value]
>>> cc1: all warnings being treated as errors
>>> make[4]: *** [mod_opus.lo] Error 1
>>> make[3]: *** [all] Error 1
>>> make[2]: *** [mod_opus-all] Error 1
>>> make[1]: *** [mod_opus] Error 2
>>> make: *** [mod_opus] Error 2
>>> 
>>> 
>>> 2015-04-14 14:52 GMT+06:00 Dragos Oancea <droancea at yahoo.com>:
>>>> 
>>>> Hi Kamil,
>>>> 
>>>> I've done a patch for Opus with FEC , but it has not been merged:
>>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/184/overview
>>>> I was inspired by how they've done it with SILK.
>>>> 
>>>> 
>>>> Regards,
>>>> Dragos Oancea
>>>> 
>>>> 
>>>> 
>>>> On 2015-04-13 16:49, Kamil Nigmatullin wrote:
>>>> > Hello. 
>>>> > 
>>>> > Anybody from developers could please clarify if FreeSWITCH uses
>>>> > Forward Error Correction feature in opus codec? We actually are not a
>>>> > specialists in C and cannot fully understand how decode function works
>>>> > in 414 line. According to opus documentation decode function  is
>>>> > called with special flag if previous packet is lost in order to call
>>>> > fec feature. In mod_opus it is hard to say for me as i don't see i or
>>>> > i+1 packets. 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
>>> 
>>> 
>>> 
>>> -- 
>>> Kamil Nigmatullin
>>> Tel: 77272323748
>>> mob: 7 (707) 2517003
>>> Skype: kamil.nigmatullin
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> Kamil Nigmatullin
> Tel: 77272323748
> mob: 7 (707) 2517003
> Skype: kamil.nigmatullin
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From fernando at softov.com.br  Sun May 10 01:28:55 2015
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Sat, 9 May 2015 17:28:55 -0400
Subject: [Freeswitch-users] Connect two freeswitch boxes with gateway
Message-ID: <CAKy26iNmRABxf0ZVF8FwuNEShZgeyAkNtMRXYspZm5AfuxqTPQ@mail.gmail.com>

Hi, i'm trying to do this

B have a gateway registered with user 1001 in A
and of course, A have a user 1001 registered

B ----> [gateway] ------> A (1001)

So, i'm trying to bridge 1001 when i call (1xxx)

A ----> [bridge user/1001] ------> B

and have this dial-string for user 1001
{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}

In FS B i receive the call, but i want to receive the real destination
number, like 1002 or 1004

Is there a way to bridge to user, and send the destination number?
So, in box B, i can transfer to correct user.


-- 
Luiz Fernando Softov
http://www.softov.com.br
fernando at softov.com.br


From anthony.minessale at gmail.com  Sun May 10 02:46:39 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 9 May 2015 17:46:39 -0500
Subject: [Freeswitch-users] Clock precision and raw audio recording
In-Reply-To: <CACOYK=i5z_9=ssMBWQfw_e60q8fe2a-8hmMu_8vWEBk-wWtE1A@mail.gmail.com>
References: <CACOYK=i5z_9=ssMBWQfw_e60q8fe2a-8hmMu_8vWEBk-wWtE1A@mail.gmail.com>
Message-ID: <CAKbxfG8ZZtOoCEaA5ffe0tUimQ8BQwoXYAVzHtAKq-4Hx305yA@mail.gmail.com>

Guess:

Did you try using it with the jitter buffer enabled?
The raw recording cannot pre-buffer the audio so its more likely jitter may
be rendered into the file.
Sometimes the gaps that are experienced are just from late packets.

The WAV recordings have a more complicated process for buffering and mixing
the file.



On Fri, May 8, 2015 at 5:12 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> I'm observing an effect which needs explanation. Comments from the
> core developers will be appreciated. The effect was tested with
> versions 1.4.18 and today's master, on 64-bit Debian 7 and Ubuntu
> 14.04. All test calls were in PCMU or PCMA.
>
> My customer requested me to build a server for call quality assurance
> for their telephony system. I installed FreeSWITCH and set the
> following in the public dialplan to record the incoming audio:
>
>     <extension name="record">
>       <condition field="destination_number" expression="^record_(.+)$">
>         <action application="jitterbuffer" data="60:200:20"/>
>         <action application="set" data="RECORD_READ_ONLY=true"/>
>         <action application="set" data="send_silence_when_idle=400"/>
>         <action application="set" data="record_waste_resources=true"/>
>         <action application="answer"/>
>         <action application="record_session" data="/var/tmp/record_$1"/>
>         <action application="playback" data="silence_stream://-1"/>
>       </condition>
>     </extension>
>
> The first try was with a DigitalOcean (KVM) virtual machine. I started
> recording *.wav files, and sometimes there were skipped frames: 1-2
> skipped frames in a 2-minute call, one in every 10-15 calls.
>
> Then I changed the configuration to write raw audio files (removed the
> .wav extension from the record_session argument). As a result, the
> recorded input audio was quite bad: lost frames every few seconds in
> every call.
>
> Then I made test calls within the server itself, by originating a call
> to its public profile and playing the test WAV audio:
>
> fs_cli  -x 'originate sofia/external/record_03 at 111.222.222.111:5080
> &playback(/var/tmp/ITU-T_P_50_BRITISH_ENGLISH.wav)'
>
> The resulting input raw audio was also choppy. The same result was on
> another VM on a different physical server at DigitalOcean.
>
> Then I made the same self-call tests on a Xen VM and on a baremetal
> ARM server, and there the recorded audio was of perfect quality.
>
> Self-calls with recording into WAV files produced audio of perfect quality.
>
> Setting RECORD_USE_THREAD=false did not change the effect.
>
> Example of choppy received audio, converted from PCMU to WAV for
> convenience:
> http://www.k-open.com/s/record_04-in.wav
> The source audio:
> http://murmur.voxserv.ch/media/ITU-T_P_50_BRITISH_ENGLISH.wav
>
> So, it looks like the clock that is available at KVM is not precise
> enough, but that's not my question.
>
> QUESTION: why is raw recording so much more sensitive to the clock
> precision?
>
> thanks,
> stanislav
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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 william.king at quentustech.com  Sun May 10 05:40:20 2015
From: william.king at quentustech.com (William King)
Date: Sat, 09 May 2015 18:40:20 -0700
Subject: [Freeswitch-users] mod_hiredis
Message-ID: <554EB704.5090605@quentustech.com>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA512

I'm working on an update Redis integration module that will use the C
library hiredis:
http://redis.io/clients#c
https://github.com/redis/hiredis

I've pushed an alpha version of the module to a branch here:
https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits?until=
refs%2Fheads%2Fmod_hiredis

The current module has a dialplan app and an api for 'hiredis_raw'
which allows any single line Redis command, and executes it in a
blocking manner, then supports returning string and integer responses.

If anyone on this list has any use cases for FreeSWITCH+Redis, please
reply to this thread. Currently the two main use cases are:
1. Call per second limits
2. Concurrent call limits

Possible additional functionality:
1. Support for fail-over connections
2. Asynchronous commands(is there a use case for this?)
- -- 
William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com
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From ssinyagin at gmail.com  Sun May 10 08:16:45 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 10 May 2015 06:16:45 +0200
Subject: [Freeswitch-users] Clock precision and raw audio recording
In-Reply-To: <CAKbxfG8ZZtOoCEaA5ffe0tUimQ8BQwoXYAVzHtAKq-4Hx305yA@mail.gmail.com>
References: <CACOYK=i5z_9=ssMBWQfw_e60q8fe2a-8hmMu_8vWEBk-wWtE1A@mail.gmail.com>
	<CAKbxfG8ZZtOoCEaA5ffe0tUimQ8BQwoXYAVzHtAKq-4Hx305yA@mail.gmail.com>
Message-ID: <CACOYK=iycM38daVjXJXbpoYo3au3diuza1cZC=zMUMCjwwS5qA@mail.gmail.com>

Jitterbuffer was always enabled on the receiving side in all tests, like in
the piece of  config in my email. I can try disabling it.

I can also try giving raw audio to playback, instead of 16-bit wav.

But basically, it's not a bug, but rather a feature allowing to check the
clock precision. But it will be great if we understand its nature :)



On May 10, 2015 12:47 AM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:

> Guess:
>
> Did you try using it with the jitter buffer enabled?
> The raw recording cannot pre-buffer the audio so its more likely jitter
> may be rendered into the file.
> Sometimes the gaps that are experienced are just from late packets.
>
> The WAV recordings have a more complicated process for buffering and
> mixing the file.
>
>
>
> On Fri, May 8, 2015 at 5:12 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> I'm observing an effect which needs explanation. Comments from the
>> core developers will be appreciated. The effect was tested with
>> versions 1.4.18 and today's master, on 64-bit Debian 7 and Ubuntu
>> 14.04. All test calls were in PCMU or PCMA.
>>
>> My customer requested me to build a server for call quality assurance
>> for their telephony system. I installed FreeSWITCH and set the
>> following in the public dialplan to record the incoming audio:
>>
>>     <extension name="record">
>>       <condition field="destination_number" expression="^record_(.+)$">
>>         <action application="jitterbuffer" data="60:200:20"/>
>>         <action application="set" data="RECORD_READ_ONLY=true"/>
>>         <action application="set" data="send_silence_when_idle=400"/>
>>         <action application="set" data="record_waste_resources=true"/>
>>         <action application="answer"/>
>>         <action application="record_session" data="/var/tmp/record_$1"/>
>>         <action application="playback" data="silence_stream://-1"/>
>>       </condition>
>>     </extension>
>>
>> The first try was with a DigitalOcean (KVM) virtual machine. I started
>> recording *.wav files, and sometimes there were skipped frames: 1-2
>> skipped frames in a 2-minute call, one in every 10-15 calls.
>>
>> Then I changed the configuration to write raw audio files (removed the
>> .wav extension from the record_session argument). As a result, the
>> recorded input audio was quite bad: lost frames every few seconds in
>> every call.
>>
>> Then I made test calls within the server itself, by originating a call
>> to its public profile and playing the test WAV audio:
>>
>> fs_cli  -x 'originate sofia/external/record_03 at 111.222.222.111:5080
>> &playback(/var/tmp/ITU-T_P_50_BRITISH_ENGLISH.wav)'
>>
>> The resulting input raw audio was also choppy. The same result was on
>> another VM on a different physical server at DigitalOcean.
>>
>> Then I made the same self-call tests on a Xen VM and on a baremetal
>> ARM server, and there the recorded audio was of perfect quality.
>>
>> Self-calls with recording into WAV files produced audio of perfect
>> quality.
>>
>> Setting RECORD_USE_THREAD=false did not change the effect.
>>
>> Example of choppy received audio, converted from PCMU to WAV for
>> convenience:
>> http://www.k-open.com/s/record_04-in.wav
>> The source audio:
>> http://murmur.voxserv.ch/media/ITU-T_P_50_BRITISH_ENGLISH.wav
>>
>> So, it looks like the clock that is available at KVM is not precise
>> enough, but that's not my question.
>>
>> QUESTION: why is raw recording so much more sensitive to the clock
>> precision?
>>
>> thanks,
>> stanislav
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From freeswitch at peely.com  Sun May 10 14:54:37 2015
From: freeswitch at peely.com (peely)
Date: Sun, 10 May 2015 03:54:37 -0700 (MST)
Subject: [Freeswitch-users] mod_sangoma_codec in debian repo?
Message-ID: <1431255277628-7596164.post@n2.nabble.com>

Hi,

I'm testing out the possibility of moving to the Debian repo (we actually
use Ubuntu) and have installed FreeSWITCH from the wheezy version. There's
no sight however of mod_sangoma_codec which we use for our G.723 and G.729
interconnects.

I can see other libraries relating to Sangoma stuff but I can't see anything
which goes as far as to install mod_sangoma_codec.

If I want to continue using Sangoma with FreeSWITCH go I have to continue
building from source or has something changed that means I use Sangoma
transcoding through a different module?



Thanks,



Neil.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-sangoma-codec-in-debian-repo-tp7596164.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ssinyagin at gmail.com  Sun May 10 18:42:12 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 10 May 2015 16:42:12 +0200
Subject: [Freeswitch-users] Clock precision and raw audio recording
In-Reply-To: <CACOYK=i5z_9=ssMBWQfw_e60q8fe2a-8hmMu_8vWEBk-wWtE1A@mail.gmail.com>
References: <CACOYK=i5z_9=ssMBWQfw_e60q8fe2a-8hmMu_8vWEBk-wWtE1A@mail.gmail.com>
Message-ID: <CACOYK=j21U+d=PmUK=7etaUt6DcX_aJOxMBNedKyTkVUnDKhCA@mail.gmail.com>

I've run it on a different Xen server which is more powerful and more
busy, and got also the same chopping audio in native recording.



On Sat, May 9, 2015 at 12:12 AM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> I'm observing an effect which needs explanation. Comments from the
> core developers will be appreciated. The effect was tested with
> versions 1.4.18 and today's master, on 64-bit Debian 7 and Ubuntu
> 14.04. All test calls were in PCMU or PCMA.
>
> My customer requested me to build a server for call quality assurance
> for their telephony system. I installed FreeSWITCH and set the
> following in the public dialplan to record the incoming audio:
>
>     <extension name="record">
>       <condition field="destination_number" expression="^record_(.+)$">
>         <action application="jitterbuffer" data="60:200:20"/>
>         <action application="set" data="RECORD_READ_ONLY=true"/>
>         <action application="set" data="send_silence_when_idle=400"/>
>         <action application="set" data="record_waste_resources=true"/>
>         <action application="answer"/>
>         <action application="record_session" data="/var/tmp/record_$1"/>
>         <action application="playback" data="silence_stream://-1"/>
>       </condition>
>     </extension>
>
> The first try was with a DigitalOcean (KVM) virtual machine. I started
> recording *.wav files, and sometimes there were skipped frames: 1-2
> skipped frames in a 2-minute call, one in every 10-15 calls.
>
> Then I changed the configuration to write raw audio files (removed the
> .wav extension from the record_session argument). As a result, the
> recorded input audio was quite bad: lost frames every few seconds in
> every call.
>
> Then I made test calls within the server itself, by originating a call
> to its public profile and playing the test WAV audio:
>
> fs_cli  -x 'originate sofia/external/record_03 at 111.222.222.111:5080
> &playback(/var/tmp/ITU-T_P_50_BRITISH_ENGLISH.wav)'
>
> The resulting input raw audio was also choppy. The same result was on
> another VM on a different physical server at DigitalOcean.
>
> Then I made the same self-call tests on a Xen VM and on a baremetal
> ARM server, and there the recorded audio was of perfect quality.
>
> Self-calls with recording into WAV files produced audio of perfect quality.
>
> Setting RECORD_USE_THREAD=false did not change the effect.
>
> Example of choppy received audio, converted from PCMU to WAV for convenience:
> http://www.k-open.com/s/record_04-in.wav
> The source audio:
> http://murmur.voxserv.ch/media/ITU-T_P_50_BRITISH_ENGLISH.wav
>
> So, it looks like the clock that is available at KVM is not precise
> enough, but that's not my question.
>
> QUESTION: why is raw recording so much more sensitive to the clock precision?
>
> thanks,
> stanislav


From s.safarov at gmail.com  Sun May 10 19:49:28 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 10 May 2015 18:49:28 +0300
Subject: [Freeswitch-users] Connect two freeswitch boxes with gateway
In-Reply-To: <CAKy26iNmRABxf0ZVF8FwuNEShZgeyAkNtMRXYspZm5AfuxqTPQ@mail.gmail.com>
References: <CAKy26iNmRABxf0ZVF8FwuNEShZgeyAkNtMRXYspZm5AfuxqTPQ@mail.gmail.com>
Message-ID: <CAHtxdDeg6-qyN_HbCuQcSgrAkZumD-BT3ti9uiVsiFB2ja2JJQ@mail.gmail.com>

Try variable use sip_invite_to_uri
Example
<action application="bridge" data="{sip_invite_to_uri=<sip:${
destination_number}@mydomain.org}user/1000 at mydomain.org"/>

On Sun, May 10, 2015 at 12:28 AM, Luiz Fernando Softov <
fernando at softov.com.br> wrote:

> Hi, i'm trying to do this
>
> B have a gateway registered with user 1001 in A
> and of course, A have a user 1001 registered
>
> B ----> [gateway] ------> A (1001)
>
> So, i'm trying to bridge 1001 when i call (1xxx)
>
> A ----> [bridge user/1001] ------> B
>
> and have this dial-string for user 1001
> {presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}
>
> In FS B i receive the call, but i want to receive the real destination
> number, like 1002 or 1004
>
> Is there a way to bridge to user, and send the destination number?
> So, in box B, i can transfer to correct user.
>
>
> --
> Luiz Fernando Softov
> http://www.softov.com.br
> fernando at softov.com.br
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From william.king at quentustech.com  Sun May 10 20:48:35 2015
From: william.king at quentustech.com (William King)
Date: Sun, 10 May 2015 09:48:35 -0700
Subject: [Freeswitch-users] mod_sangoma_codec in debian repo?
In-Reply-To: <1431255277628-7596164.post@n2.nabble.com>
References: <1431255277628-7596164.post@n2.nabble.com>
Message-ID: <554F8BE3.3050308@quentustech.com>

I'm working on adding that module to the build. I don't have one of the
transcoding cards, so I don't have a way to test. Feel free to file a
Jira feature request for the module and Jira will notify you when it's
ready to test.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 5/10/15 3:54 AM, peely wrote:
> Hi,
> 
> I'm testing out the possibility of moving to the Debian repo (we actually
> use Ubuntu) and have installed FreeSWITCH from the wheezy version. There's
> no sight however of mod_sangoma_codec which we use for our G.723 and G.729
> interconnects.
> 
> I can see other libraries relating to Sangoma stuff but I can't see anything
> which goes as far as to install mod_sangoma_codec.
> 
> If I want to continue using Sangoma with FreeSWITCH go I have to continue
> building from source or has something changed that means I use Sangoma
> transcoding through a different module?
> 
> 
> 
> Thanks,
> 
> 
> 
> Neil.
> 
> 
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-sangoma-codec-in-debian-repo-tp7596164.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
> 


From fernando at softov.com.br  Mon May 11 00:26:00 2015
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Sun, 10 May 2015 16:26:00 -0400
Subject: [Freeswitch-users] Connect two freeswitch boxes with gateway
In-Reply-To: <CAHtxdDeg6-qyN_HbCuQcSgrAkZumD-BT3ti9uiVsiFB2ja2JJQ@mail.gmail.com>
References: <CAKy26iNmRABxf0ZVF8FwuNEShZgeyAkNtMRXYspZm5AfuxqTPQ@mail.gmail.com>
	<CAHtxdDeg6-qyN_HbCuQcSgrAkZumD-BT3ti9uiVsiFB2ja2JJQ@mail.gmail.com>
Message-ID: <CAKy26iMe+F=jz=ei2r3UfLn4CwZQpH2J=vpXPExoNghq=jDp8Q@mail.gmail.com>

Thank you,

I found this

https://wiki.freeswitch.org/wiki/Sofia-SIP#Modifying_the_To:_header

* All inbound SIP calls will install any X- headers into local variables.
* This means you can easily bridge any X- header from one FreeSWITCH
instance to another.
* To access the header above on a 2nd box, use the channel variable
${sip_h_X-Answer}
* It is important to note that the syntax ${sip_h_customer-header}
can't be used to retrieve any custom header not starting with X-.
* It is because Sofia only reads and puts into variables custom
headers starting with X-.


But, i will try sip_invite_to_uri, and send reponse here







2015-05-10 11:49 GMT-04:00 Sergey Safarov <s.safarov at gmail.com>:
> Try variable use sip_invite_to_uri
> Example
> <action application="bridge"
> data="{sip_invite_to_uri=<sip:${destination_number}@mydomain.org}user/1000 at mydomain.org"/>
>
> On Sun, May 10, 2015 at 12:28 AM, Luiz Fernando Softov
> <fernando at softov.com.br> wrote:
>>
>> Hi, i'm trying to do this
>>
>> B have a gateway registered with user 1001 in A
>> and of course, A have a user 1001 registered
>>
>> B ----> [gateway] ------> A (1001)
>>
>> So, i'm trying to bridge 1001 when i call (1xxx)
>>
>> A ----> [bridge user/1001] ------> B
>>
>> and have this dial-string for user 1001
>>
>> {presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}
>>
>> In FS B i receive the call, but i want to receive the real destination
>> number, like 1002 or 1004
>>
>> Is there a way to bridge to user, and send the destination number?
>> So, in box B, i can transfer to correct user.
>>
>>
>> --
>> Luiz Fernando Softov
>> http://www.softov.com.br
>> fernando at softov.com.br
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Luiz Fernando Softov
http://www.softov.com.br
fernando at softov.com.br


From ssinyagin at gmail.com  Mon May 11 00:54:11 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 10 May 2015 22:54:11 +0200
Subject: [Freeswitch-users] Clock precision and raw audio recording
In-Reply-To: <CACOYK=j21U+d=PmUK=7etaUt6DcX_aJOxMBNedKyTkVUnDKhCA@mail.gmail.com>
References: <CACOYK=i5z_9=ssMBWQfw_e60q8fe2a-8hmMu_8vWEBk-wWtE1A@mail.gmail.com>
	<CACOYK=j21U+d=PmUK=7etaUt6DcX_aJOxMBNedKyTkVUnDKhCA@mail.gmail.com>
Message-ID: <CACOYK=jKzo6eP3KjNgpbNF+VFUgxGgUq+Sd1Q8wtOZ0apuwP1w@mail.gmail.com>

the problem disappears after rolling back to December code, so I
opened a Jira ticket:
https://freeswitch.org/jira/browse/FS-7541



On Sun, May 10, 2015 at 4:42 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> I've run it on a different Xen server which is more powerful and more
> busy, and got also the same chopping audio in native recording.
>
>
>
> On Sat, May 9, 2015 at 12:12 AM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>> I'm observing an effect which needs explanation. Comments from the
>> core developers will be appreciated. The effect was tested with
>> versions 1.4.18 and today's master, on 64-bit Debian 7 and Ubuntu
>> 14.04. All test calls were in PCMU or PCMA.
>>
>> My customer requested me to build a server for call quality assurance
>> for their telephony system. I installed FreeSWITCH and set the
>> following in the public dialplan to record the incoming audio:
>>
>>     <extension name="record">
>>       <condition field="destination_number" expression="^record_(.+)$">
>>         <action application="jitterbuffer" data="60:200:20"/>
>>         <action application="set" data="RECORD_READ_ONLY=true"/>
>>         <action application="set" data="send_silence_when_idle=400"/>
>>         <action application="set" data="record_waste_resources=true"/>
>>         <action application="answer"/>
>>         <action application="record_session" data="/var/tmp/record_$1"/>
>>         <action application="playback" data="silence_stream://-1"/>
>>       </condition>
>>     </extension>
>>
>> The first try was with a DigitalOcean (KVM) virtual machine. I started
>> recording *.wav files, and sometimes there were skipped frames: 1-2
>> skipped frames in a 2-minute call, one in every 10-15 calls.
>>
>> Then I changed the configuration to write raw audio files (removed the
>> .wav extension from the record_session argument). As a result, the
>> recorded input audio was quite bad: lost frames every few seconds in
>> every call.
>>
>> Then I made test calls within the server itself, by originating a call
>> to its public profile and playing the test WAV audio:
>>
>> fs_cli  -x 'originate sofia/external/record_03 at 111.222.222.111:5080
>> &playback(/var/tmp/ITU-T_P_50_BRITISH_ENGLISH.wav)'
>>
>> The resulting input raw audio was also choppy. The same result was on
>> another VM on a different physical server at DigitalOcean.
>>
>> Then I made the same self-call tests on a Xen VM and on a baremetal
>> ARM server, and there the recorded audio was of perfect quality.
>>
>> Self-calls with recording into WAV files produced audio of perfect quality.
>>
>> Setting RECORD_USE_THREAD=false did not change the effect.
>>
>> Example of choppy received audio, converted from PCMU to WAV for convenience:
>> http://www.k-open.com/s/record_04-in.wav
>> The source audio:
>> http://murmur.voxserv.ch/media/ITU-T_P_50_BRITISH_ENGLISH.wav
>>
>> So, it looks like the clock that is available at KVM is not precise
>> enough, but that's not my question.
>>
>> QUESTION: why is raw recording so much more sensitive to the clock precision?
>>
>> thanks,
>> stanislav


From anthony.minessale at gmail.com  Mon May 11 02:29:42 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 10 May 2015 17:29:42 -0500
Subject: [Freeswitch-users] Clock precision and raw audio recording
In-Reply-To: <CACOYK=jKzo6eP3KjNgpbNF+VFUgxGgUq+Sd1Q8wtOZ0apuwP1w@mail.gmail.com>
References: <CACOYK=i5z_9=ssMBWQfw_e60q8fe2a-8hmMu_8vWEBk-wWtE1A@mail.gmail.com>
	<CACOYK=j21U+d=PmUK=7etaUt6DcX_aJOxMBNedKyTkVUnDKhCA@mail.gmail.com>
	<CACOYK=jKzo6eP3KjNgpbNF+VFUgxGgUq+Sd1Q8wtOZ0apuwP1w@mail.gmail.com>
Message-ID: <CAKbxfG8465HvCKdw+ednOCMqes4qcqi=hzFAbYxPs7wOc_QJjw@mail.gmail.com>

You'll have to do a git bisect if you propose another version works better.



On Sunday, May 10, 2015, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> the problem disappears after rolling back to December code, so I
> opened a Jira ticket:
> https://freeswitch.org/jira/browse/FS-7541
>
>
>
> On Sun, May 10, 2015 at 4:42 PM, Stanislav Sinyagin <ssinyagin at gmail.com
> <javascript:;>> wrote:
> > I've run it on a different Xen server which is more powerful and more
> > busy, and got also the same chopping audio in native recording.
> >
> >
> >
> > On Sat, May 9, 2015 at 12:12 AM, Stanislav Sinyagin <ssinyagin at gmail.com
> <javascript:;>> wrote:
> >> I'm observing an effect which needs explanation. Comments from the
> >> core developers will be appreciated. The effect was tested with
> >> versions 1.4.18 and today's master, on 64-bit Debian 7 and Ubuntu
> >> 14.04. All test calls were in PCMU or PCMA.
> >>
> >> My customer requested me to build a server for call quality assurance
> >> for their telephony system. I installed FreeSWITCH and set the
> >> following in the public dialplan to record the incoming audio:
> >>
> >>     <extension name="record">
> >>       <condition field="destination_number" expression="^record_(.+)$">
> >>         <action application="jitterbuffer" data="60:200:20"/>
> >>         <action application="set" data="RECORD_READ_ONLY=true"/>
> >>         <action application="set" data="send_silence_when_idle=400"/>
> >>         <action application="set" data="record_waste_resources=true"/>
> >>         <action application="answer"/>
> >>         <action application="record_session" data="/var/tmp/record_$1"/>
> >>         <action application="playback" data="silence_stream://-1"/>
> >>       </condition>
> >>     </extension>
> >>
> >> The first try was with a DigitalOcean (KVM) virtual machine. I started
> >> recording *.wav files, and sometimes there were skipped frames: 1-2
> >> skipped frames in a 2-minute call, one in every 10-15 calls.
> >>
> >> Then I changed the configuration to write raw audio files (removed the
> >> .wav extension from the record_session argument). As a result, the
> >> recorded input audio was quite bad: lost frames every few seconds in
> >> every call.
> >>
> >> Then I made test calls within the server itself, by originating a call
> >> to its public profile and playing the test WAV audio:
> >>
> >> fs_cli  -x 'originate sofia/external/record_03 at 111.222.222.111:5080
> >> &playback(/var/tmp/ITU-T_P_50_BRITISH_ENGLISH.wav)'
> >>
> >> The resulting input raw audio was also choppy. The same result was on
> >> another VM on a different physical server at DigitalOcean.
> >>
> >> Then I made the same self-call tests on a Xen VM and on a baremetal
> >> ARM server, and there the recorded audio was of perfect quality.
> >>
> >> Self-calls with recording into WAV files produced audio of perfect
> quality.
> >>
> >> Setting RECORD_USE_THREAD=false did not change the effect.
> >>
> >> Example of choppy received audio, converted from PCMU to WAV for
> convenience:
> >> http://www.k-open.com/s/record_04-in.wav
> >> The source audio:
> >> http://murmur.voxserv.ch/media/ITU-T_P_50_BRITISH_ENGLISH.wav
> >>
> >> So, it looks like the clock that is available at KVM is not precise
> >> enough, but that's not my question.
> >>
> >> QUESTION: why is raw recording so much more sensitive to the clock
> precision?
> >>
> >> thanks,
> >> stanislav
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From mike at jerris.com  Mon May 11 06:43:12 2015
From: mike at jerris.com (Michael Jerris)
Date: Sun, 10 May 2015 22:43:12 -0400
Subject: [Freeswitch-users] Trickle ICE with SIP
In-Reply-To: <CADNqFCkKLqRQVg11aGof3WAUzVpnMZryiejBB6mWoZTc2RqAOw@mail.gmail.com>
References: <CADNqFCmOfu_Nq56N9rc6H2DD=9OG7=MTqB1D4dT+kJ9GFcaBSA@mail.gmail.com>
	<C56F753C-D2E5-4072-BDC2-43BCE6E367AB@jerris.com>
	<CADNqFCkDy2OtzeTqF_mDzW4hA75qambrEGxs5JpwL9QwYseh9g@mail.gmail.com>
	<C70FE012-59AB-4022-A72A-14C110A03571@jerris.com>
	<CADNqFCn1giYz0N9TSx4gKvwqid-0dJh7C3jW1O8KDbhfktJ8XA@mail.gmail.com>
	<4D2BB9C4-50F6-4D72-A022-24D25A952633@jerris.com>
	<CADNqFCnR22Ptwb_e8M3qmOgQEjH4CV2j_j=wZx2xcuXGPs5pKQ@mail.gmail.com>
	<D15E3188-08F7-45A4-82D5-3044BB2E07BB@jerris.com>
	<CADNqFCkKLqRQVg11aGof3WAUzVpnMZryiejBB6mWoZTc2RqAOw@mail.gmail.com>
Message-ID: <CB3E7087-A2EC-4F2D-B4A6-F0D1E3D1570C@jerris.com>

Not currently, but its a feature we are interested in maybe supporting when ortc comes out with revised tricke standards that can more usefully allow handoff.

> On May 8, 2015, at 7:26 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:
> 
> Got it. Any way to do trickle ice with Freeswitch?
> 
> On 8 May 2015 at 20:51, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> The TCP case is the only case, and I had not thought of it.  Not sure it ever really makes sense to do it if we could add TCP \support, but we don't have that at the moment.
> 
>> On May 8, 2015, at 1:19 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
>> 
>> Can you clarify? Do you think there is no reason to use TURN in this case?
>> 
>> On 8 May 2015 at 20:06, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> I had not thought of tcp.  
>> 
>>> On May 8, 2015, at 1:04 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
>>> 
>>> Can you explain how can I enable clients with UDP blocked if I don't use TURN? RTP ports on Freeswitch are UDP.. I haven't tried Verto yet, actually we are developing a mobile app, in this case it's easy for us to use something like pjsip for SIP signalling..
>>> 
>>> On 8 May 2015 at 20:00, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>> okay.. so to confirm, there is no reason to ever be using turn, so we only are dealing w/ 2-4 seconds.  What results do you get using sip.js and what results do you get using verto.  I know there is some overhead in the browser and there is nothing we can do to get around that issue.
>>> 
>>>> On May 8, 2015, at 11:43 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
>>>> 
>>>> Well, it can get up to 30 seconds when using TURN, in other cases it's usually 2-4 seconds.
>>>> 
>>>> On 8 May 2015 at 18:35, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>>> >From your other post you said thats only when using turn?
>>>> 
>>>> > On May 8, 2015, at 7:36 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com <mailto:adam.ben.ayoun1 at gmail.com>> wrote:
>>>> >
>>>> > Hi,
>>>> >
>>>> > We are currently using Freeswitch for voice conferencing, we use SIP for signalling. Call setup times are really slow at times (3sec-30sec), can we somehow do Trickle ICE with Freeswitch?
>>>> >
>>>> > Thanks,
>>>> > Adam
>>>> 
>>> 

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From mike at jerris.com  Mon May 11 06:57:53 2015
From: mike at jerris.com (Michael Jerris)
Date: Sun, 10 May 2015 22:57:53 -0400
Subject: [Freeswitch-users] FEC feature in mod_opus
In-Reply-To: <52FE48A8-0573-4B59-BC86-B07B43FBE356@freeswitch.org>
References: <2091369665.2474442.1429001528976.JavaMail.yahoo@mail.yahoo.com>
	<CA+Xtiee7K3mjCQVoxNWfQiunhr8zUsRmaSxLODSKJvAkzyDuWQ@mail.gmail.com>
	<74103379-A16C-487D-9CEB-ABD039E2205B@jerris.com>
	<CA+Xtiec6Aj+ZgHq3Snpmv7uwTGzGpd5kJ9yLkG=nAB6h1n3scg@mail.gmail.com>
	<52FE48A8-0573-4B59-BC86-B07B43FBE356@freeswitch.org>
Message-ID: <EFFF19A0-96BE-4ED4-9531-12E0D82EFF7F@jerris.com>

We no longer label branches as "stable" like this.  !.4 is the current maintained release, and 1.2 no longer gets even security patches.  If there are any issues in 1.4, they need to be filed in jira ASAP.


> On May 9, 2015, at 8:26 AM, Ken Rice <krice at freeswitch.org> wrote:
> 
> If there are regressions in the 1.4 release versions you should make sure there have been jira tickets opened on them.
> 
> Sent from my iPhone
> 
> On May 9, 2015, at 3:49 AM, Kamil Nigmatullin <kamil.nigmatullin at gmail.com <mailto:kamil.nigmatullin at gmail.com>> wrote:
> 
>> Yes, it is old but it is stable. A lot of things it does it does better than any development version of 1.4. Thanks I will try.
>> 
>> 2015-05-09 1:26 GMT+06:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
>> 1.2 is quite old, why would you be using it?  Have you tried on top of 1.4?
>> 
>>> On May 8, 2015, at 2:11 PM, Kamil Nigmatullin <kamil.nigmatullin at gmail.com <mailto:kamil.nigmatullin at gmail.com>> wrote:
>>> 
>>> Hi Dragos, 
>>> I tried to use your patch but have no luck with compiling it. 
>>> 
>>> I clonne stable 1.2 and then replace mod_opus.c with your code. Maybe do you know how to fix it?
>>> 
>>> This is what I get:
>>> d_opus/mod_opus.c  -fPIC -DPIC -o .libs/mod_opus.o
>>> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function 'switch_opus_fmtp_parse':
>>> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:155:18: error: 'switch_codec_fmtp_t' has no member named 'stereo'
>>> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c: In function 'switch_opus_decode':
>>> /usr/src/freeswitch/src/mod/codecs/mod_opus/mod_opus.c:423:3: error: statement with no effect [-Werror=unused-value]
>>> cc1: all warnings being treated as errors
>>> make[4]: *** [mod_opus.lo] Error 1
>>> make[3]: *** [all] Error 1
>>> make[2]: *** [mod_opus-all] Error 1
>>> make[1]: *** [mod_opus] Error 2
>>> make: *** [mod_opus] Error 2
>>> 
>>> 
>>> 2015-04-14 14:52 GMT+06:00 Dragos Oancea <droancea at yahoo.com <mailto:droancea at yahoo.com>>:
>>> 
>>> Hi Kamil,
>>> 
>>> I've done a patch for Opus with FEC , but it has not been merged:
>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/184/overview <https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/184/overview>
>>> I was inspired by how they've done it with SILK.
>>> 
>>> 
>>> Regards,
>>> Dragos Oancea
>>> 
>>> 
>>> 
>>> On 2015-04-13 16:49, Kamil Nigmatullin wrote:
>>> > Hello. 
>>> > 
>>> > Anybody from developers could please clarify if FreeSWITCH uses
>>> > Forward Error Correction feature in opus codec? We actually are not a
>>> > specialists in C and cannot fully understand how decode function works
>>> > in 414 line. According to opus documentation decode function  is
>>> > called with special flag if previous packet is lost in order to call
>>> > fec feature. In mod_opus it is hard to say for me as i don't see i or
>>> > i+1 packets. Thanks    
>>> > _________________________________________________________________________

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From steveayre at gmail.com  Mon May 11 11:48:40 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 May 2015 08:48:40 +0100
Subject: [Freeswitch-users] need SIP IP / RTP recommendation
In-Reply-To: <1DD529F81979496BADBE1B52EB5D334B@bob>
References: <1DD529F81979496BADBE1B52EB5D334B@bob>
Message-ID: <CAFiqYu=9diVPV1+UHFiZF3dQhUcnKZgZpLMH-GuAyrCWZ20sFA@mail.gmail.com>

Set the sip-ip and rtp-ip parameters on the profiles directly rather than
using ${local_ipv4} (which guesses your main WAN IP) or setting the
external_ip variables in vars.xml. The later are only to simplify matters
on simple setups (ie one WAN IP), for multihomed hosts such as yours you
should set it on the profile directly.

On 8 May 2015 at 13:58, Jason Holden <jason.holden at start.ca> wrote:

>  Currently I have a client who is using multiple WAN IP addresses.
>
> On route fail over FS still tries to use the original IP details from the
> external profile and not the secondary circuit details from the secondary
> external profile.
>
> What have others done to address this to ensure failover works propperly?
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jason.holden at start.ca  Mon May 11 16:38:44 2015
From: jason.holden at start.ca (Jason Holden)
Date: Mon, 11 May 2015 08:38:44 -0400
Subject: [Freeswitch-users] need SIP IP / RTP recommendation
In-Reply-To: <CAFiqYu=9diVPV1+UHFiZF3dQhUcnKZgZpLMH-GuAyrCWZ20sFA@mail.gmail.com>
References: <1DD529F81979496BADBE1B52EB5D334B@bob>
	<CAFiqYu=9diVPV1+UHFiZF3dQhUcnKZgZpLMH-GuAyrCWZ20sFA@mail.gmail.com>
Message-ID: <044E255B31C44F7E8FF217DCF89330C7@bob>

I am, the problem is when the route changes on the FS server due to an
outage on the primary link.

Is there a way to force FS to switch between the IPS dependant on the
default route?

I have two profiles and each is setup with a separate SIP EXT, RTP EXT IP
dependant on its WAN link details.

When the primary link goes down and the failover circuit is operational
calls are still trying to route over the primary profile and SIP is broken
since those public IPS are currently not in use.

 

 

 

  _____  

From: Steven Ayre [mailto:steveayre at gmail.com] 
Sent: Monday, May 11, 2015 3:49 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] need SIP IP / RTP recommendation

 

Set the sip-ip and rtp-ip parameters on the profiles directly rather than
using ${local_ipv4} (which guesses your main WAN IP) or setting the
external_ip variables in vars.xml. The later are only to simplify matters on
simple setups (ie one WAN IP), for multihomed hosts such as yours you should
set it on the profile directly.

 

On 8 May 2015 at 13:58, Jason Holden <jason.holden at start.ca> wrote:

Currently I have a client who is using multiple WAN IP addresses.

On route fail over FS still tries to use the original IP details from the
external profile and not the secondary circuit details from the secondary
external profile.

What have others done to address this to ensure failover works propperly?

 

 

 

 


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

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

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

 

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From michel.brabants at gmail.com  Mon May 11 17:28:35 2015
From: michel.brabants at gmail.com (Michel Brabants)
Date: Mon, 11 May 2015 15:28:35 +0200
Subject: [Freeswitch-users] info about sofia sip dns cache and
	graylisting?
In-Reply-To: <CAEJMVkA=Gw5XHtvv89o4r1X95sjGFwbcyLFAJNixDAuqJtLrYA@mail.gmail.com>
References: <CAHt+dNGgTiN9eNJXFdig03t5o2FEtPPZt6ksAcPMNN_RvWk=Ng@mail.gmail.com>
	<CAEJMVkA=Gw5XHtvv89o4r1X95sjGFwbcyLFAJNixDAuqJtLrYA@mail.gmail.com>
Message-ID: <CAHt+dNEKhw3jsEiBVjXngUS8oTx6KmEa+ztosKsVaJ-AsWf2mw@mail.gmail.com>

Hey Brian,

thanks for the reply. I created a new ticket "
https://freeswitch.org/jira/browse/FS-7543" with my take on the issue and
how to resolve it.

Michel

On Fri, May 8, 2015 at 4:38 PM, Brian West <brian at freeswitch.org> wrote:

> can you get some logs, sofia loglevel all 9, and file a JIRA and attach
> the logs and detailed info to the JIRA please.
>
> Also verify which rev of FreeSWITCH you're running when filing the JIRA.
>
> On Fri, May 8, 2015 at 9:34 AM, Michel Brabants <michel.brabants at gmail.com
> > wrote:
>
>> Hello all,
>>
>> we have a problem with a 1.4-version of Freeswitch that FS goes to 100%
>> cpu. The problem has been traced to a dns-cache-problem. The dns-cache
>> contains srv-records, which should have 2 a-records, but they have 100's of
>> a-records, which are mostly the same (some with a different priority). This
>> causes FS do to a lot of compares (to sort the records) and it goes to 100%
>> cpu in the end.
>> The priority almost never is the original priority, so I was thinking
>> that the cache is maybe growing because of graylisting of entries, causing
>> the priority to increase and the ttl to be reset.
>>
>> The code is however very difficult to follow as it is all event-based and
>> part of big C-structs in sofia-sip. A bug in the gdb-debugger causing
>> static variables to be almost unreadable makes life not much easier. The
>> graylisting policy, as well as the dns cache policy is not really
>> documented as far as I can find, so any info about this would be great.
>>
>> Anyway, my current thought is that the dns-cache is maybe growing,
>> becuase the graylisting-code sets the priority to a higher value and resets
>> the ttl to a certain value (no idea which one, but the default is 10
>> minutes and the maximum a day). The srv-record-compare-function however
>> checks also using priority and weight and returns a negative response if
>> they are different. I would think that fs maybe then readds the original
>> srv-records (with priority 0) when it requeries the dns. Those entries will
>> also increase in priority and ttl when they are added, causing the cache to
>> keep growing ...
>> Nscd is also being used for dns-caching, which doesn't help, but I'm not
>> sure if that is the problem as sofia_dig seems to return valid values
>> (except maybe ttl).
>>
>>
>>  I would think that ttl is never touched and that the cached is cleared
>> when the ttl expires, but I have no idea when the dns-cache is cleared in
>> FS...
>>
>> I know this is maybe a lot of text, but any info on this topic (sofia dns
>> cache) is welcome.
>>
>> Thanks,
>>
>> Michel
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 aqsyounas at gmail.com  Mon May 11 22:22:11 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Mon, 11 May 2015 11:22:11 -0700
Subject: [Freeswitch-users] bind_digit_action for lua
In-Reply-To: <CAC8JMO+YwH0mQkfAOPiADqLkitR4Bf6Avg0W59nasA0epeDU-w@mail.gmail.com>
References: <CAGVOjVXXZ9=gvaXxVSu5DoXViO6-Y1QPxhrZyEbuHKbKJA3m6g@mail.gmail.com>
	<CAC8JMOLH91fz_mQ07wg_gZ9uLCOWJ3qr1yrWz9VTwEWBe2B24A@mail.gmail.com>
	<CAGVOjVV69HdRqKE2tipETzNqdHGS+=Xh0SA9mjFEC10o+f3Nzg@mail.gmail.com>
	<CAC8JMO+YwH0mQkfAOPiADqLkitR4Bf6Avg0W59nasA0epeDU-w@mail.gmail.com>
Message-ID: <CAGVOjVWjBnCYoo+FwKbkXKt-8PEUJPLnipiw9BM8kM1XchjPkw@mail.gmail.com>

Many thanks for your answer.

On 8 May 2015 at 13:43, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> There are several ways.  One is to use a regex to match any digit and
> execute a lua script.  Inside the script check the last_matching_digits
> session variable.
>
> I have not tried it myself, so not sure how well it will work (regex and
> the variable).  If for some reason it does not, you can still use separate
> bindings for each digit, call the same lua script from each and pass the
> digit in as a parameter like this:
>
> <action application="bind_digit_action" data="my_digits,0,exec:lua,
> your_script.lua,0"/>
> <action application="bind_digit_action" data="my_digits,1,exec:lua,
> your_script.lua,1"/>
> <action application="bind_digit_action" data="my_digits,2,exec:lua,
> your_script.lua,2"/>
>
>
> On Fri, May 8, 2015 at 1:35 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Thanks for your reply.
>>
>> I think i didn't explain my question well. I don't want to execute a
>> separate lua script upon each regex match.
>> I want a single script with different functions for corresponding inputs.
>>
>> <include>
>>   <context name="default">
>>    <extension name="radio">
>>         <condition>
>>                 <action application="answer"/>
>>                 <action application="lua" data="main.lua"/>
>>   </condition>
>>   </extension>
>> </context>
>>
>>
>> --main.lua
>> function next()
>>   logic
>> end
>> function previous()
>>  logic
>> end
>>
>>
>> *logic for calling above functions goes here.  *
>> Is it possible to right above dialplan in a single script with different
>> fucntions?
>>
>> Thanks.
>>
>>
>> On 8 May 2015 at 13:11, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>
>>> I just finished playing with it.  You can just do this:
>>>
>>> <action application="bind_digit_action"
>>> data="my_digits,0,exec:lua,your_script.lua,param1 param2 param3"/>
>>>
>>>
>>>
>>> On Fri, May 8, 2015 at 1:05 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>>>
>>>> Hi, users.
>>>>
>>>> Below is my default.xml snippet. I am using bind_digit_action for
>>>> different inputs bindings and moves the call to that context upon regex
>>>> match.
>>>>
>>>> <condition>
>>>> <action application="bind_digit_action"
>>>> data="moderator,~\*,exec:execute_extension,previous XML Previous"/>
>>>>         <action application="bind_digit_action"
>>>> data="moderator,~\#,exec:execute_extension,next XML Next"/>
>>>>         <action application="bind_digit_action"
>>>> data="moderator,~^\d$,exec:execute_extension,exten XML Exten"/>
>>>>                 <action application="digit_action_set_realm"
>>>> data="moderator"/>
>>>>         <action application="answer"/>
>>>>         <action application="playback" data="shout://127.0.0.1/radio"/>
>>>> </condition>
>>>>
>>>> Now I want to write my dialplan in lua which I think is more handy than
>>>> xml.
>>>> Upon regex match call to be executed in corresponding functions.
>>>>
>>>> Like, for *
>>>> *function next()*
>>>> for #,
>>>> *function previous()*
>>>>
>>>>
>>>>
>>>> I know there is a application
>>>> *session.setHangupHook(hangup_function_name)*;
>>>> Which executes the function(hangup_function_name) upon call hangup.
>>>>
>>>> Is there anything available in lua witch executes functions upon regex
>>>> match.
>>>> Or any other way for doing this.
>>>>
>>>> Any pointer would be much appreciated.
>>>>
>>>> Thanks.
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From s.safarov at gmail.com  Mon May 11 22:32:08 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 11 May 2015 21:32:08 +0300
Subject: [Freeswitch-users] need SIP IP / RTP recommendation
In-Reply-To: <044E255B31C44F7E8FF217DCF89330C7@bob>
References: <1DD529F81979496BADBE1B52EB5D334B@bob>
	<CAFiqYu=9diVPV1+UHFiZF3dQhUcnKZgZpLMH-GuAyrCWZ20sFA@mail.gmail.com>
	<044E255B31C44F7E8FF217DCF89330C7@bob>
Message-ID: <CAHtxdDf9M0K6nU2P9y_YxNndv1S5QU3F14EyO7tzJS71bk_B7w@mail.gmail.com>

Try dummy interface (equal loopback) as external FS profile and configure
routing packets from/to this interface.

https://www.centos.org/docs/5/html/Virtualization-en-US/ch-virt-laptop-configurations.html

On Mon, May 11, 2015 at 3:38 PM, Jason Holden <jason.holden at start.ca> wrote:

>  I am, the problem is when the route changes on the FS server due to an
> outage on the primary link.
>
> Is there a way to force FS to switch between the IPS dependant on the
> default route?
>
> I have two profiles and each is setup with a separate SIP EXT, RTP EXT IP
> dependant on its WAN link details.
>
> When the primary link goes down and the failover circuit is operational
> calls are still trying to route over the primary profile and SIP is broken
> since those public IPS are currently not in use.
>
>
>
>
>
>
>   ------------------------------
>
> *From:* Steven Ayre [mailto:steveayre at gmail.com]
> *Sent:* Monday, May 11, 2015 3:49 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] need SIP IP / RTP recommendation
>
>
>
> Set the sip-ip and rtp-ip parameters on the profiles directly rather than
> using ${local_ipv4} (which guesses your main WAN IP) or setting the
> external_ip variables in vars.xml. The later are only to simplify matters
> on simple setups (ie one WAN IP), for multihomed hosts such as yours you
> should set it on the profile directly.
>
>
>
> On 8 May 2015 at 13:58, Jason Holden <jason.holden at start.ca> wrote:
>
> Currently I have a client who is using multiple WAN IP addresses.
>
> On route fail over FS still tries to use the original IP details from the
> external profile and not the secondary circuit details from the secondary
> external profile.
>
> What have others done to address this to ensure failover works propperly?
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 11 22:40:17 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 11 May 2015 18:40:17 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) May
	2nd-May 8th
Message-ID: <5550f791f2f2f_d1aa11b733452b7@resque-worker-high.4.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1bKCY9S
FreeSWITCH Week in Review (Master Branch) May 2nd-May 8th
Hello, again. This passed week in the FreeSWITCH master branch we had 6 commits. Some more work was done to mod_amqp this week as well as some bug fixes.

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

FS-7526 Add enable_fallback_format_fields for mod_amqp producer profiles if the profile param is set and create the amqp exchange on the first startup of a clean platform.

The following bugs were squashed:

FS-7425 Fixed a bug when using a cert with missing dhparams resulting in a segfault.

FS-7523 [mod_json_cdr] Fixed a segfault caused by a missing config file.

FS-7357 FAX now tolerates EOP and PPS messages being incorrectly echoed.

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From k.pabijanskas at gmail.com  Tue May 12 10:45:13 2015
From: k.pabijanskas at gmail.com (Karolis Pabijanskas)
Date: Tue, 12 May 2015 07:45:13 +0100
Subject: [Freeswitch-users] mod_callcenter: wrap_up_time
In-Reply-To: <CAP9PqkT1S9stcsNS7cJ_WXwt=eB40v+k=mdK+GzLH0fsW7YpLw@mail.gmail.com>
References: <CANYNSSL9s1puLh=qNa=6N2qbVhM9gKi+ttebxGnhNq0jqPta3g@mail.gmail.com>
	<CAP9PqkT1S9stcsNS7cJ_WXwt=eB40v+k=mdK+GzLH0fsW7YpLw@mail.gmail.com>
Message-ID: <CANYNSS+-D9KD33Qqi8Yu5B5kn0Rqbd_-10h7i1kBodcNTLqO5A@mail.gmail.com>

Hi Ali,

All of these have been tried before I posted here, it's mentioned in the
original post too :-)

Thanks

On 6 May 2015 at 15:58, Ali Jibran <ali.jibran44 at gmail.com> wrote:

> Maybe try
>
> callcenter_config agent set [key(contact|status|state|type|max_no_answer|wrap_up_time|ready_time|reject_delay_time|busy_delay_time)] [agent name] [value]
>
>
> On Wed, May 6, 2015 at 5:49 PM, Karolis Pabijanskas <
> k.pabijanskas at gmail.com> wrote:
>
>> Hi All,
>>
>> I am using a mod_callcenter for this particular set-up.
>>
>> I need the agents to have a wrap_up_time set to, say, 300 seconds, which
>> works great. In some cases, however, the agent needs to have a code he can
>> dial so he resets his wrap_up_time (say, he receives a call to the wrong
>> number).
>>
>> It does seem that if I call:
>>
>> callcenter_config agent set state AGENT Waiting
>>
>> It sets the call center to available and the wrap-up time is cancelled,
>> but the wrap-up never kicks in again!
>>
>> Is there a way to reset the agent wrap_up_time via the API (which I can
>> then call via the dialplan) so that the agent is available for a new call
>> immediately, but that the wrap-up time kicks in afterwards as normal?
>>
>> Many thanks for your help!
>>
>> Karolis
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 12 11:50:26 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 12 May 2015 10:50:26 +0300
Subject: [Freeswitch-users] mod_callcenter: wrap_up_time
In-Reply-To: <CANYNSS+-D9KD33Qqi8Yu5B5kn0Rqbd_-10h7i1kBodcNTLqO5A@mail.gmail.com>
References: <CANYNSSL9s1puLh=qNa=6N2qbVhM9gKi+ttebxGnhNq0jqPta3g@mail.gmail.com>
	<CAP9PqkT1S9stcsNS7cJ_WXwt=eB40v+k=mdK+GzLH0fsW7YpLw@mail.gmail.com>
	<CANYNSS+-D9KD33Qqi8Yu5B5kn0Rqbd_-10h7i1kBodcNTLqO5A@mail.gmail.com>
Message-ID: <CAHtxdDcYcP3MDE83Dq9i8OpFVk_z7LyUC3LP_HLtwoamaJsxdg@mail.gmail.com>

Are you tried set wrap_time to 0 and again to 300?

??????

On Tue, May 12, 2015 at 9:45 AM, Karolis Pabijanskas <
k.pabijanskas at gmail.com> wrote:

> Hi Ali,
>
> All of these have been tried before I posted here, it's mentioned in the
> original post too :-)
>
> Thanks
>
> On 6 May 2015 at 15:58, Ali Jibran <ali.jibran44 at gmail.com> wrote:
>
>> Maybe try
>>
>> callcenter_config agent set [key(contact|status|state|type|max_no_answer|wrap_up_time|ready_time|reject_delay_time|busy_delay_time)] [agent name] [value]
>>
>>
>> On Wed, May 6, 2015 at 5:49 PM, Karolis Pabijanskas <
>> k.pabijanskas at gmail.com> wrote:
>>
>>> Hi All,
>>>
>>> I am using a mod_callcenter for this particular set-up.
>>>
>>> I need the agents to have a wrap_up_time set to, say, 300 seconds, which
>>> works great. In some cases, however, the agent needs to have a code he can
>>> dial so he resets his wrap_up_time (say, he receives a call to the wrong
>>> number).
>>>
>>> It does seem that if I call:
>>>
>>> callcenter_config agent set state AGENT Waiting
>>>
>>> It sets the call center to available and the wrap-up time is cancelled,
>>> but the wrap-up never kicks in again!
>>>
>>> Is there a way to reset the agent wrap_up_time via the API (which I can
>>> then call via the dialplan) so that the agent is available for a new call
>>> immediately, but that the wrap-up time kicks in afterwards as normal?
>>>
>>> Many thanks for your help!
>>>
>>> Karolis
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lists at telefaks.de  Tue May 12 15:34:42 2015
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 12 May 2015 13:34:42 +0200
Subject: [Freeswitch-users] mod_hiredis
In-Reply-To: <554EB704.5090605@quentustech.com>
References: <554EB704.5090605@quentustech.com>
Message-ID: <5551E552.9090307@telefaks.de>

Hello William,

this is great, the idea of integrating Redis. We currently use Memcache
in raw mode as a method of externally controlling dialplans and failover
scenarios.
Redis, of course, brings much more features here.

>Currently the two main use cases are:
>1. Call per second limits
>2. Concurrent call limits
>
>Possible additional functionality:
>1. Support for fail-over connections
>2. Asynchronous commands(is there a use case for this?)

Another idea for your list would be to route calls according to
prefixes. You may lookup Redis with a part of the phone number and it
returns the gateway for this part of the number (redis DB is then
preloaded from another application).
And - as Redis has a publish/subscribe method - you will be able to
publish call informations from the dialplan to multiple external
subscribers (e.g. announce an incoming call to a CRM) without the use of
ESL. Is there a chance to run the redis dialplan app in a non blocking
manner for this scenario, in order to speed up the dialplan?



Best regards
Peter

On 05/10/15 03:40, William King wrote:
> I'm working on an update Redis integration module that will use the C
> library hiredis:
> http://redis.io/clients#c
> https://github.com/redis/hiredis
>
> I've pushed an alpha version of the module to a branch here:
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits?until=
> refs%2Fheads%2Fmod_hiredis
>
> The current module has a dialplan app and an api for 'hiredis_raw'
> which allows any single line Redis command, and executes it in a
> blocking manner, then supports returning string and integer responses.
>
> If anyone on this list has any use cases for FreeSWITCH+Redis, please
> reply to this thread. Currently the two main use cases are:
> 1. Call per second limits
> 2. Concurrent call limits
>
> Possible additional functionality:
> 1. Support for fail-over connections
> 2. Asynchronous commands(is there a use case for this?)
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 yadenis at seznam.cz  Tue May 12 15:50:25 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 12 May 2015 13:50:25 +0200
Subject: [Freeswitch-users] 1.6 install to Centos
Message-ID: <17898630.20150512135025@seznam.cz>

Hi all,

Is there a chance to put on CentOS the new version 1.6?

I've tried. But I have a problem on ./configure -C 
checking for libyuv >= 0.0.1280... configure: error: You need to install libyuv-dev. Required library

Where can I find this library? I tried to put it from here https://code.google.com/p/libyuv/ But FreeSwitch does not see it.

Is there a possibility to find this library to CentOS?

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From vipkilla at gmail.com  Tue May 12 16:24:33 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Tue, 12 May 2015 08:24:33 -0400
Subject: [Freeswitch-users] sending custom verto events to client
Message-ID: <CAC-LwPPOcMzzytZKS8bzRWfOrAoEfevu8maf_JGPVjy2KLOvAQ@mail.gmail.com>

Hello,
We have verto setup and working.
It is great btw.
We want to be able to send custom json data back to the client (hopefully
using dialplan if possble)
I don't really know if this is even possible or if it is, where to start.
I change mod_verto to verbose debugging and I can see it sends json data
back to the verto client, but I am lost as to how to do this on my own.
Here is an example json sent from FS to verto client:

2015-05-12 05:18:04.470641 [ALERT] mod_verto.c:604 WRITE
192.168.10.101:48165 [{
        "jsonrpc":      "2.0",
        "id":   8,
        "result":       {
                "message":      "CALL CREATED",
                "callID":       "be3e35a0-ec02-5ca7-cbf8-68a8c1cd2c4c",
                "sessid":       "48a502b3-740a-5b3d-21a4-bf9244bafb90"
        }
}]



I see there is a "json" API command in FS console but when i try to run
something like

json {"message":"CALL
CREATED","callID":"be3e35a0-ec02-5ca7-cbf8-68a8c1cd2c4c","sessid":"48a502b3-740a-5b3d-21a4-bf9244bafb90"}

I get

{"message":"CALL
CREATED","callID":"be3e35a0-ec02-5ca7-cbf8-68a8c1cd2c4c","sessid":"48a502b3-740a-5b3d-21a4-bf9244bafb90","status":"error","message":"Invalid
request or non-existant command","response":null}


If someone know how I can accomplish this, I'd be happy to update
confluence with details.
Thanks,
V
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From asdfgh1234 at freemail.hu  Tue May 12 17:03:33 2015
From: asdfgh1234 at freemail.hu (Asd)
Date: Tue, 12 May 2015 15:03:33 +0200 (CEST)
Subject: [Freeswitch-users] FreeSWITCH PROXY:0
Message-ID: <freemail.20150512150333.44759.1@fmxmldata05.freemail.hu>

Hi all Freeswitch version: 1.4.18 Debian stable with TLS if i try to call internal: RINGING <<== OKby receive of call <<== NORMAL_CLEARING then KILL with version 1.4.12 worked what  am i doing wrong? my log file: tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803aad0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803aad0)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803aad0): tls_read() returned 1143
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803aad0) msg 0x7fac6805d660 from (tls/XXX.XXX.XXX.XXX:22035) has 1143 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fac6803aad0): msg 0x7fac6805d660 (1143 bytes) from tls/XXX.XXX.XXX.XXX:22035/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received INVITE sips:1004 at sip.domain.com:9061 SIP/2.0 (CSeq 15796)
nta.c:3174 agent_check_request_via() nta: Via check: received=XXX.XXX.XXX.XXX
nta.c:3085 agent_recv_request() nta: INVITE (15796) going to a default leg
nta.c:1348 set_timeout() nta: timer shortened to 2000 ms
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
nua_stack.c:899 nh_create() nua: nh_create: entering
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fac68002100, 0x7fac680015a0, 0x7fac68077650) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fac68073910, ...) called
nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fac680711d0)
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7fac68073910) called
soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fac68073910, (nil), 0x7fac68062468, 431) called
nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fac68077650): adding session usage
tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn = TLS/XXX.XXX.XXX.XXX:22035
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803b270 0x7fac6804e0f0 385 (385)
tport.c:3492 tport_send_msg() tport_vsend returned 385
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (15796)
nua_session.c:4139 signal_call_state_change() nua(0x7fac68077650): call state changed: init -> received, received offer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fac68073910, [0x7fac7f7018b8], [0x7fac7f7018c0], [(nil)]) called
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2015-04-26 19:27:42.024708 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1001 at sip.domain.com [b847625c-43a8-4ece-9f12-9f7b5e84be32]                                                                                                                                        
2015-04-26 19:27:42.024708 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:42.024708 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-04-26 19:27:42.024708 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_NEW
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:42.024708 [DEBUG] sofia.c:8844 sofia/internal/1001 at sip.domain.com receiving invite from XXX.XXX.XXX.XXX:22035 version: 1.4.18 -3-1 64bit                                                                                                                                
2015-04-26 19:27:42.024708 [DEBUG] switch_core_media.c:344 Found audio zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10 b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe                                                                                                 
2015-04-26 19:27:42.024708 [DEBUG] sofia.c:9011 IP XXX.XXX.XXX.XXX Rejected by acl "domains". Falling back to Digest auth.
nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:529 nua_signal() nua(0x7fac68077650): sent signal r_respond
2015-04-26 19:27:42.024708 [WARNING] sofia_reg.c:1742 SIP auth challenge (INVITE) on sofia profile &#39;internal&#39; for [1004 at sip.domain.com] from ip XXX.XXX.XXX.XXX                                                                                                                          
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
2015-04-26 19:27:42.024708 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1001 at sip.domain.com) State NEW
soa.c:403 soa_set_params() soa_set_params(static::0x7fac68073910, ...) called
nua_session.c:2320 nua_invite_server_respond() nua: nua_invite_server_respond: entering
soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0x7fac68073910) called
tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn = TLS/XXX.XXX.XXX.XXX:22035
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803b270 0x7fac6805a8a0 889 (889)
tport.c:3492 tport_send_msg() tport_vsend returned 889
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
nta.c:6791 incoming_reply() nta: sent 407 Proxy Authentication Required for INVITE (15796)
nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7fac68077650): removing session usage
nua_session.c:4139 signal_call_state_change() nua(0x7fac68077650): call state changed: received -> terminated
soa.c:356 soa_destroy() soa_destroy(static::0x7fac68073910) called
nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7fac680711d0)
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:42.024708 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:42.024708 [DEBUG] sofia.c:2065 detaching session b847625c-43a8-4ece-9f12-9f7b5e84be32
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:529 nua_signal() nua(0x7fac68077650): sent signal r_destroy
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803aad0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803aad0)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803aad0): tls_read() returned 412
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803aad0) msg 0x7fac6804e0f0 from (tls/XXX.XXX.XXX.XXX:22035) has 412 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fac6803aad0): msg 0x7fac6804e0f0 (412 bytes) from tls/XXX.XXX.XXX.XXX:22035/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received ACK sips:1004 at sip.domain.com:9061 SIP/2.0 (CSeq 15796)
nta.c:3174 agent_check_request_via() nta: Via check: received=XXX.XXX.XXX.XXX
nta.c:3019 agent_recv_request() nta: ACK (15796) is going to INVITE (15796)
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803aad0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803aad0)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803aad0): tls_read() returned 1419
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803aad0) msg 0x7fac6804e0f0 from (tls/XXX.XXX.XXX.XXX:22035) has 1419 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fac6803aad0): msg 0x7fac6804e0f0 (1419 bytes) from tls/XXX.XXX.XXX.XXX:22035/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received INVITE sips:1004 at sip.domain.com:9061 SIP/2.0 (CSeq 15797)
nta.c:3174 agent_check_request_via() nta: Via check: received=XXX.XXX.XXX.XXX
nta.c:3085 agent_recv_request() nta: INVITE (15797) going to a default leg
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
nua_stack.c:899 nh_create() nua: nh_create: entering
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fac68002100, 0x7fac680015a0, 0x7fac680706f0) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fac680598d0, ...) called
nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fac68073b20)
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7fac680598d0) called
soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fac680598d0, (nil), 0x7fac6805d1ac, 431) called
nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fac680706f0): adding session usage
tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn = TLS/XXX.XXX.XXX.XXX:22035
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803b270 0x7fac6805b1d0 385 (385)
tport.c:3492 tport_send_msg() tport_vsend returned 385
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (15797)
nua_session.c:4139 signal_call_state_change() nua(0x7fac680706f0): call state changed: init -> received, received offer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fac680598d0, [0x7fac7f7018b8], [0x7fac7f7018c0], [(nil)]) called
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2015-04-26 19:27:42.204725 [DEBUG] sofia.c:2173 Re-attaching to session b847625c-43a8-4ece-9f12-9f7b5e84be32
2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:42.204725 [DEBUG] sofia.c:8844 sofia/internal/1001 at sip.domain.com receiving invite from XXX.XXX.XXX.XXX:22035 version: 1.4.18 -3-1 64bit                                                                                                                                
2015-04-26 19:27:42.204725 [DEBUG] switch_core_media.c:344 Found audio zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10 b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe                                                                                                 
2015-04-26 19:27:42.204725 [DEBUG] sofia.c:9011 IP XXX.XXX.XXX.XXX Rejected by acl "domains". Falling back to Digest auth.
nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil)
2015-04-26 19:27:42.204725 [DEBUG] sofia.c:10109 Setting NAT mode based on via received
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:42.204725 [DEBUG] sofia.c:6623 Channel sofia/internal/1001 at sip.domain.com entering state [received][100]
2015-04-26 19:27:42.204725 [DEBUG] sofia.c:6633 Remote SDP:
v=0                                                                                                                                        
o=- 3639058062 3639058062 IN IP4 192.168.51.103                                                                                            
s=pjmedia                                                                                                                                  
c=IN IP4 192.168.51.103                                                                                                                    
t=0 0                                                                                                                                      
m=audio 4002 RTP/AVP 99 0 8 101                                                                                                            
c=IN IP4 192.168.51.103                                                                                                                    
a=rtpmap:99 SILK/24000                                                                                                                     
a=fmtp:99 useinbandfec=0                                                                                                                   
a=rtpmap:0 PCMU/8000                                                                                                                       
a=rtpmap:8 PCMA/8000                                                                                                                       
a=rtpmap:101 telephone-event/8000                                                                                                          
a=fmtp:101 0-16                                                                                                                            
a=rtcp:4003 IN IP4 192.168.51.103                                                                                                          
a=zrtp-hash:1.10 b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe                                                          2015-04-26 19:27:42.204725 [DEBUG] sofia.c:6893 (sofia/internal/1001 at sip.domain.com) State Change CS_NEW -> CS_INIT
2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_INIT
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1001 at sip.domain.com) State INIT
2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:87 sofia/internal/1001 at sip.domain.com SOFIA INIT
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1001 at sip.domain.com Standard INIT
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1001 at sip.domain.com) State Change CS_INIT -> CS_ROUTING
2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1001 at sip.domain.com) State INIT going to sleep
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_ROUTING
2015-04-26 19:27:42.204725 [DEBUG] switch_channel.c:2184 (sofia/internal/1001 at sip.domain.com) Callstate Change DOWN -> RINGING
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1001 at sip.domain.com) State ROUTING
2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:123 sofia/internal/1001 at sip.domain.com SOFIA ROUTING
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1001 at sip.domain.com Standard ROUTING
2015-04-26 19:27:42.204725 [INFO] mod_dialplan_xml.c:635 Processing 1001 <1001>->1004 in context sip.domain.com
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->PHONE-FAX_b_1001_49123456789100] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [PHONE-FAX_b_1001_49123456789100] context(sip.domain.com) =~ /sip.domain.com/ break=on-false                                                                                                                                
Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL) [PHONE-FAX_b_1001_49123456789100] destination_number(1004) =~ /^(1234567e0)$/ break=on-false                                                                                                                              
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->PHONE-FAX_l_1002_49123456789101] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [PHONE-FAX_l_1002_49123456789101] context(sip.domain.com) =~ /sip.domain.com/ break=on-false                                                                                                                               
Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL) [PHONE-FAX_l_1002_49123456789101] destination_number(1004) =~ /^(1234567e1)$/ break=on-false                                                                                                                             
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->PHONE-FAX_m_1003_49123456789101-copy] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [PHONE-FAX_m_1003_49123456789102] context(sip.domain.com) =~ /sip.domain.com/ break=on-false                                                                                                                         
Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL) [PHONE-FAX_m_1003_49123456789102] destination_number(1004) =~ /^(1234567e2)$/ break=on-false                                                                                                                       
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->internal-voicemail-com] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal-voicemail-com] context(sip.domain.com) =~ /sip.domain.com/ break=on-false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal-voicemail-com] caller_id_number(1001) =~ /^(((\+|00)49)|1\d{3})$/ break=on-false                                                                                                                                    
Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL) [internal-voicemail-com] ${user_data(${destination_number}@${domain} param vm-enabled)}(false) =~ /true/ break=on-false                                                                                                       
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->internal-voicemail-com2] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal-voicemail-com2] context(sip.domain.com) =~ /sip.domain.com/ break=on-false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL) [internal-voicemail-com2] ${user_data(${destination_number}@${domain} param vm-enabled)}(false) =~ /true/ break=on-false                                                                                                       
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->internal] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal] context(sip.domain.com) =~ /sip.domain.com/ break=on-false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal] destination_number(1004) =~ /^(1\d{3})$/ break=on-false
Dialplan: sofia/internal/1001 at sip.domain.com Action set(zrtp_enrollment=true) 
Dialplan: sofia/internal/1001 at sip.domain.com Action set(hangup_after_bridge=false) 
Dialplan: sofia/internal/1001 at sip.domain.com Action set(continue_on_fail=true) 
Dialplan: sofia/internal/1001 at sip.domain.com Action set(intcallid=1004) 
Dialplan: sofia/internal/1001 at sip.domain.com Action bridge(sofia/internal/${intcallid}%${domain}) 
Dialplan: sofia/internal/1001 at sip.domain.com Action set(eml=${user_data(${intcallid}@${domain} param vm-mailto)}) 
Dialplan: sofia/internal/1001 at sip.domain.com Action set(datetime=${strftime(%Y.%m.%d)} | ${strftime(%W)}. HET | ${strftime(%H:%M)}) 
Dialplan: sofia/internal/1001 at sip.domain.com Action set(smtp_from=root) 
Dialplan: sofia/internal/1001 at sip.domain.com Action lua(NoAns.lua &#39;${originate_disposition}&#39; &#39;${eml}&#39; &#39;${smtp_from}&#39; &#39;aaa&#39; &#39;aaa&#39; &#39;bbb&#39;)                                                                                                                                      
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1001 at sip.domain.com) State Change CS_ROUTING -> CS_EXECUTE                                                                                                                                          
2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1001 at sip.domain.com) State ROUTING going to sleep
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_EXECUTE
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1001 at sip.domain.com) State EXECUTE
2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:178 sofia/internal/1001 at sip.domain.com SOFIA EXECUTE
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1001 at sip.domain.com Standard EXECUTE
EXECUTE sofia/internal/1001 at sip.domain.com set(zrtp_enrollment=true)
2015-04-26 19:27:42.204725 [DEBUG] mod_dptools.c:1445 sofia/internal/1001 at sip.domain.com SET [zrtp_enrollment]=[true]
EXECUTE sofia/internal/1001 at sip.domain.com set(hangup_after_bridge=false)
2015-04-26 19:27:42.204725 [DEBUG] mod_dptools.c:1445 sofia/internal/1001 at sip.domain.com SET [hangup_after_bridge]=[false]
EXECUTE sofia/internal/1001 at sip.domain.com set(continue_on_fail=true)
2015-04-26 19:27:42.204725 [DEBUG] mod_dptools.c:1445 sofia/internal/1001 at sip.domain.com SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1001 at sip.domain.com set(intcallid=1004)
2015-04-26 19:27:42.204725 [DEBUG] mod_dptools.c:1445 sofia/internal/1001 at sip.domain.com SET [intcallid]=[1004]
EXECUTE sofia/internal/1001 at sip.domain.com bridge(sofia/internal/1004%sip.domain.com)
2015-04-26 19:27:42.204725 [DEBUG] switch_channel.c:1201 sofia/internal/1001 at sip.domain.com EXPORTING[export_vars] [domain_name]=[sip.domain.com] to event                                                                                                                              
2015-04-26 19:27:42.204725 [DEBUG] switch_ivr_originate.c:2100 Parsing global variables
2015-04-26 19:27:42.204725 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1004 [f8a9e60a-a497-4363-bed4-039df448d26d]
2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:4701 (sofia/internal/1004) State Change CS_NEW -> CS_INIT
2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-04-26 19:27:42.204725 [DEBUG] switch_core_media.c:266 Passing a-leg remote zrtp-hash (audio) to b-leg
2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:4771 [zrtp_passthru] Setting a-leg inherit_codec=true
2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:4774 [zrtp_passthru] Setting b-leg absolute_codec_string=&#39;PCMA at 8000h@20i at 64000b&#39;
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_INIT
2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1004) State INIT
2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:87 sofia/internal/1004 SOFIA INIT
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2015-04-26 19:27:42.224657 [DEBUG] switch_core_media.c:7687 sofia/internal/1004 Patched SDP
---                                                                                                                                        
v=0                                                                                                                                        
o=- 3639058062 3639058062 IN IP4 192.168.51.103                                                                                            
s=pjmedia                                                                                                                                  
c=IN IP4 192.168.51.103                                                                                                                    
t=0 0                                                                                                                                      
m=audio 4002 RTP/AVP 99 0 8 101                                                                                                            
c=IN IP4 192.168.51.103                                                                                                                    
a=rtpmap:99 SILK/24000                                                                                                                     
a=fmtp:99 useinbandfec=0                                                                                                                   
a=rtpmap:0 PCMU/8000                                                                                                                       
a=rtpmap:8 PCMA/8000                                                                                                                       
a=rtpmap:101 telephone-event/8000                                                                                                          
a=fmtp:101 0-16                                                                                                                            
a=rtcp:4003 IN IP4 192.168.51.103                                                                                                          
a=zrtp-hash:1.10 b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe                                                          +++                                                                                                                                        
v=0                                                                                                                                        
o=FreeSWITCH 1055498831 1055498832 IN IP4 YYY.YYY.YYY.YYY                                                                                   
s=FreeSWITCH                                                                                                                               
c=IN IP4 YYY.YYY.YYY.YYY                                                                                                                    
t=0 0                                                                                                                                      
m=audio 20654 RTP/AVP 99 0 8 101                                                                                                           
c=IN IP4 YYY.YYY.YYY.YYY                                                                                                                    
a=rtpmap:99 SILK/24000                                                                                                                     
a=fmtp:99 useinbandfec=0                                                                                                                   
a=rtpmap:0 PCMU/8000                                                                                                                       
a=rtpmap:8 PCMA/8000                                                                                                                       
a=rtpmap:101 telephone-event/8000                                                                                                          
a=fmtp:101 0-16                                                                                                                            
a=rtcp:4003 IN IP4 192.168.51.103                                                                                                          
a=zrtp-hash:1.10 b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe                                                          2015-04-26 19:27:42.224657 [DEBUG] sofia_glue.c:1203 sip:1004 at XXX.XXX.XXX.XXX:22036;transport=tls;registering_acc=sip_domain_com Setting proxy route to sofia/internal/1004                                                                                                              
2015-04-26 19:27:42.224657 [DEBUG] sofia_glue.c:1232 sofia/internal/1004 sending invite version: 1.4.18 -3-1 64bit
Local SDP:                                                                                                                                 
v=0                                                                                                                                        
o=FreeSWITCH 1055498831 1055498832 IN IP4 YYY.YYY.YYY.YYY                                                                                   
s=FreeSWITCH                                                                                                                               
c=IN IP4 YYY.YYY.YYY.YYY                                                                                                                    
t=0 0                                                                                                                                      
m=audio 20654 RTP/AVP 99 0 8 101                                                                                                           
c=IN IP4 YYY.YYY.YYY.YYY                                                                                                                    
a=rtpmap:99 SILK/24000                                                                                                                     
a=fmtp:99 useinbandfec=0                                                                                                                   
a=rtpmap:0 PCMU/8000                                                                                                                       
a=rtpmap:8 PCMA/8000                                                                                                                       
a=rtpmap:101 telephone-event/8000                                                                                                          
a=fmtp:101 0-16                                                                                                                            
a=rtcp:4003 IN IP4 192.168.51.103                                                                                                          
a=zrtp-hash:1.10 b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe                                                          nua.c:633 nua_invite() nua: nua_invite: entering
nua_stack.c:529 nua_signal() nua(0x7fac38005d80): sent signal r_invite
2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1004 Standard INIT
2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1004) State Change CS_INIT -> CS_ROUTING
2015-04-26 19:27:42.224657 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1004) State INIT going to sleep
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fac68002100, 0x7fac680015a0, 0x7fac38005d80) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fac6805ba70, ...) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fac6805ba70, ...) called
soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fac6805ba70, (nil), 0x7fac3802d316, -1) called
soa.c:890 soa_set_capability_sdp() soa_set_capability_sdp(static::0x7fac6805ba70, (nil), 0x7fac3802d316, -1) called
nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fac38005d80): adding session usage
nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fac680596f0)
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7fac6805ba70) called
soa.c:1426 soa_generate_offer() soa_generate_offer(static::0x7fac6805ba70, 0) called
soa_static.c:1146 offer_answer_step() soa_static_offer_answer_action(0x7fac6805ba70, soa_generate_offer): called
soa_static.c:1187 offer_answer_step() soa_static(0x7fac6805ba70, soa_generate_offer): generating local description
soa_static.c:1215 offer_answer_step() soa_static(0x7fac6805ba70, soa_generate_offer): upgrade with local description
soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fac7f6ffa70, (nil), ""): called
soa_static.c:1444 offer_answer_step() soa_static(0x7fac6805ba70, soa_generate_offer): storing local description
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fac6805ba70, [(nil)], [0x7fac7f701bf8], [0x7fac7f701bf4]) called
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
tport.c:4588 tport_by_name() tport(0x7fac680044d0): found 0x7fac6803c380 by name tls/XXX.XXX.XXX.XXX:22036
tport.c:3257 tport_tsend() tport_tsend(0x7fac6803c380) tpn = tls/XXX.XXX.XXX.XXX:22036
2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_ROUTING
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803c670 0x7fac6807af60 1018 (1018)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803c670 0x7fac6807a870 90 (90)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803c670 0x7fac6807bfc0 432 (432)
tport.c:3492 tport_send_msg() tport_vsend returned 1540
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
nta.c:8304 outgoing_send() nta: sent INVITE (74698567) to tls/XXX.XXX.XXX.XXX:22036
tport.c:4160 tport_pend() tport_pend(0x7fac6803c380): pending 0x7fac68079d20 for tls/XXX.XXX.XXX.XXX:22036 (already 0)
nua_session.c:4139 signal_call_state_change() nua(0x7fac38005d80): call state changed: init -> calling, sent offer
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fac6805ba70, [0x7fac7f701bd8], [0x7fac7f701be0], [(nil)]) called
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:42.224657 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:42.224657 [DEBUG] sofia.c:6623 Channel sofia/internal/1004 entering state [calling][0]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1004) State ROUTING
2015-04-26 19:27:42.224657 [DEBUG] mod_sofia.c:123 sofia/internal/1004 SOFIA ROUTING
2015-04-26 19:27:42.224657 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/1004) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2015-04-26 19:27:42.224657 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1004) State ROUTING going to sleep
2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_CONSUME_MEDIA
2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/1004) State CONSUME_MEDIA
2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/1004) State CONSUME_MEDIA going to sleep
tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380): tls_read() returned 1
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg 0x7fac6807cc00 from (tls/XXX.XXX.XXX.XXX:22036) has 1 bytes, veclen = 1
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380): tls_read() returned 497
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg 0x7fac6807cc00 from (tls/XXX.XXX.XXX.XXX:22036) has 497 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fac6803c380): msg 0x7fac6807cc00 (498 bytes) from tls/XXX.XXX.XXX.XXX:22036/sips next=(nil)
nta.c:3299 agent_recv_response() nta: received 180 Ringing for INVITE (74698567)
nta.c:3366 agent_recv_response() nta: 180 Ringing is going to a transaction
nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 93.106 ms
tport.c:4222 tport_release() tport_release(0x7fac6803c380): 0x7fac68079d20 by 0x7fac6807c180 with 0x7fac6807cc00 (preliminary)
nua_session.c:4139 signal_call_state_change() nua(0x7fac38005d80): call state changed: calling -> proceeding
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:42.304700 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:42.304700 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:42.304700 [DEBUG] sofia.c:6623 Channel sofia/internal/1004 entering state [proceeding][180]
2015-04-26 19:27:42.304700 [NOTICE] sofia.c:6725 Ring-Ready sofia/internal/1004!
2015-04-26 19:27:42.304700 [DEBUG] switch_channel.c:3277 (sofia/internal/1004) Callstate Change DOWN -> RINGING
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:529 nua_signal() nua(0x7fac680706f0): sent signal r_respond
2015-04-26 19:27:42.324677 [NOTICE] mod_sofia.c:2107 Ring-Ready sofia/internal/1001 at sip.domain.com!
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fac680598d0, ...) called
nua_session.c:2320 nua_invite_server_respond() nua: nua_invite_server_respond: entering
tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn = TLS/XXX.XXX.XXX.XXX:22035
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803b270 0x7fac6807e4e0 799 (799)
tport.c:3492 tport_send_msg() tport_vsend returned 799
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
nta.c:6791 incoming_reply() nta: sent 180 Ringing for INVITE (15797)
nua_session.c:4139 signal_call_state_change() nua(0x7fac680706f0): call state changed: received -> early
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:42.324677 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:42.324677 [DEBUG] sofia.c:6623 Channel sofia/internal/1001 at sip.domain.com entering state [early][180]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:42.324677 [DEBUG] switch_core_session.c:912 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-04-26 19:27:42.324677 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/1001 at sip.domain.com!
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fac7f701c60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 0/0 term, 1/2 free
nta.c:1296 agent_timer() nta: timer set next to 2657 ms
nta.c:9101 outgoing_timer_dk() nta: timer D fired, terminate INVITE (74698551)
nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7fac7f701d40)
nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/1 term, 1/3 free
nta.c:1296 agent_timer() nta: timer set next to 1 ms
nta.c:8982 outgoing_timer_bf() nta: timer F fired, terminating ACK (74698551)
nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7fac7f701d40)
nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 1/2 free
nta.c:1296 agent_timer() nta: timer set next to 55631 ms
tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380): tls_read() returned 1
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg 0x7fac6805d660 from (tls/XXX.XXX.XXX.XXX:22036) has 1 bytes, veclen = 1
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380): tls_read() returned 655
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg 0x7fac6805d660 from (tls/XXX.XXX.XXX.XXX:22036) has 655 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fac6803c380): msg 0x7fac6805d660 (656 bytes) from tls/XXX.XXX.XXX.XXX:22036/sips next=(nil)
nta.c:3299 agent_recv_response() nta: received 200 OK for INVITE (74698567)
nta.c:3366 agent_recv_response() nta: 200 OK is going to a transaction
tport.c:4222 tport_release() tport_release(0x7fac6803c380): 0x7fac68079d20 by 0x7fac6807c180 with 0x7fac6805d660
nta.c:1348 set_timeout() nta: timer shortened to 32000 ms
soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fac6805ba70, (nil), 0x7fac6805aa2e, 130) called
soa.c:1595 soa_process_answer() soa_process_answer(static::0x7fac6805ba70) called
soa_static.c:1146 offer_answer_step() soa_static_offer_answer_action(0x7fac6805ba70, soa_process_answer): called
soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fac6807b530, 0x7fac6806f530, ""): called
soa_static.c:1302 offer_answer_step() soa_static(0x7fac6805ba70, soa_process_answer): upgrade codecs with remote description
soa_static.c:1444 offer_answer_step() soa_static(0x7fac6805ba70, soa_process_answer): storing local description
soa.c:1730 soa_activate() soa_activate(static::0x7fac6805ba70, (nil)) called
nua_session.c:988 nua_session_client_response() nua(0x7fac38005d80): INVITE: processed SDP answer in 200 OK
nua_session.c:4139 signal_call_state_change() nua(0x7fac38005d80): call state changed: proceeding -> completing, received answer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fac6805ba70, [0x7fac7f7015d8], [0x7fac7f7015e0], [(nil)]) called
soa.c:616 soa_get_params() soa_get_params(static::0x7fac6805ba70, ...) called
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:54.204709 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:54.204709 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:54.204709 [DEBUG] switch_core_media.c:272 Passing b-leg remote zrtp-hash (audio) to a-leg
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:54.204709 [DEBUG] sofia.c:6623 Channel sofia/internal/1004 entering state [completing][200]
2015-04-26 19:27:54.204709 [DEBUG] sofia.c:6633 Remote SDP:
v=0                                                                                                                                        
o=1004-jitsi.org 0 0 IN IP4 192.168.51.11                                                                                                  
s=-                                                                                                                                        
c=IN IP4 192.168.51.11                                                                                                                     
t=0 0                                                                                                                                      
m=audio 5002 RTP/AVP 8                                                                                                                     
a=rtpmap:8 PCMA/8000                                                                                                                       nua.c:639 nua_ack() nua: nua_ack: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
nua_stack.c:529 nua_signal() nua(0x7fac38005d80): sent signal r_ack
soa.c:403 soa_set_params() soa_set_params(static::0x7fac6805ba70, ...) called
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
soa.c:1730 soa_activate() soa_activate(static::0x7fac6805ba70, (nil)) called
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
tport.c:4588 tport_by_name() tport(0x7fac680044d0): found 0x7fac6803c380 by name tls/XXX.XXX.XXX.XXX:22036
tport.c:3257 tport_tsend() tport_tsend(0x7fac6803c380) tpn = tls/XXX.XXX.XXX.XXX:22036
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803c670 0x7fac68071360 475 (475)
tport.c:3492 tport_send_msg() tport_vsend returned 475
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
nta.c:8304 outgoing_send() nta: sent ACK (74698567) to tls/XXX.XXX.XXX.XXX:22036
nua_session.c:4139 signal_call_state_change() nua(0x7fac38005d80): call state changed: completing -> ready
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:54.204709 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:54.204709 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:54.204709 [DEBUG] sofia.c:6623 Channel sofia/internal/1004 entering state [ready][200]
2015-04-26 19:27:54.204709 [DEBUG] switch_channel.c:3635 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-04-26 19:27:54.204709 [NOTICE] sofia.c:7446 Channel [sofia/internal/1004] has been answered
2015-04-26 19:27:54.204709 [DEBUG] switch_channel.c:3689 (sofia/internal/1004) Callstate Change RINGING -> ACTIVE
2015-04-26 19:27:54.204709 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/1004 PROXY/0 0 ms 160 samples 0 bits 1 channels
2015-04-26 19:27:54.204709 [DEBUG] switch_core_codec.c:111 sofia/internal/1004 Original read codec set to PROXY:0
2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5212 PROXY AUDIO RTP [sofia/internal/1004] 192.168.51.11:5002->192.168.51.11:5002 codec: 0 ms: 20                                                                                                                              
2015-04-26 19:27:54.224752 [DEBUG] switch_rtp.c:3571 Not using a timer
2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5445 Set 2833 dtmf send payload to 101
2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5451 Set 2833 dtmf receive payload to 101
2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5479 Set comfort noise payload to 13
2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:978 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:54.224752 [DEBUG] switch_ivr_originate.c:415 Codec string PROXY at 8000h@20i not supported on sofia/internal/1001 at sip.domain.com, skipping inheritance                                                                                                                   
2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:7687 sofia/internal/1001 at sip.domain.com Patched SDP
---                                                                                                                                        
v=0                                                                                                                                        
o=1004-jitsi.org 0 0 IN IP4 192.168.51.11                                                                                                  
s=-                                                                                                                                        
c=IN IP4 192.168.51.11                                                                                                                     
t=0 0                                                                                                                                      
m=audio 5002 RTP/AVP 8                                                                                                                     
a=rtpmap:8 PCMA/8000                                                                                                                       +++                                                                                                                                        
v=0                                                                                                                                        
o=FreeSWITCH 1055880683 1055880684 IN IP4 YYY.YYY.YYY.YYY                                                                                   
s=FreeSWITCH                                                                                                                               
c=IN IP4 YYY.YYY.YYY.YYY                                                                                                                    
t=0 0                                                                                                                                      
m=audio 20962 RTP/AVP 8                                                                                                                    
a=rtpmap:8 PCMA/8000                                                                                                                       2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/1001 at sip.domain.com PROXY/0 0 ms 160 samples 0 bits 1 channels                                                                                                                                    
2015-04-26 19:27:54.224752 [DEBUG] switch_core_codec.c:111 sofia/internal/1001 at sip.domain.com Original read codec set to PROXY:0
2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5212 PROXY AUDIO RTP [sofia/internal/1001 at sip.domain.com] 192.168.51.103:4002->192.168.51.103:4002 codec: 0 ms: 20                                                                                                              
2015-04-26 19:27:54.224752 [DEBUG] switch_rtp.c:3571 Not using a timer
2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5445 Set 2833 dtmf send payload to 101
2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5451 Set 2833 dtmf receive payload to 101
2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5479 Set comfort noise payload to 13
nua.c:879 nua_respond() nua: nua_respond: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fac680598d0, ...) called
soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fac680598d0, (nil), 0x7faca000cd90, -1) called
nua_stack.c:529 nua_signal() nua(0x7fac680706f0): sent signal r_respond
soa.c:890 soa_set_capability_sdp() soa_set_capability_sdp(static::0x7fac680598d0, (nil), 0x7faca000cd90, -1) called
2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:912 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua_session.c:2320 nua_invite_server_respond() nua: nua_invite_server_respond: entering
soa.c:1515 soa_generate_answer() soa_generate_answer(static::0x7fac680598d0) called
soa_static.c:1146 offer_answer_step() soa_static_offer_answer_action(0x7fac680598d0, soa_generate_answer): called
soa_static.c:1187 offer_answer_step() soa_static(0x7fac680598d0, soa_generate_answer): generating local description
soa_static.c:1228 offer_answer_step() soa_static(0x7fac680598d0, soa_generate_answer): upgrade with remote description
2015-04-26 19:27:54.224752 [INFO] switch_channel.c:3321 sofia/internal/1001 at sip.domain.com ZRTP not negotiated on both sides; disabling ZRTP passthru mode.                                                                                                                            
soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fac7f6ffab0, 0x7fac68078200, ""): called
soa_static.c:1444 offer_answer_step() soa_static(0x7fac680598d0, soa_generate_answer): storing local description
soa.c:1730 soa_activate() soa_activate(static::0x7fac680598d0, (nil)) called
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fac680598d0, [(nil)], [0x7fac7f701c38], [0x7fac7f701c34]) called
tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn = TLS/XXX.XXX.XXX.XXX:22035
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803b270 0x7fac680621a0 884 (884)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803b270 0x7fac68072050 156 (156)
tport.c:3492 tport_send_msg() tport_vsend returned 1040
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
2015-04-26 19:27:54.224752 [NOTICE] switch_ivr_originate.c:3519 Channel [sofia/internal/1001 at sip.domain.com] has been answered
nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (15797)
nta.c:1348 set_timeout() nta: timer shortened to 500 ms
nua_session.c:4139 signal_call_state_change() nua(0x7fac680706f0): call state changed: early -> completed, sent answer
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fac680598d0, [0x7fac7f701ce8], [0x7fac7f701cf0], [(nil)]) called
soa.c:616 soa_get_params() soa_get_params(static::0x7fac680598d0, ...) called
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-04-26 19:27:54.224752 [DEBUG] switch_channel.c:3689 (sofia/internal/1001 at sip.domain.com) Callstate Change RINGING -> ACTIVE
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:54.224752 [DEBUG] sofia.c:6623 Channel sofia/internal/1001 at sip.domain.com entering state [completed][200]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:54.224752 [DEBUG] switch_ivr_originate.c:3577 Originate Resulted in Success: [sofia/internal/1004]
2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:912 Send signal sofia/internal/1004 [BREAK]
2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:912 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-04-26 19:27:54.224752 [DEBUG] switch_ivr_bridge.c:1465 (sofia/internal/1004) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-04-26 19:27:54.224752 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_EXCHANGE_MEDIA
2015-04-26 19:27:54.224752 [DEBUG] switch_core_state_machine.c:538 (sofia/internal/1004) State EXCHANGE_MEDIA
2015-04-26 19:27:54.224752 [DEBUG] mod_sofia.c:594 SOFIA EXCHANGE_MEDIA
tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803aad0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803aad0)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803aad0): tls_read() returned 415
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803aad0) msg 0x7fac68073670 from (tls/XXX.XXX.XXX.XXX:22035) has 415 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fac6803aad0): msg 0x7fac68073670 (415 bytes) from tls/XXX.XXX.XXX.XXX:22035/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received BYE sip:1004 at YYY.YYY.YYY.YYY:9061;transport=tls SIP/2.0 (CSeq 15798)
nta.c:3174 agent_check_request_via() nta: Via check: received=XXX.XXX.XXX.XXX
nta.c:3060 agent_recv_request() nta: BYE (15798) going to existing leg
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-04-26 19:27:54.384702 [NOTICE] sofia.c:952 Hangup sofia/internal/1001 at sip.domain.com [CS_EXECUTE] [NORMAL_CLEARING]
2015-04-26 19:27:54.384702 [DEBUG] switch_channel.c:3222 Send signal sofia/internal/1001 at sip.domain.com [KILL]
2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua.c:879 nua_respond() nua: nua_respond: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fac680598d0, ...) called
tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn = TLS/XXX.XXX.XXX.XXX:22035
nua_stack.c:529 nua_signal() nua(0x7fac680706f0): sent signal r_respond
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:529 nua_signal() nua(0x7fac680706f0): sent signal r_destroy
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803b270 0x7fac68074270 540 (540)
tport.c:3492 tport_send_msg() tport_vsend returned 540
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
nta.c:6791 incoming_reply() nta: sent 200 OK for BYE (15798)
nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7fac680706f0): removing session usage
nua_session.c:4139 signal_call_state_change() nua(0x7fac680706f0): call state changed: completed -> terminated
soa.c:356 soa_destroy() soa_destroy(static::0x7fac680598d0) called
nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7fac68073b20)
nta.c:5744 incoming_free() nta: incoming_free(0x7fac6805c910)
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD DONE [sofia/internal/1001 at sip.domain.com]
2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:690 Send signal sofia/internal/1004 [BREAK]
2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD DONE [sofia/internal/1004]
2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:690 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-04-26 19:27:54.384702 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/internal/1004 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-04-26 19:27:54.384702 [DEBUG] switch_channel.c:3222 Send signal sofia/internal/1004 [KILL]
2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:538 (sofia/internal/1004) State EXCHANGE_MEDIA going to sleep
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_HANGUP
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/1004) Callstate Change ACTIVE -> HANGUP
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/1004) State HANGUP
2015-04-26 19:27:54.384702 [DEBUG] mod_sofia.c:407 sofia/internal/1004 Overriding SIP cause 480 with 200 from the other leg
2015-04-26 19:27:54.384702 [DEBUG] mod_sofia.c:413 Channel sofia/internal/1004 hanging up, cause: NORMAL_CLEARING
2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:1563 sofia/internal/1004 skip receive message [UNBRIDGE] (channel is comngup already)
2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:1566 sofia/internal/1001 at sip.domain.com skip receive message [UNBRIDGE] (channel is comngup already)                                                                                                                             
2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:2901 sofia/internal/1001 at sip.domain.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is comngup already)                                                                                                          
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1001 at sip.domain.com) State EXECUTE going to sleep
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_HANGUP
2015-04-26 19:27:54.384702 [DEBUG] mod_sofia.c:465 Sending BYE to sofia/internal/1004
nua.c:645 nua_bye() nua: nua_bye: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fac6805ba70, ...) called
nua_stack.c:529 nua_signal() nua(0x7fac38005d80): sent signal r_bye
soa.c:1784 soa_terminate() soa_terminate(static::0x7fac6805ba70) called
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7fac6805ba70) called
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1004 Standard HANGUP, cause: NORMAL_CLEARING
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/1004) State HANGUP going to sleep
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004) State Change CS_HANGUP -> CS_REPORTING
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/1001 at sip.domain.com) Callstate Change ACTIVE -> HANGUP
tport.c:4588 tport_by_name() tport(0x7fac680044d0): found 0x7fac6803c380 by name tls/XXX.XXX.XXX.XXX:22036
2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
tport.c:3257 tport_tsend() tport_tsend(0x7fac6803c380) tpn = tls/XXX.XXX.XXX.XXX:22036
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803c670 0x7fac6805b0f0 657 (657)
tport.c:3492 tport_send_msg() tport_vsend returned 657
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
nta.c:8304 outgoing_send() nta: sent BYE (74698568) to tls/XXX.XXX.XXX.XXX:22036
tport.c:4160 tport_pend() tport_pend(0x7fac6803c380): pending 0x7fac6807bb60 for tls/XXX.XXX.XXX.XXX:22036 (already 0)
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/1001 at sip.domain.com) State HANGUP
2015-04-26 19:27:54.384702 [DEBUG] mod_sofia.c:413 Channel sofia/internal/1001 at sip.domain.com hanging up, cause: NORMAL_CLEARING
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_REPORTING
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/1004) State REPORTING
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1001 at sip.domain.com Standard HANGUP, cause: NORMAL_CLEARING
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/1001 at sip.domain.com) State HANGUP going to sleep
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1001 at sip.domain.com) State Change CS_HANGUP -> CS_REPORTING                                                                                                                                         
2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_REPORTING
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/1001 at sip.domain.com) State REPORTING
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1004 Standard REPORTING, cause: NORMAL_CLEARING
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/1004) State REPORTING going to sleep
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1004) State Change CS_REPORTING -> CS_DESTROY
2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1623 Session 6 (sofia/internal/1004) Locked, Waiting on external entities
2015-04-26 19:27:54.384702 [NOTICE] switch_core_session.c:1641 Session 6 (sofia/internal/1004) Ended
2015-04-26 19:27:54.384702 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1004 [CS_DESTROY]
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/1004) Running State Change CS_DESTROY
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/1004) State DESTROY
2015-04-26 19:27:54.384702 [DEBUG] mod_sofia.c:323 sofia/internal/1004 SOFIA DESTROY
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1004 Standard DESTROY
2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/1004) State DESTROY going to sleep
2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1001 at sip.domain.com Standard REPORTING, cause: NORMAL_CLEARING                                                                                                                                       
2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/1001 at sip.domain.com) State REPORTING going to sleep
2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1001 at sip.domain.com) State Change CS_REPORTING -> CS_DESTROY                                                                                                                                        
2015-04-26 19:27:54.404702 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-04-26 19:27:54.404702 [DEBUG] switch_core_session.c:1623 Session 5 (sofia/internal/1001 at sip.domain.com) Locked, Waiting on external entities                                                                                                                                      
2015-04-26 19:27:54.404702 [NOTICE] switch_core_session.c:1641 Session 5 (sofia/internal/1001 at sip.domain.com) Ended
2015-04-26 19:27:54.404702 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1001 at sip.domain.com [CS_DESTROY]
2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/1001 at sip.domain.com) Running State Change CS_DESTROY
2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/1001 at sip.domain.com) State DESTROY
2015-04-26 19:27:54.404702 [DEBUG] mod_sofia.c:323 sofia/internal/1001 at sip.domain.com SOFIA DESTROY
2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1001 at sip.domain.com Standard DESTROY
2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/1001 at sip.domain.com) State DESTROY going to sleep
tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380): tls_read() returned 1
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg 0x7fac680598a0 from (tls/XXX.XXX.XXX.XXX:22036) has 1 bytes, veclen = 1
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380): tls_read() returned 489
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg 0x7fac680598a0 from (tls/XXX.XXX.XXX.XXX:22036) has 489 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fac6803c380): msg 0x7fac680598a0 (490 bytes) from tls/XXX.XXX.XXX.XXX:22036/sips next=(nil)
nta.c:3299 agent_recv_response() nta: received 200 OK for BYE (74698568)
nta.c:3366 agent_recv_response() nta: 200 OK is going to a transaction
nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 62.433 ms
tport.c:4222 tport_release() tport_release(0x7fac6803c380): 0x7fac6807bb60 by 0x7fac6807d640 with 0x7fac680598a0
nua_session.c:4139 signal_call_state_change() nua(0x7fac38005d80): call state changed: terminating -> terminated
nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7fac38005d80): removing session usage
soa.c:356 soa_destroy() soa_destroy(static::0x7fac6805ba70) called
nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7fac680596f0)
nua_session.c:351 nua_session_usage_destroy() nua: terminated session 0x7fac38005d80
nta.c:8722 outgoing_free() nta: outgoing_free(0x7fac6807d640)
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
nua_stack.c:529 nua_signal() nua(0x7fac38005d80): sent signal r_destroy
nta.c:6996 _nta_incoming_timer() nta: timer G fired, retransmitting 200 reply
tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn = TLS/XXX.XXX.XXX.XXX:22035
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803b270 0x7fac680621a0 884 (884)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fac6803b270 0x7fac68072050 156 (156)
tport.c:3492 tport_send_msg() tport_vsend returned 1040
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
nta.c:1296 agent_timer() nta: timer set next to 1000 ms
tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803aad0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803aad0)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803aad0): tls_read() returned 375
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803aad0) msg 0x7fac6805b0f0 from (tls/XXX.XXX.XXX.XXX:22035) has 375 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fac6803aad0): msg 0x7fac6805b0f0 (375 bytes) from tls/XXX.XXX.XXX.XXX:22035/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received ACK sip:1004 at YYY.YYY.YYY.YYY:9061;transport=tls SIP/2.0 (CSeq 15797)
nta.c:3174 agent_check_request_via() nta: Via check: received=XXX.XXX.XXX.XXX
nta.c:3019 agent_recv_request() nta: ACK (15797) is going to INVITE (15797)
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fac7f701c60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 0/0 term, 1/1 free
nta.c:1296 agent_timer() nta: timer set next to 30478 ms
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From aqsyounas at gmail.com  Tue May 12 17:15:28 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 12 May 2015 06:15:28 -0700
Subject: [Freeswitch-users] Re-write of mod_vlc
Message-ID: <CAGVOjVVyMjbss+QZ52d1B8q+W0eYLOK9GdbS-WhT7wSyz8ha4A@mail.gmail.com>

Hi,

We are using mod_vlc for quite a long time for playing streams(urls) for
our radio application, which have more than 250 concurrent listeners.

Currently we are facing multiple issues while playing non mp3 streams (aac,
flv).

i) Freeswitch starts pooling ram untill our server is crashed due to ram
starving or we manually restart our freeswitch to release ram.So, most of
the time we manually restart freeswitch instances.
If we play only mp3 streams with mod_shout. Freeswitch ram pooling problem
is resolved. But we want to play streams other than mp3, that's why we are
using mod_vlc.

ii) AGC (Auto gain controller) some streams are higher in volume than
others.There are more than 1000 stream, daily new streams are being added
and old one deleted. So, we can't manually adjust their volumes by setting
channel variables.

iii) Couldn't play anything while mod_vlc is trying to buffer the stream.
(Please wait while we are trying to connect your requested radio)

So, I am here to ask is there any chance of re writing this module. Or any
modification in up coming freeswitch releases.


We are using FreeSWITCH Version 1.4.18~64bit ( 64bit) on Linux 66-226-75-55
3.2.0-4-amd64 #1 SMP Debian 3.2.63-2+deb7u2 x86_64 GNU/Linux

Your replies would be much appreciated.

Thanks.
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From mike at jerris.com  Tue May 12 18:51:03 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 May 2015 10:51:03 -0400
Subject: [Freeswitch-users] Re-write of mod_vlc
In-Reply-To: <CAGVOjVVyMjbss+QZ52d1B8q+W0eYLOK9GdbS-WhT7wSyz8ha4A@mail.gmail.com>
References: <CAGVOjVVyMjbss+QZ52d1B8q+W0eYLOK9GdbS-WhT7wSyz8ha4A@mail.gmail.com>
Message-ID: <CAEwTsAj-6OSQimaV605tkDqL7r+zcgj=dKBJ3=D0T9+j7m9yTg@mail.gmail.com>

There are major changes to that module coming in 1.6, but I don't think any
of your issues are covered in those.  You could test it out and if not, get
some jiras filed with the issues.

That being said, 1 is likely a vlc not a FreeSWITCH bug, you should test
under valgrind and possibly report this bug to vlc instead of us if they
are leaking.

2. I think vlc already has this feature.


On Tuesday, May 12, 2015, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi,
>
> We are using mod_vlc for quite a long time for playing streams(urls) for
> our radio application, which have more than 250 concurrent listeners.
>
> Currently we are facing multiple issues while playing non mp3 streams
> (aac, flv).
>
> i) Freeswitch starts pooling ram untill our server is crashed due to ram
> starving or we manually restart our freeswitch to release ram.So, most of
> the time we manually restart freeswitch instances.
> If we play only mp3 streams with mod_shout. Freeswitch ram pooling problem
> is resolved. But we want to play streams other than mp3, that's why we are
> using mod_vlc.
>
> ii) AGC (Auto gain controller) some streams are higher in volume than
> others.There are more than 1000 stream, daily new streams are being added
> and old one deleted. So, we can't manually adjust their volumes by setting
> channel variables.
>
> iii) Couldn't play anything while mod_vlc is trying to buffer the stream.
> (Please wait while we are trying to connect your requested radio)
>
> So, I am here to ask is there any chance of re writing this module. Or any
> modification in up coming freeswitch releases.
>
>
> We are using FreeSWITCH Version 1.4.18~64bit ( 64bit) on Linux
> 66-226-75-55 3.2.0-4-amd64 #1 SMP Debian 3.2.63-2+deb7u2 x86_64 GNU/Linux
>
> Your replies would be much appreciated.
>
> Thanks.
>
>
>
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From brian at freeswitch.org  Tue May 12 19:49:25 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 May 2015 10:49:25 -0500
Subject: [Freeswitch-users] FreeSWITCH PROXY:0
In-Reply-To: <freemail.20150512150333.44759.1@fmxmldata05.freemail.hu>
References: <freemail.20150512150333.44759.1@fmxmldata05.freemail.hu>
Message-ID: <CAEJMVkC2fnJ+vEdFPuBr_7DAqW1fqbMn02yVoM+E0K27Y+Qmnw@mail.gmail.com>

Its either you've enabled proxy media mode, or auto proxy due to the
zrtp-hash in the sdp's hard to tell 100% but I suspect thats what is taking
place.

On Tue, May 12, 2015 at 8:03 AM, Asd <asdfgh1234 at freemail.hu> wrote:

> Hi all
>
> Freeswitch version: 1.4.18 Debian stable
>
> with TLS
>
> if i try to call internal:
>
> RINGING <<== OK
> by receive of call <<== NORMAL_CLEARING then KILL
>
> with version 1.4.12 worked
>
> what  am i doing wrong?
>
> my log file:
>
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803aad0): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803aad0)
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803aad0):
> tls_read() returned 1143
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803aad0) msg
> 0x7fac6805d660 from (tls/XXX.XXX.XXX.XXX:22035) has 1143 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7fac6803aad0): msg
> 0x7fac6805d660 (1143 bytes) from tls/XXX.XXX.XXX.XXX:22035/sips next=(nil)
> nta.c:2880 agent_recv_request() nta: received INVITE
> sips:1004 at sip.domain.com:9061 SIP/2.0 (CSeq 15796)
> nta.c:3174 agent_check_request_via() nta: Via check:
> received=XXX.XXX.XXX.XXX
> nta.c:3085 agent_recv_request() nta: INVITE (15796) going to a default leg
> nta.c:1348 set_timeout() nta: timer shortened to 2000 ms
> nua_server.c:102 nua_stack_process_request() nua:
> nua_stack_process_request: entering
> nua_stack.c:899 nh_create() nua: nh_create: entering
> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:280 soa_clone() soa_clone(static::0x7fac68002100, 0x7fac680015a0,
> 0x7fac68077650) called
> soa.c:403 soa_set_params() soa_set_params(static::0x7fac68073910, ...)
> called
> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fac680711d0)
> soa.c:1302 soa_init_offer_answer()
> soa_init_offer_answer(static::0x7fac68073910) called
> soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fac68073910,
> (nil), 0x7fac68062468, 431) called
> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fac68077650): adding
> session usage
> tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn =
> TLS/XXX.XXX.XXX.XXX:22035
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803b270 0x7fac6804e0f0 385 (385)
> tport.c:3492 tport_send_msg() tport_vsend returned 385
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (15796)
> nua_session.c:4139 signal_call_state_change() nua(0x7fac68077650): call
> state changed: init -> received, received offer
> soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fac68073910,
> [0x7fac7f7018b8], [0x7fac7f7018c0], [(nil)]) called
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
> 2015-04-26 19:27:42.024708 [NOTICE] switch_channel.c:1055 New Channel
> sofia/internal/1001 at sip.domain.com [b847625c-43a8-4ece-9f12-9f7b5e84be32]
>
>
> 2015-04-26 19:27:42.024708 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:42.024708 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> 2015-04-26 19:27:42.024708 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1001 at sip.domain.com) Running State Change CS_NEW
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:42.024708 [DEBUG] sofia.c:8844 sofia/internal/
> 1001 at sip.domain.com receiving invite from XXX.XXX.XXX.XXX:22035 version:
> 1.4.18 -3-1 64bit
>
> 2015-04-26 19:27:42.024708 [DEBUG] switch_core_media.c:344 Found audio
> zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10
> b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe
>
>
> 2015-04-26 19:27:42.024708 [DEBUG] sofia.c:9011 IP XXX.XXX.XXX.XXX
> Rejected by acl "domains". Falling back to Digest auth.
> nua.c:879 nua_respond() nua: nua_respond: entering
> nua_stack.c:529 nua_signal() nua(0x7fac68077650): sent signal r_respond
> 2015-04-26 19:27:42.024708 [WARNING] sofia_reg.c:1742 SIP auth challenge
> (INVITE) on sofia profile 'internal' for [1004 at sip.domain.com] from ip
> XXX.XXX.XXX.XXX
>
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> 2015-04-26 19:27:42.024708 [DEBUG] switch_core_state_machine.c:491
> (sofia/internal/1001 at sip.domain.com) State NEW
> soa.c:403 soa_set_params() soa_set_params(static::0x7fac68073910, ...)
> called
> nua_session.c:2320 nua_invite_server_respond() nua:
> nua_invite_server_respond: entering
> soa.c:1214 soa_clear_remote_sdp()
> soa_clear_remote_sdp(static::0x7fac68073910) called
> tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn =
> TLS/XXX.XXX.XXX.XXX:22035
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803b270 0x7fac6805a8a0 889 (889)
> tport.c:3492 tport_send_msg() tport_vsend returned 889
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> nta.c:6791 incoming_reply() nta: sent 407 Proxy Authentication Required
> for INVITE (15796)
> nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7fac68077650):
> removing session usage
> nua_session.c:4139 signal_call_state_change() nua(0x7fac68077650): call
> state changed: received -> terminated
> soa.c:356 soa_destroy() soa_destroy(static::0x7fac68073910) called
> nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7fac680711d0)
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:42.024708 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:42.024708 [DEBUG] sofia.c:2065 detaching session
> b847625c-43a8-4ece-9f12-9f7b5e84be32
> nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
> nua_stack.c:529 nua_signal() nua(0x7fac68077650): sent signal r_destroy
> nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803aad0): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803aad0)
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803aad0):
> tls_read() returned 412
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803aad0) msg
> 0x7fac6804e0f0 from (tls/XXX.XXX.XXX.XXX:22035) has 412 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7fac6803aad0): msg
> 0x7fac6804e0f0 (412 bytes) from tls/XXX.XXX.XXX.XXX:22035/sips next=(nil)
> nta.c:2880 agent_recv_request() nta: received ACK
> sips:1004 at sip.domain.com:9061 SIP/2.0 (CSeq 15796)
> nta.c:3174 agent_check_request_via() nta: Via check:
> received=XXX.XXX.XXX.XXX
> nta.c:3019 agent_recv_request() nta: ACK (15796) is going to INVITE (15796)
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803aad0): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803aad0)
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803aad0):
> tls_read() returned 1419
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803aad0) msg
> 0x7fac6804e0f0 from (tls/XXX.XXX.XXX.XXX:22035) has 1419 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7fac6803aad0): msg
> 0x7fac6804e0f0 (1419 bytes) from tls/XXX.XXX.XXX.XXX:22035/sips next=(nil)
> nta.c:2880 agent_recv_request() nta: received INVITE
> sips:1004 at sip.domain.com:9061 SIP/2.0 (CSeq 15797)
> nta.c:3174 agent_check_request_via() nta: Via check:
> received=XXX.XXX.XXX.XXX
> nta.c:3085 agent_recv_request() nta: INVITE (15797) going to a default leg
> nua_server.c:102 nua_stack_process_request() nua:
> nua_stack_process_request: entering
> nua_stack.c:899 nh_create() nua: nh_create: entering
> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:280 soa_clone() soa_clone(static::0x7fac68002100, 0x7fac680015a0,
> 0x7fac680706f0) called
> soa.c:403 soa_set_params() soa_set_params(static::0x7fac680598d0, ...)
> called
> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fac68073b20)
> soa.c:1302 soa_init_offer_answer()
> soa_init_offer_answer(static::0x7fac680598d0) called
> soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fac680598d0,
> (nil), 0x7fac6805d1ac, 431) called
> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fac680706f0): adding
> session usage
> tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn =
> TLS/XXX.XXX.XXX.XXX:22035
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803b270 0x7fac6805b1d0 385 (385)
> tport.c:3492 tport_send_msg() tport_vsend returned 385
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (15797)
> nua_session.c:4139 signal_call_state_change() nua(0x7fac680706f0): call
> state changed: init -> received, received offer
> soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fac680598d0,
> [0x7fac7f7018b8], [0x7fac7f7018c0], [(nil)]) called
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
> 2015-04-26 19:27:42.204725 [DEBUG] sofia.c:2173 Re-attaching to session
> b847625c-43a8-4ece-9f12-9f7b5e84be32
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:42.204725 [DEBUG] sofia.c:8844 sofia/internal/
> 1001 at sip.domain.com receiving invite from XXX.XXX.XXX.XXX:22035 version:
> 1.4.18 -3-1 64bit
>
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_media.c:344 Found audio
> zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10
> b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe
>
>
> 2015-04-26 19:27:42.204725 [DEBUG] sofia.c:9011 IP XXX.XXX.XXX.XXX
> Rejected by acl "domains". Falling back to Digest auth.
> nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
> nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil)
> 2015-04-26 19:27:42.204725 [DEBUG] sofia.c:10109 Setting NAT mode based on
> via received
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:42.204725 [DEBUG] sofia.c:6623 Channel sofia/internal/
> 1001 at sip.domain.com entering state [received][100]
> 2015-04-26 19:27:42.204725 [DEBUG] sofia.c:6633 Remote SDP:
> v=0
>
> o=- 3639058062 3639058062 IN IP4 192.168.51.103
>
> s=pjmedia
>
> c=IN IP4 192.168.51.103
>
> t=0 0
>
> m=audio 4002 RTP/AVP 99 0 8 101
>
> c=IN IP4 192.168.51.103
>
> a=rtpmap:99 SILK/24000
>
> a=fmtp:99 useinbandfec=0
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=rtcp:4003 IN IP4 192.168.51.103
>
> a=zrtp-hash:1.10
> b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe
>
> 2015-04-26 19:27:42.204725 [DEBUG] sofia.c:6893 (sofia/internal/
> 1001 at sip.domain.com) State Change CS_NEW -> CS_INIT
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1001 at sip.domain.com) Running State Change CS_INIT
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:512
> (sofia/internal/1001 at sip.domain.com) State INIT
> 2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:87 sofia/internal/
> 1001 at sip.domain.com SOFIA INIT
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:40
> sofia/internal/1001 at sip.domain.com Standard INIT
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:48
> (sofia/internal/1001 at sip.domain.com) State Change CS_INIT -> CS_ROUTING
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:512
> (sofia/internal/1001 at sip.domain.com) State INIT going to sleep
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1001 at sip.domain.com) Running State Change CS_ROUTING
> 2015-04-26 19:27:42.204725 [DEBUG] switch_channel.c:2184 (sofia/internal/
> 1001 at sip.domain.com) Callstate Change DOWN -> RINGING
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:528
> (sofia/internal/1001 at sip.domain.com) State ROUTING
> 2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:123 sofia/internal/
> 1001 at sip.domain.com SOFIA ROUTING
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:166
> sofia/internal/1001 at sip.domain.com Standard ROUTING
> 2015-04-26 19:27:42.204725 [INFO] mod_dialplan_xml.c:635 Processing 1001
> <1001>->1004 in context sip.domain.com
> Dialplan: sofia/internal/1001 at sip.domain.com parsing
> [sip.domain.com->PHONE-FAX_b_1001_49123456789100] continue=false
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS)
> [PHONE-FAX_b_1001_49123456789100] context(sip.domain.com) =~ /
> sip.domain.com/ break=on-false
>
>
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL)
> [PHONE-FAX_b_1001_49123456789100] destination_number(1004) =~
> /^(1234567e0)$/ break=on-false
>
>
> Dialplan: sofia/internal/1001 at sip.domain.com parsing
> [sip.domain.com->PHONE-FAX_l_1002_49123456789101] continue=false
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS)
> [PHONE-FAX_l_1002_49123456789101] context(sip.domain.com) =~ /
> sip.domain.com/ break=on-false
>
>
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL)
> [PHONE-FAX_l_1002_49123456789101] destination_number(1004) =~
> /^(1234567e1)$/ break=on-false
>
>
> Dialplan: sofia/internal/1001 at sip.domain.com parsing
> [sip.domain.com->PHONE-FAX_m_1003_49123456789101-copy] continue=false
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS)
> [PHONE-FAX_m_1003_49123456789102] context(sip.domain.com) =~ /
> sip.domain.com/ break=on-false
>
>
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL)
> [PHONE-FAX_m_1003_49123456789102] destination_number(1004) =~
> /^(1234567e2)$/ break=on-false
>
> Dialplan: sofia/internal/1001 at sip.domain.com parsing
> [sip.domain.com->internal-voicemail-com] continue=false
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS)
> [internal-voicemail-com] context(sip.domain.com) =~ /sip.domain.com/
> break=on-false
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS)
> [internal-voicemail-com] caller_id_number(1001) =~ /^(((\+|00)49)|1\d{3})$/
> break=on-false
>
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL)
> [internal-voicemail-com] ${user_data(${destination_number}@${domain}
> param vm-enabled)}(false) =~ /true/ break=on-false
>
>
> Dialplan: sofia/internal/1001 at sip.domain.com parsing
> [sip.domain.com->internal-voicemail-com2] continue=false
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS)
> [internal-voicemail-com2] context(sip.domain.com) =~ /sip.domain.com/
> break=on-false
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL)
> [internal-voicemail-com2] ${user_data(${destination_number}@${domain}
> param vm-enabled)}(false) =~ /true/ break=on-false
>
>
> Dialplan: sofia/internal/1001 at sip.domain.com parsing
> [sip.domain.com->internal] continue=false
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal]
> context(sip.domain.com) =~ /sip.domain.com/ break=on-false
> Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal]
> destination_number(1004) =~ /^(1\d{3})$/ break=on-false
> Dialplan: sofia/internal/1001 at sip.domain.com Action
> set(zrtp_enrollment=true)
> Dialplan: sofia/internal/1001 at sip.domain.com Action
> set(hangup_after_bridge=false)
> Dialplan: sofia/internal/1001 at sip.domain.com Action
> set(continue_on_fail=true)
> Dialplan: sofia/internal/1001 at sip.domain.com Action set(intcallid=1004)
> Dialplan: sofia/internal/1001 at sip.domain.com Action
> bridge(sofia/internal/${intcallid}%${domain})
> Dialplan: sofia/internal/1001 at sip.domain.com Action
> set(eml=${user_data(${intcallid}@${domain} param vm-mailto)})
> Dialplan: sofia/internal/1001 at sip.domain.com Action
> set(datetime=${strftime(%Y.%m.%d)} | ${strftime(%W)}. HET |
> ${strftime(%H:%M)})
> Dialplan: sofia/internal/1001 at sip.domain.com Action set(smtp_from=root)
> Dialplan: sofia/internal/1001 at sip.domain.com Action lua(NoAns.lua
> '${originate_disposition}' '${eml}' '${smtp_from}' 'aaa' 'aaa' 'bbb')
>
>
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:216
> (sofia/internal/1001 at sip.domain.com) State Change CS_ROUTING ->
> CS_EXECUTE
>
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:528
> (sofia/internal/1001 at sip.domain.com) State ROUTING going to sleep
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1001 at sip.domain.com) Running State Change CS_EXECUTE
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:535
> (sofia/internal/1001 at sip.domain.com) State EXECUTE
> 2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:178 sofia/internal/
> 1001 at sip.domain.com SOFIA EXECUTE
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:258
> sofia/internal/1001 at sip.domain.com Standard EXECUTE
> EXECUTE sofia/internal/1001 at sip.domain.com set(zrtp_enrollment=true)
> 2015-04-26 19:27:42.204725 [DEBUG] mod_dptools.c:1445 sofia/internal/
> 1001 at sip.domain.com SET [zrtp_enrollment]=[true]
> EXECUTE sofia/internal/1001 at sip.domain.com set(hangup_after_bridge=false)
> 2015-04-26 19:27:42.204725 [DEBUG] mod_dptools.c:1445 sofia/internal/
> 1001 at sip.domain.com SET [hangup_after_bridge]=[false]
> EXECUTE sofia/internal/1001 at sip.domain.com set(continue_on_fail=true)
> 2015-04-26 19:27:42.204725 [DEBUG] mod_dptools.c:1445 sofia/internal/
> 1001 at sip.domain.com SET [continue_on_fail]=[true]
> EXECUTE sofia/internal/1001 at sip.domain.com set(intcallid=1004)
> 2015-04-26 19:27:42.204725 [DEBUG] mod_dptools.c:1445 sofia/internal/
> 1001 at sip.domain.com SET [intcallid]=[1004]
> EXECUTE sofia/internal/1001 at sip.domain.com bridge(sofia/internal/1004%
> sip.domain.com)
> 2015-04-26 19:27:42.204725 [DEBUG] switch_channel.c:1201 sofia/internal/
> 1001 at sip.domain.com EXPORTING[export_vars] [domain_name]=[sip.domain.com]
> to event
>
> 2015-04-26 19:27:42.204725 [DEBUG] switch_ivr_originate.c:2100 Parsing
> global variables
> 2015-04-26 19:27:42.204725 [NOTICE] switch_channel.c:1055 New Channel
> sofia/internal/1004 [f8a9e60a-a497-4363-bed4-039df448d26d]
> 2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:4701 (sofia/internal/1004)
> State Change CS_NEW -> CS_INIT
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1004 [BREAK]
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_media.c:266 Passing a-leg
> remote zrtp-hash (audio) to b-leg
> 2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:4771 [zrtp_passthru]
> Setting a-leg inherit_codec=true
> 2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:4774 [zrtp_passthru]
> Setting b-leg absolute_codec_string='PCMA at 8000h@20i at 64000b'
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1004) Running State Change CS_INIT
> 2015-04-26 19:27:42.204725 [DEBUG] switch_core_state_machine.c:512
> (sofia/internal/1004) State INIT
> 2015-04-26 19:27:42.204725 [DEBUG] mod_sofia.c:87 sofia/internal/1004
> SOFIA INIT
> nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_media.c:7687
> sofia/internal/1004 Patched SDP
> ---
>
> v=0
>
> o=- 3639058062 3639058062 IN IP4 192.168.51.103
>
> s=pjmedia
>
> c=IN IP4 192.168.51.103
>
> t=0 0
>
> m=audio 4002 RTP/AVP 99 0 8 101
>
> c=IN IP4 192.168.51.103
>
> a=rtpmap:99 SILK/24000
>
> a=fmtp:99 useinbandfec=0
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=rtcp:4003 IN IP4 192.168.51.103
>
> a=zrtp-hash:1.10
> b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe
>
> +++
>
> v=0
>
> o=FreeSWITCH 1055498831 1055498832 IN IP4 YYY.YYY.YYY.YYY
>
> s=FreeSWITCH
>
> c=IN IP4 YYY.YYY.YYY.YYY
>
> t=0 0
>
> m=audio 20654 RTP/AVP 99 0 8 101
>
> c=IN IP4 YYY.YYY.YYY.YYY
>
> a=rtpmap:99 SILK/24000
>
> a=fmtp:99 useinbandfec=0
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=rtcp:4003 IN IP4 192.168.51.103
>
> a=zrtp-hash:1.10
> b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe
>
> 2015-04-26 19:27:42.224657 [DEBUG] sofia_glue.c:1203
> sip:1004 at XXX.XXX.XXX.XXX:22036;transport=tls;registering_acc=sip_domain_com
> Setting proxy route to sofia/internal/1004
>
>
> 2015-04-26 19:27:42.224657 [DEBUG] sofia_glue.c:1232 sofia/internal/1004
> sending invite version: 1.4.18 -3-1 64bit
> Local SDP:
>
> v=0
>
> o=FreeSWITCH 1055498831 1055498832 IN IP4 YYY.YYY.YYY.YYY
>
> s=FreeSWITCH
>
> c=IN IP4 YYY.YYY.YYY.YYY
>
> t=0 0
>
> m=audio 20654 RTP/AVP 99 0 8 101
>
> c=IN IP4 YYY.YYY.YYY.YYY
>
> a=rtpmap:99 SILK/24000
>
> a=fmtp:99 useinbandfec=0
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=rtcp:4003 IN IP4 192.168.51.103
>
> a=zrtp-hash:1.10
> b7ef73966e5720fe5736350d2d57cb35460f4add4c6675e8d4221a13c7fc5dbe
>
> nua.c:633 nua_invite() nua: nua_invite: entering
> nua_stack.c:529 nua_signal() nua(0x7fac38005d80): sent signal r_invite
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:40
> sofia/internal/1004 Standard INIT
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:48
> (sofia/internal/1004) State Change CS_INIT -> CS_ROUTING
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1004 [BREAK]
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:512
> (sofia/internal/1004) State INIT going to sleep
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:280 soa_clone() soa_clone(static::0x7fac68002100, 0x7fac680015a0,
> 0x7fac38005d80) called
> soa.c:403 soa_set_params() soa_set_params(static::0x7fac6805ba70, ...)
> called
> soa.c:403 soa_set_params() soa_set_params(static::0x7fac6805ba70, ...)
> called
> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fac6805ba70,
> (nil), 0x7fac3802d316, -1) called
> soa.c:890 soa_set_capability_sdp()
> soa_set_capability_sdp(static::0x7fac6805ba70, (nil), 0x7fac3802d316, -1)
> called
> nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fac38005d80): adding
> session usage
> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fac680596f0)
> soa.c:1302 soa_init_offer_answer()
> soa_init_offer_answer(static::0x7fac6805ba70) called
> soa.c:1426 soa_generate_offer() soa_generate_offer(static::0x7fac6805ba70,
> 0) called
> soa_static.c:1146 offer_answer_step()
> soa_static_offer_answer_action(0x7fac6805ba70, soa_generate_offer): called
> soa_static.c:1187 offer_answer_step() soa_static(0x7fac6805ba70,
> soa_generate_offer): generating local description
> soa_static.c:1215 offer_answer_step() soa_static(0x7fac6805ba70,
> soa_generate_offer): upgrade with local description
> soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fac7f6ffa70,
> (nil), ""): called
> soa_static.c:1444 offer_answer_step() soa_static(0x7fac6805ba70,
> soa_generate_offer): storing local description
> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fac6805ba70,
> [(nil)], [0x7fac7f701bf8], [0x7fac7f701bf4]) called
> nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
> tport.c:4588 tport_by_name() tport(0x7fac680044d0): found 0x7fac6803c380
> by name tls/XXX.XXX.XXX.XXX:22036
> tport.c:3257 tport_tsend() tport_tsend(0x7fac6803c380) tpn =
> tls/XXX.XXX.XXX.XXX:22036
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1004) Running State Change CS_ROUTING
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803c670 0x7fac6807af60 1018 (1018)
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803c670 0x7fac6807a870 90 (90)
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803c670 0x7fac6807bfc0 432 (432)
> tport.c:3492 tport_send_msg() tport_vsend returned 1540
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
> nta.c:8304 outgoing_send() nta: sent INVITE (74698567) to
> tls/XXX.XXX.XXX.XXX:22036
> tport.c:4160 tport_pend() tport_pend(0x7fac6803c380): pending
> 0x7fac68079d20 for tls/XXX.XXX.XXX.XXX:22036 (already 0)
> nua_session.c:4139 signal_call_state_change() nua(0x7fac38005d80): call
> state changed: init -> calling, sent offer
> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fac6805ba70,
> [0x7fac7f701bd8], [0x7fac7f701be0], [(nil)]) called
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1004 [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:42.224657 [DEBUG] sofia.c:6623 Channel
> sofia/internal/1004 entering state [calling][0]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:528
> (sofia/internal/1004) State ROUTING
> 2015-04-26 19:27:42.224657 [DEBUG] mod_sofia.c:123 sofia/internal/1004
> SOFIA ROUTING
> 2015-04-26 19:27:42.224657 [DEBUG] switch_ivr_originate.c:67
> (sofia/internal/1004) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1004 [BREAK]
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:528
> (sofia/internal/1004) State ROUTING going to sleep
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1004) Running State Change CS_CONSUME_MEDIA
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:547
> (sofia/internal/1004) State CONSUME_MEDIA
> 2015-04-26 19:27:42.224657 [DEBUG] switch_core_state_machine.c:547
> (sofia/internal/1004) State CONSUME_MEDIA going to sleep
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380):
> tls_read() returned 1
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg
> 0x7fac6807cc00 from (tls/XXX.XXX.XXX.XXX:22036) has 1 bytes, veclen = 1
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380):
> tls_read() returned 497
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg
> 0x7fac6807cc00 from (tls/XXX.XXX.XXX.XXX:22036) has 497 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7fac6803c380): msg
> 0x7fac6807cc00 (498 bytes) from tls/XXX.XXX.XXX.XXX:22036/sips next=(nil)
> nta.c:3299 agent_recv_response() nta: received 180 Ringing for INVITE
> (74698567)
> nta.c:3366 agent_recv_response() nta: 180 Ringing is going to a transaction
> nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 93.106 ms
> tport.c:4222 tport_release() tport_release(0x7fac6803c380): 0x7fac68079d20
> by 0x7fac6807c180 with 0x7fac6807cc00 (preliminary)
> nua_session.c:4139 signal_call_state_change() nua(0x7fac38005d80): call
> state changed: calling -> proceeding
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:42.304700 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1004 [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:42.304700 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1004 [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:42.304700 [DEBUG] sofia.c:6623 Channel
> sofia/internal/1004 entering state [proceeding][180]
> 2015-04-26 19:27:42.304700 [NOTICE] sofia.c:6725 Ring-Ready
> sofia/internal/1004!
> 2015-04-26 19:27:42.304700 [DEBUG] switch_channel.c:3277
> (sofia/internal/1004) Callstate Change DOWN -> RINGING
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:879 nua_respond() nua: nua_respond: entering
> nua_stack.c:529 nua_signal() nua(0x7fac680706f0): sent signal r_respond
> 2015-04-26 19:27:42.324677 [NOTICE] mod_sofia.c:2107 Ring-Ready
> sofia/internal/1001 at sip.domain.com!
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fac680598d0, ...)
> called
> nua_session.c:2320 nua_invite_server_respond() nua:
> nua_invite_server_respond: entering
> tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn =
> TLS/XXX.XXX.XXX.XXX:22035
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803b270 0x7fac6807e4e0 799 (799)
> tport.c:3492 tport_send_msg() tport_vsend returned 799
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> nta.c:6791 incoming_reply() nta: sent 180 Ringing for INVITE (15797)
> nua_session.c:4139 signal_call_state_change() nua(0x7fac680706f0): call
> state changed: received -> early
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:42.324677 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:42.324677 [DEBUG] sofia.c:6623 Channel sofia/internal/
> 1001 at sip.domain.com entering state [early][180]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:42.324677 [DEBUG] switch_core_session.c:912 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> 2015-04-26 19:27:42.324677 [NOTICE] switch_ivr_originate.c:527 Ring Ready
> sofia/internal/1001 at sip.domain.com!
> nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil),
> 0x7fac7f701c60)
> nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout,
> 0/0 term, 1/2 free
> nta.c:1296 agent_timer() nta: timer set next to 2657 ms
> nta.c:9101 outgoing_timer_dk() nta: timer D fired, terminate INVITE
> (74698551)
> nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil),
> 0x7fac7f701d40)
> nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout,
> 1/1 term, 1/3 free
> nta.c:1296 agent_timer() nta: timer set next to 1 ms
> nta.c:8982 outgoing_timer_bf() nta: timer F fired, terminating ACK
> (74698551)
> nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil),
> 0x7fac7f701d40)
> nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout,
> 0/0 term, 1/2 free
> nta.c:1296 agent_timer() nta: timer set next to 55631 ms
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380):
> tls_read() returned 1
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg
> 0x7fac6805d660 from (tls/XXX.XXX.XXX.XXX:22036) has 1 bytes, veclen = 1
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380):
> tls_read() returned 655
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg
> 0x7fac6805d660 from (tls/XXX.XXX.XXX.XXX:22036) has 655 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7fac6803c380): msg
> 0x7fac6805d660 (656 bytes) from tls/XXX.XXX.XXX.XXX:22036/sips next=(nil)
> nta.c:3299 agent_recv_response() nta: received 200 OK for INVITE (74698567)
> nta.c:3366 agent_recv_response() nta: 200 OK is going to a transaction
> tport.c:4222 tport_release() tport_release(0x7fac6803c380): 0x7fac68079d20
> by 0x7fac6807c180 with 0x7fac6805d660
> nta.c:1348 set_timeout() nta: timer shortened to 32000 ms
> soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fac6805ba70,
> (nil), 0x7fac6805aa2e, 130) called
> soa.c:1595 soa_process_answer() soa_process_answer(static::0x7fac6805ba70)
> called
> soa_static.c:1146 offer_answer_step()
> soa_static_offer_answer_action(0x7fac6805ba70, soa_process_answer): called
> soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fac6807b530,
> 0x7fac6806f530, ""): called
> soa_static.c:1302 offer_answer_step() soa_static(0x7fac6805ba70,
> soa_process_answer): upgrade codecs with remote description
> soa_static.c:1444 offer_answer_step() soa_static(0x7fac6805ba70,
> soa_process_answer): storing local description
> soa.c:1730 soa_activate() soa_activate(static::0x7fac6805ba70, (nil))
> called
> nua_session.c:988 nua_session_client_response() nua(0x7fac38005d80):
> INVITE: processed SDP answer in 200 OK
> nua_session.c:4139 signal_call_state_change() nua(0x7fac38005d80): call
> state changed: proceeding -> completing, received answer
> soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fac6805ba70,
> [0x7fac7f7015d8], [0x7fac7f7015e0], [(nil)]) called
> soa.c:616 soa_get_params() soa_get_params(static::0x7fac6805ba70, ...)
> called
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:54.204709 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1004 [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:54.204709 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1004 [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:54.204709 [DEBUG] switch_core_media.c:272 Passing b-leg
> remote zrtp-hash (audio) to a-leg
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:54.204709 [DEBUG] sofia.c:6623 Channel
> sofia/internal/1004 entering state [completing][200]
> 2015-04-26 19:27:54.204709 [DEBUG] sofia.c:6633 Remote SDP:
> v=0
>
> o=1004-jitsi.org 0 0 IN IP4 192.168.51.11
>
> s=-
>
> c=IN IP4 192.168.51.11
>
> t=0 0
>
> m=audio 5002 RTP/AVP 8
>
> a=rtpmap:8 PCMA/8000
>
> nua.c:639 nua_ack() nua: nua_ack: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> nua_stack.c:529 nua_signal() nua(0x7fac38005d80): sent signal r_ack
> soa.c:403 soa_set_params() soa_set_params(static::0x7fac6805ba70, ...)
> called
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> soa.c:1730 soa_activate() soa_activate(static::0x7fac6805ba70, (nil))
> called
> nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
> tport.c:4588 tport_by_name() tport(0x7fac680044d0): found 0x7fac6803c380
> by name tls/XXX.XXX.XXX.XXX:22036
> tport.c:3257 tport_tsend() tport_tsend(0x7fac6803c380) tpn =
> tls/XXX.XXX.XXX.XXX:22036
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803c670 0x7fac68071360 475 (475)
> tport.c:3492 tport_send_msg() tport_vsend returned 475
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
> nta.c:8304 outgoing_send() nta: sent ACK (74698567) to
> tls/XXX.XXX.XXX.XXX:22036
> nua_session.c:4139 signal_call_state_change() nua(0x7fac38005d80): call
> state changed: completing -> ready
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:54.204709 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1004 [BREAK]
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:54.204709 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1004 [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:54.204709 [DEBUG] sofia.c:6623 Channel
> sofia/internal/1004 entering state [ready][200]
> 2015-04-26 19:27:54.204709 [DEBUG] switch_channel.c:3635 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> 2015-04-26 19:27:54.204709 [NOTICE] sofia.c:7446 Channel
> [sofia/internal/1004] has been answered
> 2015-04-26 19:27:54.204709 [DEBUG] switch_channel.c:3689
> (sofia/internal/1004) Callstate Change RINGING -> ACTIVE
> 2015-04-26 19:27:54.204709 [DEBUG] switch_core_media.c:2473 Set Codec
> sofia/internal/1004 PROXY/0 0 ms 160 samples 0 bits 1 channels
> 2015-04-26 19:27:54.204709 [DEBUG] switch_core_codec.c:111
> sofia/internal/1004 Original read codec set to PROXY:0
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5212 PROXY AUDIO
> RTP [sofia/internal/1004] 192.168.51.11:5002->192.168.51.11:5002 codec: 0
> ms: 20
>
> 2015-04-26 19:27:54.224752 [DEBUG] switch_rtp.c:3571 Not using a timer
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5445 Set 2833 dtmf
> send payload to 101
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5451 Set 2833 dtmf
> receive payload to 101
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5479 Set comfort
> noise payload to 13
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:978 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:54.224752 [DEBUG] switch_ivr_originate.c:415 Codec string
> PROXY at 8000h@20i not supported on sofia/internal/1001 at sip.domain.com,
> skipping inheritance
>
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:7687 sofia/internal/
> 1001 at sip.domain.com Patched SDP
> ---
>
> v=0
>
> o=1004-jitsi.org 0 0 IN IP4 192.168.51.11
>
> s=-
>
> c=IN IP4 192.168.51.11
>
> t=0 0
>
> m=audio 5002 RTP/AVP 8
>
> a=rtpmap:8 PCMA/8000
>
> +++
>
> v=0
>
> o=FreeSWITCH 1055880683 1055880684 IN IP4 YYY.YYY.YYY.YYY
>
> s=FreeSWITCH
>
> c=IN IP4 YYY.YYY.YYY.YYY
>
> t=0 0
>
> m=audio 20962 RTP/AVP 8
>
> a=rtpmap:8 PCMA/8000
>
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:2473 Set Codec
> sofia/internal/1001 at sip.domain.com PROXY/0 0 ms 160 samples 0 bits 1
> channels
>
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_codec.c:111 sofia/internal/
> 1001 at sip.domain.com Original read codec set to PROXY:0
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5212 PROXY AUDIO
> RTP [sofia/internal/1001 at sip.domain.com] 192.168.51.103:4002->
> 192.168.51.103:4002 codec: 0 ms: 20
>
> 2015-04-26 19:27:54.224752 [DEBUG] switch_rtp.c:3571 Not using a timer
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5445 Set 2833 dtmf
> send payload to 101
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5451 Set 2833 dtmf
> receive payload to 101
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_media.c:5479 Set comfort
> noise payload to 13
> nua.c:879 nua_respond() nua: nua_respond: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fac680598d0, ...)
> called
> soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fac680598d0,
> (nil), 0x7faca000cd90, -1) called
> nua_stack.c:529 nua_signal() nua(0x7fac680706f0): sent signal r_respond
> soa.c:890 soa_set_capability_sdp()
> soa_set_capability_sdp(static::0x7fac680598d0, (nil), 0x7faca000cd90, -1)
> called
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:912 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> nua_session.c:2320 nua_invite_server_respond() nua:
> nua_invite_server_respond: entering
> soa.c:1515 soa_generate_answer()
> soa_generate_answer(static::0x7fac680598d0) called
> soa_static.c:1146 offer_answer_step()
> soa_static_offer_answer_action(0x7fac680598d0, soa_generate_answer): called
> soa_static.c:1187 offer_answer_step() soa_static(0x7fac680598d0,
> soa_generate_answer): generating local description
> soa_static.c:1228 offer_answer_step() soa_static(0x7fac680598d0,
> soa_generate_answer): upgrade with remote description
> 2015-04-26 19:27:54.224752 [INFO] switch_channel.c:3321 sofia/internal/
> 1001 at sip.domain.com ZRTP not negotiated on both sides; disabling ZRTP
> passthru mode.
>
> soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fac7f6ffab0,
> 0x7fac68078200, ""): called
> soa_static.c:1444 offer_answer_step() soa_static(0x7fac680598d0,
> soa_generate_answer): storing local description
> soa.c:1730 soa_activate() soa_activate(static::0x7fac680598d0, (nil))
> called
> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fac680598d0,
> [(nil)], [0x7fac7f701c38], [0x7fac7f701c34]) called
> tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn =
> TLS/XXX.XXX.XXX.XXX:22035
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803b270 0x7fac680621a0 884 (884)
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803b270 0x7fac68072050 156 (156)
> tport.c:3492 tport_send_msg() tport_vsend returned 1040
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> 2015-04-26 19:27:54.224752 [NOTICE] switch_ivr_originate.c:3519 Channel
> [sofia/internal/1001 at sip.domain.com] has been answered
> nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (15797)
> nta.c:1348 set_timeout() nta: timer shortened to 500 ms
> nua_session.c:4139 signal_call_state_change() nua(0x7fac680706f0): call
> state changed: early -> completed, sent answer
> soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fac680598d0,
> [0x7fac7f701ce8], [0x7fac7f701cf0], [(nil)]) called
> soa.c:616 soa_get_params() soa_get_params(static::0x7fac680598d0, ...)
> called
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> 2015-04-26 19:27:54.224752 [DEBUG] switch_channel.c:3689 (sofia/internal/
> 1001 at sip.domain.com) Callstate Change RINGING -> ACTIVE
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:54.224752 [DEBUG] sofia.c:6623 Channel sofia/internal/
> 1001 at sip.domain.com entering state [completed][200]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:54.224752 [DEBUG] switch_ivr_originate.c:3577 Originate
> Resulted in Success: [sofia/internal/1004]
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:912 Send signal
> sofia/internal/1004 [BREAK]
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:912 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> 2015-04-26 19:27:54.224752 [DEBUG] switch_ivr_bridge.c:1465
> (sofia/internal/1004) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1004 [BREAK]
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1004) Running State Change CS_EXCHANGE_MEDIA
> 2015-04-26 19:27:54.224752 [DEBUG] switch_core_state_machine.c:538
> (sofia/internal/1004) State EXCHANGE_MEDIA
> 2015-04-26 19:27:54.224752 [DEBUG] mod_sofia.c:594 SOFIA EXCHANGE_MEDIA
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803aad0): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803aad0)
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803aad0):
> tls_read() returned 415
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803aad0) msg
> 0x7fac68073670 from (tls/XXX.XXX.XXX.XXX:22035) has 415 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7fac6803aad0): msg
> 0x7fac68073670 (415 bytes) from tls/XXX.XXX.XXX.XXX:22035/sips next=(nil)
> nta.c:2880 agent_recv_request() nta: received BYE sip:1004 at YYY.YYY.YYY.YYY:9061;transport=tls
> SIP/2.0 (CSeq 15798)
> nta.c:3174 agent_check_request_via() nta: Via check:
> received=XXX.XXX.XXX.XXX
> nta.c:3060 agent_recv_request() nta: BYE (15798) going to existing leg
> nua_server.c:102 nua_stack_process_request() nua:
> nua_stack_process_request: entering
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2015-04-26 19:27:54.384702 [NOTICE] sofia.c:952 Hangup sofia/internal/
> 1001 at sip.domain.com [CS_EXECUTE] [NORMAL_CLEARING]
> 2015-04-26 19:27:54.384702 [DEBUG] switch_channel.c:3222 Send signal
> sofia/internal/1001 at sip.domain.com [KILL]
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> nua.c:879 nua_respond() nua: nua_respond: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fac680598d0, ...)
> called
> tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn =
> TLS/XXX.XXX.XXX.XXX:22035
> nua_stack.c:529 nua_signal() nua(0x7fac680706f0): sent signal r_respond
> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
> nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
> nua_stack.c:529 nua_signal() nua(0x7fac680706f0): sent signal r_destroy
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803b270 0x7fac68074270 540 (540)
> tport.c:3492 tport_send_msg() tport_vsend returned 540
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> nta.c:6791 incoming_reply() nta: sent 200 OK for BYE (15798)
> nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7fac680706f0):
> removing session usage
> nua_session.c:4139 signal_call_state_change() nua(0x7fac680706f0): call
> state changed: completed -> terminated
> soa.c:356 soa_destroy() soa_destroy(static::0x7fac680598d0) called
> nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7fac68073b20)
> nta.c:5744 incoming_free() nta: incoming_free(0x7fac6805c910)
> nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
> 2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD
> DONE [sofia/internal/1001 at sip.domain.com]
> 2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:690 Send signal
> sofia/internal/1004 [BREAK]
> 2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD
> DONE [sofia/internal/1004]
> 2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:690 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> 2015-04-26 19:27:54.384702 [NOTICE] switch_ivr_bridge.c:754 Hangup
> sofia/internal/1004 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2015-04-26 19:27:54.384702 [DEBUG] switch_channel.c:3222 Send signal
> sofia/internal/1004 [KILL]
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1004 [BREAK]
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:538
> (sofia/internal/1004) State EXCHANGE_MEDIA going to sleep
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1004) Running State Change CS_HANGUP
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:735
> (sofia/internal/1004) Callstate Change ACTIVE -> HANGUP
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:737
> (sofia/internal/1004) State HANGUP
> 2015-04-26 19:27:54.384702 [DEBUG] mod_sofia.c:407 sofia/internal/1004
> Overriding SIP cause 480 with 200 from the other leg
> 2015-04-26 19:27:54.384702 [DEBUG] mod_sofia.c:413 Channel
> sofia/internal/1004 hanging up, cause: NORMAL_CLEARING
> 2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:1563
> sofia/internal/1004 skip receive message [UNBRIDGE] (channel is comngup
> already)
> 2015-04-26 19:27:54.384702 [DEBUG] switch_ivr_bridge.c:1566 sofia/internal/
> 1001 at sip.domain.com skip receive message [UNBRIDGE] (channel is comngup
> already)
>
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:2901
> sofia/internal/1001 at sip.domain.com skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is comngup already)
>
>
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:535
> (sofia/internal/1001 at sip.domain.com) State EXECUTE going to sleep
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1001 at sip.domain.com) Running State Change CS_HANGUP
> 2015-04-26 19:27:54.384702 [DEBUG] mod_sofia.c:465 Sending BYE to
> sofia/internal/1004
> nua.c:645 nua_bye() nua: nua_bye: entering
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0x7fac6805ba70, ...)
> called
> nua_stack.c:529 nua_signal() nua(0x7fac38005d80): sent signal r_bye
> soa.c:1784 soa_terminate() soa_terminate(static::0x7fac6805ba70) called
> soa.c:1302 soa_init_offer_answer()
> soa_init_offer_answer(static::0x7fac6805ba70) called
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1004 Standard HANGUP, cause: NORMAL_CLEARING
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:737
> (sofia/internal/1004) State HANGUP going to sleep
> nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/1004) State Change CS_HANGUP -> CS_REPORTING
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:735
> (sofia/internal/1001 at sip.domain.com) Callstate Change ACTIVE -> HANGUP
> tport.c:4588 tport_by_name() tport(0x7fac680044d0): found 0x7fac6803c380
> by name tls/XXX.XXX.XXX.XXX:22036
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1004 [BREAK]
> tport.c:3257 tport_tsend() tport_tsend(0x7fac6803c380) tpn =
> tls/XXX.XXX.XXX.XXX:22036
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803c670 0x7fac6805b0f0 657 (657)
> tport.c:3492 tport_send_msg() tport_vsend returned 657
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
> nta.c:8304 outgoing_send() nta: sent BYE (74698568) to
> tls/XXX.XXX.XXX.XXX:22036
> tport.c:4160 tport_pend() tport_pend(0x7fac6803c380): pending
> 0x7fac6807bb60 for tls/XXX.XXX.XXX.XXX:22036 (already 0)
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:737
> (sofia/internal/1001 at sip.domain.com) State HANGUP
> 2015-04-26 19:27:54.384702 [DEBUG] mod_sofia.c:413 Channel sofia/internal/
> 1001 at sip.domain.com hanging up, cause: NORMAL_CLEARING
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1004) Running State Change CS_REPORTING
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:823
> (sofia/internal/1004) State REPORTING
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/1001 at sip.domain.com Standard HANGUP, cause: NORMAL_CLEARING
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:737
> (sofia/internal/1001 at sip.domain.com) State HANGUP going to sleep
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:504
> (sofia/internal/1001 at sip.domain.com) State Change CS_HANGUP ->
> CS_REPORTING
>
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/1001 at sip.domain.com) Running State Change CS_REPORTING
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:823
> (sofia/internal/1001 at sip.domain.com) State REPORTING
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:104
> sofia/internal/1004 Standard REPORTING, cause: NORMAL_CLEARING
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:823
> (sofia/internal/1004) State REPORTING going to sleep
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:498
> (sofia/internal/1004) State Change CS_REPORTING -> CS_DESTROY
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1004 [BREAK]
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_session.c:1623 Session 6
> (sofia/internal/1004) Locked, Waiting on external entities
> 2015-04-26 19:27:54.384702 [NOTICE] switch_core_session.c:1641 Session 6
> (sofia/internal/1004) Ended
> 2015-04-26 19:27:54.384702 [NOTICE] switch_core_session.c:1645 Close
> Channel sofia/internal/1004 [CS_DESTROY]
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:626
> (sofia/internal/1004) Running State Change CS_DESTROY
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:636
> (sofia/internal/1004) State DESTROY
> 2015-04-26 19:27:54.384702 [DEBUG] mod_sofia.c:323 sofia/internal/1004
> SOFIA DESTROY
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:111
> sofia/internal/1004 Standard DESTROY
> 2015-04-26 19:27:54.384702 [DEBUG] switch_core_state_machine.c:636
> (sofia/internal/1004) State DESTROY going to sleep
> 2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:104
> sofia/internal/1001 at sip.domain.com Standard REPORTING, cause:
> NORMAL_CLEARING
>
> 2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:823
> (sofia/internal/1001 at sip.domain.com) State REPORTING going to sleep
> 2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:498
> (sofia/internal/1001 at sip.domain.com) State Change CS_REPORTING ->
> CS_DESTROY
>
> 2015-04-26 19:27:54.404702 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/internal/1001 at sip.domain.com [BREAK]
> 2015-04-26 19:27:54.404702 [DEBUG] switch_core_session.c:1623 Session 5
> (sofia/internal/1001 at sip.domain.com) Locked, Waiting on external entities
>
>
> 2015-04-26 19:27:54.404702 [NOTICE] switch_core_session.c:1641 Session 5
> (sofia/internal/1001 at sip.domain.com) Ended
> 2015-04-26 19:27:54.404702 [NOTICE] switch_core_session.c:1645 Close
> Channel sofia/internal/1001 at sip.domain.com [CS_DESTROY]
> 2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:626
> (sofia/internal/1001 at sip.domain.com) Running State Change CS_DESTROY
> 2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:636
> (sofia/internal/1001 at sip.domain.com) State DESTROY
> 2015-04-26 19:27:54.404702 [DEBUG] mod_sofia.c:323 sofia/internal/
> 1001 at sip.domain.com SOFIA DESTROY
> 2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:111
> sofia/internal/1001 at sip.domain.com Standard DESTROY
> 2015-04-26 19:27:54.404702 [DEBUG] switch_core_state_machine.c:636
> (sofia/internal/1001 at sip.domain.com) State DESTROY going to sleep
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380):
> tls_read() returned 1
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg
> 0x7fac680598a0 from (tls/XXX.XXX.XXX.XXX:22036) has 1 bytes, veclen = 1
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803c380): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803c380)
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803c380):
> tls_read() returned 489
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803c380) msg
> 0x7fac680598a0 from (tls/XXX.XXX.XXX.XXX:22036) has 489 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7fac6803c380): msg
> 0x7fac680598a0 (490 bytes) from tls/XXX.XXX.XXX.XXX:22036/sips next=(nil)
> nta.c:3299 agent_recv_response() nta: received 200 OK for BYE (74698568)
> nta.c:3366 agent_recv_response() nta: 200 OK is going to a transaction
> nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 62.433 ms
> tport.c:4222 tport_release() tport_release(0x7fac6803c380): 0x7fac6807bb60
> by 0x7fac6807d640 with 0x7fac680598a0
> nua_session.c:4139 signal_call_state_change() nua(0x7fac38005d80): call
> state changed: terminating -> terminated
> nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7fac38005d80):
> removing session usage
> soa.c:356 soa_destroy() soa_destroy(static::0x7fac6805ba70) called
> nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7fac680596f0)
> nua_session.c:351 nua_session_usage_destroy() nua: terminated session
> 0x7fac38005d80
> nta.c:8722 outgoing_free() nta: outgoing_free(0x7fac6807d640)
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803c380): reset timer
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
> nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
> nua_stack.c:529 nua_signal() nua(0x7fac38005d80): sent signal r_destroy
> nta.c:6996 _nta_incoming_timer() nta: timer G fired, retransmitting 200
> reply
> tport.c:3257 tport_tsend() tport_tsend(0x7fac6803aad0) tpn =
> TLS/XXX.XXX.XXX.XXX:22035
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803b270 0x7fac680621a0 884 (884)
> tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
> 0x7fac6803b270 0x7fac68072050 156 (156)
> tport.c:3492 tport_send_msg() tport_vsend returned 1040
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 1/1 resent, 0/1 tout,
> 0/0 term, 0/1 free
> nta.c:1296 agent_timer() nta: timer set next to 1000 ms
> tport.c:2773 tport_wakeup() tport_wakeup(0x7fac6803aad0): events IN
> tport.c:2864 tport_recv_event() tport_recv_event(0x7fac6803aad0)
> tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fac6803aad0):
> tls_read() returned 375
> tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fac6803aad0) msg
> 0x7fac6805b0f0 from (tls/XXX.XXX.XXX.XXX:22035) has 375 bytes, veclen = 1
> tport.c:3023 tport_deliver() tport_deliver(0x7fac6803aad0): msg
> 0x7fac6805b0f0 (375 bytes) from tls/XXX.XXX.XXX.XXX:22035/sips next=(nil)
> nta.c:2880 agent_recv_request() nta: received ACK sip:1004 at YYY.YYY.YYY.YYY:9061;transport=tls
> SIP/2.0 (CSeq 15797)
> nta.c:3174 agent_check_request_via() nta: Via check:
> received=XXX.XXX.XXX.XXX
> nta.c:3019 agent_recv_request() nta: ACK (15797) is going to INVITE (15797)
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> tport.c:2296 tport_set_secondary_timer() tport(0x7fac6803aad0): reset timer
> nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil),
> 0x7fac7f701c60)
> nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout,
> 0/0 term, 1/1 free
> nta.c:1296 agent_timer() nta: timer set next to 30478 ms
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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  Tue May 12 20:22:50 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 12 May 2015 11:22:50 -0500
Subject: [Freeswitch-users] =?iso-8859-1?q?ClueCon_Weekly_-_May_13=2C_2015?=
 =?iso-8859-1?q?_-_Nenad_Corbic_and_Last_mile_troubleshooting_with_Sangoma?=
 =?iso-8859-1?q?_Enterprise_SBC=B9s?=
Message-ID: <D177930A.9DA35%krice@freeswitch.org>

Hey FreeSWITCHers,

We would like to invite you all to join us tomorrow May 13th for ClueCon
Weekly!

This week Nenad Corbic of Sangoma will be joining us with the topic of ?Last
mile troubleshooting with Sangoma Enterprise SBC?s?.

This week, we will discuss how Carriers use CPE equipment to improve their
last mile debugging.
We will provide a quick overview of Sangoma SBC product line and outline
features and scenarios
that can help providers and enterprise keep their VoIP networks running
effectively and sustain high quality of service.

Join us at https://conference.freeswitch.org/verto/ and call 888 to join in
live. Or watch on YouTube @ https://youtu.be/LTxnUPLxaRE

Call starts at 1PM Eastern (1700 GMT) Wed May 13th!

And If you have not registered for ClueCon, Visit ClueCon.com and register
today!

K

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From adam.ben.ayoun1 at gmail.com  Tue May 12 20:59:17 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Tue, 12 May 2015 19:59:17 +0300
Subject: [Freeswitch-users] Improving voice quality
Message-ID: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>

Hi guys,

We are using Freeswitch as a audio "MCU" for WebRTC using mod_conference,
we are currently using mobile clients on both Android and iOS. The voice
quality is good but not as good as Hangouts for example, there are small
cut-offs every now and then that affects the overall experience. We are
trying to understand the reason for that, maybe it's Freeswitch mixing
algorithm, our servers infrastructure, or something else. We did tried
Janus audio conference demo and it was very good (we used chrome on Android
which should use pretty much the same stack). We really need help figuring
this out. I want to first make sure it's not our infrastructure (mainly
network/CPU). We are using virtualized m3.large on AWS (2 vCPUs and 7.5GB
RAM). Maybe there's a certain instance type on a certain provider that is
known to give the best quality? Any ideas on how to go about this?

Thanks,
Adam
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From aqsyounas at gmail.com  Tue May 12 21:25:04 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 12 May 2015 10:25:04 -0700
Subject: [Freeswitch-users] Re-write of mod_vlc
In-Reply-To: <CAEwTsAj-6OSQimaV605tkDqL7r+zcgj=dKBJ3=D0T9+j7m9yTg@mail.gmail.com>
References: <CAGVOjVVyMjbss+QZ52d1B8q+W0eYLOK9GdbS-WhT7wSyz8ha4A@mail.gmail.com>
	<CAEwTsAj-6OSQimaV605tkDqL7r+zcgj=dKBJ3=D0T9+j7m9yTg@mail.gmail.com>
Message-ID: <CAGVOjVU4yjFcGdgjEZu_tPw=K6RiytehsvgOh3pN0-zb6VziaQ@mail.gmail.com>

Thanks micheal for your reply. I will file the query there.

On 12 May 2015 at 07:51, Michael Jerris <mike at jerris.com> wrote:

> There are major changes to that module coming in 1.6, but I don't think
> any of your issues are covered in those.  You could test it out and if not,
> get some jiras filed with the issues.
>
> That being said, 1 is likely a vlc not a FreeSWITCH bug, you should test
> under valgrind and possibly report this bug to vlc instead of us if they
> are leaking.
>
> 2. I think vlc already has this feature.
>
>
> On Tuesday, May 12, 2015, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi,
>>
>> We are using mod_vlc for quite a long time for playing streams(urls) for
>> our radio application, which have more than 250 concurrent listeners.
>>
>> Currently we are facing multiple issues while playing non mp3 streams
>> (aac, flv).
>>
>> i) Freeswitch starts pooling ram untill our server is crashed due to ram
>> starving or we manually restart our freeswitch to release ram.So, most of
>> the time we manually restart freeswitch instances.
>> If we play only mp3 streams with mod_shout. Freeswitch ram pooling
>> problem is resolved. But we want to play streams other than mp3, that's why
>> we are using mod_vlc.
>>
>> ii) AGC (Auto gain controller) some streams are higher in volume than
>> others.There are more than 1000 stream, daily new streams are being added
>> and old one deleted. So, we can't manually adjust their volumes by setting
>> channel variables.
>>
>> iii) Couldn't play anything while mod_vlc is trying to buffer the stream.
>> (Please wait while we are trying to connect your requested radio)
>>
>> So, I am here to ask is there any chance of re writing this module. Or
>> any modification in up coming freeswitch releases.
>>
>>
>> We are using FreeSWITCH Version 1.4.18~64bit ( 64bit) on Linux
>> 66-226-75-55 3.2.0-4-amd64 #1 SMP Debian 3.2.63-2+deb7u2 x86_64 GNU/Linux
>>
>> Your replies would be much appreciated.
>>
>> Thanks.
>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From covici at ccs.covici.com  Tue May 12 21:27:51 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 12 May 2015 13:27:51 -0400
Subject: [Freeswitch-users] Improving voice quality
In-Reply-To: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
References: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
Message-ID: <7444.1431451671@ccs.covici.com>

How about using physical hardware?

Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:

> Hi guys,
> 
> We are using Freeswitch as a audio "MCU" for WebRTC using mod_conference,
> we are currently using mobile clients on both Android and iOS. The voice
> quality is good but not as good as Hangouts for example, there are small
> cut-offs every now and then that affects the overall experience. We are
> trying to understand the reason for that, maybe it's Freeswitch mixing
> algorithm, our servers infrastructure, or something else. We did tried
> Janus audio conference demo and it was very good (we used chrome on Android
> which should use pretty much the same stack). We really need help figuring
> this out. I want to first make sure it's not our infrastructure (mainly
> network/CPU). We are using virtualized m3.large on AWS (2 vCPUs and 7.5GB
> RAM). Maybe there's a certain instance type on a certain provider that is
> known to give the best quality? Any ideas on how to go about this?
> 
> Thanks,
> Adam
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From olegstolyar at gmail.com  Tue May 12 22:04:16 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 12 May 2015 11:04:16 -0700
Subject: [Freeswitch-users] Improving voice quality
In-Reply-To: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
References: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
Message-ID: <CAC8JMOJ2AkJtyFbpz_NmCOeG3YO6mrc=Jkq40_0Yu23_4pieNw@mail.gmail.com>

Have you tried it with non-mobile WebRTC clients?  Is there the same
problem?

I am using mod_conference on FS hosted on AWS m3.xlarge instance and I have
not noticed audio problem.  But my WebRTC clients are not mobile but rather
Chrome on PCs.

On Tue, May 12, 2015 at 9:59 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
wrote:

> Hi guys,
>
> We are using Freeswitch as a audio "MCU" for WebRTC using mod_conference,
> we are currently using mobile clients on both Android and iOS. The voice
> quality is good but not as good as Hangouts for example, there are small
> cut-offs every now and then that affects the overall experience. We are
> trying to understand the reason for that, maybe it's Freeswitch mixing
> algorithm, our servers infrastructure, or something else. We did tried
> Janus audio conference demo and it was very good (we used chrome on Android
> which should use pretty much the same stack). We really need help figuring
> this out. I want to first make sure it's not our infrastructure (mainly
> network/CPU). We are using virtualized m3.large on AWS (2 vCPUs and 7.5GB
> RAM). Maybe there's a certain instance type on a certain provider that is
> known to give the best quality? Any ideas on how to go about this?
>
> Thanks,
> Adam
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From william.king at quentustech.com  Tue May 12 22:15:18 2015
From: william.king at quentustech.com (William King)
Date: Tue, 12 May 2015 11:15:18 -0700
Subject: [Freeswitch-users] mod_hiredis
In-Reply-To: <5551E552.9090307@telefaks.de>
References: <554EB704.5090605@quentustech.com> <5551E552.9090307@telefaks.de>
Message-ID: <55524336.4050208@quentustech.com>

Peter,

The pubsub actions should be available now with the raw app, though this
would work in a blocking manner. You do raise a good use case for
asynchronous commands, specifically for fire and forget cases like pubsub.

For the prefix routing, currently the array return type isn't supported,
but if redis returns a single result that should be able to work and be
tested right now. Can you provide a few sample redis commands and
responses for how you'd setup this scenario?

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 05/12/2015 04:34 AM, Peter Steinbach wrote:
> Hello William,
> 
> this is great, the idea of integrating Redis. We currently use Memcache
> in raw mode as a method of externally controlling dialplans and failover
> scenarios.
> Redis, of course, brings much more features here.
> 
>>Currently the two main use cases are:
>>1. Call per second limits
>>2. Concurrent call limits
>>
>>Possible additional functionality:
>>1. Support for fail-over connections
>>2. Asynchronous commands(is there a use case for this?)
> 
> Another idea for your list would be to route calls according to
> prefixes. You may lookup Redis with a part of the phone number and it
> returns the gateway for this part of the number (redis DB is then
> preloaded from another application).
> And - as Redis has a publish/subscribe method - you will be able to
> publish call informations from the dialplan to multiple external
> subscribers (e.g. announce an incoming call to a CRM) without the use of
> ESL. Is there a chance to run the redis dialplan app in a non blocking
> manner for this scenario, in order to speed up the dialplan?
> 
> 
> 
> Best regards
> Peter
> 
> On 05/10/15 03:40, William King wrote:
>> I'm working on an update Redis integration module that will use the C
>> library hiredis:
>> http://redis.io/clients#c
>> https://github.com/redis/hiredis
>>
>> I've pushed an alpha version of the module to a branch here:
>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits?until=
>> refs%2Fheads%2Fmod_hiredis
>>
>> The current module has a dialplan app and an api for 'hiredis_raw'
>> which allows any single line Redis command, and executes it in a
>> blocking manner, then supports returning string and integer responses.
>>
>> If anyone on this list has any use cases for FreeSWITCH+Redis, please
>> reply to this thread. Currently the two main use cases are:
>> 1. Call per second limits
>> 2. Concurrent call limits
>>
>> Possible additional functionality:
>> 1. Support for fail-over connections
>> 2. Asynchronous commands(is there a use case for this?)
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> 


From tilt at temp3r.com  Tue May 12 22:17:38 2015
From: tilt at temp3r.com (tilt)
Date: Tue, 12 May 2015 11:17:38 -0700
Subject: [Freeswitch-users] freeswitch systemd unit file for debian?
In-Reply-To: <CAOHpEota04EeGXSE9_x2ohq2S4m9GbCMAQpZkUbWVYEvSFY8YQ@mail.gmail.com>
References: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>	<CAC-LwPPtPBmZpYeRpBZqg22KYuDKJDbRhZnhvg9S_HNXtA1oGA@mail.gmail.com>	<A706245D-22A6-48FB-9361-B3317A7F30AA@jerris.com>
	<CAOHpEota04EeGXSE9_x2ohq2S4m9GbCMAQpZkUbWVYEvSFY8YQ@mail.gmail.com>
Message-ID: <555243C2.8040706@temp3r.com>

Hi Joseph, Any chance you can share your service file, I am having 
issues myself trying to get freeswitch running with systemd.

Thank You,
-John

On 05/01/2015 11:07 AM, Joseph Dickson wrote:
> I've got one that works now after discovering that RuntimeDirectory= 
> will create a directory for you under /run.. I'll try to get you the 
> pull request next week
>
> On Fri, May 1, 2015 at 1:56 PM, Michael Jerris <mike at jerris.com 
> <mailto:mike at jerris.com>> wrote:
>
>     If you have a working one, please get a pull request to us so we
>     can review the changes.
>
>     Thanks
>     Mike
>
>>     On May 1, 2015, at 1:40 PM, Vik Killa <vipkilla at gmail.com
>>     <mailto:vipkilla at gmail.com>> wrote:
>>
>>     Hello,
>>     I modified the path variables in the systemd init file.
>>     My file looks like this:
>>
>>     ;;;;; Author: Travis Cross <tc at traviscross.com
>>     <mailto:tc at traviscross.com>>
>>
>>     [Unit]
>>     Description=freeswitch
>>     After=syslog.target network.target local-fs.target
>>
>>     [Service]
>>     ; service
>>     Type=forking
>>     PIDFile=/usr/local/freeswitch/run/freeswitch.pid
>>     PermissionsStartOnly=true
>>     ExecStartPre=/bin/mkdir -p /usr/local/freeswitch/run
>>     ExecStartPre=/bin/chown freeswitch:freeswitch
>>     /usr/local/freeswitch/run
>>     ExecStart=/usr/bin/freeswitch -ncwait -nonat
>>     TimeoutSec=45s
>>     Restart=always
>>     ; exec
>>     WorkingDirectory=/usr/local/freeswitch/run
>>     User=freeswitch
>>     Group=freeswitch
>>     LimitCORE=infinity
>>     LimitNOFILE=100000
>>     LimitNPROC=60000
>>     ;LimitSTACK=240
>>     LimitRTPRIO=infinity
>>     LimitRTTIME=7000000
>>     IOSchedulingClass=realtime
>>     IOSchedulingPriority=2
>>     CPUSchedulingPolicy=rr
>>     CPUSchedulingPriority=89
>>     UMask=0007
>>
>>     [Install]
>>     WantedBy=multi-user.target
>>
>>
>>     On Fri, May 1, 2015 at 12:50 PM, Joseph Dickson
>>     <jdickson at evolvetsi.com <mailto:jdickson at evolvetsi.com>> wrote:
>>
>>         Happy Friday!
>>
>>         I'm having trouble using latest release systemd unit file on
>>         Debian Jessie..  It looks like the unit file is the same in
>>         master, so I imagine the issue exists there too..
>>
>>         On my system (fresh Debian 8 install), I get the following
>>         failure when trying to start using the included unit file:
>>
>>         May 01 12:48:22 XXX systemd[9119]: Failed at step CHDIR
>>         spawning /bin/mkdir: No such file or directory
>>
>>         I'm new to systemd, but it looks like the problem is that the
>>         WorkingDirectory is set to /run/freeswitch.  Trouble is that
>>         /run/freeswitch is created in an ExecStartPre statement. 
>>         That's the best explanation I have for the CHDIR failure that
>>         systemd is complaining about.
>>
>>         It looks like the only way to get /run/freeswitch created
>>         soon enough to be used as a WorkingDirectory is the
>>         tmpfiles.d mechanism that systemd has.  Am I on the right
>>         track, or am I missing an obvious solution?
>>
>>         Thanks!
>>
>>         Joseph Dickson
>>         jdickson at evolvetsi.com <mailto:jdickson at evolvetsi.com>
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>         <http://www.freeswitchsolutions.com/>
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>         http://confluence.freeswitch.org
>>         <http://confluence.freeswitch.org/>
>>         http://www.cluecon.com <http://www.cluecon.com/>
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From simpot at gmail.com  Tue May 12 22:27:44 2015
From: simpot at gmail.com (Dmitry Saratsky)
Date: Tue, 12 May 2015 21:27:44 +0300
Subject: [Freeswitch-users] 503 internal error for gateway calls....
Message-ID: <CAM2iYy6kan4sP6a7RJ+DR38r-e6AS4QQaZapydygrB4JRbwkbQ@mail.gmail.com>

Hi All!

I have very strange problem on one of my switches...

when I calling through 1 specific gateway (gateway's config well be shown
later) - I get internal error with code 503 and freeswitch even do not try
to send invite to the remote gateway...
This gateway have very special configured as shown below (after debug
output) and digest registration. Digest registration is passed fine for me!

part of relevant console debug:
------------------------------------------------------------------
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:40
sofia/ext-ipv4/12121231122 Standard INIT
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:48
(sofia/ext-ipv4/12121231122) State Change CS_INIT -> CS_ROUTING
2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
sofia/ext-ipv4/12121231122 [BREAK]
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:512
(sofia/ext-ipv4/12121231122) State INIT going to sleep
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
(sofia/ext-ipv4/12121231122) Running State Change CS_ROUTING
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:528
(sofia/ext-ipv4/12121231122) State ROUTING
2015-05-12 21:15:36.462865 [DEBUG] mod_sofia.c:123
sofia/ext-ipv4/12121231122 SOFIA ROUTING
2015-05-12 21:15:36.462865 [DEBUG] switch_ivr_originate.c:67
(sofia/ext-ipv4/12121231122) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
sofia/ext-ipv4/12121231122 [BREAK]
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:528
(sofia/ext-ipv4/12121231122) State ROUTING going to sleep
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
(sofia/ext-ipv4/12121231122) Running State Change CS_CONSUME_MEDIA
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:547
(sofia/ext-ipv4/12121231122) State CONSUME_MEDIA
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:547
(sofia/ext-ipv4/12121231122) State CONSUME_MEDIA going to sleep
2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
sofia/ext-ipv4/12121231122 [BREAK]
2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
sofia/ext-ipv4/12121231122 [BREAK]
2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
sofia/ext-ipv4/12121231122 [BREAK]
2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
sofia/ext-ipv4/12121231122 [BREAK]
2015-05-12 21:15:36.462865 [DEBUG] sofia.c:6623 Channel
sofia/ext-ipv4/12121231122 entering state [calling][0]
2015-05-12 21:15:36.462865 [DEBUG] sofia.c:6623 Channel
sofia/ext-ipv4/12121231122 entering state [terminated][503]
2015-05-12 21:15:36.462865 [NOTICE] sofia.c:7539 Hangup
sofia/ext-ipv4/12121231122 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2015-05-12 21:15:36.462865 [DEBUG] switch_channel.c:3222 Send signal
sofia/ext-ipv4/12121231122 [KILL]
2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
sofia/ext-ipv4/12121231122 [BREAK]
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
(sofia/ext-ipv4/12121231122) Running State Change CS_HANGUP
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:735
(sofia/ext-ipv4/12121231122) Callstate Change DOWN -> HANGUP
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:737
(sofia/ext-ipv4/12121231122) State HANGUP
2015-05-12 21:15:36.462865 [DEBUG] mod_sofia.c:413 Channel
sofia/ext-ipv4/12121231122 hanging up, cause: NORMAL_TEMPORARY_FAILURE
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:60
sofia/ext-ipv4/12121231122 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:737
(sofia/ext-ipv4/12121231122) State HANGUP going to sleep
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:504
(sofia/ext-ipv4/12121231122) State Change CS_HANGUP -> CS_REPORTING
2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
sofia/ext-ipv4/12121231122 [BREAK]
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
(sofia/ext-ipv4/12121231122) Running State Change CS_REPORTING
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:823
(sofia/ext-ipv4/12121231122) State REPORTING
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:104
sofia/ext-ipv4/12121231122 Standard REPORTING, cause:
NORMAL_TEMPORARY_FAILURE
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:823
(sofia/ext-ipv4/12121231122) State REPORTING going to sleep
2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:498
(sofia/ext-ipv4/12121231122) State Change CS_REPORTING -> CS_DESTROY
2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
sofia/ext-ipv4/12121231122 [BREAK]
2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1623 Session 4
(sofia/ext-ipv4/12121231122) Locked, Waiting on external entities
2015-05-12 21:15:36.462865 [DEBUG] switch_ivr_originate.c:3720 Originate
Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]
2015-05-12 21:15:36.462865 [INFO] mod_dptools.c:3244 Originate Failed.
Cause: NORMAL_TEMPORARY_FAILURE
------------------------------------------------------------------

no packet was sent to remote gateway (checked with tcpdump), so it is
internal error...
relevant gateway is registered well:
------------------------------------------------------------------
=================================================================================================
Name            gateway.name
Profile         ext-ipv4
Scheme          Digest
Realm           realm.gateway.name
Username        +12223334455
Password        yes
>From            <sip:+12223334455 at realm.gateway.name>
Contact         <sip:+12223334455 at 1.1.1.1:5060
;expires=3600;+g.oma.sip-im;+g.oma.sip-im.large-message;transport=udp;gw=
gateway.name>
Exten           +12223334455
To              sip:+12223334455 at realm.gateway.name
Proxy           sip:realm.gateway.name
Context         ext
Expires         3600
Freq            3600
Ping            0
PingFreq        0
PingTime        0.00
PingState       0/0/0
State           REGED
Status          UP
Uptime          554s
CallsIN         0
CallsOUT        2
FailedCallsIN   0
FailedCallsOUT  2
=================================================================================================
------------------------------------------------------------------

gateway config:
------------------------------------------------------------------
<include>
  <gateway name="gateway.name">
    <param name="proxy" value="realm.gateway.name"/>
    <param name="register-proxy" value="10.10.10.10"/>
    <param name="contact-params"
value="expires=3600;+g.oma.sip-im;+g.oma.sip-im.large-message"/>
    <param name="username" value="+12223334455"/>
    <param name="auth-username" value="+12223334455 at realm.gateway.name"/>
    <param name="from-user" value="+12223334455"/>
    <param name="from-domain" value="realm.gateway.name"/>
    <param name="password" value="password"/>
    <param name="expire-seconds" value="3600"/>
    <param name="register" value="true"/>
    <param name="register-transport" value="udp"/>
    <param name="caller-id-in-from" value="false"/>
    <param name="extension-in-contact" value="+12223334455"/>
    <param name="retry-seconds" value="180"/>
  </gateway>
</include>

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

what is special in this gateway configuration, is that gateway name is
differers from proxy and both differs from register-proxy. this is required
for my for successfully registration (bcz realm.gateway.name is not
resolvable with DNS).
In addition username is differs from auth-username - for some reason it is
required by vendor...

PS:
FreeSWITCH (Version 1.5.15b git a41505f 2015-02-23 22:38:20Z 64bit) is ready


Any ideas?

Thanks,
Dmitry.
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From jdickson at evolvetsi.com  Tue May 12 22:39:22 2015
From: jdickson at evolvetsi.com (Joseph Dickson)
Date: Tue, 12 May 2015 14:39:22 -0400
Subject: [Freeswitch-users] freeswitch systemd unit file for debian?
In-Reply-To: <555243C2.8040706@temp3r.com>
References: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>
	<CAC-LwPPtPBmZpYeRpBZqg22KYuDKJDbRhZnhvg9S_HNXtA1oGA@mail.gmail.com>
	<A706245D-22A6-48FB-9361-B3317A7F30AA@jerris.com>
	<CAOHpEota04EeGXSE9_x2ohq2S4m9GbCMAQpZkUbWVYEvSFY8YQ@mail.gmail.com>
	<555243C2.8040706@temp3r.com>
Message-ID: <CAOHpEotuq7LhSmU0rZUw7d0tAD+TO3JNbbh6GOcevbNvnVGwLA@mail.gmail.com>

John,

This is what I ended up with that works.. I've been busy and haven't been
able to get all the stuff in place to put in a pull request :-/  This still
requires you to link binaries to /usr/bin from wherever you compile your
Freeswitch to..

[Unit]
Description=freeswitch
After=syslog.target network.target local-fs.target

[Service]
; service
Type=forking
RuntimeDirectory=freeswitch
PIDFile=/run/freeswitch/freeswitch.pid
PermissionsStartOnly=true
ExecStart=/usr/bin/freeswitch -ncwait -nonat -run /run/freeswitch
TimeoutSec=45s
Restart=always
; exec
WorkingDirectory=/run/freeswitch
User=freeswitch
Group=daemon
LimitCORE=infinity
LimitNOFILE=100000
LimitNPROC=60000
;LimitSTACK=240
LimitRTPRIO=infinity
LimitRTTIME=7000000
IOSchedulingClass=realtime
IOSchedulingPriority=2
CPUSchedulingPolicy=rr
CPUSchedulingPriority=89
UMask=0007

[Install]
WantedBy=multi-user.target


Cheers,

Joe

On Tue, May 12, 2015 at 2:17 PM, tilt <tilt at temp3r.com> wrote:

>  Hi Joseph, Any chance you can share your service file, I am having issues
> myself trying to get freeswitch running with systemd.
>
> Thank You,
> -John
>
>
> On 05/01/2015 11:07 AM, Joseph Dickson wrote:
>
> I've got one that works now after discovering that RuntimeDirectory= will
> create a directory for you under /run..  I'll try to get you the pull
> request next week
>
> On Fri, May 1, 2015 at 1:56 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> If you have a working one, please get a pull request to us so we can
>> review the changes.
>>
>>  Thanks
>> Mike
>>
>>  On May 1, 2015, at 1:40 PM, Vik Killa <vipkilla at gmail.com> wrote:
>>
>>  Hello,
>> I modified the path variables in the systemd init file.
>> My file looks like this:
>>
>> ;;;;; Author: Travis Cross <tc at traviscross.com>
>>
>>  [Unit]
>> Description=freeswitch
>> After=syslog.target network.target local-fs.target
>>
>>  [Service]
>> ; service
>> Type=forking
>> PIDFile=/usr/local/freeswitch/run/freeswitch.pid
>> PermissionsStartOnly=true
>> ExecStartPre=/bin/mkdir -p /usr/local/freeswitch/run
>> ExecStartPre=/bin/chown freeswitch:freeswitch /usr/local/freeswitch/run
>> ExecStart=/usr/bin/freeswitch -ncwait -nonat
>> TimeoutSec=45s
>> Restart=always
>> ; exec
>> WorkingDirectory=/usr/local/freeswitch/run
>> User=freeswitch
>> Group=freeswitch
>> LimitCORE=infinity
>> LimitNOFILE=100000
>> LimitNPROC=60000
>> ;LimitSTACK=240
>> LimitRTPRIO=infinity
>> LimitRTTIME=7000000
>> IOSchedulingClass=realtime
>> IOSchedulingPriority=2
>> CPUSchedulingPolicy=rr
>> CPUSchedulingPriority=89
>> UMask=0007
>>
>>  [Install]
>> WantedBy=multi-user.target
>>
>>
>> On Fri, May 1, 2015 at 12:50 PM, Joseph Dickson <jdickson at evolvetsi.com>
>> wrote:
>>
>>> Happy Friday!
>>>
>>>  I'm having trouble using latest release systemd unit file on Debian
>>> Jessie..  It looks like the unit file is the same in master, so I imagine
>>> the issue exists there too..
>>>
>>>  On my system (fresh Debian 8 install), I get the following failure
>>> when trying to start using the included unit file:
>>>
>>>  May 01 12:48:22 XXX systemd[9119]: Failed at step CHDIR spawning
>>> /bin/mkdir: No such file or directory
>>>
>>>  I'm new to systemd, but it looks like the problem is that the
>>> WorkingDirectory is set to /run/freeswitch.  Trouble is that
>>> /run/freeswitch is created in an ExecStartPre statement.  That's the best
>>> explanation I have for the CHDIR failure that systemd is complaining about.
>>>
>>>  It looks like the only way to get /run/freeswitch created soon enough
>>> to be used as a WorkingDirectory is the tmpfiles.d mechanism that systemd
>>> has.  Am I on the right track, or am I missing an obvious solution?
>>>
>>>  Thanks!
>>>
>>>    Joseph Dickson
>>> jdickson at evolvetsi.com
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>  _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.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 covici at ccs.covici.com  Tue May 12 22:58:44 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 12 May 2015 14:58:44 -0400
Subject: [Freeswitch-users] freeswitch systemd unit file for debian?
In-Reply-To: <CAOHpEotuq7LhSmU0rZUw7d0tAD+TO3JNbbh6GOcevbNvnVGwLA@mail.gmail.com>
References: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>
	<CAC-LwPPtPBmZpYeRpBZqg22KYuDKJDbRhZnhvg9S_HNXtA1oGA@mail.gmail.com>
	<A706245D-22A6-48FB-9361-B3317A7F30AA@jerris.com>
	<CAOHpEota04EeGXSE9_x2ohq2S4m9GbCMAQpZkUbWVYEvSFY8YQ@mail.gmail.com>
	<555243C2.8040706@temp3r.com>
	<CAOHpEotuq7LhSmU0rZUw7d0tAD+TO3JNbbh6GOcevbNvnVGwLA@mail.gmail.com>
Message-ID: <10167.1431457124@ccs.covici.com>

I could never get type=forking to function, I wonder how you got that to
work, I had to say -nf and use type=simple.

Joseph Dickson <jdickson at evolvetsi.com> wrote:

> John,
> 
> This is what I ended up with that works.. I've been busy and haven't been
> able to get all the stuff in place to put in a pull request :-/  This still
> requires you to link binaries to /usr/bin from wherever you compile your
> Freeswitch to..
> 
> [Unit]
> Description=freeswitch
> After=syslog.target network.target local-fs.target
> 
> [Service]
> ; service
> Type=forking
> RuntimeDirectory=freeswitch
> PIDFile=/run/freeswitch/freeswitch.pid
> PermissionsStartOnly=true
> ExecStart=/usr/bin/freeswitch -ncwait -nonat -run /run/freeswitch
> TimeoutSec=45s
> Restart=always
> ; exec
> WorkingDirectory=/run/freeswitch
> User=freeswitch
> Group=daemon
> LimitCORE=infinity
> LimitNOFILE=100000
> LimitNPROC=60000
> ;LimitSTACK=240
> LimitRTPRIO=infinity
> LimitRTTIME=7000000
> IOSchedulingClass=realtime
> IOSchedulingPriority=2
> CPUSchedulingPolicy=rr
> CPUSchedulingPriority=89
> UMask=0007
> 
> [Install]
> WantedBy=multi-user.target
> 
> 
> Cheers,
> 
> Joe
> 
> On Tue, May 12, 2015 at 2:17 PM, tilt <tilt at temp3r.com> wrote:
> 
> >  Hi Joseph, Any chance you can share your service file, I am having issues
> > myself trying to get freeswitch running with systemd.
> >
> > Thank You,
> > -John
> >
> >
> > On 05/01/2015 11:07 AM, Joseph Dickson wrote:
> >
> > I've got one that works now after discovering that RuntimeDirectory= will
> > create a directory for you under /run..  I'll try to get you the pull
> > request next week
> >
> > On Fri, May 1, 2015 at 1:56 PM, Michael Jerris <mike at jerris.com> wrote:
> >
> >> If you have a working one, please get a pull request to us so we can
> >> review the changes.
> >>
> >>  Thanks
> >> Mike
> >>
> >>  On May 1, 2015, at 1:40 PM, Vik Killa <vipkilla at gmail.com> wrote:
> >>
> >>  Hello,
> >> I modified the path variables in the systemd init file.
> >> My file looks like this:
> >>
> >> ;;;;; Author: Travis Cross <tc at traviscross.com>
> >>
> >>  [Unit]
> >> Description=freeswitch
> >> After=syslog.target network.target local-fs.target
> >>
> >>  [Service]
> >> ; service
> >> Type=forking
> >> PIDFile=/usr/local/freeswitch/run/freeswitch.pid
> >> PermissionsStartOnly=true
> >> ExecStartPre=/bin/mkdir -p /usr/local/freeswitch/run
> >> ExecStartPre=/bin/chown freeswitch:freeswitch /usr/local/freeswitch/run
> >> ExecStart=/usr/bin/freeswitch -ncwait -nonat
> >> TimeoutSec=45s
> >> Restart=always
> >> ; exec
> >> WorkingDirectory=/usr/local/freeswitch/run
> >> User=freeswitch
> >> Group=freeswitch
> >> LimitCORE=infinity
> >> LimitNOFILE=100000
> >> LimitNPROC=60000
> >> ;LimitSTACK=240
> >> LimitRTPRIO=infinity
> >> LimitRTTIME=7000000
> >> IOSchedulingClass=realtime
> >> IOSchedulingPriority=2
> >> CPUSchedulingPolicy=rr
> >> CPUSchedulingPriority=89
> >> UMask=0007
> >>
> >>  [Install]
> >> WantedBy=multi-user.target
> >>
> >>
> >> On Fri, May 1, 2015 at 12:50 PM, Joseph Dickson <jdickson at evolvetsi.com>
> >> wrote:
> >>
> >>> Happy Friday!
> >>>
> >>>  I'm having trouble using latest release systemd unit file on Debian
> >>> Jessie..  It looks like the unit file is the same in master, so I imagine
> >>> the issue exists there too..
> >>>
> >>>  On my system (fresh Debian 8 install), I get the following failure
> >>> when trying to start using the included unit file:
> >>>
> >>>  May 01 12:48:22 XXX systemd[9119]: Failed at step CHDIR spawning
> >>> /bin/mkdir: No such file or directory
> >>>
> >>>  I'm new to systemd, but it looks like the problem is that the
> >>> WorkingDirectory is set to /run/freeswitch.  Trouble is that
> >>> /run/freeswitch is created in an ExecStartPre statement.  That's the best
> >>> explanation I have for the CHDIR failure that systemd is complaining about.
> >>>
> >>>  It looks like the only way to get /run/freeswitch created soon enough
> >>> to be used as a WorkingDirectory is the tmpfiles.d mechanism that systemd
> >>> has.  Am I on the right track, or am I missing an obvious solution?
> >>>
> >>>  Thanks!
> >>>
> >>>    Joseph Dickson
> >>> jdickson at evolvetsi.com
> >>>
> >>>
> >>> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>  _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services: consulting at freeswitch.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
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From s.safarov at gmail.com  Tue May 12 22:59:18 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 12 May 2015 21:59:18 +0300
Subject: [Freeswitch-users] 503 internal error for gateway calls....
In-Reply-To: <CAM2iYy6kan4sP6a7RJ+DR38r-e6AS4QQaZapydygrB4JRbwkbQ@mail.gmail.com>
References: <CAM2iYy6kan4sP6a7RJ+DR38r-e6AS4QQaZapydygrB4JRbwkbQ@mail.gmail.com>
Message-ID: <CAHtxdDdQEUUY-HLYcgC_aVyA7QL9=unvjxbxs7Up=a9jA9iAuw@mail.gmail.com>

I has similar error message on cisco gateway when T310 timer expires.

Sergey Safarov

On Tue, May 12, 2015 at 9:27 PM, Dmitry Saratsky <simpot at gmail.com> wrote:

> Hi All!
>
> I have very strange problem on one of my switches...
>
> when I calling through 1 specific gateway (gateway's config well be shown
> later) - I get internal error with code 503 and freeswitch even do not try
> to send invite to the remote gateway...
> This gateway have very special configured as shown below (after debug
> output) and digest registration. Digest registration is passed fine for me!
>
> part of relevant console debug:
> ------------------------------------------------------------------
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:40
> sofia/ext-ipv4/12121231122 Standard INIT
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:48
> (sofia/ext-ipv4/12121231122) State Change CS_INIT -> CS_ROUTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:512
> (sofia/ext-ipv4/12121231122) State INIT going to sleep
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
> (sofia/ext-ipv4/12121231122) Running State Change CS_ROUTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:528
> (sofia/ext-ipv4/12121231122) State ROUTING
> 2015-05-12 21:15:36.462865 [DEBUG] mod_sofia.c:123
> sofia/ext-ipv4/12121231122 SOFIA ROUTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_ivr_originate.c:67
> (sofia/ext-ipv4/12121231122) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:528
> (sofia/ext-ipv4/12121231122) State ROUTING going to sleep
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
> (sofia/ext-ipv4/12121231122) Running State Change CS_CONSUME_MEDIA
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:547
> (sofia/ext-ipv4/12121231122) State CONSUME_MEDIA
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:547
> (sofia/ext-ipv4/12121231122) State CONSUME_MEDIA going to sleep
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] sofia.c:6623 Channel
> sofia/ext-ipv4/12121231122 entering state [calling][0]
> 2015-05-12 21:15:36.462865 [DEBUG] sofia.c:6623 Channel
> sofia/ext-ipv4/12121231122 entering state [terminated][503]
> 2015-05-12 21:15:36.462865 [NOTICE] sofia.c:7539 Hangup
> sofia/ext-ipv4/12121231122 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_channel.c:3222 Send signal
> sofia/ext-ipv4/12121231122 [KILL]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
> (sofia/ext-ipv4/12121231122) Running State Change CS_HANGUP
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:735
> (sofia/ext-ipv4/12121231122) Callstate Change DOWN -> HANGUP
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:737
> (sofia/ext-ipv4/12121231122) State HANGUP
> 2015-05-12 21:15:36.462865 [DEBUG] mod_sofia.c:413 Channel
> sofia/ext-ipv4/12121231122 hanging up, cause: NORMAL_TEMPORARY_FAILURE
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:60
> sofia/ext-ipv4/12121231122 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:737
> (sofia/ext-ipv4/12121231122) State HANGUP going to sleep
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:504
> (sofia/ext-ipv4/12121231122) State Change CS_HANGUP -> CS_REPORTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
> (sofia/ext-ipv4/12121231122) Running State Change CS_REPORTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:823
> (sofia/ext-ipv4/12121231122) State REPORTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:104
> sofia/ext-ipv4/12121231122 Standard REPORTING, cause:
> NORMAL_TEMPORARY_FAILURE
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:823
> (sofia/ext-ipv4/12121231122) State REPORTING going to sleep
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:498
> (sofia/ext-ipv4/12121231122) State Change CS_REPORTING -> CS_DESTROY
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1623 Session 4
> (sofia/ext-ipv4/12121231122) Locked, Waiting on external entities
> 2015-05-12 21:15:36.462865 [DEBUG] switch_ivr_originate.c:3720 Originate
> Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]
> 2015-05-12 21:15:36.462865 [INFO] mod_dptools.c:3244 Originate Failed.
> Cause: NORMAL_TEMPORARY_FAILURE
> ------------------------------------------------------------------
>
> no packet was sent to remote gateway (checked with tcpdump), so it is
> internal error...
> relevant gateway is registered well:
> ------------------------------------------------------------------
>
> =================================================================================================
> Name            gateway.name
> Profile         ext-ipv4
> Scheme          Digest
> Realm           realm.gateway.name
> Username        +12223334455
> Password        yes
> From            <sip:+12223334455 at realm.gateway.name>
> Contact         <sip:+12223334455 at 1.1.1.1:5060
> ;expires=3600;+g.oma.sip-im;+g.oma.sip-im.large-message;transport=udp;gw=
> gateway.name>
> Exten           +12223334455
> To              sip:+12223334455 at realm.gateway.name
> Proxy           sip:realm.gateway.name
> Context         ext
> Expires         3600
> Freq            3600
> Ping            0
> PingFreq        0
> PingTime        0.00
> PingState       0/0/0
> State           REGED
> Status          UP
> Uptime          554s
> CallsIN         0
> CallsOUT        2
> FailedCallsIN   0
> FailedCallsOUT  2
>
> =================================================================================================
> ------------------------------------------------------------------
>
> gateway config:
> ------------------------------------------------------------------
> <include>
>   <gateway name="gateway.name">
>     <param name="proxy" value="realm.gateway.name"/>
>     <param name="register-proxy" value="10.10.10.10"/>
>     <param name="contact-params"
> value="expires=3600;+g.oma.sip-im;+g.oma.sip-im.large-message"/>
>     <param name="username" value="+12223334455"/>
>     <param name="auth-username" value="+12223334455 at realm.gateway.name"/>
>     <param name="from-user" value="+12223334455"/>
>     <param name="from-domain" value="realm.gateway.name"/>
>     <param name="password" value="password"/>
>     <param name="expire-seconds" value="3600"/>
>     <param name="register" value="true"/>
>     <param name="register-transport" value="udp"/>
>     <param name="caller-id-in-from" value="false"/>
>     <param name="extension-in-contact" value="+12223334455"/>
>     <param name="retry-seconds" value="180"/>
>   </gateway>
> </include>
>
> ------------------------------------------------------------------
>
> what is special in this gateway configuration, is that gateway name is
> differers from proxy and both differs from register-proxy. this is required
> for my for successfully registration (bcz realm.gateway.name is not
> resolvable with DNS).
> In addition username is differs from auth-username - for some reason it is
> required by vendor...
>
> PS:
> FreeSWITCH (Version 1.5.15b git a41505f 2015-02-23 22:38:20Z 64bit) is
> ready
>
>
> Any ideas?
>
> Thanks,
> Dmitry.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jdickson at evolvetsi.com  Tue May 12 23:03:48 2015
From: jdickson at evolvetsi.com (Joseph Dickson)
Date: Tue, 12 May 2015 15:03:48 -0400
Subject: [Freeswitch-users] freeswitch systemd unit file for debian?
In-Reply-To: <10167.1431457124@ccs.covici.com>
References: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>
	<CAC-LwPPtPBmZpYeRpBZqg22KYuDKJDbRhZnhvg9S_HNXtA1oGA@mail.gmail.com>
	<A706245D-22A6-48FB-9361-B3317A7F30AA@jerris.com>
	<CAOHpEota04EeGXSE9_x2ohq2S4m9GbCMAQpZkUbWVYEvSFY8YQ@mail.gmail.com>
	<555243C2.8040706@temp3r.com>
	<CAOHpEotuq7LhSmU0rZUw7d0tAD+TO3JNbbh6GOcevbNvnVGwLA@mail.gmail.com>
	<10167.1431457124@ccs.covici.com>
Message-ID: <CAOHpEouMpuCdcS_yoMuJT7s-Afe07TN+fVD_rYmSqrfJ6BqG8Q@mail.gmail.com>

I think the key is to make sure that the pid file gets created in the right
spot -- note the -run flag being passed to override the run directory to be
/run/freeswitch

Joe

On Tue, May 12, 2015 at 2:58 PM, <covici at ccs.covici.com> wrote:

> I could never get type=forking to function, I wonder how you got that to
> work, I had to say -nf and use type=simple.
>
>
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From s.safarov at gmail.com  Tue May 12 23:04:43 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 12 May 2015 22:04:43 +0300
Subject: [Freeswitch-users] 503 internal error for gateway calls....
In-Reply-To: <CAM2iYy6kan4sP6a7RJ+DR38r-e6AS4QQaZapydygrB4JRbwkbQ@mail.gmail.com>
References: <CAM2iYy6kan4sP6a7RJ+DR38r-e6AS4QQaZapydygrB4JRbwkbQ@mail.gmail.com>
Message-ID: <CAHtxdDf02+4d9NW4LBu6a=-Q+_Fug4ym_cBdDjzwSg5qbcqfAw@mail.gmail.com>

The previous message is off topic. May be it similar this case FS-7540
<https://freeswitch.org/jira/browse/FS-7540>
https://freeswitch.org/jira/browse/FS-7540

On Tue, May 12, 2015 at 9:27 PM, Dmitry Saratsky <simpot at gmail.com> wrote:

> Hi All!
>
> I have very strange problem on one of my switches...
>
> when I calling through 1 specific gateway (gateway's config well be shown
> later) - I get internal error with code 503 and freeswitch even do not try
> to send invite to the remote gateway...
> This gateway have very special configured as shown below (after debug
> output) and digest registration. Digest registration is passed fine for me!
>
> part of relevant console debug:
> ------------------------------------------------------------------
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:40
> sofia/ext-ipv4/12121231122 Standard INIT
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:48
> (sofia/ext-ipv4/12121231122) State Change CS_INIT -> CS_ROUTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:512
> (sofia/ext-ipv4/12121231122) State INIT going to sleep
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
> (sofia/ext-ipv4/12121231122) Running State Change CS_ROUTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:528
> (sofia/ext-ipv4/12121231122) State ROUTING
> 2015-05-12 21:15:36.462865 [DEBUG] mod_sofia.c:123
> sofia/ext-ipv4/12121231122 SOFIA ROUTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_ivr_originate.c:67
> (sofia/ext-ipv4/12121231122) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:528
> (sofia/ext-ipv4/12121231122) State ROUTING going to sleep
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
> (sofia/ext-ipv4/12121231122) Running State Change CS_CONSUME_MEDIA
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:547
> (sofia/ext-ipv4/12121231122) State CONSUME_MEDIA
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:547
> (sofia/ext-ipv4/12121231122) State CONSUME_MEDIA going to sleep
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1061 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] sofia.c:6623 Channel
> sofia/ext-ipv4/12121231122 entering state [calling][0]
> 2015-05-12 21:15:36.462865 [DEBUG] sofia.c:6623 Channel
> sofia/ext-ipv4/12121231122 entering state [terminated][503]
> 2015-05-12 21:15:36.462865 [NOTICE] sofia.c:7539 Hangup
> sofia/ext-ipv4/12121231122 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_channel.c:3222 Send signal
> sofia/ext-ipv4/12121231122 [KILL]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
> (sofia/ext-ipv4/12121231122) Running State Change CS_HANGUP
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:735
> (sofia/ext-ipv4/12121231122) Callstate Change DOWN -> HANGUP
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:737
> (sofia/ext-ipv4/12121231122) State HANGUP
> 2015-05-12 21:15:36.462865 [DEBUG] mod_sofia.c:413 Channel
> sofia/ext-ipv4/12121231122 hanging up, cause: NORMAL_TEMPORARY_FAILURE
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:60
> sofia/ext-ipv4/12121231122 Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:737
> (sofia/ext-ipv4/12121231122) State HANGUP going to sleep
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:504
> (sofia/ext-ipv4/12121231122) State Change CS_HANGUP -> CS_REPORTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:472
> (sofia/ext-ipv4/12121231122) Running State Change CS_REPORTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:823
> (sofia/ext-ipv4/12121231122) State REPORTING
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:104
> sofia/ext-ipv4/12121231122 Standard REPORTING, cause:
> NORMAL_TEMPORARY_FAILURE
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:823
> (sofia/ext-ipv4/12121231122) State REPORTING going to sleep
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_state_machine.c:498
> (sofia/ext-ipv4/12121231122) State Change CS_REPORTING -> CS_DESTROY
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1396 Send signal
> sofia/ext-ipv4/12121231122 [BREAK]
> 2015-05-12 21:15:36.462865 [DEBUG] switch_core_session.c:1623 Session 4
> (sofia/ext-ipv4/12121231122) Locked, Waiting on external entities
> 2015-05-12 21:15:36.462865 [DEBUG] switch_ivr_originate.c:3720 Originate
> Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]
> 2015-05-12 21:15:36.462865 [INFO] mod_dptools.c:3244 Originate Failed.
> Cause: NORMAL_TEMPORARY_FAILURE
> ------------------------------------------------------------------
>
> no packet was sent to remote gateway (checked with tcpdump), so it is
> internal error...
> relevant gateway is registered well:
> ------------------------------------------------------------------
>
> =================================================================================================
> Name            gateway.name
> Profile         ext-ipv4
> Scheme          Digest
> Realm           realm.gateway.name
> Username        +12223334455
> Password        yes
> From            <sip:+12223334455 at realm.gateway.name>
> Contact         <sip:+12223334455 at 1.1.1.1:5060
> ;expires=3600;+g.oma.sip-im;+g.oma.sip-im.large-message;transport=udp;gw=
> gateway.name>
> Exten           +12223334455
> To              sip:+12223334455 at realm.gateway.name
> Proxy           sip:realm.gateway.name
> Context         ext
> Expires         3600
> Freq            3600
> Ping            0
> PingFreq        0
> PingTime        0.00
> PingState       0/0/0
> State           REGED
> Status          UP
> Uptime          554s
> CallsIN         0
> CallsOUT        2
> FailedCallsIN   0
> FailedCallsOUT  2
>
> =================================================================================================
> ------------------------------------------------------------------
>
> gateway config:
> ------------------------------------------------------------------
> <include>
>   <gateway name="gateway.name">
>     <param name="proxy" value="realm.gateway.name"/>
>     <param name="register-proxy" value="10.10.10.10"/>
>     <param name="contact-params"
> value="expires=3600;+g.oma.sip-im;+g.oma.sip-im.large-message"/>
>     <param name="username" value="+12223334455"/>
>     <param name="auth-username" value="+12223334455 at realm.gateway.name"/>
>     <param name="from-user" value="+12223334455"/>
>     <param name="from-domain" value="realm.gateway.name"/>
>     <param name="password" value="password"/>
>     <param name="expire-seconds" value="3600"/>
>     <param name="register" value="true"/>
>     <param name="register-transport" value="udp"/>
>     <param name="caller-id-in-from" value="false"/>
>     <param name="extension-in-contact" value="+12223334455"/>
>     <param name="retry-seconds" value="180"/>
>   </gateway>
> </include>
>
> ------------------------------------------------------------------
>
> what is special in this gateway configuration, is that gateway name is
> differers from proxy and both differs from register-proxy. this is required
> for my for successfully registration (bcz realm.gateway.name is not
> resolvable with DNS).
> In addition username is differs from auth-username - for some reason it is
> required by vendor...
>
> PS:
> FreeSWITCH (Version 1.5.15b git a41505f 2015-02-23 22:38:20Z 64bit) is
> ready
>
>
> Any ideas?
>
> Thanks,
> Dmitry.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tilt at temp3r.com  Tue May 12 23:28:16 2015
From: tilt at temp3r.com (tilt)
Date: Tue, 12 May 2015 12:28:16 -0700
Subject: [Freeswitch-users] freeswitch systemd unit file for debian?
In-Reply-To: <10167.1431457124@ccs.covici.com>
References: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>	<CAC-LwPPtPBmZpYeRpBZqg22KYuDKJDbRhZnhvg9S_HNXtA1oGA@mail.gmail.com>	<A706245D-22A6-48FB-9361-B3317A7F30AA@jerris.com>	<CAOHpEota04EeGXSE9_x2ohq2S4m9GbCMAQpZkUbWVYEvSFY8YQ@mail.gmail.com>	<555243C2.8040706@temp3r.com>	<CAOHpEotuq7LhSmU0rZUw7d0tAD+TO3JNbbh6GOcevbNvnVGwLA@mail.gmail.com>
	<10167.1431457124@ccs.covici.com>
Message-ID: <55525450.2030602@temp3r.com>

Thanks Joseph, this seems to be working for me!

On 05/12/2015 11:58 AM, covici at ccs.covici.com wrote:
> [Unit]
> >Description=freeswitch
> >After=syslog.target network.target local-fs.target
> >
> >[Service]
> >; service
> >Type=forking
> >RuntimeDirectory=freeswitch
> >PIDFile=/run/freeswitch/freeswitch.pid
> >PermissionsStartOnly=true
> >ExecStart=/usr/bin/freeswitch -ncwait -nonat -run /run/freeswitch
> >TimeoutSec=45s
> >Restart=always
> >; exec
> >WorkingDirectory=/run/freeswitch
> >User=freeswitch
> >Group=daemon
> >LimitCORE=infinity
> >LimitNOFILE=100000
> >LimitNPROC=60000
> >;LimitSTACK=240
> >LimitRTPRIO=infinity
> >LimitRTTIME=7000000
> >IOSchedulingClass=realtime
> >IOSchedulingPriority=2
> >CPUSchedulingPolicy=rr
> >CPUSchedulingPriority=89
> >UMask=0007
> >
> >[Install]
> >WantedBy=multi-user.target
> >



From adam.ben.ayoun1 at gmail.com  Tue May 12 23:44:07 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Tue, 12 May 2015 22:44:07 +0300
Subject: [Freeswitch-users] Improving voice quality
In-Reply-To: <CAC8JMOJ2AkJtyFbpz_NmCOeG3YO6mrc=Jkq40_0Yu23_4pieNw@mail.gmail.com>
References: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
	<CAC8JMOJ2AkJtyFbpz_NmCOeG3YO6mrc=Jkq40_0Yu23_4pieNw@mail.gmail.com>
Message-ID: <CADNqFCkeB4Hun5U6Xze0fZv-pYVff4u=7y6jDiGnoK236CkaHQ@mail.gmail.com>

We tried on c4.4xlarge with dedicated tenancy and it's pretty much the
same. Any special config you applied to the machine? I am using CentOS 6.

On 12 May 2015 at 21:04, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Have you tried it with non-mobile WebRTC clients?  Is there the same
> problem?
>
> I am using mod_conference on FS hosted on AWS m3.xlarge instance and I
> have not noticed audio problem.  But my WebRTC clients are not mobile but
> rather Chrome on PCs.
>
> On Tue, May 12, 2015 at 9:59 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com
> > wrote:
>
>> Hi guys,
>>
>> We are using Freeswitch as a audio "MCU" for WebRTC using mod_conference,
>> we are currently using mobile clients on both Android and iOS. The voice
>> quality is good but not as good as Hangouts for example, there are small
>> cut-offs every now and then that affects the overall experience. We are
>> trying to understand the reason for that, maybe it's Freeswitch mixing
>> algorithm, our servers infrastructure, or something else. We did tried
>> Janus audio conference demo and it was very good (we used chrome on Android
>> which should use pretty much the same stack). We really need help figuring
>> this out. I want to first make sure it's not our infrastructure (mainly
>> network/CPU). We are using virtualized m3.large on AWS (2 vCPUs and 7.5GB
>> RAM). Maybe there's a certain instance type on a certain provider that is
>> known to give the best quality? Any ideas on how to go about this?
>>
>> Thanks,
>> Adam
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From covici at ccs.covici.com  Tue May 12 23:46:53 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 12 May 2015 15:46:53 -0400
Subject: [Freeswitch-users] freeswitch systemd unit file for debian?
In-Reply-To: <CAOHpEouMpuCdcS_yoMuJT7s-Afe07TN+fVD_rYmSqrfJ6BqG8Q@mail.gmail.com>
References: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>
	<CAC-LwPPtPBmZpYeRpBZqg22KYuDKJDbRhZnhvg9S_HNXtA1oGA@mail.gmail.com>
	<A706245D-22A6-48FB-9361-B3317A7F30AA@jerris.com>
	<CAOHpEota04EeGXSE9_x2ohq2S4m9GbCMAQpZkUbWVYEvSFY8YQ@mail.gmail.com>
	<555243C2.8040706@temp3r.com>
	<CAOHpEotuq7LhSmU0rZUw7d0tAD+TO3JNbbh6GOcevbNvnVGwLA@mail.gmail.com>
	<10167.1431457124@ccs.covici.com>
	<CAOHpEouMpuCdcS_yoMuJT7s-Afe07TN+fVD_rYmSqrfJ6BqG8Q@mail.gmail.com>
Message-ID: <11656.1431460013@ccs.covici.com>

I had mine as /var/run/freeswitch and a Pidfile directive, but it did
not work, but -nf does, so I guess its OK.

Joseph Dickson <jdickson at evolvetsi.com> wrote:

> I think the key is to make sure that the pid file gets created in the right
> spot -- note the -run flag being passed to override the run directory to be
> /run/freeswitch
> 
> Joe
> 
> On Tue, May 12, 2015 at 2:58 PM, <covici at ccs.covici.com> wrote:
> 
> > I could never get type=forking to function, I wonder how you got that to
> > work, I had to say -nf and use type=simple.
> >
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From olegstolyar at gmail.com  Tue May 12 23:59:14 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 12 May 2015 12:59:14 -0700
Subject: [Freeswitch-users] Improving voice quality
In-Reply-To: <CADNqFCkeB4Hun5U6Xze0fZv-pYVff4u=7y6jDiGnoK236CkaHQ@mail.gmail.com>
References: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
	<CAC8JMOJ2AkJtyFbpz_NmCOeG3YO6mrc=Jkq40_0Yu23_4pieNw@mail.gmail.com>
	<CADNqFCkeB4Hun5U6Xze0fZv-pYVff4u=7y6jDiGnoK236CkaHQ@mail.gmail.com>
Message-ID: <CAC8JMOKSupg2Cc41QdyEbMa7fgx98s6W1=KV22aQxpUD6m0NEQ@mail.gmail.com>

Nothing special except maybe I am using PCMU instead of Opus.  I am using
Ubuntu 14 now but it also worked fine with CentOS 5.9.

Just for my own education - are you seeing the same issues with non-mobile
clients?

On Tue, May 12, 2015 at 12:44 PM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
wrote:

> We tried on c4.4xlarge with dedicated tenancy and it's pretty much the
> same. Any special config you applied to the machine? I am using CentOS 6.
>
> On 12 May 2015 at 21:04, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>> Have you tried it with non-mobile WebRTC clients?  Is there the same
>> problem?
>>
>> I am using mod_conference on FS hosted on AWS m3.xlarge instance and I
>> have not noticed audio problem.  But my WebRTC clients are not mobile but
>> rather Chrome on PCs.
>>
>> On Tue, May 12, 2015 at 9:59 AM, Adam Ben-Ayoun <
>> adam.ben.ayoun1 at gmail.com> wrote:
>>
>>> Hi guys,
>>>
>>> We are using Freeswitch as a audio "MCU" for WebRTC using
>>> mod_conference, we are currently using mobile clients on both Android and
>>> iOS. The voice quality is good but not as good as Hangouts for example,
>>> there are small cut-offs every now and then that affects the overall
>>> experience. We are trying to understand the reason for that, maybe it's
>>> Freeswitch mixing algorithm, our servers infrastructure, or something else.
>>> We did tried Janus audio conference demo and it was very good (we used
>>> chrome on Android which should use pretty much the same stack). We really
>>> need help figuring this out. I want to first make sure it's not our
>>> infrastructure (mainly network/CPU). We are using virtualized m3.large on
>>> AWS (2 vCPUs and 7.5GB RAM). Maybe there's a certain instance type on a
>>> certain provider that is known to give the best quality? Any ideas on how
>>> to go about this?
>>>
>>> Thanks,
>>> Adam
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed May 13 00:02:08 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 May 2015 16:02:08 -0400
Subject: [Freeswitch-users] freeswitch systemd unit file for debian?
In-Reply-To: <11656.1431460013@ccs.covici.com>
References: <CAOHpEouD+4aZajPp3amFWRbdvgEGp4g-GeK87jNkPRKgmaimBA@mail.gmail.com>
	<CAC-LwPPtPBmZpYeRpBZqg22KYuDKJDbRhZnhvg9S_HNXtA1oGA@mail.gmail.com>
	<A706245D-22A6-48FB-9361-B3317A7F30AA@jerris.com>
	<CAOHpEota04EeGXSE9_x2ohq2S4m9GbCMAQpZkUbWVYEvSFY8YQ@mail.gmail.com>
	<555243C2.8040706@temp3r.com>
	<CAOHpEotuq7LhSmU0rZUw7d0tAD+TO3JNbbh6GOcevbNvnVGwLA@mail.gmail.com>
	<10167.1431457124@ccs.covici.com>
	<CAOHpEouMpuCdcS_yoMuJT7s-Afe07TN+fVD_rYmSqrfJ6BqG8Q@mail.gmail.com>
	<11656.1431460013@ccs.covici.com>
Message-ID: <EEF94A10-0054-4089-9CBC-E9070B48CF96@jerris.com>

Can someone please open a jira and a pull request to add this to tree, and to include it in jesse packages?

> On May 12, 2015, at 3:46 PM, covici at ccs.covici.com wrote:
> 
> I had mine as /var/run/freeswitch and a Pidfile directive, but it did
> not work, but -nf does, so I guess its OK.
> 
> Joseph Dickson <jdickson at evolvetsi.com> wrote:
> 
>> I think the key is to make sure that the pid file gets created in the right
>> spot -- note the -run flag being passed to override the run directory to be
>> /run/freeswitch
>> 
>> Joe
>> 
>> On Tue, May 12, 2015 at 2:58 PM, <covici at ccs.covici.com> wrote:
>> 
>>> I could never get type=forking to function, I wonder how you got that to
>>> work, I had to say -nf and use type=simple.
>>> 
>>> 
>> 
>> ----------------------------------------------------
>> Alternatives:
>> 
>> ----------------------------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>         John Covici
>         covici at ccs.covici.com
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 shogun at tausys.de  Wed May 13 00:16:06 2015
From: shogun at tausys.de (Jens Tautenhahn)
Date: Tue, 12 May 2015 22:16:06 +0200
Subject: [Freeswitch-users] CDR problem
Message-ID: <55525F86.6080306@tausys.de>

Hi,

I have a very initial FusionPBX/FreeSwitch 1.4.18 setup. I have added
two extensions and multiple SIP trunks. Firewall config was a little bit
tricky because FreeSwitch is NATed. But now everything works great.

Now the problem: I got only CDRs for the first extension. Either for
internal or external calls. Outgoing calls from the second extension are
not logged.

How can I find the cause?


Greetings


From mishehu at freeswitch.org  Wed May 13 01:13:15 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Tue, 12 May 2015 16:13:15 -0500
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <17898630.20150512135025@seznam.cz>
References: <17898630.20150512135025@seznam.cz>
Message-ID: <55526CEB.3090900@freeswitch.org>

What version of Centos?  If it's not Centos 7, I wouldn't be very 
hopeful.  Even 7 is probably going to be a path that you will need to 
tread yourself, since I don't think anybody else at this time has done 
this yet.  I could be mistaken, of course.

-Yossi

On 05/12/2015 06:50 AM, Denis Jakovlev wrote:
> 1.6 install to Centos Hi all,
>
> Is there a chance to put on CentOS the new version 1.6?
>
> I've tried. But I have a problem on ./configure -C
> checking for libyuv >= 0.0.1280... configure: error: You need to 
> install libyuv-dev. Required library
>
> Where can I find this library? I tried to put it from here 
> https://code.google.com/p/libyuv/But FreeSwitch does not see it.
>
> Is there a possibility to find this library to CentOS?
>
> /--
> S pozdravem,
> Ing.Denis Jakovlev
> mob.tel. 775-415-382
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> /

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From adam.ben.ayoun1 at gmail.com  Wed May 13 01:42:55 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Wed, 13 May 2015 00:42:55 +0300
Subject: [Freeswitch-users] Improving voice quality
In-Reply-To: <CAC8JMOJ2AkJtyFbpz_NmCOeG3YO6mrc=Jkq40_0Yu23_4pieNw@mail.gmail.com>
References: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
	<CAC8JMOJ2AkJtyFbpz_NmCOeG3YO6mrc=Jkq40_0Yu23_4pieNw@mail.gmail.com>
Message-ID: <CADNqFCkRO0Jyesdi84HRKb9Df-ci_1CvFdfp_9uzk2j-Buoeig@mail.gmail.com>

Oleg, we did try Chrome on PC in the past, indeed it gave better results. I
will retry again soon.

On 12 May 2015 at 21:04, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Have you tried it with non-mobile WebRTC clients?  Is there the same
> problem?
>
> I am using mod_conference on FS hosted on AWS m3.xlarge instance and I
> have not noticed audio problem.  But my WebRTC clients are not mobile but
> rather Chrome on PCs.
>
> On Tue, May 12, 2015 at 9:59 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com
> > wrote:
>
>> Hi guys,
>>
>> We are using Freeswitch as a audio "MCU" for WebRTC using mod_conference,
>> we are currently using mobile clients on both Android and iOS. The voice
>> quality is good but not as good as Hangouts for example, there are small
>> cut-offs every now and then that affects the overall experience. We are
>> trying to understand the reason for that, maybe it's Freeswitch mixing
>> algorithm, our servers infrastructure, or something else. We did tried
>> Janus audio conference demo and it was very good (we used chrome on Android
>> which should use pretty much the same stack). We really need help figuring
>> this out. I want to first make sure it's not our infrastructure (mainly
>> network/CPU). We are using virtualized m3.large on AWS (2 vCPUs and 7.5GB
>> RAM). Maybe there's a certain instance type on a certain provider that is
>> known to give the best quality? Any ideas on how to go about this?
>>
>> Thanks,
>> Adam
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed May 13 01:56:35 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 May 2015 17:56:35 -0400
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <55526CEB.3090900@freeswitch.org>
References: <17898630.20150512135025@seznam.cz>
	<55526CEB.3090900@freeswitch.org>
Message-ID: <401F1BB8-7EF6-4591-909D-3B6ABD4502DB@jerris.com>

We have a community member working on rpms for the deps, and we are working on getting tarballs of them up.  For now you can get them from: https://freeswitch.org/stash/projects/SD <https://freeswitch.org/stash/projects/SD> .  You will need some of these and not others, and may need to build newer versions of other system packages to get some features to work.  Stay tuned.


> On May 12, 2015, at 5:13 PM, I put the Who? in Mishehu <mishehu at freeswitch.org> wrote:
> 
> What version of Centos?  If it's not Centos 7, I wouldn't be very hopeful.  Even 7 is probably going to be a path that you will need to tread yourself, since I don't think anybody else at this time has done this yet.  I could be mistaken, of course.
> 
> -Yossi
> 
> On 05/12/2015 06:50 AM, Denis Jakovlev wrote:
>> Hi all,
>> 
>> Is there a chance to put on CentOS the new version 1.6?
>> 
>> I've tried. But I have a problem on ./configure -C 
>> checking for libyuv >= 0.0.1280... configure: error: You need to install libyuv-dev. Required library
>> 
>> Where can I find this library? I tried to put it from here https://code.google.com/p/libyuv/ <https://code.google.com/p/libyuv/> But FreeSwitch does not see it.
>> 
>> Is there a possibility to find this library to CentOS?
>> 

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From krice at freeswitch.org  Wed May 13 02:00:01 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 12 May 2015 17:00:01 -0500
Subject: [Freeswitch-users] Improving voice quality
In-Reply-To: <CADNqFCkeB4Hun5U6Xze0fZv-pYVff4u=7y6jDiGnoK236CkaHQ@mail.gmail.com>
Message-ID: <D177E211.9DA6B%krice@freeswitch.org>

Are you sure what you are hearing is not just the noise gate? (squelch in
old school radio terms)

As far as audio quality goes it should blow hangouts away.... Of course as
with anything running on commodity shared virtual server hosting your
mileage will vary. 


On 5/12/15, 2:44 PM, "Adam Ben-Ayoun" <adam.ben.ayoun1 at gmail.com> wrote:

> We tried on c4.4xlarge with dedicated tenancy and it's pretty much the same.
> Any special config you applied to the machine? I am using CentOS 6.
> 
> On 12 May 2015 at 21:04, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>> Have you tried it with non-mobile WebRTC clients?? Is there the same problem?
>> 
>> I am using mod_conference on FS hosted on AWS m3.xlarge instance and I have
>> not noticed audio problem.? But my WebRTC clients are not mobile but rather
>> Chrome on PCs.
>> 
>> On Tue, May 12, 2015 at 9:59 AM, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com>
>> wrote:
>>> Hi guys,
>>> 
>>> We are using Freeswitch as a audio "MCU" for WebRTC using mod_conference, we
>>> are currently using mobile clients on both Android and iOS. The voice
>>> quality is good but not as good as Hangouts for example, there are small
>>> cut-offs every now and then that affects the overall experience. We are
>>> trying to understand the reason for that, maybe it's Freeswitch mixing
>>> algorithm, our servers infrastructure, or something else. We did tried Janus
>>> audio conference demo and it was very good (we used chrome on Android which
>>> should use pretty much the same stack). We really need help figuring this
>>> out. I want to first make sure it's not our infrastructure (mainly
>>> network/CPU). We are using virtualized m3.large on AWS (2 vCPUs and 7.5GB
>>> RAM). Maybe there's a certain instance type on a certain provider that is
>>> known to give the best quality? Any ideas on how to go about this?
>>> 
>>> Thanks,
>>> Adam
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From bobjectsfreeswitch at gmail.com  Wed May 13 02:29:59 2015
From: bobjectsfreeswitch at gmail.com (Bob Hartwig)
Date: Tue, 12 May 2015 17:29:59 -0500
Subject: [Freeswitch-users] Improving voice quality
In-Reply-To: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
References: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
Message-ID: <CAAemr3zp2=iwozjSRxvMbVKR=zBn6fpEHuPbTx8YSeA9J9Craw@mail.gmail.com>

"for example, there are small cut-offs every now and then"

Have you tried setting "conference-flags" to "audio-always" in
conference.conf.xml?
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From manavid at gmail.com  Wed May 13 04:05:21 2015
From: manavid at gmail.com (Mohammad Amin Navid)
Date: Tue, 12 May 2015 17:05:21 -0700
Subject: [Freeswitch-users] [SOS] Help Needed with Strange Behavior
Message-ID: <BC8448F5-A6EE-4AE9-9930-454C7053350C@gmail.com>

To all the crazy ones, all the gurus:

TLDR: FreeSWITCH Server looses connectivity with main router (gateway), can ping any other server but cannot reach router.

We have a cluster of FreeSWITCH servers running behind Kamailio.

For 3 years we have been up and running smoothly without any glitches.

Last week we upgraded our FreeSWITCH servers, the upgrade included:

1) Upgraded Debian Squeeze to Ubuntu LTS 14.04

2) Upgraded FreeSWITCH from version 1.2 to 1.4.18 (git: 4eed221)

Ever since last week, a very strange problem happened 3 times. The FreeSWITCH servers randomly loose connectivity with our main router (gateway).

FS servers can be reached and pinged from the rest of servers, they also can ping and reach all other servers but when we ping router from inside FS servers they cannot ping and reach the router.

The only way we can get them working again is by:

ifconfig eth1 down
sleep 3
ifconfig eth1 up
route add default gw 208.65.xxx.xxx eth1

The rest of our 20 servers are all running Ubuntu 14.04 and most of them are under heavy load, this only happens with FS servers.

Does any one have any clue about this bizarre situation? Any one know under what circumstances a server might loose connectivity and routing to its gateway?

Any pointer for debugging or suggestion is highly appreciated.

Thank you,

Moe

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From olegstolyar at gmail.com  Wed May 13 04:15:08 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 12 May 2015 17:15:08 -0700
Subject: [Freeswitch-users] Improving voice quality
In-Reply-To: <CAAemr3zp2=iwozjSRxvMbVKR=zBn6fpEHuPbTx8YSeA9J9Craw@mail.gmail.com>
References: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
	<CAAemr3zp2=iwozjSRxvMbVKR=zBn6fpEHuPbTx8YSeA9J9Craw@mail.gmail.com>
Message-ID: <CAC8JMO+ex15eFrvocUaDqRoahooY3hr3jy6MpG=PyE45KP=jHg@mail.gmail.com>

Bob has a good point.  Not sure about the flag but I do have energy-level
set to 0 in my conference.conf.xml

<param name="energy-level" value="0"/>


On Tue, May 12, 2015 at 3:29 PM, Bob Hartwig <bobjectsfreeswitch at gmail.com>
wrote:

> "for example, there are small cut-offs every now and then"
>
> Have you tried setting "conference-flags" to "audio-always" in
> conference.conf.xml?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Wed May 13 05:04:46 2015
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 13 May 2015 09:04:46 +0800
Subject: [Freeswitch-users] Video record problem
In-Reply-To: <1414775449.20150506134751@seznam.cz>
References: <1414775449.20150506134751@seznam.cz>
Message-ID: <CAJs+-5q4vRFoBa_H2eam13aESLYi7VvyCJR6nLTW0DWZFwC9nA@mail.gmail.com>

what version are you using?
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From dujinfang at gmail.com  Wed May 13 05:13:13 2015
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 13 May 2015 09:13:13 +0800
Subject: [Freeswitch-users] mod_hiredis
In-Reply-To: <55524336.4050208@quentustech.com>
References: <554EB704.5090605@quentustech.com> <5551E552.9090307@telefaks.de>
	<55524336.4050208@quentustech.com>
Message-ID: <CAJs+-5pN1KHKrPTKUnET2rbkj0Efe0-y+2VnRyzyfZiMeq7pBg@mail.gmail.com>

maybe use as a central storage for http sessions and ws sessions. or even
SSO if run FreeSWITCH as a cluster if we extend the http support in
FreeSWITCH.



On Wed, May 13, 2015 at 2:15 AM, William King <william.king at quentustech.com>
wrote:

> Peter,
>
> The pubsub actions should be available now with the raw app, though this
> would work in a blocking manner. You do raise a good use case for
> asynchronous commands, specifically for fire and forget cases like pubsub.
>
> For the prefix routing, currently the array return type isn't supported,
> but if redis returns a single result that should be able to work and be
> tested right now. Can you provide a few sample redis commands and
> responses for how you'd setup this scenario?
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 05/12/2015 04:34 AM, Peter Steinbach wrote:
> > Hello William,
> >
> > this is great, the idea of integrating Redis. We currently use Memcache
> > in raw mode as a method of externally controlling dialplans and failover
> > scenarios.
> > Redis, of course, brings much more features here.
> >
> >>Currently the two main use cases are:
> >>1. Call per second limits
> >>2. Concurrent call limits
> >>
> >>Possible additional functionality:
> >>1. Support for fail-over connections
> >>2. Asynchronous commands(is there a use case for this?)
> >
> > Another idea for your list would be to route calls according to
> > prefixes. You may lookup Redis with a part of the phone number and it
> > returns the gateway for this part of the number (redis DB is then
> > preloaded from another application).
> > And - as Redis has a publish/subscribe method - you will be able to
> > publish call informations from the dialplan to multiple external
> > subscribers (e.g. announce an incoming call to a CRM) without the use of
> > ESL. Is there a chance to run the redis dialplan app in a non blocking
> > manner for this scenario, in order to speed up the dialplan?
> >
> >
> >
> > Best regards
> > Peter
> >
> > On 05/10/15 03:40, William King wrote:
> >> I'm working on an update Redis integration module that will use the C
> >> library hiredis:
> >> http://redis.io/clients#c
> >> https://github.com/redis/hiredis
> >>
> >> I've pushed an alpha version of the module to a branch here:
> >>
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits?until=
> >> refs%2Fheads%2Fmod_hiredis
> >>
> >> The current module has a dialplan app and an api for 'hiredis_raw'
> >> which allows any single line Redis command, and executes it in a
> >> blocking manner, then supports returning string and integer responses.
> >>
> >> If anyone on this list has any use cases for FreeSWITCH+Redis, please
> >> reply to this thread. Currently the two main use cases are:
> >> 1. Call per second limits
> >> 2. Concurrent call limits
> >>
> >> Possible additional functionality:
> >> 1. Support for fail-over connections
> >> 2. Asynchronous commands(is there a use case for this?)
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at 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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
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From anthony.minessale at gmail.com  Wed May 13 08:48:50 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 May 2015 23:48:50 -0500
Subject: [Freeswitch-users] [SOS] Help Needed with Strange Behavior
In-Reply-To: <BC8448F5-A6EE-4AE9-9930-454C7053350C@gmail.com>
References: <BC8448F5-A6EE-4AE9-9930-454C7053350C@gmail.com>
Message-ID: <CAKbxfG8Ntqb7CwZmq38njhx2N1zpOOokD8JGiTBpinFsRv+Erw@mail.gmail.com>

Whoever convinced you to use ubuntu must have been mad at you.

Probably the problem is elementary networking mistake and nothing to do
with FS.  Look closer at the details is all I can suggest.  If you do find
FS has the power to derail you ethernet it will be a story long told.

On Tuesday, May 12, 2015, Mohammad Amin Navid <manavid at gmail.com> wrote:

> To all the crazy ones, all the gurus:
>
> *TLDR: FreeSWITCH Server looses connectivity with main router (gateway),
> can ping any other server but cannot reach router.*
>
> We have a cluster of FreeSWITCH servers running behind Kamailio.
>
> For 3 years we have been up and running smoothly without any glitches.
>
> Last week we upgraded our FreeSWITCH servers, the upgrade included:
>
> 1) Upgraded Debian Squeeze to Ubuntu LTS 14.04
>
> 2) Upgraded FreeSWITCH from version 1.2 to 1.4.18 (git: 4eed221)
>
> Ever since last week, a very strange problem happened 3 times. The
> FreeSWITCH servers randomly loose connectivity with our main router
> (gateway).
>
> FS servers can be reached and pinged from the rest of servers, they also
> can ping and reach all other servers but when we ping router from inside FS
> servers they cannot ping and reach the router.
>
> The only way we can get them working again is by:
>
> *ifconfig eth1 down*
> *sleep 3*
> *ifconfig eth1 up*
> *route add default gw 208.65.xxx.xxx eth1*
>
> The rest of our 20 servers are all running Ubuntu 14.04 and most of them
> are under heavy load, this only happens with FS servers.
>
> Does any one have any clue about this bizarre situation? Any one know
> under what circumstances a server might loose connectivity and routing to
> its gateway?
>
> Any pointer for debugging or suggestion is highly appreciated.
>
> Thank you,
>
> Moe
>
>

-- 
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 jlrichesin at gmail.com  Wed May 13 09:13:45 2015
From: jlrichesin at gmail.com (Josh Richesin)
Date: Tue, 12 May 2015 22:13:45 -0700
Subject: [Freeswitch-users] [SOS] Help Needed with Strange Behavior
In-Reply-To: <CAKbxfG8Ntqb7CwZmq38njhx2N1zpOOokD8JGiTBpinFsRv+Erw@mail.gmail.com>
References: <BC8448F5-A6EE-4AE9-9930-454C7053350C@gmail.com>
	<CAKbxfG8Ntqb7CwZmq38njhx2N1zpOOokD8JGiTBpinFsRv+Erw@mail.gmail.com>
Message-ID: <5552DD89.3070107@gmail.com>

I would move the FreeSwitch back to debian.  Ubuntu is really for 
desktops if you ask me.  I would never put anything production on 
Ubuntu.  Way to much bloat.

Josh Richesin


On 5/12/2015 9:48 PM, Anthony Minessale wrote:
> ne have any clue abou



From ali.jibran44 at gmail.com  Wed May 13 10:04:44 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Wed, 13 May 2015 11:04:44 +0500
Subject: [Freeswitch-users] CDR problem
In-Reply-To: <55525F86.6080306@tausys.de>
References: <55525F86.6080306@tausys.de>
Message-ID: <CAP9PqkTXX0Enna6na1ACMFCVcEiCnuXLiak7r4gvRHz2AjJZsg@mail.gmail.com>

Make a new file test.php and configure xml_cdr.conf to use this file.
Place this as its contents.
file_put_contents("outputfile.txt", file_get_contents("php://input"));

This will dump all the data into outputfile.txt. Try it with both the first
and second extensions. And then compare the logs to see what difference if
there. Check for urlencoding and tags.

If the logs are the same then In the fusionPBX folder search for the file
v_xml_cdr_import.
This is the file that parses the POSTed XML from freeswitch and dumps it
into the database. Try logging at different places to see why the second
extension is failing to dump data.

On Wednesday, May 13, 2015, Jens Tautenhahn <shogun at tausys.de> wrote:

> Hi,
>
> I have a very initial FusionPBX/FreeSwitch 1.4.18 setup. I have added
> two extensions and multiple SIP trunks. Firewall config was a little bit
> tricky because FreeSwitch is NATed. But now everything works great.
>
> Now the problem: I got only CDRs for the first extension. Either for
> internal or external calls. Outgoing calls from the second extension are
> not logged.
>
> How can I find the cause?
>
>
> Greetings
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yadenis at seznam.cz  Wed May 13 10:48:28 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 13 May 2015 08:48:28 +0200
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <55526CEB.3090900@freeswitch.org>
References: <17898630.20150512135025@seznam.cz>
	<55526CEB.3090900@freeswitch.org>
Message-ID: <1251442930.20150513084828@seznam.cz>

Dobr? den,

This version 7, yes. This version is somehow different?

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

?ter? 12. kv?tna 2015, 23:13:15, napsal jste:


What version of Centos?  If it's not Centos 7, I wouldn't be very hopeful.  Even 7 is probably going to be a path that you will need to tread yourself, since I don't think anybody else at this time has done this yet.  I could be mistaken, of course.

-Yossi

On 05/12/2015 06:50 AM, Denis Jakovlev wrote:
1.6 install to Centos Hi all,

Is there a chance to put on CentOS the new version 1.6?

I've tried. But I have a problem on ./configure -C 
checking for libyuv >= 0.0.1280... configure: error: You need to install libyuv-dev. Required library

Where can I find this library? I tried to put it from here https://code.google.com/p/libyuv/ But FreeSwitch does not see it.

Is there a possibility to find this library to CentOS?

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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 May 13 10:53:16 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 13 May 2015 08:53:16 +0200
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <401F1BB8-7EF6-4591-909D-3B6ABD4502DB@jerris.com>
References: <17898630.20150512135025@seznam.cz>
	<55526CEB.3090900@freeswitch.org> 
	<401F1BB8-7EF6-4591-909D-3B6ABD4502DB@jerris.com>
Message-ID: <175969258.20150513085316@seznam.cz>

Dobr? den,

Yes, I have tried to compile a library from here. It went without error. But the command configuration ./configure -C still does not pass, and wants.

checking for libyuv >= 0.0.1280... configure: error: You need to install libyuv-dev. Required library


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

?ter? 12. kv?tna 2015, 23:56:35, napsal jste:


We have a community member working on rpms for the deps, and we are working on getting tarballs of them up.  For now you can get them from: https://freeswitch.org/stash/projects/SD .  You will need some of these and not others, and may need to build newer versions of other system packages to get some features to work.  Stay tuned.


On May 12, 2015, at 5:13 PM, I put the Who? in Mishehu <mishehu at freeswitch.org> wrote:

What version of Centos?  If it's not Centos 7, I wouldn't be very hopeful.  Even 7 is probably going to be a path that you will need to tread yourself, since I don't think anybody else at this time has done this yet.  I could be mistaken, of course.

-Yossi

On 05/12/2015 06:50 AM, Denis Jakovlev wrote:
Hi all,

Is there a chance to put on CentOS the new version 1.6?

I've tried. But I have a problem on ./configure -C 
checking for libyuv >= 0.0.1280... configure: error: You need to install libyuv-dev. Required library

Where can I find this library? I tried to put it from here https://code.google.com/p/libyuv/ But FreeSwitch does not see it.

Is there a possibility to find this library to CentOS?
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From bilaln018 at gmail.com  Wed May 13 11:53:12 2015
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Wed, 13 May 2015 12:53:12 +0500
Subject: [Freeswitch-users] [mod_event_socket][pyswitch][B leg execute
	before A leg is answered]
Message-ID: <CADSVerC8kggEDLB4TARpsfu5XhFjyeOX32vEb4MWcY4dVqq=kQ@mail.gmail.com>

Hi users,

Currently i am using pyswitch to connect to freeswitch mod_event_socket.
I am originating a call and mapping its B-Leg to an IVR.
So that is A-party picks up the call,he/she will listen some IVR.
issue is before actually answering the call on A-leg  IVR starts on B-leg.
My originate command:

self.eventsocket.apiOriginate("sofia/gateway/aptcl/"+NUM+"",application="bridge",appargs=
"[DTMF="+DTMF+"]loopback/9665/default/XML",  cidname="godson", cidnum="123"
,channelvars={"origination_uuid":uid,"DTMF":DTMF})

I am bridging the B-Leg with local extension 9665 where the IVR is mapped.
can some one help me to sort-out the issue.
Your help is highly appropriated.

Freeswitch version:
FreeSWITCH Version 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git
4eed221 2015-03-12 18:55:23Z 64bit)



Regards
Bilal Abbasi
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From tculjaga at gmail.com  Wed May 13 13:02:14 2015
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 13 May 2015 11:02:14 +0200
Subject: [Freeswitch-users] T.30 faxing - V.27ter
In-Reply-To: <CAJoKKdX+L9GjLasaecL5L1SrJt7UbVqitLk8Syng8mZTAyL9tw@mail.gmail.com>
References: <CAJoKKdXF9v6BKsiJQrOov9AD4aktFT4wMA+xHH6XuMnJAYu=Dg@mail.gmail.com>
	<CAEJMVkBAvp9Vxv3-cZgVKdxVi1m7H_03GDTL9iVpM9fUiSQqXg@mail.gmail.com>
	<CAJoKKdUZq-6nJ+XPXCMXLbPaxNz3bfkZMw=_cAs7xT0FzZn63A@mail.gmail.com>
	<CAJoKKdX+L9GjLasaecL5L1SrJt7UbVqitLk8Syng8mZTAyL9tw@mail.gmail.com>
Message-ID: <CAJoKKdXO5jnLLuK9Wo=vjY9roPxi4QHLViBjA7qdqLbdanAjyA@mail.gmail.com>

can anyone help here ?

please

On Fri, May 1, 2015 at 1:29 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> does anyone have a clue ?
>
> On Wed, Apr 22, 2015 at 11:53 AM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
>
>> you are right!
>>
>> trouble making fax machine is: Samsung Xpress M2875FD
>>
>> On Wed, Apr 22, 2015 at 10:40 AM, Brian West <brian at freeswitch.org>
>> wrote:
>>
>>> JIRA FS-7453:  It's helpful to reference it in your post so others can
>>> comment on it too, keeping all relevant info in one place.
>>>
>>> Thanks,
>>> /b
>>>
>>>
>>> On Wednesday, April 22, 2015, Tihomir Culjaga <tculjaga at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> i got an issue with faxing between a fax machine and FreeSWITCH T.30
>>>> protocol.
>>>>
>>>> I'm able to send and get faxes everywhere except to this machine.
>>>> The fax machine uses V.27ter protocol at 4800 and no ECM.
>>>> faxing from a different fax machine works perfectly but not when i send
>>>> it from FS.
>>>>
>>>> FS logs are here: https://pastebin.freeswitch.org/24139
>>>>
>>>> I extracted audio
>>>>  and tried to decode it with fax_decode ( from spends/tests ) and it
>>>> fails to decode as well.
>>>> Also, i tried to decode the audio between two fax machines when the fax
>>>> went ok and it fax_decode fails on this one as well.
>>>>
>>>> fax_decode logs:
>>>> https://pastebin.freeswitch.org/24140
>>>>
>>>>
>>>> so maybe we have an issue with V.27ter perhaps.
>>>>
>>>> can anyone look say a bit more on this issue ?
>>>>
>>>> P.S.: FreeSWITCH Version 1.4.18+git~20150312T185523Z~4eed221b69~32bit
>>>> (git 4eed221 2015-03-12 18:55:23Z 32bit)
>>>>
>>>>
>>>> regards,
>>>> Tihomir.
>>>>
>>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 manavid at gmail.com  Wed May 13 13:05:40 2015
From: manavid at gmail.com (Moe Navid)
Date: Wed, 13 May 2015 02:05:40 -0700
Subject: [Freeswitch-users] [SOS] Help Needed with Strange Behavior
In-Reply-To: <CAKbxfG8Ntqb7CwZmq38njhx2N1zpOOokD8JGiTBpinFsRv+Erw@mail.gmail.com>
References: <BC8448F5-A6EE-4AE9-9930-454C7053350C@gmail.com>
	<CAKbxfG8Ntqb7CwZmq38njhx2N1zpOOokD8JGiTBpinFsRv+Erw@mail.gmail.com>
Message-ID: <CACOMF+1CHHWNW+_4NDfKSK74C6RyN1i+2S=DuNCeNKhn21dTHg@mail.gmail.com>

Hi Anthony,

Thank you very much for your response.

After almost 9 hours of debugging I was able to reproduce the issue.

I went through all our logs and found out whenever a particular customer
calls out, we loose connectivity with our router.

I found out the customer had moved his IP Phone from his office to another
location. That location has same private IP address as our servers.

On all our servers we use 10.0.0.0/24 on eth0 and eth1 has public IP
address.

What I did next was isolated a FreeSWITCH server and spawn up a dedicated
Kamailio to only send traffic to this FS server. I also changed my lab's
LAN from 172.16.16.0/24 to 10.0.0.0/24.

Here is what happens when I make a call:

First call is fine and I get:

switch_rtp.c:5846 Auto Changing port from 10.0.0.179:61856 to
108.178.144.243:10249

On second call, here is what happens:

The value of rtp_session->flags[SWITCH_RTP_FLAG_AUTOADJ] is correctly set
to 1 but the ?bytes" variable is always at zero for following if statement
on line 5833 in switch_rtp.c:

if (bytes && rtp_session->flags[SWITCH_RTP_FLAG_AUTOADJ] &&
switch_sockaddr_get_port(rtp_session->from_addr))

I also did raw packet captures on eth1, I found out right after my FS
receives SIP 183 with SDP from carrier, it sends out following:

1) an ICMP to my public IP address which is 108.178.144.243

2) an ARP broadcast ?Who has 10.0.0.179? Tell 208.65.xxx.xxx?
(208.65.xxx.xxx is the public IP of the FS server)

Right after that, the router is no longer reachable and 10.0.0.179 will
show up in arp table of FS with mac address of the router.

I tested above scenario zillion times to make sure it?s reproducible before
reporting back.

What I did next was installing rtpproxy and made few changes to our
Kamailio to rewrite SDP and pass the traffic through rtpproxy and when I
make calls with rtpproxy in the middle, the problem does not occur.

Any thoughts?

Thank you,

Moe
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From adam.ben.ayoun1 at gmail.com  Wed May 13 13:33:24 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Wed, 13 May 2015 12:33:24 +0300
Subject: [Freeswitch-users] Improving voice quality
In-Reply-To: <CAC8JMO+ex15eFrvocUaDqRoahooY3hr3jy6MpG=PyE45KP=jHg@mail.gmail.com>
References: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
	<CAAemr3zp2=iwozjSRxvMbVKR=zBn6fpEHuPbTx8YSeA9J9Craw@mail.gmail.com>
	<CAC8JMO+ex15eFrvocUaDqRoahooY3hr3jy6MpG=PyE45KP=jHg@mail.gmail.com>
Message-ID: <CADNqFCksvCst1xW4=5ZZxMHkKP8WiCdi-KkBZ7iwpC1zqy5uqw@mail.gmail.com>

Yes, it's a good point, energy level does produce choppiness, but we are
using 0 for some time now, we also tried audio-always and it's pretty much
the same. In addition, we are using iSAC at 16000h since it gave the least
choppiness compared to OPUS/others.

On 13 May 2015 at 03:15, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Bob has a good point.  Not sure about the flag but I do have energy-level
> set to 0 in my conference.conf.xml
>
> <param name="energy-level" value="0"/>
>
>
> On Tue, May 12, 2015 at 3:29 PM, Bob Hartwig <bobjectsfreeswitch at gmail.com
> > wrote:
>
>> "for example, there are small cut-offs every now and then"
>>
>> Have you tried setting "conference-flags" to "audio-always" in
>> conference.conf.xml?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From adam.ben.ayoun1 at gmail.com  Wed May 13 13:54:06 2015
From: adam.ben.ayoun1 at gmail.com (Adam Ben-Ayoun)
Date: Wed, 13 May 2015 12:54:06 +0300
Subject: [Freeswitch-users] Improving voice quality
In-Reply-To: <CADNqFCksvCst1xW4=5ZZxMHkKP8WiCdi-KkBZ7iwpC1zqy5uqw@mail.gmail.com>
References: <CADNqFCnJ036AUGFjs2h0TsKajXEiXQzpPT7LbJOhFs1u0NWaOw@mail.gmail.com>
	<CAAemr3zp2=iwozjSRxvMbVKR=zBn6fpEHuPbTx8YSeA9J9Craw@mail.gmail.com>
	<CAC8JMO+ex15eFrvocUaDqRoahooY3hr3jy6MpG=PyE45KP=jHg@mail.gmail.com>
	<CADNqFCksvCst1xW4=5ZZxMHkKP8WiCdi-KkBZ7iwpC1zqy5uqw@mail.gmail.com>
Message-ID: <CADNqFCk6xLhZwmc5eEPzVtGCtgSxAyXOYEzSLNwvoyznmgRXiQ@mail.gmail.com>

Here is the conference config in case there are some low hanging fruits in
this area:

<profile name="test">
      <param name="domain" value="$${domain}"/>
      <param name="rate" value="16000"/>
      <param name="interval" value="30"/>
      <param name="energy-level" value="0"/>
      <param name="muted-sound" value="conference/conf-muted.wav"/>
      <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>
      <param name="moh-sound" value="$${hold_music}"/>
      <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/>
      <param name="exit-sound"
value="tone_stream://%(500,0,300,200,100,50,25)"/>
      <param name="kicked-sound" value="conference/conf-kicked.wav"/>
      <param name="locked-sound" value="conference/conf-locked.wav"/>
      <param name="is-locked-sound" value="conference/conf-is-locked.wav"/>
      <param name="is-unlocked-sound"
value="conference/conf-is-unlocked.wav"/>
      <param name="pin-sound" value="conference/conf-pin.wav"/>
      <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/>
      <param name="caller-id-name" value="$${outbound_caller_name}"/>
      <param name="caller-id-number" value="$${outbound_caller_id}"/>
      <param name="comfort-noise" value="0"/>
      <param name="conference-flags" value="audio-always"/>
    </profile>

On 13 May 2015 at 12:33, Adam Ben-Ayoun <adam.ben.ayoun1 at gmail.com> wrote:

> Yes, it's a good point, energy level does produce choppiness, but we are
> using 0 for some time now, we also tried audio-always and it's pretty much
> the same. In addition, we are using iSAC at 16000h since it gave the least
> choppiness compared to OPUS/others.
>
> On 13 May 2015 at 03:15, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>> Bob has a good point.  Not sure about the flag but I do have energy-level
>> set to 0 in my conference.conf.xml
>>
>> <param name="energy-level" value="0"/>
>>
>>
>> On Tue, May 12, 2015 at 3:29 PM, Bob Hartwig <
>> bobjectsfreeswitch at gmail.com> wrote:
>>
>>> "for example, there are small cut-offs every now and then"
>>>
>>> Have you tried setting "conference-flags" to "audio-always" in
>>> conference.conf.xml?
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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 May 13 13:59:40 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 13 May 2015 12:59:40 +0300
Subject: [Freeswitch-users] [SOS] Help Needed with Strange Behavior
In-Reply-To: <CACOMF+1CHHWNW+_4NDfKSK74C6RyN1i+2S=DuNCeNKhn21dTHg@mail.gmail.com>
References: <BC8448F5-A6EE-4AE9-9930-454C7053350C@gmail.com>
	<CAKbxfG8Ntqb7CwZmq38njhx2N1zpOOokD8JGiTBpinFsRv+Erw@mail.gmail.com>
	<CACOMF+1CHHWNW+_4NDfKSK74C6RyN1i+2S=DuNCeNKhn21dTHg@mail.gmail.com>
Message-ID: <CAHtxdDe+L=2mQhTcMkukUi+JZ5YVxXTyhgC=01ar696-vXe3NQ@mail.gmail.com>

According you routing table 10.0.0.0/24 network is reachable via eth0. ARP
request must be send via eth0 interface.
Thinking is required to find why arp is send via "external" LAN segment to
router.
May be firewall rules, VLAN configuration mistake in LAN switch and etc.

Also I recommend disable ARP proxy feature on router. Router must not
respond to ARP request for not owned IP address.

Sergey Safarov

On Wed, May 13, 2015 at 12:05 PM, Moe Navid <manavid at gmail.com> wrote:

> Hi Anthony,
>
> Thank you very much for your response.
>
> After almost 9 hours of debugging I was able to reproduce the issue.
>
> I went through all our logs and found out whenever a particular customer
> calls out, we loose connectivity with our router.
>
> I found out the customer had moved his IP Phone from his office to another
> location. That location has same private IP address as our servers.
>
> On all our servers we use 10.0.0.0/24 on eth0 and eth1 has public IP
> address.
>
> What I did next was isolated a FreeSWITCH server and spawn up a dedicated
> Kamailio to only send traffic to this FS server. I also changed my lab's
> LAN from 172.16.16.0/24 to 10.0.0.0/24.
>
> Here is what happens when I make a call:
>
> First call is fine and I get:
>
> switch_rtp.c:5846 Auto Changing port from 10.0.0.179:61856 to
> 108.178.144.243:10249
>
> On second call, here is what happens:
>
> The value of rtp_session->flags[SWITCH_RTP_FLAG_AUTOADJ] is correctly set
> to 1 but the ?bytes" variable is always at zero for following if statement
> on line 5833 in switch_rtp.c:
>
> if (bytes && rtp_session->flags[SWITCH_RTP_FLAG_AUTOADJ] &&
> switch_sockaddr_get_port(rtp_session->from_addr))
>
> I also did raw packet captures on eth1, I found out right after my FS
> receives SIP 183 with SDP from carrier, it sends out following:
>
> 1) an ICMP to my public IP address which is 108.178.144.243
>
> 2) an ARP broadcast ?Who has 10.0.0.179? Tell 208.65.xxx.xxx?
> (208.65.xxx.xxx is the public IP of the FS server)
>
> Right after that, the router is no longer reachable and 10.0.0.179 will
> show up in arp table of FS with mac address of the router.
>
> I tested above scenario zillion times to make sure it?s reproducible
> before reporting back.
>
> What I did next was installing rtpproxy and made few changes to our
> Kamailio to rewrite SDP and pass the traffic through rtpproxy and when I
> make calls with rtpproxy in the middle, the problem does not occur.
>
> Any thoughts?
>
> Thank you,
>
> Moe
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mietekszczesniak2503 at gmail.com  Wed May 13 14:58:24 2015
From: mietekszczesniak2503 at gmail.com (=?UTF-8?Q?Mietek_Sze=C5=9Bniak?=)
Date: Wed, 13 May 2015 12:58:24 +0200
Subject: [Freeswitch-users] running mod_gsmopen on a separate system
Message-ID: <CADiUVxsqshYbO3M8cx6mX9q+a=piYNpvjtp+j8UOd555EufMwQ@mail.gmail.com>

I'd like to use FreeSWITCH as my private bridge between SIP and GSM.

However, I consider USB modems to be a security threat, since they
frequently have their own storage (can work as pendrives), have a black-box
baseband and are hardly ever patched.

For that reason, I'd like to connect my USB modem to a separate machine
(maybe an OpenWRT router) and connect that machine to my main FreeSWITCH
machine over Ethernet.

How would I set that up? Can mod_gsmopen be run alone, or would I have to
install a separate instance of FreeSWITCH on that second machine?

Thanks!

PS. I'm still trying to wrap my head around FreeSWITCH, so please be
forgiving. :)
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From xyangni at gmail.com  Wed May 13 15:24:33 2015
From: xyangni at gmail.com (Eric Ni)
Date: Wed, 13 May 2015 12:24:33 +0100
Subject: [Freeswitch-users] multiple registration not working
Message-ID: <CAO1DH1-QtBz9iHjP2hvQi9VL5PypYOxyvZQh+NxeB=mNeaRMUA@mail.gmail.com>

Hi,

I have just upgraded form old freeswith 1.2 to the current release version:

FreeSWITCH Version 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git
4eed221 2015-03-12 18:55:23Z 64bit)

However I can not turn on multiple registration now. After setting <param
name="multiple-registrations" value="ture"/> in sip profile. I can still
only see one contact in list_users. Can anyone help with this? Thanks.
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From s.safarov at gmail.com  Wed May 13 15:31:34 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 13 May 2015 14:31:34 +0300
Subject: [Freeswitch-users] running mod_gsmopen on a separate system
In-Reply-To: <CADiUVxsqshYbO3M8cx6mX9q+a=piYNpvjtp+j8UOd555EufMwQ@mail.gmail.com>
References: <CADiUVxsqshYbO3M8cx6mX9q+a=piYNpvjtp+j8UOd555EufMwQ@mail.gmail.com>
Message-ID: <CAHtxdDdStYndWXxeV-xa4YGo5xoiEq_VfwrpcYRCV0vzZ86bow@mail.gmail.com>

Thinking "install a separate instance of FreeSWITCH on that second machine"
is preferred.


On Wed, May 13, 2015 at 1:58 PM, Mietek Sze?niak <
mietekszczesniak2503 at gmail.com> wrote:

> I'd like to use FreeSWITCH as my private bridge between SIP and GSM.
>
> However, I consider USB modems to be a security threat, since they
> frequently have their own storage (can work as pendrives), have a black-box
> baseband and are hardly ever patched.
>
> For that reason, I'd like to connect my USB modem to a separate machine
> (maybe an OpenWRT router) and connect that machine to my main FreeSWITCH
> machine over Ethernet.
>
> How would I set that up? Can mod_gsmopen be run alone, or would I have to
> install a separate instance of FreeSWITCH on that second machine?
>
> Thanks!
>
> PS. I'm still trying to wrap my head around FreeSWITCH, so please be
> forgiving. :)
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed May 13 15:32:02 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 13 May 2015 13:32:02 +0200
Subject: [Freeswitch-users] running mod_gsmopen on a separate system
In-Reply-To: <CADiUVxsqshYbO3M8cx6mX9q+a=piYNpvjtp+j8UOd555EufMwQ@mail.gmail.com>
References: <CADiUVxsqshYbO3M8cx6mX9q+a=piYNpvjtp+j8UOd555EufMwQ@mail.gmail.com>
Message-ID: <CALXCt0qaR_GSg2GN7v7nwx22S3DpUR3ra-xa+rhtZnb0VhKHhw@mail.gmail.com>

A separate instance of Freeswitch on second machine, and two machines talk
each other via sip.
-giovanni

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 13, 2015 1:22 PM, "Mietek Sze?niak" <mietekszczesniak2503 at gmail.com>
wrote:

> I'd like to use FreeSWITCH as my private bridge between SIP and GSM.
>
> However, I consider USB modems to be a security threat, since they
> frequently have their own storage (can work as pendrives), have a black-box
> baseband and are hardly ever patched.
>
> For that reason, I'd like to connect my USB modem to a separate machine
> (maybe an OpenWRT router) and connect that machine to my main FreeSWITCH
> machine over Ethernet.
>
> How would I set that up? Can mod_gsmopen be run alone, or would I have to
> install a separate instance of FreeSWITCH on that second machine?
>
> Thanks!
>
> PS. I'm still trying to wrap my head around FreeSWITCH, so please be
> forgiving. :)
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From raphael.lechner at gmail.com  Wed May 13 16:19:33 2015
From: raphael.lechner at gmail.com (Raphael Lechner)
Date: Wed, 13 May 2015 14:19:33 +0200
Subject: [Freeswitch-users] multiple registration not working
In-Reply-To: <CAO1DH1-QtBz9iHjP2hvQi9VL5PypYOxyvZQh+NxeB=mNeaRMUA@mail.gmail.com>
References: <CAO1DH1-QtBz9iHjP2hvQi9VL5PypYOxyvZQh+NxeB=mNeaRMUA@mail.gmail.com>
Message-ID: <60E561C2-A6EA-4EA9-945F-22E2326DB2D1@gmail.com>

Hi Eric,

Is this only a typing error in your mail or have you really set value=?ture??


> On 13 May 2015, at 13:24, Eric Ni <xyangni at gmail.com> wrote:
> 
> Hi,
> 
> I have just upgraded form old freeswith 1.2 to the current release version:
> 
> FreeSWITCH Version 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12 18:55:23Z 64bit)
> 
> However I can not turn on multiple registration now. After setting <param name="multiple-registrations" value="ture"/> in sip profile. I can still only see one contact in list_users. Can anyone help with this? Thanks.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From raphael.lechner at gmail.com  Wed May 13 16:19:33 2015
From: raphael.lechner at gmail.com (Raphael Lechner)
Date: Wed, 13 May 2015 14:19:33 +0200
Subject: [Freeswitch-users] multiple registration not working
In-Reply-To: <CAO1DH1-QtBz9iHjP2hvQi9VL5PypYOxyvZQh+NxeB=mNeaRMUA@mail.gmail.com>
References: <CAO1DH1-QtBz9iHjP2hvQi9VL5PypYOxyvZQh+NxeB=mNeaRMUA@mail.gmail.com>
Message-ID: <7D801F69-5826-4499-A085-2A911ED4283B@gmail.com>

Hi Eric,

Is this only a typing error in your mail or have you really set value=?ture??


> On 13 May 2015, at 13:24, Eric Ni <xyangni at gmail.com> wrote:
> 
> Hi,
> 
> I have just upgraded form old freeswith 1.2 to the current release version:
> 
> FreeSWITCH Version 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12 18:55:23Z 64bit)
> 
> However I can not turn on multiple registration now. After setting <param name="multiple-registrations" value="ture"/> in sip profile. I can still only see one contact in list_users. Can anyone help with this? Thanks.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From shogun at tausys.de  Wed May 13 16:33:20 2015
From: shogun at tausys.de (Jens Tautenhahn)
Date: Wed, 13 May 2015 14:33:20 +0200
Subject: [Freeswitch-users] CDR problem
In-Reply-To: <CAP9PqkTXX0Enna6na1ACMFCVcEiCnuXLiak7r4gvRHz2AjJZsg@mail.gmail.com>
References: <55525F86.6080306@tausys.de>
	<CAP9PqkTXX0Enna6na1ACMFCVcEiCnuXLiak7r4gvRHz2AjJZsg@mail.gmail.com>
Message-ID: <55534490.4090500@tausys.de>

Thanks for your help! I have found the error.

In "v_xml_cdr_import.php" the variable "sip_P-Preferred-Identity" would
not be masked. The user can pass there anything. Without masking this
leads to not correct XML.

Where can I submit a bug report or patch?


From xyangni at gmail.com  Wed May 13 16:37:24 2015
From: xyangni at gmail.com (Eric Ni)
Date: Wed, 13 May 2015 13:37:24 +0100
Subject: [Freeswitch-users] multiple registration not working
In-Reply-To: <60E561C2-A6EA-4EA9-945F-22E2326DB2D1@gmail.com>
References: <CAO1DH1-QtBz9iHjP2hvQi9VL5PypYOxyvZQh+NxeB=mNeaRMUA@mail.gmail.com>
	<60E561C2-A6EA-4EA9-945F-22E2326DB2D1@gmail.com>
Message-ID: <CAO1DH1-zxWi8bzhEZS73XiJZ7Y-KD6e4EmZ_S5iDrQecv01JEA@mail.gmail.com>

Hi Raphae,

It is my typo in the xml file. Will correct it and test again tonight.
Thanks for pointing out.

On Wed, May 13, 2015 at 1:19 PM, Raphael Lechner <raphael.lechner at gmail.com>
wrote:

> Hi Eric,
>
> Is this only a typing error in your mail or have you really set
> value=?ture??
>
>
> > On 13 May 2015, at 13:24, Eric Ni <xyangni at gmail.com> wrote:
> >
> > Hi,
> >
> > I have just upgraded form old freeswith 1.2 to the current release
> version:
> >
> > FreeSWITCH Version 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git
> 4eed221 2015-03-12 18:55:23Z 64bit)
> >
> > However I can not turn on multiple registration now. After setting
> <param name="multiple-registrations" value="ture"/> in sip profile. I can
> still only see one contact in list_users. Can anyone help with this? Thanks.
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ashwinrath at gmail.com  Wed May 13 16:52:14 2015
From: ashwinrath at gmail.com (Ashwin Rath)
Date: Wed, 13 May 2015 18:22:14 +0530
Subject: [Freeswitch-users] Invoke SOAP service on mod_callcenter ring
Message-ID: <CAA_Nwu4wjAEgpaWBTVvQKmjFW46XA3sCqwUmROkbF1fUJM6r+Q@mail.gmail.com>

I was wondering is there some way to call a SOAP webservice using curl with
info such as the agent number / name to which mod_callcenter connects a
call ?



-- 
Ashwin Kumar Rath
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From krice at freeswitch.org  Wed May 13 17:00:00 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 13 May 2015 08:00:00 -0500
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <1251442930.20150513084828@seznam.cz>
Message-ID: <D178B500.9DA9B%krice@freeswitch.org>

There is a massive shift in the number of dependencies for compiling
FreeSWITCH. We are working on getting that dep chain working on non-debian
platforms but that work is not yet complete.

If you wish to try and report back to the list the steps you took to get it
working on Centos7 so that others may benefit, most of the deps that you
cant get from yum are located at https://freeswitch.org/stash/projects/SD/

This may not be an all inclusive list for building on non-debian systems but
it will get you close.




On 5/13/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz> wrote:

> Dobr? den,
> 
> This version 7, yes. This version is somehow different?

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From sertys at gmail.com  Wed May 13 17:02:04 2015
From: sertys at gmail.com (Daniel Ivanov)
Date: Wed, 13 May 2015 16:02:04 +0300
Subject: [Freeswitch-users] running mod_gsmopen on a separate system
In-Reply-To: <CALXCt0qaR_GSg2GN7v7nwx22S3DpUR3ra-xa+rhtZnb0VhKHhw@mail.gmail.com>
References: <CADiUVxsqshYbO3M8cx6mX9q+a=piYNpvjtp+j8UOd555EufMwQ@mail.gmail.com>
	<CALXCt0qaR_GSg2GN7v7nwx22S3DpUR3ra-xa+rhtZnb0VhKHhw@mail.gmail.com>
Message-ID: <CAMeZp63Mg9EyNYduYCYrzBoSn2JwqdX9HYWTcyecrvu6avV9gQ@mail.gmail.com>

Yeah, that is what we do basically. Get our modems hooked to a raspberry pi
with freeswitch and route the calls&sms via sip to the endpoints.

Giving it a little thought though gave me that :
http://usbip.sourceforge.net/

It's a simple encapsulation of usb messaging over IP which is transferred
to a kernel module acting as a virtual host controller. Haven't tested it,
but be sure to post some feedback if you give it a spin.

On Wed, May 13, 2015 at 2:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> A separate instance of Freeswitch on second machine, and two machines talk
> each other via sip.
> -giovanni
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On May 13, 2015 1:22 PM, "Mietek Sze?niak" <mietekszczesniak2503 at gmail.com>
> wrote:
>
>> I'd like to use FreeSWITCH as my private bridge between SIP and GSM.
>>
>> However, I consider USB modems to be a security threat, since they
>> frequently have their own storage (can work as pendrives), have a black-box
>> baseband and are hardly ever patched.
>>
>> For that reason, I'd like to connect my USB modem to a separate machine
>> (maybe an OpenWRT router) and connect that machine to my main FreeSWITCH
>> machine over Ethernet.
>>
>> How would I set that up? Can mod_gsmopen be run alone, or would I have to
>> install a separate instance of FreeSWITCH on that second machine?
>>
>> Thanks!
>>
>> PS. I'm still trying to wrap my head around FreeSWITCH, so please be
>> forgiving. :)
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 13 17:17:05 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 May 2015 09:17:05 -0400
Subject: [Freeswitch-users] T.30 faxing - V.27ter
In-Reply-To: <CAJoKKdXO5jnLLuK9Wo=vjY9roPxi4QHLViBjA7qdqLbdanAjyA@mail.gmail.com>
References: <CAJoKKdXF9v6BKsiJQrOov9AD4aktFT4wMA+xHH6XuMnJAYu=Dg@mail.gmail.com>
	<CAEJMVkBAvp9Vxv3-cZgVKdxVi1m7H_03GDTL9iVpM9fUiSQqXg@mail.gmail.com>
	<CAJoKKdUZq-6nJ+XPXCMXLbPaxNz3bfkZMw=_cAs7xT0FzZn63A@mail.gmail.com>
	<CAJoKKdX+L9GjLasaecL5L1SrJt7UbVqitLk8Syng8mZTAyL9tw@mail.gmail.com>
	<CAJoKKdXO5jnLLuK9Wo=vjY9roPxi4QHLViBjA7qdqLbdanAjyA@mail.gmail.com>
Message-ID: <3CCEFDD7-F6E0-4090-AC05-7F14AEFE10D3@jerris.com>

we will address this on the jira.  No movement yet, we may need more debug and such so stay tuned.

> On May 13, 2015, at 5:02 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> 
> can anyone help here ?
> 
> please
> 
> On Fri, May 1, 2015 at 1:29 PM, Tihomir Culjaga <tculjaga at gmail.com <mailto:tculjaga at gmail.com>> wrote:
> does anyone have a clue ?
> 
> On Wed, Apr 22, 2015 at 11:53 AM, Tihomir Culjaga <tculjaga at gmail.com <mailto:tculjaga at gmail.com>> wrote:
> you are right!
> 
> trouble making fax machine is: Samsung Xpress M2875FD 
> 
> On Wed, Apr 22, 2015 at 10:40 AM, Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>> wrote:
> JIRA FS-7453:  It's helpful to reference it in your post so others can comment on it too, keeping all relevant info in one place.
> 
> Thanks,
> /b
> 
> 
> On Wednesday, April 22, 2015, Tihomir Culjaga <tculjaga at gmail.com <>> wrote:
> Hi,
> 
> i got an issue with faxing between a fax machine and FreeSWITCH T.30 protocol.
> 
> I'm able to send and get faxes everywhere except to this machine.
> The fax machine uses V.27ter protocol at 4800 and no ECM.
> faxing from a different fax machine works perfectly but not when i send it from FS.
> 
> FS logs are here: https://pastebin.freeswitch.org/24139 <https://pastebin.freeswitch.org/24139>
> 
> I extracted audio 
>  and tried to decode it with fax_decode ( from spends/tests ) and it fails to decode as well.
> Also, i tried to decode the audio between two fax machines when the fax went ok and it fax_decode fails on this one as well.
> 
> fax_decode logs:
> https://pastebin.freeswitch.org/24140 <https://pastebin.freeswitch.org/24140>
> 
> 
> so maybe we have an issue with V.27ter perhaps.
> 
> can anyone look say a bit more on this issue ?
> 
> P.S.: FreeSWITCH Version 1.4.18+git~20150312T185523Z~4eed221b69~32bit (git 4eed221 2015-03-12 18:55:23Z 32bit)

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From s.safarov at gmail.com  Wed May 13 17:19:46 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 13 May 2015 16:19:46 +0300
Subject: [Freeswitch-users] running mod_gsmopen on a separate system
In-Reply-To: <CAMeZp63Mg9EyNYduYCYrzBoSn2JwqdX9HYWTcyecrvu6avV9gQ@mail.gmail.com>
References: <CADiUVxsqshYbO3M8cx6mX9q+a=piYNpvjtp+j8UOd555EufMwQ@mail.gmail.com>
	<CALXCt0qaR_GSg2GN7v7nwx22S3DpUR3ra-xa+rhtZnb0VhKHhw@mail.gmail.com>
	<CAMeZp63Mg9EyNYduYCYrzBoSn2JwqdX9HYWTcyecrvu6avV9gQ@mail.gmail.com>
Message-ID: <CAHtxdDfnSsZsG=bQcRe+y+SWG5On6nKdqygt6=+1d3V1XMEruQ@mail.gmail.com>

Passing USB messages via USBIP equal inserting dongle to FS host. It is not
solves security questions.

Sergey

On Wed, May 13, 2015 at 4:02 PM, Daniel Ivanov <sertys at gmail.com> wrote:

> Yeah, that is what we do basically. Get our modems hooked to a raspberry
> pi with freeswitch and route the calls&sms via sip to the endpoints.
>
> Giving it a little thought though gave me that :
> http://usbip.sourceforge.net/
>
> It's a simple encapsulation of usb messaging over IP which is transferred
> to a kernel module acting as a virtual host controller. Haven't tested it,
> but be sure to post some feedback if you give it a spin.
>
> On Wed, May 13, 2015 at 2:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> A separate instance of Freeswitch on second machine, and two machines
>> talk each other via sip.
>> -giovanni
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On May 13, 2015 1:22 PM, "Mietek Sze?niak" <
>> mietekszczesniak2503 at gmail.com> wrote:
>>
>>> I'd like to use FreeSWITCH as my private bridge between SIP and GSM.
>>>
>>> However, I consider USB modems to be a security threat, since they
>>> frequently have their own storage (can work as pendrives), have a black-box
>>> baseband and are hardly ever patched.
>>>
>>> For that reason, I'd like to connect my USB modem to a separate machine
>>> (maybe an OpenWRT router) and connect that machine to my main FreeSWITCH
>>> machine over Ethernet.
>>>
>>> How would I set that up? Can mod_gsmopen be run alone, or would I have
>>> to install a separate instance of FreeSWITCH on that second machine?
>>>
>>> Thanks!
>>>
>>> PS. I'm still trying to wrap my head around FreeSWITCH, so please be
>>> forgiving. :)
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sertys at gmail.com  Wed May 13 17:33:04 2015
From: sertys at gmail.com (Daniel Ivanov)
Date: Wed, 13 May 2015 16:33:04 +0300
Subject: [Freeswitch-users] running mod_gsmopen on a separate system
In-Reply-To: <CAHtxdDfnSsZsG=bQcRe+y+SWG5On6nKdqygt6=+1d3V1XMEruQ@mail.gmail.com>
References: <CADiUVxsqshYbO3M8cx6mX9q+a=piYNpvjtp+j8UOd555EufMwQ@mail.gmail.com>
	<CALXCt0qaR_GSg2GN7v7nwx22S3DpUR3ra-xa+rhtZnb0VhKHhw@mail.gmail.com>
	<CAMeZp63Mg9EyNYduYCYrzBoSn2JwqdX9HYWTcyecrvu6avV9gQ@mail.gmail.com>
	<CAHtxdDfnSsZsG=bQcRe+y+SWG5On6nKdqygt6=+1d3V1XMEruQ@mail.gmail.com>
Message-ID: <CAMeZp616r=ejd6-Xm8Se3xzpK3k1bZg8FykYAZX8v7xpF-xFqg@mail.gmail.com>

Yeah, you're correct but i can believe there can be some ACL or
whitelisting implemented on kernel module level. Keep in mind the USB modem
comm is just a bunch of serial consoles and you mitigate the problem of
DMA(direct memory access) if you just choose to relay that class.

On Wed, May 13, 2015 at 4:19 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Passing USB messages via USBIP equal inserting dongle to FS host. It is
> not solves security questions.
>
> Sergey
>
> On Wed, May 13, 2015 at 4:02 PM, Daniel Ivanov <sertys at gmail.com> wrote:
>
>> Yeah, that is what we do basically. Get our modems hooked to a raspberry
>> pi with freeswitch and route the calls&sms via sip to the endpoints.
>>
>> Giving it a little thought though gave me that :
>> http://usbip.sourceforge.net/
>>
>> It's a simple encapsulation of usb messaging over IP which is transferred
>> to a kernel module acting as a virtual host controller. Haven't tested it,
>> but be sure to post some feedback if you give it a spin.
>>
>> On Wed, May 13, 2015 at 2:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> A separate instance of Freeswitch on second machine, and two machines
>>> talk each other via sip.
>>> -giovanni
>>>
>>> sent from my mobile,
>>> Giovanni Maruzzelli
>>> cell: +39 347 266 56 18
>>> On May 13, 2015 1:22 PM, "Mietek Sze?niak" <
>>> mietekszczesniak2503 at gmail.com> wrote:
>>>
>>>> I'd like to use FreeSWITCH as my private bridge between SIP and GSM.
>>>>
>>>> However, I consider USB modems to be a security threat, since they
>>>> frequently have their own storage (can work as pendrives), have a black-box
>>>> baseband and are hardly ever patched.
>>>>
>>>> For that reason, I'd like to connect my USB modem to a separate machine
>>>> (maybe an OpenWRT router) and connect that machine to my main FreeSWITCH
>>>> machine over Ethernet.
>>>>
>>>> How would I set that up? Can mod_gsmopen be run alone, or would I have
>>>> to install a separate instance of FreeSWITCH on that second machine?
>>>>
>>>> Thanks!
>>>>
>>>> PS. I'm still trying to wrap my head around FreeSWITCH, so please be
>>>> forgiving. :)
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 13 17:34:55 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 13 May 2015 16:34:55 +0300
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <D178B500.9DA9B%krice@freeswitch.org>
References: <1251442930.20150513084828@seznam.cz>
	<D178B500.9DA9B%krice@freeswitch.org>
Message-ID: <CAHtxdDft6M-mF+=+L7jGLxPAD+EzHohYBou4O_f6rsYwCofYqg@mail.gmail.com>

I has rpm repo with all required packages to install current FS master via
yum package manager on CentOS 7. All packages ported from fedora repo.
Thinking 1.6 branch can be packaged similarly.
If anybody write Jira tiket and this tiket will be assigned to me, then I
create pull request to automate build dependences on CentOS 7.

Sergey Safarov


On Wed, May 13, 2015 at 4:00 PM, Ken Rice <krice at freeswitch.org> wrote:

>  There is a massive shift in the number of dependencies for compiling
> FreeSWITCH. We are working on getting that dep chain working on non-debian
> platforms but that work is not yet complete.
>
> If you wish to try and report back to the list the steps you took to get
> it working on Centos7 so that others may benefit, most of the deps that you
> cant get from yum are located at https://freeswitch.org/stash/projects/SD/
>
> This may not be an all inclusive list for building on non-debian systems
> but it will get you close.
>
>
>
>
> On 5/13/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz> wrote:
>
> Dobr? den,
>
> This version 7, yes. This version is somehow different?
>
>
> --
> Ken
>
>
>
> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 13 17:48:51 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 13 May 2015 15:48:51 +0200
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <CAHtxdDft6M-mF+=+L7jGLxPAD+EzHohYBou4O_f6rsYwCofYqg@mail.gmail.com>
References: <1251442930.20150513084828@seznam.cz>
	<D178B500.9DA9B%krice@freeswitch.org> 
	<CAHtxdDft6M-mF+=+L7jGLxPAD+EzHohYBou4O_f6rsYwCofYqg@mail.gmail.com>
Message-ID: <1403537106.20150513154851@seznam.cz>

Dobr? den,

I never did, but I can try. How does it work?

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

st?eda 13. kv?tna 2015, 15:34:55, napsal jste:


I has rpm repo with all required packages to install current FS master via yum package manager on CentOS 7. All packages ported from fedora repo.
Thinking 1.6 branch can be packaged similarly.
If anybody write Jira tiket and this tiket will be assigned to me, then I create pull request to automate build dependences on CentOS 7.

Sergey Safarov
 

On Wed, May 13, 2015 at 4:00 PM, Ken Rice <krice at freeswitch.org> wrote:
There is a massive shift in the number of dependencies for compiling FreeSWITCH. We are working on getting that dep chain working on non-debian platforms but that work is not yet complete.

If you wish to try and report back to the list the steps you took to get it working on Centos7 so that others may benefit, most of the deps that you cant get from yum are located at https://freeswitch.org/stash/projects/SD/

This may not be an all inclusive list for building on non-debian systems but it will get you close.




On 5/13/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz> wrote:

Dobr? den,

This version 7, yes. This version is somehow different?

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 May 13 17:54:59 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 13 May 2015 08:54:59 -0500
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <CAHtxdDft6M-mF+=+L7jGLxPAD+EzHohYBou4O_f6rsYwCofYqg@mail.gmail.com>
Message-ID: <D178C1E3.9DAA4%krice@freeswitch.org>

Theres already stuff in the works... If you have spec files for these
packages by all means please visit the projects in stash and create pull
requests for the spec files! You can also create your own jira.


On 5/13/15, 8:34 AM, "Sergey Safarov" <s.safarov at gmail.com> wrote:

> I has rpm repo with all required packages to install current FS master via yum
> package manager on CentOS 7. All packages ported from fedora repo.
> Thinking 1.6 branch can be packaged similarly.
> If anybody write Jira tiket and this tiket will be assigned to me, then I
> create pull request to automate build dependences on CentOS 7.
> 
> Sergey Safarov
> ?
> 
> On Wed, May 13, 2015 at 4:00 PM, Ken Rice <krice at freeswitch.org> wrote:
>> There is a massive shift in the number of dependencies for compiling
>> FreeSWITCH. We are working on getting that dep chain working on non-debian
>> platforms but that work is not yet complete.
>> 
>> If you wish to try and report back to the list the steps you took to get it
>> working on Centos7 so that others may benefit, most of the deps that you cant
>> get from yum are located at https://freeswitch.org/stash/projects/SD/
>> 
>> This may not be an all inclusive list for building on non-debian systems but
>> it will get you close.
>> 
>> 
>> 
>> 
>> On 5/13/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz
>> <http://yadenis at seznam.cz> > wrote:
>> 
>>> Dobr? den,
>>> 
>>> This version 7, yes. This version is somehow different?

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From yadenis at seznam.cz  Wed May 13 17:59:08 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Wed, 13 May 2015 15:59:08 +0200
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <CAHtxdDft6M-mF+=+L7jGLxPAD+EzHohYBou4O_f6rsYwCofYqg@mail.gmail.com>
References: <1251442930.20150513084828@seznam.cz>
	<D178B500.9DA9B%krice@freeswitch.org> 
	<CAHtxdDft6M-mF+=+L7jGLxPAD+EzHohYBou4O_f6rsYwCofYqg@mail.gmail.com>
Message-ID: <1641553624.20150513155908@seznam.cz>

Dobr? den,

I hope I did it right

https://freeswitch.org/jira/browse/FS-7553 

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

st?eda 13. kv?tna 2015, 15:34:55, napsal jste:


I has rpm repo with all required packages to install current FS master via yum package manager on CentOS 7. All packages ported from fedora repo.
Thinking 1.6 branch can be packaged similarly.
If anybody write Jira tiket and this tiket will be assigned to me, then I create pull request to automate build dependences on CentOS 7.

Sergey Safarov
 

On Wed, May 13, 2015 at 4:00 PM, Ken Rice <krice at freeswitch.org> wrote:
There is a massive shift in the number of dependencies for compiling FreeSWITCH. We are working on getting that dep chain working on non-debian platforms but that work is not yet complete.

If you wish to try and report back to the list the steps you took to get it working on Centos7 so that others may benefit, most of the deps that you cant get from yum are located at https://freeswitch.org/stash/projects/SD/

This may not be an all inclusive list for building on non-debian systems but it will get you close.




On 5/13/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz> wrote:

Dobr? den,

This version 7, yes. This version is somehow different?

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From mietekszczesniak2503 at gmail.com  Wed May 13 18:10:28 2015
From: mietekszczesniak2503 at gmail.com (=?UTF-8?Q?Mietek_Sze=C5=9Bniak?=)
Date: Wed, 13 May 2015 16:10:28 +0200
Subject: [Freeswitch-users] running mod_gsmopen on a separate system
In-Reply-To: <CAMeZp616r=ejd6-Xm8Se3xzpK3k1bZg8FykYAZX8v7xpF-xFqg@mail.gmail.com>
References: <CADiUVxsqshYbO3M8cx6mX9q+a=piYNpvjtp+j8UOd555EufMwQ@mail.gmail.com>
	<CALXCt0qaR_GSg2GN7v7nwx22S3DpUR3ra-xa+rhtZnb0VhKHhw@mail.gmail.com>
	<CAMeZp63Mg9EyNYduYCYrzBoSn2JwqdX9HYWTcyecrvu6avV9gQ@mail.gmail.com>
	<CAHtxdDfnSsZsG=bQcRe+y+SWG5On6nKdqygt6=+1d3V1XMEruQ@mail.gmail.com>
	<CAMeZp616r=ejd6-Xm8Se3xzpK3k1bZg8FykYAZX8v7xpF-xFqg@mail.gmail.com>
Message-ID: <CADiUVxs-AgyDghVw12WSMKtTRhG5g3j0c8XitGbsA0uXrjG+hA@mail.gmail.com>

Thanks a lot, guys!



On Wed, May 13, 2015 at 3:33 PM, Daniel Ivanov <sertys at gmail.com> wrote:

> Yeah, you're correct but i can believe there can be some ACL or
> whitelisting implemented on kernel module level. Keep in mind the USB modem
> comm is just a bunch of serial consoles and you mitigate the problem of
> DMA(direct memory access) if you just choose to relay that class.
>
> On Wed, May 13, 2015 at 4:19 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Passing USB messages via USBIP equal inserting dongle to FS host. It is
>> not solves security questions.
>>
>> Sergey
>>
>> On Wed, May 13, 2015 at 4:02 PM, Daniel Ivanov <sertys at gmail.com> wrote:
>>
>>> Yeah, that is what we do basically. Get our modems hooked to a raspberry
>>> pi with freeswitch and route the calls&sms via sip to the endpoints.
>>>
>>> Giving it a little thought though gave me that :
>>> http://usbip.sourceforge.net/
>>>
>>> It's a simple encapsulation of usb messaging over IP which is
>>> transferred to a kernel module acting as a virtual host controller. Haven't
>>> tested it, but be sure to post some feedback if you give it a spin.
>>>
>>> On Wed, May 13, 2015 at 2:32 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> A separate instance of Freeswitch on second machine, and two machines
>>>> talk each other via sip.
>>>> -giovanni
>>>>
>>>> sent from my mobile,
>>>> Giovanni Maruzzelli
>>>> cell: +39 347 266 56 18
>>>> On May 13, 2015 1:22 PM, "Mietek Sze?niak" <
>>>> mietekszczesniak2503 at gmail.com> wrote:
>>>>
>>>>> I'd like to use FreeSWITCH as my private bridge between SIP and GSM.
>>>>>
>>>>> However, I consider USB modems to be a security threat, since they
>>>>> frequently have their own storage (can work as pendrives), have a black-box
>>>>> baseband and are hardly ever patched.
>>>>>
>>>>> For that reason, I'd like to connect my USB modem to a separate
>>>>> machine (maybe an OpenWRT router) and connect that machine to my main
>>>>> FreeSWITCH machine over Ethernet.
>>>>>
>>>>> How would I set that up? Can mod_gsmopen be run alone, or would I have
>>>>> to install a separate instance of FreeSWITCH on that second machine?
>>>>>
>>>>> Thanks!
>>>>>
>>>>> PS. I'm still trying to wrap my head around FreeSWITCH, so please be
>>>>> forgiving. :)
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From simpot at gmail.com  Wed May 13 18:13:45 2015
From: simpot at gmail.com (Dmitry Saratsky)
Date: Wed, 13 May 2015 17:13:45 +0300
Subject: [Freeswitch-users] DNS resolving is not checking /etc/hosts
Message-ID: <CAM2iYy5yH+GK4j1QH6dAKfV=D5qn55_q_dB7Vk=DgaFyN94DEQ@mail.gmail.com>

Hi,

I have some gateway configured on my FS by hostname:
17:08 fs:~# grep proxy /usr/local/freeswitch/conf/sip_profiles/ext-ipv4/20*
    <param name="proxy" value="sip.domain.com"/>
17:09 fs:~#

This domain is not resolvable over internet, so I configured ip for this
domain in system's (CentOS 6.6 x64) /etc/hosts file.

For some reason it looks like FS is not checking this file, but trying to
resolve the host directly from DNS servers configured in /etc/resolv.conf...

Is there any way to tell FS to check /etc/hosts as well?

Thanks,
Dmitry.
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From brian at freeswitch.org  Wed May 13 19:21:47 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 May 2015 10:21:47 -0500
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <1641553624.20150513155908@seznam.cz>
References: <1251442930.20150513084828@seznam.cz>
	<D178B500.9DA9B%krice@freeswitch.org>
	<CAHtxdDft6M-mF+=+L7jGLxPAD+EzHohYBou4O_f6rsYwCofYqg@mail.gmail.com>
	<1641553624.20150513155908@seznam.cz>
Message-ID: <CAEJMVkBk-+YMRYGXQnH+Eu_GT571BzKNfrkp4zGDRmVabKDSgg@mail.gmail.com>

I've assigned it to Sergey Safarov, So lets get to busy ;)

On Wed, May 13, 2015 at 8:59 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> I hope I did it right
>
> https://freeswitch.org/jira/browse/FS-7553
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 st?eda 13. kv?tna 2015, 15:34:55, napsal
> jste: *
>
>  I has rpm repo with all required packages to install current FS master
> via yum package manager on CentOS 7. All packages ported from fedora repo.
> Thinking 1.6 branch can be packaged similarly.
> If anybody write Jira tiket and this tiket will be assigned to me, then I
> create pull request to automate build dependences on CentOS 7.
>
> Sergey Safarov
>
>
> On Wed, May 13, 2015 at 4:00 PM, Ken Rice <krice at freeswitch.org> wrote:
> There is a massive shift in the number of dependencies for compiling
> FreeSWITCH. We are working on getting that dep chain working on non-debian
> platforms but that work is not yet complete.
>
> If you wish to try and report back to the list the steps you took to get
> it working on Centos7 so that others may benefit, most of the deps that you
> cant get from yum are located at https://freeswitch.org/stash/projects/SD/
>
> This may not be an all inclusive list for building on non-debian systems
> but it will get you close.
>
>
>
>
> On 5/13/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz> wrote:
>
> Dobr? den,
>
> This version 7, yes. This version is somehow different?
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.FreeSWITCH.org>
> http://confluence.freeswitch.org
> http://www.cluecon.com <http://www.ClueCon.com>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> 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
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From ssinyagin at gmail.com  Wed May 13 21:29:08 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 13 May 2015 19:29:08 +0200
Subject: [Freeswitch-users] DNS resolving is not checking /etc/hosts
In-Reply-To: <CAM2iYy5yH+GK4j1QH6dAKfV=D5qn55_q_dB7Vk=DgaFyN94DEQ@mail.gmail.com>
References: <CAM2iYy5yH+GK4j1QH6dAKfV=D5qn55_q_dB7Vk=DgaFyN94DEQ@mail.gmail.com>
Message-ID: <CACOYK=iwv9HFS2xvmb9QsHGHK6=2T-K1+TNDzje_OZY0OaaNiw@mail.gmail.com>

You can use "realm" parameter as domain name, and put the IP address in
"proxy".
On May 13, 2015 4:14 PM, "Dmitry Saratsky" <simpot at gmail.com> wrote:

> Hi,
>
> I have some gateway configured on my FS by hostname:
> 17:08 fs:~# grep proxy /usr/local/freeswitch/conf/sip_profiles/ext-ipv4/20*
>     <param name="proxy" value="sip.domain.com"/>
> 17:09 fs:~#
>
> This domain is not resolvable over internet, so I configured ip for this
> domain in system's (CentOS 6.6 x64) /etc/hosts file.
>
> For some reason it looks like FS is not checking this file, but trying to
> resolve the host directly from DNS servers configured in /etc/resolv.conf...
>
> Is there any way to tell FS to check /etc/hosts as well?
>
> Thanks,
> Dmitry.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed May 13 22:44:05 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Wed, 13 May 2015 11:44:05 -0700
Subject: [Freeswitch-users] Error during the installation of esl luamod
Message-ID: <CAGVOjVVmMN7r1gKP-HLv+6JuRd4Z5-Fp0F4JvvZEBYHZQbzd_w@mail.gmail.com>

Hi users.
During esl luamod installation I see this error.







*esl_wrap.cpp:746:17: fatal error: lua.h: No such file or
directory #include "lua.h"                 ^compilation terminated.make[1]:
*** [esl_wrap.o] Error 1make[1]: Leaving directory
`/usr/src/freeswitch/libs/esl/lua'make: *** [luamod] Error 2*

But running '*find lua.h*' I see file is located at '
*/usr/include/lua5.2/lua.h*'

I guess some of you guys had been gone through this.

How did you resolve this?
Any pointer would be much appreciated.
Thanks.
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From mike at jerris.com  Wed May 13 23:22:06 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 May 2015 15:22:06 -0400
Subject: [Freeswitch-users] Error during the installation of esl luamod
In-Reply-To: <CAGVOjVVmMN7r1gKP-HLv+6JuRd4Z5-Fp0F4JvvZEBYHZQbzd_w@mail.gmail.com>
References: <CAGVOjVVmMN7r1gKP-HLv+6JuRd4Z5-Fp0F4JvvZEBYHZQbzd_w@mail.gmail.com>
Message-ID: <FEB7FD32-A79C-4BFE-BE0F-2B92AD0C13D5@jerris.com>

in master, try building with the --enable-system-lua configure flag.  This will use pkg-config to find the cflags to get to the headers.  In 1.6 this will be the behavior always.

  
> On May 13, 2015, at 2:44 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
> 
> Hi users. 
> During esl luamod installation I see this error. 
> 
> esl_wrap.cpp:746:17: fatal error: lua.h: No such file or directory
>  #include "lua.h"
>                  ^
> compilation terminated.
> make[1]: *** [esl_wrap.o] Error 1
> make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/lua'
> make: *** [luamod] Error 2
> 
> But running 'find lua.h' I see file is located at '/usr/include/lua5.2/lua.h'
> 
> I guess some of you guys had been gone through this.
> 
> How did you resolve this? 
> Any pointer would be much appreciated.
> Thanks.

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From steven.szeto at mitel.com  Thu May 14 01:26:37 2015
From: steven.szeto at mitel.com (Szeto, Steven)
Date: Wed, 13 May 2015 17:26:37 -0400
Subject: [Freeswitch-users] How to disable DTMF digit processing when a call
	is valet parked?
Message-ID: <CALi6oUzeu-usM=qHQipvroD5-fMV5ZppWyfqrt=h7E6pcofMFw@mail.gmail.com>

Suppose I have a public dial plan entry that looks like this:

    <extension name="1201">
      <condition field="destination_number" expression="1201">
        <action application="export" data="x-MitelIsACD = true" />
        <action application="playback"
data="{loops=1}tone_stream://${us-ring}" />
    <action application="valet_park" data="QueuedCalls auto in 900000
902000" />
      </condition>
    </extension>

I have an IVR application that listens to DTMF keystrokes from the valet
parked call.

I want to disable DTMF processing by freeswitch for calls that are valet
parked.

The main reason for disabling DTMF is that the caller can press "#" (end of
dial character), and freeswitch will terminate the call for the following
reason:

2015-05-13 17:13:28.729128 [NOTICE] switch_core_state_machine.c:315
sofia/internal/3006 at 10.47.32.159 has executed the last dialplan
instruction, hanging up.
2015-05-13 17:13:28.729128 [NOTICE] switch_core_state_machine.c:317 Hangup
sofia/internal/3006 at 10.47.32.159 [CS_EXECUTE] [NORMAL_CLEARING]


I have tried to add these entries to the dial plan, but they do not work:

<action application="stop_dtmf"/>
<action application="set" data="drop_dtmf=true"/>

Any suggestions?

/Steve

-- 
This e-mail (including any attachments) is for the sole use of the intended 
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disclosure or distribution of this e-mail is strictly prohibited. If you 
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From mike at jerris.com  Thu May 14 02:17:35 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 13 May 2015 18:17:35 -0400
Subject: [Freeswitch-users] How to disable DTMF digit processing when a
	call is valet parked?
In-Reply-To: <CALi6oUzeu-usM=qHQipvroD5-fMV5ZppWyfqrt=h7E6pcofMFw@mail.gmail.com>
References: <CALi6oUzeu-usM=qHQipvroD5-fMV5ZppWyfqrt=h7E6pcofMFw@mail.gmail.com>
Message-ID: <36457F17-93C1-4382-AC51-E9470348F2CD@jerris.com>

This likely requires a modification to mod_valet_parking to ignore the dtmf.

> On May 13, 2015, at 5:26 PM, Szeto, Steven <steven.szeto at mitel.com> wrote:
> 
> Suppose I have a public dial plan entry that looks like this:
> 
>     <extension name="1201">
>       <condition field="destination_number" expression="1201">
>         <action application="export" data="x-MitelIsACD = true" />
>         <action application="playback" data="{loops=1}tone_stream://${us-ring}" />
> 	    <action application="valet_park" data="QueuedCalls auto in 900000 902000" />
>       </condition>
>     </extension>
> 
> I have an IVR application that listens to DTMF keystrokes from the valet parked call.
> 
> I want to disable DTMF processing by freeswitch for calls that are valet parked.
> 
> The main reason for disabling DTMF is that the caller can press "#" (end of dial character), and freeswitch will terminate the call for the following reason:
> 
> 2015-05-13 17:13:28.729128 [NOTICE] switch_core_state_machine.c:315 sofia/internal/3006 at 10.47.32.159 <mailto:3006 at 10.47.32.159> has executed the last dialplan instruction, hanging up.
> 2015-05-13 17:13:28.729128 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/internal/3006 at 10.47.32.159 <mailto:3006 at 10.47.32.159> [CS_EXECUTE] [NORMAL_CLEARING]
> 
> 
> I have tried to add these entries to the dial plan, but they do not work:
> 
> 		<action application="stop_dtmf"/>
> 		<action application="set" data="drop_dtmf=true"/>
> 
> Any suggestions?
> 
> /Steve
> 

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From schoch+freeswitch.org at xwin32.com  Thu May 14 02:54:06 2015
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Wed, 13 May 2015 15:54:06 -0700
Subject: [Freeswitch-users] Latest (compiled) Freeswitch Version
Message-ID: <CAOkuKddBc91tjVpj=4MLZhjJT79TPg99EvP-9hs6j5U6T6cVDw@mail.gmail.com>

I noticed my CentOS system hadn't had a new version of Freeswitch in a
while. So I dug into the yum repository, which led me to
http://files.freeswitch.org/yum/6/x86_64/. The latest version there is
1.4.15 (December 2014). Should I expect a newer version?

-- 
Steve
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From 0x6e6562 at gmail.com  Thu May 14 13:07:29 2015
From: 0x6e6562 at gmail.com (Ben Hood)
Date: Thu, 14 May 2015 10:07:29 +0100
Subject: [Freeswitch-users] IVR dialplan inheritance
Message-ID: <CAB-G1jhbJW=M26FntW2XEY5q1XO3c5JLgjpn3=+Yk5CLaxSV=g@mail.gmail.com>

Hi all,

I was wondering under what circumstances an IVR menu would loop back
on itself when a transfer is triggered by user input.

I have the following menu:

<document type="freeswitch/xml">
<section name="configuration">
<configuration name="ivr.conf">
<menus>
<include>
<menu name="1"
greet-long="http_cache://http://$${user}:$${password}@$${url}/audio/76568832.wav"
invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
exit-sound="voicemail/vm-goodbye.wav"
confirm-macro=""
confirm-key=""
confirm-attempts="3"
timeout="10000"
inter-digit-timeout="2000"
max-failures="3"
max-timeouts="3"
digit-len="1">
 <entry action="menu-exec-app" digits="3" param="transfer +37950280022
XML public"/>
</menu>
</include>
</menus>
</configuration>
</section>
</document>

When the user dials 3, the following output in the log occurs:

2015-05-14 10:43:00.861297 [NOTICE] switch_ivr.c:1861 Transfer
sofia/external/+37950290088 at fs to XML[+37950280022 at public]
...
2015-05-14 10:43:00.861297 [INFO] mod_dialplan_xml.c:635 Processing
+37950290088 <+37950290088>->+37950280022 in context public

So at this point in time, I would expect the dialplan that matches
+37950280022 to get executed.

However, what appears to happen is that the dialplan that originally
invoked the ivr application is executed again:

Dialplan: sofia/external/+37950290088 at fs parsing [public->ivr_route]
continue=false
Dialplan: sofia/external/+37950290088 at fs Absolute Condition [ivr_route]
Dialplan: sofia/external/+37950290088 at fs Action answer()
Dialplan: sofia/external/+37950290088 at fs Action sleep(2000)
Dialplan: sofia/external/+37950290088 at fs Action ivr(1)

This creates a loop.

This is the original dialplan that invokes the ivr menu:

<document type="freeswitch/xml">
<section name="dialplan">
<context name="public">
<extension name="ivr_route">
<condition>
<action application="answer"/>
<action application="sleep" data="2000"/>
<action application="ivr" data="1"/>
</condition>
</extension>
</context>
</section>
</document>

Is there potentially something in my menu definition that is causing
the original dialplan to executed, as opposed to the dialplan
specified by the transfer application?

Many thanks in advance,

Ben


From dm at dwide.com  Thu May 14 14:06:06 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Thu, 14 May 2015 14:06:06 +0400
Subject: [Freeswitch-users] SIP Call transfer- how to
In-Reply-To: <5554729B.3090906@dwide.com>
References: <5554729B.3090906@dwide.com>
Message-ID: <5554738E.7070309@dwide.com>

Friends,

Could someone explain me how this feature works?

Using FS default configuration and free sip softphones.

Call transfer case
     - 1001 user call to 1002 and talks
     - then 1002 user transfer 1001 user to 1003 user
     - 1002 hang up
     - 1001 and 1003 continue talks


Is it possible?

How can do it?

Which technology (protocol, messages) using for it?

DTMF, FS application, SIP messages or else

Thank you!

Dmitr


From mike at jerris.com  Thu May 14 18:19:18 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 May 2015 10:19:18 -0400
Subject: [Freeswitch-users] SIP Call transfer- how to
In-Reply-To: <5554738E.7070309@dwide.com>
References: <5554729B.3090906@dwide.com> <5554738E.7070309@dwide.com>
Message-ID: <63F1F8F4-1BE6-4319-988D-2890A36DA16C@jerris.com>

Normal sip methods for transfer should "just work"

> On May 14, 2015, at 6:06 AM, Dmitry Mordovin <dm at dwide.com> wrote:
> 
> Friends,
> 
> Could someone explain me how this feature works?
> 
> Using FS default configuration and free sip softphones.
> 
> Call transfer case
>     - 1001 user call to 1002 and talks
>     - then 1002 user transfer 1001 user to 1003 user
>     - 1002 hang up
>     - 1001 and 1003 continue talks
> 
> 
> Is it possible?
> 
> How can do it?
> 
> Which technology (protocol, messages) using for it?
> 
> DTMF, FS application, SIP messages or else
> 
> Thank you!
> 
> Dmitr



From steveayre at gmail.com  Thu May 14 20:10:23 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 May 2015 17:10:23 +0100
Subject: [Freeswitch-users] IVR dialplan inheritance
In-Reply-To: <CAB-G1jhbJW=M26FntW2XEY5q1XO3c5JLgjpn3=+Yk5CLaxSV=g@mail.gmail.com>
References: <CAB-G1jhbJW=M26FntW2XEY5q1XO3c5JLgjpn3=+Yk5CLaxSV=g@mail.gmail.com>
Message-ID: <CAFiqYumounn7QXFSgJRYJWXt8TPUFSZrtHN8BW_DZeu+vT9AzA@mail.gmail.com>

Dialplan: sofia/external/+37950290088 at fs parsing [public->ivr_route]
continue=false
Dialplan: sofia/external/+37950290088 at fs Absolute Condition [ivr_route]
Dialplan: sofia/external/+37950290088 at fs Action answer()
Dialplan: sofia/external/+37950290088 at fs Action sleep(2000)
Dialplan: sofia/external/+37950290088 at fs Action ivr(1)

You have an absolute condition, so this will match every call that reaches
that part of the dialplan context. That includes after the transfer.

You need to either 1) change your condition so it only matches the calls
you want to go to the IVR menu, 2) transfer to a different context from the
menu or 3) handle the +37950290088 call before the ivr_route extension so
it doesn't reach that one.

On 14 May 2015 at 10:07, Ben Hood <0x6e6562 at gmail.com> wrote:

> Hi all,
>
> I was wondering under what circumstances an IVR menu would loop back
> on itself when a transfer is triggered by user input.
>
> I have the following menu:
>
> <document type="freeswitch/xml">
> <section name="configuration">
> <configuration name="ivr.conf">
> <menus>
> <include>
> <menu name="1"
> greet-long="http_cache://http://$${user}:$${password}@
> $${url}/audio/76568832.wav"
> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
> exit-sound="voicemail/vm-goodbye.wav"
> confirm-macro=""
> confirm-key=""
> confirm-attempts="3"
> timeout="10000"
> inter-digit-timeout="2000"
> max-failures="3"
> max-timeouts="3"
> digit-len="1">
>  <entry action="menu-exec-app" digits="3" param="transfer +37950280022
> XML public"/>
> </menu>
> </include>
> </menus>
> </configuration>
> </section>
> </document>
>
> When the user dials 3, the following output in the log occurs:
>
> 2015-05-14 10:43:00.861297 [NOTICE] switch_ivr.c:1861 Transfer
> sofia/external/+37950290088 at fs to XML[+37950280022 at public]
> ...
> 2015-05-14 10:43:00.861297 [INFO] mod_dialplan_xml.c:635 Processing
> +37950290088 <+37950290088>->+37950280022 in context public
>
> So at this point in time, I would expect the dialplan that matches
> +37950280022 to get executed.
>
> However, what appears to happen is that the dialplan that originally
> invoked the ivr application is executed again:
>
> Dialplan: sofia/external/+37950290088 at fs parsing [public->ivr_route]
> continue=false
> Dialplan: sofia/external/+37950290088 at fs Absolute Condition [ivr_route]
> Dialplan: sofia/external/+37950290088 at fs Action answer()
> Dialplan: sofia/external/+37950290088 at fs Action sleep(2000)
> Dialplan: sofia/external/+37950290088 at fs Action ivr(1)
>
> This creates a loop.
>
> This is the original dialplan that invokes the ivr menu:
>
> <document type="freeswitch/xml">
> <section name="dialplan">
> <context name="public">
> <extension name="ivr_route">
> <condition>
> <action application="answer"/>
> <action application="sleep" data="2000"/>
> <action application="ivr" data="1"/>
> </condition>
> </extension>
> </context>
> </section>
> </document>
>
> Is there potentially something in my menu definition that is causing
> the original dialplan to executed, as opposed to the dialplan
> specified by the transfer application?
>
> Many thanks in advance,
>
> Ben
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Thu May 14 20:12:04 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 May 2015 17:12:04 +0100
Subject: [Freeswitch-users] IVR dialplan inheritance
In-Reply-To: <CAFiqYumounn7QXFSgJRYJWXt8TPUFSZrtHN8BW_DZeu+vT9AzA@mail.gmail.com>
References: <CAB-G1jhbJW=M26FntW2XEY5q1XO3c5JLgjpn3=+Yk5CLaxSV=g@mail.gmail.com>
	<CAFiqYumounn7QXFSgJRYJWXt8TPUFSZrtHN8BW_DZeu+vT9AzA@mail.gmail.com>
Message-ID: <CAFiqYumzoLko7KS_mfeJDYk6cqpoVkEudSZh20Sqjd+1UNqx0w@mail.gmail.com>

>
> 3) handle the +37950290088 call before the ivr_route extension so it
> doesn't reach that one.


That should have read 37950280022 (destination number) rather than 37950290088
(caller id)

On 14 May 2015 at 17:10, Steven Ayre <steveayre at gmail.com> wrote:

> Dialplan: sofia/external/+37950290088 at fs parsing [public->ivr_route]
> continue=false
> Dialplan: sofia/external/+37950290088 at fs Absolute Condition [ivr_route]
> Dialplan: sofia/external/+37950290088 at fs Action answer()
> Dialplan: sofia/external/+37950290088 at fs Action sleep(2000)
> Dialplan: sofia/external/+37950290088 at fs Action ivr(1)
>
> You have an absolute condition, so this will match every call that reaches
> that part of the dialplan context. That includes after the transfer.
>
> You need to either 1) change your condition so it only matches the calls
> you want to go to the IVR menu, 2) transfer to a different context from the
> menu or 3) handle the +37950290088 call before the ivr_route extension so
> it doesn't reach that one.
>
> On 14 May 2015 at 10:07, Ben Hood <0x6e6562 at gmail.com> wrote:
>
>> Hi all,
>>
>> I was wondering under what circumstances an IVR menu would loop back
>> on itself when a transfer is triggered by user input.
>>
>> I have the following menu:
>>
>> <document type="freeswitch/xml">
>> <section name="configuration">
>> <configuration name="ivr.conf">
>> <menus>
>> <include>
>> <menu name="1"
>> greet-long="http_cache://http://$${user}:$${password}@
>> $${url}/audio/76568832.wav"
>> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
>> exit-sound="voicemail/vm-goodbye.wav"
>> confirm-macro=""
>> confirm-key=""
>> confirm-attempts="3"
>> timeout="10000"
>> inter-digit-timeout="2000"
>> max-failures="3"
>> max-timeouts="3"
>> digit-len="1">
>>  <entry action="menu-exec-app" digits="3" param="transfer +37950280022
>> XML public"/>
>> </menu>
>> </include>
>> </menus>
>> </configuration>
>> </section>
>> </document>
>>
>> When the user dials 3, the following output in the log occurs:
>>
>> 2015-05-14 10:43:00.861297 [NOTICE] switch_ivr.c:1861 Transfer
>> sofia/external/+37950290088 at fs to XML[+37950280022 at public]
>> ...
>> 2015-05-14 10:43:00.861297 [INFO] mod_dialplan_xml.c:635 Processing
>> +37950290088 <+37950290088>->+37950280022 in context public
>>
>> So at this point in time, I would expect the dialplan that matches
>> +37950280022 to get executed.
>>
>> However, what appears to happen is that the dialplan that originally
>> invoked the ivr application is executed again:
>>
>> Dialplan: sofia/external/+37950290088 at fs parsing [public->ivr_route]
>> continue=false
>> Dialplan: sofia/external/+37950290088 at fs Absolute Condition [ivr_route]
>> Dialplan: sofia/external/+37950290088 at fs Action answer()
>> Dialplan: sofia/external/+37950290088 at fs Action sleep(2000)
>> Dialplan: sofia/external/+37950290088 at fs Action ivr(1)
>>
>> This creates a loop.
>>
>> This is the original dialplan that invokes the ivr menu:
>>
>> <document type="freeswitch/xml">
>> <section name="dialplan">
>> <context name="public">
>> <extension name="ivr_route">
>> <condition>
>> <action application="answer"/>
>> <action application="sleep" data="2000"/>
>> <action application="ivr" data="1"/>
>> </condition>
>> </extension>
>> </context>
>> </section>
>> </document>
>>
>> Is there potentially something in my menu definition that is causing
>> the original dialplan to executed, as opposed to the dialplan
>> specified by the transfer application?
>>
>> Many thanks in advance,
>>
>> Ben
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From lesley.pervis at gmail.com  Thu May 14 20:43:49 2015
From: lesley.pervis at gmail.com (Lesley Pervis)
Date: Thu, 14 May 2015 10:43:49 -0600
Subject: [Freeswitch-users] Barracuda Phone System
Message-ID: <CANybnk6hMoAgeP=wYciOb585aEEBmSZvV3fXYdppUcgAk8Hzgg@mail.gmail.com>

When I search around, I can see that the core FS developers were actively
engaged on this product in 2011. Are you still involved, or have you washed
your hands of it?
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From giggsey at gmail.com  Thu May 14 21:09:13 2015
From: giggsey at gmail.com (Joshua Gigg)
Date: Thu, 14 May 2015 18:09:13 +0100
Subject: [Freeswitch-users] Re - British Telecom (BT) SIP test list
 before allowing a server on their IP Telephony network
In-Reply-To: <CAMxcU5JADH+0L_be8DKh3hKr+653Fcry+JAHzeacvjz1qGY9xQ@mail.gmail.com>
References: <HKNPR04MB04497AE73E615A432B0FB269F3380@HKNPR04MB0449.apcprd04.prod.outlook.com>
	<CAMxcU5JADH+0L_be8DKh3hKr+653Fcry+JAHzeacvjz1qGY9xQ@mail.gmail.com>
Message-ID: <CADcg8iUKq5pY4GaeG8CyRy=2+432oBQGY-ngwzyTZCFfPAQM_w@mail.gmail.com>

Has anyone had issues with BT IPX and Presentation Number?

For calls with a Presentation Number, it seems the Presentation Number
comes in the From header, and the Network Number comes in the
P-Asserted-Identity header. (See FS-7554).
On 21 Feb 2015 20:54, "Vladimir Getmanshchuk" <vladget at gmail.com> wrote:

> Hello!
>
> I've successfully done interconnection with BT using Freeswitch, so it is
> possible.
>
> Here are some not from another guy who did it too:
> http://blog.aeriandi.com/2012/10/08/bt
> -interoperability-testing-a-guide-to-jumping-the-hoops/
>
> On Fri, Feb 6, 2015 at 11:53 AM, Andrew Keil <andrew.keil at visytel.com>
> wrote:
>
>>  To FreeSWITCH users,
>>
>>
>>
>> Re - British Telecom (BT) SIP test list before allowing a server on their
>> IP Telephony network.
>>
>>
>>
>> I wondered if someone can take a quick look at this test case spreadsheet
>> from BT in the UK to see if a standard FreeSWITCH installation (current
>> production release) will cover off all these test cases.  I assume the
>> answer is yes!  However I thought that I would ask in advance.
>>
>>
>>
>> Kind Regards,
>>
>>
>>
>> Andrew Keil
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Yours sincerely,
> Vladimir Getmanshchuk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Thu May 14 21:23:50 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 May 2015 12:23:50 -0500
Subject: [Freeswitch-users] Barracuda Phone System
In-Reply-To: <CANybnk6hMoAgeP=wYciOb585aEEBmSZvV3fXYdppUcgAk8Hzgg@mail.gmail.com>
References: <CANybnk6hMoAgeP=wYciOb585aEEBmSZvV3fXYdppUcgAk8Hzgg@mail.gmail.com>
Message-ID: <CAKbxfG_WVKoeOVMMEVHcti+rmrSy5_cx8Re3XrTmEX3mrxfq4w@mail.gmail.com>

Hi,

Correct, The FreeSWITCH project is longer involved with Barracuda Networks.

We designed and implemented the phone system product using FreeSWITCH as
the core telephony engine in 2008 as part of a mutual agreement.
I, along with my founding FreeSWITCH partners, managed the product
development and the engineering team for 6 years.
My team and I chose to discontinue that relationship in 2014 to move on to
new challenges and to focus more on the FreeSWITCH project.
We no longer have any insight or knowledge of how they choose to develop
and maintain their product.



On Thu, May 14, 2015 at 11:43 AM, Lesley Pervis <lesley.pervis at gmail.com>
wrote:

> When I search around, I can see that the core FS developers were actively
> engaged on this product in 2011. Are you still involved, or have you washed
> your hands of 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
>



-- 
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 manavid at gmail.com  Thu May 14 21:25:03 2015
From: manavid at gmail.com (Moe Navid)
Date: Thu, 14 May 2015 10:25:03 -0700
Subject: [Freeswitch-users] Invoke SOAP service on mod_callcenter ring
In-Reply-To: <CAA_Nwu4wjAEgpaWBTVvQKmjFW46XA3sCqwUmROkbF1fUJM6r+Q@mail.gmail.com>
References: <CAA_Nwu4wjAEgpaWBTVvQKmjFW46XA3sCqwUmROkbF1fUJM6r+Q@mail.gmail.com>
Message-ID: <CACOMF+1s3QKtxNL5ROtW1vK7H3KxsmDrCL0yQB9Rma75Jj44Pg@mail.gmail.com>

Can you elaborate more on what you are trying to achieve?

On Wednesday, May 13, 2015, Ashwin Rath <ashwinrath at gmail.com> wrote:

> I was wondering is there some way to call a SOAP webservice using curl
> with info such as the agent number / name to which mod_callcenter connects
> a call ?
>
>
>
> --
> Ashwin Kumar Rath
>
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From bpriddy at bryantschools.org  Thu May 14 21:40:14 2015
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Thu, 14 May 2015 12:40:14 -0500
Subject: [Freeswitch-users] Barracuda Phone System
In-Reply-To: <CANybnk6hMoAgeP=wYciOb585aEEBmSZvV3fXYdppUcgAk8Hzgg@mail.gmail.com>
References: <CANybnk6hMoAgeP=wYciOb585aEEBmSZvV3fXYdppUcgAk8Hzgg@mail.gmail.com>
Message-ID: <CADK-Bjvt08+tbDfW_i5-HDx17OEi791iHsUnEjF0db1f3Grx8w@mail.gmail.com>

I believe the developers have gone full on FreeSWITCH Solutions. That would
be a question for them though ;)
On May 14, 2015 12:25 PM, "Lesley Pervis" <lesley.pervis at gmail.com> wrote:

> When I search around, I can see that the core FS developers were actively
> engaged on this product in 2011. Are you still involved, or have you washed
> your hands of 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
>
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From 0x6e6562 at gmail.com  Thu May 14 21:40:48 2015
From: 0x6e6562 at gmail.com (Ben Hood)
Date: Thu, 14 May 2015 18:40:48 +0100
Subject: [Freeswitch-users] IVR dialplan inheritance
In-Reply-To: <CAFiqYumzoLko7KS_mfeJDYk6cqpoVkEudSZh20Sqjd+1UNqx0w@mail.gmail.com>
References: <CAB-G1jhbJW=M26FntW2XEY5q1XO3c5JLgjpn3=+Yk5CLaxSV=g@mail.gmail.com>
	<CAFiqYumounn7QXFSgJRYJWXt8TPUFSZrtHN8BW_DZeu+vT9AzA@mail.gmail.com>
	<CAFiqYumzoLko7KS_mfeJDYk6cqpoVkEudSZh20Sqjd+1UNqx0w@mail.gmail.com>
Message-ID: <CAB-G1jjgBgBNeNnm18zdXMyHByw-2fFUT3FG70mFU2AbwiHuoA@mail.gmail.com>

Hi Steven,

On Thu, May 14, 2015 at 5:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> 3) handle the +37950290088 call before the ivr_route extension so it
>> doesn't reach that one.
>
>
> That should have read 37950280022 (destination number) rather than
> 37950290088 (caller id)

Many thanks for taking the time to look into this issue - much appreciated.

I've tried out your suggestions, but they didn't resolve the issue -
but it turns out that there was a shortcoming in my knowledge of the
way mod_xml_curl works and now that I've fixed that, my problem has
been solved.

What I discovered is the server processing my mod_xml_curl dialplan
requests was matching the destination using the variable_sip_req_user
parameter. This is fine for the initial inbound SIP transaction - the
server returns the appropriate dialplan. For example, in this
instance, when variable_sip_req_user is set to +37950290088, the
server returns:

<document type="freeswitch/xml">
<section name="dialplan">
<context name="public">
<extension name="ivr_route">
<condition>
<action application="answer"/>
<action application="sleep" data="2000"/>
<action application="ivr" data="1"/>
</condition>
</extension>
</context>
</section>
</document>

Once consumed by Freeswitch, this correctly triggers the subsequent
loading an IVR menu definition with the option:

<entry action="menu-exec-app" digits="3" param="transfer +37950280022
XML public"/>

So far so good.

So this is where my own misunderstanding started.

I had assumed that the next mod_xml_curl request would populate the
variable_sip_req_user parameter with +37950280022, i.e. the transfer
destination.

It turns out that it does not, rather variable_sip_req_user remains
set to +37950290088 (which is the original destination) and instead
the variable Hunt-Destination-Number is set to the transfer target,
i.e. +37950280022.

So the fix for this issue is for the mod_xml_curl server to match
dialplan requests based on Hunt-Destination-Number rather than
variable_sip_req_user. I've tried this out and it seems to work.

Taking a step back to think about the mechanics a bit more, this
delineation between variable_sip_req_user and Hunt-Destination-Number
kind of makes sense: variable_sip_req_user can be regarded as the SIP
header that was received on the original INVITE, whereas the
Hunt-Destination-Number is the result of routing logic applied to the
call after it has entered the dialplan context. In order words,
variable_sip_req_user is what was received from the wire and
Hunt-Destination-Number is a product of applying business rules to the
call event.

That said, this is only a lay person's explanation of what is going on
- feel free to correct me if I'm seeing this incorrectly.

Thanks for your help.

Cheers,

Ben


From emplant2000 at gmail.com  Thu May 14 21:41:08 2015
From: emplant2000 at gmail.com (Masakazu Nakano)
Date: Fri, 15 May 2015 02:41:08 +0900
Subject: [Freeswitch-users] not make a phone call both extensions
Message-ID: <CAPwL2MDwW_LEWziqsQE5nX2+bw74+kmCWCKmT2S2Ve7A3R4wpg@mail.gmail.com>

Hi there

I'm Freeswitch newbie. 15 years ago, usually used asterisk as old
extension.conf style.

so I got a cloud server like a EC2.

I made a couple of users 5630 and 5631.xml and so on, specified default in
context and in /conf/directory/default.

and registered in np.

freeswitch at internal> show registrations
reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata
5631,my
cloud,76122ZWRhNDRmMDY0MzA2YWU5NzJkNWIxZWJiNmZmNTJmMjI,sofia/external/
sip:5631 at 27.140.246.3:58996,1431626535,27.140.246.3,58996,udp,my cloud,
5630,my cloud,958199470722 at 10.210.59.150
,sofia/external/sip:5630 at 10.210.59.150:47420;transport=udp,1431626875,49.97.4.168,47420,udp,my
cloud,

2 total.

but fs_cli says 2015-05-15 02:31:47.752779 [INFO] mod_dialplan_xml.c:635
Processing 5631 <5631>->5630 in context public

why ??

BR

mack
https://www.facebook.com/nakano.masakazu
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From lesley.pervis at gmail.com  Thu May 14 23:50:48 2015
From: lesley.pervis at gmail.com (Lesley Pervis)
Date: Thu, 14 May 2015 13:50:48 -0600
Subject: [Freeswitch-users] Barracuda Phone System
In-Reply-To: <CAKbxfG_WVKoeOVMMEVHcti+rmrSy5_cx8Re3XrTmEX3mrxfq4w@mail.gmail.com>
References: <CANybnk6hMoAgeP=wYciOb585aEEBmSZvV3fXYdppUcgAk8Hzgg@mail.gmail.com>
	<CAKbxfG_WVKoeOVMMEVHcti+rmrSy5_cx8Re3XrTmEX3mrxfq4w@mail.gmail.com>
Message-ID: <CANybnk6YxzYYkZC4CgL-B8DO1UOn4fU+WLTpTV9afa-zPrY1TA@mail.gmail.com>

Good to know, thanks.

On Thu, May 14, 2015 at 11:23 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Hi,
>
> Correct, The FreeSWITCH project is longer involved with Barracuda Networks.
>
> We designed and implemented the phone system product using FreeSWITCH as
> the core telephony engine in 2008 as part of a mutual agreement.
> I, along with my founding FreeSWITCH partners, managed the product
> development and the engineering team for 6 years.
> My team and I chose to discontinue that relationship in 2014 to move on to
> new challenges and to focus more on the FreeSWITCH project.
> We no longer have any insight or knowledge of how they choose to develop
> and maintain their product.
>
>
>
> On Thu, May 14, 2015 at 11:43 AM, Lesley Pervis <lesley.pervis at gmail.com>
> wrote:
>
>> When I search around, I can see that the core FS developers were actively
>> engaged on this product in 2011. Are you still involved, or have you washed
>> your hands of 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
>>
>
>
>
> --
> 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
> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From william.king at quentustech.com  Fri May 15 00:16:18 2015
From: william.king at quentustech.com (William King)
Date: Thu, 14 May 2015 13:16:18 -0700
Subject: [Freeswitch-users] mod_hiredis
In-Reply-To: <CAJs+-5pN1KHKrPTKUnET2rbkj0Efe0-y+2VnRyzyfZiMeq7pBg@mail.gmail.com>
References: <554EB704.5090605@quentustech.com>
	<5551E552.9090307@telefaks.de>	<55524336.4050208@quentustech.com>
	<CAJs+-5pN1KHKrPTKUnET2rbkj0Efe0-y+2VnRyzyfZiMeq7pBg@mail.gmail.com>
Message-ID: <55550292.2090604@quentustech.com>

Very interesting use case. This should be currently supported with the
raw get and set actions.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 05/12/2015 06:13 PM, Seven Du wrote:
> maybe use as a central storage for http sessions and ws sessions. or
> even SSO if run FreeSWITCH as a cluster if we extend the http support in
> FreeSWITCH.
> 
> 
> 
> On Wed, May 13, 2015 at 2:15 AM, William King
> <william.king at quentustech.com <mailto:william.king at quentustech.com>> wrote:
> 
>     Peter,
> 
>     The pubsub actions should be available now with the raw app, though this
>     would work in a blocking manner. You do raise a good use case for
>     asynchronous commands, specifically for fire and forget cases like
>     pubsub.
> 
>     For the prefix routing, currently the array return type isn't supported,
>     but if redis returns a single result that should be able to work and be
>     tested right now. Can you provide a few sample redis commands and
>     responses for how you'd setup this scenario?
> 
>     William King
>     Senior Engineer
>     Quentus Technologies, INC
>     1037 NE 65th St Suite 273
>     Seattle, WA 98115
>     Main:   (877) 211-9337 <tel:%28877%29%20211-9337>
>     Office: (206) 388-4772 <tel:%28206%29%20388-4772>
>     Cell:   (253) 686-5518 <tel:%28253%29%20686-5518>
>     william.king at quentustech.com <mailto:william.king at quentustech.com>
> 
>     On 05/12/2015 04:34 AM, Peter Steinbach wrote:
>     > Hello William,
>     >
>     > this is great, the idea of integrating Redis. We currently use
>     Memcache
>     > in raw mode as a method of externally controlling dialplans and
>     failover
>     > scenarios.
>     > Redis, of course, brings much more features here.
>     >
>     >>Currently the two main use cases are:
>     >>1. Call per second limits
>     >>2. Concurrent call limits
>     >>
>     >>Possible additional functionality:
>     >>1. Support for fail-over connections
>     >>2. Asynchronous commands(is there a use case for this?)
>     >
>     > Another idea for your list would be to route calls according to
>     > prefixes. You may lookup Redis with a part of the phone number and it
>     > returns the gateway for this part of the number (redis DB is then
>     > preloaded from another application).
>     > And - as Redis has a publish/subscribe method - you will be able to
>     > publish call informations from the dialplan to multiple external
>     > subscribers (e.g. announce an incoming call to a CRM) without the
>     use of
>     > ESL. Is there a chance to run the redis dialplan app in a non blocking
>     > manner for this scenario, in order to speed up the dialplan?
>     >
>     >
>     >
>     > Best regards
>     > Peter
>     >
>     > On 05/10/15 03:40, William King wrote:
>     >> I'm working on an update Redis integration module that will use the C
>     >> library hiredis:
>     >> http://redis.io/clients#c
>     >> https://github.com/redis/hiredis
>     >>
>     >> I've pushed an alpha version of the module to a branch here:
>     >>
>     https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits?until=
>     >> refs%2Fheads%2Fmod_hiredis
>     >>
>     >> The current module has a dialplan app and an api for 'hiredis_raw'
>     >> which allows any single line Redis command, and executes it in a
>     >> blocking manner, then supports returning string and integer
>     responses.
>     >>
>     >> If anyone on this list has any use cases for FreeSWITCH+Redis, please
>     >> reply to this thread. Currently the two main use cases are:
>     >> 1. Call per second limits
>     >> 2. Concurrent call limits
>     >>
>     >> Possible additional functionality:
>     >> 1. Support for fail-over connections
>     >> 2. Asynchronous commands(is there a use case for this?)
>     >>
>     >>
>     >>
>     _________________________________________________________________________
>     >> 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
>     >
>     > --
>     > With kind regards
>     > Peter Steinbach
>     >
>     > Telefaks Services GmbH
>     > mailto:lists <mailto:lists> (att) telefaks.de <http://telefaks.de>
>     > Internet: www.telefaks.de <http://www.telefaks.de>
>     >
>     >
>     >
>     >
>     >
>     _________________________________________________________________________
>     > 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
> 
> 
> 
> 
> -- 
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 blasterjr at gmail.com  Fri May 15 01:11:49 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Thu, 14 May 2015 15:11:49 -0600
Subject: [Freeswitch-users] Invoke SOAP service on mod_callcenter ring
In-Reply-To: <CACOMF+1s3QKtxNL5ROtW1vK7H3KxsmDrCL0yQB9Rma75Jj44Pg@mail.gmail.com>
References: <CAA_Nwu4wjAEgpaWBTVvQKmjFW46XA3sCqwUmROkbF1fUJM6r+Q@mail.gmail.com>
	<CACOMF+1s3QKtxNL5ROtW1vK7H3KxsmDrCL0yQB9Rma75Jj44Pg@mail.gmail.com>
Message-ID: <CAO3hyzGE=XKMP7XwW7HZBw1A_sv4Yi3zi8b6wCLnTM1dO9DvKw@mail.gmail.com>

You can achieve something like this with an ESL client that does the CURL
requests for you from listening for events specific to mod_callcenter.

On Thu, May 14, 2015 at 11:25 AM, Moe Navid <manavid at gmail.com> wrote:

> Can you elaborate more on what you are trying to achieve?
>
>
> On Wednesday, May 13, 2015, Ashwin Rath <ashwinrath at gmail.com> wrote:
>
>> I was wondering is there some way to call a SOAP webservice using curl
>> with info such as the agent number / name to which mod_callcenter connects
>> a call ?
>>
>>
>>
>> --
>> Ashwin Kumar Rath
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From govoiper at gmail.com  Fri May 15 01:50:36 2015
From: govoiper at gmail.com (SamyGo)
Date: Thu, 14 May 2015 17:50:36 -0400
Subject: [Freeswitch-users] mod_perl perl IVR using FreeSwitch DSN
Message-ID: <CAJUJwtjrQ7y7CuJZvqfCSU_7cJkdwcgMQFR+e=xtCU_wwnfXPg@mail.gmail.com>

Hi List,
I've my FreeSwitch configured with ODBC and I've a perl IVR example
running. I want to use the FreeSwitch's DSN to query DB. Can anyone share
an example with me !

Thanks,
Sammy'
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From bilaln018 at gmail.com  Fri May 15 08:27:22 2015
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Fri, 15 May 2015 09:27:22 +0500
Subject: [Freeswitch-users] [Bridge/Originate][Hangup call with Single Leg]
Message-ID: <CADSVerA0Ac5o2QmfC4-sf8jwC2tURF13wwFwY1LSCS65kES5gQ@mail.gmail.com>

Hi Users,
Currently i am using mod event socket to generate a outbound call,
using pyswitch

({execute_on_answer='bridge
[DTMF="+DTMF+"]loopback/9665/default/XML'}sofia/gateway/aptcl/XXXXXXXX")

Now when call on originate channel is answered it is bridged to extension
9665 in default context.
Problem is when originate channel hangups the call,call is not terminated
on bridge's channel.
I have tried hangup after bridge variable(no luck).

Please suggest me some solution.

Regards
Bilal Abbasi
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From boros at vmtele.com  Fri May 15 12:29:39 2015
From: boros at vmtele.com (=?UTF-8?B?VG9tw6HFoSBCb3Jvcw==?=)
Date: Fri, 15 May 2015 10:29:39 +0200
Subject: [Freeswitch-users] Privacy and P-asserted-id header in responses
Message-ID: <5555AE73.5090700@vmtele.com>

Hi,

We are having troubles with some cases.

Case is the following:


[Provider A] --------[FS]-------[user redirects call to number back to 
operator A]-----|
[Provider A] 
--------[FS]--------------------------------------------------------------|

So a there is an incoming call from Provider A, we use Freeswitch for 
interconnection. The call goes to our network, where client redirects 
the call back to provider A (to different number then caller).
In such a case Diversion header is added with the user's phone number.

Provider A  (on the second leg..third?!) when accepts the call sends an 
200 OK message with P-Asserted-identity and Privacy: ID header.
My question is, it is possible to pass these headers back to our network 
and again back to the Provider A?
Provider wants to see on their first leg, that the call was answered in 
their network using p-asserted-identity header.

copy_custom_headers does not solves this issue, as P-Asserted and 
Privacy headers are not custom headers.

Thank you,
-- 
Tom?? Boros
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From andrew at cassidywebservices.co.uk  Fri May 15 14:05:48 2015
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 15 May 2015 11:05:48 +0100
Subject: [Freeswitch-users] Re - British Telecom (BT) SIP test list
 before allowing a server on their IP Telephony network
In-Reply-To: <CADcg8iUKq5pY4GaeG8CyRy=2+432oBQGY-ngwzyTZCFfPAQM_w@mail.gmail.com>
References: <HKNPR04MB04497AE73E615A432B0FB269F3380@HKNPR04MB0449.apcprd04.prod.outlook.com>
	<CAMxcU5JADH+0L_be8DKh3hKr+653Fcry+JAHzeacvjz1qGY9xQ@mail.gmail.com>
	<CADcg8iUKq5pY4GaeG8CyRy=2+432oBQGY-ngwzyTZCFfPAQM_w@mail.gmail.com>
Message-ID: <CAMbW9oLywP1gEahiBVySH6nzYKQhhd-rtTF41DSV7K5TZZdDDw@mail.gmail.com>

Is there a specific need for that variable to contain the number? You can
read out the P-Asserted-Identity header using ?{sip_h_P-Asserted-Identity}
as a workaround?

On 14 May 2015 at 18:09, Joshua Gigg <giggsey at gmail.com> wrote:

> Has anyone had issues with BT IPX and Presentation Number?
>
> For calls with a Presentation Number, it seems the Presentation Number
> comes in the From header, and the Network Number comes in the
> P-Asserted-Identity header. (See FS-7554).
> On 21 Feb 2015 20:54, "Vladimir Getmanshchuk" <vladget at gmail.com> wrote:
>
>> Hello!
>>
>> I've successfully done interconnection with BT using Freeswitch, so it
>> is possible.
>>
>> Here are some not from another guy who did it too:
>> http://blog.aeriandi.com/2012/10/08/bt
>> -interoperability-testing-a-guide-to-jumping-the-hoops/
>>
>> On Fri, Feb 6, 2015 at 11:53 AM, Andrew Keil <andrew.keil at visytel.com>
>> wrote:
>>
>>>  To FreeSWITCH users,
>>>
>>>
>>>
>>> Re - British Telecom (BT) SIP test list before allowing a server on
>>> their IP Telephony network.
>>>
>>>
>>>
>>> I wondered if someone can take a quick look at this test case
>>> spreadsheet from BT in the UK to see if a standard FreeSWITCH installation
>>> (current production release) will cover off all these test cases.  I assume
>>> the answer is yes!  However I thought that I would ask in advance.
>>>
>>>
>>>
>>> Kind Regards,
>>>
>>>
>>>
>>> Andrew Keil
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Yours sincerely,
>> Vladimir Getmanshchuk
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 andrew at cassidywebservices.co.uk  Fri May 15 14:06:41 2015
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 15 May 2015 11:06:41 +0100
Subject: [Freeswitch-users] Re - British Telecom (BT) SIP test list
 before allowing a server on their IP Telephony network
In-Reply-To: <CAMbW9oLywP1gEahiBVySH6nzYKQhhd-rtTF41DSV7K5TZZdDDw@mail.gmail.com>
References: <HKNPR04MB04497AE73E615A432B0FB269F3380@HKNPR04MB0449.apcprd04.prod.outlook.com>
	<CAMxcU5JADH+0L_be8DKh3hKr+653Fcry+JAHzeacvjz1qGY9xQ@mail.gmail.com>
	<CADcg8iUKq5pY4GaeG8CyRy=2+432oBQGY-ngwzyTZCFfPAQM_w@mail.gmail.com>
	<CAMbW9oLywP1gEahiBVySH6nzYKQhhd-rtTF41DSV7K5TZZdDDw@mail.gmail.com>
Message-ID: <CAMbW9oK+qkAaJVav_ohOeErzr3wsm3Csov_THue_congMCgbxQ@mail.gmail.com>

Except it's a $ not a ? :S

On 15 May 2015 at 11:05, Andrew Cassidy <andrew at cassidywebservices.co.uk>
wrote:

> Is there a specific need for that variable to contain the number? You can
> read out the P-Asserted-Identity header using ?{sip_h_P-Asserted-Identity}
> as a workaround?
>
> On 14 May 2015 at 18:09, Joshua Gigg <giggsey at gmail.com> wrote:
>
>> Has anyone had issues with BT IPX and Presentation Number?
>>
>> For calls with a Presentation Number, it seems the Presentation Number
>> comes in the From header, and the Network Number comes in the
>> P-Asserted-Identity header. (See FS-7554).
>> On 21 Feb 2015 20:54, "Vladimir Getmanshchuk" <vladget at gmail.com> wrote:
>>
>>> Hello!
>>>
>>> I've successfully done interconnection with BT using Freeswitch, so it
>>> is possible.
>>>
>>> Here are some not from another guy who did it too:
>>> http://blog.aeriandi.com/2012/10/08/bt
>>> -interoperability-testing-a-guide-to-jumping-the-hoops/
>>>
>>> On Fri, Feb 6, 2015 at 11:53 AM, Andrew Keil <andrew.keil at visytel.com>
>>> wrote:
>>>
>>>>  To FreeSWITCH users,
>>>>
>>>>
>>>>
>>>> Re - British Telecom (BT) SIP test list before allowing a server on
>>>> their IP Telephony network.
>>>>
>>>>
>>>>
>>>> I wondered if someone can take a quick look at this test case
>>>> spreadsheet from BT in the UK to see if a standard FreeSWITCH installation
>>>> (current production release) will cover off all these test cases.  I assume
>>>> the answer is yes!  However I thought that I would ask in advance.
>>>>
>>>>
>>>>
>>>> Kind Regards,
>>>>
>>>>
>>>>
>>>> Andrew Keil
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Yours sincerely,
>>> Vladimir Getmanshchuk
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>



-- 
*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 giggsey at gmail.com  Fri May 15 14:46:08 2015
From: giggsey at gmail.com (Joshua Gigg)
Date: Fri, 15 May 2015 11:46:08 +0100
Subject: [Freeswitch-users] Re - British Telecom (BT) SIP test list
 before allowing a server on their IP Telephony network
In-Reply-To: <CAMbW9oK+qkAaJVav_ohOeErzr3wsm3Csov_THue_congMCgbxQ@mail.gmail.com>
References: <HKNPR04MB04497AE73E615A432B0FB269F3380@HKNPR04MB0449.apcprd04.prod.outlook.com>
	<CAMxcU5JADH+0L_be8DKh3hKr+653Fcry+JAHzeacvjz1qGY9xQ@mail.gmail.com>
	<CADcg8iUKq5pY4GaeG8CyRy=2+432oBQGY-ngwzyTZCFfPAQM_w@mail.gmail.com>
	<CAMbW9oLywP1gEahiBVySH6nzYKQhhd-rtTF41DSV7K5TZZdDDw@mail.gmail.com>
	<CAMbW9oK+qkAaJVav_ohOeErzr3wsm3Csov_THue_congMCgbxQ@mail.gmail.com>
Message-ID: <CADcg8iU5ae7V7jZv7i18xfj2u7847asO5HZDUFH-scd-kKJQ4Q@mail.gmail.com>

FreeSWITCH detects the PAID, and sets the Caller-ID-Number as the PAID,
instead of From. This means it sends the incorrect number onwards.

I've found documentation from NICC Standards supporting what BT are doing,
which I've added to the issue.

On 15 May 2015 at 11:06, Andrew Cassidy <andrew at cassidywebservices.co.uk>
wrote:

> Except it's a $ not a ? :S
>
> On 15 May 2015 at 11:05, Andrew Cassidy <andrew at cassidywebservices.co.uk>
> wrote:
>
>> Is there a specific need for that variable to contain the number? You can
>> read out the P-Asserted-Identity header using ?{sip_h_P-Asserted-Identity}
>> as a workaround?
>>
>> On 14 May 2015 at 18:09, Joshua Gigg <giggsey at gmail.com> wrote:
>>
>>> Has anyone had issues with BT IPX and Presentation Number?
>>>
>>> For calls with a Presentation Number, it seems the Presentation Number
>>> comes in the From header, and the Network Number comes in the
>>> P-Asserted-Identity header. (See FS-7554).
>>> On 21 Feb 2015 20:54, "Vladimir Getmanshchuk" <vladget at gmail.com> wrote:
>>>
>>>> Hello!
>>>>
>>>> I've successfully done interconnection with BT using Freeswitch, so it
>>>> is possible.
>>>>
>>>> Here are some not from another guy who did it too:
>>>> http://blog.aeriandi.com/2012/10/08/bt
>>>> -interoperability-testing-a-guide-to-jumping-the-hoops/
>>>>
>>>> On Fri, Feb 6, 2015 at 11:53 AM, Andrew Keil <andrew.keil at visytel.com>
>>>> wrote:
>>>>
>>>>>  To FreeSWITCH users,
>>>>>
>>>>>
>>>>>
>>>>> Re - British Telecom (BT) SIP test list before allowing a server on
>>>>> their IP Telephony network.
>>>>>
>>>>>
>>>>>
>>>>> I wondered if someone can take a quick look at this test case
>>>>> spreadsheet from BT in the UK to see if a standard FreeSWITCH installation
>>>>> (current production release) will cover off all these test cases.  I assume
>>>>> the answer is yes!  However I thought that I would ask in advance.
>>>>>
>>>>>
>>>>>
>>>>> Kind Regards,
>>>>>
>>>>>
>>>>>
>>>>> Andrew Keil
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Yours sincerely,
>>>> Vladimir Getmanshchuk
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>
>
>
> --
> *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
>



-- 
Joshua Gigg
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From steveayre at gmail.com  Fri May 15 16:21:06 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 15 May 2015 13:21:06 +0100
Subject: [Freeswitch-users] Privacy and P-asserted-id header in responses
In-Reply-To: <5555AE73.5090700@vmtele.com>
References: <5555AE73.5090700@vmtele.com>
Message-ID: <CAFiqYun6mvJWNtshc4gFpJLr4KwgXP+e8AtoCwN59w7vD926Pg@mail.gmail.com>

If they're sending them to you then they're presumably part of your trust
domain and so it's ok to send them back.

${sip_cid_type} (or the sip-cid-type sofia profile param) control whether
to send them back or not.

On 15 May 2015 at 09:29, Tom?? Boros <boros at vmtele.com> wrote:

>  Hi,
>
> We are having troubles with some cases.
>
> Case is the following:
>
>
> [Provider A]  --------[FS]-------[user redirects call to number back to
> operator A]-----|
> [Provider A]
> --------[FS]--------------------------------------------------------------|
>
> So a there is an incoming call from Provider A, we use Freeswitch for
> interconnection. The call goes to our network, where client redirects the
> call back to provider A (to different number then caller).
> In such a case Diversion header is added with the user's phone number.
>
> Provider A  (on the second leg..third?!) when accepts the call sends an
> 200 OK message with P-Asserted-identity and Privacy: ID header.
> My question is, it is possible to pass these headers back to our network
> and again back to the Provider A?
> Provider wants to see on their first leg, that the call was answered in
> their network using p-asserted-identity header.
>
> copy_custom_headers does not solves this issue, as P-Asserted and Privacy
> headers are not custom headers.
>
> Thank you,
> --
>  Tom?? Boros
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
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>
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>
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From nicoleta_lazar2006 at yahoo.com  Fri May 15 15:55:15 2015
From: nicoleta_lazar2006 at yahoo.com (Nicoleta Lazar)
Date: Fri, 15 May 2015 11:55:15 +0000 (UTC)
Subject: [Freeswitch-users] Cannot collect digits from lua script
Message-ID: <1686549188.919732.1431690915783.JavaMail.yahoo@mail.yahoo.com>

Hello all,

I have a mod_erlang_event based application and I need to run a lua script within for playing prompt and colecting digits (I use freeswith api and luarun from erlang code for running lua script)
The script should collect digits, similar to:?
freeswitch.consoleLog("notice" , "Before GetDigits\n");?digits = session:getDigits(1, "#", 10000);?freeswitch.consoleLog("notice" , "AFTER GetDigits\n");?session:execute('flush_dtmf'); ?freeswitch.consoleLog("notice" , "flush DTMF 2\n");?freeswitch.consoleLog("notice" , "After Get Digits" .. "\n");?freeswitch.consoleLog("notice" , "digits = " .. digits .. "\n");


However, the digits are not collected (sometimes this scenario works OK but I cannot identify a pattern), as can be seen from the logs below:

EXECUTE sofia/test/2001 at test flush_dtmf()2015-05-15 07:09:39.694302?[DEBUG] mod_erlang_event.c:169 Ignoring event CHANNEL_EXECUTE for attached session d7e76764-faf2-11e4-8f50- 771905ed12f62015-05-15 07:09:39.694302?[DEBUG] mod_erlang_event.c:169 Ignoring event CHANNEL_EXECUTE_COMPLETE for attached session d7e76764-faf2-11e4-8f50- 771905ed12f62015-05-15 07:09:39.694302?[INFO] switch_cpp.cpp:1291 UUID: d7e76764-faf2-11e4-8f50- 771905ed12f62015-05-15 07:09:39.714301?[NOTICE] switch_cpp.cpp:1291 1234 Before GetDigits2015-05-15 07:09:42.034319?[DEBUG] switch_rtp.c:5788 RTP RECV DTMF 1:11202015-05-15 07:09:42.034319?[DEBUG] switch_channel.c:486 RECV DTMF 1:11202015-05-15 07:09:42.034319?[DEBUG] mod_dptools.c:2138 Digit 12015-05-15 07:09:42.034319?[DEBUG] mod_erlang_event.c:157 Sending event DTMF to attached session d7e76764-faf2-11e4-8f50- 771905ed12f62015-05-15 07:09:49.714345?[DEBUG] switch_cpp.cpp:838 getDigits dtmf_buf:?2015-05-15 07:09:49.714345?[NOTICE] switch_cpp.cpp:1291 1234 AFTER GetDigitsEXECUTE sofia/test/2001 at test flush_dtmf()2015-05-15 07:09:49.714345?[DEBUG] mod_erlang_event.c:169 Ignoring event CHANNEL_EXECUTE for attached session d7e76764-faf2-11e4-8f50- 771905ed12f62015-05-15 07:09:49.714345?[NOTICE] switch_cpp.cpp:1291 flush DTMF 22015-05-15 07:09:49.714345?[NOTICE] switch_cpp.cpp:1291 After Get Digits2015-05-15 07:09:49.714345?[NOTICE] switch_cpp.cpp:1291 digits =?
I also looked in the freeswitch code:
https://freeswitch.org/ fisheye/browse/freeswitch/src/ mod/event_handlers/mod_erlang_ event/mod_erlang_event.c?hb= true#to159

and it seems that the DTMF event is properly cloned and added to current session queue.
Please advise
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From yadenis at seznam.cz  Fri May 15 10:48:22 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Fri, 15 May 2015 08:48:22 +0200
Subject: [Freeswitch-users] MP4 record in 1.6
Message-ID: <1852348436.20150515084822@seznam.cz>

Hi All,

New FreeSWITCH 1.6 Video on Debian 8.

I try to record MP4 file in the conference..  
Simple dialplan

  <extension name="nb_conferences">
      <condition field="destination_number" expression="^(30\d{2})$">
        <action application="answer"/>
        <action application="set" data="enable_file_write_buffering=false"/>
        <action application="record" data="/usr/local/freeswitch/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.mp4"/>
        <action application="conference" data="$1-${domain_name}@default"/>
      </condition>
    </extension>

It works. Even something writes. But I can not play this file (The file in the attachment). What am I doing wrong?

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From asdfgh1234 at freemail.hu  Fri May 15 16:47:05 2015
From: asdfgh1234 at freemail.hu (Asd)
Date: Fri, 15 May 2015 14:47:05 +0200 (CEST)
Subject: [Freeswitch-users] FreeSWITCH PROXY:0
In-Reply-To: <CAEJMVkC2fnJ+vEdFPuBr_7DAqW1fqbMn02yVoM+E0K27Y+Qmnw@mail.gmail.com>
Message-ID: <freemail.20150515144705.49318.1@fmxmldata00.freemail.hu>

internal profile with disabled proxy-media <param name="inbound-proxy-media" value="false"/> what should i set my profile to internal freeswitch? my internal profile:                 <param name="debug" value="0"/>
                <param name="sip-trace" value="no"/>
                <param name="sip-capture" value="no"/>
                <param name="watchdog-enabled" value="no"/>
                <param name="watchdog-step-timeout" value="30000"/>
                <param name="watchdog-event-timeout" value="30000"/>
                <param name="log-auth-failures" value="true"/>
                <param name="forward-unsolicited-mwi-notify" value="false"/>
                <param name="rfc2833-pt" value="101"/>
                <param name="sip-port" value="$${internal_sip_port}"/>
                <param name="dialplan" value="XML"/>
                <param name="dtmf-duration" value="2000"/>
                <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
                <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
                <param name="rtp-ip" value="$${local_ip_v4}"/>
                <param name="sip-ip" value="$${local_ip_v4}"/>
                <param name="hold-music" value="$${hold_music}"/>
                <param name="apply-nat-acl" value="nat.auto"/>
                <param name="apply-inbound-acl" value="domains"/>
                <param name="local-network-acl" value="localnet.auto"/>
                <param name="context" value="sip.domain.com"/>
                <param name="record-path" value="$${recordings_dir}"/>
                <param name="record-template" value="${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.wav"/>
                <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>
                <param name="presence-privacy" value="$${presence_privacy}"/>
                <param name="inbound-codec-negotiation" value="generous"/>
                <param name="tls" value="$${internal_ssl_enable}"/>
                <param name="tls-bind-params" value="transport=tls"/>
                <param name="tls-sip-port" value="$${internal_tls_port}"/>
                <param name="tls-cert-dir" value="$${internal_ssl_dir}"/>
                <param name="tls-passphrase" value=""/>
                <param name="tls-verify-date" value="true"/>
                <param name="tls-verify-in-subjects" value=""/>
                <param name="tls-version" value="$${sip_tls_version}"/>
                <param name="nonce-ttl" value="60"/>
                <param name="auth-calls" value="$${internal_auth_calls}"/>
                <param name="inbound-reg-force-matching-username" value="true"/>
                <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
                <param name="ext-sip-ip" value="$${external_sip_ip}"/>
                <param name="rtp-timeout-sec" value="300"/>
                <param name="rtp-hold-timeout-sec" value="1800"/>
                <param name="force-register-domain" value="$${domain}"/>
                <param name="force-subscription-domain" value="$${domain}"/>
                <param name="force-register-db-domain" value="$${domain}"/>
                <param name="challenge-realm" value="auto_from"/>
                <param name="tls-only" value="true"/>
                <param name="tls-verify-depth" value="3"/>
                <param name="manage-presence" value="true"/>
                <param name="rtp-timer-name" value="soft"/>
                <param name="tls-ciphers" value="$${sip_tls_ciphers}"/>
                <param name="aggressive-nat-detection" value="true"/>
                <param name="inbound-late-negotiation" value="true"/>
                <param name="inbound-zrtp-passthru" value="true"/>
                <param name="tls-verify-policy" value="all"/>
                <param name="send-presence-on-register" value="false"/>
                <param name="auth-all-packets" value="false"/>
                <param name="inbound-proxy-media" value="false"/>the log file: tport.c:2773 tport_wakeup() tport_wakeup(0x7fccf00a2eb0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fccf00a2eb0)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fccf00a2eb0): tls_read() returned 1079
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fccf00a2eb0) msg 0x7fccf00a8080 from (tls/XXX.XXX.XXX.XXX:15193) has 1079 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fccf00a2eb0): msg 0x7fccf00a8080 (1099 bytes) from tls/XXX.XXX.XXX.XXX:15193/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received INVITE sips:1004 at sip.domain.com SIP/2.0 (CSeq 28198)
nta.c:3174 agent_check_request_via() nta: Via check: received=XXX.XXX.XXX.XXX
nta.c:3085 agent_recv_request() nta: INVITE (28198) going to a default leg
nta.c:1350 set_timeout() nta: timer set to 2000 ms
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
nua_stack.c:899 nh_create() nua: nh_create: entering
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fccf0001be0, 0x7fccf0000cb0, 0x7fccf0092180) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fccf0090bc0, ...) called
nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fccf00a1320)
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7fccf0090bc0) called
soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fccf0090bc0, (nil), 0x7fccf0091e32, 373) called
nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fccf0092180): adding session usage
tport.c:3257 tport_tsend() tport_tsend(0x7fccf00a2eb0) tpn = TLS/XXX.XXX.XXX.XXX:15193
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0079900 0x7fccf008ec40 384 (384)
tport.c:3492 tport_send_msg() tport_vsend returned 384
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (28198)
nua_session.c:4139 signal_call_state_change() nua(0x7fccf0092180): call state changed: init -> received, received offer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fccf0090bc0, [0x7fcd0bdfc8b8], [0x7fcd0bdfc8c0], [(nil)]) called
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2015-05-15 13:54:57.434956 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1001 at sip.domain.com [a5330703-4ba1-48e2-bb32-649b0bd8d6c8]                                                                                                                        
2015-05-15 13:54:57.434956 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-15 13:54:57.434956 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-05-15 13:54:57.434956 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_NEW
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:54:57.434956 [DEBUG] sofia.c:8844 sofia/internal/1001 at sip.domain.com receiving invite from XXX.XXX.XXX.XXX:15193 version: 1.4.18 -3-1 64bit                                                                                                                
2015-05-15 13:54:57.434956 [DEBUG] switch_core_media.c:344 Found audio zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10 cefe05149bdf61156a54615b2de164b405ca49c2eb2b7e8adb8a09ea3f0714b0                                                                                 
2015-05-15 13:54:57.434956 [DEBUG] sofia.c:9011 IP XXX.XXX.XXX.XXX Rejected by acl "domains". Falling back to Digest auth.
nua.c:879 nua_respond() nua: nua_respond: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fccf0090bc0, ...) called
nua_session.c:2320 nua_invite_server_respond() nua: nua_invite_server_respond: entering
soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0x7fccf0090bc0) called
tport.c:3257 tport_tsend() tport_tsend(0x7fccf00a2eb0) tpn = TLS/XXX.XXX.XXX.XXX:15193
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0079900 0x7fccf0093b30 888 (888)
tport.c:3492 tport_send_msg() tport_vsend returned 888
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
nta.c:6791 incoming_reply() nta: sent 407 Proxy Authentication Required for INVITE (28198)
nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7fccf0092180): removing session usage
nua_session.c:4139 signal_call_state_change() nua(0x7fccf0092180): call state changed: received -> terminated
soa.c:356 soa_destroy() soa_destroy(static::0x7fccf0090bc0) called
nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7fccf00a1320)
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-15 13:54:57.434956 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_stack.c:529 nua_signal() nua(0x7fccf0092180): sent signal r_respond
2015-05-15 13:54:57.434956 [DEBUG] sofia.c:2065 detaching session a5330703-4ba1-48e2-bb32-649b0bd8d6c8
2015-05-15 13:54:57.434956 [WARNING] sofia_reg.c:1742 SIP auth challenge (INVITE) on sofia profile &#39;internal&#39; for [1004 at sip.domain.com] from ip XXX.XXX.XXX.XXX                                                                                                          
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:529 nua_signal() nua(0x7fccf0092180): sent signal r_destroy
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
2015-05-15 13:54:57.434956 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1001 at sip.domain.com) State NEW
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
tport.c:2773 tport_wakeup() tport_wakeup(0x7fccf00a2eb0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fccf00a2eb0)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fccf00a2eb0): tls_read() returned 406
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fccf00a2eb0) msg 0x7fccf008dfe0 from (tls/XXX.XXX.XXX.XXX:15193) has 406 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fccf00a2eb0): msg 0x7fccf008dfe0 (406 bytes) from tls/XXX.XXX.XXX.XXX:15193/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received ACK sips:1004 at sip.domain.com SIP/2.0 (CSeq 28198)
nta.c:3174 agent_check_request_via() nta: Via check: received=XXX.XXX.XXX.XXX
nta.c:3019 agent_recv_request() nta: ACK (28198) is going to INVITE (28198)
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7fccf00a2eb0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fccf00a2eb0)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fccf00a2eb0): tls_read() returned 1350
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fccf00a2eb0) msg 0x7fccf008dfe0 from (tls/XXX.XXX.XXX.XXX:15193) has 1350 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fccf00a2eb0): msg 0x7fccf008dfe0 (1350 bytes) from tls/XXX.XXX.XXX.XXX:15193/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received INVITE sips:1004 at sip.domain.com SIP/2.0 (CSeq 28199)
nta.c:3174 agent_check_request_via() nta: Via check: received=XXX.XXX.XXX.XXX
nta.c:3085 agent_recv_request() nta: INVITE (28199) going to a default leg
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
nua_stack.c:899 nh_create() nua: nh_create: entering
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fccf0001be0, 0x7fccf0000cb0, 0x7fccf0091540) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fccf0091660, ...) called
nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fccf0093530)
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7fccf0091660) called
soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fccf0091660, (nil), 0x7fccf0092551, 373) called
nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fccf0091540): adding session usage
tport.c:3257 tport_tsend() tport_tsend(0x7fccf00a2eb0) tpn = TLS/XXX.XXX.XXX.XXX:15193
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0079900 0x7fccf008c970 384 (384)
tport.c:3492 tport_send_msg() tport_vsend returned 384
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
nta.c:6791 incoming_reply() nta: sent 100 Trying for INVITE (28199)
nua_session.c:4139 signal_call_state_change() nua(0x7fccf0091540): call state changed: init -> received, received offer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fccf0091660, [0x7fcd0bdfc8b8], [0x7fcd0bdfc8c0], [(nil)]) called
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2015-05-15 13:54:57.614920 [DEBUG] sofia.c:2173 Re-attaching to session a5330703-4ba1-48e2-bb32-649b0bd8d6c8
2015-05-15 13:54:57.614920 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-15 13:54:57.614920 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:54:57.634956 [DEBUG] sofia.c:8844 sofia/internal/1001 at sip.domain.com receiving invite from XXX.XXX.XXX.XXX:15193 version: 1.4.18 -3-1 64bit                                                                                                                
2015-05-15 13:54:57.634956 [DEBUG] switch_core_media.c:344 Found audio zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10 cefe05149bdf61156a54615b2de164b405ca49c2eb2b7e8adb8a09ea3f0714b0                                                                                 
2015-05-15 13:54:57.634956 [DEBUG] sofia.c:9011 IP XXX.XXX.XXX.XXX Rejected by acl "domains". Falling back to Digest auth.
nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering
nua.c:610 nua_set_hparams() nua: nua_r_set_params with invalid handle (nil)
2015-05-15 13:54:57.634956 [DEBUG] sofia.c:10109 Setting NAT mode based on via received
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:54:57.634956 [DEBUG] sofia.c:6623 Channel sofia/internal/1001 at sip.domain.com entering state [received][100]
2015-05-15 13:54:57.634956 [DEBUG] sofia.c:6633 Remote SDP:
v=0                                                                                                                                
o=- 3640679697 3640679697 IN IP4 192.168.51.35                                                                                     
s=pjmedia                                                                                                                          
c=IN IP4 192.168.51.35                                                                                                             
t=0 0                                                                                                                              
m=audio 4000 RTP/AVP 8 3 101                                                                                                       
c=IN IP4 192.168.51.35                                                                                                             
a=rtpmap:8 PCMA/8000                                                                                                               
a=rtpmap:3 GSM/8000                                                                                                                
a=rtpmap:101 telephone-event/8000                                                                                                  
a=fmtp:101 0-16                                                                                                                    
a=rtcp:4001 IN IP4 192.168.51.35                                                                                                   
a=zrtp-hash:1.10 cefe05149bdf61156a54615b2de164b405ca49c2eb2b7e8adb8a09ea3f0714b0                                                  2015-05-15 13:54:57.634956 [DEBUG] sofia.c:6899 (sofia/internal/1001 at sip.domain.com) State Change CS_NEW -> CS_INIT
2015-05-15 13:54:57.634956 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_INIT
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1001 at sip.domain.com) State INIT
2015-05-15 13:54:57.634956 [DEBUG] mod_sofia.c:87 sofia/internal/1001 at sip.domain.com SOFIA INIT
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1001 at sip.domain.com Standard INIT
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1001 at sip.domain.com) State Change CS_INIT -> CS_ROUTING                                                                                                                              
2015-05-15 13:54:57.634956 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1001 at sip.domain.com) State INIT going to sleep
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_ROUTING                                                                                                                                
2015-05-15 13:54:57.634956 [DEBUG] switch_channel.c:2184 (sofia/internal/1001 at sip.domain.com) Callstate Change DOWN -> RINGING
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1001 at sip.domain.com) State ROUTING
2015-05-15 13:54:57.634956 [DEBUG] mod_sofia.c:123 sofia/internal/1001 at sip.domain.com SOFIA ROUTING
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:166 sofia/internal/1001 at sip.domain.com Standard ROUTING
2015-05-15 13:54:57.634956 [INFO] mod_dialplan_xml.c:635 Processing 1001 <1001>->1004 in context sip.domain.com
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->PHONE-FAX_b_1001_49123456789100] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [PHONE-FAX_b_1001_49123456789100] context(sip.domain.com) =~ /sip.domain.com/ break=on-false                                                                                                                
Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL) [PHONE-FAX_b_1001_49123456789100] destination_number(1004) =~ /^(1234567e0)$/ break=on-false                                                                                                              
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->PHONE-FAX_l_1002_49123456789101] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [PHONE-FAX_l_1002_49123456789101] context(sip.domain.com) =~ /sip.domain.com/ break=on-false                                                                                                               
Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL) [PHONE-FAX_l_1002_49123456789101] destination_number(1004) =~ /^(1234567e1)$/ break=on-false                                                                                                             
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->PHONE-FAX_m_1003_49123456789101-copy] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [PHONE-FAX_m_1003_49123456789101-copy] context(sip.domain.com) =~ /sip.domain.com/ break=on-false                                                                                                         
Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL) [PHONE-FAX_m_1003_49123456789101-copy] destination_number(1004) =~ /^(1234567e2)$/ break=on-false                                                                                                       
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->internal-voicemail-com] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal-voicemail-com] context(sip.domain.com) =~ /sip.domain.com/ break=on-false                                                                                                                             
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal-voicemail-com] caller_id_number(1001) =~ /^(((\+|00)YY)|1\d{3})$/ break=on-false                                                                                                                    
Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL) [internal-voicemail-com] ${user_data(${destination_number}@${domain} param vm-enabled)}(false) =~ /true/ break=on-false                                                                                       
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->internal-voicemail-de] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal-voicemail-de] context(sip.domain.com) =~ /sip.domain.com/ break=on-false                                                                                                                             
Dialplan: sofia/internal/1001 at sip.domain.com Regex (FAIL) [internal-voicemail-de] ${user_data(${destination_number}@${domain} param vm-enabled)}(false) =~ /true/ break=on-false                                                                                       
Dialplan: sofia/internal/1001 at sip.domain.com parsing [sip.domain.com->internal] continue=false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal] context(sip.domain.com) =~ /sip.domain.com/ break=on-false
Dialplan: sofia/internal/1001 at sip.domain.com Regex (PASS) [internal] destination_number(1004) =~ /^(1\d{3})$/ break=on-false
Dialplan: sofia/internal/1001 at sip.domain.com Action set(zrtp_enrollment=true)
Dialplan: sofia/internal/1001 at sip.domain.com Action set(hangup_after_bridge=false)
Dialplan: sofia/internal/1001 at sip.domain.com Action set(continue_on_fail=true)
Dialplan: sofia/internal/1001 at sip.domain.com Action set(intcallid=1004)
Dialplan: sofia/internal/1001 at sip.domain.com Action bridge(sofia/internal/${intcallid}%${domain})
Dialplan: sofia/internal/1001 at sip.domain.com Action set(eml=${user_data(${intcallid}@${domain} param vm-mailto)})
Dialplan: sofia/internal/1001 at sip.domain.com Action set(datetime=${strftime(%Y.%m.%d)} | ${strftime(%W)}. HET | ${strftime(%H:%M)})
Dialplan: sofia/internal/1001 at sip.domain.com Action set(smtp_from=root)
Dialplan: sofia/internal/1001 at sip.domain.com Action lua(NoAns.lua &#39;${originate_disposition}&#39; &#39;${eml}&#39; &#39;${smtp_from}&#39; &#39;aaa&#39; &#39;bbb&#39; &#39;ccc&#39;)                                                                                                      
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1001 at sip.domain.com) State Change CS_ROUTING -> CS_EXECUTE                                                                                                                          
2015-05-15 13:54:57.634956 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1001 at sip.domain.com) State ROUTING going to sleep
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_EXECUTE                                                                                                                                
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1001 at sip.domain.com) State EXECUTE
2015-05-15 13:54:57.634956 [DEBUG] mod_sofia.c:178 sofia/internal/1001 at sip.domain.com SOFIA EXECUTE
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1001 at sip.domain.com Standard EXECUTE
EXECUTE sofia/internal/1001 at sip.domain.com set(zrtp_enrollment=true)
2015-05-15 13:54:57.634956 [DEBUG] mod_dptools.c:1445 sofia/internal/1001 at sip.domain.com SET [zrtp_enrollment]=[true]
EXECUTE sofia/internal/1001 at sip.domain.com set(hangup_after_bridge=false)
2015-05-15 13:54:57.634956 [DEBUG] mod_dptools.c:1445 sofia/internal/1001 at sip.domain.com SET [hangup_after_bridge]=[false]
EXECUTE sofia/internal/1001 at sip.domain.com set(continue_on_fail=true)
2015-05-15 13:54:57.634956 [DEBUG] mod_dptools.c:1445 sofia/internal/1001 at sip.domain.com SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1001 at sip.domain.com set(intcallid=1004)
2015-05-15 13:54:57.634956 [DEBUG] mod_dptools.c:1445 sofia/internal/1001 at sip.domain.com SET [intcallid]=[1004]
EXECUTE sofia/internal/1001 at sip.domain.com bridge(sofia/internal/1004%sip.domain.com)
2015-05-15 13:54:57.634956 [DEBUG] switch_channel.c:1201 sofia/internal/1001 at sip.domain.com EXPORTING[export_vars] [domain_name]=[sip.domain.com] to event                                                                                                              
2015-05-15 13:54:57.634956 [DEBUG] switch_ivr_originate.c:2100 Parsing global variables
2015-05-15 13:54:57.634956 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/1004 [e6f3933c-e8f7-4590-af6c-4de82995ee46]
2015-05-15 13:54:57.634956 [DEBUG] mod_sofia.c:4701 (sofia/internal/1004) State Change CS_NEW -> CS_INIT
2015-05-15 13:54:57.634956 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-05-15 13:54:57.634956 [DEBUG] switch_core_media.c:266 Passing a-leg remote zrtp-hash (audio) to b-leg
2015-05-15 13:54:57.634956 [DEBUG] mod_sofia.c:4771 [zrtp_passthru] Setting a-leg inherit_codec=true
2015-05-15 13:54:57.634956 [DEBUG] mod_sofia.c:4774 [zrtp_passthru] Setting b-leg absolute_codec_string=&#39;PCMA at 8000h@20i at 64000b,GSM at 8000h@20i at 13200b&#39;                                                                                                                  
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_INIT
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1004) State INIT
2015-05-15 13:54:57.634956 [DEBUG] mod_sofia.c:87 sofia/internal/1004 SOFIA INIT
2015-05-15 13:54:57.634956 [DEBUG] switch_core_media.c:6137 Adding audio a=zrtp-hash:1.10 cefe05149bdf61156a54615b2de164b405ca49c2eb2b7e8adb8a09ea3f0714b0                                                                                                            
nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
2015-05-15 13:54:57.634956 [DEBUG] sofia_glue.c:1203 sip:1004 at XXX.XXX.XXX.XXX:15194;transport=tls;registering_acc=sip_domain_com Setting proxy route to sofia/internal/1004                                                                                              
2015-05-15 13:54:57.634956 [DEBUG] sofia_glue.c:1232 sofia/internal/1004 sending invite version: 1.4.18 -3-1 64bit
Local SDP:                                                                                                                         
v=0                                                                                                                                
o=FreeSWITCH 1431670287 1431670288 IN IP4 YYY.YYY.YYY.YYY                                                                           
s=FreeSWITCH                                                                                                                       
c=IN IP4 YYY.YYY.YYY.YYY                                                                                                            
t=0 0                                                                                                                              
m=audio 20610 RTP/AVP 8 3 101 13                                                                                                   
a=rtpmap:8 PCMA/8000                                                                                                               
a=rtpmap:3 GSM/8000                                                                                                                
a=rtpmap:101 telephone-event/8000                                                                                                  
a=fmtp:101 0-16                                                                                                                    
a=ptime:20                                                                                                                         
a=zrtp-hash:1.10 cefe05149bdf61156a54615b2de164b405ca49c2eb2b7e8adb8a09ea3f0714b0                                                  
a=sendrecv                                                                                                                         nua.c:633 nua_invite() nua: nua_invite: entering
nua_stack.c:529 nua_signal() nua(0x7fcccc00a070): sent signal r_invite
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:40 sofia/internal/1004 Standard INIT
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/1004) State Change CS_INIT -> CS_ROUTING
2015-05-15 13:54:57.634956 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:512 (sofia/internal/1004) State INIT going to sleep
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0x7fccf0001be0, 0x7fccf0000cb0, 0x7fcccc00a070) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fccf008b650, ...) called
soa.c:403 soa_set_params() soa_set_params(static::0x7fccf008b650, ...) called
soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fccf008b650, (nil), 0x7fcccc0062b6, -1) called
soa.c:890 soa_set_capability_sdp() soa_set_capability_sdp(static::0x7fccf008b650, (nil), 0x7fcccc0062b6, -1) called
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_ROUTING
nua_dialog.c:338 nua_dialog_usage_add() nua(0x7fcccc00a070): adding session usage
nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7fccf00b0bb0)
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7fccf008b650) called
soa.c:1426 soa_generate_offer() soa_generate_offer(static::0x7fccf008b650, 0) called
soa_static.c:1146 offer_answer_step() soa_static_offer_answer_action(0x7fccf008b650, soa_generate_offer): called
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1004) State ROUTING
soa_static.c:1187 offer_answer_step() soa_static(0x7fccf008b650, soa_generate_offer): generating local description
soa_static.c:1215 offer_answer_step() soa_static(0x7fccf008b650, soa_generate_offer): upgrade with local description
2015-05-15 13:54:57.634956 [DEBUG] mod_sofia.c:123 sofia/internal/1004 SOFIA ROUTING
2015-05-15 13:54:57.634956 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/1004) State Change CS_ROUTING -> CS_CONSUME_MEDIA
soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fcd0bdfaa70, (nil), ""): called
2015-05-15 13:54:57.634956 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:528 (sofia/internal/1004) State ROUTING going to sleep
soa_static.c:1444 offer_answer_step() soa_static(0x7fccf008b650, soa_generate_offer): storing local description
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fccf008b650, [(nil)], [0x7fcd0bdfcbf8], [0x7fcd0bdfcbf4]) called
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
tport.c:4588 tport_by_name() tport(0x7fccf0004620): found 0x7fccf0077460 by name tls/XXX.XXX.XXX.XXX:15194
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_CONSUME_MEDIA
tport.c:3257 tport_tsend() tport_tsend(0x7fccf0077460) tpn = tls/XXX.XXX.XXX.XXX:15194
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/1004) State CONSUME_MEDIA
2015-05-15 13:54:57.634956 [DEBUG] switch_core_state_machine.c:547 (sofia/internal/1004) State CONSUME_MEDIA going to sleep
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0057610 0x7fccf00b3af0 1018 (1018)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0057610 0x7fccf00b13f0 90 (90)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0057610 0x7fccf00b2ff0 333 (333)
tport.c:3492 tport_send_msg() tport_vsend returned 1441
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf0077460): reset timer
nta.c:8304 outgoing_send() nta: sent INVITE (75509384) to tls/XXX.XXX.XXX.XXX:15194
tport.c:4160 tport_pend() tport_pend(0x7fccf0077460): pending 0x7fccf00b1040 for tls/XXX.XXX.XXX.XXX:15194 (already 0)
nua_session.c:4139 signal_call_state_change() nua(0x7fcccc00a070): call state changed: init -> calling, sent offer
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fccf008b650, [0x7fcd0bdfcbd8], [0x7fcd0bdfcbe0], [(nil)]) called
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-15 13:54:57.634956 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:54:57.634956 [DEBUG] sofia.c:6623 Channel sofia/internal/1004 entering state [calling][0]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
tport.c:2773 tport_wakeup() tport_wakeup(0x7fccf0077460): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fccf0077460)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fccf0077460): tls_read() returned 1
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fccf0077460) msg 0x7fccf00b40c0 from (tls/XXX.XXX.XXX.XXX:15194) has 1 bytes, veclen = 1
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf0077460): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7fccf0077460): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fccf0077460)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fccf0077460): tls_read() returned 497
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fccf0077460) msg 0x7fccf00b40c0 from (tls/XXX.XXX.XXX.XXX:15194) has 497 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fccf0077460): msg 0x7fccf00b40c0 (498 bytes) from tls/XXX.XXX.XXX.XXX:15194/sips next=(nil)
nta.c:3299 agent_recv_response() nta: received 180 Ringing for INVITE (75509384)
nta.c:3366 agent_recv_response() nta: 180 Ringing is going to a transaction
nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 180.054 ms
tport.c:4222 tport_release() tport_release(0x7fccf0077460): 0x7fccf00b1040 by 0x7fccf00b3150 with 0x7fccf00b40c0 (preliminary)
nua_session.c:4139 signal_call_state_change() nua(0x7fcccc00a070): call state changed: calling -> proceeding
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf0077460): reset timer
2015-05-15 13:54:57.814956 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-15 13:54:57.814956 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:54:57.814956 [DEBUG] sofia.c:6623 Channel sofia/internal/1004 entering state [proceeding][180]
2015-05-15 13:54:57.814956 [NOTICE] sofia.c:6725 Ring-Ready sofia/internal/1004!
2015-05-15 13:54:57.814956 [DEBUG] switch_channel.c:3277 (sofia/internal/1004) Callstate Change DOWN -> RINGING
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:54:57.854927 [INFO] switch_ivr_originate.c:1192 Sending early media
2015-05-15 13:54:57.854927 [DEBUG] switch_core_media.c:344 Found audio zrtp-hash; setting r_sdp_audio_zrtp_hash=1.10 cefe05149bdf61156a54615b2de164b405ca49c2eb2b7e8adb8a09ea3f0714b0                                                                                 
2015-05-15 13:54:57.854927 [DEBUG] switch_core_media.c:266 Passing a-leg remote zrtp-hash (audio) to b-leg
2015-05-15 13:54:57.854927 [DEBUG] switch_core_media.c:3632 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-05-15 13:54:57.854927 [DEBUG] switch_core_media.c:3687 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-05-15 13:54:57.854927 [DEBUG] switch_core_media.c:3632 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-05-15 13:54:57.854927 [DEBUG] switch_core_media.c:3632 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1]
2015-05-15 13:54:57.854927 [DEBUG] switch_core_media.c:3632 Audio Codec Compare [GSM:3:8000:20:13200:1]/[GSM:3:8000:20:13200:1]
2015-05-15 13:54:57.854927 [DEBUG] switch_core_media.c:3687 Audio Codec Compare [GSM:3:8000:20:13200:1] ++++ is saved as a match
2015-05-15 13:54:57.854927 [DEBUG] switch_core_media.c:3548 Set telephone-event payload to 101
2015-05-15 13:54:57.854927 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/1001 at sip.domain.com PCMA/8000 20 ms 160 samples 64000 bits 1 channels                                                                                                             
2015-05-15 13:54:57.854927 [DEBUG] switch_core_codec.c:111 sofia/internal/1001 at sip.domain.com Original read codec set to PCMA:8
2015-05-15 13:54:57.854927 [DEBUG] switch_core_media.c:3895 Set 2833 dtmf send/recv payload to 101
2015-05-15 13:54:57.874951 [DEBUG] switch_core_media.c:5147 AUDIO RTP [sofia/internal/1001 at sip.domain.com] YYY.YYY.YYY.YYY port 20514 -> 192.168.51.35 port 4000 codec: 8 ms: 20                                                                                        
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:3562 Starting timer [soft] 160 bytes per 20ms
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]: START SESSION INITIALIZATION. sID=4.
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]: ZID=346661636436616664366363.
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]: Loading User&#39;s profile:
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:    allowclear: OFF
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:    autosecure: ON
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:  disclose_bit: OFF
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:  signal. role: Unknown
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:           TTL: 4294967295
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:   SAS schemes:
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370 B256
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370 B32  
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:      Ciphers:
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370 AES3
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370 AES1
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:    PK schemes:
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370 EC25
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370 DH3k
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370 DH2k
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370 Mult
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:           ATL:
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370 HS32
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:       Hashes:
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370 S256
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]: Session initialization - DONE. sID=4.2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]: ATTACH NEW STREAM to sID=4:
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=0 UNKNOWN switching <NONE> ---> <ACTIVE>.
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:   Empty slot was found - initializing new stream with ID=4.
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]:   Preparing ZRTP Hello according to the Session profile.
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp main]: ATTACH NEW STREAM - DONE.
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [   zrtp mitm]: START REGISTRATION STREAM ID=4 mode=CLEAR state=ACTIVE.
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: START STREAM ID=4 mode=CLEAR state=ACTIVE.
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=4 CLEAR switching <ACTIVE> ---> <START>.
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Send <HELLO> ssrc=2237203113 seq=28453 size=144. Stream 4:CLEAR:START                                                                                                                         
2015-05-15 13:54:57.874951 [INFO] switch_core_media.c:5364 Activating RTCP PORT 4001
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:3912 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 4001
2015-05-15 13:54:57.874951 [DEBUG] switch_rtp.c:2367 Setting RTCP remote addr to 192.168.51.35:4001
2015-05-15 13:54:57.874951 [DEBUG] switch_core_media.c:5445 Set 2833 dtmf send payload to 101
2015-05-15 13:54:57.874951 [DEBUG] switch_core_media.c:5451 Set 2833 dtmf receive payload to 101
2015-05-15 13:54:57.874951 [INFO] switch_channel.c:3321 sofia/internal/1001 at sip.domain.com ZRTP not negotiated on both sides; disabling ZRTP passthru mode.                                                                                                            
2015-05-15 13:54:57.874951 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1001 at sip.domain.com!
2015-05-15 13:54:57.874951 [DEBUG] switch_channel.c:3399 (sofia/internal/1001 at sip.domain.com) Callstate Change RINGING -> EARLY
2015-05-15 13:54:57.874951 [DEBUG] mod_sofia.c:2268 Ring SDP:
v=0                                                                                                                                
o=FreeSWITCH 1431670383 1431670384 IN IP4 YYY.YYY.YYY.YYY                                                                           
s=FreeSWITCH                                                                                                                       
c=IN IP4 YYY.YYY.YYY.YYY                                                                                                            
t=0 0                                                                                                                              
m=audio 20514 RTP/AVP 8 101                                                                                                        
a=rtpmap:8 PCMA/8000                                                                                                               
a=rtpmap:101 telephone-event/8000                                                                                                  
a=fmtp:101 0-16                                                                                                                    
a=ptime:20                                                                                                                         
a=sendrecv                                                                                                                         
a=rtcp:20515 IN IP4 YYY.YYY.YYY.YYY                                                                                                 nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:529 nua_signal() nua(0x7fccf0091540): sent signal r_respond
2015-05-15 13:54:57.874951 [DEBUG] switch_core_session.c:912 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-05-15 13:54:57.874951 [DEBUG] switch_ivr_originate.c:1249 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
2015-05-15 13:54:57.874951 [DEBUG] switch_core_codec.c:221 sofia/internal/1001 at sip.domain.com Push codec L16:100
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fccf0091660, ...) called
soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fccf0091660, (nil), 0x7fcd30003af8, -1) called
soa.c:890 soa_set_capability_sdp() soa_set_capability_sdp(static::0x7fccf0091660, (nil), 0x7fcd30003af8, -1) called
2015-05-15 13:54:57.874951 [DEBUG] switch_ivr_originate.c:1317 Play Ringback Tone [%(1000,4000,425,0)]
nua_session.c:2320 nua_invite_server_respond() nua: nua_invite_server_respond: entering
soa.c:1515 soa_generate_answer() soa_generate_answer(static::0x7fccf0091660) called
soa_static.c:1146 offer_answer_step() soa_static_offer_answer_action(0x7fccf0091660, soa_generate_answer): called
soa_static.c:1187 offer_answer_step() soa_static(0x7fccf0091660, soa_generate_answer): generating local description
soa_static.c:1228 offer_answer_step() soa_static(0x7fccf0091660, soa_generate_answer): upgrade with remote description
soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fcd0bdfaab0, 0x7fccf008b8c0, ""): called
soa_static.c:1444 offer_answer_step() soa_static(0x7fccf0091660, soa_generate_answer): storing local description
soa.c:1730 soa_activate() soa_activate(static::0x7fccf0091660, (nil)) called
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fccf0091660, [(nil)], [0x7fcd0bdfcc38], [0x7fcd0bdfcc34]) called
tport.c:3257 tport_tsend() tport_tsend(0x7fccf00a2eb0) tpn = TLS/XXX.XXX.XXX.XXX:15193
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0079900 0x7fccf00b8780 956 (956)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0079900 0x7fccf00b7e30 260 (260)
tport.c:3492 tport_send_msg() tport_vsend returned 1216
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
nta.c:6791 incoming_reply() nta: sent 183 Session Progress for INVITE (28199)
nua_session.c:4139 signal_call_state_change() nua(0x7fccf0091540): call state changed: received -> early, sent answer
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fccf0091660, [0x7fcd0bdfcce8], [0x7fcd0bdfccf0], [(nil)]) called
soa.c:616 soa_get_params() soa_get_params(static::0x7fccf0091660, ...) called
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-15 13:54:57.874951 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:54:57.894987 [DEBUG] sofia.c:6623 Channel sofia/internal/1001 at sip.domain.com entering state [early][183]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:54:57.934958 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Send <HELLO> ssrc=2237203113 seq=28454 size=144. Stream 4:CLEAR:START                                                                                                                         
2015-05-15 13:54:57.994988 [DEBUG] switch_rtp.c:1955 rtcp_stats_init: ssrc[644801348] base_seq[30552]
2015-05-15 13:54:58.014989 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Send <HELLO> ssrc=2237203113 seq=28455 size=144. Stream 4:CLEAR:START                                                                                                                         
2015-05-15 13:54:58.155010 [INFO] switch_rtp.c:5846 Auto Changing port from 192.168.51.35:4000 to XXX.XXX.XXX.XXX:14554
2015-05-15 13:54:58.155010 [DEBUG] switch_rtp.c:2367 Setting RTCP remote addr to XXX.XXX.XXX.XXX:14555
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [  zrtp utils]: Received <Hello   > packet with ssrc=644801348 seq=1/1 size=168. Stream4:CLEAR:START.                                                                                                           
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [ zrtp engine]:   Processing HELLO from PJS ZRTP 3.0.0   V=1.10, P=0, M=0.
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [ zrtp engine]:           ac=4 cc=4 sc=1 kc=5
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [ zrtp engine]:           S384S2562FS3AES32FS1AES1SK32SK64HS32HS80EC25DH3kEC38DH2kMultB32                                                                                                                       
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [ zrtp engine]:   Received HELLO had the same protocol V.
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   _zrtp_choose_best_comp() for PKT. local=EC25 remote=EC25, choosen=EC25                                                                                                                        
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [ zrtp engine]:   Received HELLO Accepted
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   ache_get(): zid1=346661636436616664366363, zis2=1b1f26a336ab4d86859e557d MiTM=NO                                                                                                              
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   ache_get(): zid1=346661636436616664366363, zis2=1b1f26a336ab4d86859e557d MiTM=NO                                                                                                              
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   ache_get(): zid1=346661636436616664366363, zis2=1b1f26a336ab4d86859e557d MiTM=YES                                                                                                             
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Restoring Secrets: lZID=346661636436616664366363 rZID=1b1f26a336ab4d86859e557d. V=1 sID=4                                                                                                     
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:           RS1 <8b9b28eb023b7ddc32ef8690ddf28649115d721b22be9d81f24b772184093b6d>                                                                                                                
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:           RS2 <523e305c538d5e2c5b971ab21d1a29ad378bef1615515130e022735fe25006bd>                                                                                                                
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:           PBX <9d3658c6a665804d913f7b8552897605ed76c691a081332a5ff9d4d441abf482>                                                                                                                
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Send <HELLOACK> ssrc=2237203113 seq=28456 size=28. Stream 4:CLEAR:START                                                                                                                       
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=4 CLEAR switching <START> ---> <W4HACK>.
2015-05-15 13:54:58.234936 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Send <HELLO> ssrc=2237203113 seq=28457 size=144. Stream 4:CLEAR:W4HACK                                                                                                                        
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1955 rtcp_stats_init: ssrc[644801348] base_seq[2]
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [  zrtp utils]: Received <HelloACK> packet with ssrc=644801348 seq=2/2 size=28. Stream4:CLEAR:W4HACK.                                                                                                           
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=4 CLEAR switching <W4HACK> ---> <SINITSEC>.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [ zrtp engine]:   Initiating Secure iteration... ID=4.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [ zrtp engine]:   Got mode=DH. Check approval of starting.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [ zrtp engine]:   Mode=DH Cccepted. Starting ZRTP Initiator Protocol.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:   ENTER STATE INITIATING SECURE for ID=4 mode=DH state=SINITSEC.                                                                                                                                
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:   Initiator selected following options:
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:         Hash: S256
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:       Cipher: AES3
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:          ATL: HS32
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:   VAD scheme: B32
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Init INITIATOR&#39;s Protocol ID=4 mode=DH...
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [   zrtp ecdh]:   DH TEST: zrtp_ecdh_initialize() for EC25 was executed by 10ms.                                                                                                                                
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Attach RS id=e616852a885a3e89.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Attach RS peer_id=4aa5f81d3b9b6f2e.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Attach RS id=4cd99f03438d3ebc.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Attach RS peer_id=b9bf9e8c9f89cc58.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Attach RS id=ecf6dd533343a217.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Attach RS peer_id=15d7a9d63a9da7f8.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Attach RS id=c9bad87c4fb5bb8b.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Attach RS peer_id=1695ac21336ec842.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=4 DH switching <SINITSEC> ---> <INITSEC>.
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:   Start Sending COMMIT ID=4 mode=DH state=INITSEC:
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:         Hash: S256
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:       Cipher: AES3
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:          ATL: HS32
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:    PK scheme: EC25
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:   VAD scheme: B32
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [zrtp initiat]:           hv: 9b03f61329f4e5badc74b034b0dfde4615eacb5a8a9b68d5ecf3194ea20bf45d                                                                                                                  
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Send <COMMIT> ssrc=2237203113 seq=28458 size=132. Stream 4:DH:INITSEC                                                                                                                         
2015-05-15 13:54:58.334911 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.354913 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.374938 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.394966 [DEBUG] switch_rtp.c:1955 rtcp_stats_init: ssrc[644801348] base_seq[30570]
2015-05-15 13:54:58.394966 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.394966 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [  zrtp utils]: Received <DHPart1 > packet with ssrc=644801348 seq=3/3 size=164. Stream4:DH:INITSEC.                                                                                                            
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [   zrtp ecdh]:   DH TEST: zrtp_ecdh_validate() for EC25 was executed by 0ms.
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Send <DH2> ssrc=2237203113 seq=28459 size=164. Stream 4:DH:INITSEC                                                                                                                            
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]: ---------------------------------------------------
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   SWITCHING TO SRTP. ID=1092774504
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   I Initiator
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   DERIVE S0 from DH exchange and RS secrets...
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:          my rs1ID:e616852a885a3e89
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:         his rs1ID:4aa5f81d3b9b6f2e
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:    his rs1ID comp:4aa5f81d3b9b6f2e
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:          my rs2ID:4cd99f03438d3ebc
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:         his rs2ID:41ee3d31e442e2db
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:    his rs2ID comp:b9bf9e8c9f89cc58
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:         my pbxsID:c9bad87c4fb5bb8b
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:        his pbxsID:f7d894dc7a1b0333
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   his pbxsID comp:1695ac21336ec842
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]: ECDH comp_length=32
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [   zrtp ecdh]:   DH TEST: zrtp_ecdh_compute() for EC25 was executed by 9ms.
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Use S1 in calculations.
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=4 DH switching <INITSEC> ---> <WCONFIRM>.
2015-05-15 13:54:58.434968 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.475050 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.494970 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.514973 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.514973 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.554962 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.554962 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.594931 [DEBUG] switch_rtp.c:1955 rtcp_stats_init: ssrc[644801348] base_seq[4]
2015-05-15 13:54:58.594931 [DEBUG] switch_rtp.c:1370  [  zrtp utils]: Received <Confirm1> packet with ssrc=644801348 seq=4/4 size=92. Stream4:DH:WCONFIRM.                                                                                                            
2015-05-15 13:54:58.594931 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=4 DH switching <WCONFIRM> ---> <W4CONFACK>.
2015-05-15 13:54:58.594931 [DEBUG] switch_rtp.c:1370  [zrtp protoco]: HMAC TRACE. COMPUTE.
2015-05-15 13:54:58.594931 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   cipher text:593352f2f8a9e76f6c2f61f8cb5f4cc9e2627db9fa2e660a198a30fe7f529e847829a83d11b9ca66. size=40                                                                                         
2015-05-15 13:54:58.594931 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:           key:24626b9d001ba9aafc3be5c9cdc872753f5d109589e4e0a3e2d22c3ffd77068a.                                                                                                                 
2015-05-15 13:54:58.594931 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:    comp hmac:9d4563b8b476288993c4ae0cd340a8da6267ffec857580c1949e0ee025ed0ef7.                                                                                                                  
2015-05-15 13:54:58.594931 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:         hmac:9d4563b8b4762889.
2015-05-15 13:54:58.594931 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Send <CONFIRM2> ssrc=2237203113 seq=28460 size=92. Stream 4:DH:W4CONFACK                                                                                                                      
2015-05-15 13:54:58.594931 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.614967 [DEBUG] switch_rtp.c:1370  [zrtp protoco]: ERROR! Decrypt failed. ID=4:DH s=SRTP authentication failure (RTP size=172 ssrc=644801348 seq=30579/30579 pt=8)                                                                                 
2015-05-15 13:54:58.614967 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.614967 [DEBUG] switch_rtp.c:1955 rtcp_stats_init: ssrc[644801348] base_seq[30580]
2015-05-15 13:54:58.614967 [DEBUG] switch_rtp.c:1370  [zrtp protoco]: ERROR! Decrypt failed. ID=4:DH s=SRTP authentication failure (RTP size=172 ssrc=644801348 seq=30580/30580 pt=8)                                                                                 
2015-05-15 13:54:58.614967 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.654948 [DEBUG] switch_rtp.c:1370  [zrtp protoco]: ERROR! Decrypt failed. ID=4:DH s=SRTP authentication failure (RTP size=172 ssrc=644801348 seq=30581/30581 pt=8)                                                                                 
2015-05-15 13:54:58.654948 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.674985 [DEBUG] switch_rtp.c:1370  [zrtp protoco]: ERROR! Decrypt failed. ID=4:DH s=SRTP authentication failure (RTP size=172 ssrc=644801348 seq=30582/30582 pt=8)                                                                                 
2015-05-15 13:54:58.674985 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.694967 [DEBUG] switch_rtp.c:1370  [zrtp protoco]: ERROR! Decrypt failed. ID=4:DH s=SRTP authentication failure (RTP size=172 ssrc=644801348 seq=30583/30583 pt=8)                                                                                 
2015-05-15 13:54:58.694967 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.694967 [DEBUG] switch_rtp.c:1370  [zrtp protoco]: ERROR! Decrypt failed. ID=4:DH s=SRTP authentication failure (RTP size=172 ssrc=644801348 seq=30584/30584 pt=8)                                                                                 
2015-05-15 13:54:58.694967 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.734933 [DEBUG] switch_rtp.c:1370  [zrtp protoco]: ERROR! Decrypt failed. ID=4:DH s=SRTP authentication failure (RTP size=172 ssrc=644801348 seq=30585/30585 pt=8)                                                                                 
2015-05-15 13:54:58.734933 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: Add 172 bytes of entropy to the RNG pool.
2015-05-15 13:54:58.734933 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Send <CONFIRM2> ssrc=2237203113 seq=28461 size=92. Stream 4:DH:W4CONFACK                                                                                                                      
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp utils]: Received <Conf2ACK> packet with ssrc=644801348 seq=5/5 size=28. Stream4:DH:W4CONFACK.                                                                                                           
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Enter state SECURE (DH).
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   This is the very first stream in sID GENERATING SAS value.
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   SAS computed: <45zt> <>.
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Check expiration interval: last_use=1431564469 ttl=4294967295 new_ttl=4294967295 exp=1431564468 now=1431690898                                                                                
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   Flags C=26 M=2 W=0 ID=4
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=4 DH switching <W4CONFACK> ---> <SECURE>.
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   Storing ZRTP cache to </var/lib/freeswitch/db/zrtp.dat>...
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   INFO! The user requires new un-enrolment - the nedpint may clear the cache or perform other action. ID=4                                                                                      
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1341 User unenrolled!
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   INFO! The user requires new enrolment - generate new MiTM secret. ID=4                                                                                                                        
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [   zrtp mitm]: MARKING this call as REGISTRATION ID=4
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   cache_put() zid1=346661636436616664366363, zis2=1b1f26a336ab4d86859e557d MiTM=YES                                                                                                             
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   cache_put() Just update existing value.
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   Storing ZRTP cache to </var/lib/freeswitch/db/zrtp.dat>...
2015-05-15 13:54:58.754926 [DEBUG] switch_core_sqldb.c:2599 Secure Type: zrtp:45zt:
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   1 out of 1 MiTM cache entries have been flushed successfully.                                                                                                                                 
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [   zrtp mitm]: Makring this call as REGISTRATION - DONE
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1334 New user enrolled!
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   cache_put() zid1=346661636436616664366363, zis2=1b1f26a336ab4d86859e557d MiTM=NO                                                                                                              
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   cache_put() Just update existing value.
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   Storing ZRTP cache to </var/lib/freeswitch/db/zrtp.dat>...
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   1 out of 3 regular cache entries have been flushed successfully.                                                                                                                              
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:   New secret was generated:
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:           RS1 value:<c450a01dfff94ed1e4f06c0171cfc2a63ec32c0e43535a6782b814afb96d4dfa>                                                                                                          
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [zrtp protoco]:           TTL=4294967295, flags C=26 M=22 W=0 V=1
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   Storing ZRTP cache to </var/lib/freeswitch/db/zrtp.dat>...
2015-05-15 13:54:58.754926 [DEBUG] switch_rtp.c:1370  [  zrtp cache]:   1 out of 3 regular cache entries have been flushed successfully.                                                                                                                              
2015-05-15 13:54:58.915009 [DEBUG] switch_rtp.c:1955 rtcp_stats_init: ssrc[644801348] base_seq[6]
2015-05-15 13:54:58.915009 [DEBUG] switch_rtp.c:1370  [  zrtp utils]: Received <Conf2ACK> packet with ssrc=644801348 seq=6/6 size=28. Stream4:DH:SECURE.                                                                                                              
2015-05-15 13:54:58.974962 [DEBUG] switch_rtp.c:1955 rtcp_stats_init: ssrc[644801348] base_seq[30595]
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fcd0bdfcc60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 0/0 term, 1/2 free
nta.c:1296 agent_timer() nta: timer set next to 58413 ms
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
freeswitch at internal>
tport.c:2773 tport_wakeup() tport_wakeup(0x7fccf0077460): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fccf0077460)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fccf0077460): tls_read() returned 1
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fccf0077460) msg 0x7fccf00a8080 from (tls/XXX.XXX.XXX.XXX:15194) has 1 bytes, veclen = 1
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf0077460): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7fccf0077460): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fccf0077460)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fccf0077460): tls_read() returned 678
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fccf0077460) msg 0x7fccf00a8080 from (tls/XXX.XXX.XXX.XXX:15194) has 678 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fccf0077460): msg 0x7fccf00a8080 (679 bytes) from tls/XXX.XXX.XXX.XXX:15194/sips next=(nil)
nta.c:3299 agent_recv_response() nta: received 200 OK for INVITE (75509384)
nta.c:3366 agent_recv_response() nta: 200 OK is going to a transaction
tport.c:4222 tport_release() tport_release(0x7fccf0077460): 0x7fccf00b1040 by 0x7fccf00b3150 with 0x7fccf00a8080
nta.c:1348 set_timeout() nta: timer shortened to 32000 ms
soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7fccf008b650, (nil), 0x7fccf0092c9e, 153) called
soa.c:1595 soa_process_answer() soa_process_answer(static::0x7fccf008b650) called
soa_static.c:1146 offer_answer_step() soa_static_offer_answer_action(0x7fccf008b650, soa_process_answer): called
soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7fccf00b2620, 0x7fccf00a1400, ""): called
soa_static.c:1302 offer_answer_step() soa_static(0x7fccf008b650, soa_process_answer): upgrade codecs with remote description
soa.c:1730 soa_activate() soa_activate(static::0x7fccf008b650, (nil)) called
nua_session.c:988 nua_session_client_response() nua(0x7fcccc00a070): INVITE: processed SDP answer in 200 OK
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_session.c:4139 signal_call_state_change() nua(0x7fcccc00a070): call state changed: proceeding -> completing, received answer
soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7fccf008b650, [0x7fcd0bdfc5d8], [0x7fcd0bdfc5e0], [(nil)]) called
soa.c:616 soa_get_params() soa_get_params(static::0x7fccf008b650, ...) called
2015-05-15 13:55:05.694950 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf0077460): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-15 13:55:05.694950 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:55:05.694950 [DEBUG] switch_core_media.c:272 Passing b-leg remote zrtp-hash (audio) to a-leg
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:55:05.694950 [DEBUG] sofia.c:6623 Channel sofia/internal/1004 entering state [completing][200]
2015-05-15 13:55:05.694950 [DEBUG] sofia.c:6633 Remote SDP:
v=0                                                                                                                                
o=1004-jitsi.org 0 0 IN IP4 192.168.51.31                                                                                          
s=-                                                                                                                                
c=IN IP4 192.168.51.31                                                                                                             
t=0 0                                                                                                                              
m=audio 5000 RTP/AVP 8 3                                                                                                           
a=rtpmap:8 PCMA/8000                                                                                                               
a=rtpmap:3 GSM/8000                                                                                                                nua.c:639 nua_ack() nua: nua_ack: entering
nua_stack.c:529 nua_signal() nua(0x7fcccc00a070): sent signal r_ack
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fccf008b650, ...) called
soa.c:1730 soa_activate() soa_activate(static::0x7fccf008b650, (nil)) called
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
tport.c:4588 tport_by_name() tport(0x7fccf0004620): found 0x7fccf0077460 by name tls/XXX.XXX.XXX.XXX:15194
tport.c:3257 tport_tsend() tport_tsend(0x7fccf0077460) tpn = tls/XXX.XXX.XXX.XXX:15194
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0057610 0x7fccf00aaf20 475 (475)
tport.c:3492 tport_send_msg() tport_vsend returned 475
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf0077460): reset timer
nta.c:8304 outgoing_send() nta: sent ACK (75509384) to tls/XXX.XXX.XXX.XXX:15194
nua_session.c:4139 signal_call_state_change() nua(0x7fcccc00a070): call state changed: completing -> ready
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-15 13:55:05.694950 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-15 13:55:05.694950 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1004 [BREAK]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:55:05.714923 [DEBUG] sofia.c:6623 Channel sofia/internal/1004 entering state [ready][200]
2015-05-15 13:55:05.714923 [DEBUG] switch_core_media.c:272 Passing b-leg remote zrtp-hash (audio) to a-leg
2015-05-15 13:55:05.714923 [DEBUG] switch_core_media.c:3632 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2015-05-15 13:55:05.714923 [DEBUG] switch_core_media.c:3687 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2015-05-15 13:55:05.714923 [DEBUG] switch_core_media.c:3632 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1]
2015-05-15 13:55:05.714923 [DEBUG] switch_core_media.c:3632 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1]
2015-05-15 13:55:05.714923 [DEBUG] switch_core_media.c:3632 Audio Codec Compare [GSM:3:8000:20:13200:1]/[GSM:3:8000:20:13200:1]
2015-05-15 13:55:05.714923 [DEBUG] switch_core_media.c:3687 Audio Codec Compare [GSM:3:8000:20:13200:1] ++++ is saved as a match
2015-05-15 13:55:05.714923 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/1004 PCMA/8000 20 ms 160 samples 64000 bits 1 channels                                                                                                                           
2015-05-15 13:55:05.714923 [DEBUG] switch_core_codec.c:111 sofia/internal/1004 Original read codec set to PCMA:8
2015-05-15 13:55:05.714923 [DEBUG] switch_core_media.c:3908 No 2833 in SDP.  Disable 2833 dtmf and switch to INFO
2015-05-15 13:55:05.714923 [DEBUG] switch_core_media.c:5147 AUDIO RTP [sofia/internal/1004] YYY.YYY.YYY.YYY port 20610 -> 192.168.51.31 port 5000 codec: 8 ms: 20                                                                                                      
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:3562 Starting timer [soft] 160 bytes per 20ms
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]: START SESSION INITIALIZATION. sID=5.
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]: ZID=346661636436616664366363.
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]: Loading User&#39;s profile:
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:    allowclear: OFF
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:    autosecure: ON
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:  disclose_bit: OFF
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:  signal. role: Initiator
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:           TTL: 4294967295
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:   SAS schemes:
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370 B256
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370 B32  
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:      Ciphers:
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370 AES3
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370 AES1
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:    PK schemes:
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370 EC25
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370 DH3k
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370 DH2k
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370 Mult
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:           ATL:
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370 HS32
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:       Hashes:
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370 S256
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]: Session initialization - DONE. sID=5.2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]: ATTACH NEW STREAM to sID=5:
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=0 UNKNOWN switching <NONE> ---> <ACTIVE>.
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:   Empty slot was found - initializing new stream with ID=5.
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]:   Preparing ZRTP Hello according to the Session profile.
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp main]: ATTACH NEW STREAM - DONE.
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: START STREAM ID=5 mode=CLEAR state=ACTIVE.
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=5 CLEAR switching <ACTIVE> ---> <START>.
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Send <HELLO> ssrc=559348913 seq=37145 size=144. Stream 5:CLEAR:START                                                                                                                          
2015-05-15 13:55:05.714923 [DEBUG] switch_channel.c:3635 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-05-15 13:55:05.714923 [NOTICE] sofia.c:7484 Channel [sofia/internal/1004] has been answered
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp mitm]: RESOLVE MITM CALL s1=5, s2=4...
2015-05-15 13:55:05.714923 [DEBUG] switch_rtp.c:1370  [   zrtp mitm]: RESOLVE MITM CALL s1=5, s2=4...
2015-05-15 13:55:05.714923 [DEBUG] switch_core_codec.c:246 sofia/internal/1001 at sip.domain.com Restore previous codec PCMA:8.
2015-05-15 13:55:05.714923 [DEBUG] switch_channel.c:3689 (sofia/internal/1004) Callstate Change RINGING -> ACTIVE
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:55:05.714923 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1001 at sip.domain.com:
v=0                                                                                                                                
o=FreeSWITCH 1431670383 1431670385 IN IP4 YYY.YYY.YYY.YYY                                                                           
s=FreeSWITCH                                                                                                                       
c=IN IP4 YYY.YYY.YYY.YYY                                                                                                            
t=0 0                                                                                                                              
m=audio 20514 RTP/AVP 8 101                                                                                                        
a=rtpmap:8 PCMA/8000                                                                                                               
a=rtpmap:101 telephone-event/8000                                                                                                  
a=fmtp:101 0-16                                                                                                                    
a=ptime:20                                                                                                                         
a=sendrecv                                                                                                                         
a=rtcp:20515 IN IP4 YYY.YYY.YYY.YYY                                                                                                 nua.c:879 nua_respond() nua: nua_respond: entering
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
nua_stack.c:529 nua_signal() nua(0x7fccf0091540): sent signal r_respond
soa.c:403 soa_set_params() soa_set_params(static::0x7fccf0091660, ...) called
2015-05-15 13:55:05.714923 [DEBUG] switch_core_session.c:912 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7fccf0091660, (nil), 0x7fcd3000f130, -1) called
nua_session.c:2320 nua_invite_server_respond() nua: nua_invite_server_respond: entering
2015-05-15 13:55:05.714923 [NOTICE] switch_ivr_originate.c:3519 Channel [sofia/internal/1001 at sip.domain.com] has been answered
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fccf0091660, [(nil)], [0x7fcd0bdfcc38], [0x7fcd0bdfcc34]) called
tport.c:3257 tport_tsend() tport_tsend(0x7fccf00a2eb0) tpn = TLS/XXX.XXX.XXX.XXX:15193
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0079900 0x7fccf00b5420 883 (883)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0079900 0x7fccf008e3e0 260 (260)
tport.c:3492 tport_send_msg() tport_vsend returned 1143
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
nta.c:6791 incoming_reply() nta: sent 200 OK for INVITE (28199)
nta.c:1348 set_timeout() nta: timer shortened to 500 ms
nua_session.c:4139 signal_call_state_change() nua(0x7fccf0091540): call state changed: early -> completed, sent answer
soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7fccf0091660, [0x7fcd0bdfcce8], [0x7fcd0bdfccf0], [(nil)]) called
soa.c:616 soa_get_params() soa_get_params(static::0x7fccf0091660, ...) called
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-15 13:55:05.714923 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-05-15 13:55:05.714923 [DEBUG] switch_channel.c:3689 (sofia/internal/1001 at sip.domain.com) Callstate Change EARLY -> ACTIVE
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:55:05.714923 [DEBUG] sofia.c:6623 Channel sofia/internal/1001 at sip.domain.com entering state [completed][200]
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:55:05.714923 [DEBUG] switch_ivr_originate.c:3577 Originate Resulted in Success: [sofia/internal/1004]
2015-05-15 13:55:05.714923 [DEBUG] switch_core_session.c:912 Send signal sofia/internal/1004 [BREAK]
2015-05-15 13:55:05.714923 [DEBUG] switch_core_session.c:912 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-05-15 13:55:05.714923 [DEBUG] switch_ivr_bridge.c:1465 (sofia/internal/1004) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA                                                                                                                                  
2015-05-15 13:55:05.714923 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-05-15 13:55:05.714923 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_EXCHANGE_MEDIA
2015-05-15 13:55:05.714923 [DEBUG] switch_core_state_machine.c:538 (sofia/internal/1004) State EXCHANGE_MEDIA
2015-05-15 13:55:05.714923 [DEBUG] mod_sofia.c:594 SOFIA EXCHANGE_MEDIA
tport.c:2773 tport_wakeup() tport_wakeup(0x7fccf00a2eb0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fccf00a2eb0)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fccf00a2eb0): tls_read() returned 414
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fccf00a2eb0) msg 0x7fccf0092fb0 from (tls/XXX.XXX.XXX.XXX:15193) has 414 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fccf00a2eb0): msg 0x7fccf0092fb0 (414 bytes) from tls/XXX.XXX.XXX.XXX:15193/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received BYE sip:1004 at YYY.YYY.YYY.YYY:9061;transport=tls SIP/2.0 (CSeq 28200)
nta.c:3174 agent_check_request_via() nta: Via check: received=XXX.XXX.XXX.XXX
nta.c:3060 agent_recv_request() nta: BYE (28200) going to existing leg
nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
2015-05-15 13:55:05.754963 [DEBUG] switch_core_session.c:1061 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-05-15 13:55:05.754963 [DEBUG] switch_rtp.c:1370  [  zrtp utils]:   Send <HELLO> ssrc=559348913 seq=37146 size=144. Stream 5:CLEAR:START                                                                                                                          
2015-05-15 13:55:05.794920 [DEBUG] switch_rtp.c:1370  [   zrtp mitm]: RESOLVE MITM CALL s1=5, s2=4...
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2015-05-15 13:55:05.794920 [NOTICE] sofia.c:952 Hangup sofia/internal/1001 at sip.domain.com [CS_EXECUTE] [NORMAL_CLEARING]
2015-05-15 13:55:05.794920 [DEBUG] switch_channel.c:3222 Send signal sofia/internal/1001 at sip.domain.com [KILL]
2015-05-15 13:55:05.794920 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
nua.c:879 nua_respond() nua: nua_respond: entering
nua_stack.c:529 nua_signal() nua(0x7fccf0091540): sent signal r_respond
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:529 nua_signal() nua(0x7fccf0091540): sent signal r_destroy
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fccf0091660, ...) called
tport.c:3257 tport_tsend() tport_tsend(0x7fccf00a2eb0) tpn = TLS/XXX.XXX.XXX.XXX:15193
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0079900 0x7fccf00b8f60 539 (539)
tport.c:3492 tport_send_msg() tport_vsend returned 539
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
nta.c:6791 incoming_reply() nta: sent 200 OK for BYE (28200)
nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7fccf0091540): removing session usage
nua_session.c:4139 signal_call_state_change() nua(0x7fccf0091540): call state changed: completed -> terminated
soa.c:356 soa_destroy() soa_destroy(static::0x7fccf0091660) called
nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7fccf0093530)
nta.c:5744 incoming_free() nta: incoming_free(0x7fccf00b6550)
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
2015-05-15 13:55:05.794920 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD DONE [sofia/internal/1001 at sip.domain.com]
2015-05-15 13:55:05.794920 [DEBUG] switch_ivr_bridge.c:690 Send signal sofia/internal/1004 [BREAK]
2015-05-15 13:55:05.814915 [DEBUG] switch_ivr_bridge.c:660 BRIDGE THREAD DONE [sofia/internal/1004]
2015-05-15 13:55:05.814915 [DEBUG] switch_ivr_bridge.c:690 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-05-15 13:55:05.814915 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/internal/1004 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2015-05-15 13:55:05.814915 [DEBUG] switch_channel.c:3222 Send signal sofia/internal/1004 [KILL]
2015-05-15 13:55:05.814915 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:538 (sofia/internal/1004) State EXCHANGE_MEDIA going to sleep
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_HANGUP
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/1004) Callstate Change ACTIVE -> HANGUP
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/1004) State HANGUP
2015-05-15 13:55:05.814915 [DEBUG] switch_ivr_bridge.c:1563 sofia/internal/1004 skip receive message [UNBRIDGE] (channel is comngup already)                                                                                                                           
2015-05-15 13:55:05.814915 [DEBUG] switch_ivr_bridge.c:1566 sofia/internal/1001 at sip.domain.com skip receive message [UNBRIDGE] (channel is comngup already)                                                                                                             
2015-05-15 13:55:05.814915 [DEBUG] mod_sofia.c:407 sofia/internal/1004 Overriding SIP cause 480 with 200 from the other leg
2015-05-15 13:55:05.814915 [DEBUG] mod_sofia.c:413 Channel sofia/internal/1004 hanging up, cause: NORMAL_CLEARING
2015-05-15 13:55:05.814915 [DEBUG] switch_core_session.c:2901 sofia/internal/1001 at sip.domain.com skip receive message [APPLICATION_EXEC_COMPLETE] (channel is comngup already)                                                                                          
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:535 (sofia/internal/1001 at sip.domain.com) State EXECUTE going to sleep
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_HANGUP                                                                                                                                 
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:735 (sofia/internal/1001 at sip.domain.com) Callstate Change ACTIVE -> HANGUP                                                                                                                              
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/1001 at sip.domain.com) State HANGUP
2015-05-15 13:55:05.814915 [DEBUG] mod_sofia.c:465 Sending BYE to sofia/internal/1004
2015-05-15 13:55:05.814915 [DEBUG] mod_sofia.c:413 Channel sofia/internal/1001 at sip.domain.com hanging up, cause: NORMAL_CLEARING
nua.c:645 nua_bye() nua: nua_bye: entering
nua_stack.c:529 nua_signal() nua(0x7fcccc00a070): sent signal r_bye
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1004 Standard HANGUP, cause: NORMAL_CLEARING
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/1004) State HANGUP going to sleep
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0x7fccf008b650, ...) called
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004) State Change CS_HANGUP -> CS_REPORTING
2015-05-15 13:55:05.814915 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
soa.c:1784 soa_terminate() soa_terminate(static::0x7fccf008b650) called
soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7fccf008b650) called
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
tport.c:4588 tport_by_name() tport(0x7fccf0004620): found 0x7fccf0077460 by name tls/XXX.XXX.XXX.XXX:15194
tport.c:3257 tport_tsend() tport_tsend(0x7fccf0077460) tpn = tls/XXX.XXX.XXX.XXX:15194
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1004) Running State Change CS_REPORTING
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0057610 0x7fccf008e9c0 657 (657)
tport.c:3492 tport_send_msg() tport_vsend returned 657
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf0077460): reset timer
nta.c:8304 outgoing_send() nta: sent BYE (75509385) to tls/XXX.XXX.XXX.XXX:15194
tport.c:4160 tport_pend() tport_pend(0x7fccf0077460): pending 0x7fccf008b8c0 for tls/XXX.XXX.XXX.XXX:15194 (already 0)
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/1004) State REPORTING
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1001 at sip.domain.com Standard HANGUP, cause: NORMAL_CLEARING                                                                                                                           
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/1001 at sip.domain.com) State HANGUP going to sleep
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1001 at sip.domain.com) State Change CS_HANGUP -> CS_REPORTING                                                                                                                         
2015-05-15 13:55:05.814915 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/1001 at sip.domain.com) Running State Change CS_REPORTING                                                                                                                              
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/1001 at sip.domain.com) State REPORTING
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1004 Standard REPORTING, cause: NORMAL_CLEARING
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/1004) State REPORTING going to sleep
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1004) State Change CS_REPORTING -> CS_DESTROY
2015-05-15 13:55:05.814915 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1004 [BREAK]
2015-05-15 13:55:05.814915 [DEBUG] switch_core_session.c:1623 Session 6 (sofia/internal/1004) Locked, Waiting on external entities
2015-05-15 13:55:05.814915 [NOTICE] switch_core_session.c:1641 Session 6 (sofia/internal/1004) Ended
2015-05-15 13:55:05.814915 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1004 [CS_DESTROY]
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/1004) Running State Change CS_DESTROY
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/1004) State DESTROY
2015-05-15 13:55:05.814915 [DEBUG] mod_sofia.c:323 sofia/internal/1004 SOFIA DESTROY
2015-05-15 13:55:05.814915 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: STOP STREAM ID=5 mode=CLEAR state=START.
2015-05-15 13:55:05.814915 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=0 UNKNOWN switching <NONE> ---> <NONE>.
2015-05-15 13:55:05.814915 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: STOP STREAM ID=0 mode=UNKNOWN state=NONE.
2015-05-15 13:55:05.814915 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: STOP STREAM ID=0 mode=UNKNOWN state=NONE.
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1004 Standard DESTROY
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/1004) State DESTROY going to sleep
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1001 at sip.domain.com Standard REPORTING, cause: NORMAL_CLEARING                                                                                                                       
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/1001 at sip.domain.com) State REPORTING going to sleep                                                                                                                                 
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/1001 at sip.domain.com) State Change CS_REPORTING -> CS_DESTROY                                                                                                                        
2015-05-15 13:55:05.814915 [DEBUG] switch_core_session.c:1396 Send signal sofia/internal/1001 at sip.domain.com [BREAK]
2015-05-15 13:55:05.814915 [DEBUG] switch_core_session.c:1623 Session 5 (sofia/internal/1001 at sip.domain.com) Locked, Waiting on external entities                                                                                                                      
2015-05-15 13:55:05.814915 [NOTICE] switch_core_session.c:1641 Session 5 (sofia/internal/1001 at sip.domain.com) Ended
2015-05-15 13:55:05.814915 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1001 at sip.domain.com [CS_DESTROY]
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/1001 at sip.domain.com) Running State Change CS_DESTROY                                                                                                                                
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/1001 at sip.domain.com) State DESTROY
2015-05-15 13:55:05.814915 [DEBUG] mod_sofia.c:323 sofia/internal/1001 at sip.domain.com SOFIA DESTROY
2015-05-15 13:55:05.814915 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: STOP STREAM ID=4 mode=DH state=SECURE.
2015-05-15 13:55:05.814915 [DEBUG] switch_rtp.c:1370  [        zrtp]:   Stream ID=0 UNKNOWN switching <NONE> ---> <NONE>.
2015-05-15 13:55:05.814915 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: STOP STREAM ID=0 mode=UNKNOWN state=NONE.
2015-05-15 13:55:05.814915 [DEBUG] switch_rtp.c:1370  [ zrtp engine]: STOP STREAM ID=0 mode=UNKNOWN state=NONE.
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1001 at sip.domain.com Standard DESTROY
2015-05-15 13:55:05.814915 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/1001 at sip.domain.com) State DESTROY going to sleep
nta.c:6996 _nta_incoming_timer() nta: timer G fired, retransmitting 200 reply
tport.c:3257 tport_tsend() tport_tsend(0x7fccf00a2eb0) tpn = TLS/XXX.XXX.XXX.XXX:15193
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0079900 0x7fccf00b5420 883 (883)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec 0x7fccf0079900 0x7fccf008e3e0 260 (260)
tport.c:3492 tport_send_msg() tport_vsend returned 1143
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free
nta.c:1296 agent_timer() nta: timer set next to 1000 ms
tport.c:2773 tport_wakeup() tport_wakeup(0x7fccf0077460): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fccf0077460)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fccf0077460): tls_read() returned 1
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fccf0077460) msg 0x7fccf00b6550 from (tls/XXX.XXX.XXX.XXX:15194) has 1 bytes, veclen = 1
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf0077460): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7fccf0077460): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fccf0077460)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fccf0077460): tls_read() returned 489
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fccf0077460) msg 0x7fccf00b6550 from (tls/XXX.XXX.XXX.XXX:15194) has 489 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fccf0077460): msg 0x7fccf00b6550 (490 bytes) from tls/XXX.XXX.XXX.XXX:15194/sips next=(nil)
nta.c:3299 agent_recv_response() nta: received 200 OK for BYE (75509385)
nta.c:3366 agent_recv_response() nta: 200 OK is going to a transaction
nta.c:9564 outgoing_estimate_delay() nta_outgoing: RTT is 505.558 ms
tport.c:4222 tport_release() tport_release(0x7fccf0077460): 0x7fccf008b8c0 by 0x7fccf00a1400 with 0x7fccf00b6550
nua_session.c:4139 signal_call_state_change() nua(0x7fcccc00a070): call state changed: terminating -> terminated
nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7fcccc00a070): removing session usage
soa.c:356 soa_destroy() soa_destroy(static::0x7fccf008b650) called
nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7fccf00b0bb0)
nua_session.c:351 nua_session_usage_destroy() nua: terminated session 0x7fcccc00a070
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nta.c:8722 outgoing_free() nta: outgoing_free(0x7fccf00a1400)
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf0077460): reset timer
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:529 nua_signal() nua(0x7fcccc00a070): sent signal r_destroy
nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil))
tport.c:2773 tport_wakeup() tport_wakeup(0x7fccf00a2eb0): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7fccf00a2eb0)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7fccf00a2eb0): tls_read() returned 374
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7fccf00a2eb0) msg 0x7fccf008b450 from (tls/XXX.XXX.XXX.XXX:15193) has 374 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7fccf00a2eb0): msg 0x7fccf008b450 (374 bytes) from tls/XXX.XXX.XXX.XXX:15193/sips next=(nil)
nta.c:2880 agent_recv_request() nta: received ACK sip:1004 at YYY.YYY.YYY.YYY:9061;transport=tls SIP/2.0 (CSeq 28199)
nta.c:3174 agent_check_request_via() nta: Via check: received=XXX.XXX.XXX.XXX
nta.c:3019 agent_recv_request() nta: ACK (28199) is going to INVITE (28199)
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
tport.c:2296 tport_set_secondary_timer() tport(0x7fccf00a2eb0): reset timer
nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7fcd0bdfcc60)
nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 0/0 term, 1/1 free
nta.c:1296 agent_timer() nta: timer set next to 30493 ms
Brian West <brian at freeswitch.org> ?rta:
>Its either you&#39;ve enabled proxy media mode, or auto proxy due to the zrtp-hash in the sdp&#39;s hard to tell 100% but I suspect thats what is taking place. 
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From krice at freeswitch.org  Fri May 15 16:59:45 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 15 May 2015 07:59:45 -0500
Subject: [Freeswitch-users] MP4 record in 1.6
In-Reply-To: <1852348436.20150515084822@seznam.cz>
Message-ID: <D17B57F1.9DD92%krice@freeswitch.org>

Don?t set that from the dialplan, trigger it with the conference recording
command, else you will record each conference participant and if you have 10
different people in the conf you?ll get 10 redundant recordings


On 5/15/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz> wrote:

> Hi All,
> 
> New FreeSWITCH 1.6 Video on Debian 8.
> 
> I try to record MP4 file in the conference..
> Simple dialplan
> 
>   <extension name="nb_conferences">
>       <condition field="destination_number" expression="^(30\d{2})$">
>         <action application="answer"/>
>         <action application="set" data="enable_file_write_buffering=false"/>
>         <action application="record"
> data="/usr/local/freeswitch/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.
> mp4"/>
>         <action application="conference" data="$1-${domain_name}@default"/>
>       </condition>
>     </extension>
> 
> It works. Even something writes. But I can not play this file (The file in the
> attachment). What am I doing wrong?

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From mietekszczesniak2503 at gmail.com  Fri May 15 17:15:46 2015
From: mietekszczesniak2503 at gmail.com (=?UTF-8?Q?Mietek_Sze=C5=9Bniak?=)
Date: Fri, 15 May 2015 15:15:46 +0200
Subject: [Freeswitch-users] will FS work on Debian 8?
Message-ID: <CADiUVxtzK9Zor8uDk1eA5C+bv0=cJf-EkQk_5BQe9x5Ffm3cJQ@mail.gmail.com>

The question really is: will it Just Work*?*, fully, out of the box, on
Debian 8?

Because I'm sure an experienced linux hacker can get it to work, but I'm
asking from a novice user point of view.

This relates to the changes introduced by Debian 8 - primarily systemd.

Thanks.
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From yadenis at seznam.cz  Fri May 15 17:18:58 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Fri, 15 May 2015 15:18:58 +0200
Subject: [Freeswitch-users] MP4 record in 1.6
In-Reply-To: <D17B57F1.9DD92%krice@freeswitch.org>
References: <1852348436.20150515084822@seznam.cz>
	<D17B57F1.9DD92%krice@freeswitch.org>
Message-ID: <197574130.20150515151858@seznam.cz>

Dobr? den,

Ok. And how to do it right?

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

p?tek 15. kv?tna 2015, 14:59:45, napsal jste:


Don?t set that from the dialplan, trigger it with the conference recording command, else you will record each conference participant and if you have 10 different people in the conf you?ll get 10 redundant recordings


On 5/15/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz> wrote:

Hi All,

New FreeSWITCH 1.6 Video on Debian 8.

I try to record MP4 file in the conference..  
Simple dialplan

  <extension name="nb_conferences">
      <condition field="destination_number" expression="^(30\d{2})$">
        <action application="answer"/>
        <action application="set" data="enable_file_write_buffering=false"/>
        <action application="record" data="/usr/local/freeswitch/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.mp4"/>
        <action application="conference" data="$1-${domain_name}@default"/>
      </condition>
    </extension>

It works. Even something writes. But I can not play this file (The file in the attachment). What am I doing wrong?

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH
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From krice at freeswitch.org  Fri May 15 17:54:25 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 15 May 2015 08:54:25 -0500
Subject: [Freeswitch-users] will FS work on Debian 8?
In-Reply-To: <CADiUVxtzK9Zor8uDk1eA5C+bv0=cJf-EkQk_5BQe9x5Ffm3cJQ@mail.gmail.com>
Message-ID: <D17B64C1.9DD99%krice@freeswitch.org>

Hows this for an answer, the core FreeSWITCH Dev Team primarily uses Debian
8


On 5/15/15, 8:15 AM, "Mietek Sze?niak" <mietekszczesniak2503 at gmail.com>
wrote:

> The question really is: will it Just Work(TM), fully, out of the box, on Debian
> 8?
> 
> Because I'm sure an experienced linux hacker can get it to work, but I'm
> asking from a novice user point of view.
> 
> This relates to the changes introduced by Debian 8 - primarily systemd.
> 
> 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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From krice at freeswitch.org  Fri May 15 18:01:15 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 15 May 2015 14:01:15 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <5555fc2b1e6cc_3c09f2f314163e6@resque-worker-high.3.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From tony at intelecenter.com  Fri May 15 18:17:26 2015
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Fri, 15 May 2015 07:17:26 -0700
Subject: [Freeswitch-users] Freeswitch mod_http_cache with Azure Blob Storage
Message-ID: <CAANhKXLF1oYHmjjC8A_j8Y+Ef83PfJRqOCtQ7F+p363D_HFkTg@mail.gmail.com>

Hello:


does anyone have a working example/success with using mod_http_cache with
Azure Blob Storage?

Thanks

-- 

Tony Bourdeaux

*Intelecenter, LLC*

Skype: tony.bourdeaux





"This message and any attachments are solely for the intended recipient and
may contain confidential or privileged information. If you are not the
intended recipient, any disclosure, copying, use, or distribution of the
information included in this message and any attachments is prohibited. If
you have received this communication in error, please notify me by reply
e-mail and immediately and permanently delete this message and any
attachments."
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From aqsyounas at gmail.com  Fri May 15 18:32:15 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 15 May 2015 07:32:15 -0700
Subject: [Freeswitch-users] Error during the installation of esl luamod
In-Reply-To: <FEB7FD32-A79C-4BFE-BE0F-2B92AD0C13D5@jerris.com>
References: <CAGVOjVVmMN7r1gKP-HLv+6JuRd4Z5-Fp0F4JvvZEBYHZQbzd_w@mail.gmail.com>
	<FEB7FD32-A79C-4BFE-BE0F-2B92AD0C13D5@jerris.com>
Message-ID: <CAGVOjVXo0zO7M9T2e20J0N+Pf=C56X-K-2P=k-_Pd4reMBgvUQ@mail.gmail.com>

It worked. Thanks

On 13 May 2015 at 12:22, Michael Jerris <mike at jerris.com> wrote:

> in master, try building with the --enable-system-lua configure flag.
> This will use pkg-config to find the cflags to get to the headers.  In 1.6
> this will be the behavior always.
>
>
>
> On May 13, 2015, at 2:44 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
> Hi users.
> During esl luamod installation I see this error.
>
>
>
>
>
>
>
> *esl_wrap.cpp:746:17: fatal error: lua.h: No such file or
> directory #include "lua.h"                 ^compilation terminated.make[1]:
> *** [esl_wrap.o] Error 1make[1]: Leaving directory
> `/usr/src/freeswitch/libs/esl/lua'make: *** [luamod] Error 2*
>
> But running '*find lua.h*' I see file is located at '
> */usr/include/lua5.2/lua.h*'
>
> I guess some of you guys had been gone through this.
>
> How did you resolve this?
> Any pointer would be much appreciated.
> Thanks.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From leonidnasedkin at gmail.com  Fri May 15 17:31:58 2015
From: leonidnasedkin at gmail.com (Leonid Nasedkin)
Date: Fri, 15 May 2015 16:31:58 +0300
Subject: [Freeswitch-users] XML_RPX connection limit
Message-ID: <08646886-1503-414A-AAB1-46B28B3DDBA8@gmail.com>

Hi, All!
Im trying to use freeswitch for automated outbound calls. 
Im creating outbound calls via xml_rpc interface with api call. But it seems like somewhere in freeswitch hardcoded connection limit to xml_rpc . I cant create more then 16 concurrent connections. 
How I can increase this value? 
Im using:
FreeSWITCH version: 1.4.15~64bit ( 64bit) 2.6.32-504.16.2.el6.x86_64
CentOS 6.5

Thanks.

From leonidnasedkin at gmail.com  Fri May 15 18:22:18 2015
From: leonidnasedkin at gmail.com (Leonid Nasedkin)
Date: Fri, 15 May 2015 17:22:18 +0300
Subject: [Freeswitch-users] XML_RPC connections limit
Message-ID: <B96DD723-6467-468D-9A8D-E28C0D020801@gmail.com>

Hi, All!
Im trying to use freeswitch for automated outbound calls. 
Im init calls via xml_rpc interface with api call. But it seems like somewhere in freeswitch hardcoded connection limit to xml_rpc . I cant create more then 16 concurrent connections. 
How I can increase this value? 
Im using:
FreeSWITCH version: 1.4.15~64bit ( 64bit) 
CentOS 6.5
2.6.32-504.16.2.el6.x86_64

Thanks.

From yadenis at seznam.cz  Fri May 15 19:12:43 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Fri, 15 May 2015 17:12:43 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.6 Video MP4 recording
Message-ID: <61489624.20150515171243@seznam.cz>

Hi All

Finally I managed to get record MP4. He writes beautifully with the module mod_vlc. But without sound.

[00007ff65c027408] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007ff65c027408] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007ff65c027408] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007ff65c027408] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007ff65c027408] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)

I checked. Indeed there is a module libvpx.so not in place. What do I need to do to this module come from?

Thank you

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From krice at freeswitch.org  Fri May 15 19:13:45 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 15 May 2015 10:13:45 -0500
Subject: [Freeswitch-users] ***NOTICE: FreeSWITCH File Distribution server
	slight path update.
Message-ID: <D17B7759.9DDA4%krice@freeswitch.org>

Greetings FreeSWITCHers!

Just a quick note, we?ve done a little spring cleaning on
http://files.freeswitch.org/

The root of the web server was getting a bit messy with all the old releases
so we?ve gone through and reorganized a little bit.

FreeSWITCH Source Tarballs have been to
http://files.freeswitch.org/releases/freeswitch and the sounds tarballs have
been moved to http://files.freeswitch.org/releases/sounds/

There is a 301 redirect in place so all your old links to the sources will
still work, but you?ll want to update your scripts to bypass the 301 and
look in the new locations.

Have a Great Day!
K
-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From mike at jerris.com  Fri May 15 19:18:50 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 May 2015 11:18:50 -0400
Subject: [Freeswitch-users] MP4 record in 1.6
In-Reply-To: <197574130.20150515151858@seznam.cz>
References: <1852348436.20150515084822@seznam.cz>
	<D17B57F1.9DD92%krice@freeswitch.org>
	<197574130.20150515151858@seznam.cz>
Message-ID: <0A405FEA-024F-4B62-A8D5-00111BC1D40C@jerris.com>

just like you do to record audio in a conference.  Check out the conference confluence page.

> On May 15, 2015, at 9:18 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> 
> Dobr? den,
> 
> Ok. And how to do it right?
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel. 775-415-382
> 
> p?tek 15. kv?tna 2015, 14:59:45, napsal jste:
> 
> 
> Don?t set that from the dialplan, trigger it with the conference recording command, else you will record each conference participant and if you have 10 different people in the conf you?ll get 10 redundant recordings
> 
> 
> On 5/15/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz <x-msg://51/yadenis at seznam.cz>> wrote:
> 
> Hi All,
> 
> New FreeSWITCH 1.6 Video on Debian 8.
> 
> I try to record MP4 file in the conference..  
> Simple dialplan
> 
>  <extension name="nb_conferences">
>      <condition field="destination_number" expression="^(30\d{2})$">
>        <action application="answer"/>
>        <action application="set" data="enable_file_write_buffering=false"/>
>        <action application="record" data="/usr/local/freeswitch/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.mp4"/>
>        <action application="conference" data="$1-${domain_name}@default"/>
>      </condition>
>    </extension>
> 
> It works. Even something writes. But I can not play this file (The file in the attachment). What am I doing wrong?

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From mike at jerris.com  Fri May 15 19:20:02 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 May 2015 11:20:02 -0400
Subject: [Freeswitch-users] will FS work on Debian 8?
In-Reply-To: <D17B64C1.9DD99%krice@freeswitch.org>
References: <D17B64C1.9DD99%krice@freeswitch.org>
Message-ID: <A4AC2FCB-895F-45B1-AEF4-8BBDA4DD8539@jerris.com>

Caveat... We are about to merge video code into master. If you want any hope of those features working, you will need to use our repo to get a number of required deps, including a few things that we override.

> On May 15, 2015, at 9:54 AM, Ken Rice <krice at freeswitch.org> wrote:
> 
> Hows this for an answer, the core FreeSWITCH Dev Team primarily uses Debian 8
> 
> 
> On 5/15/15, 8:15 AM, "Mietek Sze?niak" <mietekszczesniak2503 at gmail.com <x-msg://53/mietekszczesniak2503 at gmail.com>> wrote:
> 
>> The question really is: will it Just Work?, fully, out of the box, on Debian 8?
>> 
>> Because I'm sure an experienced linux hacker can get it to work, but I'm asking from a novice user point of view.
>> 
>> This relates to the changes introduced by Debian 8 - primarily systemd.
>> 
>> Thanks.

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From mike at jerris.com  Fri May 15 19:21:55 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 May 2015 11:21:55 -0400
Subject: [Freeswitch-users] FreeSWITCH 1.6 Video MP4 recording
In-Reply-To: <61489624.20150515171243@seznam.cz>
References: <61489624.20150515171243@seznam.cz>
Message-ID: <4F311309-FC56-48D4-8C49-B7BA7E42129D@jerris.com>

There is an open bug on this related to issues from our updated libvpx.  We are working to resolve this issue, hopefully something will go in today.

> On May 15, 2015, at 11:12 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> 
> Hi All
> 
> Finally I managed to get record MP4. He writes beautifully with the module mod_vlc. But without sound.
> 
> [00007ff65c027408] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007ff65c027408] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007ff65c027408] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007ff65c027408] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007ff65c027408] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
> 
> I checked. Indeed there is a module libvpx.so not in place. What do I need to do to this module come from?

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From anthony.minessale at gmail.com  Fri May 15 20:01:03 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 15 May 2015 11:01:03 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.6 Video MP4 recording
In-Reply-To: <4F311309-FC56-48D4-8C49-B7BA7E42129D@jerris.com>
References: <61489624.20150515171243@seznam.cz>
	<4F311309-FC56-48D4-8C49-B7BA7E42129D@jerris.com>
Message-ID: <CAKbxfG_MiA=8Jqa6GV6NBWwDSr7Mm6MNTMNhWhaGdLekva3_ew@mail.gmail.com>

you have no sound because you are alone in the conference.

set param min-required-recording-participants to 1 on your conference
profile.




On Fri, May 15, 2015 at 10:21 AM, Michael Jerris <mike at jerris.com> wrote:

> There is an open bug on this related to issues from our updated libvpx.
> We are working to resolve this issue, hopefully something will go in today.
>
> On May 15, 2015, at 11:12 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>  Hi All
>
> Finally I managed to get record MP4. He writes beautifully with the module
> mod_vlc. But without sound.
>
> [00007ff65c027408] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007ff65c027408] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007ff65c027408] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
> open shared object file: No such file or directory)
> [00007ff65c027408] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007ff65c027408] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
>
> I checked. Indeed there is a module libvpx.so not in place. What do I need
> to do to this module come from?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cmrienzo at gmail.com  Fri May 15 21:03:07 2015
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Fri, 15 May 2015 13:03:07 -0400
Subject: [Freeswitch-users] Freeswitch mod_http_cache with Azure Blob
	Storage
In-Reply-To: <CAANhKXLF1oYHmjjC8A_j8Y+Ef83PfJRqOCtQ7F+p363D_HFkTg@mail.gmail.com>
References: <CAANhKXLF1oYHmjjC8A_j8Y+Ef83PfJRqOCtQ7F+p363D_HFkTg@mail.gmail.com>
Message-ID: <CAD-qRkgJFNCDJ-n6YWhAh8xsEO_Ry6PRYCji7x0WASw6e3Cu4w@mail.gmail.com>

mod_http_cache does not currently support Azure storage services
authentication.  Reviewing the blob REST API, it looks similar to S3 (which
mod_http_cache support), so it could be added if somebody wants to
pay/contribute.

On Fri, May 15, 2015 at 10:17 AM, Tony Bourdeaux <tony at intelecenter.com>
wrote:

> Hello:
>
>
> does anyone have a working example/success with using mod_http_cache with
> Azure Blob Storage?
>
> Thanks
>
> --
>
> Tony Bourdeaux
>
> *Intelecenter, LLC*
>
> Skype: tony.bourdeaux
>
>
>
>
>
> "This message and any attachments are solely for the intended recipient
> and may contain confidential or privileged information. If you are not the
> intended recipient, any disclosure, copying, use, or distribution of the
> information included in this message and any attachments is prohibited. If
> you have received this communication in error, please notify me by reply
> e-mail and immediately and permanently delete this message and any
> attachments."
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at gmail.com  Fri May 15 21:41:13 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 15 May 2015 19:41:13 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.6 Video MP4 recording
In-Reply-To: <CAKbxfG_MiA=8Jqa6GV6NBWwDSr7Mm6MNTMNhWhaGdLekva3_ew@mail.gmail.com>
References: <61489624.20150515171243@seznam.cz>
	<4F311309-FC56-48D4-8C49-B7BA7E42129D@jerris.com>
	<CAKbxfG_MiA=8Jqa6GV6NBWwDSr7Mm6MNTMNhWhaGdLekva3_ew@mail.gmail.com>
Message-ID: <CACOYK=grt-LWA_J_VTSkZHp8S2uuF91j2C56coSFFQ=7trQOBg@mail.gmail.com>

Yes, talking to yourself is a typical thing for a voice engineer. And now -
with a picture!  :)
On May 15, 2015 6:01 PM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:

> you have no sound because you are alone in the conference.
>
> set param min-required-recording-participants to 1 on your conference
> profile.
>
>
>
>
> On Fri, May 15, 2015 at 10:21 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> There is an open bug on this related to issues from our updated libvpx.
>> We are working to resolve this issue, hopefully something will go in today.
>>
>> On May 15, 2015, at 11:12 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>>
>>  Hi All
>>
>> Finally I managed to get record MP4. He writes beautifully with the
>> module mod_vlc. But without sound.
>>
>> [00007ff65c027408] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
>> (libvpx.so.1: cannot open shared object file: No such file or directory)
>> [00007ff65c027408] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>> [00007ff65c027408] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
>> open shared object file: No such file or directory)
>> [00007ff65c027408] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>> [00007ff65c027408] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>>
>> I checked. Indeed there is a module libvpx.so not in place. What do I
>> need to do to this module come from?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tony at intelecenter.com  Fri May 15 21:49:41 2015
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Fri, 15 May 2015 10:49:41 -0700
Subject: [Freeswitch-users] Freeswitch mod_http_cache with Azure Blob
	Storage
In-Reply-To: <CAD-qRkgJFNCDJ-n6YWhAh8xsEO_Ry6PRYCji7x0WASw6e3Cu4w@mail.gmail.com>
References: <CAANhKXLF1oYHmjjC8A_j8Y+Ef83PfJRqOCtQ7F+p363D_HFkTg@mail.gmail.com>
	<CAD-qRkgJFNCDJ-n6YWhAh8xsEO_Ry6PRYCji7x0WASw6e3Cu4w@mail.gmail.com>
Message-ID: <CAANhKXLn_rJSF6XGKpAN1NkrP6=TUogGUgUfwkXS7MR-yyx-_Q@mail.gmail.com>

ok thank you for the reply.

On Fri, May 15, 2015 at 10:03 AM, Christopher Rienzo <cmrienzo at gmail.com>
wrote:

> mod_http_cache does not currently support Azure storage services
> authentication.  Reviewing the blob REST API, it looks similar to S3 (which
> mod_http_cache support), so it could be added if somebody wants to
> pay/contribute.
>
> On Fri, May 15, 2015 at 10:17 AM, Tony Bourdeaux <tony at intelecenter.com>
> wrote:
>
>> Hello:
>>
>>
>> does anyone have a working example/success with using mod_http_cache with
>> Azure Blob Storage?
>>
>> Thanks
>>
>> --
>>
>> Tony Bourdeaux
>>
>> *Intelecenter, LLC*
>>
>> Skype: tony.bourdeaux
>>
>>
>>
>>
>>
>> "This message and any attachments are solely for the intended recipient
>> and may contain confidential or privileged information. If you are not the
>> intended recipient, any disclosure, copying, use, or distribution of the
>> information included in this message and any attachments is prohibited. If
>> you have received this communication in error, please notify me by reply
>> e-mail and immediately and permanently delete this message and any
>> attachments."
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Tony Bourdeaux

*Intelecenter, LLC*

ph: 805-428-3031

Skype: tony.bourdeaux





"This message and any attachments are solely for the intended recipient and
may contain confidential or privileged information. If you are not the
intended recipient, any disclosure, copying, use, or distribution of the
information included in this message and any attachments is prohibited. If
you have received this communication in error, please notify me by reply
e-mail and immediately and permanently delete this message and any
attachments."
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From victor.medina at cibersys.com  Fri May 15 22:56:23 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Fri, 15 May 2015 14:26:23 -0430
Subject: [Freeswitch-users] no friday free for all this week?
Message-ID: <CACXsapMzHuGSfKfruAdPQUH-MawnX58PxizZ7j1pnxYj2Fr2NA@mail.gmail.com>

-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From Sharath.Kumar at meZocliq.com  Fri May 15 23:15:45 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Fri, 15 May 2015 19:15:45 +0000
Subject: [Freeswitch-users] Setting loglevel beyond 7 ?
Message-ID: <CY1PR0501MB10982A84F72B060A6AD0C30084C70@CY1PR0501MB1098.namprd05.prod.outlook.com>

All,

I noticed some of the logs in the code are using SWITCH_LOG_DEBUG1 =101  - SWITCH_LOG_DEBUG10 = 110. Is there a way to enable these logs from the fs_cli  ? I can recompile the ones I want with a lower value but just wanted to avoid that.

Thanks
Sharath
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From Sharath.Kumar at meZocliq.com  Fri May 15 23:18:24 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Fri, 15 May 2015 19:18:24 +0000
Subject: [Freeswitch-users] will FS work on Debian 8?
In-Reply-To: <D17B64C1.9DD99%krice@freeswitch.org>
References: <CADiUVxtzK9Zor8uDk1eA5C+bv0=cJf-EkQk_5BQe9x5Ffm3cJQ@mail.gmail.com>
	<D17B64C1.9DD99%krice@freeswitch.org>
Message-ID: <CY1PR0501MB1098CB95C09F2A16D5D9743F84C70@CY1PR0501MB1098.namprd05.prod.outlook.com>

A few weeks ago, I got it working on Deb 8 without any hacks. I was doing it for the first time and the instructions were super easy to follow. I compiled from source using the Makefile provided by the FS team. Go for it!

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Friday, May 15, 2015 9:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] will FS work on Debian 8?

Hows this for an answer, the core FreeSWITCH Dev Team primarily uses Debian 8


On 5/15/15, 8:15 AM, "Mietek Sze?niak" <mietekszczesniak2503 at gmail.com> wrote:
The question really is: will it Just Work(tm), fully, out of the box, on Debian 8?

Because I'm sure an experienced linux hacker can get it to work, but I'm asking from a novice user point of view.

This relates to the changes introduced by Debian 8 - primarily systemd.

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

--
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH
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From krice at freeswitch.org  Fri May 15 23:31:10 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 15 May 2015 14:31:10 -0500
Subject: [Freeswitch-users] no friday free for all this week?
In-Reply-To: <CACXsapMzHuGSfKfruAdPQUH-MawnX58PxizZ7j1pnxYj2Fr2NA@mail.gmail.com>
Message-ID: <D17BB3AE.9DDC2%krice@freeswitch.org>

Theres people popping in and out all they time, hang out for a few and see
what happens

K


On 5/15/15, 1:56 PM, "Victor Medina" <victor.medina at cibersys.com> wrote:

> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From tony at intelecenter.com  Sat May 16 01:04:04 2015
From: tony at intelecenter.com (Tony Bourdeaux)
Date: Fri, 15 May 2015 14:04:04 -0700
Subject: [Freeswitch-users] Freeswitch mod_http_cache with Azure Blob
	Storage
In-Reply-To: <CAD-qRkgJFNCDJ-n6YWhAh8xsEO_Ry6PRYCji7x0WASw6e3Cu4w@mail.gmail.com>
References: <CAANhKXLF1oYHmjjC8A_j8Y+Ef83PfJRqOCtQ7F+p363D_HFkTg@mail.gmail.com>
	<CAD-qRkgJFNCDJ-n6YWhAh8xsEO_Ry6PRYCji7x0WASw6e3Cu4w@mail.gmail.com>
Message-ID: <CAANhKXJxHkqNwG=FhbpuuJ1E3ivFsKVDb=cDHwzLxEDugm-TYA@mail.gmail.com>

What do you think something like this would cost?

On Fri, May 15, 2015 at 10:03 AM, Christopher Rienzo <cmrienzo at gmail.com>
wrote:

> mod_http_cache does not currently support Azure storage services
> authentication.  Reviewing the blob REST API, it looks similar to S3 (which
> mod_http_cache support), so it could be added if somebody wants to
> pay/contribute.
>
> On Fri, May 15, 2015 at 10:17 AM, Tony Bourdeaux <tony at intelecenter.com>
> wrote:
>
>> Hello:
>>
>>
>> does anyone have a working example/success with using mod_http_cache with
>> Azure Blob Storage?
>>
>> Thanks
>>
>> --
>>
>> Tony Bourdeaux
>>
>> *Intelecenter, LLC*
>>
>> Skype: tony.bourdeaux
>>
>>
>>
>>
>>
>> "This message and any attachments are solely for the intended recipient
>> and may contain confidential or privileged information. If you are not the
>> intended recipient, any disclosure, copying, use, or distribution of the
>> information included in this message and any attachments is prohibited. If
>> you have received this communication in error, please notify me by reply
>> e-mail and immediately and permanently delete this message and any
>> attachments."
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Tony Bourdeaux

*Intelecenter, LLC*

ph: 805-428-3031

Skype: tony.bourdeaux





"This message and any attachments are solely for the intended recipient and
may contain confidential or privileged information. If you are not the
intended recipient, any disclosure, copying, use, or distribution of the
information included in this message and any attachments is prohibited. If
you have received this communication in error, please notify me by reply
e-mail and immediately and permanently delete this message and any
attachments."
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From voransoy at gmail.com  Sat May 16 14:08:51 2015
From: voransoy at gmail.com (Volkan Oransoy)
Date: Sat, 16 May 2015 13:08:51 +0300
Subject: [Freeswitch-users] not make a phone call both extensions
In-Reply-To: <CAPwL2MDwW_LEWziqsQE5nX2+bw74+kmCWCKmT2S2Ve7A3R4wpg@mail.gmail.com>
References: <CAPwL2MDwW_LEWziqsQE5nX2+bw74+kmCWCKmT2S2Ve7A3R4wpg@mail.gmail.com>
Message-ID: <CAM_EGOERev5AaAG1PYoZKMNm8DmTfGWA-oz=hqWVz5SrCS4fdA@mail.gmail.com>

Freeswitch is very flexible, so you can set many sip profiles for many
different scenarios.
The starting point of a context is a sip profile. When a call hits the
dialplan, the relevant sip profile sets the initial context.
You  should check your sip profile settings if you want to change the
initial context.

/Volkan

2015-05-14 20:41 GMT+03:00 Masakazu Nakano <emplant2000 at gmail.com>:

> Hi there
>
> I'm Freeswitch newbie. 15 years ago, usually used asterisk as old
> extension.conf style.
>
> so I got a cloud server like a EC2.
>
> I made a couple of users 5630 and 5631.xml and so on, specified default in
> context and in /conf/directory/default.
>
> and registered in np.
>
> freeswitch at internal> show registrations
>
> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata
> 5631,my
> cloud,76122ZWRhNDRmMDY0MzA2YWU5NzJkNWIxZWJiNmZmNTJmMjI,sofia/external/
> sip:5631 at 27.140.246.3:58996,1431626535,27.140.246.3,58996,udp,my cloud,
> 5630,my cloud,958199470722 at 10.210.59.150
> ,sofia/external/sip:5630 at 10.210.59.150:47420;transport=udp,1431626875,49.97.4.168,47420,udp,my
> cloud,
>
> 2 total.
>
> but fs_cli says 2015-05-15 02:31:47.752779 [INFO] mod_dialplan_xml.c:635
> Processing 5631 <5631>->5630 in context public
>
> why ??
>
> BR
>
> mack
> https://www.facebook.com/nakano.masakazu
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anuragrana31189 at gmail.com  Sat May 16 14:21:16 2015
From: anuragrana31189 at gmail.com (Anurag Rana)
Date: Sat, 16 May 2015 15:51:16 +0530
Subject: [Freeswitch-users] UnstisfiedLinkError
Message-ID: <CAFjUrj4OkkQT39MJvYv9RneYSeqpmU1PwGYW9vXaiM4z8qghEQ@mail.gmail.com>

Hi ,

I am getting this error when trying to run the java code

---error start------
java.lang.UnsatisfiedLinkError:
org.freeswitch.swig.freeswitchJNI.new_JavaSession__SWIG_0()J
    at org.freeswitch.swig.freeswitchJNI.new_JavaSession__SWIG_0(Native
Method)
    at org.freeswitch.swig.JavaSession.<init>(JavaSession.java:37)
    at Ping_MakeCall.call(Ping_MakeCall.java:68)
    at Ping_MakeCall.run(Ping_MakeCall.java:61)
    at java.lang.Thread.run(Thread.java:744)
----error ends ------

---
?Java code line where I am getting the error----
  JavaSession sessionAPI = new JavaSession();
 org.freeswitch.swig.API a = new API(sessionAPI);
------
I have already included freeswitch.jar file in path.

?
?What could be the possible reason for this error.?


Anurag Rana <https://sites.google.com/site/homepagerana/>
M.Tech CSE, IIIT-Delhi,
newbie42.com
https://sites.google.com/site/homepagerana/
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From emplant2000 at gmail.com  Sat May 16 14:27:46 2015
From: emplant2000 at gmail.com (Masakazu Nakano)
Date: Sat, 16 May 2015 19:27:46 +0900
Subject: [Freeswitch-users] not make a phone call both extensions
In-Reply-To: <CAM_EGOERev5AaAG1PYoZKMNm8DmTfGWA-oz=hqWVz5SrCS4fdA@mail.gmail.com>
References: <CAPwL2MDwW_LEWziqsQE5nX2+bw74+kmCWCKmT2S2Ve7A3R4wpg@mail.gmail.com>
	<CAM_EGOERev5AaAG1PYoZKMNm8DmTfGWA-oz=hqWVz5SrCS4fdA@mail.gmail.com>
Message-ID: <CAPwL2MD5TotyjKPVcwC14mJStS--7VpH0PNuesxjrVHAiWJc1w@mail.gmail.com>

Hi Volkan,

thank you for your reply.

I think I forgot setting up dialplan, and make it.but result is under like
that.

2015-05-16 19:15:57.652782 [INFO] mod_dialplan_xml.c:635 Processing 5631
<5631>->5630 in context public
Dialplan: sofia/external/5631 at XXX parsing [public->unloop] continue=false
Dialplan: sofia/external/5631 at XXX Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/5631 at XXX Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/5631 at XXX parsing [public->outside_call]
continue=true
Dialplan: sofia/external/5631 at XXX Absolute Condition [outside_call]
Dialplan: sofia/external/5631 at XXX Action set(outside_call=true)
Dialplan: sofia/external/5631 at XXX Action export(RFC2822_DATE=${strftime(%a,
%d %b %Y %T %z)})
Dialplan: sofia/external/5631 at XXX parsing [public->public_extensions]
continue=false
Dialplan: sofia/external/5631 at XXX Regex (PASS) [public_extensions]
destination_number(5630) =~ /^(563[0-9])$/ break=on-false
Dialplan: sofia/external/5631 at XXX Action
bridge(sofia/doublenat/5630%27.XX.104.17 )
2015-05-16 19:15:57.652782 [DEBUG] switch_core_state_machine.c:216
(sofia/external/5631 at XXX) State Change CS_ROUTING -> CS_EXECUTE

Can I make a phone call by fs_cli to 5631 ??

PS: I'm understand old style extension.conf is enought. but xml one is bit
difficult... :(

BR

mack



2015-05-16 19:08 GMT+09:00 Volkan Oransoy <voransoy at gmail.com>:

> Freeswitch is very flexible, so you can set many sip profiles for many
> different scenarios.
> The starting point of a context is a sip profile. When a call hits the
> dialplan, the relevant sip profile sets the initial context.
> You  should check your sip profile settings if you want to change the
> initial context.
>
> /Volkan
>
> 2015-05-14 20:41 GMT+03:00 Masakazu Nakano <emplant2000 at gmail.com>:
>
>> Hi there
>>
>> I'm Freeswitch newbie. 15 years ago, usually used asterisk as old
>> extension.conf style.
>>
>> so I got a cloud server like a EC2.
>>
>> I made a couple of users 5630 and 5631.xml and so on, specified default
>> in context and in /conf/directory/default.
>>
>> and registered in np.
>>
>> freeswitch at internal> show registrations
>>
>> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata
>> 5631,my
>> cloud,76122ZWRhNDRmMDY0MzA2YWU5NzJkNWIxZWJiNmZmNTJmMjI,sofia/external/
>> sip:5631 at 27.140.246.3:58996,1431626535,27.140.246.3,58996,udp,my cloud,
>> 5630,my cloud,958199470722 at 10.210.59.150
>> ,sofia/external/sip:5630 at 10.210.59.150:47420;transport=udp,1431626875,49.97.4.168,47420,udp,my
>> cloud,
>>
>> 2 total.
>>
>> but fs_cli says 2015-05-15 02:31:47.752779 [INFO] mod_dialplan_xml.c:635
>> Processing 5631 <5631>->5630 in context public
>>
>> why ??
>>
>> BR
>>
>> mack
>> https://www.facebook.com/nakano.masakazu
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at gmail.com  Sat May 16 14:37:09 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 16 May 2015 12:37:09 +0200
Subject: [Freeswitch-users] not make a phone call both extensions
In-Reply-To: <CAPwL2MD5TotyjKPVcwC14mJStS--7VpH0PNuesxjrVHAiWJc1w@mail.gmail.com>
References: <CAPwL2MDwW_LEWziqsQE5nX2+bw74+kmCWCKmT2S2Ve7A3R4wpg@mail.gmail.com>
	<CAM_EGOERev5AaAG1PYoZKMNm8DmTfGWA-oz=hqWVz5SrCS4fdA@mail.gmail.com>
	<CAPwL2MD5TotyjKPVcwC14mJStS--7VpH0PNuesxjrVHAiWJc1w@mail.gmail.com>
Message-ID: <CACOYK=jVLK2NMyPgz9xD_HfxwyfkXAu--CnffRN7xUfV6VsyEA@mail.gmail.com>

Hopefully this will help you as a starting point:
https://github.com/voxserv/freeswitch_conf_minimal/blob/master/docs/tutorial_01_simple_pbx.md
On May 16, 2015 12:28 PM, "Masakazu Nakano" <emplant2000 at gmail.com> wrote:

> Hi Volkan,
>
> thank you for your reply.
>
> I think I forgot setting up dialplan, and make it.but result is under like
> that.
>
> 2015-05-16 19:15:57.652782 [INFO] mod_dialplan_xml.c:635 Processing 5631
> <5631>->5630 in context public
> Dialplan: sofia/external/5631 at XXX parsing [public->unloop] continue=false
> Dialplan: sofia/external/5631 at XXX Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/external/5631 at XXX Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/external/5631 at XXX parsing [public->outside_call]
> continue=true
> Dialplan: sofia/external/5631 at XXX Absolute Condition [outside_call]
> Dialplan: sofia/external/5631 at XXX Action set(outside_call=true)
> Dialplan: sofia/external/5631 at XXX Action
> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
> Dialplan: sofia/external/5631 at XXX parsing [public->public_extensions]
> continue=false
> Dialplan: sofia/external/5631 at XXX Regex (PASS) [public_extensions]
> destination_number(5630) =~ /^(563[0-9])$/ break=on-false
> Dialplan: sofia/external/5631 at XXX Action
> bridge(sofia/doublenat/5630%27.XX.104.17 )
> 2015-05-16 19:15:57.652782 [DEBUG] switch_core_state_machine.c:216
> (sofia/external/5631 at XXX) State Change CS_ROUTING -> CS_EXECUTE
>
> Can I make a phone call by fs_cli to 5631 ??
>
> PS: I'm understand old style extension.conf is enought. but xml one is bit
> difficult... :(
>
> BR
>
> mack
>
>
>
> 2015-05-16 19:08 GMT+09:00 Volkan Oransoy <voransoy at gmail.com>:
>
>> Freeswitch is very flexible, so you can set many sip profiles for many
>> different scenarios.
>> The starting point of a context is a sip profile. When a call hits the
>> dialplan, the relevant sip profile sets the initial context.
>> You  should check your sip profile settings if you want to change the
>> initial context.
>>
>> /Volkan
>>
>> 2015-05-14 20:41 GMT+03:00 Masakazu Nakano <emplant2000 at gmail.com>:
>>
>>> Hi there
>>>
>>> I'm Freeswitch newbie. 15 years ago, usually used asterisk as old
>>> extension.conf style.
>>>
>>> so I got a cloud server like a EC2.
>>>
>>> I made a couple of users 5630 and 5631.xml and so on, specified default
>>> in context and in /conf/directory/default.
>>>
>>> and registered in np.
>>>
>>> freeswitch at internal> show registrations
>>>
>>> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata
>>> 5631,my
>>> cloud,76122ZWRhNDRmMDY0MzA2YWU5NzJkNWIxZWJiNmZmNTJmMjI,sofia/external/
>>> sip:5631 at 27.140.246.3:58996,1431626535,27.140.246.3,58996,udp,my cloud,
>>> 5630,my cloud,958199470722 at 10.210.59.150
>>> ,sofia/external/sip:5630 at 10.210.59.150:47420;transport=udp,1431626875,49.97.4.168,47420,udp,my
>>> cloud,
>>>
>>> 2 total.
>>>
>>> but fs_cli says 2015-05-15 02:31:47.752779 [INFO] mod_dialplan_xml.c:635
>>> Processing 5631 <5631>->5630 in context public
>>>
>>> why ??
>>>
>>> BR
>>>
>>> mack
>>> https://www.facebook.com/nakano.masakazu
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mitchelle.bit at gmail.com  Sat May 16 09:55:14 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Sat, 16 May 2015 11:25:14 +0530
Subject: [Freeswitch-users] XML parsing error
Message-ID: <CANBm6Syqy-f24AH97-xYefPP89BKM0dNkTzjpS_ZN9jC_4kM5w@mail.gmail.com>

Hi,
When I am using the web server to handle xml CDR's the xml file which it
sends gives an error.
The error being:

XML Parsing Error: not well-formed
Location: http://www.w3schools.com/xml/xml_validator.asp
Line Number 138, Column 25:
    <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
------------------------^

Please help me resolve this.


Thanks
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From mitchelle.bit at gmail.com  Sat May 16 11:44:02 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Sat, 16 May 2015 13:14:02 +0530
Subject: [Freeswitch-users] [XML parsing error]
Message-ID: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>

Hi,
When I am using the web server to handle xml CDR's the xml file which it
sends gives an error.
The error being:

XML Parsing Error: not well-formed
Location: http://www.w3schools.com/xml/xml_validator.asp
Line Number 138, Column 25:
    <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
------------------------^

Please help me resolve this.


Thanks,
Mitchelle
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From mitchelle.bit at gmail.com  Sat May 16 12:00:29 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Sat, 16 May 2015 13:30:29 +0530
Subject: [Freeswitch-users] XML parsing error
In-Reply-To: <CANBm6Syqy-f24AH97-xYefPP89BKM0dNkTzjpS_ZN9jC_4kM5w@mail.gmail.com>
References: <CANBm6Syqy-f24AH97-xYefPP89BKM0dNkTzjpS_ZN9jC_4kM5w@mail.gmail.com>
Message-ID: <CANBm6Szr97Y8-fjL3py6anQVT0ga8G82H6W_k7r-JNuVdsx-Xw@mail.gmail.com>

Hi,
Please find more details on http://pastebin.com/y5Gx3gPF .

Thanks,
Mitchelle

On Sat, May 16, 2015 at 11:25 AM, Mitchelle Johnson <mitchelle.bit at gmail.com
> wrote:

> Hi,
> When I am using the web server to handle xml CDR's the xml file which it
> sends gives an error.
> The error being:
>
> XML Parsing Error: not well-formed
> Location: http://www.w3schools.com/xml/xml_validator.asp
> Line Number 138, Column 25:
>     <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
> ------------------------^
>
> Please help me resolve this.
>
>
> Thanks
>
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From s.safarov at gmail.com  Sat May 16 16:20:13 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 16 May 2015 15:20:13 +0300
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
Message-ID: <CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>

See https://freeswitch.org/jira/browse/FS-7258

On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <mitchelle.bit at gmail.com
> wrote:

> Hi,
> When I am using the web server to handle xml CDR's the xml file which it
> sends gives an error.
> The error being:
>
> XML Parsing Error: not well-formed
> Location: http://www.w3schools.com/xml/xml_validator.asp
> Line Number 138, Column 25:
>     <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
> ------------------------^
>
> Please help me resolve this.
>
>
> Thanks,
> Mitchelle
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From iqbal.b.abdullah at gmail.com  Sat May 16 18:30:24 2015
From: iqbal.b.abdullah at gmail.com (Iqbal Abdullah)
Date: Sat, 16 May 2015 23:30:24 +0900
Subject: [Freeswitch-users] ESL python library for python3
Message-ID: <CAB6AXdnyX9Zsu73i+C7qtx=OSzkqo4fpTMrK0LiSaAD_8VX7Vg@mail.gmail.com>

Hello,

I've managed to compile, install and use the python esl library from
freeswitch 1.4.18, but am now wondering is there a way to get the
python esl library for python3?

I've tried changing the Makefiles in libs/esl/python to point to
python3 and made sure my dev packages are installed, but the
compilation failed.

Does the python esl module supports python3?


From voransoy at gmail.com  Sun May 17 00:29:29 2015
From: voransoy at gmail.com (Volkan Oransoy)
Date: Sat, 16 May 2015 23:29:29 +0300
Subject: [Freeswitch-users] not make a phone call both extensions
In-Reply-To: <CACOYK=jVLK2NMyPgz9xD_HfxwyfkXAu--CnffRN7xUfV6VsyEA@mail.gmail.com>
References: <CAPwL2MDwW_LEWziqsQE5nX2+bw74+kmCWCKmT2S2Ve7A3R4wpg@mail.gmail.com>
	<CAM_EGOERev5AaAG1PYoZKMNm8DmTfGWA-oz=hqWVz5SrCS4fdA@mail.gmail.com>
	<CAPwL2MD5TotyjKPVcwC14mJStS--7VpH0PNuesxjrVHAiWJc1w@mail.gmail.com>
	<CACOYK=jVLK2NMyPgz9xD_HfxwyfkXAu--CnffRN7xUfV6VsyEA@mail.gmail.com>
Message-ID: <CAM_EGOGD1Ehu-Mw2GwvO14Ld-jZW+bt2KcYESPLvPzqcWzQ1Gg@mail.gmail.com>

To start a call on freeswitch console, please check this link.
https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate

2015-05-16 13:37 GMT+03:00 Stanislav Sinyagin <ssinyagin at gmail.com>:

> Hopefully this will help you as a starting point:
> https://github.com/voxserv/freeswitch_conf_minimal/blob/master/docs/tutorial_01_simple_pbx.md
> On May 16, 2015 12:28 PM, "Masakazu Nakano" <emplant2000 at gmail.com> wrote:
>
>> Hi Volkan,
>>
>> thank you for your reply.
>>
>> I think I forgot setting up dialplan, and make it.but result is under
>> like that.
>>
>> 2015-05-16 19:15:57.652782 [INFO] mod_dialplan_xml.c:635 Processing 5631
>> <5631>->5630 in context public
>> Dialplan: sofia/external/5631 at XXX parsing [public->unloop] continue=false
>> Dialplan: sofia/external/5631 at XXX Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/external/5631 at XXX Regex (FAIL) [unloop]
>> ${sip_looped_call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/external/5631 at XXX parsing [public->outside_call]
>> continue=true
>> Dialplan: sofia/external/5631 at XXX Absolute Condition [outside_call]
>> Dialplan: sofia/external/5631 at XXX Action set(outside_call=true)
>> Dialplan: sofia/external/5631 at XXX Action
>> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>> Dialplan: sofia/external/5631 at XXX parsing [public->public_extensions]
>> continue=false
>> Dialplan: sofia/external/5631 at XXX Regex (PASS) [public_extensions]
>> destination_number(5630) =~ /^(563[0-9])$/ break=on-false
>> Dialplan: sofia/external/5631 at XXX Action
>> bridge(sofia/doublenat/5630%27.XX.104.17 )
>> 2015-05-16 19:15:57.652782 [DEBUG] switch_core_state_machine.c:216
>> (sofia/external/5631 at XXX) State Change CS_ROUTING -> CS_EXECUTE
>>
>> Can I make a phone call by fs_cli to 5631 ??
>>
>> PS: I'm understand old style extension.conf is enought. but xml one is
>> bit difficult... :(
>>
>> BR
>>
>> mack
>>
>>
>>
>> 2015-05-16 19:08 GMT+09:00 Volkan Oransoy <voransoy at gmail.com>:
>>
>>> Freeswitch is very flexible, so you can set many sip profiles for many
>>> different scenarios.
>>> The starting point of a context is a sip profile. When a call hits the
>>> dialplan, the relevant sip profile sets the initial context.
>>> You  should check your sip profile settings if you want to change the
>>> initial context.
>>>
>>> /Volkan
>>>
>>> 2015-05-14 20:41 GMT+03:00 Masakazu Nakano <emplant2000 at gmail.com>:
>>>
>>>> Hi there
>>>>
>>>> I'm Freeswitch newbie. 15 years ago, usually used asterisk as old
>>>> extension.conf style.
>>>>
>>>> so I got a cloud server like a EC2.
>>>>
>>>> I made a couple of users 5630 and 5631.xml and so on, specified default
>>>> in context and in /conf/directory/default.
>>>>
>>>> and registered in np.
>>>>
>>>> freeswitch at internal> show registrations
>>>>
>>>> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata
>>>> 5631,my
>>>> cloud,76122ZWRhNDRmMDY0MzA2YWU5NzJkNWIxZWJiNmZmNTJmMjI,sofia/external/
>>>> sip:5631 at 27.140.246.3:58996,1431626535,27.140.246.3,58996,udp,my cloud,
>>>> 5630,my cloud,958199470722 at 10.210.59.150
>>>> ,sofia/external/sip:5630 at 10.210.59.150:47420;transport=udp,1431626875,49.97.4.168,47420,udp,my
>>>> cloud,
>>>>
>>>> 2 total.
>>>>
>>>> but fs_cli says 2015-05-15 02:31:47.752779 [INFO]
>>>> mod_dialplan_xml.c:635 Processing 5631 <5631>->5630 in context public
>>>>
>>>> why ??
>>>>
>>>> BR
>>>>
>>>> mack
>>>> https://www.facebook.com/nakano.masakazu
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From juanito1982 at gmail.com  Sun May 17 01:05:30 2015
From: juanito1982 at gmail.com (=?UTF-8?Q?Juan_Antonio_Iba=C3=B1ez_Santorum?=)
Date: Sat, 16 May 2015 23:05:30 +0200
Subject: [Freeswitch-users] No audio on 180 Ringing receipt
Message-ID: <CAHLYDCvD4fuRM_zCCTE1K5M2EJNhkvyyAvUH++sVfyktM3kNPw@mail.gmail.com>

Hello,

  I am doing some test in a callback scenario. We make a call from the
server to the caller through A provider which is answered. Then we make a
sencod call to the calle through B provider. When callee ring is ready, B
provider send to our server a 180 Ringing but the server does not start
generate that ringing tone to the caller.

You can see full flow here:

http://es.tinypic.com/r/vsj1w2/8

Is there any way to force FS to generate ringing for A leg when a 180 gets
received from B leg?

Regards
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From s.safarov at gmail.com  Sun May 17 02:07:46 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 17 May 2015 01:07:46 +0300
Subject: [Freeswitch-users] No audio on 180 Ringing receipt
In-Reply-To: <CAHLYDCvD4fuRM_zCCTE1K5M2EJNhkvyyAvUH++sVfyktM3kNPw@mail.gmail.com>
References: <CAHLYDCvD4fuRM_zCCTE1K5M2EJNhkvyyAvUH++sVfyktM3kNPw@mail.gmail.com>
Message-ID: <CAHtxdDf4y=SPM-aYM4PebvB9kY3A0Xvm7PFa7rexzT+mbG_MQg@mail.gmail.com>

Are is RTP media is proxied by FS?

On Sun, May 17, 2015 at 12:05 AM, Juan Antonio Iba?ez Santorum <
juanito1982 at gmail.com> wrote:

> Hello,
>
>   I am doing some test in a callback scenario. We make a call from the
> server to the caller through A provider which is answered. Then we make a
> sencod call to the calle through B provider. When callee ring is ready, B
> provider send to our server a 180 Ringing but the server does not start
> generate that ringing tone to the caller.
>
> You can see full flow here:
>
> http://es.tinypic.com/r/vsj1w2/8
>
> Is there any way to force FS to generate ringing for A leg when a 180 gets
> received from B leg?
>
> 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 juanito1982 at gmail.com  Sun May 17 02:53:29 2015
From: juanito1982 at gmail.com (=?UTF-8?Q?Juan_Antonio_Iba=C3=B1ez_Santorum?=)
Date: Sun, 17 May 2015 00:53:29 +0200
Subject: [Freeswitch-users] No audio on 180 Ringing receipt
In-Reply-To: <CAHtxdDf4y=SPM-aYM4PebvB9kY3A0Xvm7PFa7rexzT+mbG_MQg@mail.gmail.com>
References: <CAHLYDCvD4fuRM_zCCTE1K5M2EJNhkvyyAvUH++sVfyktM3kNPw@mail.gmail.com>
	<CAHtxdDf4y=SPM-aYM4PebvB9kY3A0Xvm7PFa7rexzT+mbG_MQg@mail.gmail.com>
Message-ID: <CAHLYDCsbXpxN=zPBiNOM68nGP-e27m362+-TkJmf0uWDzfyPAw@mail.gmail.com>

Yes, RTP goes through FS
El 17/05/2015 00:10, "Sergey Safarov" <s.safarov at gmail.com> escribi?:

> Are is RTP media is proxied by FS?
>
> On Sun, May 17, 2015 at 12:05 AM, Juan Antonio Iba?ez Santorum <
> juanito1982 at gmail.com> wrote:
>
>> Hello,
>>
>>   I am doing some test in a callback scenario. We make a call from the
>> server to the caller through A provider which is answered. Then we make a
>> sencod call to the calle through B provider. When callee ring is ready, B
>> provider send to our server a 180 Ringing but the server does not start
>> generate that ringing tone to the caller.
>>
>> You can see full flow here:
>>
>> http://es.tinypic.com/r/vsj1w2/8
>>
>> Is there any way to force FS to generate ringing for A leg when a 180
>> gets received from B leg?
>>
>> Regards
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at gmail.com  Sun May 17 09:49:19 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 17 May 2015 07:49:19 +0200
Subject: [Freeswitch-users] No audio on 180 Ringing receipt
In-Reply-To: <CAHLYDCsbXpxN=zPBiNOM68nGP-e27m362+-TkJmf0uWDzfyPAw@mail.gmail.com>
References: <CAHLYDCvD4fuRM_zCCTE1K5M2EJNhkvyyAvUH++sVfyktM3kNPw@mail.gmail.com>
	<CAHtxdDf4y=SPM-aYM4PebvB9kY3A0Xvm7PFa7rexzT+mbG_MQg@mail.gmail.com>
	<CAHLYDCsbXpxN=zPBiNOM68nGP-e27m362+-TkJmf0uWDzfyPAw@mail.gmail.com>
Message-ID: <CACOYK=ixSLJNgkiQUvDZWGQZLpPAToiNLiidwgrNMbd-ymp1kg@mail.gmail.com>

did you check it here? https://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones
You need to have ringback variable, and the behavior is different
depending on ignore_early_media set to true or false.



On Sun, May 17, 2015 at 12:53 AM, Juan Antonio Iba?ez Santorum
<juanito1982 at gmail.com> wrote:
> Yes, RTP goes through FS
>
> El 17/05/2015 00:10, "Sergey Safarov" <s.safarov at gmail.com> escribi?:
>>
>> Are is RTP media is proxied by FS?
>>
>> On Sun, May 17, 2015 at 12:05 AM, Juan Antonio Iba?ez Santorum
>> <juanito1982 at gmail.com> wrote:
>>>
>>> Hello,
>>>
>>>   I am doing some test in a callback scenario. We make a call from the
>>> server to the caller through A provider which is answered. Then we make a
>>> sencod call to the calle through B provider. When callee ring is ready, B
>>> provider send to our server a 180 Ringing but the server does not start
>>> generate that ringing tone to the caller.
>>>
>>> You can see full flow here:
>>>
>>> http://es.tinypic.com/r/vsj1w2/8
>>>
>>> Is there any way to force FS to generate ringing for A leg when a 180
>>> gets received from B leg?
>>>
>>> Regards
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mitchelle.bit at gmail.com  Sun May 17 11:57:18 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Sun, 17 May 2015 13:27:18 +0530
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
Message-ID: <CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>

I am sorry, I am not able to understand to how use the link provided by you
to resolve my issue...could you please explain me the process in detail.

Thanks,
Mitchelle

On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> See https://freeswitch.org/jira/browse/FS-7258
>
> On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <
> mitchelle.bit at gmail.com> wrote:
>
>> Hi,
>> When I am using the web server to handle xml CDR's the xml file which it
>> sends gives an error.
>> The error being:
>>
>> XML Parsing Error: not well-formed
>> Location: http://www.w3schools.com/xml/xml_validator.asp
>> Line Number 138, Column 25:
>>     <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
>> ------------------------^
>>
>> Please help me resolve this.
>>
>>
>> Thanks,
>> Mitchelle
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ali.jibran44 at gmail.com  Sun May 17 12:02:56 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Sun, 17 May 2015 13:02:56 +0500
Subject: [Freeswitch-users] XML parsing error
In-Reply-To: <CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
Message-ID: <CAP9PqkQywKjHmxjtMS5_A+2W0FugfhCy=KYqsz1_FCY_A2iEDQ@mail.gmail.com>

I had the same issue with FusionPBX. The issue was that the data wasn't
being urlencoded correctly.
As a work around I had to manually encode every tag and then pass it on to
the XML parser.

That is very hectic. Turned out it was a FS issue. 1.5 was giving me this
issue. I reverted back to 1.4.15 and bam. Issue resolved.

On Sunday, May 17, 2015, Mitchelle Johnson <mitchelle.bit at gmail.com> wrote:

> I am sorry, I am not able to understand to how use the link provided by
> you to resolve my issue...could you please explain me the process in detail.
>
> Thanks,
> Mitchelle
>
> On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov <s.safarov at gmail.com
> <javascript:_e(%7B%7D,'cvml','s.safarov at gmail.com');>> wrote:
>
>> See https://freeswitch.org/jira/browse/FS-7258
>>
>> On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <
>> mitchelle.bit at gmail.com
>> <javascript:_e(%7B%7D,'cvml','mitchelle.bit at gmail.com');>> wrote:
>>
>>> Hi,
>>> When I am using the web server to handle xml CDR's the xml file which it
>>> sends gives an error.
>>> The error being:
>>>
>>> XML Parsing Error: not well-formed
>>> Location: http://www.w3schools.com/xml/xml_validator.asp
>>> Line Number 138, Column 25:
>>>     <sip_full_from><sip:1001 at 10.0.0.8
>>> <javascript:_e(%7B%7D,'cvml','sip:1001 at 10.0.0.8');>
>>> >;tag=661a086d</sip_full_from>
>>> ------------------------^
>>>
>>> Please help me resolve this.
>>>
>>>
>>> Thanks,
>>> Mitchelle
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From ali.jibran44 at gmail.com  Sun May 17 12:06:32 2015
From: ali.jibran44 at gmail.com (Ali Jibran)
Date: Sun, 17 May 2015 13:06:32 +0500
Subject: [Freeswitch-users] CDR problem
In-Reply-To: <55534490.4090500@tausys.de>
References: <55525F86.6080306@tausys.de>
	<CAP9PqkTXX0Enna6na1ACMFCVcEiCnuXLiak7r4gvRHz2AjJZsg@mail.gmail.com>
	<55534490.4090500@tausys.de>
Message-ID: <CAP9PqkRKkwxhwozFUHO7KJQUV1SPTedBM7aio3WACPmESNT6zw@mail.gmail.com>

You're welcome :)
Try fusions site maybe. I haven't really submitted a bug so I can't help
you there.

Happy switching!

On Wednesday, May 13, 2015, Jens Tautenhahn <shogun at tausys.de> wrote:

> Thanks for your help! I have found the error.
>
> In "v_xml_cdr_import.php" the variable "sip_P-Preferred-Identity" would
> not be masked. The user can pass there anything. Without masking this
> leads to not correct XML.
>
> Where can I submit a bug report or patch?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From juanito1982 at gmail.com  Sun May 17 12:12:40 2015
From: juanito1982 at gmail.com (=?UTF-8?Q?Juan_Antonio_Iba=C3=B1ez_Santorum?=)
Date: Sun, 17 May 2015 10:12:40 +0200
Subject: [Freeswitch-users] No audio on 180 Ringing receipt
In-Reply-To: <CACOYK=ixSLJNgkiQUvDZWGQZLpPAToiNLiidwgrNMbd-ymp1kg@mail.gmail.com>
References: <CAHLYDCvD4fuRM_zCCTE1K5M2EJNhkvyyAvUH++sVfyktM3kNPw@mail.gmail.com>
	<CAHtxdDf4y=SPM-aYM4PebvB9kY3A0Xvm7PFa7rexzT+mbG_MQg@mail.gmail.com>
	<CAHLYDCsbXpxN=zPBiNOM68nGP-e27m362+-TkJmf0uWDzfyPAw@mail.gmail.com>
	<CACOYK=ixSLJNgkiQUvDZWGQZLpPAToiNLiidwgrNMbd-ymp1kg@mail.gmail.com>
Message-ID: <CAHLYDCtirUBrmtGrwBk4AqOmn3qLZ36FJsU_373TBooKet=nAA@mail.gmail.com>

You are the man Stanislav! Thank you very much

2015-05-17 7:49 GMT+02:00 Stanislav Sinyagin <ssinyagin at gmail.com>:

> did you check it here?
> https://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones
> You need to have ringback variable, and the behavior is different
> depending on ignore_early_media set to true or false.
>
>
>
> On Sun, May 17, 2015 at 12:53 AM, Juan Antonio Iba?ez Santorum
> <juanito1982 at gmail.com> wrote:
> > Yes, RTP goes through FS
> >
> > El 17/05/2015 00:10, "Sergey Safarov" <s.safarov at gmail.com> escribi?:
> >>
> >> Are is RTP media is proxied by FS?
> >>
> >> On Sun, May 17, 2015 at 12:05 AM, Juan Antonio Iba?ez Santorum
> >> <juanito1982 at gmail.com> wrote:
> >>>
> >>> Hello,
> >>>
> >>>   I am doing some test in a callback scenario. We make a call from the
> >>> server to the caller through A provider which is answered. Then we
> make a
> >>> sencod call to the calle through B provider. When callee ring is
> ready, B
> >>> provider send to our server a 180 Ringing but the server does not start
> >>> generate that ringing tone to the caller.
> >>>
> >>> You can see full flow here:
> >>>
> >>> http://es.tinypic.com/r/vsj1w2/8
> >>>
> >>> Is there any way to force FS to generate ringing for A leg when a 180
> >>> gets received from B leg?
> >>>
> >>> Regards
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From simpot at gmail.com  Sun May 17 12:17:27 2015
From: simpot at gmail.com (Dmitry Saratsky)
Date: Sun, 17 May 2015 11:17:27 +0300
Subject: [Freeswitch-users] need correct SDP media examples for all
	variances of g729
Message-ID: <CAM2iYy5gKrJEswBMQBKM49oXa3Dj8vrGV-Nn2XmQMCQfhY1Z0g@mail.gmail.com>

Hi all,

Can you please show me CORRECT SDP media examples for all possible G.729
annexes (need only G.729, G.729A, G.729B and G.729AB).

How should correct SDP look like for each case of the above including fmtp
parameters and maybe other parameters?

Thanks guys!

BR,
Dmitry.
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From mitchelle.bit at gmail.com  Sun May 17 12:20:04 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Sun, 17 May 2015 13:50:04 +0530
Subject: [Freeswitch-users] XML parsing error
In-Reply-To: <CAP9PqkQywKjHmxjtMS5_A+2W0FugfhCy=KYqsz1_FCY_A2iEDQ@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
	<CAP9PqkQywKjHmxjtMS5_A+2W0FugfhCy=KYqsz1_FCY_A2iEDQ@mail.gmail.com>
Message-ID: <CANBm6Sz3TG3VaNT5T0UPcOsBfZZwt7fZKt3krouCbSfxo+hHew@mail.gmail.com>

Hi Ali,
The issue I think is not with FreeSWITCH coz I have a production system
that has FS installed on a centOS environment. and when I took its xml cdr
to the check for the same issue... FS 1.5 on centOS didnt have the same
issue.

Thanks,
Mitchelle

On Sun, May 17, 2015 at 1:32 PM, Ali Jibran <ali.jibran44 at gmail.com> wrote:

> I had the same issue with FusionPBX. The issue was that the data wasn't
> being urlencoded correctly.
> As a work around I had to manually encode every tag and then pass it on to
> the XML parser.
>
> That is very hectic. Turned out it was a FS issue. 1.5 was giving me this
> issue. I reverted back to 1.4.15 and bam. Issue resolved.
>
>
> On Sunday, May 17, 2015, Mitchelle Johnson <mitchelle.bit at gmail.com>
> wrote:
>
>> I am sorry, I am not able to understand to how use the link provided by
>> you to resolve my issue...could you please explain me the process in detail.
>>
>> Thanks,
>> Mitchelle
>>
>> On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> See https://freeswitch.org/jira/browse/FS-7258
>>>
>>> On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <
>>> mitchelle.bit at gmail.com> wrote:
>>>
>>>> Hi,
>>>> When I am using the web server to handle xml CDR's the xml file which
>>>> it sends gives an error.
>>>> The error being:
>>>>
>>>> XML Parsing Error: not well-formed
>>>> Location: http://www.w3schools.com/xml/xml_validator.asp
>>>> Line Number 138, Column 25:
>>>>     <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
>>>> ------------------------^
>>>>
>>>> Please help me resolve this.
>>>>
>>>>
>>>> Thanks,
>>>> Mitchelle
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun May 17 14:52:24 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 17 May 2015 13:52:24 +0300
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
Message-ID: <CAHtxdDccdRxC=AuU_kpwAthhvT4wmR8foVEVve1hT3ZL=A9xbQ@mail.gmail.com>

Mitchelle manually or using patch utility apply changes in off #192 pull
request.
https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/192/diff

On Sun, May 17, 2015 at 10:57 AM, Mitchelle Johnson <mitchelle.bit at gmail.com
> wrote:

> I am sorry, I am not able to understand to how use the link provided by
> you to resolve my issue...could you please explain me the process in detail.
>
> Thanks,
> Mitchelle
>
> On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> See https://freeswitch.org/jira/browse/FS-7258
>>
>> On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <
>> mitchelle.bit at gmail.com> wrote:
>>
>>> Hi,
>>> When I am using the web server to handle xml CDR's the xml file which it
>>> sends gives an error.
>>> The error being:
>>>
>>> XML Parsing Error: not well-formed
>>> Location: http://www.w3schools.com/xml/xml_validator.asp
>>> Line Number 138, Column 25:
>>>     <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
>>> ------------------------^
>>>
>>> Please help me resolve this.
>>>
>>>
>>> Thanks,
>>> Mitchelle
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mitchelle.bit at gmail.com  Sun May 17 15:18:17 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Sun, 17 May 2015 16:48:17 +0530
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <CAHtxdDccdRxC=AuU_kpwAthhvT4wmR8foVEVve1hT3ZL=A9xbQ@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
	<CAHtxdDccdRxC=AuU_kpwAthhvT4wmR8foVEVve1hT3ZL=A9xbQ@mail.gmail.com>
Message-ID: <CANBm6Sze77vWO+GxO6KD22mii9JSvgVVmj+_HxA9n1FB47_CaQ@mail.gmail.com>

Thanks Sergey,
Could you please tell me how to apply changes in off #192 pull request?

Thanks,
Mitchelle

On Sun, May 17, 2015 at 4:22 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Mitchelle manually or using patch utility apply changes in off #192 pull
> request.
>
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/192/diff
>
> On Sun, May 17, 2015 at 10:57 AM, Mitchelle Johnson <
> mitchelle.bit at gmail.com> wrote:
>
>> I am sorry, I am not able to understand to how use the link provided by
>> you to resolve my issue...could you please explain me the process in detail.
>>
>> Thanks,
>> Mitchelle
>>
>> On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> See https://freeswitch.org/jira/browse/FS-7258
>>>
>>> On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <
>>> mitchelle.bit at gmail.com> wrote:
>>>
>>>> Hi,
>>>> When I am using the web server to handle xml CDR's the xml file which
>>>> it sends gives an error.
>>>> The error being:
>>>>
>>>> XML Parsing Error: not well-formed
>>>> Location: http://www.w3schools.com/xml/xml_validator.asp
>>>> Line Number 138, Column 25:
>>>>     <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
>>>> ------------------------^
>>>>
>>>> Please help me resolve this.
>>>>
>>>>
>>>> Thanks,
>>>> Mitchelle
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From saumar at uol.com.br  Sun May 17 17:51:18 2015
From: saumar at uol.com.br (Saumar Hajjar)
Date: Sun, 17 May 2015 10:51:18 -0300
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <CANBm6Sze77vWO+GxO6KD22mii9JSvgVVmj+_HxA9n1FB47_CaQ@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>	<CAHtxdDccdRxC=AuU_kpwAthhvT4wmR8foVEVve1hT3ZL=A9xbQ@mail.gmail.com>
	<CANBm6Sze77vWO+GxO6KD22mii9JSvgVVmj+_HxA9n1FB47_CaQ@mail.gmail.com>
Message-ID: <55589CD6.4060303@uol.com.br>

Hi Mitchelle,

I've also faced this issue when I started developing with 1.4 release - 
at the time I gave the master branch a try - and then got lucky.
Shortly after that, I updated my setup to a more recent master and xml 
cdrs got broken again.

broken 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 
2015-03-12 18:55:23Z 64bit)
works 1.5.15b+git~20150117T062211Z~46cf8a4dce~64bit (git 46cf8a4 
2015-01-17 06:22:11Z 64bit)
broken 1.5.15b+git~20150421T235828Z~a4d877c189~64bit (git a4d877c 
2015-04-21 23:58:28Z 64bit)

Below you'll find what I'm using in all versions I have installed now:

// switch_utils.c
SWITCH_DECLARE(char *) switch_url_encode(const char *url, char *buf, 
size_t len)
{
     const char *p;
     size_t x = 0;
     const char urlunsafe[] = "\r\n \"#%&+:;<=>?@[\\]^`{|}";
     const char hex[] = "0123456789ABCDEF";

     if (!buf) {
         return 0;
     }

     if (!url) {
         return 0;
     }

     len--;

     for (p = url; *p; p++) {
         if (x >= len) {
             break;
         }
         if (*p < ' ' || *p > '~' || strchr(urlunsafe, *p)) {
             if ((x + 3) > len) {
                 break;
             }
             buf[x++] = '%';
             buf[x++] = hex[(*p >> 4) & 0x0f];
             buf[x++] = hex[*p & 0x0f];
         } else {
             buf[x++] = *p;
         }
     }
     buf[x] = '\0';

     return buf;
}

On 17/05/2015 08:18, Mitchelle Johnson wrote:
> Thanks Sergey,
> Could you please tell me how to apply changes in off #192 pull request?
>
> Thanks,
> Mitchelle
>
> On Sun, May 17, 2015 at 4:22 PM, Sergey Safarov <s.safarov at gmail.com 
> <mailto:s.safarov at gmail.com>> wrote:
>
>     Mitchelle manually or using patch utility apply changes in off
>     #192 pull request.
>     https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/192/diff
>
>     On Sun, May 17, 2015 at 10:57 AM, Mitchelle Johnson
>     <mitchelle.bit at gmail.com <mailto:mitchelle.bit at gmail.com>> wrote:
>
>         I am sorry, I am not able to understand to how use the link
>         provided by you to resolve my issue...could you please explain
>         me the process in detail.
>
>         Thanks,
>         Mitchelle
>
>         On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov
>         <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>
>             See https://freeswitch.org/jira/browse/FS-7258
>
>             On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson
>             <mitchelle.bit at gmail.com <mailto:mitchelle.bit at gmail.com>>
>             wrote:
>
>                 Hi,
>                 When I am using the web server to handle xml CDR's the
>                 xml file which it sends gives an error.
>                 The error being:
>
>                 XML Parsing Error: not well-formed
>                 Location: http://www.w3schools.com/xml/xml_validator.asp
>                 Line Number 138, Column 25:
>                 <sip_full_from><sip:1001 at 10.0.0.8
>                 <mailto:sip%3A1001 at 10.0.0.8>>;tag=661a086d</sip_full_from>
>                 ------------------------^
>
>                 Please help me resolve this.
>
>
>                 Thanks,
>                 Mitchelle
>
>                 _________________________________________________________________________
>                 Professional FreeSWITCH Consulting Services:
>                 consulting at freeswitch.org
>                 <mailto:consulting at freeswitch.org>
>                 http://www.freeswitchsolutions.com
>
>                 Official FreeSWITCH Sites
>                 http://www.freeswitch.org
>                 http://confluence.freeswitch.org
>                 http://www.cluecon.com
>
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 http://www.freeswitch.org
>
>
>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>             http://www.freeswitchsolutions.com
>
>             Official FreeSWITCH Sites
>             http://www.freeswitch.org
>             http://confluence.freeswitch.org
>             http://www.cluecon.com
>
>             FreeSWITCH-users mailing list
>             FreeSWITCH-users at lists.freeswitch.org
>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>             http://www.freeswitch.org
>
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://confluence.freeswitch.org
>         http://www.cluecon.com
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From nasida at live.ru  Sun May 17 18:44:53 2015
From: nasida at live.ru (Yuriy Nasida)
Date: Sun, 17 May 2015 17:44:53 +0300
Subject: [Freeswitch-users] unexpected segfault with latest debian and
 libmyodbc.so
In-Reply-To: <CAJR7f2FdFFqUEdTpQSN59uzVg_p3nq_kBj5FvDYtM2pUieDeOA@mail.gmail.com>
References: <DUB122-W36F85086474E2280B9ED15D5090@phx.gbl>,
	<CAJR7f2HCFX6wOTxNStRVO8by4x7HiSUo9zQf-8566SzqqsDFvA@mail.gmail.com>,
	<CAKbxfG8+=HVT_D7FqUwdynRa8-8KEvkOVzuPnjF6tnmZipuvQA@mail.gmail.com>,
	<DUB122-W720E4C2E410A9EA55250CD5F20@phx.gbl>,
	<CAJR7f2FdFFqUEdTpQSN59uzVg_p3nq_kBj5FvDYtM2pUieDeOA@mail.gmail.com>
Message-ID: <DUB122-W425F87363588D683B60CC6D5C50@phx.gbl>

Guys, unfortunately updating of unixODBC didn't help and I getting segfaults again after some quiet period.

Now I have: 
1) unixODBC 2.3.1
2) FreeSWITCH Version 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 18:59:51Z 64bit)
3) Debian GNU/Linux 7.8 (wheezy)
4) libmyodbc  5.1.10-2+deb7u1

Please advice
Thanks




Date: Thu, 2 Apr 2015 16:19:13 -0300
From: italorossib at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] unexpected segfault with latest debian and	libmyodbc.so

Great! :)

On Thu, Apr 2, 2015 at 2:17 PM, Yuriy Nasida <nasida at live.ru> wrote:



I have updated unixodbc to 2.3 (from source) and looks like it fixed the problem. Thanks guys!



Date: Fri, 27 Mar 2015 17:39:55 -0500
From: anthony.minessale at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] unexpected segfault with latest debian and	libmyodbc.so

We actually recommend Threading=0 for postgres and mysql for the most part.

On Fri, Mar 27, 2015 at 7:48 AM, ?talo Rossi <italorossib at gmail.com> wrote:
I have seen a lot of threading issues with mysql + odbc, make sure you're using unixodbc >= 2.3. 
If you're using an older version you can set Threading = 2 in your /etc/odbcinst.ini as a workaround, but this is *not* recommended for production/high volume, upgrade as soon as possible.
On Fri, Mar 27, 2015 at 9:41 AM, Yuriy Nasida <nasida at live.ru> wrote:



Hi guys,

I just got unexpected segfault I try to understand if anybody had similar problems.

Mar 26 07:00:19 kernel: [226389.252971] freeswitch[7972]: segfault at 500 ip 00007fd3a8362252 sp 00007fd34a7d9f70 error 4 in libmyodbc.so[7fd3a8340000+3c000]


# lsb_release -a
Distributor ID:    Debian
Description:    Debian GNU/Linux 7.8 (wheezy)
Release:    7.8
Codename:    wheezy

freeswitch at internal> version 
FreeSWITCH Version 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 18:59:51Z 64bit)

# apt-cache show libmyodbc
Package: libmyodbc
Source: myodbc
Version: 5.1.10-2+deb7u1

Please advice,
Thanks.

 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

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-- 
?talo Rossi



_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

http://www.cluecon.com



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http://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


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

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

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org


-- 
?talo Rossi



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

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

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From s.safarov at gmail.com  Sun May 17 23:13:17 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 17 May 2015 22:13:17 +0300
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <CAEJMVkBk-+YMRYGXQnH+Eu_GT571BzKNfrkp4zGDRmVabKDSgg@mail.gmail.com>
References: <1251442930.20150513084828@seznam.cz>
	<D178B500.9DA9B%krice@freeswitch.org>
	<CAHtxdDft6M-mF+=+L7jGLxPAD+EzHohYBou4O_f6rsYwCofYqg@mail.gmail.com>
	<1641553624.20150513155908@seznam.cz>
	<CAEJMVkBk-+YMRYGXQnH+Eu_GT571BzKNfrkp4zGDRmVabKDSgg@mail.gmail.com>
Message-ID: <CAHtxdDc7URbEp0F8conEycByR32iQckqa-o3ekHAjuUPqXxB7g@mail.gmail.com>

I has created dependency packages and has placed on
http://fs-repo.network-engineer.ru/
A is anybody can help to place this rpms on
http://files.freeswitch.org/repo/ ?

Sergey Safarov


On Wed, May 13, 2015 at 6:21 PM, Brian West <brian at freeswitch.org> wrote:

> I've assigned it to Sergey Safarov, So lets get to busy ;)
>
> On Wed, May 13, 2015 at 8:59 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>>  Dobr? den,
>>
>> I hope I did it right
>>
>> https://freeswitch.org/jira/browse/FS-7553
>>
>>
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382 st?eda 13. kv?tna 2015, 15:34:55, napsal
>> jste: *
>>
>>  I has rpm repo with all required packages to install current FS master
>> via yum package manager on CentOS 7. All packages ported from fedora repo.
>> Thinking 1.6 branch can be packaged similarly.
>> If anybody write Jira tiket and this tiket will be assigned to me, then I
>> create pull request to automate build dependences on CentOS 7.
>>
>> Sergey Safarov
>>
>>
>> On Wed, May 13, 2015 at 4:00 PM, Ken Rice <krice at freeswitch.org> wrote:
>> There is a massive shift in the number of dependencies for compiling
>> FreeSWITCH. We are working on getting that dep chain working on non-debian
>> platforms but that work is not yet complete.
>>
>> If you wish to try and report back to the list the steps you took to get
>> it working on Centos7 so that others may benefit, most of the deps that you
>> cant get from yum are located at
>> https://freeswitch.org/stash/projects/SD/
>>
>> This may not be an all inclusive list for building on non-debian systems
>> but it will get you close.
>>
>>
>>
>>
>> On 5/13/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz> wrote:
>>
>> Dobr? den,
>>
>> This version 7, yes. This version is somehow different?
>>
>> --
>> Ken
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> irc.freenode.net #freeswitch
>> Twitter: @FreeSWITCH
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.FreeSWITCH.org>
>> http://confluence.freeswitch.org
>> http://www.cluecon.com <http://www.ClueCon.com>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 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
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
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> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
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From mitchelle.bit at gmail.com  Mon May 18 09:37:32 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Mon, 18 May 2015 11:07:32 +0530
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <55589CD6.4060303@uol.com.br>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
	<CAHtxdDccdRxC=AuU_kpwAthhvT4wmR8foVEVve1hT3ZL=A9xbQ@mail.gmail.com>
	<CANBm6Sze77vWO+GxO6KD22mii9JSvgVVmj+_HxA9n1FB47_CaQ@mail.gmail.com>
	<55589CD6.4060303@uol.com.br>
Message-ID: <CANBm6SyXEfkyFkReYkrB+=LKBkYZv5-0hWuHqWBY-gMbMWLYgQ@mail.gmail.com>

Hi Saumar,
Could you please tell me where to incorporate the code snippet that you
posted? It will be a lot of help if you could tell me in detail as to what
file to make changes in and the location of the file, etc..

Thanks,
Mitchelle

On Sun, May 17, 2015 at 7:21 PM, Saumar Hajjar <saumar at uol.com.br> wrote:

>  Hi Mitchelle,
>
> I've also faced this issue when I started developing with 1.4 release - at
> the time I gave the master branch a try - and then got lucky.
> Shortly after that, I updated my setup to a more recent master and xml
> cdrs got broken again.
>
> broken 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221
> 2015-03-12 18:55:23Z 64bit)
> works 1.5.15b+git~20150117T062211Z~46cf8a4dce~64bit (git 46cf8a4
> 2015-01-17 06:22:11Z 64bit)
> broken 1.5.15b+git~20150421T235828Z~a4d877c189~64bit (git a4d877c
> 2015-04-21 23:58:28Z 64bit)
>
> Below you'll find what I'm using in all versions I have installed now:
>
> // switch_utils.c
> SWITCH_DECLARE(char *) switch_url_encode(const char *url, char *buf,
> size_t len)
> {
>     const char *p;
>     size_t x = 0;
>     const char urlunsafe[] = "\r\n \"#%&+:;<=>?@[\\]^`{|}";
>     const char hex[] = "0123456789ABCDEF";
>
>     if (!buf) {
>         return 0;
>     }
>
>     if (!url) {
>         return 0;
>     }
>
>     len--;
>
>     for (p = url; *p; p++) {
>         if (x >= len) {
>             break;
>         }
>         if (*p < ' ' || *p > '~' || strchr(urlunsafe, *p)) {
>             if ((x + 3) > len) {
>                 break;
>             }
>             buf[x++] = '%';
>             buf[x++] = hex[(*p >> 4) & 0x0f];
>             buf[x++] = hex[*p & 0x0f];
>         } else {
>             buf[x++] = *p;
>         }
>     }
>     buf[x] = '\0';
>
>     return buf;
>
> }
>
> On 17/05/2015 08:18, Mitchelle Johnson wrote:
>
>   Thanks Sergey,
>  Could you please tell me how to apply changes in off #192 pull request?
>
>  Thanks,
>  Mitchelle
>
> On Sun, May 17, 2015 at 4:22 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Mitchelle manually or using patch utility apply changes in off #192 pull
>> request.
>>
>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/192/diff
>>
>> On Sun, May 17, 2015 at 10:57 AM, Mitchelle Johnson <
>> mitchelle.bit at gmail.com> wrote:
>>
>>>  I am sorry, I am not able to understand to how use the link provided
>>> by you to resolve my issue...could you please explain me the process in
>>> detail.
>>>
>>>  Thanks,
>>>  Mitchelle
>>>
>>> On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov <s.safarov at gmail.com>
>>> wrote:
>>>
>>>> See https://freeswitch.org/jira/browse/FS-7258
>>>>
>>>> On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <
>>>> mitchelle.bit at gmail.com> wrote:
>>>>
>>>>>     Hi,
>>>>>  When I am using the web server to handle xml CDR's the xml file which
>>>>> it sends gives an error.
>>>>>  The error being:
>>>>>
>>>>> XML Parsing Error: not well-formed
>>>>> Location: http://www.w3schools.com/xml/xml_validator.asp
>>>>> Line Number 138, Column 25:
>>>>>     <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
>>>>> ------------------------^
>>>>>
>>>>>  Please help me resolve this.
>>>>>
>>>>>
>>>>>  Thanks,
>>>>>  Mitchelle
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/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 bordmi at rarus.ru  Mon May 18 11:23:43 2015
From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=)
Date: Mon, 18 May 2015 10:23:43 +0300
Subject: [Freeswitch-users] How to create an event for B-leg?
Message-ID: <CAL2h29jM9s26f04YcUh3HdeTkXKsuF8jeBGBxpTEQ1wtm-UcPg@mail.gmail.com>

Hi, All!

I need to receive some event when phone on leg B become into RINGING state.
I need domain_name information, original-caller-id, callee-id, channel-uuid
and destination-number.

Now I wrote script which creates and fires this event, but it started by
     session:setVariable("bridge_pre_execute_bleg_app","lua")
     session:setVariable("bridge_pre_execute_bleg_data","lua/event_fire.lua
" .. event_id .." uuid")
after B-leg was answered :(

How can I run this script when channel receive RINGING?

-- 
with best regards,
Dmitriy Borisov
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From bordmi at rarus.ru  Mon May 18 11:40:43 2015
From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=)
Date: Mon, 18 May 2015 10:40:43 +0300
Subject: [Freeswitch-users] How to create an event for B-leg?
In-Reply-To: <CAL2h29jM9s26f04YcUh3HdeTkXKsuF8jeBGBxpTEQ1wtm-UcPg@mail.gmail.com>
References: <CAL2h29jM9s26f04YcUh3HdeTkXKsuF8jeBGBxpTEQ1wtm-UcPg@mail.gmail.com>
Message-ID: <CAL2h29jtgZcvw9GX=9OEnfToA73jYFOMYZZB2Bemz=CZXMTMJw@mail.gmail.com>

That`s all works fine when I changed to:

session:execute("export","nolocal:execute_on_ring=lua lua/event_fire.lua "
.. event_id .." uuid")

Sorry :)

2015-05-18 10:23 GMT+03:00 ???????, ??????? / Dmitriy Borisov <
bordmi at rarus.ru>:

> Hi, All!
>
> I need to receive some event when phone on leg B become into RINGING
> state. I need domain_name information, original-caller-id, callee-id,
> channel-uuid and destination-number.
>
> Now I wrote script which creates and fires this event, but it started by
>      session:setVariable("bridge_pre_execute_bleg_app","lua")
>
>  session:setVariable("bridge_pre_execute_bleg_data","lua/event_fire.lua "
> .. event_id .." uuid")
> after B-leg was answered :(
>
> How can I run this script when channel receive RINGING?
>
> --
> with best regards,
> Dmitriy Borisov
>



-- 
with best regards,
Dmitriy Borisov
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From emplant2000 at gmail.com  Mon May 18 12:13:55 2015
From: emplant2000 at gmail.com (Masakazu Nakano)
Date: Mon, 18 May 2015 17:13:55 +0900
Subject: [Freeswitch-users] not make a phone call both extensions
In-Reply-To: <CAM_EGOGD1Ehu-Mw2GwvO14Ld-jZW+bt2KcYESPLvPzqcWzQ1Gg@mail.gmail.com>
References: <CAPwL2MDwW_LEWziqsQE5nX2+bw74+kmCWCKmT2S2Ve7A3R4wpg@mail.gmail.com>
	<CAM_EGOERev5AaAG1PYoZKMNm8DmTfGWA-oz=hqWVz5SrCS4fdA@mail.gmail.com>
	<CAPwL2MD5TotyjKPVcwC14mJStS--7VpH0PNuesxjrVHAiWJc1w@mail.gmail.com>
	<CACOYK=jVLK2NMyPgz9xD_HfxwyfkXAu--CnffRN7xUfV6VsyEA@mail.gmail.com>
	<CAM_EGOGD1Ehu-Mw2GwvO14Ld-jZW+bt2KcYESPLvPzqcWzQ1Gg@mail.gmail.com>
Message-ID: <CAPwL2MD+LH0TeFr-WDp=MrfkbCzReWhaNJZC65N0jfSKWOE2ng@mail.gmail.com>

ok,I got it :)
about dialplan , store the file to dialplan/default and/or public and trial
to be done.

BR

mack

2015-05-17 5:29 GMT+09:00 Volkan Oransoy <voransoy at gmail.com>:

> To start a call on freeswitch console, please check this link.
> https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate
>
> 2015-05-16 13:37 GMT+03:00 Stanislav Sinyagin <ssinyagin at gmail.com>:
>
>> Hopefully this will help you as a starting point:
>> https://github.com/voxserv/freeswitch_conf_minimal/blob/master/docs/tutorial_01_simple_pbx.md
>> On May 16, 2015 12:28 PM, "Masakazu Nakano" <emplant2000 at gmail.com>
>> wrote:
>>
>>> Hi Volkan,
>>>
>>> thank you for your reply.
>>>
>>> I think I forgot setting up dialplan, and make it.but result is under
>>> like that.
>>>
>>> 2015-05-16 19:15:57.652782 [INFO] mod_dialplan_xml.c:635 Processing 5631
>>> <5631>->5630 in context public
>>> Dialplan: sofia/external/5631 at XXX parsing [public->unloop]
>>> continue=false
>>> Dialplan: sofia/external/5631 at XXX Regex (PASS) [unloop]
>>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>> Dialplan: sofia/external/5631 at XXX Regex (FAIL) [unloop]
>>> ${sip_looped_call}() =~ /^true$/ break=on-false
>>> Dialplan: sofia/external/5631 at XXX parsing [public->outside_call]
>>> continue=true
>>> Dialplan: sofia/external/5631 at XXX Absolute Condition [outside_call]
>>> Dialplan: sofia/external/5631 at XXX Action set(outside_call=true)
>>> Dialplan: sofia/external/5631 at XXX Action
>>> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
>>> Dialplan: sofia/external/5631 at XXX parsing [public->public_extensions]
>>> continue=false
>>> Dialplan: sofia/external/5631 at XXX Regex (PASS) [public_extensions]
>>> destination_number(5630) =~ /^(563[0-9])$/ break=on-false
>>> Dialplan: sofia/external/5631 at XXX Action
>>> bridge(sofia/doublenat/5630%27.XX.104.17 )
>>> 2015-05-16 19:15:57.652782 [DEBUG] switch_core_state_machine.c:216
>>> (sofia/external/5631 at XXX) State Change CS_ROUTING -> CS_EXECUTE
>>>
>>> Can I make a phone call by fs_cli to 5631 ??
>>>
>>> PS: I'm understand old style extension.conf is enought. but xml one is
>>> bit difficult... :(
>>>
>>> BR
>>>
>>> mack
>>>
>>>
>>>
>>> 2015-05-16 19:08 GMT+09:00 Volkan Oransoy <voransoy at gmail.com>:
>>>
>>>> Freeswitch is very flexible, so you can set many sip profiles for many
>>>> different scenarios.
>>>> The starting point of a context is a sip profile. When a call hits the
>>>> dialplan, the relevant sip profile sets the initial context.
>>>> You  should check your sip profile settings if you want to change the
>>>> initial context.
>>>>
>>>> /Volkan
>>>>
>>>> 2015-05-14 20:41 GMT+03:00 Masakazu Nakano <emplant2000 at gmail.com>:
>>>>
>>>>> Hi there
>>>>>
>>>>> I'm Freeswitch newbie. 15 years ago, usually used asterisk as old
>>>>> extension.conf style.
>>>>>
>>>>> so I got a cloud server like a EC2.
>>>>>
>>>>> I made a couple of users 5630 and 5631.xml and so on, specified
>>>>> default in context and in /conf/directory/default.
>>>>>
>>>>> and registered in np.
>>>>>
>>>>> freeswitch at internal> show registrations
>>>>>
>>>>> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata
>>>>> 5631,my
>>>>> cloud,76122ZWRhNDRmMDY0MzA2YWU5NzJkNWIxZWJiNmZmNTJmMjI,sofia/external/
>>>>> sip:5631 at 27.140.246.3:58996,1431626535,27.140.246.3,58996,udp,my
>>>>> cloud,
>>>>> 5630,my cloud,958199470722 at 10.210.59.150
>>>>> ,sofia/external/sip:5630 at 10.210.59.150:47420;transport=udp,1431626875,49.97.4.168,47420,udp,my
>>>>> cloud,
>>>>>
>>>>> 2 total.
>>>>>
>>>>> but fs_cli says 2015-05-15 02:31:47.752779 [INFO]
>>>>> mod_dialplan_xml.c:635 Processing 5631 <5631>->5630 in context public
>>>>>
>>>>> why ??
>>>>>
>>>>> BR
>>>>>
>>>>> mack
>>>>> https://www.facebook.com/nakano.masakazu
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From saumar at uol.com.br  Mon May 18 19:39:02 2015
From: saumar at uol.com.br (Saumar Hajjar)
Date: Mon, 18 May 2015 12:39:02 -0300
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <CANBm6SyXEfkyFkReYkrB+=LKBkYZv5-0hWuHqWBY-gMbMWLYgQ@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>	<CAHtxdDccdRxC=AuU_kpwAthhvT4wmR8foVEVve1hT3ZL=A9xbQ@mail.gmail.com>	<CANBm6Sze77vWO+GxO6KD22mii9JSvgVVmj+_HxA9n1FB47_CaQ@mail.gmail.com>	<55589CD6.4060303@uol.com.br>
	<CANBm6SyXEfkyFkReYkrB+=LKBkYZv5-0hWuHqWBY-gMbMWLYgQ@mail.gmail.com>
Message-ID: <555A0796.5060503@uol.com.br>

The file is  "src/switch_utils.c"
Edit this file, replace the function "switch_url_encode" and rebuild.

https://freeswitch.org/confluence/display/FREESWITCH/Installation


On 18/05/2015 02:37, Mitchelle Johnson wrote:
> Hi Saumar,
> Could you please tell me where to incorporate the code snippet that 
> you posted? It will be a lot of help if you could tell me in detail as 
> to what file to make changes in and the location of the file, etc..
>
> Thanks,
> Mitchelle
>
> On Sun, May 17, 2015 at 7:21 PM, Saumar Hajjar <saumar at uol.com.br 
> <mailto:saumar at uol.com.br>> wrote:
>
>     Hi Mitchelle,
>
>     I've also faced this issue when I started developing with 1.4
>     release - at the time I gave the master branch a try - and then
>     got lucky.
>     Shortly after that, I updated my setup to a more recent master and
>     xml cdrs got broken again.
>
>     broken 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221
>     2015-03-12 18:55:23Z 64bit)
>     works 1.5.15b+git~20150117T062211Z~46cf8a4dce~64bit (git 46cf8a4
>     2015-01-17 06:22:11Z 64bit)
>     broken 1.5.15b+git~20150421T235828Z~a4d877c189~64bit (git a4d877c
>     2015-04-21 23:58:28Z 64bit)
>
>     Below you'll find what I'm using in all versions I have installed now:
>
>     // switch_utils.c
>     SWITCH_DECLARE(char *) switch_url_encode(const char *url, char
>     *buf, size_t len)
>     {
>         const char *p;
>         size_t x = 0;
>         const char urlunsafe[] = "\r\n \"#%&+:;<=>?@[\\]^`{|}";
>         const char hex[] = "0123456789ABCDEF";
>
>         if (!buf) {
>             return 0;
>         }
>
>         if (!url) {
>             return 0;
>         }
>
>         len--;
>
>         for (p = url; *p; p++) {
>             if (x >= len) {
>                 break;
>             }
>             if (*p < ' ' || *p > '~' || strchr(urlunsafe, *p)) {
>                 if ((x + 3) > len) {
>                     break;
>                 }
>                 buf[x++] = '%';
>                 buf[x++] = hex[(*p >> 4) & 0x0f];
>                 buf[x++] = hex[*p & 0x0f];
>             } else {
>                 buf[x++] = *p;
>             }
>         }
>         buf[x] = '\0';
>
>         return buf;
>
>     }
>
>     On 17/05/2015 08:18, Mitchelle Johnson wrote:
>>     Thanks Sergey,
>>     Could you please tell me how to apply changes in off #192 pull
>>     request?
>>
>>     Thanks,
>>     Mitchelle
>>
>>     On Sun, May 17, 2015 at 4:22 PM, Sergey Safarov
>>     <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>>
>>         Mitchelle manually or using patch utility apply changes in
>>         off #192 pull request.
>>         https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/192/diff
>>
>>         On Sun, May 17, 2015 at 10:57 AM, Mitchelle Johnson
>>         <mitchelle.bit at gmail.com <mailto:mitchelle.bit at gmail.com>> wrote:
>>
>>             I am sorry, I am not able to understand to how use the
>>             link provided by you to resolve my issue...could you
>>             please explain me the process in detail.
>>
>>             Thanks,
>>             Mitchelle
>>
>>             On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov
>>             <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>>
>>                 See https://freeswitch.org/jira/browse/FS-7258
>>
>>                 On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson
>>                 <mitchelle.bit at gmail.com
>>                 <mailto:mitchelle.bit at gmail.com>> wrote:
>>
>>                     Hi,
>>                     When I am using the web server to handle xml
>>                     CDR's the xml file which it sends gives an error.
>>                     The error being:
>>
>>                     XML Parsing Error: not well-formed
>>                     Location:
>>                     http://www.w3schools.com/xml/xml_validator.asp
>>                     Line Number 138, Column 25:
>>                     <sip_full_from><sip:1001 at 10.0.0.8
>>                     <mailto:sip%3A1001 at 10.0.0.8>>;tag=661a086d</sip_full_from>
>>                     ------------------------^
>>
>>                     Please help me resolve this.
>>
>>
>>                     Thanks,
>>                     Mitchelle
>>
>>                     _________________________________________________________________________
>>                     Professional FreeSWITCH Consulting Services:
>>                     consulting at freeswitch.org
>>                     <mailto:consulting at freeswitch.org>
>>                     http://www.freeswitchsolutions.com
>>
>>                     Official FreeSWITCH Sites
>>                     http://www.freeswitch.org
>>                     http://confluence.freeswitch.org
>>                     http://www.cluecon.com
>>
>>                     FreeSWITCH-users mailing list
>>                     FreeSWITCH-users at lists.freeswitch.org
>>                     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>                     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>                     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>                     http://www.freeswitch.org
>>
>>
>>
>>                 _________________________________________________________________________
>>                 Professional FreeSWITCH Consulting Services:
>>                 consulting at freeswitch.org
>>                 <mailto:consulting at freeswitch.org>
>>                 http://www.freeswitchsolutions.com
>>
>>                 Official FreeSWITCH Sites
>>                 http://www.freeswitch.org
>>                 http://confluence.freeswitch.org
>>                 http://www.cluecon.com
>>
>>                 FreeSWITCH-users mailing list
>>                 FreeSWITCH-users at lists.freeswitch.org
>>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>                 http://www.freeswitch.org
>>
>>
>>
>>             _________________________________________________________________________
>>             Professional FreeSWITCH Consulting Services:
>>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>             http://www.freeswitchsolutions.com
>>
>>             Official FreeSWITCH Sites
>>             http://www.freeswitch.org
>>             http://confluence.freeswitch.org
>>             http://www.cluecon.com
>>
>>             FreeSWITCH-users mailing list
>>             FreeSWITCH-users at lists.freeswitch.org
>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>             http://www.freeswitch.org
>>
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://confluence.freeswitch.org
>>         http://www.cluecon.com
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://confluence.freeswitch.org
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mitchelle.bit at gmail.com  Mon May 18 20:40:26 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Mon, 18 May 2015 22:10:26 +0530
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <555A0796.5060503@uol.com.br>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
	<CAHtxdDccdRxC=AuU_kpwAthhvT4wmR8foVEVve1hT3ZL=A9xbQ@mail.gmail.com>
	<CANBm6Sze77vWO+GxO6KD22mii9JSvgVVmj+_HxA9n1FB47_CaQ@mail.gmail.com>
	<55589CD6.4060303@uol.com.br>
	<CANBm6SyXEfkyFkReYkrB+=LKBkYZv5-0hWuHqWBY-gMbMWLYgQ@mail.gmail.com>
	<555A0796.5060503@uol.com.br>
Message-ID: <CANBm6SzYgjxboCahh1OSNXU=iHZYwmxMhZUrm8UHxdh14NrSow@mail.gmail.com>

Hi Saumar,

Thanks a lot for your help. I will do as you have said and would report if
the issue is not resolved.

Regards,
Mitchelle

On Mon, May 18, 2015 at 9:09 PM, Saumar Hajjar <saumar at uol.com.br> wrote:

>  The file is  "src/switch_utils.c"
> Edit this file, replace the function "switch_url_encode" and rebuild.
>
> https://freeswitch.org/confluence/display/FREESWITCH/Installation
>
>
>
> On 18/05/2015 02:37, Mitchelle Johnson wrote:
>
>   Hi Saumar,
>  Could you please tell me where to incorporate the code snippet that you
> posted? It will be a lot of help if you could tell me in detail as to what
> file to make changes in and the location of the file, etc..
>
>  Thanks,
>  Mitchelle
>
> On Sun, May 17, 2015 at 7:21 PM, Saumar Hajjar <saumar at uol.com.br> wrote:
>
>>  Hi Mitchelle,
>>
>> I've also faced this issue when I started developing with 1.4 release -
>> at the time I gave the master branch a try - and then got lucky.
>> Shortly after that, I updated my setup to a more recent master and xml
>> cdrs got broken again.
>>
>> broken 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221
>> 2015-03-12 18:55:23Z 64bit)
>> works 1.5.15b+git~20150117T062211Z~46cf8a4dce~64bit (git 46cf8a4
>> 2015-01-17 06:22:11Z 64bit)
>> broken 1.5.15b+git~20150421T235828Z~a4d877c189~64bit (git a4d877c
>> 2015-04-21 23:58:28Z 64bit)
>>
>> Below you'll find what I'm using in all versions I have installed now:
>>
>> // switch_utils.c
>> SWITCH_DECLARE(char *) switch_url_encode(const char *url, char *buf,
>> size_t len)
>> {
>>     const char *p;
>>     size_t x = 0;
>>     const char urlunsafe[] = "\r\n \"#%&+:;<=>?@[\\]^`{|}";
>>     const char hex[] = "0123456789ABCDEF";
>>
>>     if (!buf) {
>>         return 0;
>>     }
>>
>>     if (!url) {
>>         return 0;
>>     }
>>
>>     len--;
>>
>>     for (p = url; *p; p++) {
>>         if (x >= len) {
>>             break;
>>         }
>>         if (*p < ' ' || *p > '~' || strchr(urlunsafe, *p)) {
>>             if ((x + 3) > len) {
>>                 break;
>>             }
>>             buf[x++] = '%';
>>             buf[x++] = hex[(*p >> 4) & 0x0f];
>>             buf[x++] = hex[*p & 0x0f];
>>         } else {
>>             buf[x++] = *p;
>>         }
>>     }
>>     buf[x] = '\0';
>>
>>     return buf;
>>
>> }
>>
>> On 17/05/2015 08:18, Mitchelle Johnson wrote:
>>
>>   Thanks Sergey,
>>  Could you please tell me how to apply changes in off #192 pull request?
>>
>>  Thanks,
>>  Mitchelle
>>
>> On Sun, May 17, 2015 at 4:22 PM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> Mitchelle manually or using patch utility apply changes in off #192 pull
>>> request.
>>>
>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/192/diff
>>>
>>> On Sun, May 17, 2015 at 10:57 AM, Mitchelle Johnson <
>>> mitchelle.bit at gmail.com> wrote:
>>>
>>>>  I am sorry, I am not able to understand to how use the link provided
>>>> by you to resolve my issue...could you please explain me the process in
>>>> detail.
>>>>
>>>>  Thanks,
>>>>  Mitchelle
>>>>
>>>> On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov <s.safarov at gmail.com>
>>>> wrote:
>>>>
>>>>> See https://freeswitch.org/jira/browse/FS-7258
>>>>>
>>>>> On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <
>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>
>>>>>>     Hi,
>>>>>>  When I am using the web server to handle xml CDR's the xml file
>>>>>> which it sends gives an error.
>>>>>>  The error being:
>>>>>>
>>>>>> XML Parsing Error: not well-formed
>>>>>> Location: http://www.w3schools.com/xml/xml_validator.asp
>>>>>> Line Number 138, Column 25:
>>>>>>     <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
>>>>>> ------------------------^
>>>>>>
>>>>>>  Please help me resolve this.
>>>>>>
>>>>>>
>>>>>>  Thanks,
>>>>>>  Mitchelle
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/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
>>
>
>
>
> _________________________________________________________________________
> 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 brian at freeswitch.org  Mon May 18 20:45:11 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 May 2015 11:45:11 -0500
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <CANBm6SzYgjxboCahh1OSNXU=iHZYwmxMhZUrm8UHxdh14NrSow@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
	<CAHtxdDccdRxC=AuU_kpwAthhvT4wmR8foVEVve1hT3ZL=A9xbQ@mail.gmail.com>
	<CANBm6Sze77vWO+GxO6KD22mii9JSvgVVmj+_HxA9n1FB47_CaQ@mail.gmail.com>
	<55589CD6.4060303@uol.com.br>
	<CANBm6SyXEfkyFkReYkrB+=LKBkYZv5-0hWuHqWBY-gMbMWLYgQ@mail.gmail.com>
	<555A0796.5060503@uol.com.br>
	<CANBm6SzYgjxboCahh1OSNXU=iHZYwmxMhZUrm8UHxdh14NrSow@mail.gmail.com>
Message-ID: <CAEJMVkBojvhvNXcf3vwmxsqhQZ6uQyp9e=uaWFz5KMvPJ6Rb5Q@mail.gmail.com>

The issue is already reported, and has a patch pending testing.

https://freeswitch.org/jira/browse/FS-7258

That patch needs some testing!  It will get merged and fix this issue but
not without more testing.

On Mon, May 18, 2015 at 11:40 AM, Mitchelle Johnson <mitchelle.bit at gmail.com
> wrote:

> Hi Saumar,
>
> Thanks a lot for your help. I will do as you have said and would report if
> the issue is not resolved.
>
> Regards,
> Mitchelle
>
> On Mon, May 18, 2015 at 9:09 PM, Saumar Hajjar <saumar at uol.com.br> wrote:
>
>>  The file is  "src/switch_utils.c"
>> Edit this file, replace the function "switch_url_encode" and rebuild.
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Installation
>>
>>
>>
>> On 18/05/2015 02:37, Mitchelle Johnson wrote:
>>
>>   Hi Saumar,
>>  Could you please tell me where to incorporate the code snippet that you
>> posted? It will be a lot of help if you could tell me in detail as to what
>> file to make changes in and the location of the file, etc..
>>
>>  Thanks,
>>  Mitchelle
>>
>> On Sun, May 17, 2015 at 7:21 PM, Saumar Hajjar <saumar at uol.com.br> wrote:
>>
>>>  Hi Mitchelle,
>>>
>>> I've also faced this issue when I started developing with 1.4 release -
>>> at the time I gave the master branch a try - and then got lucky.
>>> Shortly after that, I updated my setup to a more recent master and xml
>>> cdrs got broken again.
>>>
>>> broken 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221
>>> 2015-03-12 18:55:23Z 64bit)
>>> works 1.5.15b+git~20150117T062211Z~46cf8a4dce~64bit (git 46cf8a4
>>> 2015-01-17 06:22:11Z 64bit)
>>> broken 1.5.15b+git~20150421T235828Z~a4d877c189~64bit (git a4d877c
>>> 2015-04-21 23:58:28Z 64bit)
>>>
>>> Below you'll find what I'm using in all versions I have installed now:
>>>
>>> // switch_utils.c
>>> SWITCH_DECLARE(char *) switch_url_encode(const char *url, char *buf,
>>> size_t len)
>>> {
>>>     const char *p;
>>>     size_t x = 0;
>>>     const char urlunsafe[] = "\r\n \"#%&+:;<=>?@[\\]^`{|}";
>>>     const char hex[] = "0123456789ABCDEF";
>>>
>>>     if (!buf) {
>>>         return 0;
>>>     }
>>>
>>>     if (!url) {
>>>         return 0;
>>>     }
>>>
>>>     len--;
>>>
>>>     for (p = url; *p; p++) {
>>>         if (x >= len) {
>>>             break;
>>>         }
>>>         if (*p < ' ' || *p > '~' || strchr(urlunsafe, *p)) {
>>>             if ((x + 3) > len) {
>>>                 break;
>>>             }
>>>             buf[x++] = '%';
>>>             buf[x++] = hex[(*p >> 4) & 0x0f];
>>>             buf[x++] = hex[*p & 0x0f];
>>>         } else {
>>>             buf[x++] = *p;
>>>         }
>>>     }
>>>     buf[x] = '\0';
>>>
>>>     return buf;
>>>
>>> }
>>>
>>> On 17/05/2015 08:18, Mitchelle Johnson wrote:
>>>
>>>   Thanks Sergey,
>>>  Could you please tell me how to apply changes in off #192 pull request?
>>>
>>>  Thanks,
>>>  Mitchelle
>>>
>>> On Sun, May 17, 2015 at 4:22 PM, Sergey Safarov <s.safarov at gmail.com>
>>> wrote:
>>>
>>>> Mitchelle manually or using patch utility apply changes in off #192
>>>> pull request.
>>>>
>>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/192/diff
>>>>
>>>> On Sun, May 17, 2015 at 10:57 AM, Mitchelle Johnson <
>>>> mitchelle.bit at gmail.com> wrote:
>>>>
>>>>>  I am sorry, I am not able to understand to how use the link provided
>>>>> by you to resolve my issue...could you please explain me the process in
>>>>> detail.
>>>>>
>>>>>  Thanks,
>>>>>  Mitchelle
>>>>>
>>>>> On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov <s.safarov at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> See https://freeswitch.org/jira/browse/FS-7258
>>>>>>
>>>>>> On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <
>>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>>
>>>>>>>     Hi,
>>>>>>>  When I am using the web server to handle xml CDR's the xml file
>>>>>>> which it sends gives an error.
>>>>>>>  The error being:
>>>>>>>
>>>>>>> XML Parsing Error: not well-formed
>>>>>>> Location: http://www.w3schools.com/xml/xml_validator.asp
>>>>>>> Line Number 138, Column 25:
>>>>>>>     <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
>>>>>>> ------------------------^
>>>>>>>
>>>>>>>  Please help me resolve this.
>>>>>>>
>>>>>>>
>>>>>>>  Thanks,
>>>>>>>  Mitchelle
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/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
>>>
>>
>>
>>
>> _________________________________________________________________________
>> 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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 mitchelle.bit at gmail.com  Mon May 18 20:50:04 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Mon, 18 May 2015 22:20:04 +0530
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <CAEJMVkBojvhvNXcf3vwmxsqhQZ6uQyp9e=uaWFz5KMvPJ6Rb5Q@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
	<CAHtxdDccdRxC=AuU_kpwAthhvT4wmR8foVEVve1hT3ZL=A9xbQ@mail.gmail.com>
	<CANBm6Sze77vWO+GxO6KD22mii9JSvgVVmj+_HxA9n1FB47_CaQ@mail.gmail.com>
	<55589CD6.4060303@uol.com.br>
	<CANBm6SyXEfkyFkReYkrB+=LKBkYZv5-0hWuHqWBY-gMbMWLYgQ@mail.gmail.com>
	<555A0796.5060503@uol.com.br>
	<CANBm6SzYgjxboCahh1OSNXU=iHZYwmxMhZUrm8UHxdh14NrSow@mail.gmail.com>
	<CAEJMVkBojvhvNXcf3vwmxsqhQZ6uQyp9e=uaWFz5KMvPJ6Rb5Q@mail.gmail.com>
Message-ID: <CANBm6Sx00Lzu=1ZZ_2Y1DsfTCBws06nZg0Lh5exo=DdkBQpKTw@mail.gmail.com>

Hi Brian,

I also wanted to confirm about the 'billsec' in the xml cdr. It behaves
abnormally. Even if the call is not answered it shows some value inspite of
showing 0.

Please help.

Thanks,
Mitchelle


On Mon, May 18, 2015 at 10:15 PM, Brian West <brian at freeswitch.org> wrote:

> The issue is already reported, and has a patch pending testing.
>
> https://freeswitch.org/jira/browse/FS-7258
>
> That patch needs some testing!  It will get merged and fix this issue but
> not without more testing.
>
> On Mon, May 18, 2015 at 11:40 AM, Mitchelle Johnson <
> mitchelle.bit at gmail.com> wrote:
>
>> Hi Saumar,
>>
>> Thanks a lot for your help. I will do as you have said and would report
>> if the issue is not resolved.
>>
>> Regards,
>> Mitchelle
>>
>> On Mon, May 18, 2015 at 9:09 PM, Saumar Hajjar <saumar at uol.com.br> wrote:
>>
>>>  The file is  "src/switch_utils.c"
>>> Edit this file, replace the function "switch_url_encode" and rebuild.
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/Installation
>>>
>>>
>>>
>>> On 18/05/2015 02:37, Mitchelle Johnson wrote:
>>>
>>>   Hi Saumar,
>>>  Could you please tell me where to incorporate the code snippet that you
>>> posted? It will be a lot of help if you could tell me in detail as to what
>>> file to make changes in and the location of the file, etc..
>>>
>>>  Thanks,
>>>  Mitchelle
>>>
>>> On Sun, May 17, 2015 at 7:21 PM, Saumar Hajjar <saumar at uol.com.br>
>>> wrote:
>>>
>>>>  Hi Mitchelle,
>>>>
>>>> I've also faced this issue when I started developing with 1.4 release -
>>>> at the time I gave the master branch a try - and then got lucky.
>>>> Shortly after that, I updated my setup to a more recent master and xml
>>>> cdrs got broken again.
>>>>
>>>> broken 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221
>>>> 2015-03-12 18:55:23Z 64bit)
>>>> works 1.5.15b+git~20150117T062211Z~46cf8a4dce~64bit (git 46cf8a4
>>>> 2015-01-17 06:22:11Z 64bit)
>>>> broken 1.5.15b+git~20150421T235828Z~a4d877c189~64bit (git a4d877c
>>>> 2015-04-21 23:58:28Z 64bit)
>>>>
>>>> Below you'll find what I'm using in all versions I have installed now:
>>>>
>>>> // switch_utils.c
>>>> SWITCH_DECLARE(char *) switch_url_encode(const char *url, char *buf,
>>>> size_t len)
>>>> {
>>>>     const char *p;
>>>>     size_t x = 0;
>>>>     const char urlunsafe[] = "\r\n \"#%&+:;<=>?@[\\]^`{|}";
>>>>     const char hex[] = "0123456789ABCDEF";
>>>>
>>>>     if (!buf) {
>>>>         return 0;
>>>>     }
>>>>
>>>>     if (!url) {
>>>>         return 0;
>>>>     }
>>>>
>>>>     len--;
>>>>
>>>>     for (p = url; *p; p++) {
>>>>         if (x >= len) {
>>>>             break;
>>>>         }
>>>>         if (*p < ' ' || *p > '~' || strchr(urlunsafe, *p)) {
>>>>             if ((x + 3) > len) {
>>>>                 break;
>>>>             }
>>>>             buf[x++] = '%';
>>>>             buf[x++] = hex[(*p >> 4) & 0x0f];
>>>>             buf[x++] = hex[*p & 0x0f];
>>>>         } else {
>>>>             buf[x++] = *p;
>>>>         }
>>>>     }
>>>>     buf[x] = '\0';
>>>>
>>>>     return buf;
>>>>
>>>> }
>>>>
>>>> On 17/05/2015 08:18, Mitchelle Johnson wrote:
>>>>
>>>>   Thanks Sergey,
>>>>  Could you please tell me how to apply changes in off #192 pull request?
>>>>
>>>>  Thanks,
>>>>  Mitchelle
>>>>
>>>> On Sun, May 17, 2015 at 4:22 PM, Sergey Safarov <s.safarov at gmail.com>
>>>> wrote:
>>>>
>>>>> Mitchelle manually or using patch utility apply changes in off #192
>>>>> pull request.
>>>>>
>>>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/192/diff
>>>>>
>>>>> On Sun, May 17, 2015 at 10:57 AM, Mitchelle Johnson <
>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>
>>>>>>  I am sorry, I am not able to understand to how use the link
>>>>>> provided by you to resolve my issue...could you please explain me the
>>>>>> process in detail.
>>>>>>
>>>>>>  Thanks,
>>>>>>  Mitchelle
>>>>>>
>>>>>> On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov <s.safarov at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> See https://freeswitch.org/jira/browse/FS-7258
>>>>>>>
>>>>>>> On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <
>>>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>>>
>>>>>>>>     Hi,
>>>>>>>>  When I am using the web server to handle xml CDR's the xml file
>>>>>>>> which it sends gives an error.
>>>>>>>>  The error being:
>>>>>>>>
>>>>>>>> XML Parsing Error: not well-formed
>>>>>>>> Location: http://www.w3schools.com/xml/xml_validator.asp
>>>>>>>> Line Number 138, Column 25:
>>>>>>>>     <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
>>>>>>>> ------------------------^
>>>>>>>>
>>>>>>>>  Please help me resolve this.
>>>>>>>>
>>>>>>>>
>>>>>>>>  Thanks,
>>>>>>>>  Mitchelle
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/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
>>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 mitchelle.bit at gmail.com  Mon May 18 20:51:30 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Mon, 18 May 2015 22:21:30 +0530
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <CANBm6Sx00Lzu=1ZZ_2Y1DsfTCBws06nZg0Lh5exo=DdkBQpKTw@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
	<CAHtxdDccdRxC=AuU_kpwAthhvT4wmR8foVEVve1hT3ZL=A9xbQ@mail.gmail.com>
	<CANBm6Sze77vWO+GxO6KD22mii9JSvgVVmj+_HxA9n1FB47_CaQ@mail.gmail.com>
	<55589CD6.4060303@uol.com.br>
	<CANBm6SyXEfkyFkReYkrB+=LKBkYZv5-0hWuHqWBY-gMbMWLYgQ@mail.gmail.com>
	<555A0796.5060503@uol.com.br>
	<CANBm6SzYgjxboCahh1OSNXU=iHZYwmxMhZUrm8UHxdh14NrSow@mail.gmail.com>
	<CAEJMVkBojvhvNXcf3vwmxsqhQZ6uQyp9e=uaWFz5KMvPJ6Rb5Q@mail.gmail.com>
	<CANBm6Sx00Lzu=1ZZ_2Y1DsfTCBws06nZg0Lh5exo=DdkBQpKTw@mail.gmail.com>
Message-ID: <CANBm6Sym1Ci1bcPjE9NzYWBzgiPf2nW2OY=QhXSC6juom6x-Pg@mail.gmail.com>

Hi,

I will generate a new thread on billsec tag of xml cdr.

Thanks,
Mitchelle

On Mon, May 18, 2015 at 10:20 PM, Mitchelle Johnson <mitchelle.bit at gmail.com
> wrote:

> Hi Brian,
>
> I also wanted to confirm about the 'billsec' in the xml cdr. It behaves
> abnormally. Even if the call is not answered it shows some value inspite of
> showing 0.
>
> Please help.
>
> Thanks,
> Mitchelle
>
>
> On Mon, May 18, 2015 at 10:15 PM, Brian West <brian at freeswitch.org> wrote:
>
>> The issue is already reported, and has a patch pending testing.
>>
>> https://freeswitch.org/jira/browse/FS-7258
>>
>> That patch needs some testing!  It will get merged and fix this issue but
>> not without more testing.
>>
>> On Mon, May 18, 2015 at 11:40 AM, Mitchelle Johnson <
>> mitchelle.bit at gmail.com> wrote:
>>
>>> Hi Saumar,
>>>
>>> Thanks a lot for your help. I will do as you have said and would report
>>> if the issue is not resolved.
>>>
>>> Regards,
>>> Mitchelle
>>>
>>> On Mon, May 18, 2015 at 9:09 PM, Saumar Hajjar <saumar at uol.com.br>
>>> wrote:
>>>
>>>>  The file is  "src/switch_utils.c"
>>>> Edit this file, replace the function "switch_url_encode" and rebuild.
>>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/Installation
>>>>
>>>>
>>>>
>>>> On 18/05/2015 02:37, Mitchelle Johnson wrote:
>>>>
>>>>   Hi Saumar,
>>>>  Could you please tell me where to incorporate the code snippet that
>>>> you posted? It will be a lot of help if you could tell me in detail as to
>>>> what file to make changes in and the location of the file, etc..
>>>>
>>>>  Thanks,
>>>>  Mitchelle
>>>>
>>>> On Sun, May 17, 2015 at 7:21 PM, Saumar Hajjar <saumar at uol.com.br>
>>>> wrote:
>>>>
>>>>>  Hi Mitchelle,
>>>>>
>>>>> I've also faced this issue when I started developing with 1.4 release
>>>>> - at the time I gave the master branch a try - and then got lucky.
>>>>> Shortly after that, I updated my setup to a more recent master and xml
>>>>> cdrs got broken again.
>>>>>
>>>>> broken 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221
>>>>> 2015-03-12 18:55:23Z 64bit)
>>>>> works 1.5.15b+git~20150117T062211Z~46cf8a4dce~64bit (git 46cf8a4
>>>>> 2015-01-17 06:22:11Z 64bit)
>>>>> broken 1.5.15b+git~20150421T235828Z~a4d877c189~64bit (git a4d877c
>>>>> 2015-04-21 23:58:28Z 64bit)
>>>>>
>>>>> Below you'll find what I'm using in all versions I have installed now:
>>>>>
>>>>> // switch_utils.c
>>>>> SWITCH_DECLARE(char *) switch_url_encode(const char *url, char *buf,
>>>>> size_t len)
>>>>> {
>>>>>     const char *p;
>>>>>     size_t x = 0;
>>>>>     const char urlunsafe[] = "\r\n \"#%&+:;<=>?@[\\]^`{|}";
>>>>>     const char hex[] = "0123456789ABCDEF";
>>>>>
>>>>>     if (!buf) {
>>>>>         return 0;
>>>>>     }
>>>>>
>>>>>     if (!url) {
>>>>>         return 0;
>>>>>     }
>>>>>
>>>>>     len--;
>>>>>
>>>>>     for (p = url; *p; p++) {
>>>>>         if (x >= len) {
>>>>>             break;
>>>>>         }
>>>>>         if (*p < ' ' || *p > '~' || strchr(urlunsafe, *p)) {
>>>>>             if ((x + 3) > len) {
>>>>>                 break;
>>>>>             }
>>>>>             buf[x++] = '%';
>>>>>             buf[x++] = hex[(*p >> 4) & 0x0f];
>>>>>             buf[x++] = hex[*p & 0x0f];
>>>>>         } else {
>>>>>             buf[x++] = *p;
>>>>>         }
>>>>>     }
>>>>>     buf[x] = '\0';
>>>>>
>>>>>     return buf;
>>>>>
>>>>> }
>>>>>
>>>>> On 17/05/2015 08:18, Mitchelle Johnson wrote:
>>>>>
>>>>>   Thanks Sergey,
>>>>>  Could you please tell me how to apply changes in off #192 pull
>>>>> request?
>>>>>
>>>>>  Thanks,
>>>>>  Mitchelle
>>>>>
>>>>> On Sun, May 17, 2015 at 4:22 PM, Sergey Safarov <s.safarov at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Mitchelle manually or using patch utility apply changes in off #192
>>>>>> pull request.
>>>>>>
>>>>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/192/diff
>>>>>>
>>>>>> On Sun, May 17, 2015 at 10:57 AM, Mitchelle Johnson <
>>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>>
>>>>>>>  I am sorry, I am not able to understand to how use the link
>>>>>>> provided by you to resolve my issue...could you please explain me the
>>>>>>> process in detail.
>>>>>>>
>>>>>>>  Thanks,
>>>>>>>  Mitchelle
>>>>>>>
>>>>>>> On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov <s.safarov at gmail.com
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> See https://freeswitch.org/jira/browse/FS-7258
>>>>>>>>
>>>>>>>> On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <
>>>>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>>>>
>>>>>>>>>     Hi,
>>>>>>>>>  When I am using the web server to handle xml CDR's the xml file
>>>>>>>>> which it sends gives an error.
>>>>>>>>>  The error being:
>>>>>>>>>
>>>>>>>>> XML Parsing Error: not well-formed
>>>>>>>>> Location: http://www.w3schools.com/xml/xml_validator.asp
>>>>>>>>> Line Number 138, Column 25:
>>>>>>>>>     <sip_full_from><sip:1001 at 10.0.0.8
>>>>>>>>> >;tag=661a086d</sip_full_from>
>>>>>>>>> ------------------------^
>>>>>>>>>
>>>>>>>>>  Please help me resolve this.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>  Thanks,
>>>>>>>>>  Mitchelle
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/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
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 mitchelle.bit at gmail.com  Mon May 18 20:56:23 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Mon, 18 May 2015 22:26:23 +0530
Subject: [Freeswitch-users] billsec tag of XML CDR
Message-ID: <CANBm6SyW8Sn7quZax7cQcSOdkuEdBZPCs81nuq1aMT9qu5JaNQ@mail.gmail.com>

Hi,

I am using XML CDR and updating my database dynamically with it. But the
'billsec' tag of the XML CDR works strangely, it shows some value even if
the call is not answered.

Please help.

Thanks,
Mitchelle
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From brian at freeswitch.org  Mon May 18 20:59:25 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 May 2015 11:59:25 -0500
Subject: [Freeswitch-users] billsec tag of XML CDR
In-Reply-To: <CANBm6SyW8Sn7quZax7cQcSOdkuEdBZPCs81nuq1aMT9qu5JaNQ@mail.gmail.com>
References: <CANBm6SyW8Sn7quZax7cQcSOdkuEdBZPCs81nuq1aMT9qu5JaNQ@mail.gmail.com>
Message-ID: <CAEJMVkDacx8VDyghdCeHPvRDcEHVoh3GB+sUhZ0zMVhTv2xBtw@mail.gmail.com>

I can't see how thats possible, unless the call was actually answered,
Happen to have logs?

On Mon, May 18, 2015 at 11:56 AM, Mitchelle Johnson <mitchelle.bit at gmail.com
> wrote:

> Hi,
>
> I am using XML CDR and updating my database dynamically with it. But the
> 'billsec' tag of the XML CDR works strangely, it shows some value even if
> the call is not answered.
>
> Please help.
>
> Thanks,
> Mitchelle
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 mitchelle.bit at gmail.com  Mon May 18 21:05:29 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Mon, 18 May 2015 22:35:29 +0530
Subject: [Freeswitch-users] billsec tag of XML CDR
In-Reply-To: <CAEJMVkDacx8VDyghdCeHPvRDcEHVoh3GB+sUhZ0zMVhTv2xBtw@mail.gmail.com>
References: <CANBm6SyW8Sn7quZax7cQcSOdkuEdBZPCs81nuq1aMT9qu5JaNQ@mail.gmail.com>
	<CAEJMVkDacx8VDyghdCeHPvRDcEHVoh3GB+sUhZ0zMVhTv2xBtw@mail.gmail.com>
Message-ID: <CANBm6SwOsP+h5+-YTV-tiFTZ3oon4UaAuPXnJ1ZGZrMu3Ht9Gw@mail.gmail.com>

Hi Brian,

Shall I paste the FS console logs?

Thanks,
Mitchelle

On Mon, May 18, 2015 at 10:29 PM, Brian West <brian at freeswitch.org> wrote:

> I can't see how thats possible, unless the call was actually answered,
> Happen to have logs?
>
> On Mon, May 18, 2015 at 11:56 AM, Mitchelle Johnson <
> mitchelle.bit at gmail.com> wrote:
>
>> Hi,
>>
>> I am using XML CDR and updating my database dynamically with it. But the
>> 'billsec' tag of the XML CDR works strangely, it shows some value even if
>> the call is not answered.
>>
>> Please help.
>>
>> Thanks,
>> Mitchelle
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
> <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon May 18 21:08:03 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 May 2015 12:08:03 -0500
Subject: [Freeswitch-users] billsec tag of XML CDR
In-Reply-To: <CANBm6SwOsP+h5+-YTV-tiFTZ3oon4UaAuPXnJ1ZGZrMu3Ht9Gw@mail.gmail.com>
References: <CANBm6SyW8Sn7quZax7cQcSOdkuEdBZPCs81nuq1aMT9qu5JaNQ@mail.gmail.com>
	<CAEJMVkDacx8VDyghdCeHPvRDcEHVoh3GB+sUhZ0zMVhTv2xBtw@mail.gmail.com>
	<CANBm6SwOsP+h5+-YTV-tiFTZ3oon4UaAuPXnJ1ZGZrMu3Ht9Gw@mail.gmail.com>
Message-ID: <CAEJMVkA6-1mbwVpYrX_vdQMRhU0JE5Fp1is3WoWp8AG-A5iV6w@mail.gmail.com>

Yes please.

On Mon, May 18, 2015 at 12:05 PM, Mitchelle Johnson <mitchelle.bit at gmail.com
> wrote:

> Hi Brian,
>
> Shall I paste the FS console logs?
>
> Thanks,
> Mitchelle
>
> On Mon, May 18, 2015 at 10:29 PM, Brian West <brian at freeswitch.org> wrote:
>
>> I can't see how thats possible, unless the call was actually answered,
>> Happen to have logs?
>>
>> On Mon, May 18, 2015 at 11:56 AM, Mitchelle Johnson <
>> mitchelle.bit at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I am using XML CDR and updating my database dynamically with it. But the
>>> 'billsec' tag of the XML CDR works strangely, it shows some value even if
>>> the call is not answered.
>>>
>>> Please help.
>>>
>>> Thanks,
>>> Mitchelle
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>> <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 mitchelle.bit at gmail.com  Mon May 18 21:48:02 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Mon, 18 May 2015 23:18:02 +0530
Subject: [Freeswitch-users] billsec tag of XML CDR
In-Reply-To: <CAEJMVkA6-1mbwVpYrX_vdQMRhU0JE5Fp1is3WoWp8AG-A5iV6w@mail.gmail.com>
References: <CANBm6SyW8Sn7quZax7cQcSOdkuEdBZPCs81nuq1aMT9qu5JaNQ@mail.gmail.com>
	<CAEJMVkDacx8VDyghdCeHPvRDcEHVoh3GB+sUhZ0zMVhTv2xBtw@mail.gmail.com>
	<CANBm6SwOsP+h5+-YTV-tiFTZ3oon4UaAuPXnJ1ZGZrMu3Ht9Gw@mail.gmail.com>
	<CAEJMVkA6-1mbwVpYrX_vdQMRhU0JE5Fp1is3WoWp8AG-A5iV6w@mail.gmail.com>
Message-ID: <CANBm6SyNM=CbYLFaOpDtgLPOnF9G2RV3kdxAgJsr4FRpcRMk1Q@mail.gmail.com>

Hi Brian,

Please find the pastebin link http://pastebin.com/JhUyYSGb

Thanks,
Mitchelle

On Mon, May 18, 2015 at 10:38 PM, Brian West <brian at freeswitch.org> wrote:

> Yes please.
>
> On Mon, May 18, 2015 at 12:05 PM, Mitchelle Johnson <
> mitchelle.bit at gmail.com> wrote:
>
>> Hi Brian,
>>
>> Shall I paste the FS console logs?
>>
>> Thanks,
>> Mitchelle
>>
>> On Mon, May 18, 2015 at 10:29 PM, Brian West <brian at freeswitch.org>
>> wrote:
>>
>>> I can't see how thats possible, unless the call was actually answered,
>>> Happen to have logs?
>>>
>>> On Mon, May 18, 2015 at 11:56 AM, Mitchelle Johnson <
>>> mitchelle.bit at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I am using XML CDR and updating my database dynamically with it. But
>>>> the 'billsec' tag of the XML CDR works strangely, it shows some value even
>>>> if the call is not answered.
>>>>
>>>> Please help.
>>>>
>>>> Thanks,
>>>> Mitchelle
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>>> <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 mitchelle.bit at gmail.com  Mon May 18 22:01:16 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Mon, 18 May 2015 23:31:16 +0530
Subject: [Freeswitch-users] [XML parsing error]
In-Reply-To: <CANBm6Sym1Ci1bcPjE9NzYWBzgiPf2nW2OY=QhXSC6juom6x-Pg@mail.gmail.com>
References: <CANBm6Sx6YUsR90dYrsEsJMMmmrRYBrc+2=yQYYMnt2zc-RS-Ug@mail.gmail.com>
	<CAHtxdDfZBQ-PQ7XC6x2DUaE+hfBL8b298fCktSzUqBonPrUAqg@mail.gmail.com>
	<CANBm6Sy3C_qiCbag=tazCiXexWw0mZOHhpfEr3kWXcYwhzj6jw@mail.gmail.com>
	<CAHtxdDccdRxC=AuU_kpwAthhvT4wmR8foVEVve1hT3ZL=A9xbQ@mail.gmail.com>
	<CANBm6Sze77vWO+GxO6KD22mii9JSvgVVmj+_HxA9n1FB47_CaQ@mail.gmail.com>
	<55589CD6.4060303@uol.com.br>
	<CANBm6SyXEfkyFkReYkrB+=LKBkYZv5-0hWuHqWBY-gMbMWLYgQ@mail.gmail.com>
	<555A0796.5060503@uol.com.br>
	<CANBm6SzYgjxboCahh1OSNXU=iHZYwmxMhZUrm8UHxdh14NrSow@mail.gmail.com>
	<CAEJMVkBojvhvNXcf3vwmxsqhQZ6uQyp9e=uaWFz5KMvPJ6Rb5Q@mail.gmail.com>
	<CANBm6Sx00Lzu=1ZZ_2Y1DsfTCBws06nZg0Lh5exo=DdkBQpKTw@mail.gmail.com>
	<CANBm6Sym1Ci1bcPjE9NzYWBzgiPf2nW2OY=QhXSC6juom6x-Pg@mail.gmail.com>
Message-ID: <CANBm6SwoPyG5kt12_bdZfnphgRD25kp1e1fkjxneB=FV5=SRLQ@mail.gmail.com>

Hi Saumar,

I still get the same error even after making the changes suggested by you.

Thanks,
Mitchelle

On Mon, May 18, 2015 at 10:21 PM, Mitchelle Johnson <mitchelle.bit at gmail.com
> wrote:

> Hi,
>
> I will generate a new thread on billsec tag of xml cdr.
>
> Thanks,
> Mitchelle
>
> On Mon, May 18, 2015 at 10:20 PM, Mitchelle Johnson <
> mitchelle.bit at gmail.com> wrote:
>
>> Hi Brian,
>>
>> I also wanted to confirm about the 'billsec' in the xml cdr. It behaves
>> abnormally. Even if the call is not answered it shows some value inspite of
>> showing 0.
>>
>> Please help.
>>
>> Thanks,
>> Mitchelle
>>
>>
>> On Mon, May 18, 2015 at 10:15 PM, Brian West <brian at freeswitch.org>
>> wrote:
>>
>>> The issue is already reported, and has a patch pending testing.
>>>
>>> https://freeswitch.org/jira/browse/FS-7258
>>>
>>> That patch needs some testing!  It will get merged and fix this issue
>>> but not without more testing.
>>>
>>> On Mon, May 18, 2015 at 11:40 AM, Mitchelle Johnson <
>>> mitchelle.bit at gmail.com> wrote:
>>>
>>>> Hi Saumar,
>>>>
>>>> Thanks a lot for your help. I will do as you have said and would report
>>>> if the issue is not resolved.
>>>>
>>>> Regards,
>>>> Mitchelle
>>>>
>>>> On Mon, May 18, 2015 at 9:09 PM, Saumar Hajjar <saumar at uol.com.br>
>>>> wrote:
>>>>
>>>>>  The file is  "src/switch_utils.c"
>>>>> Edit this file, replace the function "switch_url_encode" and rebuild.
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Installation
>>>>>
>>>>>
>>>>>
>>>>> On 18/05/2015 02:37, Mitchelle Johnson wrote:
>>>>>
>>>>>   Hi Saumar,
>>>>>  Could you please tell me where to incorporate the code snippet that
>>>>> you posted? It will be a lot of help if you could tell me in detail as to
>>>>> what file to make changes in and the location of the file, etc..
>>>>>
>>>>>  Thanks,
>>>>>  Mitchelle
>>>>>
>>>>> On Sun, May 17, 2015 at 7:21 PM, Saumar Hajjar <saumar at uol.com.br>
>>>>> wrote:
>>>>>
>>>>>>  Hi Mitchelle,
>>>>>>
>>>>>> I've also faced this issue when I started developing with 1.4 release
>>>>>> - at the time I gave the master branch a try - and then got lucky.
>>>>>> Shortly after that, I updated my setup to a more recent master and
>>>>>> xml cdrs got broken again.
>>>>>>
>>>>>> broken 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221
>>>>>> 2015-03-12 18:55:23Z 64bit)
>>>>>> works 1.5.15b+git~20150117T062211Z~46cf8a4dce~64bit (git 46cf8a4
>>>>>> 2015-01-17 06:22:11Z 64bit)
>>>>>> broken 1.5.15b+git~20150421T235828Z~a4d877c189~64bit (git a4d877c
>>>>>> 2015-04-21 23:58:28Z 64bit)
>>>>>>
>>>>>> Below you'll find what I'm using in all versions I have installed now:
>>>>>>
>>>>>> // switch_utils.c
>>>>>> SWITCH_DECLARE(char *) switch_url_encode(const char *url, char *buf,
>>>>>> size_t len)
>>>>>> {
>>>>>>     const char *p;
>>>>>>     size_t x = 0;
>>>>>>     const char urlunsafe[] = "\r\n \"#%&+:;<=>?@[\\]^`{|}";
>>>>>>     const char hex[] = "0123456789ABCDEF";
>>>>>>
>>>>>>     if (!buf) {
>>>>>>         return 0;
>>>>>>     }
>>>>>>
>>>>>>     if (!url) {
>>>>>>         return 0;
>>>>>>     }
>>>>>>
>>>>>>     len--;
>>>>>>
>>>>>>     for (p = url; *p; p++) {
>>>>>>         if (x >= len) {
>>>>>>             break;
>>>>>>         }
>>>>>>         if (*p < ' ' || *p > '~' || strchr(urlunsafe, *p)) {
>>>>>>             if ((x + 3) > len) {
>>>>>>                 break;
>>>>>>             }
>>>>>>             buf[x++] = '%';
>>>>>>             buf[x++] = hex[(*p >> 4) & 0x0f];
>>>>>>             buf[x++] = hex[*p & 0x0f];
>>>>>>         } else {
>>>>>>             buf[x++] = *p;
>>>>>>         }
>>>>>>     }
>>>>>>     buf[x] = '\0';
>>>>>>
>>>>>>     return buf;
>>>>>>
>>>>>> }
>>>>>>
>>>>>> On 17/05/2015 08:18, Mitchelle Johnson wrote:
>>>>>>
>>>>>>   Thanks Sergey,
>>>>>>  Could you please tell me how to apply changes in off #192 pull
>>>>>> request?
>>>>>>
>>>>>>  Thanks,
>>>>>>  Mitchelle
>>>>>>
>>>>>> On Sun, May 17, 2015 at 4:22 PM, Sergey Safarov <s.safarov at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Mitchelle manually or using patch utility apply changes in off #192
>>>>>>> pull request.
>>>>>>>
>>>>>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/192/diff
>>>>>>>
>>>>>>> On Sun, May 17, 2015 at 10:57 AM, Mitchelle Johnson <
>>>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>>>
>>>>>>>>  I am sorry, I am not able to understand to how use the link
>>>>>>>> provided by you to resolve my issue...could you please explain me the
>>>>>>>> process in detail.
>>>>>>>>
>>>>>>>>  Thanks,
>>>>>>>>  Mitchelle
>>>>>>>>
>>>>>>>> On Sat, May 16, 2015 at 5:50 PM, Sergey Safarov <
>>>>>>>> s.safarov at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> See https://freeswitch.org/jira/browse/FS-7258
>>>>>>>>>
>>>>>>>>> On Sat, May 16, 2015 at 10:44 AM, Mitchelle Johnson <
>>>>>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>>     Hi,
>>>>>>>>>>  When I am using the web server to handle xml CDR's the xml file
>>>>>>>>>> which it sends gives an error.
>>>>>>>>>>  The error being:
>>>>>>>>>>
>>>>>>>>>> XML Parsing Error: not well-formed
>>>>>>>>>> Location: http://www.w3schools.com/xml/xml_validator.asp
>>>>>>>>>> Line Number 138, Column 25:
>>>>>>>>>>     <sip_full_from><sip:1001 at 10.0.0.8
>>>>>>>>>> >;tag=661a086d</sip_full_from>
>>>>>>>>>> ------------------------^
>>>>>>>>>>
>>>>>>>>>>  Please help me resolve this.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>  Thanks,
>>>>>>>>>>  Mitchelle
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/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
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>>> <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From brian at freeswitch.org  Mon May 18 22:27:47 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 May 2015 13:27:47 -0500
Subject: [Freeswitch-users] billsec tag of XML CDR
In-Reply-To: <CANBm6SyNM=CbYLFaOpDtgLPOnF9G2RV3kdxAgJsr4FRpcRMk1Q@mail.gmail.com>
References: <CANBm6SyW8Sn7quZax7cQcSOdkuEdBZPCs81nuq1aMT9qu5JaNQ@mail.gmail.com>
	<CAEJMVkDacx8VDyghdCeHPvRDcEHVoh3GB+sUhZ0zMVhTv2xBtw@mail.gmail.com>
	<CANBm6SwOsP+h5+-YTV-tiFTZ3oon4UaAuPXnJ1ZGZrMu3Ht9Gw@mail.gmail.com>
	<CAEJMVkA6-1mbwVpYrX_vdQMRhU0JE5Fp1is3WoWp8AG-A5iV6w@mail.gmail.com>
	<CANBm6SyNM=CbYLFaOpDtgLPOnF9G2RV3kdxAgJsr4FRpcRMk1Q@mail.gmail.com>
Message-ID: <CAEJMVkCif1AnuH=01Jc=jBXGcdhXW84nJmcy7na_o6pM+Ld70w@mail.gmail.com>

mod_dptools.c:1208 Channel [FreeTDM/1:17/016] has been answered

Also set the debug level in vars.xml to debug, you're missing tons of info
that could probably help you.  This is probably the ISDN doing a CONNECT
and answering the channel to pay the busy signal.

On Mon, May 18, 2015 at 12:48 PM, Mitchelle Johnson <mitchelle.bit at gmail.com
> wrote:

> Hi Brian,
>
> Please find the pastebin link http://pastebin.com/JhUyYSGb
>
> Thanks,
> Mitchelle
>
> On Mon, May 18, 2015 at 10:38 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Yes please.
>>
>> On Mon, May 18, 2015 at 12:05 PM, Mitchelle Johnson <
>> mitchelle.bit at gmail.com> wrote:
>>
>>> Hi Brian,
>>>
>>> Shall I paste the FS console logs?
>>>
>>> Thanks,
>>> Mitchelle
>>>
>>> On Mon, May 18, 2015 at 10:29 PM, Brian West <brian at freeswitch.org>
>>> wrote:
>>>
>>>> I can't see how thats possible, unless the call was actually answered,
>>>> Happen to have logs?
>>>>
>>>> On Mon, May 18, 2015 at 11:56 AM, Mitchelle Johnson <
>>>> mitchelle.bit at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> I am using XML CDR and updating my database dynamically with it. But
>>>>> the 'billsec' tag of the XML CDR works strangely, it shows some value even
>>>>> if the call is not answered.
>>>>>
>>>>> Please help.
>>>>>
>>>>> Thanks,
>>>>> Mitchelle
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>> ClueCon 2015 Call for Speakers
>>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>>>> <https://www.reddit.com/r/freeswitch>
>>>>
>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>> <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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  Mon May 18 22:35:39 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 May 2015 13:35:39 -0500
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <CAHtxdDc7URbEp0F8conEycByR32iQckqa-o3ekHAjuUPqXxB7g@mail.gmail.com>
References: <1251442930.20150513084828@seznam.cz>
	<D178B500.9DA9B%krice@freeswitch.org>
	<CAHtxdDft6M-mF+=+L7jGLxPAD+EzHohYBou4O_f6rsYwCofYqg@mail.gmail.com>
	<1641553624.20150513155908@seznam.cz>
	<CAEJMVkBk-+YMRYGXQnH+Eu_GT571BzKNfrkp4zGDRmVabKDSgg@mail.gmail.com>
	<CAHtxdDc7URbEp0F8conEycByR32iQckqa-o3ekHAjuUPqXxB7g@mail.gmail.com>
Message-ID: <CAEJMVkBFm-3=Y5zu=Rh7njLp5c+J5kPw51hjnxbBGHuu9o=zRg@mail.gmail.com>

The RPM spec files need to be submitted to our JIRA, so we can build them
and sign them with our key.



On Sun, May 17, 2015 at 2:13 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> I has created dependency packages and has placed on
> http://fs-repo.network-engineer.ru/
> A is anybody can help to place this rpms on
> http://files.freeswitch.org/repo/ ?
>
> Sergey Safarov
>
>
> On Wed, May 13, 2015 at 6:21 PM, Brian West <brian at freeswitch.org> wrote:
>
>> I've assigned it to Sergey Safarov, So lets get to busy ;)
>>
>> On Wed, May 13, 2015 at 8:59 AM, Denis Jakovlev <yadenis at seznam.cz>
>> wrote:
>>
>>>  Dobr? den,
>>>
>>> I hope I did it right
>>>
>>> https://freeswitch.org/jira/browse/FS-7553
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>>> <http://mob.tel>. 775-415-382 st?eda 13. kv?tna 2015, 15:34:55, napsal
>>> jste: *
>>>
>>>  I has rpm repo with all required packages to install current FS master
>>> via yum package manager on CentOS 7. All packages ported from fedora repo.
>>> Thinking 1.6 branch can be packaged similarly.
>>> If anybody write Jira tiket and this tiket will be assigned to me, then
>>> I create pull request to automate build dependences on CentOS 7.
>>>
>>> Sergey Safarov
>>>
>>>
>>> On Wed, May 13, 2015 at 4:00 PM, Ken Rice <krice at freeswitch.org> wrote:
>>> There is a massive shift in the number of dependencies for compiling
>>> FreeSWITCH. We are working on getting that dep chain working on non-debian
>>> platforms but that work is not yet complete.
>>>
>>> If you wish to try and report back to the list the steps you took to get
>>> it working on Centos7 so that others may benefit, most of the deps that you
>>> cant get from yum are located at
>>> https://freeswitch.org/stash/projects/SD/
>>>
>>> This may not be an all inclusive list for building on non-debian systems
>>> but it will get you close.
>>>
>>>
>>>
>>>
>>> On 5/13/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz> wrote:
>>>
>>> Dobr? den,
>>>
>>> This version 7, yes. This version is somehow different?
>>>
>>> --
>>> Ken
>>> http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>> irc.freenode.net #freeswitch
>>> Twitter: @FreeSWITCH
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org <http://www.FreeSWITCH.org>
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com <http://www.ClueCon.com>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> 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
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>> <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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  Mon May 18 22:48:53 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 May 2015 13:48:53 -0500
Subject: [Freeswitch-users] unexpected segfault with latest debian and
	libmyodbc.so
In-Reply-To: <DUB122-W425F87363588D683B60CC6D5C50@phx.gbl>
References: <DUB122-W36F85086474E2280B9ED15D5090@phx.gbl>
	<CAJR7f2HCFX6wOTxNStRVO8by4x7HiSUo9zQf-8566SzqqsDFvA@mail.gmail.com>
	<CAKbxfG8+=HVT_D7FqUwdynRa8-8KEvkOVzuPnjF6tnmZipuvQA@mail.gmail.com>
	<DUB122-W720E4C2E410A9EA55250CD5F20@phx.gbl>
	<CAJR7f2FdFFqUEdTpQSN59uzVg_p3nq_kBj5FvDYtM2pUieDeOA@mail.gmail.com>
	<DUB122-W425F87363588D683B60CC6D5C50@phx.gbl>
Message-ID: <CAEJMVkDTEgoa51dny8PiyvGiy6aAcvqG3vsVvJs8jyjezud==g@mail.gmail.com>

Did you try Threading = 0?

On Sun, May 17, 2015 at 9:44 AM, Yuriy Nasida <nasida at live.ru> wrote:

> Guys, unfortunately updating of unixODBC didn't help and I getting
> segfaults again after some quiet period.
>
> Now I have:
> 1) unixODBC 2.3.1
> 2) FreeSWITCH Version 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git
> 507a0f2 2014-12-29 18:59:51Z 64bit)
> 3) Debian GNU/Linux 7.8 (wheezy)
> 4) libmyodbc  5.1.10-2+deb7u1
>
> Please advice
> Thanks
>
>
>
> ------------------------------
> Date: Thu, 2 Apr 2015 16:19:13 -0300
> From: italorossib at gmail.com
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] unexpected segfault with latest debian and
> libmyodbc.so
>
> Great! :)
>
>
> On Thu, Apr 2, 2015 at 2:17 PM, Yuriy Nasida <nasida at live.ru> wrote:
>
> I have updated unixodbc to 2.3 (from source) and looks like it fixed the
> problem. Thanks guys!
>
>
> ------------------------------
> Date: Fri, 27 Mar 2015 17:39:55 -0500
> From: anthony.minessale at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] unexpected segfault with latest debian and
> libmyodbc.so
>
>
> We actually recommend Threading=0 for postgres and mysql for the most part.
>
>
> On Fri, Mar 27, 2015 at 7:48 AM, ?talo Rossi <italorossib at gmail.com>
> wrote:
>
> I have seen a lot of threading issues with mysql + odbc, make sure you're
> using unixodbc >= 2.3.
>
> If you're using an older version you can set Threading = 2 in your
> /etc/odbcinst.ini as a workaround, but this is *not* recommended for
> production/high volume, upgrade as soon as possible.
>
> On Fri, Mar 27, 2015 at 9:41 AM, Yuriy Nasida <nasida at live.ru> wrote:
>
> Hi guys,
>
> I just got unexpected segfault I try to understand if anybody had similar
> problems.
>
> *Mar 26 07:00:19 kernel: [226389.252971] freeswitch[7972]: segfault at 500
> ip 00007fd3a8362252 sp 00007fd34a7d9f70 error 4 in
> libmyodbc.so[7fd3a8340000+3c000]*
>
>
> # lsb_release -a
> Distributor ID:    Debian
> Description:    Debian GNU/Linux 7.8 (wheezy)
> Release:    7.8
> Codename:    wheezy
>
> freeswitch at internal> version
> FreeSWITCH Version 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git
> 507a0f2 2014-12-29 18:59:51Z 64bit)
>
> # apt-cache show libmyodbc
> Package: libmyodbc
> Source: myodbc
> Version: 5.1.10-2+deb7u1
>
> Please advice,
> 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
>
>
>
>
> --
> ?talo Rossi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
> [image: ?????? ???????] sip:888 at conference.freeswitch.org  [image: ??????
> ???????] +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com Official FreeSWITCH Sites
> http://www.freeswitch.org http://confluence.freeswitch.org
> http://www.cluecon.com FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com Official FreeSWITCH Sites
> http://www.freeswitch.org http://confluence.freeswitch.org
> http://www.cluecon.com FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From mike at jerris.com  Mon May 18 22:52:01 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 May 2015 14:52:01 -0400
Subject: [Freeswitch-users] Setting loglevel beyond 7 ?
In-Reply-To: <CY1PR0501MB10982A84F72B060A6AD0C30084C70@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CY1PR0501MB10982A84F72B060A6AD0C30084C70@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <5878DB33-E3AC-4C75-B775-481BEDE60E7A@jerris.com>

https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-debug_level <https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-debug_level>

> On May 15, 2015, at 3:15 PM, Sharath Kumar <Sharath.Kumar at meZocliq.com> wrote:
> 
> All,
>  
> I noticed some of the logs in the code are using SWITCH_LOG_DEBUG1 =101  - SWITCH_LOG_DEBUG10 = 110. Is there a way to enable these logs from the fs_cli  ? I can recompile the ones I want with a lower value but just wanted to avoid that.

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From mike at jerris.com  Mon May 18 22:54:06 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 May 2015 14:54:06 -0400
Subject: [Freeswitch-users] XML parsing error
In-Reply-To: <CANBm6Syqy-f24AH97-xYefPP89BKM0dNkTzjpS_ZN9jC_4kM5w@mail.gmail.com>
References: <CANBm6Syqy-f24AH97-xYefPP89BKM0dNkTzjpS_ZN9jC_4kM5w@mail.gmail.com>
Message-ID: <81FB7E30-F6B6-4551-8272-56A2A958BAF0@jerris.com>

https://freeswitch.org/jira/browse/FS-7258

> On May 16, 2015, at 1:55 AM, Mitchelle Johnson <mitchelle.bit at gmail.com> wrote:
> 
> Hi,
> When I am using the web server to handle xml CDR's the xml file which it sends gives an error.
> The error being:
> 
> XML Parsing Error: not well-formed
> Location: http://www.w3schools.com/xml/xml_validator.asp <http://www.w3schools.com/xml/xml_validator.asp>
> Line Number 138, Column 25:
>     <sip_full_from><sip:1001 at 10.0.0.8 <mailto:sip%3A1001 at 10.0.0.8>>;tag=661a086d</sip_full_from>
> ------------------------^

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From mike at jerris.com  Mon May 18 22:54:51 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 May 2015 14:54:51 -0400
Subject: [Freeswitch-users] ESL python library for python3
In-Reply-To: <CAB6AXdnyX9Zsu73i+C7qtx=OSzkqo4fpTMrK0LiSaAD_8VX7Vg@mail.gmail.com>
References: <CAB6AXdnyX9Zsu73i+C7qtx=OSzkqo4fpTMrK0LiSaAD_8VX7Vg@mail.gmail.com>
Message-ID: <B7CCE472-344E-45E6-81E0-03C81AC52E95@jerris.com>

I don't think we have ever tried it, but we would be happy to take pull requests to fix any issues with it.

> On May 16, 2015, at 10:30 AM, Iqbal Abdullah <iqbal.b.abdullah at gmail.com> wrote:
> 
> Hello,
> 
> I've managed to compile, install and use the python esl library from
> freeswitch 1.4.18, but am now wondering is there a way to get the
> python esl library for python3?
> 
> I've tried changing the Makefiles in libs/esl/python to point to
> python3 and made sure my dev packages are installed, but the
> compilation failed.
> 
> Does the python esl module supports python3?



From mike at jerris.com  Mon May 18 22:57:18 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 May 2015 14:57:18 -0400
Subject: [Freeswitch-users] need correct SDP media examples for all
	variances of g729
In-Reply-To: <CAM2iYy5gKrJEswBMQBKM49oXa3Dj8vrGV-Nn2XmQMCQfhY1Z0g@mail.gmail.com>
References: <CAM2iYy5gKrJEswBMQBKM49oXa3Dj8vrGV-Nn2XmQMCQfhY1Z0g@mail.gmail.com>
Message-ID: <EEA53A70-4B50-4143-B31A-680E647C6486@jerris.com>

some info on this: http://tools.ietf.org/html/draft-muthu-payload-offer-answer-g723-g729-00 <http://tools.ietf.org/html/draft-muthu-payload-offer-answer-g723-g729-00>

> On May 17, 2015, at 4:17 AM, Dmitry Saratsky <simpot at gmail.com> wrote:
> 
> Hi all,
> 
> Can you please show me CORRECT SDP media examples for all possible G.729  annexes (need only G.729, G.729A, G.729B and G.729AB).
> 
> How should correct SDP look like for each case of the above including fmtp parameters and maybe other parameters?

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From iqbal.b.abdullah at gmail.com  Mon May 18 23:00:17 2015
From: iqbal.b.abdullah at gmail.com (Iqbal Abdullah)
Date: Tue, 19 May 2015 04:00:17 +0900
Subject: [Freeswitch-users] ESL python library for python3
In-Reply-To: <B7CCE472-344E-45E6-81E0-03C81AC52E95@jerris.com>
References: <CAB6AXdnyX9Zsu73i+C7qtx=OSzkqo4fpTMrK0LiSaAD_8VX7Vg@mail.gmail.com>
	<B7CCE472-344E-45E6-81E0-03C81AC52E95@jerris.com>
Message-ID: <CAB6AXdm5K=hcnS3wqpnCQU03CLakSSbE3iDdPw3R9Q9kt=JCEA@mail.gmail.com>

Michael,

I have managed to build the library againts py3 and will make a pull
request later.
On May 18, 2015 11:57 AM, "Michael Jerris" <mike at jerris.com> wrote:

> I don't think we have ever tried it, but we would be happy to take pull
> requests to fix any issues with it.
>
> > On May 16, 2015, at 10:30 AM, Iqbal Abdullah <iqbal.b.abdullah at gmail.com>
> wrote:
> >
> > Hello,
> >
> > I've managed to compile, install and use the python esl library from
> > freeswitch 1.4.18, but am now wondering is there a way to get the
> > python esl library for python3?
> >
> > I've tried changing the Makefiles in libs/esl/python to point to
> > python3 and made sure my dev packages are installed, but the
> > compilation failed.
> >
> > Does the python esl module supports python3?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Mon May 18 23:05:14 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 May 2015 15:05:14 -0400
Subject: [Freeswitch-users] ESL python library for python3
In-Reply-To: <CAB6AXdm5K=hcnS3wqpnCQU03CLakSSbE3iDdPw3R9Q9kt=JCEA@mail.gmail.com>
References: <CAB6AXdnyX9Zsu73i+C7qtx=OSzkqo4fpTMrK0LiSaAD_8VX7Vg@mail.gmail.com>
	<B7CCE472-344E-45E6-81E0-03C81AC52E95@jerris.com>
	<CAB6AXdm5K=hcnS3wqpnCQU03CLakSSbE3iDdPw3R9Q9kt=JCEA@mail.gmail.com>
Message-ID: <A3B6B08D-67BF-48DC-9581-3D1C36F10A2C@jerris.com>

we will need some way that uses configure to detect the right python, and builds against old and new python.

> On May 18, 2015, at 3:00 PM, Iqbal Abdullah <iqbal.b.abdullah at gmail.com> wrote:
> 
> Michael,
> 
> I have managed to build the library againts py3 and will make a pull request later.
> 
> On May 18, 2015 11:57 AM, "Michael Jerris" <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> I don't think we have ever tried it, but we would be happy to take pull requests to fix any issues with it.
> 
> > On May 16, 2015, at 10:30 AM, Iqbal Abdullah <iqbal.b.abdullah at gmail.com <mailto:iqbal.b.abdullah at gmail.com>> wrote:
> >
> > Hello,
> >
> > I've managed to compile, install and use the python esl library from
> > freeswitch 1.4.18, but am now wondering is there a way to get the
> > python esl library for python3?
> >
> > I've tried changing the Makefiles in libs/esl/python to point to
> > python3 and made sure my dev packages are installed, but the
> > compilation failed.
> >
> > Does the python esl module supports python3?
> 

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From iqbal.b.abdullah at gmail.com  Mon May 18 23:15:30 2015
From: iqbal.b.abdullah at gmail.com (Iqbal Abdullah)
Date: Mon, 18 May 2015 12:15:30 -0700
Subject: [Freeswitch-users] ESL python library for python3
In-Reply-To: <A3B6B08D-67BF-48DC-9581-3D1C36F10A2C@jerris.com>
References: <CAB6AXdnyX9Zsu73i+C7qtx=OSzkqo4fpTMrK0LiSaAD_8VX7Vg@mail.gmail.com>
	<B7CCE472-344E-45E6-81E0-03C81AC52E95@jerris.com>
	<CAB6AXdm5K=hcnS3wqpnCQU03CLakSSbE3iDdPw3R9Q9kt=JCEA@mail.gmail.com>
	<A3B6B08D-67BF-48DC-9581-3D1C36F10A2C@jerris.com>
Message-ID: <CAB6AXd=04K0JSBBNx+AXenqyjW1348SBXu7j5quO60ug4FkVUA@mail.gmail.com>

If you run configure while in a py3 environment (i.e via virtualenv)
the configure will correctly detect the py3 variables.

I made the build against py3 on 1.4.18, and changes need to be done at
3 places though, after running configure above.

1) libs/esl/python/python-config because as it is this code is not py3
compliant. You can easily use 2to3 for this.
2) Make a slight change to the Makefile in libs/esl/python to pass the
-py3 parameter to swig instead, so it produces a py3 compliant
esl_wrap.cpp

--- a/libs/esl/python/Makefile
+++ b/libs/esl/python/Makefile
@@ -5,7 +5,7 @@ SITE_DIR=$(DESTDIR)/`python -c "from
distutils.sysconfig import get_python_lib;
 all: _ESL.so

 esl_wrap.cpp:
-       swig -module ESL -classic -python -c++ -DMULTIPLICITY -threads
-I../src/include -o esl_wrap.cpp ../ESL.i
+       swig -module ESL -py3 -python -c++ -DMULTIPLICITY -threads
-I../src/include -o esl_wrap.cpp ../ESL.i

 esl_wrap.o: esl_wrap.cpp
        $(CXX) $(CXX_CFLAGS) $(CXXFLAGS) $(LOCAL_CFLAGS) -c
esl_wrap.cpp -o esl_wrap.o

3) Finally, I'm not sure if this is needed for all environment (mine is
Ubuntu 14.04 LTS x86_64) but I made a slight change to the
libs/esl/python/python-config after putting it through 2to3:

 elif opt in ('--libs', '--ldflags'):
     libs = getvar('LIBS').split() + getvar('SYSLIBS').split()
-    libs.append('-lpython'+pyver)
+    if pyver[0] == "3":
+        libs.append('-lpython'+pyver+'m')
+    else:
+        libs.append('-lpython'+pyver)

In my environment, the python3 library is called libpython3.4m, so I
had to append an "m" at the end of the library name, i.e:


On 18 May 2015 at 12:05, Michael Jerris <mike at jerris.com> wrote:
> we will need some way that uses configure to detect the right python, and
> builds against old and new python.
>
> On May 18, 2015, at 3:00 PM, Iqbal Abdullah <iqbal.b.abdullah at gmail.com>
> wrote:
>
> Michael,
>
> I have managed to build the library againts py3 and will make a pull request
> later.
>
> On May 18, 2015 11:57 AM, "Michael Jerris" <mike at jerris.com> wrote:
>>
>> I don't think we have ever tried it, but we would be happy to take pull
>> requests to fix any issues with it.
>>
>> > On May 16, 2015, at 10:30 AM, Iqbal Abdullah
>> > <iqbal.b.abdullah at gmail.com> wrote:
>> >
>> > Hello,
>> >
>> > I've managed to compile, install and use the python esl library from
>> > freeswitch 1.4.18, but am now wondering is there a way to get the
>> > python esl library for python3?
>> >
>> > I've tried changing the Makefiles in libs/esl/python to point to
>> > python3 and made sure my dev packages are installed, but the
>> > compilation failed.
>> >
>> > Does the python esl module supports python3?
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon May 18 23:31:54 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 May 2015 15:31:54 -0400
Subject: [Freeswitch-users] ESL python library for python3
In-Reply-To: <CAB6AXd=04K0JSBBNx+AXenqyjW1348SBXu7j5quO60ug4FkVUA@mail.gmail.com>
References: <CAB6AXdnyX9Zsu73i+C7qtx=OSzkqo4fpTMrK0LiSaAD_8VX7Vg@mail.gmail.com>
	<B7CCE472-344E-45E6-81E0-03C81AC52E95@jerris.com>
	<CAB6AXdm5K=hcnS3wqpnCQU03CLakSSbE3iDdPw3R9Q9kt=JCEA@mail.gmail.com>
	<A3B6B08D-67BF-48DC-9581-3D1C36F10A2C@jerris.com>
	<CAB6AXd=04K0JSBBNx+AXenqyjW1348SBXu7j5quO60ug4FkVUA@mail.gmail.com>
Message-ID: <A9B90B0D-873D-453E-9F48-688AC6A0EDAE@jerris.com>

The changes will have to be done in a way where it will detect which version to build against in configure, and will still build against either, without requiring manual changes.  Please submit a pull request with these changes when ready and in in a way that will not break older python.

> On May 18, 2015, at 3:15 PM, Iqbal Abdullah <iqbal.b.abdullah at gmail.com> wrote:
> 
> If you run configure while in a py3 environment (i.e via virtualenv)
> the configure will correctly detect the py3 variables.
> 
> I made the build against py3 on 1.4.18, and changes need to be done at
> 3 places though, after running configure above.
> 
> 1) libs/esl/python/python-config because as it is this code is not py3
> compliant. You can easily use 2to3 for this.
> 2) Make a slight change to the Makefile in libs/esl/python to pass the
> -py3 parameter to swig instead, so it produces a py3 compliant
> esl_wrap.cpp
> 
> --- a/libs/esl/python/Makefile
> +++ b/libs/esl/python/Makefile
> @@ -5,7 +5,7 @@ SITE_DIR=$(DESTDIR)/`python -c "from
> distutils.sysconfig import get_python_lib;
> all: _ESL.so
> 
> esl_wrap.cpp:
> -       swig -module ESL -classic -python -c++ -DMULTIPLICITY -threads
> -I../src/include -o esl_wrap.cpp ../ESL.i
> +       swig -module ESL -py3 -python -c++ -DMULTIPLICITY -threads
> -I../src/include -o esl_wrap.cpp ../ESL.i
> 
> esl_wrap.o: esl_wrap.cpp
>        $(CXX) $(CXX_CFLAGS) $(CXXFLAGS) $(LOCAL_CFLAGS) -c
> esl_wrap.cpp -o esl_wrap.o
> 
> 3) Finally, I'm not sure if this is needed for all environment (mine is
> Ubuntu 14.04 LTS x86_64) but I made a slight change to the
> libs/esl/python/python-config after putting it through 2to3:
> 
> elif opt in ('--libs', '--ldflags'):
>     libs = getvar('LIBS').split() + getvar('SYSLIBS').split()
> -    libs.append('-lpython'+pyver)
> +    if pyver[0] == "3":
> +        libs.append('-lpython'+pyver+'m')
> +    else:
> +        libs.append('-lpython'+pyver)
> 
> In my environment, the python3 library is called libpython3.4m, so I
> had to append an "m" at the end of the library name, i.e:
> 
> 
> On 18 May 2015 at 12:05, Michael Jerris <mike at jerris.com> wrote:
>> we will need some way that uses configure to detect the right python, and
>> builds against old and new python.
>> 
>> On May 18, 2015, at 3:00 PM, Iqbal Abdullah <iqbal.b.abdullah at gmail.com>
>> wrote:
>> 
>> Michael,
>> 
>> I have managed to build the library againts py3 and will make a pull request
>> later.
>> 
>> On May 18, 2015 11:57 AM, "Michael Jerris" <mike at jerris.com> wrote:
>>> 
>>> I don't think we have ever tried it, but we would be happy to take pull
>>> requests to fix any issues with it.
>>> 
>>>> On May 16, 2015, at 10:30 AM, Iqbal Abdullah
>>>> <iqbal.b.abdullah at gmail.com> wrote:
>>>> 
>>>> Hello,
>>>> 
>>>> I've managed to compile, install and use the python esl library from
>>>> freeswitch 1.4.18, but am now wondering is there a way to get the
>>>> python esl library for python3?
>>>> 
>>>> I've tried changing the Makefiles in libs/esl/python to point to
>>>> python3 and made sure my dev packages are installed, but the
>>>> compilation failed.
>>>> 
>>>> Does the python esl module supports python3?
>>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ben at langfeld.co.uk  Mon May 18 23:54:08 2015
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Mon, 18 May 2015 16:54:08 -0300
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <D178B500.9DA9B%krice@freeswitch.org>
References: <1251442930.20150513084828@seznam.cz>
	<D178B500.9DA9B%krice@freeswitch.org>
Message-ID: <CAAyX+KF_LJ07ocQqqGmmudS8cnfAjWRzujsrHtkPUwNdf2v7rA@mail.gmail.com>

On that point, Ken, I wonder how you personally / the FS core team in
general views the proposals in
http://0pointer.net/blog/revisiting-how-we-put-together-linux-systems.html.
Had you seen that?

On 13 May 2015 at 10:00, Ken Rice <krice at freeswitch.org> wrote:

>  There is a massive shift in the number of dependencies for compiling
> FreeSWITCH. We are working on getting that dep chain working on non-debian
> platforms but that work is not yet complete.
>
> If you wish to try and report back to the list the steps you took to get
> it working on Centos7 so that others may benefit, most of the deps that you
> cant get from yum are located at https://freeswitch.org/stash/projects/SD/
>
> This may not be an all inclusive list for building on non-debian systems
> but it will get you close.
>
>
>
>
> On 5/13/15, 1:48 AM, "Denis Jakovlev" <yadenis at seznam.cz> wrote:
>
> Dobr? den,
>
> This version 7, yes. This version is somehow different?
>
>
> --
> Ken
>
>
>
> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Tue May 19 00:24:14 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 18 May 2015 21:24:14 +0100
Subject: [Freeswitch-users] need correct SDP media examples for all
 variances of g729
In-Reply-To: <EEA53A70-4B50-4143-B31A-680E647C6486@jerris.com>
References: <CAM2iYy5gKrJEswBMQBKM49oXa3Dj8vrGV-Nn2XmQMCQfhY1Z0g@mail.gmail.com>
	<EEA53A70-4B50-4143-B31A-680E647C6486@jerris.com>
Message-ID: <CAFiqYu=1B60DO7c+UgTV5EP5woEzx7xB3ZZN086ra80uvJz-jA@mail.gmail.com>

Sample:
  m=audio 49170 RTP/AVP 18
  a=rtpmap:18 G729/8000
  a=fmtp:18 annexb=yes

m=audio - the numeric type will be present here. The static type assigned
by IANA is 18, which means G729/8000. Another number may be used if a
a=rtpmap line defines it, but very rarely would be.

a=rtpmap is optional (see ${verbose_sdp}) but specifies the codec name
using the 18 code. It's optional since 18 is defined to be G729/8000 unless
overridden.

"a=fmtp:18 annexb=yes" defines Annex B is in use (either G729B or G729AB),
the absence of it means it is not.

Annex A (G729A or G729AB) does not appear at all in the SDP. It's an
alternative method of encoding using less CPU at the expense of quality.
The results can be decoded by any G729 implementation so they do not need
to know G729A was used.

Some phones will send G729A/8000 (Cisco/Linksys) - this is incorrect. I
believe FreeSWITCH handles this special case though.


On 18 May 2015 at 19:57, Michael Jerris <mike at jerris.com> wrote:

> some info on this:
> http://tools.ietf.org/html/draft-muthu-payload-offer-answer-g723-g729-00
>
> On May 17, 2015, at 4:17 AM, Dmitry Saratsky <simpot at gmail.com> wrote:
>
> Hi all,
>
> Can you please show me CORRECT SDP media examples for all possible G.729
> annexes (need only G.729, G.729A, G.729B and G.729AB).
>
> How should correct SDP look like for each case of the above including fmtp
> parameters and maybe other parameters?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From naveen32india at gmail.com  Tue May 19 00:30:17 2015
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Tue, 19 May 2015 02:00:17 +0530
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
Message-ID: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>

Hi,

I am wondering how to send sip signaling message from the fs console for
the particular user/caller.
I found  respond dialplan application to send sip message back to the
caller.
Like

<action application="respond" data="480 Try again later"/>

?Is there any way to send sip message back to the caller. One use case is
call rejection or playing with SIP?


-- 
Thanks & Regards,
Naveen Tamanam
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From mike at jerris.com  Tue May 19 00:49:17 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 May 2015 16:49:17 -0400
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
Message-ID: <DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>

respond is one way, what exact message are you trying to send, and at what point in the call.  There are capabilites to trigger re-invites in some situations, transfer, send notify or info or message.  It depends on what exactly you are trying to do.


> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com> wrote:
> 
> Hi, 
> 
> I am wondering how to send sip signaling message from the fs console for the particular user/caller. 
> I found  respond dialplan application to send sip message back to the caller. 
> Like 
> <action application="respond" data="480 Try again later"/>
> ?Is there any way to send sip message back to the caller. One use case is call rejection or playing with SIP?
> 

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From naveen32india at gmail.com  Tue May 19 00:55:06 2015
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Tue, 19 May 2015 02:25:06 +0530
Subject: [Freeswitch-users] Tone detection How To
Message-ID: <CABsWbRbKSSCUk_Qwx1bDHbsoD5+da1M4OSSouYfZ4ETKXqm1sA@mail.gmail.com>

Hi,

I am using following command for tone detection,

tone_detect <uuid> <key> <tone_spec> [<flags> <timeout> <app> <args>] <hits>

Many times I end with the false detection. In my case  I haven't given all
optional args.
I just gave   tone_detect uuid  key tone_specs. It would be something like,

tone_detect  8582g3b8-f7db-10e4-2ac4-c01885c1ffd3  tone_name frp1,frp2,frp3

Where frp1, frp2 and frp3 are the frequencies of  few peak amplitudes of
the tone.
Those frequencies are detected through audacity by plotting spectrum,
Spectrum specs are,
    Alogorithm: Spectrum,
    Fuction: Hanning window  or Bartlett window
    Size: 512
    Axis: Log Frequencies

Is there anything I am missing. Any suggestions are highly appreciated,
Thanks.




-- 
Thanks & Regards,
Naveen Tamanam
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From naveen32india at gmail.com  Tue May 19 00:59:25 2015
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Tue, 19 May 2015 02:29:25 +0530
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
Message-ID: <CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>

I am trying to do the following,  I would like to reject the call when it
ringing from the fs console.
And second thing is I am pretty much eager to know the way to send
sip(signaling) message manually for a
selected channel from fs console.


On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com> wrote:

> respond is one way, what exact message are you trying to send, and at what
> point in the call.  There are capabilites to trigger re-invites in some
> situations, transfer, send notify or info or message.  It depends on what
> exactly you are trying to do.
>
>
> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com>
> wrote:
>
> Hi,
>
> I am wondering how to send sip signaling message from the fs console for
> the particular user/caller.
> I found  respond dialplan application to send sip message back to the
> caller.
> Like
>
> <action application="respond" data="480 Try again later"/>
>
> ?Is there any way to send sip message back to the caller. One use case is
> call rejection or playing with SIP?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regards,
Naveen Tamanam
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From steveayre at gmail.com  Tue May 19 01:21:09 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 18 May 2015 22:21:09 +0100
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
Message-ID: <CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>

>
>  I would like to reject the call when it ringing from the fs console.


uuid_hangup


On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com> wrote:

> I am trying to do the following,  I would like to reject the call when it
> ringing from the fs console.
> And second thing is I am pretty much eager to know the way to send
> sip(signaling) message manually for a
> selected channel from fs console.
>
>
> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> respond is one way, what exact message are you trying to send, and at
>> what point in the call.  There are capabilites to trigger re-invites in
>> some situations, transfer, send notify or info or message.  It depends on
>> what exactly you are trying to do.
>>
>>
>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com>
>> wrote:
>>
>> Hi,
>>
>> I am wondering how to send sip signaling message from the fs console for
>> the particular user/caller.
>> I found  respond dialplan application to send sip message back to the
>> caller.
>> Like
>>
>> <action application="respond" data="480 Try again later"/>
>>
>> ?Is there any way to send sip message back to the caller. One use case is
>> call rejection or playing with SIP?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Thanks & Regards,
> Naveen Tamanam
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Tue May 19 01:22:50 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 18 May 2015 21:22:50 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) May
	9th-May 16th
Message-ID: <555a582aee3ff_fbfb125f31884746@resque-worker.17.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1EToOuE
FreeSWITCH Week in Review (Master Branch) May 9th-May 16th
Hello, again. This passed week in the FreeSWITCH master branch we had 14 commits. Our feature for this week is the addition of limit backend to mod_mongo. It needs some testing, so go check it out.

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

New features that were added:

FS-7557 [mod_mongo] Add limit backend

The following bugs were squashed:

FS-7529 Fixed an error setting a media bug on G722

FS-7552 [mod_amqp] Fixed a segfault on unload and when no connections were valid

FS-7463 [mod_sofia] Conditionally allow intercept of replaced call-id when processing replaces header

FS-7557 [mod_mongo] Fixed a crash when doing ?limit_usage mongo foo bar?

FS-7545 [mod_opus] Fixed RTP timestamps to prevent un-needed resampling when transcoding

FS-7184 [mod_spandsp] Fixed a fax buffer overflow in t38 on failure condition with some fax machines

FS-7546 [mod_spandsp] Fixed a crash when sending a fax when built using clang compiler

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From john.nash778 at gmail.com  Tue May 19 01:22:54 2015
From: john.nash778 at gmail.com (John Nash)
Date: Tue, 19 May 2015 02:52:54 +0530
Subject: [Freeswitch-users] Playing multiple files in lua script
Message-ID: <CAA4rGPz=PveNCvUpbGfsHy8gargvnEdff0_k+tSF+66UZnsZJQ@mail.gmail.com>

I am trying to play some native files (Preencoded) in lua script and
calling the lua script in dialpan like below..

session:execute("playback", <path to file1>);
session:execute("playback", <path to file2>);
session:execute("playback", <path to file3>);

When I make call I hear file1 clearly but after that for approx 10-15
seconds I do not hear anything (except some occasional noise like breaking
voice)

I tried commenting file1 (to check if file1 is corrupt or something) but
then I hear file2 clearly and not file 3 for 10-15 seconds. I also tried
session:sleep(100) between playbacks but issues remains same.

Am I doing something wrong?
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From abaci64 at gmail.com  Tue May 19 01:37:04 2015
From: abaci64 at gmail.com (Abaci B)
Date: Mon, 18 May 2015 17:37:04 -0400
Subject: [Freeswitch-users] Playing multiple files in lua script
In-Reply-To: <CAA4rGPz=PveNCvUpbGfsHy8gargvnEdff0_k+tSF+66UZnsZJQ@mail.gmail.com>
References: <CAA4rGPz=PveNCvUpbGfsHy8gargvnEdff0_k+tSF+66UZnsZJQ@mail.gmail.com>
Message-ID: <CAFmZFiN8rynkr+F4F2dvAO3GCqHV+ErV_XEYHiZdCTEWtSJ+zw@mail.gmail.com>

is there anything in the logs when this happens?
any reason not to play the files using
file-string:///path/to/file1!/path/to/file2!/path/to/file3 ? not sure if it
would mke a difference but you could try.
did you also try if you see the same behavior when doing
session:streamFile("/path/to/file1") instead of session:execute("playback",
"/path/to/file1")


On Mon, May 18, 2015 at 5:22 PM, John Nash <john.nash778 at gmail.com> wrote:

> I am trying to play some native files (Preencoded) in lua script and
> calling the lua script in dialpan like below..
>
> session:execute("playback", <path to file1>);
> session:execute("playback", <path to file2>);
> session:execute("playback", <path to file3>);
>
> When I make call I hear file1 clearly but after that for approx 10-15
> seconds I do not hear anything (except some occasional noise like breaking
> voice)
>
> I tried commenting file1 (to check if file1 is corrupt or something) but
> then I hear file2 clearly and not file 3 for 10-15 seconds. I also tried
> session:sleep(100) between playbacks but issues remains same.
>
> Am I doing something wrong?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From john.nash778 at gmail.com  Tue May 19 02:13:29 2015
From: john.nash778 at gmail.com (John Nash)
Date: Tue, 19 May 2015 03:43:29 +0530
Subject: [Freeswitch-users] Playing multiple files in lua script
In-Reply-To: <CAFmZFiN8rynkr+F4F2dvAO3GCqHV+ErV_XEYHiZdCTEWtSJ+zw@mail.gmail.com>
References: <CAA4rGPz=PveNCvUpbGfsHy8gargvnEdff0_k+tSF+66UZnsZJQ@mail.gmail.com>
	<CAFmZFiN8rynkr+F4F2dvAO3GCqHV+ErV_XEYHiZdCTEWtSJ+zw@mail.gmail.com>
Message-ID: <CAA4rGPzV1oNbmT=cEQW==VN4wUQOaM7nGQu5LNn1udRqz9DwRg@mail.gmail.com>

Actually streamfile and playing using "!" (file_string") also had same
issue. But this is solved when I used rtp-timer-name = soft . before I was
using rtp-timer-name = none. I read in some posts that timer = soft causes
performance issues in centos 6. Do you have any idea about this?

also is Session:streamfile different from session:playback in some
way?...which should be preferred?




On Tue, May 19, 2015 at 3:07 AM, Abaci B <abaci64 at gmail.com> wrote:

> is there anything in the logs when this happens?
> any reason not to play the files using
> file-string:///path/to/file1!/path/to/file2!/path/to/file3 ? not sure if it
> would mke a difference but you could try.
> did you also try if you see the same behavior when doing
> session:streamFile("/path/to/file1") instead of session:execute("playback",
> "/path/to/file1")
>
>
> On Mon, May 18, 2015 at 5:22 PM, John Nash <john.nash778 at gmail.com> wrote:
>
>> I am trying to play some native files (Preencoded) in lua script and
>> calling the lua script in dialpan like below..
>>
>> session:execute("playback", <path to file1>);
>> session:execute("playback", <path to file2>);
>> session:execute("playback", <path to file3>);
>>
>> When I make call I hear file1 clearly but after that for approx 10-15
>> seconds I do not hear anything (except some occasional noise like breaking
>> voice)
>>
>> I tried commenting file1 (to check if file1 is corrupt or something) but
>> then I hear file2 clearly and not file 3 for 10-15 seconds. I also tried
>> session:sleep(100) between playbacks but issues remains same.
>>
>> Am I doing something wrong?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From abaci64 at gmail.com  Tue May 19 03:15:56 2015
From: abaci64 at gmail.com (Abaci B)
Date: Mon, 18 May 2015 19:15:56 -0400
Subject: [Freeswitch-users] Playing multiple files in lua script
In-Reply-To: <CAA4rGPzV1oNbmT=cEQW==VN4wUQOaM7nGQu5LNn1udRqz9DwRg@mail.gmail.com>
References: <CAA4rGPz=PveNCvUpbGfsHy8gargvnEdff0_k+tSF+66UZnsZJQ@mail.gmail.com>
	<CAFmZFiN8rynkr+F4F2dvAO3GCqHV+ErV_XEYHiZdCTEWtSJ+zw@mail.gmail.com>
	<CAA4rGPzV1oNbmT=cEQW==VN4wUQOaM7nGQu5LNn1udRqz9DwRg@mail.gmail.com>
Message-ID: <CAFmZFiO1SBsjBfyGFk-rhNoMHY6vGhD28SyRKoE=r_v8E0S3Ww@mail.gmail.com>

no idea about timer=soft issues on centos, I switched from centos to debian
when the core devs switch as I like to be on the platform best supported.
as far as usig streamFile over playback it gives you more control if you
need handling of DTMF (or speech recognition) and iirc using playback will
give you more logs

On Mon, May 18, 2015 at 6:13 PM, John Nash <john.nash778 at gmail.com> wrote:

> Actually streamfile and playing using "!" (file_string") also had same
> issue. But this is solved when I used rtp-timer-name = soft . before I was
> using rtp-timer-name = none. I read in some posts that timer = soft causes
> performance issues in centos 6. Do you have any idea about this?
>
> also is Session:streamfile different from session:playback in some
> way?...which should be preferred?
>
>
>
>
> On Tue, May 19, 2015 at 3:07 AM, Abaci B <abaci64 at gmail.com> wrote:
>
>> is there anything in the logs when this happens?
>> any reason not to play the files using
>> file-string:///path/to/file1!/path/to/file2!/path/to/file3 ? not sure if it
>> would mke a difference but you could try.
>> did you also try if you see the same behavior when doing
>> session:streamFile("/path/to/file1") instead of session:execute("playback",
>> "/path/to/file1")
>>
>>
>> On Mon, May 18, 2015 at 5:22 PM, John Nash <john.nash778 at gmail.com>
>> wrote:
>>
>>> I am trying to play some native files (Preencoded) in lua script and
>>> calling the lua script in dialpan like below..
>>>
>>> session:execute("playback", <path to file1>);
>>> session:execute("playback", <path to file2>);
>>> session:execute("playback", <path to file3>);
>>>
>>> When I make call I hear file1 clearly but after that for approx 10-15
>>> seconds I do not hear anything (except some occasional noise like breaking
>>> voice)
>>>
>>> I tried commenting file1 (to check if file1 is corrupt or something) but
>>> then I hear file2 clearly and not file 3 for 10-15 seconds. I also tried
>>> session:sleep(100) between playbacks but issues remains same.
>>>
>>> Am I doing something wrong?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ehsanminachi at gmail.com  Tue May 19 05:06:19 2015
From: ehsanminachi at gmail.com (Ehsan Minachi)
Date: Mon, 18 May 2015 18:06:19 -0700
Subject: [Freeswitch-users] Suggestions for Linux Distro
Message-ID: <CAPDpkrBg0w4zirfALnoqE9okX6cOUvXhsSgqt906gQCYBn-X-g@mail.gmail.com>

Hello all,

We have about 7 FreeSWITCH servers running CentOS 5 and we are planning to
replace CentOS 5 with a more recent Linux Distro.

Looking at wiki and docs I see FreeSWITCH team recommending Debian 7.

Right now we have picked 3 options for our upgrade path:

1) Debian 8 (Jessie)
2) Debian 7
3) Ubuntu 14.04 LTS

Among our options we really like to have a Distro either with systemd or
upstart managing our daemons, therefore our options come to Debian 8 and
Ubuntu 14.04.

I have read some posts users suggesting avoiding Ubuntu.

Does anyone suggest Debian 8 for production use?

Thank you,

- Ehsan
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From brian at freeswitch.org  Tue May 19 05:25:11 2015
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 May 2015 20:25:11 -0500
Subject: [Freeswitch-users] Suggestions for Linux Distro
In-Reply-To: <CAPDpkrBg0w4zirfALnoqE9okX6cOUvXhsSgqt906gQCYBn-X-g@mail.gmail.com>
References: <CAPDpkrBg0w4zirfALnoqE9okX6cOUvXhsSgqt906gQCYBn-X-g@mail.gmail.com>
Message-ID: <CAEJMVkBS3WA53-J=hOBV-s0-itu5Cnwq78MVGRxMMGQsf3bd9A@mail.gmail.com>

I'd read this:

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

And go with the flow and run Debian 8! ;)

On Mon, May 18, 2015 at 8:06 PM, Ehsan Minachi <ehsanminachi at gmail.com>
wrote:

> Hello all,
>
> We have about 7 FreeSWITCH servers running CentOS 5 and we are planning to
> replace CentOS 5 with a more recent Linux Distro.
>
> Looking at wiki and docs I see FreeSWITCH team recommending Debian 7.
>
> Right now we have picked 3 options for our upgrade path:
>
> 1) Debian 8 (Jessie)
> 2) Debian 7
> 3) Ubuntu 14.04 LTS
>
> Among our options we really like to have a Distro either with systemd or
> upstart managing our daemons, therefore our options come to Debian 8 and
> Ubuntu 14.04.
>
> I have read some posts users suggesting avoiding Ubuntu.
>
> Does anyone suggest Debian 8 for production use?
>
> Thank you,
>
> - Ehsan
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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  Tue May 19 05:27:36 2015
From: jprangi at didforsale.com (Jai Rangi)
Date: Mon, 18 May 2015 18:27:36 -0700
Subject: [Freeswitch-users] Monitoring SIP Service
Message-ID: <CAPfjz3iY9pdMsKDu1OnhO+McTY-ZwaKknfc=3CZcJiuUJ5yN3g@mail.gmail.com>

Very common concerns from new Asterisk, Freeswitch, opensips and freepbx
owners, How can we monitor system, what happens if service stops
responding.

Here is a small howto on monitoring any SIP service with nagios. I am sure
there are plenty of options and suggestions, but this is one of them and
has been working out very well for us for years.

http://www.didforsale.com/monitor-sip-server

Best,
-Jai
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From ehsanminachi at gmail.com  Tue May 19 05:49:18 2015
From: ehsanminachi at gmail.com (Ehsan Minachi)
Date: Mon, 18 May 2015 18:49:18 -0700
Subject: [Freeswitch-users] Suggestions for Linux Distro
In-Reply-To: <CAEJMVkBS3WA53-J=hOBV-s0-itu5Cnwq78MVGRxMMGQsf3bd9A@mail.gmail.com>
References: <CAPDpkrBg0w4zirfALnoqE9okX6cOUvXhsSgqt906gQCYBn-X-g@mail.gmail.com>
	<CAEJMVkBS3WA53-J=hOBV-s0-itu5Cnwq78MVGRxMMGQsf3bd9A@mail.gmail.com>
Message-ID: <CAPDpkrAU1KaNTMxE7gJGOQCdq_qsaiHy+uNpk7eX8=X9LWmQ3w@mail.gmail.com>

Brian,

Thank you very much for your suggestion.

We currently have about 30K subscribers and upgrading Linux is a bit scary
for us.

Taking into account that Jessie has only be out for few weeks, do you think
it's going to be stable for production?

Thank you very much, and sorry for asking again :) a bit paranoid

- Ehsan

On Monday, May 18, 2015, Brian West <brian at freeswitch.org> wrote:

> I'd read this:
>
> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>
> And go with the flow and run Debian 8! ;)
>
> On Mon, May 18, 2015 at 8:06 PM, Ehsan Minachi <ehsanminachi at gmail.com
> <javascript:_e(%7B%7D,'cvml','ehsanminachi at gmail.com');>> wrote:
>
>> Hello all,
>>
>> We have about 7 FreeSWITCH servers running CentOS 5 and we are planning
>> to replace CentOS 5 with a more recent Linux Distro.
>>
>> Looking at wiki and docs I see FreeSWITCH team recommending Debian 7.
>>
>> Right now we have picked 3 options for our upgrade path:
>>
>> 1) Debian 8 (Jessie)
>> 2) Debian 7
>> 3) Ubuntu 14.04 LTS
>>
>> Among our options we really like to have a Distro either with systemd or
>> upstart managing our daemons, therefore our options come to Debian 8 and
>> Ubuntu 14.04.
>>
>> I have read some posts users suggesting avoiding Ubuntu.
>>
>> Does anyone suggest Debian 8 for production use?
>>
>> Thank you,
>>
>> - Ehsan
>>
>> _________________________________________________________________________
>> 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
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 mitchelle.bit at gmail.com  Tue May 19 12:04:58 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Tue, 19 May 2015 13:34:58 +0530
Subject: [Freeswitch-users] billsec tag of XML CDR
In-Reply-To: <CAEJMVkCif1AnuH=01Jc=jBXGcdhXW84nJmcy7na_o6pM+Ld70w@mail.gmail.com>
References: <CANBm6SyW8Sn7quZax7cQcSOdkuEdBZPCs81nuq1aMT9qu5JaNQ@mail.gmail.com>
	<CAEJMVkDacx8VDyghdCeHPvRDcEHVoh3GB+sUhZ0zMVhTv2xBtw@mail.gmail.com>
	<CANBm6SwOsP+h5+-YTV-tiFTZ3oon4UaAuPXnJ1ZGZrMu3Ht9Gw@mail.gmail.com>
	<CAEJMVkA6-1mbwVpYrX_vdQMRhU0JE5Fp1is3WoWp8AG-A5iV6w@mail.gmail.com>
	<CANBm6SyNM=CbYLFaOpDtgLPOnF9G2RV3kdxAgJsr4FRpcRMk1Q@mail.gmail.com>
	<CAEJMVkCif1AnuH=01Jc=jBXGcdhXW84nJmcy7na_o6pM+Ld70w@mail.gmail.com>
Message-ID: <CANBm6Sw06ZNTq94uRDeU+Omqc7Xy-ifdfTZ+3xBYdnwU6CmZWQ@mail.gmail.com>

Hi Brain,

I will portray a scenario in which 'billsec' shows a non zero value.

I am currently running a FS installation for my company, now when a call is
made to my company and it is disconnected from my company's end then
'billsec' shows a non zero value as the call is then redirected to the
company's voice-mail for the user to record the message, in this case the
user has to pay.

When the call is disconnected by the user, he/she is not redirected to the
voice-mail and hence billsec shows zero as the value.

I hope the redirection to the voice-mail is the prime cause for blllsec
showing some value other than zero.

Please share your views.

Thanks,
Mitchelle

On Mon, May 18, 2015 at 11:57 PM, Brian West <brian at freeswitch.org> wrote:

> mod_dptools.c:1208 Channel [FreeTDM/1:17/016] has been answered
>
> Also set the debug level in vars.xml to debug, you're missing tons of info
> that could probably help you.  This is probably the ISDN doing a CONNECT
> and answering the channel to pay the busy signal.
>
> On Mon, May 18, 2015 at 12:48 PM, Mitchelle Johnson <
> mitchelle.bit at gmail.com> wrote:
>
>> Hi Brian,
>>
>> Please find the pastebin link http://pastebin.com/JhUyYSGb
>>
>> Thanks,
>> Mitchelle
>>
>> On Mon, May 18, 2015 at 10:38 PM, Brian West <brian at freeswitch.org>
>> wrote:
>>
>>> Yes please.
>>>
>>> On Mon, May 18, 2015 at 12:05 PM, Mitchelle Johnson <
>>> mitchelle.bit at gmail.com> wrote:
>>>
>>>> Hi Brian,
>>>>
>>>> Shall I paste the FS console logs?
>>>>
>>>> Thanks,
>>>> Mitchelle
>>>>
>>>> On Mon, May 18, 2015 at 10:29 PM, Brian West <brian at freeswitch.org>
>>>> wrote:
>>>>
>>>>> I can't see how thats possible, unless the call was actually answered,
>>>>> Happen to have logs?
>>>>>
>>>>> On Mon, May 18, 2015 at 11:56 AM, Mitchelle Johnson <
>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I am using XML CDR and updating my database dynamically with it. But
>>>>>> the 'billsec' tag of the XML CDR works strangely, it shows some value even
>>>>>> if the call is not answered.
>>>>>>
>>>>>> Please help.
>>>>>>
>>>>>> Thanks,
>>>>>> Mitchelle
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at 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
>>>>>
>>>>> ClueCon 2015 Call for Speakers
>>>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>>>>> <https://www.reddit.com/r/freeswitch>
>>>>>
>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>>> <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 mitchelle.bit at gmail.com  Tue May 19 12:13:31 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Tue, 19 May 2015 13:43:31 +0530
Subject: [Freeswitch-users] XML parsing error
In-Reply-To: <81FB7E30-F6B6-4551-8272-56A2A958BAF0@jerris.com>
References: <CANBm6Syqy-f24AH97-xYefPP89BKM0dNkTzjpS_ZN9jC_4kM5w@mail.gmail.com>
	<81FB7E30-F6B6-4551-8272-56A2A958BAF0@jerris.com>
Message-ID: <CANBm6Sy=YX9Z5vkhkboPyeG6Dm-Mz3jH6d5MLuLo36w4PXk0bg@mail.gmail.com>

Thanks, Sergey, Ali, Brian and Michael for your support and special mention
and thanks to Saumar for helping me.

Regards,
Mitchelle

On Tue, May 19, 2015 at 12:24 AM, Michael Jerris <mike at jerris.com> wrote:

> https://freeswitch.org/jira/browse/FS-7258
>
> On May 16, 2015, at 1:55 AM, Mitchelle Johnson <mitchelle.bit at gmail.com>
> wrote:
>
> Hi,
> When I am using the web server to handle xml CDR's the xml file which it
> sends gives an error.
> The error being:
>
> XML Parsing Error: not well-formed
> Location: http://www.w3schools.com/xml/xml_validator.asp
> Line Number 138, Column 25:
>     <sip_full_from><sip:1001 at 10.0.0.8>;tag=661a086d</sip_full_from>
> ------------------------^
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at gmail.com  Tue May 19 13:02:04 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 19 May 2015 11:02:04 +0200
Subject: [Freeswitch-users] Monitoring SIP Service
In-Reply-To: <CAPfjz3iY9pdMsKDu1OnhO+McTY-ZwaKknfc=3CZcJiuUJ5yN3g@mail.gmail.com>
References: <CAPfjz3iY9pdMsKDu1OnhO+McTY-ZwaKknfc=3CZcJiuUJ5yN3g@mail.gmail.com>
Message-ID: <CACOYK=gmmo4tvgej8ZXZPEDHasmz6H5wa2Yu9Yga4pa6MMDx+A@mail.gmail.com>

I'm working on two different projects for SIP testing and monitoring,
and soon there will be more information.

https://github.com/voxserv/rring
(documentation is still missing) this will be an automated tester that
uses FreeSWITCH to originate and terminate the calls, and it will
analyze the SIP messages that are received from remote side.


https://txlab.wordpress.com/2015/05/14/quality-assurance-for-voip-calls/
some scripts and work in progress for voice quality assurance.

I will make a separate posting as soon as I'm ready.




On Tue, May 19, 2015 at 3:27 AM, Jai Rangi <jprangi at didforsale.com> wrote:
> Very common concerns from new Asterisk, Freeswitch, opensips and freepbx
> owners, How can we monitor system, what happens if service stops responding.
>
> Here is a small howto on monitoring any SIP service with nagios. I am sure
> there are plenty of options and suggestions, but this is one of them and has
> been working out very well for us for years.
>
> http://www.didforsale.com/monitor-sip-server
>
> Best,
> -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


From s.safarov at gmail.com  Tue May 19 14:10:53 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 19 May 2015 13:10:53 +0300
Subject: [Freeswitch-users] load user vars from dialplan or CLI
Message-ID: <CAHtxdDcQdZTF=yr=K198UwK_m5m37pk-wRz5FgnZbe3_vXLh3w@mail.gmail.com>

In one example, I came across the command to retrieve all user variables of
the following form
searching commands to load all vars
originate ???/1000 at 8.8.8.8 Sofia / gatevay / 1111

Can you help locate this command?
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From lexxua at gmail.com  Tue May 19 14:35:44 2015
From: lexxua at gmail.com (=?UTF-8?B?0JLQu9Cw0LTQuNC80LjRgCDQpNC10LTQvtGA0L7Qsg==?=)
Date: Tue, 19 May 2015 10:35:44 +0000
Subject: [Freeswitch-users] Monitoring SIP Service
In-Reply-To: <CACOYK=gmmo4tvgej8ZXZPEDHasmz6H5wa2Yu9Yga4pa6MMDx+A@mail.gmail.com>
References: <CAPfjz3iY9pdMsKDu1OnhO+McTY-ZwaKknfc=3CZcJiuUJ5yN3g@mail.gmail.com>
	<CACOYK=gmmo4tvgej8ZXZPEDHasmz6H5wa2Yu9Yga4pa6MMDx+A@mail.gmail.com>
Message-ID: <CAO_GGAp-4eWSvVm61wqbu_iNeCcF07K9pSdzVPE=4Q-y0TccYg@mail.gmail.com>

In my company we use Yate  analyzer module
as a call quality monitor http://docs.yate.ro/wiki/Quality_analyzer_module
. Results of tests stored in Influxdb.
For gateway monitoring we use built in Freeswitch logging of DOWN state of
gateway.



On Tue, May 19, 2015 at 12:04 PM Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> I'm working on two different projects for SIP testing and monitoring,
> and soon there will be more information.
>
> https://github.com/voxserv/rring
> (documentation is still missing) this will be an automated tester that
> uses FreeSWITCH to originate and terminate the calls, and it will
> analyze the SIP messages that are received from remote side.
>
>
> https://txlab.wordpress.com/2015/05/14/quality-assurance-for-voip-calls/
> some scripts and work in progress for voice quality assurance.
>
> I will make a separate posting as soon as I'm ready.
>
>
>
>
> On Tue, May 19, 2015 at 3:27 AM, Jai Rangi <jprangi at didforsale.com> wrote:
> > Very common concerns from new Asterisk, Freeswitch, opensips and freepbx
> > owners, How can we monitor system, what happens if service stops
> responding.
> >
> > Here is a small howto on monitoring any SIP service with nagios. I am
> sure
> > there are plenty of options and suggestions, but this is one of them and
> has
> > been working out very well for us for years.
> >
> > http://www.didforsale.com/monitor-sip-server
> >
> > Best,
> > -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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at gmail.com  Tue May 19 14:47:59 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 19 May 2015 12:47:59 +0200
Subject: [Freeswitch-users] Monitoring SIP Service
In-Reply-To: <CAO_GGAp-4eWSvVm61wqbu_iNeCcF07K9pSdzVPE=4Q-y0TccYg@mail.gmail.com>
References: <CAPfjz3iY9pdMsKDu1OnhO+McTY-ZwaKknfc=3CZcJiuUJ5yN3g@mail.gmail.com>
	<CACOYK=gmmo4tvgej8ZXZPEDHasmz6H5wa2Yu9Yga4pa6MMDx+A@mail.gmail.com>
	<CAO_GGAp-4eWSvVm61wqbu_iNeCcF07K9pSdzVPE=4Q-y0TccYg@mail.gmail.com>
Message-ID: <CACOYK=hkth4KN+moo3g1oW1bgV-i0gDaW7mnpU+yhnVyv=8tqA@mail.gmail.com>

Hmm, quite interesting. I think we can develop a testing tool based on
this, and make it royalty free. And make it suitable for transcoding and
analog paths.
On May 19, 2015 12:36 PM, "???????? ???????" <lexxua at gmail.com> wrote:

> In my company we use Yate  analyzer module
> as a call quality monitor http://docs.yate.ro/wiki/Quality_analyzer_module
> . Results of tests stored in Influxdb.
> For gateway monitoring we use built in Freeswitch logging of DOWN state of
> gateway.
>
>
>
> On Tue, May 19, 2015 at 12:04 PM Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> I'm working on two different projects for SIP testing and monitoring,
>> and soon there will be more information.
>>
>> https://github.com/voxserv/rring
>> (documentation is still missing) this will be an automated tester that
>> uses FreeSWITCH to originate and terminate the calls, and it will
>> analyze the SIP messages that are received from remote side.
>>
>>
>> https://txlab.wordpress.com/2015/05/14/quality-assurance-for-voip-calls/
>> some scripts and work in progress for voice quality assurance.
>>
>> I will make a separate posting as soon as I'm ready.
>>
>>
>>
>>
>> On Tue, May 19, 2015 at 3:27 AM, Jai Rangi <jprangi at didforsale.com>
>> wrote:
>> > Very common concerns from new Asterisk, Freeswitch, opensips and freepbx
>> > owners, How can we monitor system, what happens if service stops
>> responding.
>> >
>> > Here is a small howto on monitoring any SIP service with nagios. I am
>> sure
>> > there are plenty of options and suggestions, but this is one of them
>> and has
>> > been working out very well for us for years.
>> >
>> > http://www.didforsale.com/monitor-sip-server
>> >
>> > Best,
>> > -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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Tue May 19 17:06:16 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 19 May 2015 14:06:16 +0100
Subject: [Freeswitch-users] billsec tag of XML CDR
In-Reply-To: <CANBm6Sw06ZNTq94uRDeU+Omqc7Xy-ifdfTZ+3xBYdnwU6CmZWQ@mail.gmail.com>
References: <CANBm6SyW8Sn7quZax7cQcSOdkuEdBZPCs81nuq1aMT9qu5JaNQ@mail.gmail.com>
	<CAEJMVkDacx8VDyghdCeHPvRDcEHVoh3GB+sUhZ0zMVhTv2xBtw@mail.gmail.com>
	<CANBm6SwOsP+h5+-YTV-tiFTZ3oon4UaAuPXnJ1ZGZrMu3Ht9Gw@mail.gmail.com>
	<CAEJMVkA6-1mbwVpYrX_vdQMRhU0JE5Fp1is3WoWp8AG-A5iV6w@mail.gmail.com>
	<CANBm6SyNM=CbYLFaOpDtgLPOnF9G2RV3kdxAgJsr4FRpcRMk1Q@mail.gmail.com>
	<CAEJMVkCif1AnuH=01Jc=jBXGcdhXW84nJmcy7na_o6pM+Ld70w@mail.gmail.com>
	<CANBm6Sw06ZNTq94uRDeU+Omqc7Xy-ifdfTZ+3xBYdnwU6CmZWQ@mail.gmail.com>
Message-ID: <CAFiqYunh-3WMST6UHVbm6ZxcEevyoH0vWwbgCtmuP4YR5A+tGw@mail.gmail.com>

The call will be answered by the voicemail, so a non-zero billsec will be
correct.

On 19 May 2015 at 09:04, Mitchelle Johnson <mitchelle.bit at gmail.com> wrote:

> Hi Brain,
>
> I will portray a scenario in which 'billsec' shows a non zero value.
>
> I am currently running a FS installation for my company, now when a call
> is made to my company and it is disconnected from my company's end then
> 'billsec' shows a non zero value as the call is then redirected to the
> company's voice-mail for the user to record the message, in this case the
> user has to pay.
>
> When the call is disconnected by the user, he/she is not redirected to the
> voice-mail and hence billsec shows zero as the value.
>
> I hope the redirection to the voice-mail is the prime cause for blllsec
> showing some value other than zero.
>
> Please share your views.
>
> Thanks,
> Mitchelle
>
> On Mon, May 18, 2015 at 11:57 PM, Brian West <brian at freeswitch.org> wrote:
>
>> mod_dptools.c:1208 Channel [FreeTDM/1:17/016] has been answered
>>
>> Also set the debug level in vars.xml to debug, you're missing tons of
>> info that could probably help you.  This is probably the ISDN doing a
>> CONNECT and answering the channel to pay the busy signal.
>>
>> On Mon, May 18, 2015 at 12:48 PM, Mitchelle Johnson <
>> mitchelle.bit at gmail.com> wrote:
>>
>>> Hi Brian,
>>>
>>> Please find the pastebin link http://pastebin.com/JhUyYSGb
>>>
>>> Thanks,
>>> Mitchelle
>>>
>>> On Mon, May 18, 2015 at 10:38 PM, Brian West <brian at freeswitch.org>
>>> wrote:
>>>
>>>> Yes please.
>>>>
>>>> On Mon, May 18, 2015 at 12:05 PM, Mitchelle Johnson <
>>>> mitchelle.bit at gmail.com> wrote:
>>>>
>>>>> Hi Brian,
>>>>>
>>>>> Shall I paste the FS console logs?
>>>>>
>>>>> Thanks,
>>>>> Mitchelle
>>>>>
>>>>> On Mon, May 18, 2015 at 10:29 PM, Brian West <brian at freeswitch.org>
>>>>> wrote:
>>>>>
>>>>>> I can't see how thats possible, unless the call was actually
>>>>>> answered, Happen to have logs?
>>>>>>
>>>>>> On Mon, May 18, 2015 at 11:56 AM, Mitchelle Johnson <
>>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> I am using XML CDR and updating my database dynamically with it. But
>>>>>>> the 'billsec' tag of the XML CDR works strangely, it shows some value even
>>>>>>> if the call is not answered.
>>>>>>>
>>>>>>> Please help.
>>>>>>>
>>>>>>> Thanks,
>>>>>>> Mitchelle
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at 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
>>>>>>
>>>>>> ClueCon 2015 Call for Speakers
>>>>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>>>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
>>>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>>>
>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> ClueCon 2015 Call for Speakers
>>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>>>> <https://www.reddit.com/r/freeswitch>
>>>>
>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 emplant2000 at gmail.com  Tue May 19 17:11:40 2015
From: emplant2000 at gmail.com (Masakazu Nakano)
Date: Tue, 19 May 2015 22:11:40 +0900
Subject: [Freeswitch-users] newbie: originate dial problem
Message-ID: <CAPwL2MC--nb2s6WjL-udZZHoCAMMPBhQv8ELpS6ucRpADmNLJg@mail.gmail.com>

Hi there.

a softphone is registed called 5630 in public.

show registered says following under like that.
5630,27.112.104.17,siGhOxWCoIW0n9JjzwxSTg..,sofia/external/sip:5630 at 49.97.4.168:61682
;transport=UDP;rinstance=b1c4cf82f7d6e637,1432040746,49.97.4.168,61682,udp,
b3am.com,

and

freeswitch at internal > originate user/5630

is successfull.

but by dialplan/public/5630.xml
   <extension name="5630">
     <condition field="destination_number" expression="5630">
       <action application="bridge" data="user/5630"/>
     </condition>
   </extension>

and make a call with another phone to 5630.to be "Cause: INVALID_PROFILE"

2015-05-19 22:08:53.724519 [NOTICE] switch_ivr_originate.c:2732 Cannot
create outgoing channel of type [sofia] cause: [INVALID_PROFILE]
2015-05-19 22:08:53.724519 [DEBUG] switch_ivr_originate.c:3720 Originate
Resulted in Error Cause: 611 [INVALID_PROFILE]
2015-05-19 22:08:53.724519 [INFO] mod_dptools.c:3246 Originate Failed.
Cause: INVALID_PROFILE

why?

BR

mack
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From brian at freeswitch.org  Tue May 19 17:49:15 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 May 2015 08:49:15 -0500
Subject: [Freeswitch-users] newbie: originate dial problem
In-Reply-To: <CAPwL2MC--nb2s6WjL-udZZHoCAMMPBhQv8ELpS6ucRpADmNLJg@mail.gmail.com>
References: <CAPwL2MC--nb2s6WjL-udZZHoCAMMPBhQv8ELpS6ucRpADmNLJg@mail.gmail.com>
Message-ID: <CAEJMVkCWxk=d7roEvZ8mroQ+rjy+yeujOEcOJ3RoLCXZP6grqQ@mail.gmail.com>

What rev of FreeSWITCH and what is the output of 'sofia status',   Then you
can try sofia_contact */5630 and see what it returns, and pastbin all of
that info for us to see.



On Tue, May 19, 2015 at 8:11 AM, Masakazu Nakano <emplant2000 at gmail.com>
wrote:

> Hi there.
>
> a softphone is registed called 5630 in public.
>
> show registered says following under like that.
>
> 5630,27.112.104.17,siGhOxWCoIW0n9JjzwxSTg..,sofia/external/sip:5630 at 49.97.4.168:61682
> ;transport=UDP;rinstance=b1c4cf82f7d6e637,1432040746,49.97.4.168,61682,udp,
> b3am.com,
>
> and
>
> freeswitch at internal > originate user/5630
>
> is successfull.
>
> but by dialplan/public/5630.xml
>    <extension name="5630">
>      <condition field="destination_number" expression="5630">
>        <action application="bridge" data="user/5630"/>
>      </condition>
>    </extension>
>
> and make a call with another phone to 5630.to be "Cause: INVALID_PROFILE"
>
> 2015-05-19 22:08:53.724519 [NOTICE] switch_ivr_originate.c:2732 Cannot
> create outgoing channel of type [sofia] cause: [INVALID_PROFILE]
> 2015-05-19 22:08:53.724519 [DEBUG] switch_ivr_originate.c:3720 Originate
> Resulted in Error Cause: 611 [INVALID_PROFILE]
> 2015-05-19 22:08:53.724519 [INFO] mod_dptools.c:3246 Originate Failed.
> Cause: INVALID_PROFILE
>
> why?
>
> BR
>
> mack
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From aqsyounas at gmail.com  Tue May 19 18:07:01 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 19 May 2015 07:07:01 -0700
Subject: [Freeswitch-users] Help in freeswitch dbh query.
Message-ID: <CAGVOjVWDej=c_xRJcBTjgs1Wjxcg=HWGF1BRWA00g0VKd6Fg3w@mail.gmail.com>

Hi, users.

I am trying to fetch language from database through freeswitch dbh after
user presses some digits on main message.

Database table structure is as follow.

Id    Did  inp_1  inp_2   inp_3

1     14045872020 <14048002020> english  chinese hindi

 *input* = session:playAndGetDigits(1, 1, 3, 3000, "#",main_msg,
"/error.wav", "\\d")
session:consoleLog("info", "Caller pressed  : ".. input .."\n")

sql ="select inp_"..input.." from bible_main where Did = "..Dest
session:consoleLog("info", "SQL : ".. sql .."\n")

dbh:query(sql, function(row)
     * lang = row.inp_..input*
      session:consoleLog("info", "Caller has Selected : ".. lang .."\n")
end)

Running above code gives me below error.
2015-05-17 02:26:20.057184 [ERR] mod_lua.cpp:103
/usr/local/freeswitch/scripts/bible.lua:27: attempt to concatenate field
'inp_' (a nil value)
stack traceback:
        /usr/local/freeswitch/scripts/bible.lua:27: in function
</usr/local/freeswitch/scripts/bible.lua:25>
        [C]: in function 'query'
        /usr/local/freeswitch/scripts/bible.lua:25: in main chunk

Actually, i do not know how to concatenate a table value with string so
that it reflects the table column. So, i can get value from this.

Running it like this, gives perfect result.

dbh:query(sql, function(row)
     * lang = row.inp_1*
      session:consoleLog("info", "Caller has Selected : ".. lang .."\n")
end)

But I want user's input not some hard coded values for input.


Any help in this regard.
Thanks.
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From aqsyounas at gmail.com  Tue May 19 18:30:33 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 19 May 2015 07:30:33 -0700
Subject: [Freeswitch-users] Help in freeswitch dbh query.
In-Reply-To: <CAGVOjVWDej=c_xRJcBTjgs1Wjxcg=HWGF1BRWA00g0VKd6Fg3w@mail.gmail.com>
References: <CAGVOjVWDej=c_xRJcBTjgs1Wjxcg=HWGF1BRWA00g0VKd6Fg3w@mail.gmail.com>
Message-ID: <CAGVOjVVF5WkCCaJVOwxkV_JqodvWQ-Mq-p5+SGD9dDerY0nvdQ@mail.gmail.com>

Sorry, i resolved the issue. Thanks.

On 19 May 2015 at 07:07, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi, users.
>
> I am trying to fetch language from database through freeswitch dbh after
> user presses some digits on main message.
>
> Database table structure is as follow.
>
> Id    Did  inp_1  inp_2   inp_3
>
> 1     14045872020 <14048002020> english  chinese hindi
>
>  *input* = session:playAndGetDigits(1, 1, 3, 3000, "#",main_msg,
> "/error.wav", "\\d")
> session:consoleLog("info", "Caller pressed  : ".. input .."\n")
>
> sql ="select inp_"..input.." from bible_main where Did = "..Dest
> session:consoleLog("info", "SQL : ".. sql .."\n")
>
> dbh:query(sql, function(row)
>      * lang = row.inp_..input*
>       session:consoleLog("info", "Caller has Selected : ".. lang .."\n")
> end)
>
> Running above code gives me below error.
> 2015-05-17 02:26:20.057184 [ERR] mod_lua.cpp:103
> /usr/local/freeswitch/scripts/bible.lua:27: attempt to concatenate field
> 'inp_' (a nil value)
> stack traceback:
>         /usr/local/freeswitch/scripts/bible.lua:27: in function
> </usr/local/freeswitch/scripts/bible.lua:25>
>         [C]: in function 'query'
>         /usr/local/freeswitch/scripts/bible.lua:25: in main chunk
>
> Actually, i do not know how to concatenate a table value with string so
> that it reflects the table column. So, i can get value from this.
>
> Running it like this, gives perfect result.
>
> dbh:query(sql, function(row)
>      * lang = row.inp_1*
>       session:consoleLog("info", "Caller has Selected : ".. lang .."\n")
> end)
>
> But I want user's input not some hard coded values for input.
>
>
> Any help in this regard.
> Thanks.
>
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From emplant2000 at gmail.com  Tue May 19 18:32:01 2015
From: emplant2000 at gmail.com (Masakazu Nakano)
Date: Tue, 19 May 2015 23:32:01 +0900
Subject: [Freeswitch-users] newbie: originate dial problem
In-Reply-To: <CAEJMVkCWxk=d7roEvZ8mroQ+rjy+yeujOEcOJ3RoLCXZP6grqQ@mail.gmail.com>
References: <CAPwL2MC--nb2s6WjL-udZZHoCAMMPBhQv8ELpS6ucRpADmNLJg@mail.gmail.com>
	<CAEJMVkCWxk=d7roEvZ8mroQ+rjy+yeujOEcOJ3RoLCXZP6grqQ@mail.gmail.com>
Message-ID: <CAPwL2MDMJNQJuqYbijckyLDrqsfjBU0C_p5d1nf2ZnKFod5Eow@mail.gmail.com>

HI Brian

thank you for your reply.

'sofia status' says following like  that.

freeswitch at internal> sofia status
                     Name          Type
  Data      State
=================================================================================================
            external-ipv6       profile
sip:mod_sofia@[::2]:5060
     RUNNING (0)
            27.112.104.17         alias
internal      ALIASED
                 external       profile
sip:mod_sofia at 27.112.104.17:5060      RUNNING (0)
          external::iptel       gateway
sip:emplant2000 at sip.iptel.org      FAIL_WAIT
            internal-ipv6       profile
sip:mod_sofia@[::2]:5090
     RUNNING (0)
                 internal       profile
sip:mod_sofia at fusionpbx.b3am.com:5090      RUNNING (0)
=================================================================================================
4 profiles 1 alias

BR

mack



2015-05-19 22:49 GMT+09:00 Brian West <brian at freeswitch.org>:

> What rev of FreeSWITCH and what is the output of 'sofia status',   Then
> you can try sofia_contact */5630 and see what it returns, and pastbin all
> of that info for us to see.
>
>
>
> On Tue, May 19, 2015 at 8:11 AM, Masakazu Nakano <emplant2000 at gmail.com>
> wrote:
>
>> Hi there.
>>
>> a softphone is registed called 5630 in public.
>>
>> show registered says following under like that.
>>
>> 5630,27.112.104.17,siGhOxWCoIW0n9JjzwxSTg..,sofia/external/sip:5630 at 49.97.4.168:61682
>> ;transport=UDP;rinstance=b1c4cf82f7d6e637,1432040746,49.97.4.168,61682,udp,
>> b3am.com,
>>
>> and
>>
>> freeswitch at internal > originate user/5630
>>
>> is successfull.
>>
>> but by dialplan/public/5630.xml
>>    <extension name="5630">
>>      <condition field="destination_number" expression="5630">
>>        <action application="bridge" data="user/5630"/>
>>      </condition>
>>    </extension>
>>
>> and make a call with another phone to 5630.to be "Cause: INVALID_PROFILE"
>>
>> 2015-05-19 22:08:53.724519 [NOTICE] switch_ivr_originate.c:2732 Cannot
>> create outgoing channel of type [sofia] cause: [INVALID_PROFILE]
>> 2015-05-19 22:08:53.724519 [DEBUG] switch_ivr_originate.c:3720 Originate
>> Resulted in Error Cause: 611 [INVALID_PROFILE]
>> 2015-05-19 22:08:53.724519 [INFO] mod_dptools.c:3246 Originate Failed.
>> Cause: INVALID_PROFILE
>>
>> why?
>>
>> BR
>>
>> mack
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 abaci64 at gmail.com  Tue May 19 18:33:56 2015
From: abaci64 at gmail.com (Abaci B)
Date: Tue, 19 May 2015 10:33:56 -0400
Subject: [Freeswitch-users] Help in freeswitch dbh query.
In-Reply-To: <CAGVOjVWDej=c_xRJcBTjgs1Wjxcg=HWGF1BRWA00g0VKd6Fg3w@mail.gmail.com>
References: <CAGVOjVWDej=c_xRJcBTjgs1Wjxcg=HWGF1BRWA00g0VKd6Fg3w@mail.gmail.com>
Message-ID: <CAFmZFiO1CmW2g_ztTOb_5oMj-2YVj=N97WR6L5goYdn3+QySUA@mail.gmail.com>

try      * lang = row[inp_..input]*

On Tue, May 19, 2015 at 10:07 AM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi, users.
>
> I am trying to fetch language from database through freeswitch dbh after
> user presses some digits on main message.
>
> Database table structure is as follow.
>
> Id    Did  inp_1  inp_2   inp_3
>
> 1     14045872020 <14048002020> english  chinese hindi
>
>  *input* = session:playAndGetDigits(1, 1, 3, 3000, "#",main_msg,
> "/error.wav", "\\d")
> session:consoleLog("info", "Caller pressed  : ".. input .."\n")
>
> sql ="select inp_"..input.." from bible_main where Did = "..Dest
> session:consoleLog("info", "SQL : ".. sql .."\n")
>
> dbh:query(sql, function(row)
>      * lang = row.inp_..input*
>       session:consoleLog("info", "Caller has Selected : ".. lang .."\n")
> end)
>
> Running above code gives me below error.
> 2015-05-17 02:26:20.057184 [ERR] mod_lua.cpp:103
> /usr/local/freeswitch/scripts/bible.lua:27: attempt to concatenate field
> 'inp_' (a nil value)
> stack traceback:
>         /usr/local/freeswitch/scripts/bible.lua:27: in function
> </usr/local/freeswitch/scripts/bible.lua:25>
>         [C]: in function 'query'
>         /usr/local/freeswitch/scripts/bible.lua:25: in main chunk
>
> Actually, i do not know how to concatenate a table value with string so
> that it reflects the table column. So, i can get value from this.
>
> Running it like this, gives perfect result.
>
> dbh:query(sql, function(row)
>      * lang = row.inp_1*
>       session:consoleLog("info", "Caller has Selected : ".. lang .."\n")
> end)
>
> But I want user's input not some hard coded values for input.
>
>
> Any help in this regard.
> 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 mitchelle.bit at gmail.com  Tue May 19 18:39:31 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Tue, 19 May 2015 20:09:31 +0530
Subject: [Freeswitch-users] billsec tag of XML CDR
In-Reply-To: <CAFiqYunh-3WMST6UHVbm6ZxcEevyoH0vWwbgCtmuP4YR5A+tGw@mail.gmail.com>
References: <CANBm6SyW8Sn7quZax7cQcSOdkuEdBZPCs81nuq1aMT9qu5JaNQ@mail.gmail.com>
	<CAEJMVkDacx8VDyghdCeHPvRDcEHVoh3GB+sUhZ0zMVhTv2xBtw@mail.gmail.com>
	<CANBm6SwOsP+h5+-YTV-tiFTZ3oon4UaAuPXnJ1ZGZrMu3Ht9Gw@mail.gmail.com>
	<CAEJMVkA6-1mbwVpYrX_vdQMRhU0JE5Fp1is3WoWp8AG-A5iV6w@mail.gmail.com>
	<CANBm6SyNM=CbYLFaOpDtgLPOnF9G2RV3kdxAgJsr4FRpcRMk1Q@mail.gmail.com>
	<CAEJMVkCif1AnuH=01Jc=jBXGcdhXW84nJmcy7na_o6pM+Ld70w@mail.gmail.com>
	<CANBm6Sw06ZNTq94uRDeU+Omqc7Xy-ifdfTZ+3xBYdnwU6CmZWQ@mail.gmail.com>
	<CAFiqYunh-3WMST6UHVbm6ZxcEevyoH0vWwbgCtmuP4YR5A+tGw@mail.gmail.com>
Message-ID: <CANBm6SxpeeYDxjh1w7oRNBtMAY9T-yxjDeZaYTmfmvWG=vL6Yg@mail.gmail.com>

Hi Steven, Brian,

I also discovered the same thing. I thank you both for your views and being
so responsive in answering.

Regards,
Mitchelle

On Tue, May 19, 2015 at 6:36 PM, Steven Ayre <steveayre at gmail.com> wrote:

> The call will be answered by the voicemail, so a non-zero billsec will be
> correct.
>
> On 19 May 2015 at 09:04, Mitchelle Johnson <mitchelle.bit at gmail.com>
> wrote:
>
>> Hi Brain,
>>
>> I will portray a scenario in which 'billsec' shows a non zero value.
>>
>> I am currently running a FS installation for my company, now when a call
>> is made to my company and it is disconnected from my company's end then
>> 'billsec' shows a non zero value as the call is then redirected to the
>> company's voice-mail for the user to record the message, in this case the
>> user has to pay.
>>
>> When the call is disconnected by the user, he/she is not redirected to
>> the voice-mail and hence billsec shows zero as the value.
>>
>> I hope the redirection to the voice-mail is the prime cause for blllsec
>> showing some value other than zero.
>>
>> Please share your views.
>>
>> Thanks,
>> Mitchelle
>>
>> On Mon, May 18, 2015 at 11:57 PM, Brian West <brian at freeswitch.org>
>> wrote:
>>
>>> mod_dptools.c:1208 Channel [FreeTDM/1:17/016] has been answered
>>>
>>> Also set the debug level in vars.xml to debug, you're missing tons of
>>> info that could probably help you.  This is probably the ISDN doing a
>>> CONNECT and answering the channel to pay the busy signal.
>>>
>>> On Mon, May 18, 2015 at 12:48 PM, Mitchelle Johnson <
>>> mitchelle.bit at gmail.com> wrote:
>>>
>>>> Hi Brian,
>>>>
>>>> Please find the pastebin link http://pastebin.com/JhUyYSGb
>>>>
>>>> Thanks,
>>>> Mitchelle
>>>>
>>>> On Mon, May 18, 2015 at 10:38 PM, Brian West <brian at freeswitch.org>
>>>> wrote:
>>>>
>>>>> Yes please.
>>>>>
>>>>> On Mon, May 18, 2015 at 12:05 PM, Mitchelle Johnson <
>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>
>>>>>> Hi Brian,
>>>>>>
>>>>>> Shall I paste the FS console logs?
>>>>>>
>>>>>> Thanks,
>>>>>> Mitchelle
>>>>>>
>>>>>> On Mon, May 18, 2015 at 10:29 PM, Brian West <brian at freeswitch.org>
>>>>>> wrote:
>>>>>>
>>>>>>> I can't see how thats possible, unless the call was actually
>>>>>>> answered, Happen to have logs?
>>>>>>>
>>>>>>> On Mon, May 18, 2015 at 11:56 AM, Mitchelle Johnson <
>>>>>>> mitchelle.bit at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>> I am using XML CDR and updating my database dynamically with it.
>>>>>>>> But the 'billsec' tag of the XML CDR works strangely, it shows some value
>>>>>>>> even if the call is not answered.
>>>>>>>>
>>>>>>>> Please help.
>>>>>>>>
>>>>>>>> Thanks,
>>>>>>>> Mitchelle
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at 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
>>>>>>>
>>>>>>> ClueCon 2015 Call for Speakers
>>>>>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>>>>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
>>>>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>>>>
>>>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> *Brian West*
>>>>> brian at freeswitch.org
>>>>>
>>>>>
>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>> http://www.freeswitchbook.com
>>>>> http://www.freeswitchcookbook.com
>>>>>
>>>>> ClueCon 2015 Call for Speakers
>>>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>>>>> <https://www.reddit.com/r/freeswitch>
>>>>>
>>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>>> <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aqsyounas at gmail.com  Tue May 19 18:41:28 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 19 May 2015 07:41:28 -0700
Subject: [Freeswitch-users] Help in freeswitch dbh query.
In-Reply-To: <CAFmZFiO1CmW2g_ztTOb_5oMj-2YVj=N97WR6L5goYdn3+QySUA@mail.gmail.com>
References: <CAGVOjVWDej=c_xRJcBTjgs1Wjxcg=HWGF1BRWA00g0VKd6Fg3w@mail.gmail.com>
	<CAFmZFiO1CmW2g_ztTOb_5oMj-2YVj=N97WR6L5goYdn3+QySUA@mail.gmail.com>
Message-ID: <CAGVOjVWxKt5naBWQ5pcpHdDGYGz4NhmoOagqx8HZ6OXpNV56AA@mail.gmail.com>

Thanks for reply.

row["inp_"..input]

Solved the problem.

On 19 May 2015 at 07:33, Abaci B <abaci64 at gmail.com> wrote:

> try      * lang = row[inp_..input]*
>
> On Tue, May 19, 2015 at 10:07 AM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Hi, users.
>>
>> I am trying to fetch language from database through freeswitch dbh after
>> user presses some digits on main message.
>>
>> Database table structure is as follow.
>>
>> Id    Did  inp_1  inp_2   inp_3
>>
>> 1     14045872020 <14048002020> english  chinese hindi
>>
>>  *input* = session:playAndGetDigits(1, 1, 3, 3000, "#",main_msg,
>> "/error.wav", "\\d")
>> session:consoleLog("info", "Caller pressed  : ".. input .."\n")
>>
>> sql ="select inp_"..input.." from bible_main where Did = "..Dest
>> session:consoleLog("info", "SQL : ".. sql .."\n")
>>
>> dbh:query(sql, function(row)
>>      * lang = row.inp_..input*
>>       session:consoleLog("info", "Caller has Selected : ".. lang .."\n")
>> end)
>>
>> Running above code gives me below error.
>> 2015-05-17 02:26:20.057184 [ERR] mod_lua.cpp:103
>> /usr/local/freeswitch/scripts/bible.lua:27: attempt to concatenate field
>> 'inp_' (a nil value)
>> stack traceback:
>>         /usr/local/freeswitch/scripts/bible.lua:27: in function
>> </usr/local/freeswitch/scripts/bible.lua:25>
>>         [C]: in function 'query'
>>         /usr/local/freeswitch/scripts/bible.lua:25: in main chunk
>>
>> Actually, i do not know how to concatenate a table value with string so
>> that it reflects the table column. So, i can get value from this.
>>
>> Running it like this, gives perfect result.
>>
>> dbh:query(sql, function(row)
>>      * lang = row.inp_1*
>>       session:consoleLog("info", "Caller has Selected : ".. lang .."\n")
>> end)
>>
>> But I want user's input not some hard coded values for input.
>>
>>
>> Any help in this regard.
>> Thanks.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Tue May 19 22:10:40 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 May 2015 14:10:40 -0400
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
Message-ID: <8F2655CF-363F-45FA-9571-E2E06C356072@jerris.com>

Please review my previous question.

> On May 18, 2015, at 4:59 PM, Naveen Tamanam <naveen32india at gmail.com> wrote:
> 
> I am trying to do the following,  I would like to reject the call when it ringing from the fs console. 
> And second thing is I am pretty much eager to know the way to send sip(signaling) message manually for a 
> selected channel from fs console. 
> 
> 
> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> respond is one way, what exact message are you trying to send, and at what point in the call.  There are capabilites to trigger re-invites in some situations, transfer, send notify or info or message.  It depends on what exactly you are trying to do.
> 
> 
>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> 
>> Hi, 
>> 
>> I am wondering how to send sip signaling message from the fs console for the particular user/caller. 
>> I found  respond dialplan application to send sip message back to the caller. 
>> Like 
>> <action application="respond" data="480 Try again later"/>
>> ?Is there any way to send sip message back to the caller. One use case is call rejection or playing with SIP?
>> 
> 
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Thanks & Regards,
> Naveen Tamanam
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From marketing at cluecon.com  Tue May 19 21:14:02 2015
From: marketing at cluecon.com (ClueCon Marketing)
Date: Tue, 19 May 2015 12:14:02 -0500
Subject: [Freeswitch-users] ClueCon 2015,
	Register before June 1st for more chances to win!
Message-ID: <CAJEN5BKBM87EWWkQgK4qnEY_Ubhuwbk+T-o9T+k=4ZuCbt1Xfg@mail.gmail.com>

[image: Description: HDD:Users:anthm:Downloads:ccxx.jpg]
 *Register Today!*

August 3rd ? August 6th 2015

877-742-2583  ?  marketing at cluecon.com

Register NOW!

?   $899 Staying at the Hotel

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?   First 100 Gino?s PIZZA PARTY


*IMPORTANT DATES*

*Before June 1st*

15 Raffle Tickets Per Day

2 Gigabit Drink Coupons


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5 Raffle Tickets Per Day

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*Contact Us*

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*Speaker Submission Deadline*

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*Suggested Speaking Topics*

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Standards

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SIP

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Your Latest and Greatest



*Great Speakers, Hack-a-Thon, WebRTC And MUCH MORE at ClueCon 2015!*



Be sure to register as soon as possible for the upcoming ClueCon 2015
Developers Conference.



Not only will it give you piece of mind, the sooner you register, the more
opportunities you will get to win prizes! You?ll also get more drink
coupons for the Gigabit Reception Tuesday Night!



[image: Description: mac1]The grand prize is a laser engraved commemorative
FreeSWITCH 1.6 Edition dual-core 13" Retina MacBook Pro!  See the Important
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Stay tuned for more exciting announcements!


 Why I Think You MUST COME To ClueCon!



[image: Description: kk]Hi, I?m Kathleen. I?m the FreeSWITCH and ClueCon
Social Media Correspondent.  I?ve been working hard all year keeping you
all up to date on what?s going on with FreeSWITCH. Today I?m here to let
you know more about the upcoming ClueCon 2015 Conference!



This year we are adding an optional day on Monday with an all-day
Hack-A-Thon with great coding contests, game shows and kick-off fun!



If you are interested in WebRTC, Voice over IP or Open Source projects like
FreeSWITCH, ClueCon is the greatest opportunity you have to gain exposure
to the most knowledge and technology in one place.  Also, it?s the most fun
you can possibly have while still getting a ton of work done!  I really
look forward to seeing you all there and enjoying the amazing talks, the
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Make sure you register today so you can reserve your place among the
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Be sure to follow us on Facebook and Twitter to get my latest updates in
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[image: Inline image 1] <https://facebook.com/cluecon> [image: Inline image
2] <https://twitter.com/cluecon>



 It?s Not Too Late to Sign Up as a Speaker



Do you have a great idea or an awesome project you have been working on? We
would love to hear about it! Show off what you know!

[image: Description: fss]Sign up to speak at ClueCon this year and let us
know what you?ve been doing. The call for speakers page for ClueCon 2015 is
now open and you can visit https://cluecon.com/speaking-proposals to submit
your speaking proposal right now!





[image: Inline image 3] <https://www.youtube.com/watch?v=NLaDpGQuZDA>


 [image: Inline image 4] <https://www.youtube.com/watch?v=9XXgW34t40s>
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From mario_fs at mgtech.com  Tue May 19 22:32:21 2015
From: mario_fs at mgtech.com (Mario)
Date: Tue, 19 May 2015 11:32:21 -0700
Subject: [Freeswitch-users] Playing multiple files in lua script
In-Reply-To: <CAFmZFiO1SBsjBfyGFk-rhNoMHY6vGhD28SyRKoE=r_v8E0S3Ww@mail.gmail.com>
References: <CAA4rGPz=PveNCvUpbGfsHy8gargvnEdff0_k+tSF+66UZnsZJQ@mail.gmail.com>
	<CAFmZFiN8rynkr+F4F2dvAO3GCqHV+ErV_XEYHiZdCTEWtSJ+zw@mail.gmail.com>
	<CAA4rGPzV1oNbmT=cEQW==VN4wUQOaM7nGQu5LNn1udRqz9DwRg@mail.gmail.com>
	<CAFmZFiO1SBsjBfyGFk-rhNoMHY6vGhD28SyRKoE=r_v8E0S3Ww@mail.gmail.com>
Message-ID: <901298B3-9395-4AE4-A654-51CBD3D1CAFD@mgtech.com>

Here is how I construct a LUA variable to play multiple files in a ringback, syntax would be the same for your application. This is a complex one since it not only plays different files but sets a time limit for each:

   ringback = "file_string://"..em1.."!{timeout="..moh1_timeout.."}local_stream://mohmv!file_string://"..em2.."!local_stream://mohmv?

Also see https://wiki.freeswitch.org/wiki/Mod_local_stream <https://wiki.freeswitch.org/wiki/Mod_local_stream>, I added the timeout stuff at the bottom which shows syntax for a file.
Hope this helps a little
Mario G



> On May 18, 2015, at 4:15 PM, Abaci B <abaci64 at gmail.com> wrote:
> 
> no idea about timer=soft issues on centos, I switched from centos to debian when the core devs switch as I like to be on the platform best supported.
> as far as usig streamFile over playback it gives you more control if you need handling of DTMF (or speech recognition) and iirc using playback will give you more logs
> 
> On Mon, May 18, 2015 at 6:13 PM, John Nash <john.nash778 at gmail.com <mailto:john.nash778 at gmail.com>> wrote:
> Actually streamfile and playing using "!" (file_string") also had same issue. But this is solved when I used rtp-timer-name = soft . before I was using rtp-timer-name = none. I read in some posts that timer = soft causes performance issues in centos 6. Do you have any idea about this?
> 
> also is Session:streamfile different from session:playback in some way?...which should be preferred? 
> 
> 
> 
> 
> On Tue, May 19, 2015 at 3:07 AM, Abaci B <abaci64 at gmail.com <mailto:abaci64 at gmail.com>> wrote:
> is there anything in the logs when this happens?
> any reason not to play the files using file-string:///path/to/file1!/path/to/file2!/path/to/file3 ? not sure if it would mke a difference but you could try.
> did you also try if you see the same behavior when doing session:streamFile("/path/to/file1") instead of session:execute("playback", "/path/to/file1")
> 
> 
> On Mon, May 18, 2015 at 5:22 PM, John Nash <john.nash778 at gmail.com <mailto:john.nash778 at gmail.com>> wrote:
> I am trying to play some native files (Preencoded) in lua script and calling the lua script in dialpan like below..
> 
> session:execute("playback", <path to file1>);
> session:execute("playback", <path to file2>);
> session:execute("playback", <path to file3>);
> 
> When I make call I hear file1 clearly but after that for approx 10-15 seconds I do not hear anything (except some occasional noise like breaking voice) 
> 
> I tried commenting file1 (to check if file1 is corrupt or something) but then I hear file2 clearly and not file 3 for 10-15 seconds. I also tried session:sleep(100) between playbacks but issues remains same.
> 
> Am I doing something wrong?
> 
> _________________________________________________________________________
> 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
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From lexxua at gmail.com  Tue May 19 23:18:20 2015
From: lexxua at gmail.com (Volodymyr Fedorov)
Date: Tue, 19 May 2015 22:18:20 +0300
Subject: [Freeswitch-users] Monitoring SIP Service
In-Reply-To: <CACOYK=hkth4KN+moo3g1oW1bgV-i0gDaW7mnpU+yhnVyv=8tqA@mail.gmail.com>
References: <CAPfjz3iY9pdMsKDu1OnhO+McTY-ZwaKknfc=3CZcJiuUJ5yN3g@mail.gmail.com>
	<CACOYK=gmmo4tvgej8ZXZPEDHasmz6H5wa2Yu9Yga4pa6MMDx+A@mail.gmail.com>
	<CAO_GGAp-4eWSvVm61wqbu_iNeCcF07K9pSdzVPE=4Q-y0TccYg@mail.gmail.com>
	<CACOYK=hkth4KN+moo3g1oW1bgV-i0gDaW7mnpU+yhnVyv=8tqA@mail.gmail.com>
Message-ID: <CAO_GGAqrJO73y0TOBam6m-Rj2aA_E0xiY9XVzHeQFgeEFgRByw@mail.gmail.com>

Hello, I`m pushed our little python script to Github
https://github.com/lexxua/qcheck it`s not ideal but makes dirty job. Maybe
if I have some more time I will rewrite it for mod_bert usage.

On Tue, May 19, 2015 at 1:47 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> Hmm, quite interesting. I think we can develop a testing tool based on
> this, and make it royalty free. And make it suitable for transcoding and
> analog paths.
> On May 19, 2015 12:36 PM, "???????? ???????" <lexxua at gmail.com> wrote:
>
>> In my company we use Yate  analyzer module
>> as a call quality monitor
>> http://docs.yate.ro/wiki/Quality_analyzer_module . Results of tests
>> stored in Influxdb.
>> For gateway monitoring we use built in Freeswitch logging of DOWN state
>> of gateway.
>>
>>
>>
>> On Tue, May 19, 2015 at 12:04 PM Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>>
>>> I'm working on two different projects for SIP testing and monitoring,
>>> and soon there will be more information.
>>>
>>> https://github.com/voxserv/rring
>>> (documentation is still missing) this will be an automated tester that
>>> uses FreeSWITCH to originate and terminate the calls, and it will
>>> analyze the SIP messages that are received from remote side.
>>>
>>>
>>> https://txlab.wordpress.com/2015/05/14/quality-assurance-for-voip-calls/
>>> some scripts and work in progress for voice quality assurance.
>>>
>>> I will make a separate posting as soon as I'm ready.
>>>
>>>
>>>
>>>
>>> On Tue, May 19, 2015 at 3:27 AM, Jai Rangi <jprangi at didforsale.com>
>>> wrote:
>>> > Very common concerns from new Asterisk, Freeswitch, opensips and
>>> freepbx
>>> > owners, How can we monitor system, what happens if service stops
>>> responding.
>>> >
>>> > Here is a small howto on monitoring any SIP service with nagios. I am
>>> sure
>>> > there are plenty of options and suggestions, but this is one of them
>>> and has
>>> > been working out very well for us for years.
>>> >
>>> > http://www.didforsale.com/monitor-sip-server
>>> >
>>> > Best,
>>> > -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
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Cheers !
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From nasida at live.ru  Wed May 20 00:53:03 2015
From: nasida at live.ru (Yuriy Nasida)
Date: Tue, 19 May 2015 23:53:03 +0300
Subject: [Freeswitch-users] unexpected segfault with latest debian and
 libmyodbc.so
In-Reply-To: <CAEJMVkDTEgoa51dny8PiyvGiy6aAcvqG3vsVvJs8jyjezud==g@mail.gmail.com>
References: <DUB122-W36F85086474E2280B9ED15D5090@phx.gbl>,
	<CAJR7f2HCFX6wOTxNStRVO8by4x7HiSUo9zQf-8566SzqqsDFvA@mail.gmail.com>,
	<CAKbxfG8+=HVT_D7FqUwdynRa8-8KEvkOVzuPnjF6tnmZipuvQA@mail.gmail.com>,
	<DUB122-W720E4C2E410A9EA55250CD5F20@phx.gbl>,
	<CAJR7f2FdFFqUEdTpQSN59uzVg_p3nq_kBj5FvDYtM2pUieDeOA@mail.gmail.com>,
	<DUB122-W425F87363588D683B60CC6D5C50@phx.gbl>,
	<CAEJMVkDTEgoa51dny8PiyvGiy6aAcvqG3vsVvJs8jyjezud==g@mail.gmail.com>
Message-ID: <DUB122-W29A96178CDC4BF3EAC2355D5C30@phx.gbl>

I had Threading = 0 always. 
I have updated FS to 1.4.18 and it's good so far.



Date: Mon, 18 May 2015 13:48:53 -0500
From: brian at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] unexpected segfault with latest debian and	libmyodbc.so

Did you try Threading = 0?
On Sun, May 17, 2015 at 9:44 AM, Yuriy Nasida <nasida at live.ru> wrote:



Guys, unfortunately updating of unixODBC didn't help and I getting segfaults again after some quiet period.

Now I have: 
1) unixODBC 2.3.1
2) FreeSWITCH Version 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 18:59:51Z 64bit)
3) Debian GNU/Linux 7.8 (wheezy)
4) libmyodbc  5.1.10-2+deb7u1

Please advice
Thanks




Date: Thu, 2 Apr 2015 16:19:13 -0300
From: italorossib at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] unexpected segfault with latest debian and	libmyodbc.so

Great! :)

On Thu, Apr 2, 2015 at 2:17 PM, Yuriy Nasida <nasida at live.ru> wrote:



I have updated unixodbc to 2.3 (from source) and looks like it fixed the problem. Thanks guys!



Date: Fri, 27 Mar 2015 17:39:55 -0500
From: anthony.minessale at gmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] unexpected segfault with latest debian and	libmyodbc.so

We actually recommend Threading=0 for postgres and mysql for the most part.

On Fri, Mar 27, 2015 at 7:48 AM, ?talo Rossi <italorossib at gmail.com> wrote:
I have seen a lot of threading issues with mysql + odbc, make sure you're using unixodbc >= 2.3. 
If you're using an older version you can set Threading = 2 in your /etc/odbcinst.ini as a workaround, but this is *not* recommended for production/high volume, upgrade as soon as possible.
On Fri, Mar 27, 2015 at 9:41 AM, Yuriy Nasida <nasida at live.ru> wrote:



Hi guys,

I just got unexpected segfault I try to understand if anybody had similar problems.

Mar 26 07:00:19 kernel: [226389.252971] freeswitch[7972]: segfault at 500 ip 00007fd3a8362252 sp 00007fd34a7d9f70 error 4 in libmyodbc.so[7fd3a8340000+3c000]


# lsb_release -a
Distributor ID:    Debian
Description:    Debian GNU/Linux 7.8 (wheezy)
Release:    7.8
Codename:    wheezy

freeswitch at internal> version 
FreeSWITCH Version 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 18:59:51Z 64bit)

# apt-cache show libmyodbc
Package: libmyodbc
Source: myodbc
Version: 5.1.10-2+deb7u1

Please advice,
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


-- 
?talo Rossi



_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

http://www.cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at 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


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

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

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

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

http://www.cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at 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



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

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

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

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

http://www.cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at 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
ClueCon 2015 Call for Speakers | Register TODAY! | 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 		 	   		  
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From schoch+freeswitch.org at xwin32.com  Wed May 20 01:49:47 2015
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Tue, 19 May 2015 14:49:47 -0700
Subject: [Freeswitch-users] Should I switch to Debian?
Message-ID: <CAOkuKdf54aC+CAfxH9day2BxzCK9rNVXDqkBu26Atqc4Xfokhg@mail.gmail.com>

When I set up our office PBX using FreeSwitch, I chose CentOS because I had
used it before. However, it seems that FreeSwitch is better supported on
Debian, at least for updated binary installations. It's not that big a deal
to make a new VM and move my configuration. Would it make it easier on me
in the long run to switch to a Debian machine?

-- 
Steve
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From abalashov at evaristesys.com  Wed May 20 01:54:51 2015
From: abalashov at evaristesys.com (Alex Balashov)
Date: Tue, 19 May 2015 17:54:51 -0400
Subject: [Freeswitch-users] Should I switch to Debian?
In-Reply-To: <CAOkuKdf54aC+CAfxH9day2BxzCK9rNVXDqkBu26Atqc4Xfokhg@mail.gmail.com>
References: <CAOkuKdf54aC+CAfxH9day2BxzCK9rNVXDqkBu26Atqc4Xfokhg@mail.gmail.com>
Message-ID: <20150519215451.5419088.99834.62272@evaristesys.com>

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From carlos.ruizdiaz at gmail.com  Wed May 20 02:02:44 2015
From: carlos.ruizdiaz at gmail.com (=?UTF-8?Q?Carlos_Ruiz_D=C3=ADaz?=)
Date: Tue, 19 May 2015 17:02:44 -0500
Subject: [Freeswitch-users] Should I switch to Debian?
In-Reply-To: <20150519215451.5419088.99834.62272@evaristesys.com>
References: <CAOkuKdf54aC+CAfxH9day2BxzCK9rNVXDqkBu26Atqc4Xfokhg@mail.gmail.com>
	<20150519215451.5419088.99834.62272@evaristesys.com>
Message-ID: <CA+3gBcmxYstnLNfMi6+V0TtAe2k-PLkzJw6GbedTi9x88jHzaQ@mail.gmail.com>

+1 on Alex's approach.

I used to use RPMs, but constantly waiting for them to be updated turned
into a PITA, so I switched to source code as well.

Regards,
Carlos
On May 19, 2015 16:56, "Alex Balashov" <abalashov at evaristesys.com> wrote:

> My personal view is that distro packaging for Freeswitch, Asterisk,
> Kamailio, and most OSS VoIP elements is unreliable, inconsistent and
> lopsided, so the only viable approach is to build these things from source,
> always. This allows one to maintain granular version controland to apply
> ?patches as needed.
>
> If you just forget packaging and learn to love the source, you'll obviate
> a vast category of problems and not waste time on fighting with packages.
> It also has the positive side effect that you can support pretty much any
> server distribution. Debian, CentOS, RHEL, Ubuntu server, who cares?
>
>  --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> Sent from my BlackBerry.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 20 02:34:56 2015
From: jprangi at didforsale.com (Jai Rangi)
Date: Tue, 19 May 2015 15:34:56 -0700
Subject: [Freeswitch-users] Should I switch to Debian?
In-Reply-To: <CA+3gBcmxYstnLNfMi6+V0TtAe2k-PLkzJw6GbedTi9x88jHzaQ@mail.gmail.com>
References: <CAOkuKdf54aC+CAfxH9day2BxzCK9rNVXDqkBu26Atqc4Xfokhg@mail.gmail.com>
	<20150519215451.5419088.99834.62272@evaristesys.com>
	<CA+3gBcmxYstnLNfMi6+V0TtAe2k-PLkzJw6GbedTi9x88jHzaQ@mail.gmail.com>
Message-ID: <CAPfjz3jAu7StBDU7Fa09bVOmwxfD56gUCLtwtwBEv3+khHHWYw@mail.gmail.com>

I agree with Alex, Linux is Linux, I have used Redhat, Fedora. CentOS
Ubuntu, it really does not matter.

-J

On Tue, May 19, 2015 at 3:02 PM, Carlos Ruiz D?az <carlos.ruizdiaz at gmail.com
> wrote:

> +1 on Alex's approach.
>
> I used to use RPMs, but constantly waiting for them to be updated turned
> into a PITA, so I switched to source code as well.
>
> Regards,
> Carlos
> On May 19, 2015 16:56, "Alex Balashov" <abalashov at evaristesys.com> wrote:
>
>> My personal view is that distro packaging for Freeswitch, Asterisk,
>> Kamailio, and most OSS VoIP elements is unreliable, inconsistent and
>> lopsided, so the only viable approach is to build these things from source,
>> always. This allows one to maintain granular version controland to apply
>> ?patches as needed.
>>
>> If you just forget packaging and learn to love the source, you'll obviate
>> a vast category of problems and not waste time on fighting with packages.
>> It also has the positive side effect that you can support pretty much any
>> server distribution. Debian, CentOS, RHEL, Ubuntu server, who cares?
>>
>>  --
>> Alex Balashov | Principal | Evariste Systems LLC
>> 303 Perimeter Center North, Suite 300
>> Atlanta, GA 30346
>> United States
>>
>> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>
>> Sent from my BlackBerry.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.medina at cibersys.com  Wed May 20 02:42:04 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Tue, 19 May 2015 18:12:04 -0430
Subject: [Freeswitch-users] JsSIP -> OverSIP -> Freeswitch
Message-ID: <CACXsapPPc2jti7K0H0C4nm6Sw1bMTzAt70SoqNish=uuyBk=qw@mail.gmail.com>

Hi guys!

Im having some problems while trying to connect to a Freswitch PBX using
OverSip and JsSIP.

Using Jssip i can recive calls but when calling from the OverSIP/JsSIp I
always get a 422 error.

Softphone -> WebRTC ext calls OK
WebRTC -> Softphone fails
WebRTC -> ECHO TEST call on freeswitch also fails

Can somebody help find out what this could be?

SIP/2.0 422 Session Interval Too Small
Via: SIP/2.0/WSS vv3f75qcos24.invalid;branch=z9hG4bK1673051
From: "test" <sip:1002 at conference.cibersys.com>;tag=5or80k3oc9
To: <sip:9196 at conference.cibersys.com>;tag=Q3Xyt4X2vSX6Q
Call-ID: 4norlmjdkqok6smt0vd4
CSeq: 9416 INVITE
User-Agent:
FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Min-Se: 120
Content-Length: 0

BTW... already asked @oversip guys... But wanted to try this in here just
in case! =)

Thanks.

-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From brian at freeswitch.org  Wed May 20 02:45:19 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 May 2015 17:45:19 -0500
Subject: [Freeswitch-users] JsSIP -> OverSIP -> Freeswitch
In-Reply-To: <CACXsapPPc2jti7K0H0C4nm6Sw1bMTzAt70SoqNish=uuyBk=qw@mail.gmail.com>
References: <CACXsapPPc2jti7K0H0C4nm6Sw1bMTzAt70SoqNish=uuyBk=qw@mail.gmail.com>
Message-ID: <CAEJMVkDBv2XmAHhySS_faVTrDo6K2dgy3+OEL7k=ER33dT7kGg@mail.gmail.com>

The whole sip exchange would need to be seen for me to tell, it could be a
bug, it could be a config issue... not exactly sure yet.  Can you paste bin
it?

Thanks,

On Tue, May 19, 2015 at 5:42 PM, Victor Medina <victor.medina at cibersys.com>
wrote:

> Hi guys!
>
> Im having some problems while trying to connect to a Freswitch PBX using
> OverSip and JsSIP.
>
> Using Jssip i can recive calls but when calling from the OverSIP/JsSIp I
> always get a 422 error.
>
> Softphone -> WebRTC ext calls OK
> WebRTC -> Softphone fails
> WebRTC -> ECHO TEST call on freeswitch also fails
>
> Can somebody help find out what this could be?
>
> SIP/2.0 422 Session Interval Too Small
> Via: SIP/2.0/WSS vv3f75qcos24.invalid;branch=z9hG4bK1673051
> From: "test" <sip:1002 at conference.cibersys.com>;tag=5or80k3oc9
> To: <sip:9196 at conference.cibersys.com>;tag=Q3Xyt4X2vSX6Q
> Call-ID: 4norlmjdkqok6smt0vd4
> CSeq: 9416 INVITE
> User-Agent:
> FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
> line-seize, call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
> Min-Se: 120
> Content-Length: 0
>
> BTW... already asked @oversip guys... But wanted to try this in here just
> in case! =)
>
> Thanks.
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From brian at freeswitch.org  Wed May 20 02:50:53 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 May 2015 17:50:53 -0500
Subject: [Freeswitch-users] Should I switch to Debian?
In-Reply-To: <20150519215451.5419088.99834.62272@evaristesys.com>
References: <CAOkuKdf54aC+CAfxH9day2BxzCK9rNVXDqkBu26Atqc4Xfokhg@mail.gmail.com>
	<20150519215451.5419088.99834.62272@evaristesys.com>
Message-ID: <CAEJMVkDcjm+Y==BbHQjMRENXpeL0Po42_fKJRSfSimcfJ05wrQ@mail.gmail.com>

I build from source because I'm one of the core devs, but end users cant
NOT be expected to do that, I'd love to have FreeSWITCH in multiple linux
distros, and we're working on exactly that.  Anyone wishing to help in that
area could join us on tomorrows ClueCon Weekly,  We'll be talking about 1.6
and some of these items.

Also there are companies that will not use source due to policies.

Thanks,


On Tue, May 19, 2015 at 4:54 PM, Alex Balashov <abalashov at evaristesys.com>
wrote:

> My personal view is that distro packaging for Freeswitch, Asterisk,
> Kamailio, and most OSS VoIP elements is unreliable, inconsistent and
> lopsided, so the only viable approach is to build these things from source,
> always. This allows one to maintain granular version controland to apply
> ?patches as needed.
>
> If you just forget packaging and learn to love the source, you'll obviate
> a vast category of problems and not waste time on fighting with packages.
> It also has the positive side effect that you can support pretty much any
> server distribution. Debian, CentOS, RHEL, Ubuntu server, who cares?
>
>  --
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> Sent from my BlackBerry.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From brian at freeswitch.org  Wed May 20 02:53:25 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 May 2015 17:53:25 -0500
Subject: [Freeswitch-users] newbie: originate dial problem
In-Reply-To: <CAPwL2MDMJNQJuqYbijckyLDrqsfjBU0C_p5d1nf2ZnKFod5Eow@mail.gmail.com>
References: <CAPwL2MC--nb2s6WjL-udZZHoCAMMPBhQv8ELpS6ucRpADmNLJg@mail.gmail.com>
	<CAEJMVkCWxk=d7roEvZ8mroQ+rjy+yeujOEcOJ3RoLCXZP6grqQ@mail.gmail.com>
	<CAPwL2MDMJNQJuqYbijckyLDrqsfjBU0C_p5d1nf2ZnKFod5Eow@mail.gmail.com>
Message-ID: <CAEJMVkDMyvdU0Hoa9-_+cLadw1fHH6h0Mw+cP1P86xe1e8NYNA@mail.gmail.com>

sofia status profile internal reg
sofia status profile external reg

What do those two commands say?

On Tue, May 19, 2015 at 9:32 AM, Masakazu Nakano <emplant2000 at gmail.com>
wrote:

> HI Brian
>
> thank you for your reply.
>
> 'sofia status' says following like  that.
>
> freeswitch at internal> sofia status
>                      Name          Type
>     Data      State
>
> =================================================================================================
>             external-ipv6       profile                   sip:mod_sofia@[::2]:5060
>      RUNNING (0)
>             27.112.104.17         alias
> internal      ALIASED
>                  external       profile
> sip:mod_sofia at 27.112.104.17:5060      RUNNING (0)
>           external::iptel       gateway
> sip:emplant2000 at sip.iptel.org      FAIL_WAIT
>             internal-ipv6       profile                   sip:mod_sofia@[::2]:5090
>      RUNNING (0)
>                  internal       profile
> sip:mod_sofia at fusionpbx.b3am.com:5090      RUNNING (0)
>
> =================================================================================================
> 4 profiles 1 alias
>
> BR
>
> mack
>
>
>
> 2015-05-19 22:49 GMT+09:00 Brian West <brian at freeswitch.org>:
>
>> What rev of FreeSWITCH and what is the output of 'sofia status',   Then
>> you can try sofia_contact */5630 and see what it returns, and pastbin all
>> of that info for us to see.
>>
>>
>>
>> On Tue, May 19, 2015 at 8:11 AM, Masakazu Nakano <emplant2000 at gmail.com>
>> wrote:
>>
>>> Hi there.
>>>
>>> a softphone is registed called 5630 in public.
>>>
>>> show registered says following under like that.
>>>
>>> 5630,27.112.104.17,siGhOxWCoIW0n9JjzwxSTg..,sofia/external/sip:5630 at 49.97.4.168:61682
>>> ;transport=UDP;rinstance=b1c4cf82f7d6e637,1432040746,49.97.4.168,61682,udp,
>>> b3am.com,
>>>
>>> and
>>>
>>> freeswitch at internal > originate user/5630
>>>
>>> is successfull.
>>>
>>> but by dialplan/public/5630.xml
>>>    <extension name="5630">
>>>      <condition field="destination_number" expression="5630">
>>>        <action application="bridge" data="user/5630"/>
>>>      </condition>
>>>    </extension>
>>>
>>> and make a call with another phone to 5630.to be "Cause:
>>> INVALID_PROFILE"
>>>
>>> 2015-05-19 22:08:53.724519 [NOTICE] switch_ivr_originate.c:2732 Cannot
>>> create outgoing channel of type [sofia] cause: [INVALID_PROFILE]
>>> 2015-05-19 22:08:53.724519 [DEBUG] switch_ivr_originate.c:3720 Originate
>>> Resulted in Error Cause: 611 [INVALID_PROFILE]
>>> 2015-05-19 22:08:53.724519 [INFO] mod_dptools.c:3246 Originate Failed.
>>> Cause: INVALID_PROFILE
>>>
>>> why?
>>>
>>> BR
>>>
>>> mack
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>> <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 victor.medina at cibersys.com  Wed May 20 02:58:45 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Tue, 19 May 2015 18:28:45 -0430
Subject: [Freeswitch-users] JsSIP -> OverSIP -> Freeswitch
In-Reply-To: <CAEJMVkDBv2XmAHhySS_faVTrDo6K2dgy3+OEL7k=ER33dT7kGg@mail.gmail.com>
References: <CACXsapPPc2jti7K0H0C4nm6Sw1bMTzAt70SoqNish=uuyBk=qw@mail.gmail.com>
	<CAEJMVkDBv2XmAHhySS_faVTrDo6K2dgy3+OEL7k=ER33dT7kGg@mail.gmail.com>
Message-ID: <CACXsapPGxPT3mGvhTMeqk_QTuPXkGj7ibDJ91G8c9rGpxop7WA@mail.gmail.com>

SURE!

This is the Sofia Internal side...

http://pastebin.com/xwBjzMCX

This is Javascript console... (JsSIP -> OverSIP)

http://pastebin.com/XmXRt0nV

Thanks.

2015-05-19 18:15 GMT-04:30 Brian West <brian at freeswitch.org>:

> The whole sip exchange would need to be seen for me to tell, it could be a
> bug, it could be a config issue... not exactly sure yet.  Can you paste bin
> it?
>
> Thanks,
>
> On Tue, May 19, 2015 at 5:42 PM, Victor Medina <victor.medina at cibersys.com
> > wrote:
>
>> Hi guys!
>>
>> Im having some problems while trying to connect to a Freswitch PBX using
>> OverSip and JsSIP.
>>
>> Using Jssip i can recive calls but when calling from the OverSIP/JsSIp I
>> always get a 422 error.
>>
>> Softphone -> WebRTC ext calls OK
>> WebRTC -> Softphone fails
>> WebRTC -> ECHO TEST call on freeswitch also fails
>>
>> Can somebody help find out what this could be?
>>
>> SIP/2.0 422 Session Interval Too Small
>> Via: SIP/2.0/WSS vv3f75qcos24.invalid;branch=z9hG4bK1673051
>> From: "test" <sip:1002 at conference.cibersys.com>;tag=5or80k3oc9
>> To: <sip:9196 at conference.cibersys.com>;tag=Q3Xyt4X2vSX6Q
>> Call-ID: 4norlmjdkqok6smt0vd4
>> CSeq: 9416 INVITE
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
>> Accept: application/sdp
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, path, replaces
>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
>> line-seize, call-info, sla, include-session-description, presence.winfo,
>> message-summary, refer
>> Min-Se: 120
>> Content-Length: 0
>>
>> BTW... already asked @oversip guys... But wanted to try this in here just
>> in case! =)
>>
>> Thanks.
>>
>> --
>>
>>
>>
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561
>> BB #79A8AFA2
>> @VMCibersys
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From emplant2000 at gmail.com  Wed May 20 03:11:07 2015
From: emplant2000 at gmail.com (Masakazu Nakano)
Date: Wed, 20 May 2015 08:11:07 +0900
Subject: [Freeswitch-users] newbie: originate dial problem
In-Reply-To: <CAEJMVkDMyvdU0Hoa9-_+cLadw1fHH6h0Mw+cP1P86xe1e8NYNA@mail.gmail.com>
References: <CAPwL2MC--nb2s6WjL-udZZHoCAMMPBhQv8ELpS6ucRpADmNLJg@mail.gmail.com>
	<CAEJMVkCWxk=d7roEvZ8mroQ+rjy+yeujOEcOJ3RoLCXZP6grqQ@mail.gmail.com>
	<CAPwL2MDMJNQJuqYbijckyLDrqsfjBU0C_p5d1nf2ZnKFod5Eow@mail.gmail.com>
	<CAEJMVkDMyvdU0Hoa9-_+cLadw1fHH6h0Mw+cP1P86xe1e8NYNA@mail.gmail.com>
Message-ID: <CAPwL2MAWHSSbBn5zJ2BDnnFPtQ66OhOCto=boLbX_yjNV2v8iw@mail.gmail.com>

Hi Brian

thank you for my reply.

sorry too long result

freeswitch at internal> sofia status profile internal
=================================================================================================
Name                    internal
Domain Name             N/A
Auto-NAT                false
DBName                  sofia_reg_internal
Pres Hosts              27.112.104.17,27.112.104.17,fusionpbx.b3am.com
Dialplan                XML
Context                 public
Challenge Realm         auto_from
RTP-IP                  fusionpbx.b3am.com
SIP-IP                  fusionpbx.b3am.com
URL                     sip:mod_sofia at fusionpbx.b3am.com:5090
BIND-URL                sip:mod_sofia at fusionpbx.b3am.com:5090
;transport=udp,tcp
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               OPUS,speex at 16000h@20i,speex at 8000h@20i,G7221 at 32000h
,G7221 at 16000h,G722,PCMU,PCMA,GSM
CODECS OUT              OPUS,speex at 16000h@20i,speex at 8000h@20i,G7221 at 32000h
,G7221 at 16000h,G722,PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                true
PROXY-MEDIA             false
ZRTP-PASSTHRU           true
AGGRESSIVENAT           false
CALLS-IN                0
FAILED-CALLS-IN         0
CALLS-OUT               0
FAILED-CALLS-OUT        0
REGISTRATIONS           0

-- and

freeswitch at internal> sofia status profile external
=================================================================================================
Name                    external
Domain Name             N/A
Auto-NAT                false
DBName                  share_presence
Pres Hosts              27.112.104.17
Dialplan                XML
Context                 public
Challenge Realm         auto_to
RTP-IP                  27.112.104.17
SIP-IP                  27.112.104.17
URL                     sip:mod_sofia at 27.112.104.17:5060
BIND-URL                sip:mod_sofia at 27.112.104.17:5060;transport=udp,tcp
WS-BIND-URL             sip:mod_sofia at 27.112.104.17:5066;transport=ws
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS IN               OPUS,speex at 16000h@20i,speex at 8000h@20i,G7221 at 32000h
,G7221 at 16000h,G722,PCMU,PCMA,GSM
CODECS OUT              speex at 16000h@20i,PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               info
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                true
PROXY-MEDIA             false
ZRTP-PASSTHRU           true
AGGRESSIVENAT           false
CALLS-IN                72
FAILED-CALLS-IN         72
CALLS-OUT               0
FAILED-CALLS-OUT        0
REGISTRATIONS           3

# oops, I,I,I'm sorry to...I forget...what I can say...
# I touched asterisk so many time but 15 years spend...
# pls,pls auhhggg....

BR

mack






2015-05-20 7:53 GMT+09:00 Brian West <brian at freeswitch.org>:

> sofia status profile internal reg
> sofia status profile external reg
>
> What do those two commands say?
>
> On Tue, May 19, 2015 at 9:32 AM, Masakazu Nakano <emplant2000 at gmail.com>
> wrote:
>
>> HI Brian
>>
>> thank you for your reply.
>>
>> 'sofia status' says following like  that.
>>
>> freeswitch at internal> sofia status
>>                      Name          Type
>>     Data      State
>>
>> =================================================================================================
>>             external-ipv6       profile                   sip:mod_sofia@[::2]:5060
>>      RUNNING (0)
>>             27.112.104.17         alias
>> internal      ALIASED
>>                  external       profile
>> sip:mod_sofia at 27.112.104.17:5060      RUNNING (0)
>>           external::iptel       gateway
>> sip:emplant2000 at sip.iptel.org      FAIL_WAIT
>>             internal-ipv6       profile                   sip:mod_sofia@[::2]:5090
>>      RUNNING (0)
>>                  internal       profile
>> sip:mod_sofia at fusionpbx.b3am.com:5090      RUNNING (0)
>>
>> =================================================================================================
>> 4 profiles 1 alias
>>
>> BR
>>
>> mack
>>
>>
>>
>> 2015-05-19 22:49 GMT+09:00 Brian West <brian at freeswitch.org>:
>>
>>> What rev of FreeSWITCH and what is the output of 'sofia status',   Then
>>> you can try sofia_contact */5630 and see what it returns, and pastbin all
>>> of that info for us to see.
>>>
>>>
>>>
>>> On Tue, May 19, 2015 at 8:11 AM, Masakazu Nakano <emplant2000 at gmail.com>
>>> wrote:
>>>
>>>> Hi there.
>>>>
>>>> a softphone is registed called 5630 in public.
>>>>
>>>> show registered says following under like that.
>>>>
>>>> 5630,27.112.104.17,siGhOxWCoIW0n9JjzwxSTg..,sofia/external/sip:5630 at 49.97.4.168:61682
>>>> ;transport=UDP;rinstance=b1c4cf82f7d6e637,1432040746,49.97.4.168,61682,udp,
>>>> b3am.com,
>>>>
>>>> and
>>>>
>>>> freeswitch at internal > originate user/5630
>>>>
>>>> is successfull.
>>>>
>>>> but by dialplan/public/5630.xml
>>>>    <extension name="5630">
>>>>      <condition field="destination_number" expression="5630">
>>>>        <action application="bridge" data="user/5630"/>
>>>>      </condition>
>>>>    </extension>
>>>>
>>>> and make a call with another phone to 5630.to be "Cause:
>>>> INVALID_PROFILE"
>>>>
>>>> 2015-05-19 22:08:53.724519 [NOTICE] switch_ivr_originate.c:2732 Cannot
>>>> create outgoing channel of type [sofia] cause: [INVALID_PROFILE]
>>>> 2015-05-19 22:08:53.724519 [DEBUG] switch_ivr_originate.c:3720
>>>> Originate Resulted in Error Cause: 611 [INVALID_PROFILE]
>>>> 2015-05-19 22:08:53.724519 [INFO] mod_dptools.c:3246 Originate Failed.
>>>> Cause: INVALID_PROFILE
>>>>
>>>> why?
>>>>
>>>> BR
>>>>
>>>> mack
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>>> <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> ClueCon 2015 Call for Speakers
> <https://www.cluecon.com/call-for-speakers/> | Register
> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 mike at jerris.com  Wed May 20 03:20:23 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 May 2015 19:20:23 -0400
Subject: [Freeswitch-users] JsSIP -> OverSIP -> Freeswitch
In-Reply-To: <CACXsapPGxPT3mGvhTMeqk_QTuPXkGj7ibDJ91G8c9rGpxop7WA@mail.gmail.com>
References: <CACXsapPPc2jti7K0H0C4nm6Sw1bMTzAt70SoqNish=uuyBk=qw@mail.gmail.com>
	<CAEJMVkDBv2XmAHhySS_faVTrDo6K2dgy3+OEL7k=ER33dT7kGg@mail.gmail.com>
	<CACXsapPGxPT3mGvhTMeqk_QTuPXkGj7ibDJ91G8c9rGpxop7WA@mail.gmail.com>
Message-ID: <8A1140F3-33E1-4F91-A639-A6BCED23FC98@jerris.com>

what is oversip?  you shouldn't need any layers between FS and jssip.

> On May 19, 2015, at 6:58 PM, Victor Medina <victor.medina at cibersys.com> wrote:
> 
> SURE!
> 
> This is the Sofia Internal side...
> 
> http://pastebin.com/xwBjzMCX <http://pastebin.com/xwBjzMCX>
> 
> This is Javascript console... (JsSIP -> OverSIP)
> 
> http://pastebin.com/XmXRt0nV <http://pastebin.com/XmXRt0nV>
> 
> Thanks.
> 
> 2015-05-19 18:15 GMT-04:30 Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>>:
> The whole sip exchange would need to be seen for me to tell, it could be a bug, it could be a config issue... not exactly sure yet.  Can you paste bin it?
> 
> Thanks,
> 
> On Tue, May 19, 2015 at 5:42 PM, Victor Medina <victor.medina at cibersys.com <mailto:victor.medina at cibersys.com>> wrote:
> Hi guys!
> 
> Im having some problems while trying to connect to a Freswitch PBX using OverSip and JsSIP. 
> 
> Using Jssip i can recive calls but when calling from the OverSIP/JsSIp I always get a 422 error. 
> 
> Softphone -> WebRTC ext calls OK
> WebRTC -> Softphone fails
> WebRTC -> ECHO TEST call on freeswitch also fails
> 
> Can somebody help find out what this could be?
> 
> SIP/2.0 422 Session Interval Too Small
> Via: SIP/2.0/WSS vv3f75qcos24.invalid;branch=z9hG4bK1673051
> From: "test" <sip:1002 at conference.cibersys.com <mailto:sip%3A1002 at conference.cibersys.com>>;tag=5or80k3oc9
> To: <sip:9196 at conference.cibersys.com <mailto:sip%3A9196 at conference.cibersys.com>>;tag=Q3Xyt4X2vSX6Q
> Call-ID: 4norlmjdkqok6smt0vd4
> CSeq: 9416 INVITE
> User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
> Min-Se: 120
> Content-Length: 0
> 
> BTW... already asked @oversip guys... But wanted to try this in here just in case! =)
> 
> Thanks.
> 
> -- 
> 
> 
> 
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561 <>
> BB #79A8AFA2
> @VMCibersys
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
> ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
> T:+19184209001 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <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/>
> 
> 
> 
> -- 
> 
> 
> 
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561 <>
> BB #79A8AFA2
> @VMCibersys
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.medina at cibersys.com  Wed May 20 03:34:03 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Tue, 19 May 2015 19:04:03 -0430
Subject: [Freeswitch-users] JsSIP -> OverSIP -> Freeswitch
In-Reply-To: <8A1140F3-33E1-4F91-A639-A6BCED23FC98@jerris.com>
References: <CACXsapPPc2jti7K0H0C4nm6Sw1bMTzAt70SoqNish=uuyBk=qw@mail.gmail.com>
	<CAEJMVkDBv2XmAHhySS_faVTrDo6K2dgy3+OEL7k=ER33dT7kGg@mail.gmail.com>
	<CACXsapPGxPT3mGvhTMeqk_QTuPXkGj7ibDJ91G8c9rGpxop7WA@mail.gmail.com>
	<8A1140F3-33E1-4F91-A639-A6BCED23FC98@jerris.com>
Message-ID: <CACXsapOeMWeauuJW=1hr7FA2Aa9eKx+5yp_70-8MDB8A3eFDJw@mail.gmail.com>

Hi Michael!

This is the the sip debug on the freeswitch directly jssip -> freeswitch ->
7443

freeswitch at internal> sofia profile internal siptrace on
Enabled sip debugging on internal
recv 5122 bytes from wss/[208.84.81.64]:57975 at 19:31:56.205487:
   ------------------------------------------------------------------------
   INVITE sip:9196 at conference.cibersys.com SIP/2.0
   Via: SIP/2.0/WSS 7rrpouob7plu.invalid;branch=z9hG4bK1885537
   Max-Forwards: 69
   To: <sip:9196 at conference.cibersys.com>
   From: <sip:1000 at conference.cibersys.com>;tag=i26sin3126
   Call-ID: di8dteiqmu6ofrf75pbd
   CSeq: 6473 INVITE
   X-Can-Renegotiate: true
   Contact: <sip:e3e4td20 at 7rrpouob7plu.invalid;transport=ws;ob>
   Content-Type: application/sdp
   Session-Expires: 90
   Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
   Supported: timer,ice,outbound
   User-Agent: JsSIP 0.6.26
   Content-Length: 4574

   v=0
   o=- 3581044066039556708 2 IN IP4 127.0.0.1
   s=-
   t=0 0
   a=group:BUNDLE audio video
   a=msid-semantic: WMS iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD
   m=audio 55125 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
   c=IN IP4 208.84.81.64
   a=rtcp:55125 IN IP4 208.84.81.64
   a=candidate:1505006532 1 udp 2122255103
2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
   a=candidate:1505006532 2 udp 2122255103
2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
   a=candidate:2999745851 1 udp 2122194687 192.168.56.1 55124 typ host
generation 0
   a=candidate:2999745851 2 udp 2122194687 192.168.56.1 55124 typ host
generation 0
   a=candidate:536231733 1 udp 2122129151 10.0.80.221 55125 typ host
generation 0
   a=candidate:536231733 2 udp 2122129151 10.0.80.221 55125 typ host
generation 0
   a=candidate:389508916 1 tcp 1518275327
2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
   a=candidate:389508916 2 tcp 1518275327
2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
   a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
active generation 0
   a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host tcptype
active generation 0
   a=candidate:1366672325 1 tcp 1518149375 10.0.80.221 0 typ host tcptype
active generation 0
   a=candidate:1366672325 2 tcp 1518149375 10.0.80.221 0 typ host tcptype
active generation 0
   a=candidate:3064574176 1 udp 1685921535 208.84.81.64 55125 typ srflx
raddr 10.0.80.221 rport 55125 generation 0
   a=candidate:3064574176 2 udp 1685921535 208.84.81.64 55125 typ srflx
raddr 10.0.80.221 rport 55125 generation 0
   a=ice-ufrag:1UvloQ6OnbX0r5t4
   a=ice-pwd:mE9PaCEu/IMEtzv9L8YgDRCh
   a=fingerprint:sha-256
61:AC:51:08:4B:E0:AB:B0:89:7B:18:05:CA:49:B3:71:9B:5A:77:99:07:58:67:A9:3E:0E:72:92:13:F5:1E:F7
   a=setup:actpass
   a=mid:audio
   a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
   a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
   a=sendrecv
   a=rtcp-mux
   a=rtpmap:111 opus/48000/2
   a=fmtp:111 minptime=10; useinbandfec=1
   a=rtpmap:103 ISAC/16000
   a=rtpmap:104 ISAC/32000
   a=rtpmap:9 G722/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:106 CN/32000
   a=rtpmap:105 CN/16000
   a=rtpmap:13 CN/8000
   a=rtpmap:126 telephone-event/8000
   a=maxptime:60
   a=ssrc:691038544 cname:WW2sd3POh47MtjWG
   a=ssrc:691038544 msid:iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD
4cca9568-6c68-48b7-bac4-ed658260a073
   a=ssrc:691038544 mslabel:iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD
   a=ssrc:691038544 label:4cca9568-6c68-48b7-bac4-ed658260a073
   m=video 55125 RTP/SAVPF 100 116 117 96
   c=IN IP4 208.84.81.64
   a=rtcp:55125 IN IP4 208.84.81.64
   a=candidate:1505006532 1 udp 2122255103
2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
   a=candidate:1505006532 2 udp 2122255103
2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
   a=candidate:2999745851 1 udp 2122194687 192.168.56.1 55124 typ host
generation 0
   a=candidate:2999745851 2 udp 2122194687 192.168.56.1 55124 typ host
generation 0
   a=candidate:536231733 1 udp 2122129151 10.0.80.221 55125 typ host
generation 0
   a=candidate:536231733 2 udp 2122129151 10.0.80.221 55125 typ host
generation 0
   a=candidate:389508916 1 tcp 1518275327
2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
   a=candidate:389508916 2 tcp 1518275327
2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
   a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
active generation 0
   a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host tcptype
active generation 0
   a=candidate:1366672325 1 tcp 1518149375 10.0.80.221 0 typ host tcptype
active generation 0
   a=candidate:1366672325 2 tcp 1518149375 10.0.80.221 0 typ host tcptype
active generation 0
   a=candidate:3064574176 1 udp 1685921535 208.84.81.64 55125 typ srflx
raddr 10.0.80.221 rport 55125 generation 0
   a=candidate:3064574176 2 udp 1685921535 208.84.81.64 55125 typ srflx
raddr 10.0.80.221 rport 55125 generation 0
   a=ice-ufrag:1UvloQ6OnbX0r5t4
   a=ice-pwd:mE9PaCEu/IMEtzv9L8YgDRCh
   a=fingerprint:sha-256
61:AC:51:08:4B:E0:AB:B0:89:7B:18:05:CA:49:B3:71:9B:5A:77:99:07:58:67:A9:3E:0E:72:92:13:F5:1E:F7
   a=setup:actpass
   a=mid:video
   a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
   a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
   a=recvonly
   a=rtcp-mux
   a=rtpmap:100 VP8/90000
   a=rtcp-fb:100 ccm fir
   a=rtcp-fb:100 nack
   a=rtcp-fb:100 nack pli
   a=rtcp-fb:100 goog-remb
   a=rtpmap:116 red/90000
   a=rtpmap:117 ulpfec/90000
   a=rtpmap:96 rtx/90000
   a=fmtp:96 apt=100
   ------------------------------------------------------------------------
send 754 bytes to wss/[208.84.81.64]:57975 at 19:31:56.205982:
   ------------------------------------------------------------------------
   SIP/2.0 422 Session Interval Too Small
   Via: SIP/2.0/WSS
7rrpouob7plu.invalid;branch=z9hG4bK1885537;received=208.84.81.64;rport=57975
   From: <sip:1000 at conference.cibersys.com>;tag=i26sin3126
   To: <sip:9196 at conference.cibersys.com>;tag=ma466p1ScScUa
   Call-ID: di8dteiqmu6ofrf75pbd
   CSeq: 6473 INVITE
   User-Agent:
FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event,
dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer
   Min-SE: 120
   Content-Length: 0

   ------------------------------------------------------------------------
recv 290 bytes from wss/[208.84.81.64]:57975 at 19:31:56.483525:
   ------------------------------------------------------------------------
   ACK sip:9196 at conference.cibersys.com SIP/2.0
   Via: SIP/2.0/WSS 7rrpouob7plu.invalid;branch=z9hG4bK1885537
   To: <sip:9196 at conference.cibersys.com>;tag=ma466p1ScScUa
   From: <sip:1000 at conference.cibersys.com>;tag=i26sin3126
   Call-ID: di8dteiqmu6ofrf75pbd
   CSeq: 6473 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
freeswitch at internal>


2015-05-19 18:50 GMT-04:30 Michael Jerris <mike at jerris.com>:

> what is oversip?  you shouldn't need any layers between FS and jssip.
>
> On May 19, 2015, at 6:58 PM, Victor Medina <victor.medina at cibersys.com>
> wrote:
>
> SURE!
>
> This is the Sofia Internal side...
>
> http://pastebin.com/xwBjzMCX
>
> This is Javascript console... (JsSIP -> OverSIP)
>
> http://pastebin.com/XmXRt0nV
>
> Thanks.
>
> 2015-05-19 18:15 GMT-04:30 Brian West <brian at freeswitch.org>:
>
>> The whole sip exchange would need to be seen for me to tell, it could be
>> a bug, it could be a config issue... not exactly sure yet.  Can you paste
>> bin it?
>>
>> Thanks,
>>
>> On Tue, May 19, 2015 at 5:42 PM, Victor Medina <
>> victor.medina at cibersys.com> wrote:
>>
>>> Hi guys!
>>>
>>> Im having some problems while trying to connect to a Freswitch PBX using
>>> OverSip and JsSIP.
>>>
>>> Using Jssip i can recive calls but when calling from the OverSIP/JsSIp I
>>> always get a 422 error.
>>>
>>> Softphone -> WebRTC ext calls OK
>>> WebRTC -> Softphone fails
>>> WebRTC -> ECHO TEST call on freeswitch also fails
>>>
>>> Can somebody help find out what this could be?
>>>
>>> SIP/2.0 422 Session Interval Too Small
>>> Via: SIP/2.0/WSS vv3f75qcos24.invalid;branch=z9hG4bK1673051
>>> From: "test" <sip:1002 at conference.cibersys.com>;tag=5or80k3oc9
>>> To: <sip:9196 at conference.cibersys.com>;tag=Q3Xyt4X2vSX6Q
>>> Call-ID: 4norlmjdkqok6smt0vd4
>>> CSeq: 9416 INVITE
>>> User-Agent:
>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
>>> Accept: application/sdp
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> Supported: timer, path, replaces
>>> Allow-Events: talk, hold, conference, presence, as-feature-event,
>>> dialog, line-seize, call-info, sla, include-session-description,
>>> presence.winfo, message-summary, refer
>>> Min-Se: 120
>>> Content-Length: 0
>>>
>>> BTW... already asked @oversip guys... But wanted to try this in here
>>> just in case! =)
>>>
>>> Thanks.
>>>
>>> --
>>>
>>>
>>>
>>> V?ctor E. Medina M.
>>> Platform Architect / Chief Infrastructure
>>> +58424 291 4561
>>> BB #79A8AFA2
>>> @VMCibersys
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>>
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From krice at freeswitch.org  Wed May 20 03:41:28 2015
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 19 May 2015 18:41:28 -0500
Subject: [Freeswitch-users] JsSIP -> OverSIP -> Freeswitch
In-Reply-To: <CACXsapOeMWeauuJW=1hr7FA2Aa9eKx+5yp_70-8MDB8A3eFDJw@mail.gmail.com>
References: <CACXsapPPc2jti7K0H0C4nm6Sw1bMTzAt70SoqNish=uuyBk=qw@mail.gmail.com>
	<CAEJMVkDBv2XmAHhySS_faVTrDo6K2dgy3+OEL7k=ER33dT7kGg@mail.gmail.com>
	<CACXsapPGxPT3mGvhTMeqk_QTuPXkGj7ibDJ91G8c9rGpxop7WA@mail.gmail.com>
	<8A1140F3-33E1-4F91-A639-A6BCED23FC98@jerris.com>
	<CACXsapOeMWeauuJW=1hr7FA2Aa9eKx+5yp_70-8MDB8A3eFDJw@mail.gmail.com>
Message-ID: <D2308758-768B-413F-A88B-616D87A51941@freeswitch.org>

Your session time is too short make it longer like 300 secs

Sent from my iPhone

> On May 19, 2015, at 6:34 PM, Victor Medina <victor.medina at cibersys.com> wrote:
> 
> Hi Michael!
> 
> This is the the sip debug on the freeswitch directly jssip -> freeswitch -> 7443
> 
> freeswitch at internal> sofia profile internal siptrace on
> Enabled sip debugging on internal
> recv 5122 bytes from wss/[208.84.81.64]:57975 at 19:31:56.205487:
>    ------------------------------------------------------------------------
>    INVITE sip:9196 at conference.cibersys.com SIP/2.0
>    Via: SIP/2.0/WSS 7rrpouob7plu.invalid;branch=z9hG4bK1885537
>    Max-Forwards: 69
>    To: <sip:9196 at conference.cibersys.com>
>    From: <sip:1000 at conference.cibersys.com>;tag=i26sin3126
>    Call-ID: di8dteiqmu6ofrf75pbd
>    CSeq: 6473 INVITE
>    X-Can-Renegotiate: true
>    Contact: <sip:e3e4td20 at 7rrpouob7plu.invalid;transport=ws;ob>
>    Content-Type: application/sdp
>    Session-Expires: 90
>    Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
>    Supported: timer,ice,outbound
>    User-Agent: JsSIP 0.6.26
>    Content-Length: 4574
> 
>    v=0
>    o=- 3581044066039556708 2 IN IP4 127.0.0.1
>    s=-
>    t=0 0
>    a=group:BUNDLE audio video
>    a=msid-semantic: WMS iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD
>    m=audio 55125 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>    c=IN IP4 208.84.81.64
>    a=rtcp:55125 IN IP4 208.84.81.64
>    a=candidate:1505006532 1 udp 2122255103 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>    a=candidate:1505006532 2 udp 2122255103 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>    a=candidate:2999745851 1 udp 2122194687 192.168.56.1 55124 typ host generation 0
>    a=candidate:2999745851 2 udp 2122194687 192.168.56.1 55124 typ host generation 0
>    a=candidate:536231733 1 udp 2122129151 10.0.80.221 55125 typ host generation 0
>    a=candidate:536231733 2 udp 2122129151 10.0.80.221 55125 typ host generation 0
>    a=candidate:389508916 1 tcp 1518275327 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>    a=candidate:389508916 2 tcp 1518275327 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>    a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0
>    a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0
>    a=candidate:1366672325 1 tcp 1518149375 10.0.80.221 0 typ host tcptype active generation 0
>    a=candidate:1366672325 2 tcp 1518149375 10.0.80.221 0 typ host tcptype active generation 0
>    a=candidate:3064574176 1 udp 1685921535 208.84.81.64 55125 typ srflx raddr 10.0.80.221 rport 55125 generation 0
>    a=candidate:3064574176 2 udp 1685921535 208.84.81.64 55125 typ srflx raddr 10.0.80.221 rport 55125 generation 0
>    a=ice-ufrag:1UvloQ6OnbX0r5t4
>    a=ice-pwd:mE9PaCEu/IMEtzv9L8YgDRCh
>    a=fingerprint:sha-256 61:AC:51:08:4B:E0:AB:B0:89:7B:18:05:CA:49:B3:71:9B:5A:77:99:07:58:67:A9:3E:0E:72:92:13:F5:1E:F7
>    a=setup:actpass
>    a=mid:audio
>    a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>    a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>    a=sendrecv
>    a=rtcp-mux
>    a=rtpmap:111 opus/48000/2
>    a=fmtp:111 minptime=10; useinbandfec=1
>    a=rtpmap:103 ISAC/16000
>    a=rtpmap:104 ISAC/32000
>    a=rtpmap:9 G722/8000
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:106 CN/32000
>    a=rtpmap:105 CN/16000
>    a=rtpmap:13 CN/8000
>    a=rtpmap:126 telephone-event/8000
>    a=maxptime:60
>    a=ssrc:691038544 cname:WW2sd3POh47MtjWG
>    a=ssrc:691038544 msid:iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD 4cca9568-6c68-48b7-bac4-ed658260a073
>    a=ssrc:691038544 mslabel:iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD
>    a=ssrc:691038544 label:4cca9568-6c68-48b7-bac4-ed658260a073
>    m=video 55125 RTP/SAVPF 100 116 117 96
>    c=IN IP4 208.84.81.64
>    a=rtcp:55125 IN IP4 208.84.81.64
>    a=candidate:1505006532 1 udp 2122255103 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>    a=candidate:1505006532 2 udp 2122255103 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>    a=candidate:2999745851 1 udp 2122194687 192.168.56.1 55124 typ host generation 0
>    a=candidate:2999745851 2 udp 2122194687 192.168.56.1 55124 typ host generation 0
>    a=candidate:536231733 1 udp 2122129151 10.0.80.221 55125 typ host generation 0
>    a=candidate:536231733 2 udp 2122129151 10.0.80.221 55125 typ host generation 0
>    a=candidate:389508916 1 tcp 1518275327 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>    a=candidate:389508916 2 tcp 1518275327 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>    a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0
>    a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0
>    a=candidate:1366672325 1 tcp 1518149375 10.0.80.221 0 typ host tcptype active generation 0
>    a=candidate:1366672325 2 tcp 1518149375 10.0.80.221 0 typ host tcptype active generation 0
>    a=candidate:3064574176 1 udp 1685921535 208.84.81.64 55125 typ srflx raddr 10.0.80.221 rport 55125 generation 0
>    a=candidate:3064574176 2 udp 1685921535 208.84.81.64 55125 typ srflx raddr 10.0.80.221 rport 55125 generation 0
>    a=ice-ufrag:1UvloQ6OnbX0r5t4
>    a=ice-pwd:mE9PaCEu/IMEtzv9L8YgDRCh
>    a=fingerprint:sha-256 61:AC:51:08:4B:E0:AB:B0:89:7B:18:05:CA:49:B3:71:9B:5A:77:99:07:58:67:A9:3E:0E:72:92:13:F5:1E:F7
>    a=setup:actpass
>    a=mid:video
>    a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>    a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>    a=recvonly
>    a=rtcp-mux
>    a=rtpmap:100 VP8/90000
>    a=rtcp-fb:100 ccm fir
>    a=rtcp-fb:100 nack
>    a=rtcp-fb:100 nack pli
>    a=rtcp-fb:100 goog-remb
>    a=rtpmap:116 red/90000
>    a=rtpmap:117 ulpfec/90000
>    a=rtpmap:96 rtx/90000
>    a=fmtp:96 apt=100
>    ------------------------------------------------------------------------
> send 754 bytes to wss/[208.84.81.64]:57975 at 19:31:56.205982:
>    ------------------------------------------------------------------------
>    SIP/2.0 422 Session Interval Too Small
>    Via: SIP/2.0/WSS 7rrpouob7plu.invalid;branch=z9hG4bK1885537;received=208.84.81.64;rport=57975
>    From: <sip:1000 at conference.cibersys.com>;tag=i26sin3126
>    To: <sip:9196 at conference.cibersys.com>;tag=ma466p1ScScUa
>    Call-ID: di8dteiqmu6ofrf75pbd
>    CSeq: 6473 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
>    Min-SE: 120
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> recv 290 bytes from wss/[208.84.81.64]:57975 at 19:31:56.483525:
>    ------------------------------------------------------------------------
>    ACK sip:9196 at conference.cibersys.com SIP/2.0
>    Via: SIP/2.0/WSS 7rrpouob7plu.invalid;branch=z9hG4bK1885537
>    To: <sip:9196 at conference.cibersys.com>;tag=ma466p1ScScUa
>    From: <sip:1000 at conference.cibersys.com>;tag=i26sin3126
>    Call-ID: di8dteiqmu6ofrf75pbd
>    CSeq: 6473 ACK
>    Content-Length: 0
> 
>    ------------------------------------------------------------------------
> freeswitch at internal>
> 
> 
> 2015-05-19 18:50 GMT-04:30 Michael Jerris <mike at jerris.com>:
>> what is oversip?  you shouldn't need any layers between FS and jssip.
>> 
>>> On May 19, 2015, at 6:58 PM, Victor Medina <victor.medina at cibersys.com> wrote:
>>> 
>>> SURE!
>>> 
>>> This is the Sofia Internal side...
>>> 
>>> http://pastebin.com/xwBjzMCX
>>> 
>>> This is Javascript console... (JsSIP -> OverSIP)
>>> 
>>> http://pastebin.com/XmXRt0nV
>>> 
>>> Thanks.
>>> 
>>> 2015-05-19 18:15 GMT-04:30 Brian West <brian at freeswitch.org>:
>>>> The whole sip exchange would need to be seen for me to tell, it could be a bug, it could be a config issue... not exactly sure yet.  Can you paste bin it?
>>>> 
>>>> Thanks,
>>>> 
>>>>> On Tue, May 19, 2015 at 5:42 PM, Victor Medina <victor.medina at cibersys.com> wrote:
>>>>> Hi guys!
>>>>> 
>>>>> Im having some problems while trying to connect to a Freswitch PBX using OverSip and JsSIP. 
>>>>> 
>>>>> Using Jssip i can recive calls but when calling from the OverSIP/JsSIp I always get a 422 error. 
>>>>> 
>>>>> Softphone -> WebRTC ext calls OK
>>>>> WebRTC -> Softphone fails
>>>>> WebRTC -> ECHO TEST call on freeswitch also fails
>>>>> 
>>>>> Can somebody help find out what this could be?
>>>>> 
>>>>> SIP/2.0 422 Session Interval Too Small
>>>>> Via: SIP/2.0/WSS vv3f75qcos24.invalid;branch=z9hG4bK1673051
>>>>> From: "test" <sip:1002 at conference.cibersys.com>;tag=5or80k3oc9
>>>>> To: <sip:9196 at conference.cibersys.com>;tag=Q3Xyt4X2vSX6Q
>>>>> Call-ID: 4norlmjdkqok6smt0vd4
>>>>> CSeq: 9416 INVITE
>>>>> User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
>>>>> Accept: application/sdp
>>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>> Supported: timer, path, replaces
>>>>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
>>>>> Min-Se: 120
>>>>> Content-Length: 0
>>>>> 
>>>>> BTW... already asked @oversip guys... But wanted to try this in here just in case! =)
>>>>> 
>>>>> Thanks.
>>>>> 
>>>>> -- 
>>>>> 
>>>>> 
>>>>> 
>>>>> V?ctor E. Medina M.
>>>>> Platform Architect / Chief Infrastructure
>>>>> +58424 291 4561
>>>>> BB #79A8AFA2
>>>>> @VMCibersys
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>> 
>>>> ClueCon 2015 Call for Speakers | Register TODAY! | 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
>>> 
>>> 
>>> 
>>> -- 
>>> 
>>> 
>>> 
>>> V?ctor E. Medina M.
>>> Platform Architect / Chief Infrastructure
>>> +58424 291 4561
>>> BB #79A8AFA2
>>> @VMCibersys
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> 
> 
> 
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 20 04:14:27 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 May 2015 20:14:27 -0400
Subject: [Freeswitch-users] JsSIP -> OverSIP -> Freeswitch
In-Reply-To: <D2308758-768B-413F-A88B-616D87A51941@freeswitch.org>
References: <CACXsapPPc2jti7K0H0C4nm6Sw1bMTzAt70SoqNish=uuyBk=qw@mail.gmail.com>
	<CAEJMVkDBv2XmAHhySS_faVTrDo6K2dgy3+OEL7k=ER33dT7kGg@mail.gmail.com>
	<CACXsapPGxPT3mGvhTMeqk_QTuPXkGj7ibDJ91G8c9rGpxop7WA@mail.gmail.com>
	<8A1140F3-33E1-4F91-A639-A6BCED23FC98@jerris.com>
	<CACXsapOeMWeauuJW=1hr7FA2Aa9eKx+5yp_70-8MDB8A3eFDJw@mail.gmail.com>
	<D2308758-768B-413F-A88B-616D87A51941@freeswitch.org>
Message-ID: <0BC899C0-6842-424B-B83E-D0DCFF652CC3@jerris.com>

from the default configs:

<!--<param name="minimum-session-expires" value="120"/>-->

it looks like that got uncommented.  That being said, 90 is an insanely low session expires value, recommended value for that is 30 minutes (1800)

https://tools.ietf.org/html/rfc4028 <https://tools.ietf.org/html/rfc4028>

Mike


> On May 19, 2015, at 7:41 PM, Ken Rice <krice at freeswitch.org> wrote:
> 
> Your session time is too short make it longer like 300 secs
> 
> Sent from my iPhone
> 
> On May 19, 2015, at 6:34 PM, Victor Medina <victor.medina at cibersys.com <mailto:victor.medina at cibersys.com>> wrote:
> 
>> Hi Michael!
>> 
>> This is the the sip debug on the freeswitch directly jssip -> freeswitch -> 7443
>> 
>> freeswitch at internal> sofia profile internal siptrace on
>> Enabled sip debugging on internal
>> recv 5122 bytes from wss/[208.84.81.64]:57975 at 19:31:56.205487:
>>    ------------------------------------------------------------------------
>>    INVITE sip:9196 at conference.cibersys.com <mailto:sip%3A9196 at conference.cibersys.com> SIP/2.0
>>    Via: SIP/2.0/WSS 7rrpouob7plu.invalid;branch=z9hG4bK1885537
>>    Max-Forwards: 69
>>    To: <sip:9196 at conference.cibersys.com <mailto:sip%3A9196 at conference.cibersys.com>>
>>    From: <sip:1000 at conference.cibersys.com <mailto:sip%3A1000 at conference.cibersys.com>>;tag=i26sin3126
>>    Call-ID: di8dteiqmu6ofrf75pbd
>>    CSeq: 6473 INVITE
>>    X-Can-Renegotiate: true
>>    Contact: <sip:e3e4td20 at 7rrpouob7plu.invalid;transport=ws;ob>
>>    Content-Type: application/sdp
>>    Session-Expires: 90
>>    Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
>>    Supported: timer,ice,outbound
>>    User-Agent: JsSIP 0.6.26
>>    Content-Length: 4574
>> 
>>    v=0
>>    o=- 3581044066039556708 2 IN IP4 127.0.0.1
>>    s=-
>>    t=0 0
>>    a=group:BUNDLE audio video
>>    a=msid-semantic: WMS iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD
>>    m=audio 55125 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>    c=IN IP4 208.84.81.64
>>    a=rtcp:55125 IN IP4 208.84.81.64
>>    a=candidate:1505006532 1 udp 2122255103 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>>    a=candidate:1505006532 2 udp 2122255103 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>>    a=candidate:2999745851 1 udp 2122194687 192.168.56.1 55124 typ host generation 0
>>    a=candidate:2999745851 2 udp 2122194687 192.168.56.1 55124 typ host generation 0
>>    a=candidate:536231733 1 udp 2122129151 10.0.80.221 55125 typ host generation 0
>>    a=candidate:536231733 2 udp 2122129151 10.0.80.221 55125 typ host generation 0
>>    a=candidate:389508916 1 tcp 1518275327 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>>    a=candidate:389508916 2 tcp 1518275327 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>>    a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0
>>    a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0
>>    a=candidate:1366672325 1 tcp 1518149375 10.0.80.221 0 typ host tcptype active generation 0
>>    a=candidate:1366672325 2 tcp 1518149375 10.0.80.221 0 typ host tcptype active generation 0
>>    a=candidate:3064574176 1 udp 1685921535 208.84.81.64 55125 typ srflx raddr 10.0.80.221 rport 55125 generation 0
>>    a=candidate:3064574176 2 udp 1685921535 208.84.81.64 55125 typ srflx raddr 10.0.80.221 rport 55125 generation 0
>>    a=ice-ufrag:1UvloQ6OnbX0r5t4
>>    a=ice-pwd:mE9PaCEu/IMEtzv9L8YgDRCh
>>    a=fingerprint:sha-256 61:AC:51:08:4B:E0:AB:B0:89:7B:18:05:CA:49:B3:71:9B:5A:77:99:07:58:67:A9:3E:0E:72:92:13:F5:1E:F7
>>    a=setup:actpass
>>    a=mid:audio
>>    a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>    a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
>>    a=sendrecv
>>    a=rtcp-mux
>>    a=rtpmap:111 opus/48000/2
>>    a=fmtp:111 minptime=10; useinbandfec=1
>>    a=rtpmap:103 ISAC/16000
>>    a=rtpmap:104 ISAC/32000
>>    a=rtpmap:9 G722/8000
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:8 PCMA/8000
>>    a=rtpmap:106 CN/32000
>>    a=rtpmap:105 CN/16000
>>    a=rtpmap:13 CN/8000
>>    a=rtpmap:126 telephone-event/8000
>>    a=maxptime:60
>>    a=ssrc:691038544 cname:WW2sd3POh47MtjWG
>>    a=ssrc:691038544 msid:iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD 4cca9568-6c68-48b7-bac4-ed658260a073
>>    a=ssrc:691038544 mslabel:iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD
>>    a=ssrc:691038544 label:4cca9568-6c68-48b7-bac4-ed658260a073
>>    m=video 55125 RTP/SAVPF 100 116 117 96
>>    c=IN IP4 208.84.81.64
>>    a=rtcp:55125 IN IP4 208.84.81.64
>>    a=candidate:1505006532 1 udp 2122255103 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>>    a=candidate:1505006532 2 udp 2122255103 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>>    a=candidate:2999745851 1 udp 2122194687 192.168.56.1 55124 typ host generation 0
>>    a=candidate:2999745851 2 udp 2122194687 192.168.56.1 55124 typ host generation 0
>>    a=candidate:536231733 1 udp 2122129151 10.0.80.221 55125 typ host generation 0
>>    a=candidate:536231733 2 udp 2122129151 10.0.80.221 55125 typ host generation 0
>>    a=candidate:389508916 1 tcp 1518275327 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>>    a=candidate:389508916 2 tcp 1518275327 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>>    a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0
>>    a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host tcptype active generation 0
>>    a=candidate:1366672325 1 tcp 1518149375 10.0.80.221 0 typ host tcptype active generation 0
>>    a=candidate:1366672325 2 tcp 1518149375 10.0.80.221 0 typ host tcptype active generation 0
>>    a=candidate:3064574176 1 udp 1685921535 208.84.81.64 55125 typ srflx raddr 10.0.80.221 rport 55125 generation 0
>>    a=candidate:3064574176 2 udp 1685921535 208.84.81.64 55125 typ srflx raddr 10.0.80.221 rport 55125 generation 0
>>    a=ice-ufrag:1UvloQ6OnbX0r5t4
>>    a=ice-pwd:mE9PaCEu/IMEtzv9L8YgDRCh
>>    a=fingerprint:sha-256 61:AC:51:08:4B:E0:AB:B0:89:7B:18:05:CA:49:B3:71:9B:5A:77:99:07:58:67:A9:3E:0E:72:92:13:F5:1E:F7
>>    a=setup:actpass
>>    a=mid:video
>>    a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>>    a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
>>    a=recvonly
>>    a=rtcp-mux
>>    a=rtpmap:100 VP8/90000
>>    a=rtcp-fb:100 ccm fir
>>    a=rtcp-fb:100 nack
>>    a=rtcp-fb:100 nack pli
>>    a=rtcp-fb:100 goog-remb
>>    a=rtpmap:116 red/90000
>>    a=rtpmap:117 ulpfec/90000
>>    a=rtpmap:96 rtx/90000
>>    a=fmtp:96 apt=100
>>    ------------------------------------------------------------------------
>> send 754 bytes to wss/[208.84.81.64]:57975 at 19:31:56.205982:
>>    ------------------------------------------------------------------------
>>    SIP/2.0 422 Session Interval Too Small
>>    Via: SIP/2.0/WSS 7rrpouob7plu.invalid;branch=z9hG4bK1885537;received=208.84.81.64;rport=57975
>>    From: <sip:1000 at conference.cibersys.com <mailto:sip%3A1000 at conference.cibersys.com>>;tag=i26sin3126
>>    To: <sip:9196 at conference.cibersys.com <mailto:sip%3A9196 at conference.cibersys.com>>;tag=ma466p1ScScUa
>>    Call-ID: di8dteiqmu6ofrf75pbd
>>    CSeq: 6473 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, path, replaces
>>    Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
>>    Min-SE: 120
>>    Content-Length: 0
>> 
>>    ------------------------------------------------------------------------
>> recv 290 bytes from wss/[208.84.81.64]:57975 at 19:31:56.483525:
>>    ------------------------------------------------------------------------
>>    ACK sip:9196 at conference.cibersys.com <mailto:sip%3A9196 at conference.cibersys.com> SIP/2.0
>>    Via: SIP/2.0/WSS 7rrpouob7plu.invalid;branch=z9hG4bK1885537
>>    To: <sip:9196 at conference.cibersys.com <mailto:sip%3A9196 at conference.cibersys.com>>;tag=ma466p1ScScUa
>>    From: <sip:1000 at conference.cibersys.com <mailto:sip%3A1000 at conference.cibersys.com>>;tag=i26sin3126
>>    Call-ID: di8dteiqmu6ofrf75pbd
>>    CSeq: 6473 ACK
>>    Content-Length: 0
>> 
>>    ------------------------------------------------------------------------
>> freeswitch at internal>
>> 
>> 
>> 2015-05-19 18:50 GMT-04:30 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
>> what is oversip?  you shouldn't need any layers between FS and jssip.
>> 
>>> On May 19, 2015, at 6:58 PM, Victor Medina <victor.medina at cibersys.com <mailto:victor.medina at cibersys.com>> wrote:
>>> 
>>> SURE!
>>> 
>>> This is the Sofia Internal side...
>>> 
>>> http://pastebin.com/xwBjzMCX <http://pastebin.com/xwBjzMCX>
>>> 
>>> This is Javascript console... (JsSIP -> OverSIP)
>>> 
>>> http://pastebin.com/XmXRt0nV <http://pastebin.com/XmXRt0nV>
>>> 
>>> Thanks.
>>> 
>>> 2015-05-19 18:15 GMT-04:30 Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>>:
>>> The whole sip exchange would need to be seen for me to tell, it could be a bug, it could be a config issue... not exactly sure yet.  Can you paste bin it?
>>> 
>>> Thanks,
>>> 
>>> On Tue, May 19, 2015 at 5:42 PM, Victor Medina <victor.medina at cibersys.com <mailto:victor.medina at cibersys.com>> wrote:
>>> Hi guys!
>>> 
>>> Im having some problems while trying to connect to a Freswitch PBX using OverSip and JsSIP. 
>>> 
>>> Using Jssip i can recive calls but when calling from the OverSIP/JsSIp I always get a 422 error. 
>>> 
>>> Softphone -> WebRTC ext calls OK
>>> WebRTC -> Softphone fails
>>> WebRTC -> ECHO TEST call on freeswitch also fails
>>> 
>>> Can somebody help find out what this could be?
>>> 
>>> SIP/2.0 422 Session Interval Too Small
>>> Via: SIP/2.0/WSS vv3f75qcos24.invalid;branch=z9hG4bK1673051
>>> From: "test" <sip:1002 at conference.cibersys.com <mailto:sip%3A1002 at conference.cibersys.com>>;tag=5or80k3oc9
>>> To: <sip:9196 at conference.cibersys.com <mailto:sip%3A9196 at conference.cibersys.com>>;tag=Q3Xyt4X2vSX6Q
>>> Call-ID: 4norlmjdkqok6smt0vd4
>>> CSeq: 9416 INVITE
>>> User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
>>> Accept: application/sdp
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> Supported: timer, path, replaces
>>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
>>> Min-Se: 120
>>> Content-Length: 0
>>> 
>>> BTW... already asked @oversip guys... But wanted to try this in here just in case! =)
>>> 
>>> Thanks.
>>> 
>>> -- 
>>> 
>>> 
>>> 
>>> V?ctor E. Medina M.
>>> Platform Architect / Chief Infrastructure
>>> +58424 291 4561 <>
>>> BB #79A8AFA2
>>> @VMCibersys
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>> http://www.cluecon.com <http://www.cluecon.com/>
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> 
>>> 
>>> 
>>> -- 
>>> Brian West
>>> brian at freeswitch.org <mailto:brian at freeswitch.org>
>>> 
>>> Twitter: @FreeSWITCH , @briankwest
>>> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
>>> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
>>> ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>> T:+19184209001 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378)
>>> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com <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/>
>>> 
>>> 
>>> 
>>> -- 
>>> 
>>> 
>>> 
>>> V?ctor E. Medina M.
>>> Platform Architect / Chief Infrastructure
>>> +58424 291 4561 <>
>>> BB #79A8AFA2
>>> @VMCibersys
>>> 
>>> _________________________________________________________________________
>>> 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/>
>> 
>> 
>> 
>> -- 
>> 
>> 
>> 
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561 <>
>> BB #79A8AFA2
>> @VMCibersys
>> 
>> _________________________________________________________________________
>> 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 brian at freeswitch.org  Wed May 20 05:20:16 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 19 May 2015 20:20:16 -0500
Subject: [Freeswitch-users] newbie: originate dial problem
In-Reply-To: <CAPwL2MAWHSSbBn5zJ2BDnnFPtQ66OhOCto=boLbX_yjNV2v8iw@mail.gmail.com>
References: <CAPwL2MC--nb2s6WjL-udZZHoCAMMPBhQv8ELpS6ucRpADmNLJg@mail.gmail.com>
	<CAEJMVkCWxk=d7roEvZ8mroQ+rjy+yeujOEcOJ3RoLCXZP6grqQ@mail.gmail.com>
	<CAPwL2MDMJNQJuqYbijckyLDrqsfjBU0C_p5d1nf2ZnKFod5Eow@mail.gmail.com>
	<CAEJMVkDMyvdU0Hoa9-_+cLadw1fHH6h0Mw+cP1P86xe1e8NYNA@mail.gmail.com>
	<CAPwL2MAWHSSbBn5zJ2BDnnFPtQ66OhOCto=boLbX_yjNV2v8iw@mail.gmail.com>
Message-ID: <CAEJMVkD_q=cWe6NYGVy==prk4rz6UvKzuW11t666tFYTMfUZJQ@mail.gmail.com>

Close, you forgot the entire command, the 'reg' at the end was required, I
wanna see how registrations show up.

On Tue, May 19, 2015 at 6:11 PM, Masakazu Nakano <emplant2000 at gmail.com>
wrote:

> Hi Brian
>
> thank you for my reply.
>
> sorry too long result
>
> freeswitch at internal> sofia status profile internal
>
> =================================================================================================
> Name                    internal
> Domain Name             N/A
> Auto-NAT                false
> DBName                  sofia_reg_internal
> Pres Hosts              27.112.104.17,27.112.104.17,fusionpbx.b3am.com
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_from
> RTP-IP                  fusionpbx.b3am.com
> SIP-IP                  fusionpbx.b3am.com
> URL                     sip:mod_sofia at fusionpbx.b3am.com:5090
> BIND-URL                sip:mod_sofia at fusionpbx.b3am.com:5090
> ;transport=udp,tcp
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS IN               OPUS,speex at 16000h@20i,speex at 8000h@20i,G7221 at 32000h
> ,G7221 at 16000h,G722,PCMU,PCMA,GSM
> CODECS OUT              OPUS,speex at 16000h@20i,speex at 8000h@20i,G7221 at 32000h
> ,G7221 at 16000h,G722,PCMU,PCMA,GSM
> TEL-EVENT               101
> DTMF-MODE               rfc2833
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                true
> PROXY-MEDIA             false
> ZRTP-PASSTHRU           true
> AGGRESSIVENAT           false
> CALLS-IN                0
> FAILED-CALLS-IN         0
> CALLS-OUT               0
> FAILED-CALLS-OUT        0
> REGISTRATIONS           0
>
> -- and
>
> freeswitch at internal> sofia status profile external
>
> =================================================================================================
> Name                    external
> Domain Name             N/A
> Auto-NAT                false
> DBName                  share_presence
> Pres Hosts              27.112.104.17
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_to
> RTP-IP                  27.112.104.17
> SIP-IP                  27.112.104.17
> URL                     sip:mod_sofia at 27.112.104.17:5060
> BIND-URL                sip:mod_sofia at 27.112.104.17:5060;transport=udp,tcp
> WS-BIND-URL             sip:mod_sofia at 27.112.104.17:5066;transport=ws
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS IN               OPUS,speex at 16000h@20i,speex at 8000h@20i,G7221 at 32000h
> ,G7221 at 16000h,G722,PCMU,PCMA,GSM
> CODECS OUT              speex at 16000h@20i,PCMU,PCMA,GSM
> TEL-EVENT               101
> DTMF-MODE               info
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                true
> PROXY-MEDIA             false
> ZRTP-PASSTHRU           true
> AGGRESSIVENAT           false
> CALLS-IN                72
> FAILED-CALLS-IN         72
> CALLS-OUT               0
> FAILED-CALLS-OUT        0
> REGISTRATIONS           3
>
> # oops, I,I,I'm sorry to...I forget...what I can say...
> # I touched asterisk so many time but 15 years spend...
> # pls,pls auhhggg....
>
> BR
>
> mack
>
>
>
>
>
>
> 2015-05-20 7:53 GMT+09:00 Brian West <brian at freeswitch.org>:
>
>> sofia status profile internal reg
>> sofia status profile external reg
>>
>> What do those two commands say?
>>
>> On Tue, May 19, 2015 at 9:32 AM, Masakazu Nakano <emplant2000 at gmail.com>
>> wrote:
>>
>>> HI Brian
>>>
>>> thank you for your reply.
>>>
>>> 'sofia status' says following like  that.
>>>
>>> freeswitch at internal> sofia status
>>>                      Name          Type
>>>       Data      State
>>>
>>> =================================================================================================
>>>             external-ipv6       profile                   sip:mod_sofia@[::2]:5060
>>>      RUNNING (0)
>>>             27.112.104.17         alias
>>>   internal      ALIASED
>>>                  external       profile
>>> sip:mod_sofia at 27.112.104.17:5060      RUNNING (0)
>>>           external::iptel       gateway
>>> sip:emplant2000 at sip.iptel.org      FAIL_WAIT
>>>             internal-ipv6       profile                   sip:mod_sofia@[::2]:5090
>>>      RUNNING (0)
>>>                  internal       profile
>>> sip:mod_sofia at fusionpbx.b3am.com:5090      RUNNING (0)
>>>
>>> =================================================================================================
>>> 4 profiles 1 alias
>>>
>>> BR
>>>
>>> mack
>>>
>>>
>>>
>>> 2015-05-19 22:49 GMT+09:00 Brian West <brian at freeswitch.org>:
>>>
>>>> What rev of FreeSWITCH and what is the output of 'sofia status',   Then
>>>> you can try sofia_contact */5630 and see what it returns, and pastbin all
>>>> of that info for us to see.
>>>>
>>>>
>>>>
>>>> On Tue, May 19, 2015 at 8:11 AM, Masakazu Nakano <emplant2000 at gmail.com
>>>> > wrote:
>>>>
>>>>> Hi there.
>>>>>
>>>>> a softphone is registed called 5630 in public.
>>>>>
>>>>> show registered says following under like that.
>>>>>
>>>>> 5630,27.112.104.17,siGhOxWCoIW0n9JjzwxSTg..,sofia/external/sip:5630 at 49.97.4.168:61682
>>>>> ;transport=UDP;rinstance=b1c4cf82f7d6e637,1432040746,49.97.4.168,61682,udp,
>>>>> b3am.com,
>>>>>
>>>>> and
>>>>>
>>>>> freeswitch at internal > originate user/5630
>>>>>
>>>>> is successfull.
>>>>>
>>>>> but by dialplan/public/5630.xml
>>>>>    <extension name="5630">
>>>>>      <condition field="destination_number" expression="5630">
>>>>>        <action application="bridge" data="user/5630"/>
>>>>>      </condition>
>>>>>    </extension>
>>>>>
>>>>> and make a call with another phone to 5630.to be "Cause:
>>>>> INVALID_PROFILE"
>>>>>
>>>>> 2015-05-19 22:08:53.724519 [NOTICE] switch_ivr_originate.c:2732 Cannot
>>>>> create outgoing channel of type [sofia] cause: [INVALID_PROFILE]
>>>>> 2015-05-19 22:08:53.724519 [DEBUG] switch_ivr_originate.c:3720
>>>>> Originate Resulted in Error Cause: 611 [INVALID_PROFILE]
>>>>> 2015-05-19 22:08:53.724519 [INFO] mod_dptools.c:3246 Originate
>>>>> Failed.  Cause: INVALID_PROFILE
>>>>>
>>>>> why?
>>>>>
>>>>> BR
>>>>>
>>>>> mack
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>> ClueCon 2015 Call for Speakers
>>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>>>> <https://www.reddit.com/r/freeswitch>
>>>>
>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> ClueCon 2015 Call for Speakers
>> <https://www.cluecon.com/call-for-speakers/> | Register
>> <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch
>> <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From olegstolyar at gmail.com  Wed May 20 06:01:10 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 19 May 2015 19:01:10 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
Message-ID: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>

Hi guys,

Several weeks ago I started getting an occasional problem where FS is
sending an INVITE to the other side in the middle of a call, the other side
does not respond and FS hangs up the leg.  Below is the relevant log.  The
user experience is that they keep talking and hearing each other up to the
very end.  I have a recording of that call, so can confirm.

The call uses WebRTC and is originated by JsSip from Chrome.  Then the user
is put into a conference but I doubt it's relevant in this case since the
INVITE and disconnect are not happening from mod_conference

I suspect it's a re-INVITE but what triggers FS to send it?  I couldn't
find anything in the logs that could shed light.

send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
   ------------------------------------------------------------------------
   INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
   Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
   Max-Forwards: 70
   From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>;tag=KQecUSr12rSQp
   To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
   Call-ID: g8980rbrbk2t45oj5mru
   CSeq: 75703945 INVITE
   Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@
##.##.###.###:5080;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Session-Expires: 120;refresher=uac
   Min-SE: 120
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 825

   v=0
   o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
   s=FreeSWITCH
   c=IN IP4 ##.##.##.##
   t=0 0
   a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
   m=audio 22670 RTP/SAVPF 9 126 106
   a=rtpmap:9 G722/8000
   a=rtpmap:126 telephone-event/8000
   a=rtpmap:106 CN/8000
   a=ptime:20
   a=fingerprint:sha-256
E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
   a=rtcp-mux
   a=rtcp:22670 IN IP4 ##.##.##.##
   a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
   a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
   a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
   a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
   a=ice-ufrag:5dS3Fzx1Thrmdy9Z
   a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
   a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host
generation 0
   ------------------------------------------------------------------------
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
switch_core_session.c:1061 Send signal
sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid entering
state [calling][0]
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
switch_core_session.c:1061 Send signal
sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
switch_core_session.c:1061 Send signal
sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid entering
state [terminating][503]
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE]
sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
[CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
switch_channel.c:3242 Send signal
sofia/leia_agent/anonymous at anonymous.invalid [KILL]
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
switch_core_session.c:1396 Send signal
sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
mod_conference.c:5057 Channel leaving conference, cause:
NORMAL_TEMPORARY_FAILURE
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid skip
receive message [UNBRIDGE] (channel is hungup already)
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
switch_core_media.c:7772 sofia/leia_agent/anonymous at anonymous.invalid skip
receive message [REFER_EVENT] (channel is hungup already)
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
Restore previous codec G722:9.
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
switch_core_session.c:2901 sofia/leia_agent/anonymous at anonymous.invalid
skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
switch_core_state_machine.c:535
(sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to sleep
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
switch_core_state_machine.c:472
(sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
CS_HANGUP
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
switch_core_state_machine.c:735
(sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE ->
HANGUP
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
switch_core_state_machine.c:737
(sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
hanging up, cause: NORMAL_TEMPORARY_FAILURE
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
switch_core_state_machine.c:60 sofia/leia_agent/anonymous at anonymous.invalid
Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
switch_core_state_machine.c:737
(sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to sleep
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From ssinyagin at gmail.com  Wed May 20 06:20:59 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 20 May 2015 04:20:59 +0200
Subject: [Freeswitch-users] Should I switch to Debian?
In-Reply-To: <CA+3gBcmxYstnLNfMi6+V0TtAe2k-PLkzJw6GbedTi9x88jHzaQ@mail.gmail.com>
References: <CAOkuKdf54aC+CAfxH9day2BxzCK9rNVXDqkBu26Atqc4Xfokhg@mail.gmail.com>
	<20150519215451.5419088.99834.62272@evaristesys.com>
	<CA+3gBcmxYstnLNfMi6+V0TtAe2k-PLkzJw6GbedTi9x88jHzaQ@mail.gmail.com>
Message-ID: <CACOYK=j971EJJMb0T4LgwaU_VDH78y6V3PvFxhm=NvhDum3d2w@mail.gmail.com>

you don't need to wait for Debian packages, and you can build them
yourself, for any revision that is required. Then the whole
installation and upgrade process is well-managed, with proper startup
scripts and permissions. I guess the RPM's are easy to build as well,
I just never needed to do that.

The package builder is failing under Ubuntu, but it's not a server
platform anyway :-))
https://freeswitch.org/jira/browse/FS-7534




On Wed, May 20, 2015 at 12:02 AM, Carlos Ruiz D?az
<carlos.ruizdiaz at gmail.com> wrote:
> +1 on Alex's approach.
>
> I used to use RPMs, but constantly waiting for them to be updated turned
> into a PITA, so I switched to source code as well.
>
> Regards,
> Carlos
>
> On May 19, 2015 16:56, "Alex Balashov" <abalashov at evaristesys.com> wrote:
>>
>> My personal view is that distro packaging for Freeswitch, Asterisk,
>> Kamailio, and most OSS VoIP elements is unreliable, inconsistent and
>> lopsided, so the only viable approach is to build these things from source,
>> always. This allows one to maintain granular version controland to apply
>> ?patches as needed.
>>
>> If you just forget packaging and learn to love the source, you'll obviate
>> a vast category of problems and not waste time on fighting with packages. It
>> also has the positive side effect that you can support pretty much any
>> server distribution. Debian, CentOS, RHEL, Ubuntu server, who cares?
>>
>> --
>> Alex Balashov | Principal | Evariste Systems LLC
>> 303 Perimeter Center North, Suite 300
>> Atlanta, GA 30346
>> United States
>>
>> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>
>> Sent from my BlackBerry.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 May 20 07:05:11 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 May 2015 23:05:11 -0400
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
Message-ID: <08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>

I think the sip.js guys fixed this issue when they forked jssip.  I'd suggest using that.

> On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> Hi guys,
> 
> Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg.  Below is the relevant log.  The user experience is that they keep talking and hearing each other up to the very end.  I have a recording of that call, so can confirm.
> 
> The call uses WebRTC and is originated by JsSip from Chrome.  Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
> 
> I suspect it's a re-INVITE but what triggers FS to send it?  I couldn't find anything in the logs that could shed light.
> 
> send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>    ------------------------------------------------------------------------
>    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>    Max-Forwards: 70
>    From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid>;tag=KQecUSr12rSQp
>    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>    Call-ID: g8980rbrbk2t45oj5mru
>    CSeq: 75703945 INVITE
>    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Session-Expires: 120;refresher=uac
>    Min-SE: 120
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 825
> 
>    v=0
>    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>    s=FreeSWITCH
>    c=IN IP4 ##.##.##.##
>    t=0 0
>    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>    m=audio 22670 RTP/SAVPF 9 126 106
>    a=rtpmap:9 G722/8000
>    a=rtpmap:126 telephone-event/8000
>    a=rtpmap:106 CN/8000
>    a=ptime:20
>    a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>    a=rtcp-mux
>    a=rtcp:22670 IN IP4 ##.##.##.##
>    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
>    ------------------------------------------------------------------------
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid entering state [calling][0]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid entering state [terminating][503]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous at anonymous.invalid [KILL]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already)
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous at anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already)
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid Restore previous codec G722:9.
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous at anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to sleep
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change CS_HANGUP
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE -> HANGUP
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to sleep
> 



From mike at jerris.com  Wed May 20 07:08:30 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 May 2015 23:08:30 -0400
Subject: [Freeswitch-users] Should I switch to Debian?
In-Reply-To: <CACOYK=j971EJJMb0T4LgwaU_VDH78y6V3PvFxhm=NvhDum3d2w@mail.gmail.com>
References: <CAOkuKdf54aC+CAfxH9day2BxzCK9rNVXDqkBu26Atqc4Xfokhg@mail.gmail.com>
	<20150519215451.5419088.99834.62272@evaristesys.com>
	<CA+3gBcmxYstnLNfMi6+V0TtAe2k-PLkzJw6GbedTi9x88jHzaQ@mail.gmail.com>
	<CACOYK=j971EJJMb0T4LgwaU_VDH78y6V3PvFxhm=NvhDum3d2w@mail.gmail.com>
Message-ID: <D1658B35-DCEA-461F-99CC-576013705D40@jerris.com>

You don't need to wait for Debian packages, they are already in our public repo.  These packages will always be up to date with our latest public releases.


> On May 19, 2015, at 10:20 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> 
> you don't need to wait for Debian packages, and you can build them
> yourself, for any revision that is required. Then the whole
> installation and upgrade process is well-managed, with proper startup
> scripts and permissions. I guess the RPM's are easy to build as well,
> I just never needed to do that.
> 
> The package builder is failing under Ubuntu, but it's not a server
> platform anyway :-))
> https://freeswitch.org/jira/browse/FS-7534
> 
> 
> 
> 
> On Wed, May 20, 2015 at 12:02 AM, Carlos Ruiz D?az
> <carlos.ruizdiaz at gmail.com> wrote:
>> +1 on Alex's approach.
>> 
>> I used to use RPMs, but constantly waiting for them to be updated turned
>> into a PITA, so I switched to source code as well.
>> 
>> Regards,
>> Carlos
>> 
>> On May 19, 2015 16:56, "Alex Balashov" <abalashov at evaristesys.com> wrote:
>>> 
>>> My personal view is that distro packaging for Freeswitch, Asterisk,
>>> Kamailio, and most OSS VoIP elements is unreliable, inconsistent and
>>> lopsided, so the only viable approach is to build these things from source,
>>> always. This allows one to maintain granular version controland to apply
>>> ?patches as needed.
>>> 
>>> If you just forget packaging and learn to love the source, you'll obviate
>>> a vast category of problems and not waste time on fighting with packages. It
>>> also has the positive side effect that you can support pretty much any
>>> server distribution. Debian, CentOS, RHEL, Ubuntu server, who cares?
>>> 
>>> --
>>> Alex Balashov | Principal | Evariste Systems LLC
>>> 303 Perimeter Center North, Suite 300
>>> Atlanta, GA 30346
>>> United States
>>> 
>>> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
>>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>> 
>>> Sent from my BlackBerry.
>>> 



From olegstolyar at gmail.com  Wed May 20 07:41:28 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 19 May 2015 20:41:28 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
Message-ID: <CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>

Thanks Michael, I'll see if we can do that!

So, is the re-INVITE legit and the problem is that JsSip does not respond
to it?  Still, I am curious what is triggering the re-INVITE.

On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com> wrote:

> I think the sip.js guys fixed this issue when they forked jssip.  I'd
> suggest using that.
>
> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
> >
> > Hi guys,
> >
> > Several weeks ago I started getting an occasional problem where FS is
> sending an INVITE to the other side in the middle of a call, the other side
> does not respond and FS hangs up the leg.  Below is the relevant log.  The
> user experience is that they keep talking and hearing each other up to the
> very end.  I have a recording of that call, so can confirm.
> >
> > The call uses WebRTC and is originated by JsSip from Chrome.  Then the
> user is put into a conference but I doubt it's relevant in this case since
> the INVITE and disconnect are not happening from mod_conference
> >
> > I suspect it's a re-INVITE but what triggers FS to send it?  I couldn't
> find anything in the logs that could shed light.
> >
> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
> >
> ------------------------------------------------------------------------
> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
> >    Max-Forwards: 70
> >    From:
> <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
> >;tag=KQecUSr12rSQp
> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
> >    Call-ID: g8980rbrbk2t45oj5mru
> >    CSeq: 75703945 INVITE
> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@
> ##.##.###.###:5080;transport=udp>
> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >    Supported: timer, path, replaces
> >    Session-Expires: 120;refresher=uac
> >    Min-SE: 120
> >    Content-Type: application/sdp
> >    Content-Disposition: session
> >    Content-Length: 825
> >
> >    v=0
> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
> >    s=FreeSWITCH
> >    c=IN IP4 ##.##.##.##
> >    t=0 0
> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
> >    m=audio 22670 RTP/SAVPF 9 126 106
> >    a=rtpmap:9 G722/8000
> >    a=rtpmap:126 telephone-event/8000
> >    a=rtpmap:106 CN/8000
> >    a=ptime:20
> >    a=fingerprint:sha-256
> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
> >    a=rtcp-mux
> >    a=rtcp:22670 IN IP4 ##.##.##.##
> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host
> generation 0
> >
> ------------------------------------------------------------------------
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
> switch_core_session.c:1061 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
> sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
> entering state [calling][0]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_core_session.c:1061 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_core_session.c:1061 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
> entering state [terminating][503]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE]
> sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_channel.c:3242 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_core_session.c:1396 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> mod_conference.c:5057 Channel leaving conference, cause:
> NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid skip
> receive message [UNBRIDGE] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_media.c:7772 sofia/leia_agent/anonymous at anonymous.invalid
> skip receive message [REFER_EVENT] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
> Restore previous codec G722:9.
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_session.c:2901 sofia/leia_agent/anonymous at anonymous.invalid
> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:535
> (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to
> sleep
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:472
> (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
> CS_HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:735
> (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE ->
> HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:737
> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
> hanging up, cause: NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:60 sofia/leia_agent/anonymous at anonymous.invalid
> Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:737
> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to sleep
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jungleboogie0 at gmail.com  Wed May 20 07:57:07 2015
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Tue, 19 May 2015 20:57:07 -0700
Subject: [Freeswitch-users] Should I switch to Debian?
In-Reply-To: <20150519215451.5419088.99834.62272@evaristesys.com>
References: <CAOkuKdf54aC+CAfxH9day2BxzCK9rNVXDqkBu26Atqc4Xfokhg@mail.gmail.com>
	<20150519215451.5419088.99834.62272@evaristesys.com>
Message-ID: <CAKE2PDvWpKXR4VXdnnQ=9YSzFuAByfXPD4mA2icrWbLxTzoDxw@mail.gmail.com>

On 19 May 2015 at 14:54, Alex Balashov <abalashov at evaristesys.com> wrote:
> My personal view is that distro packaging for Freeswitch, Asterisk,
> Kamailio, and most OSS VoIP elements is unreliable, inconsistent and
> lopsided, so the only viable approach is to build these things from source,
> always. This allows one to maintain granular version controland to apply
> ?patches as needed.


I agree. This also means you can:
a) build on freeBSD!
b) have a staging area where you are following trunk/master so you're
aware of what to expect from new releases and test in advance.
c) better understand the project.



-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From ashwinrkjain at gmail.com  Wed May 20 09:07:27 2015
From: ashwinrkjain at gmail.com (Ashwin Jain)
Date: Wed, 20 May 2015 10:37:27 +0530
Subject: [Freeswitch-users] Detecting external voicemail
Message-ID: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>

Hi all,

We are using freeswitch 1.4.18. Our setup is very basic, we have an
incoming call, which we terminate on mobile phones (via gateways).

Is there anyway to detect if the call is picked up by mobile phone provider
voicemail?

-- 
Thanks and Regards,
Ashwin Jain
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From mitchelle.bit at gmail.com  Wed May 20 09:53:51 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Wed, 20 May 2015 11:23:51 +0530
Subject: [Freeswitch-users] Detecting external voicemail
In-Reply-To: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
References: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
Message-ID: <CANBm6Sxf-O8x=XoR7sCjvpyR2Lj7i8swHoKL-dVW1wNfwnxzHA@mail.gmail.com>

Hi Ankit,

I think the CDR's (Call Detail Records) can give you information about
this, the 'billsec' in XML CDR will show if the call is answered by the
voicemail or not.

Thanks,
Mitchelle

On Wed, May 20, 2015 at 10:37 AM, Ashwin Jain <ashwinrkjain at gmail.com>
wrote:

> Hi all,
>
> We are using freeswitch 1.4.18. Our setup is very basic, we have an
> incoming call, which we terminate on mobile phones (via gateways).
>
> Is there anyway to detect if the call is picked up by mobile phone
> provider voicemail?
>
> --
> Thanks and Regards,
> Ashwin Jain
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From giggsey at gmail.com  Wed May 20 10:15:18 2015
From: giggsey at gmail.com (Joshua Gigg)
Date: Wed, 20 May 2015 07:15:18 +0100
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
Message-ID: <CADcg8iUDn9-3XjerFPGki0h=x+MCwgC5dO=7-txVwVATHOOOgg@mail.gmail.com>

The session expires?
On 20 May 2015 04:44, "Oleg Stolyar" <olegstolyar at gmail.com> wrote:

> Thanks Michael, I'll see if we can do that!
>
> So, is the re-INVITE legit and the problem is that JsSip does not respond
> to it?  Still, I am curious what is triggering the re-INVITE.
>
> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I think the sip.js guys fixed this issue when they forked jssip.  I'd
>> suggest using that.
>>
>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
>> wrote:
>> >
>> > Hi guys,
>> >
>> > Several weeks ago I started getting an occasional problem where FS is
>> sending an INVITE to the other side in the middle of a call, the other side
>> does not respond and FS hangs up the leg.  Below is the relevant log.  The
>> user experience is that they keep talking and hearing each other up to the
>> very end.  I have a recording of that call, so can confirm.
>> >
>> > The call uses WebRTC and is originated by JsSip from Chrome.  Then the
>> user is put into a conference but I doubt it's relevant in this case since
>> the INVITE and disconnect are not happening from mod_conference
>> >
>> > I suspect it's a re-INVITE but what triggers FS to send it?  I couldn't
>> find anything in the logs that could shed light.
>> >
>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>> >
>> ------------------------------------------------------------------------
>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>> >    Max-Forwards: 70
>> >    From:
>> <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>> >;tag=KQecUSr12rSQp
>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>> >    Call-ID: g8980rbrbk2t45oj5mru
>> >    CSeq: 75703945 INVITE
>> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@
>> ##.##.###.###:5080;transport=udp>
>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >    Supported: timer, path, replaces
>> >    Session-Expires: 120;refresher=uac
>> >    Min-SE: 120
>> >    Content-Type: application/sdp
>> >    Content-Disposition: session
>> >    Content-Length: 825
>> >
>> >    v=0
>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>> >    s=FreeSWITCH
>> >    c=IN IP4 ##.##.##.##
>> >    t=0 0
>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>> >    m=audio 22670 RTP/SAVPF 9 126 106
>> >    a=rtpmap:9 G722/8000
>> >    a=rtpmap:126 telephone-event/8000
>> >    a=rtpmap:106 CN/8000
>> >    a=ptime:20
>> >    a=fingerprint:sha-256
>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>> >    a=rtcp-mux
>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host
>> generation 0
>> >
>> ------------------------------------------------------------------------
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
>> switch_core_session.c:1061 Send signal
>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
>> sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>> entering state [calling][0]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
>> switch_core_session.c:1061 Send signal
>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
>> switch_core_session.c:1061 Send signal
>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
>> sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>> entering state [terminating][503]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>> [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
>> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
>> switch_channel.c:3242 Send signal
>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
>> switch_core_session.c:1396 Send signal
>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> mod_conference.c:5057 Channel leaving conference, cause:
>> NORMAL_TEMPORARY_FAILURE
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid skip
>> receive message [UNBRIDGE] (channel is hungup already)
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_media.c:7772 sofia/leia_agent/anonymous at anonymous.invalid
>> skip receive message [REFER_EVENT] (channel is hungup already)
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
>> Restore previous codec G722:9.
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_session.c:2901 sofia/leia_agent/anonymous at anonymous.invalid
>> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:535
>> (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to
>> sleep
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:472
>> (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>> CS_HANGUP
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:735
>> (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE
>> -> HANGUP
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:737
>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
>> hanging up, cause: NORMAL_TEMPORARY_FAILURE
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:60 sofia/leia_agent/anonymous at anonymous.invalid
>> Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:737
>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to
>> sleep
>> >
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Wed May 20 10:26:07 2015
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 20 May 2015 06:26:07 +0000
Subject: [Freeswitch-users] Detecting external voicemail
In-Reply-To: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
References: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
Message-ID: <0000014d70002517-9a948295-a420-439e-82c4-df6e573cdb38-000000@email.amazonses.com>

Maybe some fancy usage of vm detect. When I want to ensure it can't be
picked up by the VM, I use group_confirm that you have to press 1 to take
the call.
-Avi

On Wed, May 20, 2015 at 8:07 AM, Ashwin Jain <ashwinrkjain at gmail.com> wrote:

> Hi all,
>
> We are using freeswitch 1.4.18. Our setup is very basic, we have an
> incoming call, which we terminate on mobile phones (via gateways).
>
> Is there anyway to detect if the call is picked up by mobile phone
> provider voicemail?
>
> --
> Thanks and Regards,
> Ashwin Jain
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed May 20 10:40:56 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 May 2015 02:40:56 -0400
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
Message-ID: <CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>

session timer

On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Thanks Michael, I'll see if we can do that!
>
> So, is the re-INVITE legit and the problem is that JsSip does not respond
> to it?  Still, I am curious what is triggering the re-INVITE.
>
> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>
>> I think the sip.js guys fixed this issue when they forked jssip.  I'd
>> suggest using that.
>>
>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com
>> <javascript:_e(%7B%7D,'cvml','olegstolyar at gmail.com');>> wrote:
>> >
>> > Hi guys,
>> >
>> > Several weeks ago I started getting an occasional problem where FS is
>> sending an INVITE to the other side in the middle of a call, the other side
>> does not respond and FS hangs up the leg.  Below is the relevant log.  The
>> user experience is that they keep talking and hearing each other up to the
>> very end.  I have a recording of that call, so can confirm.
>> >
>> > The call uses WebRTC and is originated by JsSip from Chrome.  Then the
>> user is put into a conference but I doubt it's relevant in this case since
>> the INVITE and disconnect are not happening from mod_conference
>> >
>> > I suspect it's a re-INVITE but what triggers FS to send it?  I couldn't
>> find anything in the logs that could shed light.
>> >
>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>> >
>> ------------------------------------------------------------------------
>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>> >    Max-Forwards: 70
>> >    From:
>> <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>> >;tag=KQecUSr12rSQp
>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>> >    Call-ID: g8980rbrbk2t45oj5mru
>> >    CSeq: 75703945 INVITE
>> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@
>> ##.##.###.###:5080;transport=udp>
>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> >    Supported: timer, path, replaces
>> >    Session-Expires: 120;refresher=uac
>> >    Min-SE: 120
>> >    Content-Type: application/sdp
>> >    Content-Disposition: session
>> >    Content-Length: 825
>> >
>> >    v=0
>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>> >    s=FreeSWITCH
>> >    c=IN IP4 ##.##.##.##
>> >    t=0 0
>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>> >    m=audio 22670 RTP/SAVPF 9 126 106
>> >    a=rtpmap:9 G722/8000
>> >    a=rtpmap:126 telephone-event/8000
>> >    a=rtpmap:106 CN/8000
>> >    a=ptime:20
>> >    a=fingerprint:sha-256
>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>> >    a=rtcp-mux
>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host
>> generation 0
>> >
>> ------------------------------------------------------------------------
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
>> switch_core_session.c:1061 Send signal
>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
>> sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>> entering state [calling][0]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
>> switch_core_session.c:1061 Send signal
>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
>> switch_core_session.c:1061 Send signal
>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
>> sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>> entering state [terminating][503]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>> [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
>> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
>> switch_channel.c:3242 Send signal
>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
>> switch_core_session.c:1396 Send signal
>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> mod_conference.c:5057 Channel leaving conference, cause:
>> NORMAL_TEMPORARY_FAILURE
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid skip
>> receive message [UNBRIDGE] (channel is hungup already)
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_media.c:7772 sofia/leia_agent/anonymous at anonymous.invalid
>> skip receive message [REFER_EVENT] (channel is hungup already)
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
>> Restore previous codec G722:9.
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_session.c:2901 sofia/leia_agent/anonymous at anonymous.invalid
>> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:535
>> (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to
>> sleep
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:472
>> (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>> CS_HANGUP
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:735
>> (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE
>> -> HANGUP
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:737
>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
>> hanging up, cause: NORMAL_TEMPORARY_FAILURE
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:60 sofia/leia_agent/anonymous at anonymous.invalid
>> Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
>> switch_core_state_machine.c:737
>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to
>> sleep
>> >
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From s.safarov at gmail.com  Wed May 20 10:42:05 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 20 May 2015 09:42:05 +0300
Subject: [Freeswitch-users] Detecting external voicemail
In-Reply-To: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
References: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
Message-ID: <CAHtxdDcqzpdw3SrWVpW4RQvw9Ho3jo5=y800WxWJ7j7ZQjNL2A@mail.gmail.com>

Try mod_avmd <https://freeswitch.org/confluence/display/FREESWITCH/mod_avmd>


On Wed, May 20, 2015 at 8:07 AM, Ashwin Jain <ashwinrkjain at gmail.com> wrote:

> Hi all,
>
> We are using freeswitch 1.4.18. Our setup is very basic, we have an
> incoming call, which we terminate on mobile phones (via gateways).
>
> Is there anyway to detect if the call is picked up by mobile phone
> provider voicemail?
>
> --
> Thanks and Regards,
> Ashwin Jain
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed May 20 10:43:29 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 May 2015 02:43:29 -0400
Subject: [Freeswitch-users] Detecting external voicemail
In-Reply-To: <CANBm6Sxf-O8x=XoR7sCjvpyR2Lj7i8swHoKL-dVW1wNfwnxzHA@mail.gmail.com>
References: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
	<CANBm6Sxf-O8x=XoR7sCjvpyR2Lj7i8swHoKL-dVW1wNfwnxzHA@mail.gmail.com>
Message-ID: <CAEwTsAghfg3CbFmrA38c7vywOhREdi_pG-tHuFSiKp3MQPrPsw@mail.gmail.com>

Billsec has nothing at all to do with if its answered by voicemail or a
person.  There is no definitive way to know, but as said elsewhere you can
use group confirm or some sort of answering machine detection to try to
figure it out, but it won't be 100%

On Wednesday, May 20, 2015, Mitchelle Johnson <mitchelle.bit at gmail.com>
wrote:

> Hi Ankit,
>
> I think the CDR's (Call Detail Records) can give you information about
> this, the 'billsec' in XML CDR will show if the call is answered by the
> voicemail or not.
>
> Thanks,
> Mitchelle
>
> On Wed, May 20, 2015 at 10:37 AM, Ashwin Jain <ashwinrkjain at gmail.com
> <javascript:_e(%7B%7D,'cvml','ashwinrkjain at gmail.com');>> wrote:
>
>> Hi all,
>>
>> We are using freeswitch 1.4.18. Our setup is very basic, we have an
>> incoming call, which we terminate on mobile phones (via gateways).
>>
>> Is there anyway to detect if the call is picked up by mobile phone
>> provider voicemail?
>>
>> --
>> Thanks and Regards,
>> Ashwin Jain
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From mitchelle.bit at gmail.com  Wed May 20 10:55:40 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Wed, 20 May 2015 12:25:40 +0530
Subject: [Freeswitch-users] Detecting external voicemail
In-Reply-To: <CAEwTsAghfg3CbFmrA38c7vywOhREdi_pG-tHuFSiKp3MQPrPsw@mail.gmail.com>
References: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
	<CANBm6Sxf-O8x=XoR7sCjvpyR2Lj7i8swHoKL-dVW1wNfwnxzHA@mail.gmail.com>
	<CAEwTsAghfg3CbFmrA38c7vywOhREdi_pG-tHuFSiKp3MQPrPsw@mail.gmail.com>
Message-ID: <CANBm6Sz_ivkWEjCJgHgF88sxi09CUmhC_ijjWJ9u+mXReB_ofg@mail.gmail.com>

Hi Michael,

Ashwin has specifically asked for the information on if the phone call is
answered by the voicemail only. If billsec is not the correct way to figure
out whether the call is answered,(irrespective of whether voicemail or a
person answers the call), then what exactly billsec is?

Please share your useful knowledge.

Thanks,
Mitchelle

On Wed, May 20, 2015 at 12:13 PM, Michael Jerris <mike at jerris.com> wrote:

> Billsec has nothing at all to do with if its answered by voicemail or a
> person.  There is no definitive way to know, but as said elsewhere you can
> use group confirm or some sort of answering machine detection to try to
> figure it out, but it won't be 100%
>
> On Wednesday, May 20, 2015, Mitchelle Johnson <mitchelle.bit at gmail.com>
> wrote:
>
>> Hi Ankit,
>>
>> I think the CDR's (Call Detail Records) can give you information about
>> this, the 'billsec' in XML CDR will show if the call is answered by the
>> voicemail or not.
>>
>> Thanks,
>> Mitchelle
>>
>> On Wed, May 20, 2015 at 10:37 AM, Ashwin Jain <ashwinrkjain at gmail.com>
>> wrote:
>>
>>> Hi all,
>>>
>>> We are using freeswitch 1.4.18. Our setup is very basic, we have an
>>> incoming call, which we terminate on mobile phones (via gateways).
>>>
>>> Is there anyway to detect if the call is picked up by mobile phone
>>> provider voicemail?
>>>
>>> --
>>> Thanks and Regards,
>>> Ashwin Jain
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed May 20 11:00:29 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 20 May 2015 09:00:29 +0200
Subject: [Freeswitch-users] Detecting external voicemail
In-Reply-To: <CANBm6Sz_ivkWEjCJgHgF88sxi09CUmhC_ijjWJ9u+mXReB_ofg@mail.gmail.com>
References: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
	<CANBm6Sxf-O8x=XoR7sCjvpyR2Lj7i8swHoKL-dVW1wNfwnxzHA@mail.gmail.com>
	<CAEwTsAghfg3CbFmrA38c7vywOhREdi_pG-tHuFSiKp3MQPrPsw@mail.gmail.com>
	<CANBm6Sz_ivkWEjCJgHgF88sxi09CUmhC_ijjWJ9u+mXReB_ofg@mail.gmail.com>
Message-ID: <CALXCt0pDrjfnpJNDwSKeHZ-YoyT+-S0L9mB8knZ1YYuTMeSmXw@mail.gmail.com>

Billsec is seconds after call has been answered (by a human or by a
machine).

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 20, 2015 8:56 AM, "Mitchelle Johnson" <mitchelle.bit at gmail.com>
wrote:

> Hi Michael,
>
> Ashwin has specifically asked for the information on if the phone call is
> answered by the voicemail only. If billsec is not the correct way to figure
> out whether the call is answered,(irrespective of whether voicemail or a
> person answers the call), then what exactly billsec is?
>
> Please share your useful knowledge.
>
> Thanks,
> Mitchelle
>
> On Wed, May 20, 2015 at 12:13 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Billsec has nothing at all to do with if its answered by voicemail or a
>> person.  There is no definitive way to know, but as said elsewhere you can
>> use group confirm or some sort of answering machine detection to try to
>> figure it out, but it won't be 100%
>>
>> On Wednesday, May 20, 2015, Mitchelle Johnson <mitchelle.bit at gmail.com>
>> wrote:
>>
>>> Hi Ankit,
>>>
>>> I think the CDR's (Call Detail Records) can give you information about
>>> this, the 'billsec' in XML CDR will show if the call is answered by the
>>> voicemail or not.
>>>
>>> Thanks,
>>> Mitchelle
>>>
>>> On Wed, May 20, 2015 at 10:37 AM, Ashwin Jain <ashwinrkjain at gmail.com>
>>> wrote:
>>>
>>>> Hi all,
>>>>
>>>> We are using freeswitch 1.4.18. Our setup is very basic, we have an
>>>> incoming call, which we terminate on mobile phones (via gateways).
>>>>
>>>> Is there anyway to detect if the call is picked up by mobile phone
>>>> provider voicemail?
>>>>
>>>> --
>>>> Thanks and Regards,
>>>> Ashwin Jain
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed May 20 11:03:05 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 20 May 2015 09:03:05 +0200
Subject: [Freeswitch-users] Detecting external voicemail
In-Reply-To: <CALXCt0pDrjfnpJNDwSKeHZ-YoyT+-S0L9mB8knZ1YYuTMeSmXw@mail.gmail.com>
References: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
	<CANBm6Sxf-O8x=XoR7sCjvpyR2Lj7i8swHoKL-dVW1wNfwnxzHA@mail.gmail.com>
	<CAEwTsAghfg3CbFmrA38c7vywOhREdi_pG-tHuFSiKp3MQPrPsw@mail.gmail.com>
	<CANBm6Sz_ivkWEjCJgHgF88sxi09CUmhC_ijjWJ9u+mXReB_ofg@mail.gmail.com>
	<CALXCt0pDrjfnpJNDwSKeHZ-YoyT+-S0L9mB8knZ1YYuTMeSmXw@mail.gmail.com>
Message-ID: <CALXCt0oSLbaiz09n1enm+SH+bkbyGKX-91ufGaP2yyCS0uquPQ@mail.gmail.com>

Actually, is "billable seconds", the part of the call considered billable.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 20, 2015 9:00 AM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

> Billsec is seconds after call has been answered (by a human or by a
> machine).
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On May 20, 2015 8:56 AM, "Mitchelle Johnson" <mitchelle.bit at gmail.com>
> wrote:
>
>> Hi Michael,
>>
>> Ashwin has specifically asked for the information on if the phone call is
>> answered by the voicemail only. If billsec is not the correct way to figure
>> out whether the call is answered,(irrespective of whether voicemail or a
>> person answers the call), then what exactly billsec is?
>>
>> Please share your useful knowledge.
>>
>> Thanks,
>> Mitchelle
>>
>> On Wed, May 20, 2015 at 12:13 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Billsec has nothing at all to do with if its answered by voicemail or a
>>> person.  There is no definitive way to know, but as said elsewhere you can
>>> use group confirm or some sort of answering machine detection to try to
>>> figure it out, but it won't be 100%
>>>
>>> On Wednesday, May 20, 2015, Mitchelle Johnson <mitchelle.bit at gmail.com>
>>> wrote:
>>>
>>>> Hi Ankit,
>>>>
>>>> I think the CDR's (Call Detail Records) can give you information about
>>>> this, the 'billsec' in XML CDR will show if the call is answered by the
>>>> voicemail or not.
>>>>
>>>> Thanks,
>>>> Mitchelle
>>>>
>>>> On Wed, May 20, 2015 at 10:37 AM, Ashwin Jain <ashwinrkjain at gmail.com>
>>>> wrote:
>>>>
>>>>> Hi all,
>>>>>
>>>>> We are using freeswitch 1.4.18. Our setup is very basic, we have an
>>>>> incoming call, which we terminate on mobile phones (via gateways).
>>>>>
>>>>> Is there anyway to detect if the call is picked up by mobile phone
>>>>> provider voicemail?
>>>>>
>>>>> --
>>>>> Thanks and Regards,
>>>>> Ashwin Jain
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
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From mitchelle.bit at gmail.com  Wed May 20 11:05:51 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Wed, 20 May 2015 12:35:51 +0530
Subject: [Freeswitch-users] Detecting external voicemail
In-Reply-To: <CALXCt0pDrjfnpJNDwSKeHZ-YoyT+-S0L9mB8knZ1YYuTMeSmXw@mail.gmail.com>
References: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
	<CANBm6Sxf-O8x=XoR7sCjvpyR2Lj7i8swHoKL-dVW1wNfwnxzHA@mail.gmail.com>
	<CAEwTsAghfg3CbFmrA38c7vywOhREdi_pG-tHuFSiKp3MQPrPsw@mail.gmail.com>
	<CANBm6Sz_ivkWEjCJgHgF88sxi09CUmhC_ijjWJ9u+mXReB_ofg@mail.gmail.com>
	<CALXCt0pDrjfnpJNDwSKeHZ-YoyT+-S0L9mB8knZ1YYuTMeSmXw@mail.gmail.com>
Message-ID: <CANBm6Swgmg=vvHhpPv0q=t3Yh4fp-Gnxnz2mS_Fp66XaiAzBJw@mail.gmail.com>

Thanks a lot Giovanni,

So if billsec in XML CDR shows a non-zero value its means the call is
answered either by machine or by human?
If yes, can it not be used as a yard stick to determine if the call is
answered?

Thanks again,
Mitchelle

On Wed, May 20, 2015 at 12:30 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Billsec is seconds after call has been answered (by a human or by a
> machine).
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On May 20, 2015 8:56 AM, "Mitchelle Johnson" <mitchelle.bit at gmail.com>
> wrote:
>
>> Hi Michael,
>>
>> Ashwin has specifically asked for the information on if the phone call is
>> answered by the voicemail only. If billsec is not the correct way to figure
>> out whether the call is answered,(irrespective of whether voicemail or a
>> person answers the call), then what exactly billsec is?
>>
>> Please share your useful knowledge.
>>
>> Thanks,
>> Mitchelle
>>
>> On Wed, May 20, 2015 at 12:13 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Billsec has nothing at all to do with if its answered by voicemail or a
>>> person.  There is no definitive way to know, but as said elsewhere you can
>>> use group confirm or some sort of answering machine detection to try to
>>> figure it out, but it won't be 100%
>>>
>>> On Wednesday, May 20, 2015, Mitchelle Johnson <mitchelle.bit at gmail.com>
>>> wrote:
>>>
>>>> Hi Ankit,
>>>>
>>>> I think the CDR's (Call Detail Records) can give you information about
>>>> this, the 'billsec' in XML CDR will show if the call is answered by the
>>>> voicemail or not.
>>>>
>>>> Thanks,
>>>> Mitchelle
>>>>
>>>> On Wed, May 20, 2015 at 10:37 AM, Ashwin Jain <ashwinrkjain at gmail.com>
>>>> wrote:
>>>>
>>>>> Hi all,
>>>>>
>>>>> We are using freeswitch 1.4.18. Our setup is very basic, we have an
>>>>> incoming call, which we terminate on mobile phones (via gateways).
>>>>>
>>>>> Is there anyway to detect if the call is picked up by mobile phone
>>>>> provider voicemail?
>>>>>
>>>>> --
>>>>> Thanks and Regards,
>>>>> Ashwin Jain
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gmaruzz at gmail.com  Wed May 20 11:12:31 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 20 May 2015 09:12:31 +0200
Subject: [Freeswitch-users] Detecting external voicemail
In-Reply-To: <CANBm6Swgmg=vvHhpPv0q=t3Yh4fp-Gnxnz2mS_Fp66XaiAzBJw@mail.gmail.com>
References: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
	<CANBm6Sxf-O8x=XoR7sCjvpyR2Lj7i8swHoKL-dVW1wNfwnxzHA@mail.gmail.com>
	<CAEwTsAghfg3CbFmrA38c7vywOhREdi_pG-tHuFSiKp3MQPrPsw@mail.gmail.com>
	<CANBm6Sz_ivkWEjCJgHgF88sxi09CUmhC_ijjWJ9u+mXReB_ofg@mail.gmail.com>
	<CALXCt0pDrjfnpJNDwSKeHZ-YoyT+-S0L9mB8knZ1YYuTMeSmXw@mail.gmail.com>
	<CANBm6Swgmg=vvHhpPv0q=t3Yh4fp-Gnxnz2mS_Fp66XaiAzBJw@mail.gmail.com>
Message-ID: <CALXCt0qfOPD1OW0zjCoRyr0HXqaxE-aquwAxWUp2Ri=YvNmWyQ@mail.gmail.com>

The question was about understanding if the call was answered by a
voicemail or a human. Billsec has nothing to do with it.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 20, 2015 9:06 AM, "Mitchelle Johnson" <mitchelle.bit at gmail.com>
wrote:

> Thanks a lot Giovanni,
>
> So if billsec in XML CDR shows a non-zero value its means the call is
> answered either by machine or by human?
> If yes, can it not be used as a yard stick to determine if the call is
> answered?
>
> Thanks again,
> Mitchelle
>
> On Wed, May 20, 2015 at 12:30 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Billsec is seconds after call has been answered (by a human or by a
>> machine).
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On May 20, 2015 8:56 AM, "Mitchelle Johnson" <mitchelle.bit at gmail.com>
>> wrote:
>>
>>> Hi Michael,
>>>
>>> Ashwin has specifically asked for the information on if the phone call
>>> is answered by the voicemail only. If billsec is not the correct way to
>>> figure out whether the call is answered,(irrespective of whether voicemail
>>> or a person answers the call), then what exactly billsec is?
>>>
>>> Please share your useful knowledge.
>>>
>>> Thanks,
>>> Mitchelle
>>>
>>> On Wed, May 20, 2015 at 12:13 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>>
>>>> Billsec has nothing at all to do with if its answered by voicemail or a
>>>> person.  There is no definitive way to know, but as said elsewhere you can
>>>> use group confirm or some sort of answering machine detection to try to
>>>> figure it out, but it won't be 100%
>>>>
>>>> On Wednesday, May 20, 2015, Mitchelle Johnson <mitchelle.bit at gmail.com>
>>>> wrote:
>>>>
>>>>> Hi Ankit,
>>>>>
>>>>> I think the CDR's (Call Detail Records) can give you information about
>>>>> this, the 'billsec' in XML CDR will show if the call is answered by the
>>>>> voicemail or not.
>>>>>
>>>>> Thanks,
>>>>> Mitchelle
>>>>>
>>>>> On Wed, May 20, 2015 at 10:37 AM, Ashwin Jain <ashwinrkjain at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi all,
>>>>>>
>>>>>> We are using freeswitch 1.4.18. Our setup is very basic, we have an
>>>>>> incoming call, which we terminate on mobile phones (via gateways).
>>>>>>
>>>>>> Is there anyway to detect if the call is picked up by mobile phone
>>>>>> provider voicemail?
>>>>>>
>>>>>> --
>>>>>> Thanks and Regards,
>>>>>> Ashwin Jain
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mishehu at freeswitch.org  Wed May 20 11:18:01 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Wed, 20 May 2015 02:18:01 -0500
Subject: [Freeswitch-users] Detecting external voicemail
In-Reply-To: <CANBm6Swgmg=vvHhpPv0q=t3Yh4fp-Gnxnz2mS_Fp66XaiAzBJw@mail.gmail.com>
References: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>	<CANBm6Sxf-O8x=XoR7sCjvpyR2Lj7i8swHoKL-dVW1wNfwnxzHA@mail.gmail.com>	<CAEwTsAghfg3CbFmrA38c7vywOhREdi_pG-tHuFSiKp3MQPrPsw@mail.gmail.com>	<CANBm6Sz_ivkWEjCJgHgF88sxi09CUmhC_ijjWJ9u+mXReB_ofg@mail.gmail.com>	<CALXCt0pDrjfnpJNDwSKeHZ-YoyT+-S0L9mB8knZ1YYuTMeSmXw@mail.gmail.com>
	<CANBm6Swgmg=vvHhpPv0q=t3Yh4fp-Gnxnz2mS_Fp66XaiAzBJw@mail.gmail.com>
Message-ID: <555C3529.5070407@freeswitch.org>

That machine can be the current switch or another switch even.  If you 
have an extension where one of the applications executed is "answer", 
then there you have it, you now have billsec that increments until the 
channel is hung up.  I would not advise trying to use billsec as any 
yardstick.

-Yossi

On 05/20/2015 02:05 AM, Mitchelle Johnson wrote:
> Thanks a lot Giovanni,
>
> So if billsec in XML CDR shows a non-zero value its means the call is 
> answered either by machine or by human?
> If yes, can it not be used as a yard stick to determine if the call is 
> answered?
>
> Thanks again,
> Mitchelle
>
> On Wed, May 20, 2015 at 12:30 PM, Giovanni Maruzzelli 
> <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>
>     Billsec is seconds after call has been answered (by a human or by
>     a machine).
>
>     sent from my mobile,
>     Giovanni Maruzzelli
>     cell: +39 347 266 56 18
>
>     On May 20, 2015 8:56 AM, "Mitchelle Johnson"
>     <mitchelle.bit at gmail.com <mailto:mitchelle.bit at gmail.com>> wrote:
>
>         Hi Michael,
>
>         Ashwin has specifically asked for the information on if the
>         phone call is answered by the voicemail only. If billsec is
>         not the correct way to figure out whether the call is
>         answered,(irrespective of whether voicemail or a person
>         answers the call), then what exactly billsec is?
>
>         Please share your useful knowledge.
>
>         Thanks,
>         Mitchelle
>
>         On Wed, May 20, 2015 at 12:13 PM, Michael Jerris
>         <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>
>             Billsec has nothing at all to do with if its answered by
>             voicemail or a person.  There is no definitive way to
>             know, but as said elsewhere you can use group confirm or
>             some sort of answering machine detection to try to figure
>             it out, but it won't be 100%
>
>             On Wednesday, May 20, 2015, Mitchelle Johnson
>             <mitchelle.bit at gmail.com <mailto:mitchelle.bit at gmail.com>>
>             wrote:
>
>                 Hi Ankit,
>
>                 I think the CDR's (Call Detail Records) can give you
>                 information about this, the 'billsec' in XML CDR will
>                 show if the call is answered by the voicemail or not.
>
>                 Thanks,
>                 Mitchelle
>
>                 On Wed, May 20, 2015 at 10:37 AM, Ashwin Jain
>                 <ashwinrkjain at gmail.com> wrote:
>
>                     Hi all,
>
>                     We are using freeswitch 1.4.18. Our setup is very
>                     basic, we have an incoming call, which we
>                     terminate on mobile phones (via gateways).
>
>                     Is there anyway to detect if the call is picked up
>                     by mobile phone provider voicemail?
>
>                     -- 
>                     Thanks and Regards,
>                     Ashwin Jain
>
>                     _________________________________________________________________________
>                     Professional FreeSWITCH Consulting Services:
>                     consulting at freeswitch.org
>                     http://www.freeswitchsolutions.com
>
>                     Official FreeSWITCH Sites
>                     http://www.freeswitch.org
>                     http://confluence.freeswitch.org
>                     http://www.cluecon.com
>
>                     FreeSWITCH-users mailing list
>                     FreeSWITCH-users at lists.freeswitch.org
>                     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                     http://www.freeswitch.org
>
>
>
>             _________________________________________________________________________
>             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 mitchelle.bit at gmail.com  Wed May 20 11:51:38 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Wed, 20 May 2015 13:21:38 +0530
Subject: [Freeswitch-users] Detecting external voicemail
In-Reply-To: <555C3529.5070407@freeswitch.org>
References: <CAM9SNVT=XgDMDpDLAJ5NXdPOy1NirrCjxTRTKvDnkH7xDkrPag@mail.gmail.com>
	<CANBm6Sxf-O8x=XoR7sCjvpyR2Lj7i8swHoKL-dVW1wNfwnxzHA@mail.gmail.com>
	<CAEwTsAghfg3CbFmrA38c7vywOhREdi_pG-tHuFSiKp3MQPrPsw@mail.gmail.com>
	<CANBm6Sz_ivkWEjCJgHgF88sxi09CUmhC_ijjWJ9u+mXReB_ofg@mail.gmail.com>
	<CALXCt0pDrjfnpJNDwSKeHZ-YoyT+-S0L9mB8knZ1YYuTMeSmXw@mail.gmail.com>
	<CANBm6Swgmg=vvHhpPv0q=t3Yh4fp-Gnxnz2mS_Fp66XaiAzBJw@mail.gmail.com>
	<555C3529.5070407@freeswitch.org>
Message-ID: <CANBm6SySUNjq=nO=WXQNh-H70KiF1+s_cRr2bD=U5mZKt_vCqw@mail.gmail.com>

Hi Yossi,
thanks a lot for the information.

Regards,
Mitchelle

On Wed, May 20, 2015 at 12:48 PM, I put the Who? in Mishehu <
mishehu at freeswitch.org> wrote:

>  That machine can be the current switch or another switch even.  If you
> have an extension where one of the applications executed is "answer", then
> there you have it, you now have billsec that increments until the channel
> is hung up.  I would not advise trying to use billsec as any yardstick.
>
> -Yossi
>
>
> On 05/20/2015 02:05 AM, Mitchelle Johnson wrote:
>
>   Thanks a lot Giovanni,
>
>  So if billsec in XML CDR shows a non-zero value its means the call is
> answered either by machine or by human?
>  If yes, can it not be used as a yard stick to determine if the call is
> answered?
>
>  Thanks again,
>  Mitchelle
>
> On Wed, May 20, 2015 at 12:30 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Billsec is seconds after call has been answered (by a human or by a
>> machine).
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>>  On May 20, 2015 8:56 AM, "Mitchelle Johnson" <mitchelle.bit at gmail.com>
>> wrote:
>>
>>>   Hi Michael,
>>>
>>>  Ashwin has specifically asked for the information on if the phone call
>>> is answered by the voicemail only. If billsec is not the correct way to
>>> figure out whether the call is answered,(irrespective of whether voicemail
>>> or a person answers the call), then what exactly billsec is?
>>>
>>>  Please share your useful knowledge.
>>>
>>>  Thanks,
>>>  Mitchelle
>>>
>>> On Wed, May 20, 2015 at 12:13 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>>
>>>> Billsec has nothing at all to do with if its answered by voicemail or a
>>>> person.  There is no definitive way to know, but as said elsewhere you can
>>>> use group confirm or some sort of answering machine detection to try to
>>>> figure it out, but it won't be 100%
>>>>
>>>> On Wednesday, May 20, 2015, Mitchelle Johnson <mitchelle.bit at gmail.com>
>>>> wrote:
>>>>
>>>>>   Hi Ankit,
>>>>>
>>>>>  I think the CDR's (Call Detail Records) can give you information
>>>>> about this, the 'billsec' in XML CDR will show if the call is answered by
>>>>> the voicemail or not.
>>>>>
>>>>>  Thanks,
>>>>>  Mitchelle
>>>>>
>>>>> On Wed, May 20, 2015 at 10:37 AM, Ashwin Jain <ashwinrkjain at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi all,
>>>>>>
>>>>>>  We are using freeswitch 1.4.18. Our setup is very basic, we have an
>>>>>> incoming call, which we terminate on mobile phones (via gateways).
>>>>>>
>>>>>>  Is there anyway to detect if the call is picked up by mobile phone
>>>>>> provider voicemail?
>>>>>>
>>>>>>  --
>>>>>>  Thanks and Regards,
>>>>>> Ashwin Jain
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/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 ajithdesilva at gmail.com  Wed May 20 11:44:19 2015
From: ajithdesilva at gmail.com (Ajith de Silva)
Date: Wed, 20 May 2015 07:44:19 +0000 (UTC)
Subject: [Freeswitch-users] =?utf-8?q?MANDATORY=5FIE=5FMISSING_when_callin?=
	=?utf-8?q?g_Linphone?=
Message-ID: <loom.20150520T093257-282@post.gmane.org>

Hi,
I am trying to send a call to FreeBAPX from freeSWITCH.
Allow the freeSWithc IP in freePABX.

Followed the following sequence to do the calls

Step 1 (staic user)
1. Register user using the xml file in the directory
2. Dial defined code "5151"
3. Bridge the call to FreePABX.
5. FreeBAPX extension rings & answer
4. Both parties can hear the voice.


Step 2 (dynamic user)
1. Register user using the xml provided by the web server. 
   fetch it using the xml_curl & I get 

<document type="freeswitch/xml">
<section name="directory">
<domain name="abc.com">
        <params>
                <param name="dial-string" value="
{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_us
er}@${dialed_domain})}"/>
        </params>
        <groups>
                <group name="default">
                <users>
                <user id="1000">
                <params>
                        <param name="password" value="123123"/>
                </params>
                <variables>
                        <variable name="user_context" value="default" />
                        <variable name="accountcode" value="1000" />
                </variables>
                </user>
                </users>
                </group>
        </groups>
</domain>
</section>
</document>


2. Dial defined code "5151"
3. Try Bridge the call to FreePABX.
4. Call fail with following issue
Cannot locate any authentication credentials to complete an 
authentication request for realm '"asterisk"'

 Originate Resulted in Error Cause: 96 [MANDATORY_IE_MISSING]

 2015-05-20 07:29:59.782326 [INFO] mod_dptools.c:3244 Originate Failed.  
Cause: MANDATORY_IE_MISSING
 
mod_sofia.c:413 Channel sofia/internal/1000 at emrostel.com hanging up, 
cause: MANDATORY_IE_MISSING
2015-05-20 07:29:59.782326 [DEBUG] mod_sofia.c:549 Responding to INVITE 
with: 480

Can you please me guide what did I miss over there.

Best Regards,
Ajith de Silva





From brian at freeswitch.org  Wed May 20 16:43:03 2015
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 May 2015 07:43:03 -0500
Subject: [Freeswitch-users] MANDATORY_IE_MISSING when calling Linphone
In-Reply-To: <loom.20150520T093257-282@post.gmane.org>
References: <loom.20150520T093257-282@post.gmane.org>
Message-ID: <CAEJMVkAjqK9s9sFd0rA-eDCZGEVrSOTx1c70c5c6F-40XF=9cw@mail.gmail.com>

It usually means you were challenged on your outbound INVITE by the remote
party and you have no way to answer that challenge.

On Wed, May 20, 2015 at 2:44 AM, Ajith de Silva <ajithdesilva at gmail.com>
wrote:

> Hi,
> I am trying to send a call to FreeBAPX from freeSWITCH.
> Allow the freeSWithc IP in freePABX.
>
> Followed the following sequence to do the calls
>
> Step 1 (staic user)
> 1. Register user using the xml file in the directory
> 2. Dial defined code "5151"
> 3. Bridge the call to FreePABX.
> 5. FreeBAPX extension rings & answer
> 4. Both parties can hear the voice.
>
>
> Step 2 (dynamic user)
> 1. Register user using the xml provided by the web server.
>    fetch it using the xml_curl & I get
>
> <document type="freeswitch/xml">
> <section name="directory">
> <domain name="abc.com">
>         <params>
>                 <param name="dial-string" value="
> {presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_us
> er}@${dialed_domain})}"/>
>         </params>
>         <groups>
>                 <group name="default">
>                 <users>
>                 <user id="1000">
>                 <params>
>                         <param name="password" value="123123"/>
>                 </params>
>                 <variables>
>                         <variable name="user_context" value="default" />
>                         <variable name="accountcode" value="1000" />
>                 </variables>
>                 </user>
>                 </users>
>                 </group>
>         </groups>
> </domain>
> </section>
> </document>
>
>
> 2. Dial defined code "5151"
> 3. Try Bridge the call to FreePABX.
> 4. Call fail with following issue
> Cannot locate any authentication credentials to complete an
> authentication request for realm '"asterisk"'
>
>  Originate Resulted in Error Cause: 96 [MANDATORY_IE_MISSING]
>
>  2015-05-20 07:29:59.782326 [INFO] mod_dptools.c:3244 Originate Failed.
> Cause: MANDATORY_IE_MISSING
>
> mod_sofia.c:413 Channel sofia/internal/1000 at emrostel.com hanging up,
> cause: MANDATORY_IE_MISSING
> 2015-05-20 07:29:59.782326 [DEBUG] mod_sofia.c:549 Responding to INVITE
> with: 480
>
> Can you please me guide what did I miss over there.
>
> Best Regards,
> Ajith de 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

ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> |
Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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 naveen32india at gmail.com  Wed May 20 17:39:44 2015
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Wed, 20 May 2015 19:09:44 +0530
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
	<CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
Message-ID: <CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>

I am aware of uuid_hangup. Indeed my intention to know the way to send (low
level)sip messages
from fs console for a selected user.


On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com> wrote:

>  I would like to reject the call when it ringing from the fs console.
>
>
> uuid_hangup
>
>
> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com> wrote:
>
>> I am trying to do the following,  I would like to reject the call when it
>> ringing from the fs console.
>> And second thing is I am pretty much eager to know the way to send
>> sip(signaling) message manually for a
>> selected channel from fs console.
>>
>>
>> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> respond is one way, what exact message are you trying to send, and at
>>> what point in the call.  There are capabilites to trigger re-invites in
>>> some situations, transfer, send notify or info or message.  It depends on
>>> what exactly you are trying to do.
>>>
>>>
>>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com>
>>> wrote:
>>>
>>> Hi,
>>>
>>> I am wondering how to send sip signaling message from the fs console for
>>> the particular user/caller.
>>> I found  respond dialplan application to send sip message back to the
>>> caller.
>>> Like
>>>
>>> <action application="respond" data="480 Try again later"/>
>>>
>>> ?Is there any way to send sip message back to the caller. One use case
>>> is call rejection or playing with SIP?
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Thanks & Regards,
>> Naveen Tamanam
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regards,
Naveen Tamanam
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From olegstolyar at gmail.com  Wed May 20 18:40:17 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 20 May 2015 07:40:17 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
Message-ID: <CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>

But isn't that based on the session-timeout param which defaults to 30
min?  My re-invites occur much sooner than 30 min into a call.  Or does
session-timeout param only control sessions initiated by FS while incoming
sessions use the minimum-session-expires param if it's not explicitly
passed by the session initiator?

On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com> wrote:

> session timer
>
>
> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>> Thanks Michael, I'll see if we can do that!
>>
>> So, is the re-INVITE legit and the problem is that JsSip does not respond
>> to it?  Still, I am curious what is triggering the re-INVITE.
>>
>> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> I think the sip.js guys fixed this issue when they forked jssip.  I'd
>>> suggest using that.
>>>
>>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>> wrote:
>>> >
>>> > Hi guys,
>>> >
>>> > Several weeks ago I started getting an occasional problem where FS is
>>> sending an INVITE to the other side in the middle of a call, the other side
>>> does not respond and FS hangs up the leg.  Below is the relevant log.  The
>>> user experience is that they keep talking and hearing each other up to the
>>> very end.  I have a recording of that call, so can confirm.
>>> >
>>> > The call uses WebRTC and is originated by JsSip from Chrome.  Then the
>>> user is put into a conference but I doubt it's relevant in this case since
>>> the INVITE and disconnect are not happening from mod_conference
>>> >
>>> > I suspect it's a re-INVITE but what triggers FS to send it?  I
>>> couldn't find anything in the logs that could shed light.
>>> >
>>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>>> >
>>> ------------------------------------------------------------------------
>>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>>> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>>> >    Max-Forwards: 70
>>> >    From:
>>> <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>>> >;tag=KQecUSr12rSQp
>>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>>> >    Call-ID: g8980rbrbk2t45oj5mru
>>> >    CSeq: 75703945 INVITE
>>> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@
>>> ##.##.###.###:5080;transport=udp>
>>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> >    Supported: timer, path, replaces
>>> >    Session-Expires: 120;refresher=uac
>>> >    Min-SE: 120
>>> >    Content-Type: application/sdp
>>> >    Content-Disposition: session
>>> >    Content-Length: 825
>>> >
>>> >    v=0
>>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>>> >    s=FreeSWITCH
>>> >    c=IN IP4 ##.##.##.##
>>> >    t=0 0
>>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>> >    m=audio 22670 RTP/SAVPF 9 126 106
>>> >    a=rtpmap:9 G722/8000
>>> >    a=rtpmap:126 telephone-event/8000
>>> >    a=rtpmap:106 CN/8000
>>> >    a=ptime:20
>>> >    a=fingerprint:sha-256
>>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>>> >    a=rtcp-mux
>>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host
>>> generation 0
>>> >
>>> ------------------------------------------------------------------------
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>> [DEBUG] switch_core_session.c:1061 Send signal
>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>> entering state [calling][0]
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>> [DEBUG] switch_core_session.c:1061 Send signal
>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>> [DEBUG] switch_core_session.c:1061 Send signal
>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>> entering state [terminating][503]
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>> [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
>>> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>> [DEBUG] switch_channel.c:3242 Send signal
>>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>> [DEBUG] switch_core_session.c:1396 Send signal
>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] mod_conference.c:5057 Channel leaving conference, cause:
>>> NORMAL_TEMPORARY_FAILURE
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid
>>> skip receive message [UNBRIDGE] (channel is hungup already)
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] switch_core_media.c:7772
>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>> [REFER_EVENT] (channel is hungup already)
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
>>> Restore previous codec G722:9.
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] switch_core_session.c:2901
>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] switch_core_state_machine.c:535
>>> (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to
>>> sleep
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] switch_core_state_machine.c:472
>>> (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>>> CS_HANGUP
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] switch_core_state_machine.c:735
>>> (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE
>>> -> HANGUP
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] switch_core_state_machine.c:737
>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>> hanging up, cause: NORMAL_TEMPORARY_FAILURE
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] switch_core_state_machine.c:60
>>> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause:
>>> NORMAL_TEMPORARY_FAILURE
>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>> [DEBUG] switch_core_state_machine.c:737
>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to
>>> sleep
>>> >
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From chirag at ncc.co.in  Wed May 20 18:36:39 2015
From: chirag at ncc.co.in (Chirag Ajmera)
Date: Wed, 20 May 2015 20:06:39 +0530
Subject: [Freeswitch-users] NAT Settings with dynamic ip & port forward
Message-ID: <CANxXOfT3rOJ6uHLgQWk=EM8=osQw8gV78UP0Pk=Vq3B2rT6pSQ@mail.gmail.com>

Kindly can anyone pls guide with how to achieve NAT settings in Freeswitch

Scenario
------------

Local PC (Bria) <-------------> FS Server (192.168.1.199)
(192.168.1.91)

Remote Iphone (Bria) <-------------> Internet <-------------> Router (Port
Fwd) <-------------> FS Server

Router is on dynamic ip - Domain - myhome.ddns.net

*ISSUE - cannot register remote extension*

*Also, the same scenario works well with freepbx and asterisk (port
forwards are ok)*

[image: Inline image 1]

Thank You
Best Regards,
Chirag A.
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From mike at jerris.com  Wed May 20 19:26:08 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 May 2015 11:26:08 -0400
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
	<CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
	<CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
Message-ID: <796198D0-5ABF-4A76-BDEF-0405CB2EF671@jerris.com>

Please review my previous question. If you can't answer that, no one can help you.

> On May 20, 2015, at 9:39 AM, Naveen Tamanam <naveen32india at gmail.com> wrote:
> 
> I am aware of uuid_hangup. Indeed my intention to know the way to send (low level)sip messages 
> from fs console for a selected user. 
> 
> 
> On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
>  I would like to reject the call when it ringing from the fs console. 
> 
> uuid_hangup
>  
> 
> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
> I am trying to do the following,  I would like to reject the call when it ringing from the fs console. 
> And second thing is I am pretty much eager to know the way to send sip(signaling) message manually for a 
> selected channel from fs console. 
> 
> 
> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> respond is one way, what exact message are you trying to send, and at what point in the call.  There are capabilites to trigger re-invites in some situations, transfer, send notify or info or message.  It depends on what exactly you are trying to do.
> 
> 
>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> 
>> Hi, 
>> 
>> I am wondering how to send sip signaling message from the fs console for the particular user/caller. 
>> I found  respond dialplan application to send sip message back to the caller. 
>> Like 
>> <action application="respond" data="480 Try again later"/>
>> ?Is there any way to send sip message back to the caller. One use case is call rejection or playing with SIP?
>> 
> 

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From giggsey at gmail.com  Wed May 20 20:18:03 2015
From: giggsey at gmail.com (Joshua Gigg)
Date: Wed, 20 May 2015 17:18:03 +0100
Subject: [Freeswitch-users] Update CLI on Freeswitch originated call
In-Reply-To: <CADcg8iUT=TybcX+9_X1+3kuHQde+Edj+2u+0=XcKf2Fccu5+sg@mail.gmail.com>
References: <CADcg8iUL1EsMdedCm6793O83E+pK=UBPFBFqkgbVbJtjvdCmZw@mail.gmail.com>
	<CAKzWOxUut2N0-o+SKcqKkEpFp6g_9UyFHrnr7y2FobutycNRhw@mail.gmail.com>
	<CADcg8iUT=TybcX+9_X1+3kuHQde+Edj+2u+0=XcKf2Fccu5+sg@mail.gmail.com>
Message-ID: <CADcg8iWyxxZjF1sesYOxT8di2L1N16=cODjjysAqNci=354+YQ@mail.gmail.com>

Digging up an old thread, but I've been looking into this a bit more.

What I want to do is described in RFC 4916. What I want to try to use an
application / api command to generate a SIP INFO/UPDATE packet that would
update the Caller ID.

On 29 January 2015 at 20:06, Joshua Gigg <giggsey at gmail.com> wrote:

> When a transfer completes, FreeSWITCH will send an UPDATE message to the
> SIP server updating the caller id.
>
> Is there a way of making FreeSWITCH send this message via a
> dialplan/command?
>
> On 29 January 2015 at 17:58, Michael Collins <msc at freeswitch.org> wrote:
>
>> Could you expound upon this question a bit? What does "update the CLI"
>> mean?
>> Thanks,
>> MC
>>
>> On Thu, Jan 29, 2015 at 1:20 AM, Joshua Gigg <giggsey at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> Is it possible to update the CLI at will once a Freeswitch originated
>>> call has been answered?
>>>
>>> I know it can update during a transfer, but I want to be able to control
>>> it directly myself.
>>>
>>> --
>>> Joshua Gigg
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Joshua Gigg
>



-- 
Joshua Gigg
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From mike at jerris.com  Wed May 20 20:26:43 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 May 2015 12:26:43 -0400
Subject: [Freeswitch-users] Update CLI on Freeswitch originated call
In-Reply-To: <CADcg8iWyxxZjF1sesYOxT8di2L1N16=cODjjysAqNci=354+YQ@mail.gmail.com>
References: <CADcg8iUL1EsMdedCm6793O83E+pK=UBPFBFqkgbVbJtjvdCmZw@mail.gmail.com>
	<CAKzWOxUut2N0-o+SKcqKkEpFp6g_9UyFHrnr7y2FobutycNRhw@mail.gmail.com>
	<CADcg8iUT=TybcX+9_X1+3kuHQde+Edj+2u+0=XcKf2Fccu5+sg@mail.gmail.com>
	<CADcg8iWyxxZjF1sesYOxT8di2L1N16=cODjjysAqNci=354+YQ@mail.gmail.com>
Message-ID: <789BFCEB-3922-4952-BF04-4651B321CC0F@jerris.com>

this should happen by default unless you have ignore_dispaly_updates=true.


> On May 20, 2015, at 12:18 PM, Joshua Gigg <giggsey at gmail.com> wrote:
> 
> Digging up an old thread, but I've been looking into this a bit more.
> 
> What I want to do is described in RFC 4916. What I want to try to use an application / api command to generate a SIP INFO/UPDATE packet that would update the Caller ID.
> 
> On 29 January 2015 at 20:06, Joshua Gigg <giggsey at gmail.com <mailto:giggsey at gmail.com>> wrote:
> When a transfer completes, FreeSWITCH will send an UPDATE message to the SIP server updating the caller id.
> 
> Is there a way of making FreeSWITCH send this message via a dialplan/command?
> 
> On 29 January 2015 at 17:58, Michael Collins <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
> Could you expound upon this question a bit? What does "update the CLI" mean?
> Thanks,
> MC
> 
> On Thu, Jan 29, 2015 at 1:20 AM, Joshua Gigg <giggsey at gmail.com <mailto:giggsey at gmail.com>> wrote:
> Hi,
> 
> Is it possible to update the CLI at will once a Freeswitch originated call has been answered?
> 
> I know it can update during a transfer, but I want to be able to control it directly myself.
> 

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From victor.medina at cibersys.com  Wed May 20 20:38:39 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Wed, 20 May 2015 12:08:39 -0430
Subject: [Freeswitch-users] JsSIP -> OverSIP -> Freeswitch
In-Reply-To: <0BC899C0-6842-424B-B83E-D0DCFF652CC3@jerris.com>
References: <CACXsapPPc2jti7K0H0C4nm6Sw1bMTzAt70SoqNish=uuyBk=qw@mail.gmail.com>
	<CAEJMVkDBv2XmAHhySS_faVTrDo6K2dgy3+OEL7k=ER33dT7kGg@mail.gmail.com>
	<CACXsapPGxPT3mGvhTMeqk_QTuPXkGj7ibDJ91G8c9rGpxop7WA@mail.gmail.com>
	<8A1140F3-33E1-4F91-A639-A6BCED23FC98@jerris.com>
	<CACXsapOeMWeauuJW=1hr7FA2Aa9eKx+5yp_70-8MDB8A3eFDJw@mail.gmail.com>
	<D2308758-768B-413F-A88B-616D87A51941@freeswitch.org>
	<0BC899C0-6842-424B-B83E-D0DCFF652CC3@jerris.com>
Message-ID: <CACXsapO36jZuCT=htN3QOGH=522pUhxhtTnn8aWtq0C=bf-8jw@mail.gmail.com>

Yeap! THAT DID IT!

Thanks you all.

I also tried disabling the timers in jssip and also worked.

2015-05-19 19:44 GMT-04:30 Michael Jerris <mike at jerris.com>:

> from the default configs:
>
> <!--<param name="minimum-session-expires" value="120"/>-->
>
> it looks like that got uncommented.  That being said, 90 is an insanely
> low session expires value, recommended value for that is 30 minutes (1800)
>
> https://tools.ietf.org/html/rfc4028
>
> Mike
>
>
> On May 19, 2015, at 7:41 PM, Ken Rice <krice at freeswitch.org> wrote:
>
> Your session time is too short make it longer like 300 secs
>
> Sent from my iPhone
>
> On May 19, 2015, at 6:34 PM, Victor Medina <victor.medina at cibersys.com>
> wrote:
>
> Hi Michael!
>
> This is the the sip debug on the freeswitch directly jssip -> freeswitch
> -> 7443
>
> freeswitch at internal> sofia profile internal siptrace on
> Enabled sip debugging on internal
> recv 5122 bytes from wss/[208.84.81.64]:57975 at 19:31:56.205487:
>    ------------------------------------------------------------------------
>    INVITE sip:9196 at conference.cibersys.com SIP/2.0
>    Via: SIP/2.0/WSS 7rrpouob7plu.invalid;branch=z9hG4bK1885537
>    Max-Forwards: 69
>    To: <sip:9196 at conference.cibersys.com>
>    From: <sip:1000 at conference.cibersys.com>;tag=i26sin3126
>    Call-ID: di8dteiqmu6ofrf75pbd
>    CSeq: 6473 INVITE
>    X-Can-Renegotiate: true
>    Contact: <sip:e3e4td20 at 7rrpouob7plu.invalid;transport=ws;ob>
>    Content-Type: application/sdp
>    Session-Expires: 90
>    Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
>    Supported: timer,ice,outbound
>    User-Agent: JsSIP 0.6.26
>    Content-Length: 4574
>
>    v=0
>    o=- 3581044066039556708 2 IN IP4 127.0.0.1
>    s=-
>    t=0 0
>    a=group:BUNDLE audio video
>    a=msid-semantic: WMS iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD
>    m=audio 55125 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>    c=IN IP4 208.84.81.64
>    a=rtcp:55125 IN IP4 208.84.81.64
>    a=candidate:1505006532 1 udp 2122255103
> 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>    a=candidate:1505006532 2 udp 2122255103
> 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>    a=candidate:2999745851 1 udp 2122194687 192.168.56.1 55124 typ host
> generation 0
>    a=candidate:2999745851 2 udp 2122194687 192.168.56.1 55124 typ host
> generation 0
>    a=candidate:536231733 1 udp 2122129151 10.0.80.221 55125 typ host
> generation 0
>    a=candidate:536231733 2 udp 2122129151 10.0.80.221 55125 typ host
> generation 0
>    a=candidate:389508916 1 tcp 1518275327
> 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>    a=candidate:389508916 2 tcp 1518275327
> 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>    a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host
> tcptype active generation 0
>    a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host
> tcptype active generation 0
>    a=candidate:1366672325 1 tcp 1518149375 10.0.80.221 0 typ host tcptype
> active generation 0
>    a=candidate:1366672325 2 tcp 1518149375 10.0.80.221 0 typ host tcptype
> active generation 0
>    a=candidate:3064574176 1 udp 1685921535 208.84.81.64 55125 typ srflx
> raddr 10.0.80.221 rport 55125 generation 0
>    a=candidate:3064574176 2 udp 1685921535 208.84.81.64 55125 typ srflx
> raddr 10.0.80.221 rport 55125 generation 0
>    a=ice-ufrag:1UvloQ6OnbX0r5t4
>    a=ice-pwd:mE9PaCEu/IMEtzv9L8YgDRCh
>    a=fingerprint:sha-256
> 61:AC:51:08:4B:E0:AB:B0:89:7B:18:05:CA:49:B3:71:9B:5A:77:99:07:58:67:A9:3E:0E:72:92:13:F5:1E:F7
>    a=setup:actpass
>    a=mid:audio
>    a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>    a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>    a=sendrecv
>    a=rtcp-mux
>    a=rtpmap:111 opus/48000/2
>    a=fmtp:111 minptime=10; useinbandfec=1
>    a=rtpmap:103 ISAC/16000
>    a=rtpmap:104 ISAC/32000
>    a=rtpmap:9 G722/8000
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:106 CN/32000
>    a=rtpmap:105 CN/16000
>    a=rtpmap:13 CN/8000
>    a=rtpmap:126 telephone-event/8000
>    a=maxptime:60
>    a=ssrc:691038544 cname:WW2sd3POh47MtjWG
>    a=ssrc:691038544 msid:iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD
> 4cca9568-6c68-48b7-bac4-ed658260a073
>    a=ssrc:691038544 mslabel:iTaMd7UVZzDvEsx9dBzv4DvSRoA6A6uwAIAD
>    a=ssrc:691038544 label:4cca9568-6c68-48b7-bac4-ed658260a073
>    m=video 55125 RTP/SAVPF 100 116 117 96
>    c=IN IP4 208.84.81.64
>    a=rtcp:55125 IN IP4 208.84.81.64
>    a=candidate:1505006532 1 udp 2122255103
> 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>    a=candidate:1505006532 2 udp 2122255103
> 2001::9d38:90d7:10d8:3345:2fab:aebf 55123 typ host generation 0
>    a=candidate:2999745851 1 udp 2122194687 192.168.56.1 55124 typ host
> generation 0
>    a=candidate:2999745851 2 udp 2122194687 192.168.56.1 55124 typ host
> generation 0
>    a=candidate:536231733 1 udp 2122129151 10.0.80.221 55125 typ host
> generation 0
>    a=candidate:536231733 2 udp 2122129151 10.0.80.221 55125 typ host
> generation 0
>    a=candidate:389508916 1 tcp 1518275327
> 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>    a=candidate:389508916 2 tcp 1518275327
> 2001::9d38:90d7:10d8:3345:2fab:aebf 0 typ host tcptype active generation 0
>    a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host
> tcptype active generation 0
>    a=candidate:4233069003 2 tcp 1518214911 192.168.56.1 0 typ host
> tcptype active generation 0
>    a=candidate:1366672325 1 tcp 1518149375 10.0.80.221 0 typ host tcptype
> active generation 0
>    a=candidate:1366672325 2 tcp 1518149375 10.0.80.221 0 typ host tcptype
> active generation 0
>    a=candidate:3064574176 1 udp 1685921535 208.84.81.64 55125 typ srflx
> raddr 10.0.80.221 rport 55125 generation 0
>    a=candidate:3064574176 2 udp 1685921535 208.84.81.64 55125 typ srflx
> raddr 10.0.80.221 rport 55125 generation 0
>    a=ice-ufrag:1UvloQ6OnbX0r5t4
>    a=ice-pwd:mE9PaCEu/IMEtzv9L8YgDRCh
>    a=fingerprint:sha-256
> 61:AC:51:08:4B:E0:AB:B0:89:7B:18:05:CA:49:B3:71:9B:5A:77:99:07:58:67:A9:3E:0E:72:92:13:F5:1E:F7
>    a=setup:actpass
>    a=mid:video
>    a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>    a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>    a=recvonly
>    a=rtcp-mux
>    a=rtpmap:100 VP8/90000
>    a=rtcp-fb:100 ccm fir
>    a=rtcp-fb:100 nack
>    a=rtcp-fb:100 nack pli
>    a=rtcp-fb:100 goog-remb
>    a=rtpmap:116 red/90000
>    a=rtpmap:117 ulpfec/90000
>    a=rtpmap:96 rtx/90000
>    a=fmtp:96 apt=100
>    ------------------------------------------------------------------------
> send 754 bytes to wss/[208.84.81.64]:57975 at 19:31:56.205982:
>    ------------------------------------------------------------------------
>    SIP/2.0 422 Session Interval Too Small
>    Via: SIP/2.0/WSS
> 7rrpouob7plu.invalid;branch=z9hG4bK1885537;received=208.84.81.64;rport=57975
>    From: <sip:1000 at conference.cibersys.com>;tag=i26sin3126
>    To: <sip:9196 at conference.cibersys.com>;tag=ma466p1ScScUa
>    Call-ID: di8dteiqmu6ofrf75pbd
>    CSeq: 6473 INVITE
>    User-Agent:
> FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, presence, as-feature-event,
> dialog, line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer
>    Min-SE: 120
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 290 bytes from wss/[208.84.81.64]:57975 at 19:31:56.483525:
>    ------------------------------------------------------------------------
>    ACK sip:9196 at conference.cibersys.com SIP/2.0
>    Via: SIP/2.0/WSS 7rrpouob7plu.invalid;branch=z9hG4bK1885537
>    To: <sip:9196 at conference.cibersys.com>;tag=ma466p1ScScUa
>    From: <sip:1000 at conference.cibersys.com>;tag=i26sin3126
>    Call-ID: di8dteiqmu6ofrf75pbd
>    CSeq: 6473 ACK
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> freeswitch at internal>
>
>
> 2015-05-19 18:50 GMT-04:30 Michael Jerris <mike at jerris.com>:
>
>> what is oversip?  you shouldn't need any layers between FS and jssip.
>>
>> On May 19, 2015, at 6:58 PM, Victor Medina <victor.medina at cibersys.com>
>> wrote:
>>
>> SURE!
>>
>> This is the Sofia Internal side...
>>
>> http://pastebin.com/xwBjzMCX
>>
>> This is Javascript console... (JsSIP -> OverSIP)
>>
>> http://pastebin.com/XmXRt0nV
>>
>> Thanks.
>>
>> 2015-05-19 18:15 GMT-04:30 Brian West <brian at freeswitch.org>:
>>
>>> The whole sip exchange would need to be seen for me to tell, it could be
>>> a bug, it could be a config issue... not exactly sure yet.  Can you paste
>>> bin it?
>>>
>>> Thanks,
>>>
>>> On Tue, May 19, 2015 at 5:42 PM, Victor Medina <
>>> victor.medina at cibersys.com> wrote:
>>>
>>>> Hi guys!
>>>>
>>>> Im having some problems while trying to connect to a Freswitch PBX
>>>> using OverSip and JsSIP.
>>>>
>>>> Using Jssip i can recive calls but when calling from the OverSIP/JsSIp
>>>> I always get a 422 error.
>>>>
>>>> Softphone -> WebRTC ext calls OK
>>>> WebRTC -> Softphone fails
>>>> WebRTC -> ECHO TEST call on freeswitch also fails
>>>>
>>>> Can somebody help find out what this could be?
>>>>
>>>> SIP/2.0 422 Session Interval Too Small
>>>> Via: SIP/2.0/WSS vv3f75qcos24.invalid;branch=z9hG4bK1673051
>>>> From: "test" <sip:1002 at conference.cibersys.com>;tag=5or80k3oc9
>>>> To: <sip:9196 at conference.cibersys.com>;tag=Q3Xyt4X2vSX6Q
>>>> Call-ID: 4norlmjdkqok6smt0vd4
>>>> CSeq: 9416 INVITE
>>>> User-Agent:
>>>> FreeSWITCH-mod_sofia/1.5.15b+git~20150511T062448Z~1baaa24f9e~64bit
>>>> Accept: application/sdp
>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>> Supported: timer, path, replaces
>>>> Allow-Events: talk, hold, conference, presence, as-feature-event,
>>>> dialog, line-seize, call-info, sla, include-session-description,
>>>> presence.winfo, message-summary, refer
>>>> Min-Se: 120
>>>> Content-Length: 0
>>>>
>>>> BTW... already asked @oversip guys... But wanted to try this in here
>>>> just in case! =)
>>>>
>>>> Thanks.
>>>>
>>>> --
>>>>
>>>>
>>>>
>>>> V?ctor E. Medina M.
>>>> Platform Architect / Chief Infrastructure
>>>> +58424 291 4561
>>>> BB #79A8AFA2
>>>> @VMCibersys
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> ClueCon 2015 Call for Speakers
>>> <https://www.cluecon.com/call-for-speakers/> | Register
>>> <https://freeswitch.com/cart.php?gid=1> TODAY! | 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
>>>
>>
>>
>>
>> --
>>
>>
>>
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561
>> BB #79A8AFA2
>> @VMCibersys
>>
>>  _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From olegstolyar at gmail.com  Thu May 21 02:44:36 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 20 May 2015 15:44:36 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
Message-ID: <CAC8JMOK7jD5sWTGn+KecrbyeOFpqPpydTFKHzKFupUPcDEJbDQ@mail.gmail.com>

I looked at my logs more carefully and noticed that FS sends such
re-INVITES for many of my calls but not all.  It is always sent about one
minute after the call is established, so it's probably not based on session
timer, right?

In most cases when it's sent JsSip answers with a 100 Trying.  In this case
the call stays on.  In some rare cases JsSip does not respond and FS
disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log above.


I will investigate why JsSip sometimes does not respond but I would still
like to figure out why FS is sending those re-INVITES to begin with?

I thought it could be happening if JsSip does not respond to OK with an ACK
but that's not the case.  The ACK is there for these calls.

There is a variable nonce-ttl in the sip profile that defaults to 60
seconds.  Perhaps it has something to do with that?  This profile does not
require authentication or registration at all, though.

On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> But isn't that based on the session-timeout param which defaults to 30
> min?  My re-invites occur much sooner than 30 min into a call.  Or does
> session-timeout param only control sessions initiated by FS while incoming
> sessions use the minimum-session-expires param if it's not explicitly
> passed by the session initiator?
>
> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> session timer
>>
>>
>> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>
>>> Thanks Michael, I'll see if we can do that!
>>>
>>> So, is the re-INVITE legit and the problem is that JsSip does not
>>> respond to it?  Still, I am curious what is triggering the re-INVITE.
>>>
>>> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> I think the sip.js guys fixed this issue when they forked jssip.  I'd
>>>> suggest using that.
>>>>
>>>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>>> wrote:
>>>> >
>>>> > Hi guys,
>>>> >
>>>> > Several weeks ago I started getting an occasional problem where FS is
>>>> sending an INVITE to the other side in the middle of a call, the other side
>>>> does not respond and FS hangs up the leg.  Below is the relevant log.  The
>>>> user experience is that they keep talking and hearing each other up to the
>>>> very end.  I have a recording of that call, so can confirm.
>>>> >
>>>> > The call uses WebRTC and is originated by JsSip from Chrome.  Then
>>>> the user is put into a conference but I doubt it's relevant in this case
>>>> since the INVITE and disconnect are not happening from mod_conference
>>>> >
>>>> > I suspect it's a re-INVITE but what triggers FS to send it?  I
>>>> couldn't find anything in the logs that could shed light.
>>>> >
>>>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>>>> >
>>>> ------------------------------------------------------------------------
>>>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>>>> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>>>> >    Max-Forwards: 70
>>>> >    From:
>>>> <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>>>> >;tag=KQecUSr12rSQp
>>>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>>>> >    Call-ID: g8980rbrbk2t45oj5mru
>>>> >    CSeq: 75703945 INVITE
>>>> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@
>>>> ##.##.###.###:5080;transport=udp>
>>>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>>>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>> >    Supported: timer, path, replaces
>>>> >    Session-Expires: 120;refresher=uac
>>>> >    Min-SE: 120
>>>> >    Content-Type: application/sdp
>>>> >    Content-Disposition: session
>>>> >    Content-Length: 825
>>>> >
>>>> >    v=0
>>>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>>>> >    s=FreeSWITCH
>>>> >    c=IN IP4 ##.##.##.##
>>>> >    t=0 0
>>>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>> >    m=audio 22670 RTP/SAVPF 9 126 106
>>>> >    a=rtpmap:9 G722/8000
>>>> >    a=rtpmap:126 telephone-event/8000
>>>> >    a=rtpmap:106 CN/8000
>>>> >    a=ptime:20
>>>> >    a=fingerprint:sha-256
>>>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>>>> >    a=rtcp-mux
>>>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>>>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>>>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>>>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>>>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>>>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>>>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host
>>>> generation 0
>>>> >
>>>> ------------------------------------------------------------------------
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>> entering state [calling][0]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>> entering state [terminating][503]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
>>>> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [DEBUG] switch_channel.c:3242 Send signal
>>>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [DEBUG] switch_core_session.c:1396 Send signal
>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] mod_conference.c:5057 Channel leaving conference, cause:
>>>> NORMAL_TEMPORARY_FAILURE
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid
>>>> skip receive message [UNBRIDGE] (channel is hungup already)
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_media.c:7772
>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>> [REFER_EVENT] (channel is hungup already)
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
>>>> Restore previous codec G722:9.
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_session.c:2901
>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:535
>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to
>>>> sleep
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:472
>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>>>> CS_HANGUP
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:735
>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE
>>>> -> HANGUP
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:737
>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>> hanging up, cause: NORMAL_TEMPORARY_FAILURE
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:60
>>>> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause:
>>>> NORMAL_TEMPORARY_FAILURE
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:737
>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to
>>>> sleep
>>>> >
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From mike at jerris.com  Thu May 21 07:19:11 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 May 2015 23:19:11 -0400
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMOK7jD5sWTGn+KecrbyeOFpqPpydTFKHzKFupUPcDEJbDQ@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAC8JMOK7jD5sWTGn+KecrbyeOFpqPpydTFKHzKFupUPcDEJbDQ@mail.gmail.com>
Message-ID: <CAEwTsAh9Rawxob9P-3tCndMfayHOuq2fW8H8hKgYW5s03k2jwA@mail.gmail.com>

See my previous suggestion earlier in this thread

On Wednesday, May 20, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> I looked at my logs more carefully and noticed that FS sends such
> re-INVITES for many of my calls but not all.  It is always sent about one
> minute after the call is established, so it's probably not based on session
> timer, right?
>
> In most cases when it's sent JsSip answers with a 100 Trying.  In this
> case the call stays on.  In some rare cases JsSip does not respond and FS
> disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log
> above.
>
> I will investigate why JsSip sometimes does not respond but I would still
> like to figure out why FS is sending those re-INVITES to begin with?
>
> I thought it could be happening if JsSip does not respond to OK with an
> ACK but that's not the case.  The ACK is there for these calls.
>
> There is a variable nonce-ttl in the sip profile that defaults to 60
> seconds.  Perhaps it has something to do with that?  This profile does not
> require authentication or registration at all, though.
>
> On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar at gmail.com
> <javascript:_e(%7B%7D,'cvml','olegstolyar at gmail.com');>> wrote:
>
>> But isn't that based on the session-timeout param which defaults to 30
>> min?  My re-invites occur much sooner than 30 min into a call.  Or does
>> session-timeout param only control sessions initiated by FS while incoming
>> sessions use the minimum-session-expires param if it's not explicitly
>> passed by the session initiator?
>>
>> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com
>> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>>
>>> session timer
>>>
>>>
>>> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','olegstolyar at gmail.com');>> wrote:
>>>
>>>> Thanks Michael, I'll see if we can do that!
>>>>
>>>> So, is the re-INVITE legit and the problem is that JsSip does not
>>>> respond to it?  Still, I am curious what is triggering the re-INVITE.
>>>>
>>>> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>>
>>>>> I think the sip.js guys fixed this issue when they forked jssip.  I'd
>>>>> suggest using that.
>>>>>
>>>>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>>>> wrote:
>>>>> >
>>>>> > Hi guys,
>>>>> >
>>>>> > Several weeks ago I started getting an occasional problem where FS
>>>>> is sending an INVITE to the other side in the middle of a call, the other
>>>>> side does not respond and FS hangs up the leg.  Below is the relevant log.
>>>>> The user experience is that they keep talking and hearing each other up to
>>>>> the very end.  I have a recording of that call, so can confirm.
>>>>> >
>>>>> > The call uses WebRTC and is originated by JsSip from Chrome.  Then
>>>>> the user is put into a conference but I doubt it's relevant in this case
>>>>> since the INVITE and disconnect are not happening from mod_conference
>>>>> >
>>>>> > I suspect it's a re-INVITE but what triggers FS to send it?  I
>>>>> couldn't find anything in the logs that could shed light.
>>>>> >
>>>>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>>>>> >
>>>>> ------------------------------------------------------------------------
>>>>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>>>>> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>>>>> >    Max-Forwards: 70
>>>>> >    From:
>>>>> <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>>>>> >;tag=KQecUSr12rSQp
>>>>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>>>>> >    Call-ID: g8980rbrbk2t45oj5mru
>>>>> >    CSeq: 75703945 INVITE
>>>>> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@
>>>>> ##.##.###.###:5080;transport=udp>
>>>>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>>>>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>> >    Supported: timer, path, replaces
>>>>> >    Session-Expires: 120;refresher=uac
>>>>> >    Min-SE: 120
>>>>> >    Content-Type: application/sdp
>>>>> >    Content-Disposition: session
>>>>> >    Content-Length: 825
>>>>> >
>>>>> >    v=0
>>>>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>>>>> >    s=FreeSWITCH
>>>>> >    c=IN IP4 ##.##.##.##
>>>>> >    t=0 0
>>>>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>> >    m=audio 22670 RTP/SAVPF 9 126 106
>>>>> >    a=rtpmap:9 G722/8000
>>>>> >    a=rtpmap:126 telephone-event/8000
>>>>> >    a=rtpmap:106 CN/8000
>>>>> >    a=ptime:20
>>>>> >    a=fingerprint:sha-256
>>>>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>>>>> >    a=rtcp-mux
>>>>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>>>>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>>>>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>>>>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>>>>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>>>>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>>>>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host
>>>>> generation 0
>>>>> >
>>>>> ------------------------------------------------------------------------
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>> entering state [calling][0]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>> entering state [terminating][503]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
>>>>> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [DEBUG] switch_channel.c:3242 Send signal
>>>>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [DEBUG] switch_core_session.c:1396 Send signal
>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] mod_conference.c:5057 Channel leaving conference, cause:
>>>>> NORMAL_TEMPORARY_FAILURE
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid
>>>>> skip receive message [UNBRIDGE] (channel is hungup already)
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_media.c:7772
>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>> [REFER_EVENT] (channel is hungup already)
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
>>>>> Restore previous codec G722:9.
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_session.c:2901
>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:535
>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to
>>>>> sleep
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:472
>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>>>>> CS_HANGUP
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:735
>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change
>>>>> ACTIVE -> HANGUP
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:737
>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>> hanging up, cause: NORMAL_TEMPORARY_FAILURE
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:60
>>>>> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause:
>>>>> NORMAL_TEMPORARY_FAILURE
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:737
>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to
>>>>> sleep
>>>>> >
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From switcherfree at gmail.com  Thu May 21 11:03:10 2015
From: switcherfree at gmail.com (Free Switcher)
Date: Thu, 21 May 2015 00:03:10 -0700
Subject: [Freeswitch-users] Updated javascript V8 engine
Message-ID: <CAM6Un23UCrRh4RuX-8qLToKNq9kXBnpiqxau=iJjGOQrg-ES+g@mail.gmail.com>

Hi Guys,
The version of V8 javascript engine currently packaged with FreeSwitch
is quite old. Is it possible to build with a more recent version? Any
pointers appreciated.

Thanks,
Andy


From giggsey at gmail.com  Thu May 21 12:05:01 2015
From: giggsey at gmail.com (Joshua Gigg)
Date: Thu, 21 May 2015 09:05:01 +0100
Subject: [Freeswitch-users] Update CLI on Freeswitch originated call
In-Reply-To: <789BFCEB-3922-4952-BF04-4651B321CC0F@jerris.com>
References: <CADcg8iUL1EsMdedCm6793O83E+pK=UBPFBFqkgbVbJtjvdCmZw@mail.gmail.com>
	<CAKzWOxUut2N0-o+SKcqKkEpFp6g_9UyFHrnr7y2FobutycNRhw@mail.gmail.com>
	<CADcg8iUT=TybcX+9_X1+3kuHQde+Edj+2u+0=XcKf2Fccu5+sg@mail.gmail.com>
	<CADcg8iWyxxZjF1sesYOxT8di2L1N16=cODjjysAqNci=354+YQ@mail.gmail.com>
	<789BFCEB-3922-4952-BF04-4651B321CC0F@jerris.com>
Message-ID: <CADcg8iVNCwUmhLBsBOVYe=r1tXwwPtwjbtxB7OyHRrKVjqOVzg@mail.gmail.com>

I do have ignore_dispaly_updates=true, but that's on outbound phones, my
displays were updating with the number being bridged out to.

Also, I want to trigger it manually rather than just automatically when a
transfer takes place.

On 20 May 2015 at 17:26, Michael Jerris <mike at jerris.com> wrote:

> this should happen by default unless you have ignore_dispaly_updates=true.
>
>
> On May 20, 2015, at 12:18 PM, Joshua Gigg <giggsey at gmail.com> wrote:
>
> Digging up an old thread, but I've been looking into this a bit more.
>
> What I want to do is described in RFC 4916. What I want to try to use an
> application / api command to generate a SIP INFO/UPDATE packet that would
> update the Caller ID.
>
> On 29 January 2015 at 20:06, Joshua Gigg <giggsey at gmail.com> wrote:
>
>> When a transfer completes, FreeSWITCH will send an UPDATE message to the
>> SIP server updating the caller id.
>>
>> Is there a way of making FreeSWITCH send this message via a
>> dialplan/command?
>>
>> On 29 January 2015 at 17:58, Michael Collins <msc at freeswitch.org> wrote:
>>
>>> Could you expound upon this question a bit? What does "update the CLI"
>>> mean?
>>> Thanks,
>>> MC
>>>
>>> On Thu, Jan 29, 2015 at 1:20 AM, Joshua Gigg <giggsey at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> Is it possible to update the CLI at will once a Freeswitch originated
>>>> call has been answered?
>>>>
>>>> I know it can update during a transfer, but I want to be able to
>>>> control it directly myself.
>>>>
>>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Joshua Gigg
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From regis.freeswitch.org at tornad.net  Thu May 21 14:29:19 2015
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Thu, 21 May 2015 12:29:19 +0200
Subject: [Freeswitch-users] Update CLI on Freeswitch originated call
In-Reply-To: <CADcg8iVNCwUmhLBsBOVYe=r1tXwwPtwjbtxB7OyHRrKVjqOVzg@mail.gmail.com>
References: <CADcg8iUL1EsMdedCm6793O83E+pK=UBPFBFqkgbVbJtjvdCmZw@mail.gmail.com>
	<CAKzWOxUut2N0-o+SKcqKkEpFp6g_9UyFHrnr7y2FobutycNRhw@mail.gmail.com>
	<CADcg8iUT=TybcX+9_X1+3kuHQde+Edj+2u+0=XcKf2Fccu5+sg@mail.gmail.com>
	<CADcg8iWyxxZjF1sesYOxT8di2L1N16=cODjjysAqNci=354+YQ@mail.gmail.com>
	<789BFCEB-3922-4952-BF04-4651B321CC0F@jerris.com>
	<CADcg8iVNCwUmhLBsBOVYe=r1tXwwPtwjbtxB7OyHRrKVjqOVzg@mail.gmail.com>
Message-ID: <CAHWtr1TQaELgHkpt35_ZMwQ+9JMLR_6RJ49zD7=szr7Jac=Jjw@mail.gmail.com>

+1   if you find how to do this, I'm interessted.

By the way, I have a similar problem to manage MWI.
I solve it by sending an event to the phone to update the message waiting
indicator (on and off).  It use mod_even_socket.
If you find the good message structure for SIP INFO/UPdate for display, it
could do the job I think.

https://wiki.freeswitch.org/wiki/Mod_lua#Sending_an_Event

MWI-Messages-Waiting (yes/no)MWI-Message-Account <any sip url you
want>MWI-Voice-Message x/y (a/b)  <-- read/unread (urgent read/urgent
unread)

And a JIRA reference to

https://freeswitch.org/jira/browse/FS-844


Please keep me informed as I need this feature too :)

Regards,



2015-05-21 10:05 GMT+02:00 Joshua Gigg <giggsey at gmail.com>:

> I do have ignore_dispaly_updates=true, but that's on outbound phones, my
> displays were updating with the number being bridged out to.
>
> Also, I want to trigger it manually rather than just automatically when a
> transfer takes place.
>
> On 20 May 2015 at 17:26, Michael Jerris <mike at jerris.com> wrote:
>
>> this should happen by default unless you have ignore_dispaly_updates=true.
>>
>>
>> On May 20, 2015, at 12:18 PM, Joshua Gigg <giggsey at gmail.com> wrote:
>>
>> Digging up an old thread, but I've been looking into this a bit more.
>>
>> What I want to do is described in RFC 4916. What I want to try to use an
>> application / api command to generate a SIP INFO/UPDATE packet that would
>> update the Caller ID.
>>
>> On 29 January 2015 at 20:06, Joshua Gigg <giggsey at gmail.com> wrote:
>>
>>> When a transfer completes, FreeSWITCH will send an UPDATE message to the
>>> SIP server updating the caller id.
>>>
>>> Is there a way of making FreeSWITCH send this message via a
>>> dialplan/command?
>>>
>>> On 29 January 2015 at 17:58, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>> Could you expound upon this question a bit? What does "update the CLI"
>>>> mean?
>>>> Thanks,
>>>> MC
>>>>
>>>> On Thu, Jan 29, 2015 at 1:20 AM, Joshua Gigg <giggsey at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> Is it possible to update the CLI at will once a Freeswitch originated
>>>>> call has been answered?
>>>>>
>>>>> I know it can update during a transfer, but I want to be able to
>>>>> control it directly myself.
>>>>>
>>>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Joshua Gigg
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 21 15:01:55 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 21 May 2015 13:01:55 +0200
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
	<CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
	<CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
Message-ID: <CALXCt0r4RDeQQ2VSWuPqbm2Umoda50c0bg9NGEUYEbvGVi+ttA@mail.gmail.com>

Naveen,

Please take note that FreeSWITCH is not intended to let the user to
manipulate in arbitrary way SIP dialogs.

You can do that for specific purposes and in.specific cases, within
specific boundaries.

If you are looking to interact with SIP directly and freely, you may want
to look at Kamailio and OpenSIPS.

You can put one of them in front of FreeSWITCH, and you can cross command
them, for example via.lua scripting (both of them and FreeSWITCH can be
scripted in.lua).

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 20, 2015 3:40 PM, "Naveen Tamanam" <naveen32india at gmail.com> wrote:

> I am aware of uuid_hangup. Indeed my intention to know the way to send
> (low level)sip messages
> from fs console for a selected user.
>
>
> On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>>  I would like to reject the call when it ringing from the fs console.
>>
>>
>> uuid_hangup
>>
>>
>> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com> wrote:
>>
>>> I am trying to do the following,  I would like to reject the call when
>>> it ringing from the fs console.
>>> And second thing is I am pretty much eager to know the way to send
>>> sip(signaling) message manually for a
>>> selected channel from fs console.
>>>
>>>
>>> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> respond is one way, what exact message are you trying to send, and at
>>>> what point in the call.  There are capabilites to trigger re-invites in
>>>> some situations, transfer, send notify or info or message.  It depends on
>>>> what exactly you are trying to do.
>>>>
>>>>
>>>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com>
>>>> wrote:
>>>>
>>>> Hi,
>>>>
>>>> I am wondering how to send sip signaling message from the fs console
>>>> for the particular user/caller.
>>>> I found  respond dialplan application to send sip message back to the
>>>> caller.
>>>> Like
>>>>
>>>> <action application="respond" data="480 Try again later"/>
>>>>
>>>> ?Is there any way to send sip message back to the caller. One use case
>>>> is call rejection or playing with SIP?
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Thanks & Regards,
>>> Naveen Tamanam
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Thanks & Regards,
> Naveen Tamanam
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vipkilla at gmail.com  Thu May 21 16:11:51 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Thu, 21 May 2015 08:11:51 -0400
Subject: [Freeswitch-users] Verto client SSL configuration
Message-ID: <CAC-LwPN-_Xv6PUuwRZUqhgqgVoN8FF6HqisY9Eqwexa4j9J07w@mail.gmail.com>

Hello,
The confluence page indicates that both the Verto Client and the Verto
Module on FS use the same SSL.
We tested using a different a different SSL on each end and it still works.
Is this a misconception I have regarding the SSL setup?
Thanks,
V
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From mike at jerris.com  Thu May 21 18:26:53 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 May 2015 10:26:53 -0400
Subject: [Freeswitch-users] Update CLI on Freeswitch originated call
In-Reply-To: <CADcg8iVNCwUmhLBsBOVYe=r1tXwwPtwjbtxB7OyHRrKVjqOVzg@mail.gmail.com>
References: <CADcg8iUL1EsMdedCm6793O83E+pK=UBPFBFqkgbVbJtjvdCmZw@mail.gmail.com>
	<CAKzWOxUut2N0-o+SKcqKkEpFp6g_9UyFHrnr7y2FobutycNRhw@mail.gmail.com>
	<CADcg8iUT=TybcX+9_X1+3kuHQde+Edj+2u+0=XcKf2Fccu5+sg@mail.gmail.com>
	<CADcg8iWyxxZjF1sesYOxT8di2L1N16=cODjjysAqNci=354+YQ@mail.gmail.com>
	<789BFCEB-3922-4952-BF04-4651B321CC0F@jerris.com>
	<CADcg8iVNCwUmhLBsBOVYe=r1tXwwPtwjbtxB7OyHRrKVjqOVzg@mail.gmail.com>
Message-ID: <46E4639F-C60C-4A72-8134-CE977B2DDBD2@jerris.com>

from mod_dptools:

    SWITCH_ADD_APP(app_interface, "send_display", "Send session a new display", "Send session a new display.", display_function, "<text>",                      
                   SAF_SUPPORT_NOMEDIA);                                                                                                                        


there is a similar api command in mod_commands.  I think if you have ignore_dispaly_updates it may not work, try it out to be sure.

> On May 21, 2015, at 4:05 AM, Joshua Gigg <giggsey at gmail.com> wrote:
> 
> I do have ignore_dispaly_updates=true, but that's on outbound phones, my displays were updating with the number being bridged out to.
> 
> Also, I want to trigger it manually rather than just automatically when a transfer takes place.
> 
> On 20 May 2015 at 17:26, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> this should happen by default unless you have ignore_dispaly_updates=true.
> 
> 
>> On May 20, 2015, at 12:18 PM, Joshua Gigg <giggsey at gmail.com <mailto:giggsey at gmail.com>> wrote:
>> 
>> Digging up an old thread, but I've been looking into this a bit more.
>> 
>> What I want to do is described in RFC 4916. What I want to try to use an application / api command to generate a SIP INFO/UPDATE packet that would update the Caller ID.
>> 
>> On 29 January 2015 at 20:06, Joshua Gigg <giggsey at gmail.com <mailto:giggsey at gmail.com>> wrote:
>> When a transfer completes, FreeSWITCH will send an UPDATE message to the SIP server updating the caller id.
>> 
>> Is there a way of making FreeSWITCH send this message via a dialplan/command?
>> 
>> On 29 January 2015 at 17:58, Michael Collins <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>> Could you expound upon this question a bit? What does "update the CLI" mean?
>> Thanks,
>> MC
>> 
>> On Thu, Jan 29, 2015 at 1:20 AM, Joshua Gigg <giggsey at gmail.com <mailto:giggsey at gmail.com>> wrote:
>> Hi,
>> 
>> Is it possible to update the CLI at will once a Freeswitch originated call has been answered?
>> 
>> I know it can update during a transfer, but I want to be able to control it directly myself.
>> 
> 

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From mike at jerris.com  Thu May 21 18:27:53 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 May 2015 10:27:53 -0400
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <CALXCt0r4RDeQQ2VSWuPqbm2Umoda50c0bg9NGEUYEbvGVi+ttA@mail.gmail.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
	<CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
	<CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
	<CALXCt0r4RDeQQ2VSWuPqbm2Umoda50c0bg9NGEUYEbvGVi+ttA@mail.gmail.com>
Message-ID: <BEAE7616-EEB0-4A3A-A5E2-12E39B60E1EE@jerris.com>

That being said, we can send refer, specific response to invite,info, message, notify, etc.  But they all use different methods, so without knowing what he wants to send, its not possible to answer the question.

> On May 21, 2015, at 7:01 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> 
> Naveen,
> 
> Please take note that FreeSWITCH is not intended to let the user to manipulate in arbitrary way SIP dialogs.
> 
> You can do that for specific purposes and in.specific cases, within specific boundaries.
> 
> If you are looking to interact with SIP directly and freely, you may want to look at Kamailio and OpenSIPS.
> 
> You can put one of them in front of FreeSWITCH, and you can cross command them, for example via.lua scripting (both of them and FreeSWITCH can be scripted in.lua).
> 
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> 
> On May 20, 2015 3:40 PM, "Naveen Tamanam" <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
> I am aware of uuid_hangup. Indeed my intention to know the way to send (low level)sip messages 
> from fs console for a selected user. 
> 
> 
> On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
>  I would like to reject the call when it ringing from the fs console. 
> 
> uuid_hangup
>  
> 
> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
> I am trying to do the following,  I would like to reject the call when it ringing from the fs console. 
> And second thing is I am pretty much eager to know the way to send sip(signaling) message manually for a 
> selected channel from fs console. 
> 
> 
> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> respond is one way, what exact message are you trying to send, and at what point in the call.  There are capabilites to trigger re-invites in some situations, transfer, send notify or info or message.  It depends on what exactly you are trying to do.
> 
> 
>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> 
>> Hi, 
>> 
>> I am wondering how to send sip signaling message from the fs console for the particular user/caller. 
>> I found  respond dialplan application to send sip message back to the caller. 
>> Like 
>> <action application="respond" data="480 Try again later"/>
>> ?Is there any way to send sip message back to the caller. One use case is call rejection or playing with SIP?
>> 
> 
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Thanks & Regards,
> Naveen Tamanam
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Thanks & Regards,
> Naveen Tamanam
> 
> _________________________________________________________________________
> 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  Thu May 21 18:30:36 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 May 2015 10:30:36 -0400
Subject: [Freeswitch-users] Verto client SSL configuration
In-Reply-To: <CAC-LwPN-_Xv6PUuwRZUqhgqgVoN8FF6HqisY9Eqwexa4j9J07w@mail.gmail.com>
References: <CAC-LwPN-_Xv6PUuwRZUqhgqgVoN8FF6HqisY9Eqwexa4j9J07w@mail.gmail.com>
Message-ID: <CD205D34-86E0-40C0-9D6A-6DE04E9F7A0D@jerris.com>

The verto client is js code, it doesn't have its own ssl cert.  Do you mean for the web page and the websocket?  If so, its strongly recommended that you use the same one.  If you are actually using different hosts for the web client and the websocket, its possible, but you are more likely to run into cert issues on some certs, particularly with mobile browsers, as some certs will require user interaction that will not happen on the websocket, but will for the web content.


> On May 21, 2015, at 8:11 AM, Vik Killa <vipkilla at gmail.com> wrote:
> 
> Hello,
> The confluence page indicates that both the Verto Client and the Verto Module on FS use the same SSL.
> We tested using a different a different SSL on each end and it still works.
> Is this a misconception I have regarding the SSL setup?
> Thanks,
> V
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rels84 at ya.ru  Thu May 21 17:33:24 2015
From: rels84 at ya.ru (Evgeniy Sementsov)
Date: Thu, 21 May 2015 16:33:24 +0300
Subject: [Freeswitch-users] Cisco SPA not responding problem with SIP TLS
Message-ID: <6584951432215204@web23o.yandex.ru>

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From olegstolyar at gmail.com  Thu May 21 18:44:07 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Thu, 21 May 2015 07:44:07 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAEwTsAh9Rawxob9P-3tCndMfayHOuq2fW8H8hKgYW5s03k2jwA@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAC8JMOK7jD5sWTGn+KecrbyeOFpqPpydTFKHzKFupUPcDEJbDQ@mail.gmail.com>
	<CAEwTsAh9Rawxob9P-3tCndMfayHOuq2fW8H8hKgYW5s03k2jwA@mail.gmail.com>
Message-ID: <CAC8JMO+Z6z8AYqfMahxV60Nip5k2dtq=KqN7Ek6ujfL8yZthJw@mail.gmail.com>

The suggestion about switching to SIP.JS?  We are considering this but it
will take a us a while to get it done and ready for production.  In the
meantime I was able to enable debugging on JsSip and it seems that we get
this error when receiving re-INVITE from FS:
rtcninja:ERROR:RTCPeerConnection setLocalDescription() | error: +0ms Failed
to set local answer sdp: Session error code: ERROR_CONTENT. Session error
description: Failed to set audio receive codecs..

It seems that FS is not including codecs info in re-INVITES.  JsSip
developers are saying that all invites including re-invites should include
all the info.  What do you think?



On Wed, May 20, 2015 at 8:19 PM, Michael Jerris <mike at jerris.com> wrote:

> See my previous suggestion earlier in this thread
>
>
> On Wednesday, May 20, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>> I looked at my logs more carefully and noticed that FS sends such
>> re-INVITES for many of my calls but not all.  It is always sent about one
>> minute after the call is established, so it's probably not based on session
>> timer, right?
>>
>> In most cases when it's sent JsSip answers with a 100 Trying.  In this
>> case the call stays on.  In some rare cases JsSip does not respond and FS
>> disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log
>> above.
>>
>> I will investigate why JsSip sometimes does not respond but I would still
>> like to figure out why FS is sending those re-INVITES to begin with?
>>
>> I thought it could be happening if JsSip does not respond to OK with an
>> ACK but that's not the case.  The ACK is there for these calls.
>>
>> There is a variable nonce-ttl in the sip profile that defaults to 60
>> seconds.  Perhaps it has something to do with that?  This profile does not
>> require authentication or registration at all, though.
>>
>> On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar at gmail.com>
>> wrote:
>>
>>> But isn't that based on the session-timeout param which defaults to 30
>>> min?  My re-invites occur much sooner than 30 min into a call.  Or does
>>> session-timeout param only control sessions initiated by FS while incoming
>>> sessions use the minimum-session-expires param if it's not explicitly
>>> passed by the session initiator?
>>>
>>> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>>
>>>> session timer
>>>>
>>>>
>>>> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>>>
>>>>> Thanks Michael, I'll see if we can do that!
>>>>>
>>>>> So, is the re-INVITE legit and the problem is that JsSip does not
>>>>> respond to it?  Still, I am curious what is triggering the re-INVITE.
>>>>>
>>>>> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>>> I think the sip.js guys fixed this issue when they forked jssip.  I'd
>>>>>> suggest using that.
>>>>>>
>>>>>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>>>>> wrote:
>>>>>> >
>>>>>> > Hi guys,
>>>>>> >
>>>>>> > Several weeks ago I started getting an occasional problem where FS
>>>>>> is sending an INVITE to the other side in the middle of a call, the other
>>>>>> side does not respond and FS hangs up the leg.  Below is the relevant log.
>>>>>> The user experience is that they keep talking and hearing each other up to
>>>>>> the very end.  I have a recording of that call, so can confirm.
>>>>>> >
>>>>>> > The call uses WebRTC and is originated by JsSip from Chrome.  Then
>>>>>> the user is put into a conference but I doubt it's relevant in this case
>>>>>> since the INVITE and disconnect are not happening from mod_conference
>>>>>> >
>>>>>> > I suspect it's a re-INVITE but what triggers FS to send it?  I
>>>>>> couldn't find anything in the logs that could shed light.
>>>>>> >
>>>>>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>>>>>> >
>>>>>> ------------------------------------------------------------------------
>>>>>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>>>>>> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>>>>>> >    Max-Forwards: 70
>>>>>> >    From:
>>>>>> <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>>>>>> >;tag=KQecUSr12rSQp
>>>>>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>>>>>> >    Call-ID: g8980rbrbk2t45oj5mru
>>>>>> >    CSeq: 75703945 INVITE
>>>>>> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@
>>>>>> ##.##.###.###:5080;transport=udp>
>>>>>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>>>>>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>> >    Supported: timer, path, replaces
>>>>>> >    Session-Expires: 120;refresher=uac
>>>>>> >    Min-SE: 120
>>>>>> >    Content-Type: application/sdp
>>>>>> >    Content-Disposition: session
>>>>>> >    Content-Length: 825
>>>>>> >
>>>>>> >    v=0
>>>>>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>>>>>> >    s=FreeSWITCH
>>>>>> >    c=IN IP4 ##.##.##.##
>>>>>> >    t=0 0
>>>>>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>>> >    m=audio 22670 RTP/SAVPF 9 126 106
>>>>>> >    a=rtpmap:9 G722/8000
>>>>>> >    a=rtpmap:126 telephone-event/8000
>>>>>> >    a=rtpmap:106 CN/8000
>>>>>> >    a=ptime:20
>>>>>> >    a=fingerprint:sha-256
>>>>>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>>>>>> >    a=rtcp-mux
>>>>>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>>>>>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>>>>>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>>>>>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>>>>>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>>>>>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>>>>>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ
>>>>>> host generation 0
>>>>>> >
>>>>>> ------------------------------------------------------------------------
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> entering state [calling][0]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> entering state [terminating][503]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [DEBUG] switch_channel.c:3242 Send signal
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [DEBUG] switch_core_session.c:1396 Send signal
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] mod_conference.c:5057 Channel leaving conference, cause:
>>>>>> NORMAL_TEMPORARY_FAILURE
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> skip receive message [UNBRIDGE] (channel is hungup already)
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_media.c:7772
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>> [REFER_EVENT] (channel is hungup already)
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> Restore previous codec G722:9.
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_session.c:2901
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:535
>>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going
>>>>>> to sleep
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:472
>>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>>>>>> CS_HANGUP
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:735
>>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change
>>>>>> ACTIVE -> HANGUP
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:737
>>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> hanging up, cause: NORMAL_TEMPORARY_FAILURE
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:60
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause:
>>>>>> NORMAL_TEMPORARY_FAILURE
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:737
>>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to
>>>>>> sleep
>>>>>> >
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mike at jerris.com  Thu May 21 18:50:53 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 May 2015 10:50:53 -0400
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMO+Z6z8AYqfMahxV60Nip5k2dtq=KqN7Ek6ujfL8yZthJw@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAC8JMOK7jD5sWTGn+KecrbyeOFpqPpydTFKHzKFupUPcDEJbDQ@mail.gmail.com>
	<CAEwTsAh9Rawxob9P-3tCndMfayHOuq2fW8H8hKgYW5s03k2jwA@mail.gmail.com>
	<CAC8JMO+Z6z8AYqfMahxV60Nip5k2dtq=KqN7Ek6ujfL8yZthJw@mail.gmail.com>
Message-ID: <BD624B11-4E01-4EE6-A052-033FB16509D2@jerris.com>

I think its incredibly easy to switch to sip.js as its pretty close API to jssip, and looking at the invite you posted, it does have a full sdp.  jssip has been mostly abandoned for many months (maybe they came back, but it has serious bitrot either way), its not worth the time to mess with something we know is broken when there is an easy alternative we know works.

> On May 21, 2015, at 10:44 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> The suggestion about switching to SIP.JS?  We are considering this but it will take a us a while to get it done and ready for production.  In the meantime I was able to enable debugging on JsSip and it seems that we get this error when receiving re-INVITE from FS:
> rtcninja:ERROR:RTCPeerConnection setLocalDescription() | error: +0ms Failed to set local answer sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set audio receive codecs..
> 
> It seems that FS is not including codecs info in re-INVITES.  JsSip developers are saying that all invites including re-invites should include all the info.  What do you think?
> 
> 
> 
> On Wed, May 20, 2015 at 8:19 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> See my previous suggestion earlier in this thread
> 
> 
> On Wednesday, May 20, 2015, Oleg Stolyar <olegstolyar at gmail.com <mailto:olegstolyar at gmail.com>> wrote:
> I looked at my logs more carefully and noticed that FS sends such re-INVITES for many of my calls but not all.  It is always sent about one minute after the call is established, so it's probably not based on session timer, right?
> 
> In most cases when it's sent JsSip answers with a 100 Trying.  In this case the call stays on.  In some rare cases JsSip does not respond and FS disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log above.  
> 
> I will investigate why JsSip sometimes does not respond but I would still like to figure out why FS is sending those re-INVITES to begin with?
> 
> I thought it could be happening if JsSip does not respond to OK with an ACK but that's not the case.  The ACK is there for these calls.
> 
> There is a variable nonce-ttl in the sip profile that defaults to 60 seconds.  Perhaps it has something to do with that?  This profile does not require authentication or registration at all, though.
> 
> On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar at gmail.com <>> wrote:
> But isn't that based on the session-timeout param which defaults to 30 min?  My re-invites occur much sooner than 30 min into a call.  Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
> 
> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com <>> wrote:
> session timer
> 
> 
> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com <>> wrote:
> Thanks Michael, I'll see if we can do that!
> 
> So, is the re-INVITE legit and the problem is that JsSip does not respond to it?  Still, I am curious what is triggering the re-INVITE.
> 
> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com <>> wrote:
> I think the sip.js guys fixed this issue when they forked jssip.  I'd suggest using that.
> 
> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com <>> wrote:
> >
> > Hi guys,
> >
> > Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg.  Below is the relevant log.  The user experience is that they keep talking and hearing each other up to the very end.  I have a recording of that call, so can confirm.
> >
> > The call uses WebRTC and is originated by JsSip from Chrome.  Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
> >
> > I suspect it's a re-INVITE but what triggers FS to send it?  I couldn't find anything in the logs that could shed light.
> >
> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
> >    ------------------------------------------------------------------------
> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
> >    Max-Forwards: 70
> >    From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid>;tag=KQecUSr12rSQp
> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
> >    Call-ID: g8980rbrbk2t45oj5mru
> >    CSeq: 75703945 INVITE
> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp>
> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >    Supported: timer, path, replaces
> >    Session-Expires: 120;refresher=uac
> >    Min-SE: 120
> >    Content-Type: application/sdp
> >    Content-Disposition: session
> >    Content-Length: 825
> >
> >    v=0
> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
> >    s=FreeSWITCH
> >    c=IN IP4 ##.##.##.##
> >    t=0 0
> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
> >    m=audio 22670 RTP/SAVPF 9 126 106
> >    a=rtpmap:9 G722/8000
> >    a=rtpmap:126 telephone-event/8000
> >    a=rtpmap:106 CN/8000
> >    a=ptime:20
> >    a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
> >    a=rtcp-mux
> >    a=rtcp:22670 IN IP4 ##.##.##.##
> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
> >    ------------------------------------------------------------------------
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid entering state [calling][0]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid entering state [terminating][503]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous at anonymous.invalid [KILL]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous at anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid Restore previous codec G722:9.
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous at anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to sleep
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change CS_HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE -> HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to sleep
> >
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
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> 
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> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <>
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> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
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> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
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> 
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From olegstolyar at gmail.com  Thu May 21 18:59:20 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Thu, 21 May 2015 07:59:20 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <BD624B11-4E01-4EE6-A052-033FB16509D2@jerris.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAC8JMOK7jD5sWTGn+KecrbyeOFpqPpydTFKHzKFupUPcDEJbDQ@mail.gmail.com>
	<CAEwTsAh9Rawxob9P-3tCndMfayHOuq2fW8H8hKgYW5s03k2jwA@mail.gmail.com>
	<CAC8JMO+Z6z8AYqfMahxV60Nip5k2dtq=KqN7Ek6ujfL8yZthJw@mail.gmail.com>
	<BD624B11-4E01-4EE6-A052-033FB16509D2@jerris.com>
Message-ID: <CAC8JMOJM6LfANc=7Z2=vxBkrN-QvxNg2opq3g3mgbnxdg4cx7w@mail.gmail.com>

Thanks Michael!

There are some internal reasons why it's not trivial for us to switch to
sip.js but as I said we'll be looking into that as well.
On a side note, JsSip did come back - they are actively releasing new
versions.

Also, I have a question about best practices.  If a client originates a
call to FS, do you think it's better to have the client be the session
refresher or FS?  Any reason one way or another?

On Thu, May 21, 2015 at 7:50 AM, Michael Jerris <mike at jerris.com> wrote:

> I think its incredibly easy to switch to sip.js as its pretty close API to
> jssip, and looking at the invite you posted, it does have a full sdp.
>  jssip has been mostly abandoned for many months (maybe they came back, but
> it has serious bitrot either way), its not worth the time to mess with
> something we know is broken when there is an easy alternative we know works.
>
> On May 21, 2015, at 10:44 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
> The suggestion about switching to SIP.JS?  We are considering this but it
> will take a us a while to get it done and ready for production.  In the
> meantime I was able to enable debugging on JsSip and it seems that we get
> this error when receiving re-INVITE from FS:
> rtcninja:ERROR:RTCPeerConnection setLocalDescription() | error: +0ms
> Failed to set local answer sdp: Session error code: ERROR_CONTENT. Session
> error description: Failed to set audio receive codecs..
>
> It seems that FS is not including codecs info in re-INVITES.  JsSip
> developers are saying that all invites including re-invites should include
> all the info.  What do you think?
>
>
>
> On Wed, May 20, 2015 at 8:19 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> See my previous suggestion earlier in this thread
>>
>>
>> On Wednesday, May 20, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>
>>> I looked at my logs more carefully and noticed that FS sends such
>>> re-INVITES for many of my calls but not all.  It is always sent about one
>>> minute after the call is established, so it's probably not based on session
>>> timer, right?
>>>
>>> In most cases when it's sent JsSip answers with a 100 Trying.  In this
>>> case the call stays on.  In some rare cases JsSip does not respond and FS
>>> disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log
>>> above.
>>>
>>> I will investigate why JsSip sometimes does not respond but I would
>>> still like to figure out why FS is sending those re-INVITES to begin with?
>>>
>>> I thought it could be happening if JsSip does not respond to OK with an
>>> ACK but that's not the case.  The ACK is there for these calls.
>>>
>>> There is a variable nonce-ttl in the sip profile that defaults to 60
>>> seconds.  Perhaps it has something to do with that?  This profile does not
>>> require authentication or registration at all, though.
>>>
>>> On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar at gmail.com>
>>> wrote:
>>>
>>>> But isn't that based on the session-timeout param which defaults to 30
>>>> min?  My re-invites occur much sooner than 30 min into a call.  Or does
>>>> session-timeout param only control sessions initiated by FS while incoming
>>>> sessions use the minimum-session-expires param if it's not explicitly
>>>> passed by the session initiator?
>>>>
>>>> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>>
>>>>> session timer
>>>>>
>>>>>
>>>>> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>>>>
>>>>>> Thanks Michael, I'll see if we can do that!
>>>>>>
>>>>>> So, is the re-INVITE legit and the problem is that JsSip does not
>>>>>> respond to it?  Still, I am curious what is triggering the re-INVITE.
>>>>>>
>>>>>> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com>
>>>>>> wrote:
>>>>>>
>>>>>>> I think the sip.js guys fixed this issue when they forked jssip.
>>>>>>> I'd suggest using that.
>>>>>>>
>>>>>>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>>>>>> wrote:
>>>>>>> >
>>>>>>> > Hi guys,
>>>>>>> >
>>>>>>> > Several weeks ago I started getting an occasional problem where FS
>>>>>>> is sending an INVITE to the other side in the middle of a call, the other
>>>>>>> side does not respond and FS hangs up the leg.  Below is the relevant log.
>>>>>>> The user experience is that they keep talking and hearing each other up to
>>>>>>> the very end.  I have a recording of that call, so can confirm.
>>>>>>> >
>>>>>>> > The call uses WebRTC and is originated by JsSip from Chrome.  Then
>>>>>>> the user is put into a conference but I doubt it's relevant in this case
>>>>>>> since the INVITE and disconnect are not happening from mod_conference
>>>>>>> >
>>>>>>> > I suspect it's a re-INVITE but what triggers FS to send it?  I
>>>>>>> couldn't find anything in the logs that could shed light.
>>>>>>> >
>>>>>>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>>>>>>> >
>>>>>>> ------------------------------------------------------------------------
>>>>>>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob
>>>>>>> SIP/2.0
>>>>>>> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>>>>>>> >    Max-Forwards: 70
>>>>>>> >    From: <
>>>>>>> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>>>>>>> >;tag=KQecUSr12rSQp
>>>>>>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>>>>>>> >    Call-ID: g8980rbrbk2t45oj5mru
>>>>>>> >    CSeq: 75703945 INVITE
>>>>>>> >    Contact: <
>>>>>>> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp
>>>>>>> >
>>>>>>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>>>>>>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
>>>>>>> UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>>> >    Supported: timer, path, replaces
>>>>>>> >    Session-Expires: 120;refresher=uac
>>>>>>> >    Min-SE: 120
>>>>>>> >    Content-Type: application/sdp
>>>>>>> >    Content-Disposition: session
>>>>>>> >    Content-Length: 825
>>>>>>> >
>>>>>>> >    v=0
>>>>>>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>>>>>>> >    s=FreeSWITCH
>>>>>>> >    c=IN IP4 ##.##.##.##
>>>>>>> >    t=0 0
>>>>>>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>>>> >    m=audio 22670 RTP/SAVPF 9 126 106
>>>>>>> >    a=rtpmap:9 G722/8000
>>>>>>> >    a=rtpmap:126 telephone-event/8000
>>>>>>> >    a=rtpmap:106 CN/8000
>>>>>>> >    a=ptime:20
>>>>>>> >    a=fingerprint:sha-256
>>>>>>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>>>>>>> >    a=rtcp-mux
>>>>>>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>>>>>>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>>>>>>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>>>>>>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>>>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>>>>>>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>>>>>>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>>>>>>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ
>>>>>>> host generation 0
>>>>>>> >
>>>>>>> ------------------------------------------------------------------------
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>>>> [DEBUG] sofia.c:6627 Channel
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid entering state
>>>>>>> [calling][0]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [DEBUG] sofia.c:6627 Channel
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid entering state
>>>>>>> [terminating][503]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [NOTICE] sofia.c:7543 Hangup
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [CS_EXECUTE]
>>>>>>> [NORMAL_TEMPORARY_FAILURE]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [DEBUG] switch_channel.c:3242 Send signal
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [DEBUG] switch_core_session.c:1396 Send signal
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] mod_conference.c:5057 Channel leaving conference, cause:
>>>>>>> NORMAL_TEMPORARY_FAILURE
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] mod_conference.c:9650
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>>> [UNBRIDGE] (channel is hungup already)
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_media.c:7772
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>>> [REFER_EVENT] (channel is hungup already)
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_codec.c:246
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid Restore previous codec
>>>>>>> G722:9.
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_session.c:2901
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:535 (
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going
>>>>>>> to sleep
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:472 (
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>>>>>>> CS_HANGUP
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:735 (
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change
>>>>>>> ACTIVE -> HANGUP
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:737 (
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] mod_sofia.c:413 Channel
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid hanging up, cause:
>>>>>>> NORMAL_TEMPORARY_FAILURE
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:60
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP,
>>>>>>> cause: NORMAL_TEMPORARY_FAILURE
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:737 (
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to
>>>>>>> sleep
>>>>>>> >
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dm at dwide.com  Thu May 21 20:07:20 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Thu, 21 May 2015 20:07:20 +0400
Subject: [Freeswitch-users] mod_event_socket - run LUA application
Message-ID: <555E02B8.6060607@dwide.com>

Hello,

Is it possible to use mod_event_socket to run lua application with some 
parameters?

originate command can make a call to URL and bridge it to application.

In my case - only run lua app without voice calls

Thenks a lot!


From mitchelle.bit at gmail.com  Thu May 21 20:10:22 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Thu, 21 May 2015 21:40:22 +0530
Subject: [Freeswitch-users] Google Hangouts Integration
Message-ID: <CANBm6Sw0FNwo1Gw5sS-AmwcjBfM5x3S9-YtZ4n4GBvCgbL3paA@mail.gmail.com>

Hi all,

Like skype can be integrated with FreeSWITCH. Is there a way to integrate
google Hangouts also?

Thanks,
Mitchelle
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From anthony.minessale at gmail.com  Thu May 21 20:15:09 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 May 2015 11:15:09 -0500
Subject: [Freeswitch-users] Google Hangouts Integration
In-Reply-To: <CANBm6Sw0FNwo1Gw5sS-AmwcjBfM5x3S9-YtZ4n4GBvCgbL3paA@mail.gmail.com>
References: <CANBm6Sw0FNwo1Gw5sS-AmwcjBfM5x3S9-YtZ4n4GBvCgbL3paA@mail.gmail.com>
Message-ID: <CAKbxfG9zKVVso0=Xp9=rdVWaxm_sXQ2zTV0gOWNaXLWaGsGA3g@mail.gmail.com>

If google published the specifications on the signalling they use, then
yes.  So far they have not as far as I know.


On Thu, May 21, 2015 at 11:10 AM, Mitchelle Johnson <mitchelle.bit at gmail.com
> wrote:

> Hi all,
>
> Like skype can be integrated with FreeSWITCH. Is there a way to integrate
> google Hangouts also?
>
> Thanks,
> Mitchelle
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 emplant2000 at gmail.com  Thu May 21 20:21:33 2015
From: emplant2000 at gmail.com (Masakazu Nakano)
Date: Fri, 22 May 2015 01:21:33 +0900
Subject: [Freeswitch-users] Google Hangouts Integration
In-Reply-To: <CAKbxfG9zKVVso0=Xp9=rdVWaxm_sXQ2zTV0gOWNaXLWaGsGA3g@mail.gmail.com>
References: <CANBm6Sw0FNwo1Gw5sS-AmwcjBfM5x3S9-YtZ4n4GBvCgbL3paA@mail.gmail.com>
	<CAKbxfG9zKVVso0=Xp9=rdVWaxm_sXQ2zTV0gOWNaXLWaGsGA3g@mail.gmail.com>
Message-ID: <CAPwL2MAdj-YmQCfgBcn0e20yBh9vZN-Br-JDaaWdq11Bo=AKzw@mail.gmail.com>

maybe in proprietary,I think so too.

BR

mack

2015-05-22 1:15 GMT+09:00 Anthony Minessale <anthony.minessale at gmail.com>:

> If google published the specifications on the signalling they use, then
> yes.  So far they have not as far as I know.
>
>
> On Thu, May 21, 2015 at 11:10 AM, Mitchelle Johnson <
> mitchelle.bit at gmail.com> wrote:
>
>> Hi all,
>>
>> Like skype can be integrated with FreeSWITCH. Is there a way to integrate
>> google Hangouts also?
>>
>> Thanks,
>> Mitchelle
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Thu May 21 20:30:09 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 May 2015 12:30:09 -0400
Subject: [Freeswitch-users] mod_event_socket - run LUA application
In-Reply-To: <555E02B8.6060607@dwide.com>
References: <555E02B8.6060607@dwide.com>
Message-ID: <071B5375-830A-49D4-8BC0-BB21094A43E8@jerris.com>

you can run lua via the api interface from event socket.

> On May 21, 2015, at 12:07 PM, Dmitry Mordovin <dm at dwide.com> wrote:
> 
> Hello,
> 
> Is it possible to use mod_event_socket to run lua application with some 
> parameters?
> 
> originate command can make a call to URL and bridge it to application.
> 
> In my case - only run lua app without voice calls
> 
> Thenks a lot!



From vipkilla at gmail.com  Thu May 21 21:17:14 2015
From: vipkilla at gmail.com (Vik Killa)
Date: Thu, 21 May 2015 13:17:14 -0400
Subject: [Freeswitch-users] Verto client SSL configuration
In-Reply-To: <CD205D34-86E0-40C0-9D6A-6DE04E9F7A0D@jerris.com>
References: <CAC-LwPN-_Xv6PUuwRZUqhgqgVoN8FF6HqisY9Eqwexa4j9J07w@mail.gmail.com>
	<CD205D34-86E0-40C0-9D6A-6DE04E9F7A0D@jerris.com>
Message-ID: <CAC-LwPNNB1kSUGH+gRoSXx1O6BtE9RNCDsLjO8eVBMAPNo11=w@mail.gmail.com>

To clarify my questions, yes, I am talking about a separate web-host using
an SSL which will connect to the websocket (FS's verto) which has a
different SSL.

I'm confused by your last comment:
...some certs will require user interaction that will not happen on the
websocket, but will for the web content....

On Thu, May 21, 2015 at 10:30 AM, Michael Jerris <mike at jerris.com> wrote:

> The verto client is js code, it doesn't have its own ssl cert.  Do you
> mean for the web page and the websocket?  If so, its strongly recommended
> that you use the same one.  If you are actually using different hosts for
> the web client and the websocket, its possible, but you are more likely to
> run into cert issues on some certs, particularly with mobile browsers, as
> some certs will require user interaction that will not happen on the
> websocket, but will for the web content.
>
>
> > On May 21, 2015, at 8:11 AM, Vik Killa <vipkilla at gmail.com> wrote:
> >
> > Hello,
> > The confluence page indicates that both the Verto Client and the Verto
> Module on FS use the same SSL.
> > We tested using a different a different SSL on each end and it still
> works.
> > Is this a misconception I have regarding the SSL setup?
> > Thanks,
> > V
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From cdgraff at gmail.com  Thu May 21 21:23:48 2015
From: cdgraff at gmail.com (Alejandro)
Date: Thu, 21 May 2015 14:23:48 -0300
Subject: [Freeswitch-users] Google Hangouts Integration
In-Reply-To: <CAPwL2MAdj-YmQCfgBcn0e20yBh9vZN-Br-JDaaWdq11Bo=AKzw@mail.gmail.com>
References: <CANBm6Sw0FNwo1Gw5sS-AmwcjBfM5x3S9-YtZ4n4GBvCgbL3paA@mail.gmail.com>
	<CAKbxfG9zKVVso0=Xp9=rdVWaxm_sXQ2zTV0gOWNaXLWaGsGA3g@mail.gmail.com>
	<CAPwL2MAdj-YmQCfgBcn0e20yBh9vZN-Br-JDaaWdq11Bo=AKzw@mail.gmail.com>
Message-ID: <CALJtzxCAK-5RgoRw-Swc6p90rLJLjife8umjP6D42G64aU8gcw@mail.gmail.com>

Is Skype integration legal? I read something about that now is not more
legal connect Skype... some advice? if possible can share some link i'm
interested in do the integration too.

2015-05-21 13:21 GMT-03:00 Masakazu Nakano <emplant2000 at gmail.com>:

> maybe in proprietary,I think so too.
>
> BR
>
> mack
>
> 2015-05-22 1:15 GMT+09:00 Anthony Minessale <anthony.minessale at gmail.com>:
>
>> If google published the specifications on the signalling they use, then
>> yes.  So far they have not as far as I know.
>>
>>
>> On Thu, May 21, 2015 at 11:10 AM, Mitchelle Johnson <
>> mitchelle.bit at gmail.com> wrote:
>>
>>> Hi all,
>>>
>>> Like skype can be integrated with FreeSWITCH. Is there a way to
>>> integrate google Hangouts also?
>>>
>>> Thanks,
>>> Mitchelle
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +19193869900
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Thu May 21 21:41:51 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 21 May 2015 12:41:51 -0500
Subject: [Freeswitch-users] Google Hangouts Integration
In-Reply-To: <CALJtzxCAK-5RgoRw-Swc6p90rLJLjife8umjP6D42G64aU8gcw@mail.gmail.com>
References: <CANBm6Sw0FNwo1Gw5sS-AmwcjBfM5x3S9-YtZ4n4GBvCgbL3paA@mail.gmail.com>
	<CAKbxfG9zKVVso0=Xp9=rdVWaxm_sXQ2zTV0gOWNaXLWaGsGA3g@mail.gmail.com>
	<CAPwL2MAdj-YmQCfgBcn0e20yBh9vZN-Br-JDaaWdq11Bo=AKzw@mail.gmail.com>
	<CALJtzxCAK-5RgoRw-Swc6p90rLJLjife8umjP6D42G64aU8gcw@mail.gmail.com>
Message-ID: <CAKbxfG9no1oiQ0inoMAZ9U6zp0kxp9Ev2PBuSt-oogwrEqxChg@mail.gmail.com>

Its mostly not legal per-se.  They developed an API then they don't let
anybody use it for anything at all interesting.  This was before Microsoft
took it over and since then its all but extinguished.
We still have mod_skypeopen which uses an older version of the actual linux
X skype client and a bridge device to simulate channels.  Its technically
allowed because its just running the client and not using the protocol
directly.
That said, the scalability suffers from the all the overhead of running X
clients in pretend x roots and using fake audio devices etc.



On Thu, May 21, 2015 at 12:23 PM, Alejandro <cdgraff at gmail.com> wrote:

> Is Skype integration legal? I read something about that now is not more
> legal connect Skype... some advice? if possible can share some link i'm
> interested in do the integration too.
>
> 2015-05-21 13:21 GMT-03:00 Masakazu Nakano <emplant2000 at gmail.com>:
>
> maybe in proprietary,I think so too.
>>
>> BR
>>
>> mack
>>
>> 2015-05-22 1:15 GMT+09:00 Anthony Minessale <anthony.minessale at gmail.com>
>> :
>>
>>> If google published the specifications on the signalling they use, then
>>> yes.  So far they have not as far as I know.
>>>
>>>
>>> On Thu, May 21, 2015 at 11:10 AM, Mitchelle Johnson <
>>> mitchelle.bit at gmail.com> wrote:
>>>
>>>> Hi all,
>>>>
>>>> Like skype can be integrated with FreeSWITCH. Is there a way to
>>>> integrate google Hangouts also?
>>>>
>>>> Thanks,
>>>> Mitchelle
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +19193869900
>>>
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 emplant2000 at gmail.com  Thu May 21 21:47:25 2015
From: emplant2000 at gmail.com (Masakazu Nakano)
Date: Fri, 22 May 2015 02:47:25 +0900
Subject: [Freeswitch-users] Google Hangouts Integration
In-Reply-To: <CALJtzxCAK-5RgoRw-Swc6p90rLJLjife8umjP6D42G64aU8gcw@mail.gmail.com>
References: <CANBm6Sw0FNwo1Gw5sS-AmwcjBfM5x3S9-YtZ4n4GBvCgbL3paA@mail.gmail.com>
	<CAKbxfG9zKVVso0=Xp9=rdVWaxm_sXQ2zTV0gOWNaXLWaGsGA3g@mail.gmail.com>
	<CAPwL2MAdj-YmQCfgBcn0e20yBh9vZN-Br-JDaaWdq11Bo=AKzw@mail.gmail.com>
	<CALJtzxCAK-5RgoRw-Swc6p90rLJLjife8umjP6D42G64aU8gcw@mail.gmail.com>
Message-ID: <CAPwL2MDLVHYTE4NO3OO1kYuuYzCVzwa6tKF1fv2oaao5wys_5g@mail.gmail.com>

google to be gets os level proprietary communicating such as NSA situation.

I dont know they are disclose or not.

when you select 8.8.8.8, it's meaning your browsers incomming streaming is
controlled by google.

and a couple of cloud cache is working today,maybe you know.

BR

mack


2015-05-22 2:23 GMT+09:00 Alejandro <cdgraff at gmail.com>:

> Is Skype integration legal? I read something about that now is not more
> legal connect Skype... some advice? if possible can share some link i'm
> interested in do the integration too.
>
> 2015-05-21 13:21 GMT-03:00 Masakazu Nakano <emplant2000 at gmail.com>:
>
>> maybe in proprietary,I think so too.
>>
>> BR
>>
>> mack
>>
>> 2015-05-22 1:15 GMT+09:00 Anthony Minessale <anthony.minessale at gmail.com>
>> :
>>
>>> If google published the specifications on the signalling they use, then
>>> yes.  So far they have not as far as I know.
>>>
>>>
>>> On Thu, May 21, 2015 at 11:10 AM, Mitchelle Johnson <
>>> mitchelle.bit at gmail.com> wrote:
>>>
>>>> Hi all,
>>>>
>>>> Like skype can be integrated with FreeSWITCH. Is there a way to
>>>> integrate google Hangouts also?
>>>>
>>>> Thanks,
>>>> Mitchelle
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +19193869900
>>>
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From olegstolyar at gmail.com  Thu May 21 23:19:11 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Thu, 21 May 2015 12:19:11 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMOJM6LfANc=7Z2=vxBkrN-QvxNg2opq3g3mgbnxdg4cx7w@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAC8JMOK7jD5sWTGn+KecrbyeOFpqPpydTFKHzKFupUPcDEJbDQ@mail.gmail.com>
	<CAEwTsAh9Rawxob9P-3tCndMfayHOuq2fW8H8hKgYW5s03k2jwA@mail.gmail.com>
	<CAC8JMO+Z6z8AYqfMahxV60Nip5k2dtq=KqN7Ek6ujfL8yZthJw@mail.gmail.com>
	<BD624B11-4E01-4EE6-A052-033FB16509D2@jerris.com>
	<CAC8JMOJM6LfANc=7Z2=vxBkrN-QvxNg2opq3g3mgbnxdg4cx7w@mail.gmail.com>
Message-ID: <CAC8JMOJG0k+srzeFkynpHwM5xP19fKmZtGFM5b24zTksnVsUgw@mail.gmail.com>

Hi Michael,

I followed your suggestion and tested the same call using sip.js.  You are
right sip.js does not disconnect the call after receiving the re-invite.
However, it does send back the message below.  I agree that it's a more
graceful way to handle the problem but the original problem on both JsSip
and SIP.JS is still that there is something wrong with the re-INVITE
message from FS.
By the way a side effect of this is that FS does not send any more
re-invites for this session after that first attempt.

I'll be happy to run any other tests that could help fix this.  For now I
will either switch to sip.js or use JsSip's new session timer capability
which makes JsSip send regular update messages, so FS does not need to send
re-invites at all.


   ------------------------------------------------------------------------
recv 379 bytes from wss/[69.53.236.236]:17084 at 19:11:35.832686:
   ------------------------------------------------------------------------
   SIP/2.0 488 Not Acceptable Here
   Via: SIP/2.0/WSS ##.##.##.##:5067;branch=z9hG4bK9rtFZNj1vK09D
   To: <sip:oleg1234 at something.com>;tag=c6tjtcraq7
   From: <sip:echo-test at something.com>;tag=c29e7F87NvQNp
   Call-ID: 1v8ll5j5dcqj1d2fimo1
   CSeq: 75781683 INVITE
   Warning: 399 sipjs "Cannot update media description"
   Supported: outbound
   User-Agent: SIP.js/0.7.0
   Content-Length: 0


On Thu, May 21, 2015 at 7:59 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Thanks Michael!
>
> There are some internal reasons why it's not trivial for us to switch to
> sip.js but as I said we'll be looking into that as well.
> On a side note, JsSip did come back - they are actively releasing new
> versions.
>
> Also, I have a question about best practices.  If a client originates a
> call to FS, do you think it's better to have the client be the session
> refresher or FS?  Any reason one way or another?
>
> On Thu, May 21, 2015 at 7:50 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> I think its incredibly easy to switch to sip.js as its pretty close API
>> to jssip, and looking at the invite you posted, it does have a full sdp.
>>  jssip has been mostly abandoned for many months (maybe they came back, but
>> it has serious bitrot either way), its not worth the time to mess with
>> something we know is broken when there is an easy alternative we know works.
>>
>> On May 21, 2015, at 10:44 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>
>> The suggestion about switching to SIP.JS?  We are considering this but it
>> will take a us a while to get it done and ready for production.  In the
>> meantime I was able to enable debugging on JsSip and it seems that we get
>> this error when receiving re-INVITE from FS:
>> rtcninja:ERROR:RTCPeerConnection setLocalDescription() | error: +0ms
>> Failed to set local answer sdp: Session error code: ERROR_CONTENT. Session
>> error description: Failed to set audio receive codecs..
>>
>> It seems that FS is not including codecs info in re-INVITES.  JsSip
>> developers are saying that all invites including re-invites should include
>> all the info.  What do you think?
>>
>>
>>
>> On Wed, May 20, 2015 at 8:19 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> See my previous suggestion earlier in this thread
>>>
>>>
>>> On Wednesday, May 20, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>>
>>>> I looked at my logs more carefully and noticed that FS sends such
>>>> re-INVITES for many of my calls but not all.  It is always sent about one
>>>> minute after the call is established, so it's probably not based on session
>>>> timer, right?
>>>>
>>>> In most cases when it's sent JsSip answers with a 100 Trying.  In this
>>>> case the call stays on.  In some rare cases JsSip does not respond and FS
>>>> disconnects the call with NORMAL_TEMPORARY_FAILURE we see in the log
>>>> above.
>>>>
>>>> I will investigate why JsSip sometimes does not respond but I would
>>>> still like to figure out why FS is sending those re-INVITES to begin with?
>>>>
>>>> I thought it could be happening if JsSip does not respond to OK with an
>>>> ACK but that's not the case.  The ACK is there for these calls.
>>>>
>>>> There is a variable nonce-ttl in the sip profile that defaults to 60
>>>> seconds.  Perhaps it has something to do with that?  This profile does not
>>>> require authentication or registration at all, though.
>>>>
>>>> On Wed, May 20, 2015 at 7:40 AM, Oleg Stolyar <olegstolyar at gmail.com>
>>>> wrote:
>>>>
>>>>> But isn't that based on the session-timeout param which defaults to 30
>>>>> min?  My re-invites occur much sooner than 30 min into a call.  Or does
>>>>> session-timeout param only control sessions initiated by FS while incoming
>>>>> sessions use the minimum-session-expires param if it's not explicitly
>>>>> passed by the session initiator?
>>>>>
>>>>> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>>> session timer
>>>>>>
>>>>>>
>>>>>> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>>>>>
>>>>>>> Thanks Michael, I'll see if we can do that!
>>>>>>>
>>>>>>> So, is the re-INVITE legit and the problem is that JsSip does not
>>>>>>> respond to it?  Still, I am curious what is triggering the re-INVITE.
>>>>>>>
>>>>>>> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> I think the sip.js guys fixed this issue when they forked jssip.
>>>>>>>> I'd suggest using that.
>>>>>>>>
>>>>>>>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>>>>>>> wrote:
>>>>>>>> >
>>>>>>>> > Hi guys,
>>>>>>>> >
>>>>>>>> > Several weeks ago I started getting an occasional problem where
>>>>>>>> FS is sending an INVITE to the other side in the middle of a call, the
>>>>>>>> other side does not respond and FS hangs up the leg.  Below is the relevant
>>>>>>>> log.  The user experience is that they keep talking and hearing each other
>>>>>>>> up to the very end.  I have a recording of that call, so can confirm.
>>>>>>>> >
>>>>>>>> > The call uses WebRTC and is originated by JsSip from Chrome.
>>>>>>>> Then the user is put into a conference but I doubt it's relevant in this
>>>>>>>> case since the INVITE and disconnect are not happening from mod_conference
>>>>>>>> >
>>>>>>>> > I suspect it's a re-INVITE but what triggers FS to send it?  I
>>>>>>>> couldn't find anything in the logs that could shed light.
>>>>>>>> >
>>>>>>>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>>>>>>>> >
>>>>>>>> ------------------------------------------------------------------------
>>>>>>>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob
>>>>>>>> SIP/2.0
>>>>>>>> >    Via: SIP/2.0/WSS 10.97.158.232:5067
>>>>>>>> ;branch=z9hG4bK7Xm4tjevU45Sr
>>>>>>>> >    Max-Forwards: 70
>>>>>>>> >    From: <
>>>>>>>> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>>>>>>>> >;tag=KQecUSr12rSQp
>>>>>>>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>>>>>>>> >    Call-ID: g8980rbrbk2t45oj5mru
>>>>>>>> >    CSeq: 75703945 INVITE
>>>>>>>> >    Contact: <
>>>>>>>> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp
>>>>>>>> >
>>>>>>>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>>>>>>>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
>>>>>>>> UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>>>> >    Supported: timer, path, replaces
>>>>>>>> >    Session-Expires: 120;refresher=uac
>>>>>>>> >    Min-SE: 120
>>>>>>>> >    Content-Type: application/sdp
>>>>>>>> >    Content-Disposition: session
>>>>>>>> >    Content-Length: 825
>>>>>>>> >
>>>>>>>> >    v=0
>>>>>>>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>>>>>>>> >    s=FreeSWITCH
>>>>>>>> >    c=IN IP4 ##.##.##.##
>>>>>>>> >    t=0 0
>>>>>>>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>>>>> >    m=audio 22670 RTP/SAVPF 9 126 106
>>>>>>>> >    a=rtpmap:9 G722/8000
>>>>>>>> >    a=rtpmap:126 telephone-event/8000
>>>>>>>> >    a=rtpmap:106 CN/8000
>>>>>>>> >    a=ptime:20
>>>>>>>> >    a=fingerprint:sha-256
>>>>>>>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>>>>>>>> >    a=rtcp-mux
>>>>>>>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>>>>>>>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>>>>>>>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>>>>>>>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>>>>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>>>>>>>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>>>>>>>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>>>>>>>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ
>>>>>>>> host generation 0
>>>>>>>> >
>>>>>>>> ------------------------------------------------------------------------
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>>>>> [DEBUG] sofia.c:6627 Channel
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid entering state
>>>>>>>> [calling][0]
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>>> [DEBUG] sofia.c:6627 Channel
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid entering state
>>>>>>>> [terminating][503]
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>>> [NOTICE] sofia.c:7543 Hangup
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [CS_EXECUTE]
>>>>>>>> [NORMAL_TEMPORARY_FAILURE]
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>>> [DEBUG] switch_channel.c:3242 Send signal
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>>> [DEBUG] switch_core_session.c:1396 Send signal
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] mod_conference.c:5057 Channel leaving conference, cause:
>>>>>>>> NORMAL_TEMPORARY_FAILURE
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] mod_conference.c:9650
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>>>> [UNBRIDGE] (channel is hungup already)
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] switch_core_media.c:7772
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>>>> [REFER_EVENT] (channel is hungup already)
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] switch_core_codec.c:246
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid Restore previous
>>>>>>>> codec G722:9.
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] switch_core_session.c:2901
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] switch_core_state_machine.c:535 (
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going
>>>>>>>> to sleep
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] switch_core_state_machine.c:472 (
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>>>>>>>> CS_HANGUP
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] switch_core_state_machine.c:735 (
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change
>>>>>>>> ACTIVE -> HANGUP
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] switch_core_state_machine.c:737 (
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] mod_sofia.c:413 Channel
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid hanging up, cause:
>>>>>>>> NORMAL_TEMPORARY_FAILURE
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] switch_core_state_machine.c:60
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP,
>>>>>>>> cause: NORMAL_TEMPORARY_FAILURE
>>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>>> [DEBUG] switch_core_state_machine.c:737 (
>>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going
>>>>>>>> to sleep
>>>>>>>> >
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From emplant2000 at gmail.com  Thu May 21 23:27:26 2015
From: emplant2000 at gmail.com (Masakazu Nakano)
Date: Fri, 22 May 2015 04:27:26 +0900
Subject: [Freeswitch-users] Google Hangouts Integration
In-Reply-To: <CAKbxfG9no1oiQ0inoMAZ9U6zp0kxp9Ev2PBuSt-oogwrEqxChg@mail.gmail.com>
References: <CANBm6Sw0FNwo1Gw5sS-AmwcjBfM5x3S9-YtZ4n4GBvCgbL3paA@mail.gmail.com>
	<CAKbxfG9zKVVso0=Xp9=rdVWaxm_sXQ2zTV0gOWNaXLWaGsGA3g@mail.gmail.com>
	<CAPwL2MAdj-YmQCfgBcn0e20yBh9vZN-Br-JDaaWdq11Bo=AKzw@mail.gmail.com>
	<CALJtzxCAK-5RgoRw-Swc6p90rLJLjife8umjP6D42G64aU8gcw@mail.gmail.com>
	<CAKbxfG9no1oiQ0inoMAZ9U6zp0kxp9Ev2PBuSt-oogwrEqxChg@mail.gmail.com>
Message-ID: <CAPwL2MCKzhNHhJTNidmP5JJBymmE7JB5DvC=a5tsNGaH2SDmFA@mail.gmail.com>

ya,it can patching each one by jackd on linux maybe.

old skype protocol was disclosed by in or outer engineer as hacker by an
pdf,my remember.it's quite interesting topic around teleco peoples at that
time.interesting high-end audio codec and P2P.

perphaps,latest HTML5 has similar P2P function.
but that is not disclosed,i think.and capsuled in crypted in each browser.

BR

mack


2015-05-22 2:41 GMT+09:00 Anthony Minessale <anthony.minessale at gmail.com>:

> Its mostly not legal per-se.  They developed an API then they don't let
> anybody use it for anything at all interesting.  This was before Microsoft
> took it over and since then its all but extinguished.
> We still have mod_skypeopen which uses an older version of the actual
> linux X skype client and a bridge device to simulate channels.  Its
> technically allowed because its just running the client and not using the
> protocol directly.
> That said, the scalability suffers from the all the overhead of running X
> clients in pretend x roots and using fake audio devices etc.
>
>
>
> On Thu, May 21, 2015 at 12:23 PM, Alejandro <cdgraff at gmail.com> wrote:
>
>> Is Skype integration legal? I read something about that now is not more
>> legal connect Skype... some advice? if possible can share some link i'm
>> interested in do the integration too.
>>
>> 2015-05-21 13:21 GMT-03:00 Masakazu Nakano <emplant2000 at gmail.com>:
>>
>> maybe in proprietary,I think so too.
>>>
>>> BR
>>>
>>> mack
>>>
>>> 2015-05-22 1:15 GMT+09:00 Anthony Minessale <anthony.minessale at gmail.com
>>> >:
>>>
>>>> If google published the specifications on the signalling they use, then
>>>> yes.  So far they have not as far as I know.
>>>>
>>>>
>>>> On Thu, May 21, 2015 at 11:10 AM, Mitchelle Johnson <
>>>> mitchelle.bit at gmail.com> wrote:
>>>>
>>>>> Hi all,
>>>>>
>>>>> Like skype can be integrated with FreeSWITCH. Is there a way to
>>>>> integrate google Hangouts also?
>>>>>
>>>>> Thanks,
>>>>> Mitchelle
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +19193869900
>>>>
>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +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
>



-- 
--- keep it bass,keep it drum.
ubuntu server 14.04 + nginx + mysql + ISPConfig3 + dovecot + postfix +
bingbluebutton + Freeswitch + FusionPBX + moodle ( + Blender + Cinelerra +
Synfig )
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From s.safarov at gmail.com  Thu May 21 23:46:45 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 21 May 2015 22:46:45 +0300
Subject: [Freeswitch-users] Cisco SPA not responding problem with SIP TLS
In-Reply-To: <6584951432215204@web23o.yandex.ru>
References: <6584951432215204@web23o.yandex.ru>
Message-ID: <CAHtxdDexP_CJ000fsKGQTyhqTGzY7Jazrzj=BHc7R_9OK_LJYw@mail.gmail.com>

We also have similar issue. Cisco SPA 303 phone cannot restore connection
after network problems (FS restart) over UDP also.
To minimize probability of this issue we use UDP transport and update phone
firmware to 7.5.7s.

On Thu, May 21, 2015 at 4:33 PM, Evgeniy Sementsov <rels84 at ya.ru> wrote:

> We using 1.4.18 (x86_64, rpm version on CentOS 6) with TLS SIP profile
> with Cisco SPA 303/504 with firmwares 7.4.9c up to 7.5.7s (no matter).
> When phones lost connect to servers (network problems) Cisco phones after
> few seconds not responding.
> And we need to reset phones with turn off/on power.
>
> Can FreeSWITCH play role on phones that not responding after loss
> connection to FreeSWITCH or this is problem only Cisco SPA phones?
> Maybe anybody know how we can solve this problem.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mbodbg at gmx.net  Thu May 21 23:57:18 2015
From: mbodbg at gmx.net (mbo)
Date: Thu, 21 May 2015 21:57:18 +0200
Subject: [Freeswitch-users] Caller id in P-Asserted-Identity
Message-ID: <4088C1E6-932D-46F1-8675-391D0F1A1280@gmx.net>

Our carrier sends us always the callerid in the P-Asserted-Identity field in the SIP header. When a caller calls anonymous, in the INVITE the relevant fields in the header look like:

From: <sip:anonymous at anonymous.invalid;user=phone;udp>
P-Asserted-Identity: <sip:+PHONENUMBER at IPADDRESS>
Privacy: header;user;id

When I bridge the call, in the INVITE which is sent to the remote server, the relevant fields in the header look like: 

From: "anonymous" <sip:+PHONENUMBER at IP-ADDRESS>;tag=K90ap6HBNQDtQ
Privacy: id
P-Asserted-Identity: "anonymous" <sip:+PHONENUMBER at IP-ADDRESS>

But I fo not want to sent the callerid at all, so the fields should be set to:

From: <sip:anonymous at anonymous.invalid;user=phone;udp>
P-Asserted-Identity: "anonymous" <sip:+anonymous at anonymous.invalid;user=phone;udp>
Privacy: header;user;id

or drop the P-Asserted-Identity field like:

From: <sip:anonymous at anonymous.invalid;user=phone;udp>
Privacy: header;user;id

Which variables do I need to set before the call is bridged to archive this (I?m still using FS 1.2)

Thanks and Regards

Markus









From naveen32india at gmail.com  Fri May 22 00:14:51 2015
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Fri, 22 May 2015 01:44:51 +0530
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <CALXCt0r4RDeQQ2VSWuPqbm2Umoda50c0bg9NGEUYEbvGVi+ttA@mail.gmail.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
	<CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
	<CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
	<CALXCt0r4RDeQQ2VSWuPqbm2Umoda50c0bg9NGEUYEbvGVi+ttA@mail.gmail.com>
Message-ID: <CABsWbRbq9a48q+dzbsehYPM6-5QmFP=EttFBwEa1r7O6OjP3WA@mail.gmail.com>

Giovanni,

Thank you very much, my intention is to know the way  to handle/manipulate
(low level) sip packets with freeswitch if it possible. As per your reply
this is obvious way to use Kamailio or OpenSIPS infront of FS.

On Thu, May 21, 2015 at 4:31 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Naveen,
>
> Please take note that FreeSWITCH is not intended to let the user to
> manipulate in arbitrary way SIP dialogs.
>
> You can do that for specific purposes and in.specific cases, within
> specific boundaries.
>
> If you are looking to interact with SIP directly and freely, you may want
> to look at Kamailio and OpenSIPS.
>
> You can put one of them in front of FreeSWITCH, and you can cross command
> them, for example via.lua scripting (both of them and FreeSWITCH can be
> scripted in.lua).
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On May 20, 2015 3:40 PM, "Naveen Tamanam" <naveen32india at gmail.com> wrote:
>
>> I am aware of uuid_hangup. Indeed my intention to know the way to send
>> (low level)sip messages
>> from fs console for a selected user.
>>
>>
>> On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>>  I would like to reject the call when it ringing from the fs console.
>>>
>>>
>>> uuid_hangup
>>>
>>>
>>> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com> wrote:
>>>
>>>> I am trying to do the following,  I would like to reject the call when
>>>> it ringing from the fs console.
>>>> And second thing is I am pretty much eager to know the way to send
>>>> sip(signaling) message manually for a
>>>> selected channel from fs console.
>>>>
>>>>
>>>> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>>
>>>>> respond is one way, what exact message are you trying to send, and at
>>>>> what point in the call.  There are capabilites to trigger re-invites in
>>>>> some situations, transfer, send notify or info or message.  It depends on
>>>>> what exactly you are trying to do.
>>>>>
>>>>>
>>>>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com>
>>>>> wrote:
>>>>>
>>>>> Hi,
>>>>>
>>>>> I am wondering how to send sip signaling message from the fs console
>>>>> for the particular user/caller.
>>>>> I found  respond dialplan application to send sip message back to the
>>>>> caller.
>>>>> Like
>>>>>
>>>>> <action application="respond" data="480 Try again later"/>
>>>>>
>>>>> ?Is there any way to send sip message back to the caller. One use case
>>>>> is call rejection or playing with SIP?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Thanks & Regards,
>>>> Naveen Tamanam
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Thanks & Regards,
>> Naveen Tamanam
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regards,
Naveen Tamanam
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From mike at jerris.com  Fri May 22 00:23:59 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 May 2015 16:23:59 -0400
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <CABsWbRbq9a48q+dzbsehYPM6-5QmFP=EttFBwEa1r7O6OjP3WA@mail.gmail.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
	<CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
	<CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
	<CALXCt0r4RDeQQ2VSWuPqbm2Umoda50c0bg9NGEUYEbvGVi+ttA@mail.gmail.com>
	<CABsWbRbq9a48q+dzbsehYPM6-5QmFP=EttFBwEa1r7O6OjP3WA@mail.gmail.com>
Message-ID: <1A821973-1FDC-41E4-8484-D89A4B3ABF4A@jerris.com>

Or its totally possible that I have asked for specifics multiple times now and you have ignored me every single time.  YOU CAN DO THIS IN FREESWITCH BUT I CAN NOT TELL YOU HOW UNLESS YOU TELL ME WHAT SIP MESSAGES YOU ARE TRYING TO SEND IN WHAT SITUATIONS!!!!!!!!


> On May 21, 2015, at 4:14 PM, Naveen Tamanam <naveen32india at gmail.com> wrote:
> 
> Giovanni, 
> 
> Thank you very much, my intention is to know the way  to handle/manipulate (low level) sip packets with freeswitch if it possible. As per your reply this is obvious way to use Kamailio or OpenSIPS infront of FS. 
> 
> On Thu, May 21, 2015 at 4:31 PM, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
> Naveen,
> 
> Please take note that FreeSWITCH is not intended to let the user to manipulate in arbitrary way SIP dialogs.
> 
> You can do that for specific purposes and in.specific cases, within specific boundaries.
> 
> If you are looking to interact with SIP directly and freely, you may want to look at Kamailio and OpenSIPS.
> 
> You can put one of them in front of FreeSWITCH, and you can cross command them, for example via.lua scripting (both of them and FreeSWITCH can be scripted in.lua).
> 
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> 
> On May 20, 2015 3:40 PM, "Naveen Tamanam" <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
> I am aware of uuid_hangup. Indeed my intention to know the way to send (low level)sip messages 
> from fs console for a selected user. 
> 
> 
> On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
>  I would like to reject the call when it ringing from the fs console. 
> 
> uuid_hangup
>  
> 
> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
> I am trying to do the following,  I would like to reject the call when it ringing from the fs console. 
> And second thing is I am pretty much eager to know the way to send sip(signaling) message manually for a 
> selected channel from fs console. 
> 
> 
> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> respond is one way, what exact message are you trying to send, and at what point in the call.  There are capabilites to trigger re-invites in some situations, transfer, send notify or info or message.  It depends on what exactly you are trying to do.
> 
> 
>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> 
>> Hi, 
>> 
>> I am wondering how to send sip signaling message from the fs console for the particular user/caller. 
>> I found  respond dialplan application to send sip message back to the caller. 
>> Like 
>> <action application="respond" data="480 Try again later"/>
>> ?Is there any way to send sip message back to the caller. One use case is call rejection or playing with SIP?
>> 

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From italorossib at gmail.com  Fri May 22 00:49:50 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 21 May 2015 17:49:50 -0300
Subject: [Freeswitch-users] Caller id in P-Asserted-Identity
In-Reply-To: <4088C1E6-932D-46F1-8675-391D0F1A1280@gmx.net>
References: <4088C1E6-932D-46F1-8675-391D0F1A1280@gmx.net>
Message-ID: <CAJR7f2G6gNG5XP-gaBehViwr9a4e4eOZdLBRafbosRAHu0rgWA@mail.gmail.com>

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

On Thu, May 21, 2015 at 4:57 PM, mbo <mbodbg at gmx.net> wrote:

> Our carrier sends us always the callerid in the P-Asserted-Identity field
> in the SIP header. When a caller calls anonymous, in the INVITE the
> relevant fields in the header look like:
>
> From: <sip:anonymous at anonymous.invalid;user=phone;udp>
> P-Asserted-Identity: <sip:+PHONENUMBER at IPADDRESS>
> Privacy: header;user;id
>
> When I bridge the call, in the INVITE which is sent to the remote server,
> the relevant fields in the header look like:
>
> From: "anonymous" <sip:+PHONENUMBER at IP-ADDRESS>;tag=K90ap6HBNQDtQ
> Privacy: id
> P-Asserted-Identity: "anonymous" <sip:+PHONENUMBER at IP-ADDRESS>
>
> But I fo not want to sent the callerid at all, so the fields should be set
> to:
>
> From: <sip:anonymous at anonymous.invalid;user=phone;udp>
> P-Asserted-Identity: "anonymous" <sip:+anonymous at anonymous.invalid
> ;user=phone;udp>
> Privacy: header;user;id
>
> or drop the P-Asserted-Identity field like:
>
> From: <sip:anonymous at anonymous.invalid;user=phone;udp>
> Privacy: header;user;id
>
> Which variables do I need to set before the call is bridged to archive
> this (I?m still using FS 1.2)
>
> Thanks and Regards
>
> Markus
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
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From olegstolyar at gmail.com  Fri May 22 02:46:23 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Thu, 21 May 2015 15:46:23 -0700
Subject: [Freeswitch-users] Session-expires overwrite in OK Response
Message-ID: <CAC8JMO+GwKnYC1rxnqh8YevZ3Y4BGd7avF_GWvDXvXQTRFYD3A@mail.gmail.com>

When I send an INVITE to FS that contains a Session-Expires header, FS
always overwrites it in the 200 OK response with this:

Session-Expires: 120;refresher=uac

How can I stop it? I saw that there is the sip-force-expires variable but I
am not setting it anywhere.

If I can't stop it from overwriting the value sent by the client, can I at
least change 120 to a different value somewhere?  Changing
minimum-session-expires and session-timeout did not seem to help.
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From fvillarroel at yahoo.com  Fri May 22 02:50:17 2015
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Thu, 21 May 2015 22:50:17 +0000 (UTC)
Subject: [Freeswitch-users] Test
Message-ID: <1088066209.44934.1432248617242.JavaMail.yahoo@mail.yahoo.com>

This is a test
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From gmaruzz at gmail.com  Fri May 22 02:58:21 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 22 May 2015 00:58:21 +0200
Subject: [Freeswitch-users] Test
In-Reply-To: <1088066209.44934.1432248617242.JavaMail.yahoo@mail.yahoo.com>
References: <1088066209.44934.1432248617242.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <CALXCt0pScK02qJXTCApctm2H1RdNAiqPsRYpHT819AjXdNKEtQ@mail.gmail.com>

Copy dat

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 22, 2015 12:51 AM, "FERNANDO VILLARROEL" <fvillarroel at yahoo.com>
wrote:

> This is a test
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fvillarroel at yahoo.com  Fri May 22 03:04:42 2015
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Thu, 21 May 2015 23:04:42 +0000 (UTC)
Subject: [Freeswitch-users] Nibblebill deducting balance two times
Message-ID: <539301371.36291.1432249482202.JavaMail.yahoo@mail.yahoo.com>

Hello All.
I am using FS with nibblebill and mod_lcr.
But i do not understand why nibblebill deduc balance two times; my log :
2015-05-21 19:56:08.846179 [DEBUG] mod_nibblebill.c:546 31 seconds passed since last bill time of 2015-05-21 19:55:372015-05-21 19:56:08.846179 [DEBUG] mod_nibblebill.c:563 Billing $16.800001 to 2 (Call: c457b3c6-6bc8-445e-9697-2ffc327c6722 / 0.000000 so far)2015-05-21 19:56:08.846179 [DEBUG] mod_nibblebill.c:393 Doing update query[UPDATE auth_user SET saldo=saldo-16.800001 WHERE id='2']2015-05-21 19:56:08.866181 [DEBUG] mod_nibblebill.c:428 Retrieved current balance for account 2 (balance = 83.199999)2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:60 sofia/external/992797744 Standard HANGUP, cause: NORMAL_CLEARING2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:737 (sofia/external/992797744) State HANGUP going to sleep2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:504 (sofia/external/992797744) State Change CS_HANGUP -> CS_REPORTING2015-05-21 19:56:08.866181 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/992797744 [BREAK]2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:472 (sofia/external/992797744) Running State Change CS_REPORTING2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:823 (sofia/external/992797744) State REPORTING2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:104 sofia/external/992797744 Standard REPORTING, cause: NORMAL_CLEARING2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:823 (sofia/external/992797744) State REPORTING going to sleep2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:498 (sofia/external/992797744) State Change CS_REPORTING -> CS_DESTROY2015-05-21 19:56:08.866181 [DEBUG] switch_core_session.c:1388 Send signal sofia/external/992797744 [BREAK]2015-05-21 19:56:08.866181 [DEBUG] switch_core_session.c:1615 Session 8547 (sofia/external/992797744) Locked, Waiting on external entities2015-05-21 19:56:08.866181 [NOTICE] switch_core_session.c:1633 Session 8547 (sofia/external/992797744) Ended2015-05-21 19:56:08.866181 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/992797744 [CS_DESTROY]2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:626 (sofia/external/992797744) Running State Change CS_DESTROY2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:636 (sofia/external/992797744) State DESTROY2015-05-21 19:56:08.866181 [DEBUG] mod_sofia.c:323 sofia/external/992797744 SOFIA DESTROY2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:111 sofia/external/992797744 Standard DESTROY2015-05-21 19:56:08.866181 [DEBUG] switch_core_state_machine.c:636 (sofia/external/992797744) State DESTROY going to sleep2015-05-21 19:56:08.886193 [DEBUG] mod_nibblebill.c:420 Doing lookup query[SELECT saldo AS nibble_balance FROM auth_user WHERE id='2']2015-05-21 19:56:08.886193 [DEBUG] mod_nibblebill.c:428 Retrieved current balance for account 2 (balance = 66.399998)2015-05-21 19:56:08.886193 [DEBUG] switch_core_state_machine.c:60 sofia/internal/448908901 at ssw.opendata.cl Standard HANGUP, cause: NORMAL_CLEARING2015-05-21 19:56:08.886193 [DEBUG] switch_core_state_machine.c:737 (sofia/internal/448908901 at ssw.opendata.cl) State HANGUP going to sleep2015-05-21 19:56:08.886193 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/448908901 at ssw.opendata.cl) State Change CS_HANGUP -> CS_REPORTING2015-05-21 19:56:08.886193 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/448908901 at ssw.opendata.cl [BREAK]2015-05-21 19:56:08.886193 [DEBUG] switch_core_state_machine.c:472 (sofia/internal/448908901 at ssw.opendata.cl) Running State Change CS_REPORTING2015-05-21 19:56:08.886193 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/448908901 at ssw.opendata.cl) State REPORTING2015-05-21 19:56:08.886193 [DEBUG] switch_scheduler.c:144 Deleting task 8550 switch_core_session_sched_heartbeat (52f0e5f6-c97e-41c1-9a39-a4d12ec0683f)2015-05-21 19:56:09.086191 [DEBUG] switch_core_state_machine.c:104 sofia/internal/448908901 at ssw.opendata.cl Standard REPORTING, cause: NORMAL_CLEARING2015-05-21 19:56:09.086191 [DEBUG] switch_core_state_machine.c:823 (sofia/internal/448908901 at ssw.opendata.cl) State REPORTING going to sleep2015-05-21 19:56:09.086191 [DEBUG] switch_core_state_machine.c:498 (sofia/internal/448908901 at ssw.opendata.cl) State Change CS_REPORTING -> CS_DESTROY2015-05-21 19:56:09.086191 [DEBUG] switch_core_session.c:1388 Send signal sofia/internal/448908901 at ssw.opendata.cl [BREAK]2015-05-21 19:56:09.086191 [DEBUG] switch_core_session.c:1615 Session 8546 (sofia/internal/448908901 at ssw.opendata.cl) Locked, Waiting on external entities2015-05-21 19:56:09.086191 [NOTICE] switch_core_session.c:1633 Session 8546 (sofia/internal/448908901 at ssw.opendata.cl) Ended2015-05-21 19:56:09.086191 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/448908901 at ssw.opendata.cl [CS_DESTROY]2015-05-21 19:56:09.086191 [DEBUG] switch_core_state_machine.c:626 (sofia/internal/448908901 at ssw.opendata.cl) Running State Change CS_DESTROY2015-05-21 19:56:09.086191 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/448908901 at ssw.opendata.cl) State DESTROY2015-05-21 19:56:09.086191 [DEBUG] mod_sofia.c:323 sofia/internal/448908901 at ssw.opendata.cl SOFIA DESTROY2015-05-21 19:56:09.086191 [DEBUG] switch_core_state_machine.c:111 sofia/internal/448908901 at ssw.opendata.cl Standard DESTROY2015-05-21 19:56:09.086191 [DEBUG] switch_core_state_machine.c:636 (sofia/internal/448908901 at ssw.opendata.cl) State DESTROY going to sleep2015-05-21 19:56:09.386178 [DEBUG] switch_scheduler.c:144 Deleting task 8549 switch_ivr_schedule_hangup (c457b3c6-6bc8-445e-9697-2ffc327c6722)

As you can see the first time say:
[UPDATE auth_user SET saldo=saldo-16.800001 WHERE id='2']2015-05-21 19:56:08.866181 [DEBUG] mod_nibblebill.c:428 Retrieved current balance for account 2 (balance = 83.199999)
The new balance is 83.199999 it's fine
and then
2015-05-21 19:56:08.886193 [DEBUG] mod_nibblebill.c:420 Doing lookup query[SELECT saldo AS nibble_balance FROM auth_user WHERE id='2']2015-05-21 19:56:08.886193 [DEBUG] mod_nibblebill.c:428 Retrieved current balance for account 2 (balance = 66.399998)
But again retrieved balance and now balance is 66.399998 it's wrong!!

My dialplan:
? ? <extension name="Todo">? ? ? <condition expression="^([2-9]\d{8})$" field="destination_number">? ? ? ? <action application="export" data="dialed_extension=$1"/>? ? ? ? <action application="set" data="hangup_after_bridge=true"/>
? ? ? ? <action application="set" data="continue_on_fail=true"/>
? ? ? ? <action application="sched_hangup" data="+1800 alloted_timeout"/>
? ? ? ? <action application="limit" data="hash fraud_protection calls_max_intl ${calls_max_intl} !NORMAL_TEMPORARY_FAILURE"/>
? ? ? ? <action application="lcr" data="$1 pg_prefix2"/>
? ? ? ? <action application="set" data="nibble_rate=${lcr_rate_1}"/>
? ? ? ? <action application="set" data="nibble_account=${accountcode}"/>
? ? ? ? <action application="set" data="lowbal_amt=${lowbal_amt}"/>
? ? ? ? <action application="set" data="nobal_amt=${nobal_amt}"/>
? ? ? ? <!--<action application="nibblebill" data="heartbeat 10"/>-->
? ? ? ? <action application="set" data="bypass_media=true"/>? ? ? ? <action application="bridge" data="${lcr_auto_route}"/>
Anyone give some idea or what i am doing wrong.
Regards
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From olegstolyar at gmail.com  Fri May 22 04:11:52 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Thu, 21 May 2015 17:11:52 -0700
Subject: [Freeswitch-users] Session-expires overwrite in OK Response
In-Reply-To: <CAC8JMO+GwKnYC1rxnqh8YevZ3Y4BGd7avF_GWvDXvXQTRFYD3A@mail.gmail.com>
References: <CAC8JMO+GwKnYC1rxnqh8YevZ3Y4BGd7avF_GWvDXvXQTRFYD3A@mail.gmail.com>
Message-ID: <CAC8JMO+Je4UMt15p3mYXP9M+vNBgkoOvevywscPRsd6D=QtoAw@mail.gmail.com>

As with most problems, there was a simple solution.

Simply disabling session timers on the FS sip profile did the trick.

On Thu, May 21, 2015 at 3:46 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> When I send an INVITE to FS that contains a Session-Expires header, FS
> always overwrites it in the 200 OK response with this:
>
> Session-Expires: 120;refresher=uac
>
> How can I stop it? I saw that there is the sip-force-expires variable but
> I am not setting it anywhere.
>
> If I can't stop it from overwriting the value sent by the client, can I at
> least change 120 to a different value somewhere?  Changing
> minimum-session-expires and session-timeout did not seem to help.
>
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From s.safarov at gmail.com  Fri May 22 08:55:30 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 22 May 2015 07:55:30 +0300
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <1A821973-1FDC-41E4-8484-D89A4B3ABF4A@jerris.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
	<CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
	<CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
	<CALXCt0r4RDeQQ2VSWuPqbm2Umoda50c0bg9NGEUYEbvGVi+ttA@mail.gmail.com>
	<CABsWbRbq9a48q+dzbsehYPM6-5QmFP=EttFBwEa1r7O6OjP3WA@mail.gmail.com>
	<1A821973-1FDC-41E4-8484-D89A4B3ABF4A@jerris.com>
Message-ID: <CAHtxdDe8POGEPFs0auaUuGJwZbudkJ42dBs6qDozSGu_a8CLHA@mail.gmail.com>

Michael, I think over how to organize the control of the call recording on
OpenSips from FS dialplan (Scheme where FS is managing callcontrol,
OpenSIPs RTP media processing)
As an option I think to make it through
1) additional headers in the messages 100, 180, 183, 200 (start recording).
Stop recording through RE-INVITE;
2) as a separate SIP INFO message (start, stop recording - more elegant
solution).

Tell me, please, as much as possible
1) send a custom SIP INFO message
2) how custom headers can be added to 100, 180, 183, 200 messages


On Thu, May 21, 2015 at 11:23 PM, Michael Jerris <mike at jerris.com> wrote:

> Or its totally possible that I have asked for specifics multiple times now
> and you have ignored me every single time.  YOU CAN DO THIS IN FREESWITCH
> BUT I CAN NOT TELL YOU HOW UNLESS YOU TELL ME WHAT SIP MESSAGES YOU ARE
> TRYING TO SEND IN WHAT SITUATIONS!!!!!!!!
>
>
> On May 21, 2015, at 4:14 PM, Naveen Tamanam <naveen32india at gmail.com>
> wrote:
>
> Giovanni,
>
> Thank you very much, my intention is to know the way  to handle/manipulate
> (low level) sip packets with freeswitch if it possible. As per your reply
> this is obvious way to use Kamailio or OpenSIPS infront of FS.
>
> On Thu, May 21, 2015 at 4:31 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Naveen,
>>
>> Please take note that FreeSWITCH is not intended to let the user to
>> manipulate in arbitrary way SIP dialogs.
>>
>> You can do that for specific purposes and in.specific cases, within
>> specific boundaries.
>>
>> If you are looking to interact with SIP directly and freely, you may want
>> to look at Kamailio and OpenSIPS.
>>
>> You can put one of them in front of FreeSWITCH, and you can cross command
>> them, for example via.lua scripting (both of them and FreeSWITCH can be
>> scripted in.lua).
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On May 20, 2015 3:40 PM, "Naveen Tamanam" <naveen32india at gmail.com>
>> wrote:
>>
>>> I am aware of uuid_hangup. Indeed my intention to know the way to send
>>> (low level)sip messages
>>> from fs console for a selected user.
>>>
>>>
>>> On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com>
>>> wrote:
>>>
>>>>  I would like to reject the call when it ringing from the fs console.
>>>>
>>>>
>>>> uuid_hangup
>>>>
>>>>
>>>> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com>
>>>> wrote:
>>>>
>>>>> I am trying to do the following,  I would like to reject the call when
>>>>> it ringing from the fs console.
>>>>> And second thing is I am pretty much eager to know the way to send
>>>>> sip(signaling) message manually for a
>>>>> selected channel from fs console.
>>>>>
>>>>>
>>>>> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>>> respond is one way, what exact message are you trying to send, and at
>>>>>> what point in the call.  There are capabilites to trigger re-invites in
>>>>>> some situations, transfer, send notify or info or message.  It depends on
>>>>>> what exactly you are trying to do.
>>>>>>
>>>>>>
>>>>>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I am wondering how to send sip signaling message from the fs console
>>>>>> for the particular user/caller.
>>>>>> I found  respond dialplan application to send sip message back to the
>>>>>> caller.
>>>>>> Like
>>>>>>
>>>>>> <action application="respond" data="480 Try again later"/>
>>>>>>
>>>>>> ?Is there any way to send sip message back to the caller. One use
>>>>>> case is call rejection or playing with SIP?
>>>>>>
>>>>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dm at dwide.com  Fri May 22 11:31:55 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Fri, 22 May 2015 11:31:55 +0400
Subject: [Freeswitch-users] Special dialplan extension
Message-ID: <555EDB6B.8070902@dwide.com>

Hello,

I want to configure that case in dialplan extension:

  - playback welcome prompt (<action application="playback" 
data="phrase:welcome_prompt"/>)
  - start call to external number (<action application="bridge" 
data="sofia/gateway/sipprovider/12122417020 at sipprovider"/>)
  - playback second prompt in incoming leg (while bridge calling to 
12122417020)
  - connect incoming leg and outgoing leg at early media will received 
(12122417020)

Is it possible to start a call and continue play second prompt?

How can legs be connected when early media received? (and stoped play 
second prompt)

Thank you!


From boros at vmtele.com  Fri May 22 12:02:07 2015
From: boros at vmtele.com (=?windows-1252?Q?Tom=E1=9A_Boros?=)
Date: Fri, 22 May 2015 10:02:07 +0200
Subject: [Freeswitch-users] Caller id in P-Asserted-Identity
In-Reply-To: <CAJR7f2G6gNG5XP-gaBehViwr9a4e4eOZdLBRafbosRAHu0rgWA@mail.gmail.com>
References: <4088C1E6-932D-46F1-8675-391D0F1A1280@gmx.net>
	<CAJR7f2G6gNG5XP-gaBehViwr9a4e4eOZdLBRafbosRAHu0rgWA@mail.gmail.com>
Message-ID: <555EE27F.7030309@vmtele.com>

Or you can use:

effective_caller_id_number = anonymous
effective_caller_id_name = anonymous
  and sip_from_uri=sip:anonymous at anonymous.invalid

and of cource sip_cid_type

all the variables should be exported or placed in the {} of the bridge.

Thomas

On 21.05.2015 22:49, ?talo Rossi wrote:
> https://wiki.freeswitch.org/wiki/Variable_sip_cid_type
>
> On Thu, May 21, 2015 at 4:57 PM, mbo <mbodbg at gmx.net 
> <mailto:mbodbg at gmx.net>> wrote:
>
>     Our carrier sends us always the callerid in the
>     P-Asserted-Identity field in the SIP header. When a caller calls
>     anonymous, in the INVITE the relevant fields in the header look like:
>
>     From: <sip:anonymous at anonymous.invalid;user=phone;udp>
>     P-Asserted-Identity: <sip:+PHONENUMBER at IPADDRESS>
>     Privacy: header;user;id
>
>     When I bridge the call, in the INVITE which is sent to the remote
>     server, the relevant fields in the header look like:
>
>     From: "anonymous" <sip:+PHONENUMBER at IP-ADDRESS>;tag=K90ap6HBNQDtQ
>     Privacy: id
>     P-Asserted-Identity: "anonymous" <sip:+PHONENUMBER at IP-ADDRESS>
>
>     But I fo not want to sent the callerid at all, so the fields
>     should be set to:
>
>     From: <sip:anonymous at anonymous.invalid;user=phone;udp>
>     P-Asserted-Identity: "anonymous"
>     <sip:+anonymous at anonymous.invalid;user=phone;udp>
>     Privacy: header;user;id
>
>     or drop the P-Asserted-Identity field like:
>
>     From: <sip:anonymous at anonymous.invalid;user=phone;udp>
>     Privacy: header;user;id
>
>     Which variables do I need to set before the call is bridged to
>     archive this (I?m still using FS 1.2)
>
>     Thanks and Regards
>
>     Markus
>
>
>
>
>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> ?talo Rossi
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From giggsey at gmail.com  Fri May 22 12:10:12 2015
From: giggsey at gmail.com (Joshua Gigg)
Date: Fri, 22 May 2015 09:10:12 +0100
Subject: [Freeswitch-users] Update CLI on Freeswitch originated call
In-Reply-To: <46E4639F-C60C-4A72-8134-CE977B2DDBD2@jerris.com>
References: <CADcg8iUL1EsMdedCm6793O83E+pK=UBPFBFqkgbVbJtjvdCmZw@mail.gmail.com>
	<CAKzWOxUut2N0-o+SKcqKkEpFp6g_9UyFHrnr7y2FobutycNRhw@mail.gmail.com>
	<CADcg8iUT=TybcX+9_X1+3kuHQde+Edj+2u+0=XcKf2Fccu5+sg@mail.gmail.com>
	<CADcg8iWyxxZjF1sesYOxT8di2L1N16=cODjjysAqNci=354+YQ@mail.gmail.com>
	<789BFCEB-3922-4952-BF04-4651B321CC0F@jerris.com>
	<CADcg8iVNCwUmhLBsBOVYe=r1tXwwPtwjbtxB7OyHRrKVjqOVzg@mail.gmail.com>
	<46E4639F-C60C-4A72-8134-CE977B2DDBD2@jerris.com>
Message-ID: <CADcg8iVx9jdfNPXJ6zNpcumtgFPWetKz6uOz8=9uhnjrs+yqJA@mail.gmail.com>

In mod_commands.c, it appears to split the argv by spaces, and pass that to
the switch. mod_sofia will take the second argument as the phone number.

When I try it from the API (uuid_display *callid* Name 1234), it sends the
sip as:


From: "Name 1234" *ThisCallToAddress*

Instead of splitting it up.

My snom does then display Name 1234 on the screen, but as a text name,
instead of updating the number.

This can work, but I'd rather be able to update the number too.

On 21 May 2015 at 15:26, Michael Jerris <mike at jerris.com> wrote:

> from mod_dptools:
>
>     SWITCH_ADD_APP(app_interface, "send_display", "Send session a new
> display", "Send session a new display.", display_function, "<text>",
>
>                    SAF_SUPPORT_NOMEDIA);
>
>
>
>
> there is a similar api command in mod_commands.  I think if you have
> ignore_dispaly_updates it may not work, try it out to be sure.
>
> On May 21, 2015, at 4:05 AM, Joshua Gigg <giggsey at gmail.com> wrote:
>
> I do have ignore_dispaly_updates=true, but that's on outbound phones, my
> displays were updating with the number being bridged out to.
>
> Also, I want to trigger it manually rather than just automatically when a
> transfer takes place.
>
> On 20 May 2015 at 17:26, Michael Jerris <mike at jerris.com> wrote:
>
>> this should happen by default unless you have ignore_dispaly_updates=true.
>>
>>
>> On May 20, 2015, at 12:18 PM, Joshua Gigg <giggsey at gmail.com> wrote:
>>
>> Digging up an old thread, but I've been looking into this a bit more.
>>
>> What I want to do is described in RFC 4916. What I want to try to use an
>> application / api command to generate a SIP INFO/UPDATE packet that would
>> update the Caller ID.
>>
>> On 29 January 2015 at 20:06, Joshua Gigg <giggsey at gmail.com> wrote:
>>
>>> When a transfer completes, FreeSWITCH will send an UPDATE message to the
>>> SIP server updating the caller id.
>>>
>>> Is there a way of making FreeSWITCH send this message via a
>>> dialplan/command?
>>>
>>> On 29 January 2015 at 17:58, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>> Could you expound upon this question a bit? What does "update the CLI"
>>>> mean?
>>>> Thanks,
>>>> MC
>>>>
>>>> On Thu, Jan 29, 2015 at 1:20 AM, Joshua Gigg <giggsey at gmail.com> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> Is it possible to update the CLI at will once a Freeswitch originated
>>>>> call has been answered?
>>>>>
>>>>> I know it can update during a transfer, but I want to be able to
>>>>> control it directly myself.
>>>>>
>>>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Joshua Gigg
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From boros at vmtele.com  Fri May 22 13:17:45 2015
From: boros at vmtele.com (=?UTF-8?B?VG9tw6HFoSBCb3Jvcw==?=)
Date: Fri, 22 May 2015 11:17:45 +0200
Subject: [Freeswitch-users] RPID to PAI translation
Message-ID: <555EF439.4070501@vmtele.com>

Hello,

we have the following problem.

On profile, which connects to the provider we use P-Asserted-Identity 
for identification.
In the internal line, we use Remote-Party-ID for identification.

The following problem occurs:
When doing calls, Invites are correctly translated from PAI to RPID, but 
when a 200 OK arrives from our Internal network, it has a 
Remote-Party-ID and is translated to a P-Asserted-Identity, but the 
number is changed.

I am attaching SIP messages:

recv 1167 bytes from udp/[192.168.5.17]:5060 at 10:26:06.319229:
------------------------------------------------------------------------
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.5.97;rport=5060;branch=z9hG4bK5mNSKt220vF5K
    Record-Route: <sip:192.168.5.66;lr=on;ftag=m29tUHH087vFr;did=01a.d76>
    Record-Route: <sip:192.168.5.18;lr=on;did=01a.bf7>
    From: "00421800123993" 
<sip:00421800123993 at 192.168.5.97>;tag=m29tUHH087vFr
    To: <sip:307000421221028600 at 192.168.5.17>;tag=p9D7Npj3amv3j
    Call-ID: 953485a8-d9b6-4c36-a54d-01fab395ad48
    CSeq: 75805518 INVITE
    Contact: <sip:307000421221028600 at 192.168.5.94:5060;transport=udp>
    User-Agent: VMTele-SBC
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, 
REGISTER, REFER, NOTIFY
    Supported: timer, path, replaces
    Allow-Events: talk, hold, conference, refer
    Content-Type: application/sdp
    Content-Disposition: session
    Content-Length: 278
    X-FS-Support: update_display,send_info
*Remote-Party-ID: 
<sip:+421221028600 at 212.232.17.77>;party=calling;screen=yes;privacy=off*

    v=0
    o=VMTele-SBC 1432264463 1432264464 IN IP4 192.168.5.94
    s=VMTele-SBC
    c=IN IP4 192.168.5.94
    t=0 0
    m=audio 18702 RTP/AVP 8 101 13
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:13 CN/8000
    a=ptime:20
    a=rtcp:18703 IN IP4 192.168.5.94

In debug logs I can see:
2015-05-22 10:26:06.328812 [ALERT] sofia.c:1198 
sofia/core-profile/307000421221028600 Same Callee ID "Outbound Call" 
<307000421221028600>

send 1030 bytes to udp/[212.232.17.60]:5060 at 10:26:06.340758:
------------------------------------------------------------------------
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 212.232.17.60:5060;branch=z9hG4bK4365798a;rport=5060
    From: <sip:+421800123993 at 212.232.17.60>;tag=as703def00
    To: <sip:+421221028600 at 212.232.17.77>;tag=XB02p6tZSjvFN
    Call-ID: 2598c37260ec91283d70ac220f32c5cf at 212.232.17.60:5060
    CSeq: 102 INVITE
    Contact: <sip:+421221028600 at 212.232.17.77:5060;transport=udp>
    User-Agent: VMTelecomSBC
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, 
REGISTER, REFER, PRACK, NOTIFY
    Require: timer
    Supported: precondition, 100rel, timer, path, replaces
    Allow-Events: talk, hold, conference, refer
    Session-Expires: 800;refresher=uac
    Content-Type: application/sdp
    Content-Disposition: session
    Content-Length: 257
*P-Asserted-Identity: "VIPTel" <sip:307000421221028600 at 212.232.17.77>*

    v=0
    o=VMTele-SBC 1432250761 1432250762 IN IP4 212.232.17.77
    s=VMTele-SBC
    c=IN IP4 212.232.17.77
    t=0 0
    m=audio 32404 RTP/AVP 0 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    a=rtcp:32405 IN IP4 212.232.17.77


I have made tests with PAI<-> PAI too, I mean I have used 
P-Asserted-Identity instead of Remote-Party-ID in 200 OK, and in such a 
case it worked correctly. Received number and name was forwarded to the 
other leg.

The problem only occurs, when the RPID is in the incoming 200 OK.

Do you think its a bug? I have pass-callee-id set to true on both 
profiles and I use pai and rpid as in sip_cid_type accordingly.

-- 
Tom?? Boros
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From mitchelle.bit at gmail.com  Fri May 22 13:45:14 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Fri, 22 May 2015 15:15:14 +0530
Subject: [Freeswitch-users] XML CDR help
Message-ID: <CANBm6SzMgrb8qNBrdU-X9WHpgoSStVg04NMtusfohGcvGsW-5g@mail.gmail.com>

Hi,
When processing XML CDRs sometimes(not all) the callee_id_number tag is
left blank. Please tell me why does this happen?

Thanks,
Mitchelle
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From mitchelle.bit at gmail.com  Fri May 22 13:48:50 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Fri, 22 May 2015 15:18:50 +0530
Subject: [Freeswitch-users] XML CDR help
In-Reply-To: <CANBm6SzMgrb8qNBrdU-X9WHpgoSStVg04NMtusfohGcvGsW-5g@mail.gmail.com>
References: <CANBm6SzMgrb8qNBrdU-X9WHpgoSStVg04NMtusfohGcvGsW-5g@mail.gmail.com>
Message-ID: <CANBm6SwqcTQR4LDukb7rAdEZsGvDQHs1mCbiVG99tmQrf48-hA@mail.gmail.com>

Please find the XML file on pastebin. The link is
http://pastebin.com/fAS7Gu2a

Thanks,
Mitchelle

On Fri, May 22, 2015 at 3:15 PM, Mitchelle Johnson <mitchelle.bit at gmail.com>
wrote:

> Hi,
> When processing XML CDRs sometimes(not all) the callee_id_number tag is
> left blank. Please tell me why does this happen?
>
> Thanks,
> Mitchelle
>
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From boros at vmtele.com  Fri May 22 14:13:16 2015
From: boros at vmtele.com (=?windows-1252?Q?Tom=E1=9A_Boros?=)
Date: Fri, 22 May 2015 12:13:16 +0200
Subject: [Freeswitch-users] RPID to PAI translation
In-Reply-To: <555EF439.4070501@vmtele.com>
References: <555EF439.4070501@vmtele.com>
Message-ID: <555F013C.7070104@vmtele.com>

Ok,
I have found it in the sources of mod_sofia.

In function sofia_update_callee_id the function parses the call ID only 
from the P-asserted-identity, so RPID is ignored.
Can be this posted to Jira as feature request?

Thank you,
Thomas

On 22.05.2015 11:17, Tom?? Boros wrote:
> Hello,
>
> we have the following problem.
>
> On profile, which connects to the provider we use P-Asserted-Identity 
> for identification.
> In the internal line, we use Remote-Party-ID for identification.
>
> The following problem occurs:
> When doing calls, Invites are correctly translated from PAI to RPID, 
> but when a 200 OK arrives from our Internal network, it has a 
> Remote-Party-ID and is translated to a P-Asserted-Identity, but the 
> number is changed.
>
> I am attaching SIP messages:
>
> recv 1167 bytes from udp/[192.168.5.17]:5060 at 10:26:06.319229:
> ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP 192.168.5.97;rport=5060;branch=z9hG4bK5mNSKt220vF5K
>    Record-Route: <sip:192.168.5.66;lr=on;ftag=m29tUHH087vFr;did=01a.d76>
>    Record-Route: <sip:192.168.5.18;lr=on;did=01a.bf7>
>    From: "00421800123993" 
> <sip:00421800123993 at 192.168.5.97>;tag=m29tUHH087vFr
>    To: <sip:307000421221028600 at 192.168.5.17>;tag=p9D7Npj3amv3j
>    Call-ID: 953485a8-d9b6-4c36-a54d-01fab395ad48
>    CSeq: 75805518 INVITE
>    Contact: <sip:307000421221028600 at 192.168.5.94:5060;transport=udp>
>    User-Agent: VMTele-SBC
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, 
> REGISTER, REFER, NOTIFY
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 278
>    X-FS-Support: update_display,send_info
> *Remote-Party-ID: 
> <sip:+421221028600 at 212.232.17.77>;party=calling;screen=yes;privacy=off*
>
>    v=0
>    o=VMTele-SBC 1432264463 1432264464 IN IP4 192.168.5.94
>    s=VMTele-SBC
>    c=IN IP4 192.168.5.94
>    t=0 0
>    m=audio 18702 RTP/AVP 8 101 13
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:13 CN/8000
>    a=ptime:20
>    a=rtcp:18703 IN IP4 192.168.5.94
>
> In debug logs I can see:
> 2015-05-22 10:26:06.328812 [ALERT] sofia.c:1198 
> sofia/core-profile/307000421221028600 Same Callee ID "Outbound Call" 
> <307000421221028600>
>
> send 1030 bytes to udp/[212.232.17.60]:5060 at 10:26:06.340758:
> ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP 212.232.17.60:5060;branch=z9hG4bK4365798a;rport=5060
>    From: <sip:+421800123993 at 212.232.17.60>;tag=as703def00
>    To: <sip:+421221028600 at 212.232.17.77>;tag=XB02p6tZSjvFN
>    Call-ID: 2598c37260ec91283d70ac220f32c5cf at 212.232.17.60:5060
>    CSeq: 102 INVITE
>    Contact: <sip:+421221028600 at 212.232.17.77:5060;transport=udp>
>    User-Agent: VMTelecomSBC
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, 
> REGISTER, REFER, PRACK, NOTIFY
>    Require: timer
>    Supported: precondition, 100rel, timer, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Session-Expires: 800;refresher=uac
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 257
> *P-Asserted-Identity: "VIPTel" <sip:307000421221028600 at 212.232.17.77>*
>
>    v=0
>    o=VMTele-SBC 1432250761 1432250762 IN IP4 212.232.17.77
>    s=VMTele-SBC
>    c=IN IP4 212.232.17.77
>    t=0 0
>    m=audio 32404 RTP/AVP 0 101
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>    a=rtcp:32405 IN IP4 212.232.17.77
>
>
> I have made tests with PAI<-> PAI too, I mean I have used 
> P-Asserted-Identity instead of Remote-Party-ID in 200 OK, and in such 
> a case it worked correctly. Received number and name was forwarded to 
> the other leg.
>
> The problem only occurs, when the RPID is in the incoming 200 OK.
>
> Do you think its a bug? I have pass-callee-id set to true on both 
> profiles and I use pai and rpid as in sip_cid_type accordingly.
>
> -- 
> Tom?? Boros
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From manish.thakor97 at gmail.com  Fri May 22 10:52:34 2015
From: manish.thakor97 at gmail.com (manish thakor)
Date: Fri, 22 May 2015 12:22:34 +0530
Subject: [Freeswitch-users] NO CDR logged on incoming calls upgrade version
	to 1.4.18
Message-ID: <CAO6=Ltm00k4u3kriTD+WLu6U80dmaVb4kDhC=ekyGa+4-s_Fjg@mail.gmail.com>

Hi,


I have to upgrade freeswitch version to 1.4.18.
Incoming calls are logged only if I set log on disk:
<param name="log-http-and-disk" value="true"/>


When i set param to false then its does not work.

-- 
Thanks & Regards

Manish Thakor
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From krice at freeswitch.org  Fri May 22 18:01:09 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 22 May 2015 14:01:09 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <555f36a576e4e_30a04a533021948@resque-worker-high.2.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From boros at vmtele.com  Fri May 22 18:39:57 2015
From: boros at vmtele.com (=?UTF-8?B?VG9tw6HFoSBCb3Jvcw==?=)
Date: Fri, 22 May 2015 16:39:57 +0200
Subject: [Freeswitch-users] Passing Privacy headers to other leg 18x 200
	responses
Message-ID: <555F3FBD.2070607@vmtele.com>

Hi,

I have an issue.
When a 200 OK response is received from the other leg, but it has its 
Callee ID restricted (Termination Identification Restriction TIR from 
etsi), the Privacy header is not appended to the 200 OK message on the 
first leg.

A 200 OK or a 180 Ringing is received with a Privacy: ID header, but 
this header will not be copied to the other channel when forwarding the 
200 OK to the caller.

I was digging in the source code and did not find anything about it. Do 
someone have a clue, where should it be implemented? Could someone post 
me a patch for it?

Thank you,
-- 
Tom?? Boros
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From mike at jerris.com  Fri May 22 19:52:27 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 22 May 2015 11:52:27 -0400
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <CAHtxdDe8POGEPFs0auaUuGJwZbudkJ42dBs6qDozSGu_a8CLHA@mail.gmail.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
	<CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
	<CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
	<CALXCt0r4RDeQQ2VSWuPqbm2Umoda50c0bg9NGEUYEbvGVi+ttA@mail.gmail.com>
	<CABsWbRbq9a48q+dzbsehYPM6-5QmFP=EttFBwEa1r7O6OjP3WA@mail.gmail.com>
	<1A821973-1FDC-41E4-8484-D89A4B3ABF4A@jerris.com>
	<CAHtxdDe8POGEPFs0auaUuGJwZbudkJ42dBs6qDozSGu_a8CLHA@mail.gmail.com>
Message-ID: <73BF366B-8CB1-4337-81B7-5237CD800147@jerris.com>


> On May 22, 2015, at 12:55 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> Michael, I think over how to organize the control of the call recording on OpenSips from FS dialplan (Scheme where FS is managing callcontrol, OpenSIPs RTP media processing)
> As an option I think to make it through
> 1) additional headers in the messages 100, 180, 183, 200 (start recording). Stop recording through RE-INVITE;
> 2) as a separate SIP INFO message (start, stop recording - more elegant solution).
> 
> Tell me, please, as much as possible
> 1) send a custom SIP INFO message 

from api interface: uuid_send_info_function
from dialplan: send_info

> 2) how custom headers can be added to 100, 180, 183, 200 messages

https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack#SofiaSIPStack-ChannelVariables <https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack#SofiaSIPStack-ChannelVariables>

> 
> On Thu, May 21, 2015 at 11:23 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Or its totally possible that I have asked for specifics multiple times now and you have ignored me every single time.  YOU CAN DO THIS IN FREESWITCH BUT I CAN NOT TELL YOU HOW UNLESS YOU TELL ME WHAT SIP MESSAGES YOU ARE TRYING TO SEND IN WHAT SITUATIONS!!!!!!!!
> 
> 
>> On May 21, 2015, at 4:14 PM, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> 
>> Giovanni, 
>> 
>> Thank you very much, my intention is to know the way  to handle/manipulate (low level) sip packets with freeswitch if it possible. As per your reply this is obvious way to use Kamailio or OpenSIPS infront of FS. 
>> 
>> On Thu, May 21, 2015 at 4:31 PM, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>> Naveen,
>> 
>> Please take note that FreeSWITCH is not intended to let the user to manipulate in arbitrary way SIP dialogs.
>> 
>> You can do that for specific purposes and in.specific cases, within specific boundaries.
>> 
>> If you are looking to interact with SIP directly and freely, you may want to look at Kamailio and OpenSIPS.
>> 
>> You can put one of them in front of FreeSWITCH, and you can cross command them, for example via.lua scripting (both of them and FreeSWITCH can be scripted in.lua).
>> 
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> 
>> On May 20, 2015 3:40 PM, "Naveen Tamanam" <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> I am aware of uuid_hangup. Indeed my intention to know the way to send (low level)sip messages 
>> from fs console for a selected user. 
>> 
>> 
>> On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
>>  I would like to reject the call when it ringing from the fs console. 
>> 
>> uuid_hangup
>>  
>> 
>> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> I am trying to do the following,  I would like to reject the call when it ringing from the fs console. 
>> And second thing is I am pretty much eager to know the way to send sip(signaling) message manually for a 
>> selected channel from fs console. 
>> 
>> 
>> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> respond is one way, what exact message are you trying to send, and at what point in the call.  There are capabilites to trigger re-invites in some situations, transfer, send notify or info or message.  It depends on what exactly you are trying to do.
>> 
>> 
>>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>>> 
>>> Hi, 
>>> 
>>> I am wondering how to send sip signaling message from the fs console for the particular user/caller. 
>>> I found  respond dialplan application to send sip message back to the caller. 
>>> Like 
>>> <action application="respond" data="480 Try again later"/>
>>> ?Is there any way to send sip message back to the caller. One use case is call rejection or playing with SIP?
>>> 
> 
> 
> _________________________________________________________________________
> 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 carlosj.gf at gmail.com  Fri May 22 16:40:52 2015
From: carlosj.gf at gmail.com (=?ISO-8859-1?Q?Carlos_Gonz=E1lez_Florido?=)
Date: Fri, 22 May 2015 14:40:52 +0200
Subject: [Freeswitch-users] Video always muted in conference (fs_video2)
Message-ID: <CAFKjMAYjKz32rE-G_aA639J5b27eNwnRAME31La25W9KiWbagA@mail.gmail.com>

Hello,

I'm testing the impressive fs_video2 branch, but I have the following
problems:

- some participants have their video always muted (and their camera is on
and working for Hangouts, for example).
- other participants start the same (video muted), but after 3-5 minutes
the video is automatically unmuted.
- if we do a layout change (using fs_cli), all participants (having video
muted or unmuted at that moment) go to muted video for the rest of the
conference.

What is the reason for fs to automatically mute the video? Is this expected
or does it look like a bug? Can this behaviour be tuned or disconnected?

Thank you,
Carlos Gonzalez
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From mike at jerris.com  Fri May 22 19:55:07 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 22 May 2015 11:55:07 -0400
Subject: [Freeswitch-users] Special dialplan extension
In-Reply-To: <555EDB6B.8070902@dwide.com>
References: <555EDB6B.8070902@dwide.com>
Message-ID: <ADA19AD6-09C6-44C3-9D00-F5D8513EF5FB@jerris.com>

use the second file as your ringback

> On May 22, 2015, at 3:31 AM, Dmitry Mordovin <dm at dwide.com> wrote:
> 
> Hello,
> 
> I want to configure that case in dialplan extension:
> 
>  - playback welcome prompt (<action application="playback" 
> data="phrase:welcome_prompt"/>)
>  - start call to external number (<action application="bridge" 
> data="sofia/gateway/sipprovider/12122417020 at sipprovider"/>)
>  - playback second prompt in incoming leg (while bridge calling to 
> 12122417020)
>  - connect incoming leg and outgoing leg at early media will received 
> (12122417020)
> 
> Is it possible to start a call and continue play second prompt?
> 
> How can legs be connected when early media received? (and stoped play 
> second prompt)
> 
> Thank you!



From mike at jerris.com  Fri May 22 20:02:14 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 22 May 2015 12:02:14 -0400
Subject: [Freeswitch-users] Update CLI on Freeswitch originated call
In-Reply-To: <CADcg8iVx9jdfNPXJ6zNpcumtgFPWetKz6uOz8=9uhnjrs+yqJA@mail.gmail.com>
References: <CADcg8iUL1EsMdedCm6793O83E+pK=UBPFBFqkgbVbJtjvdCmZw@mail.gmail.com>
	<CAKzWOxUut2N0-o+SKcqKkEpFp6g_9UyFHrnr7y2FobutycNRhw@mail.gmail.com>
	<CADcg8iUT=TybcX+9_X1+3kuHQde+Edj+2u+0=XcKf2Fccu5+sg@mail.gmail.com>
	<CADcg8iWyxxZjF1sesYOxT8di2L1N16=cODjjysAqNci=354+YQ@mail.gmail.com>
	<789BFCEB-3922-4952-BF04-4651B321CC0F@jerris.com>
	<CADcg8iVNCwUmhLBsBOVYe=r1tXwwPtwjbtxB7OyHRrKVjqOVzg@mail.gmail.com>
	<46E4639F-C60C-4A72-8134-CE977B2DDBD2@jerris.com>
	<CADcg8iVx9jdfNPXJ6zNpcumtgFPWetKz6uOz8=9uhnjrs+yqJA@mail.gmail.com>
Message-ID: <79625001-D2C3-4A59-BCEA-0798C9F2F526@jerris.com>

the command only supports sending name, it will use the destination_number from caller profile.  If you use esl or another way to generate the session message you can send both name and number.

check out the code in mod_sofia.c at 1636 if you want to follow the logic how it decides what to send.  Also, we could take a patch to send_display to let you specify both if you can figure out a backwards compatible way to specify it.  See uuid_send_message for an example of how to set multiple args in that message.


> On May 22, 2015, at 4:10 AM, Joshua Gigg <giggsey at gmail.com> wrote:
> 
> In mod_commands.c, it appears to split the argv by spaces, and pass that to the switch. mod_sofia will take the second argument as the phone number.
> 
> When I try it from the API (uuid_display callid Name 1234), it sends the sip as:
> 
> 
> From: "Name 1234" ThisCallToAddress
> 
> Instead of splitting it up.
> 
> My snom does then display Name 1234 on the screen, but as a text name, instead of updating the number.
> 
> This can work, but I'd rather be able to update the number too.
> 
> On 21 May 2015 at 15:26, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> from mod_dptools:
> 
>     SWITCH_ADD_APP(app_interface, "send_display", "Send session a new display", "Send session a new display.", display_function, "<text>",                     
>                    SAF_SUPPORT_NOMEDIA);                                                                                                                       
> 
> 
> there is a similar api command in mod_commands.  I think if you have ignore_dispaly_updates it may not work, try it out to be sure.
> 
>> On May 21, 2015, at 4:05 AM, Joshua Gigg <giggsey at gmail.com <mailto:giggsey at gmail.com>> wrote:
>> 
>> I do have ignore_dispaly_updates=true, but that's on outbound phones, my displays were updating with the number being bridged out to.
>> 
>> Also, I want to trigger it manually rather than just automatically when a transfer takes place.
>> 
>> On 20 May 2015 at 17:26, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> this should happen by default unless you have ignore_dispaly_updates=true.
>> 
>> 
>>> On May 20, 2015, at 12:18 PM, Joshua Gigg <giggsey at gmail.com <mailto:giggsey at gmail.com>> wrote:
>>> 
>>> Digging up an old thread, but I've been looking into this a bit more.
>>> 
>>> What I want to do is described in RFC 4916. What I want to try to use an application / api command to generate a SIP INFO/UPDATE packet that would update the Caller ID.
>>> 
>>> On 29 January 2015 at 20:06, Joshua Gigg <giggsey at gmail.com <mailto:giggsey at gmail.com>> wrote:
>>> When a transfer completes, FreeSWITCH will send an UPDATE message to the SIP server updating the caller id.
>>> 
>>> Is there a way of making FreeSWITCH send this message via a dialplan/command?
>>> 
>>> On 29 January 2015 at 17:58, Michael Collins <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
>>> Could you expound upon this question a bit? What does "update the CLI" mean?
>>> Thanks,
>>> MC
>>> 
>>> On Thu, Jan 29, 2015 at 1:20 AM, Joshua Gigg <giggsey at gmail.com <mailto:giggsey at gmail.com>> wrote:
>>> Hi,
>>> 
>>> Is it possible to update the CLI at will once a Freeswitch originated call has been answered?
>>> 
>>> I know it can update during a transfer, but I want to be able to control it directly myself.
>>> 
>> 
> 
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Joshua Gigg
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From dm at dwide.com  Fri May 22 20:06:00 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Fri, 22 May 2015 20:06:00 +0400
Subject: [Freeswitch-users] Special dialplan extension
In-Reply-To: <ADA19AD6-09C6-44C3-9D00-F5D8513EF5FB@jerris.com>
References: <555EDB6B.8070902@dwide.com>
	<ADA19AD6-09C6-44C3-9D00-F5D8513EF5FB@jerris.com>
Message-ID: <555F53E8.1030404@dwide.com>

Thank you Michael

Good idea!

On 05/22/2015 07:55 PM, Michael Jerris wrote:
> use the second file as your ringback
>
>> On May 22, 2015, at 3:31 AM, Dmitry Mordovin <dm at dwide.com> wrote:
>>
>> Hello,
>>
>> I want to configure that case in dialplan extension:
>>
>>   - playback welcome prompt (<action application="playback"
>> data="phrase:welcome_prompt"/>)
>>   - start call to external number (<action application="bridge"
>> data="sofia/gateway/sipprovider/12122417020 at sipprovider"/>)
>>   - playback second prompt in incoming leg (while bridge calling to
>> 12122417020)
>>   - connect incoming leg and outgoing leg at early media will received
>> (12122417020)
>>
>> Is it possible to start a call and continue play second prompt?
>>
>> How can legs be connected when early media received? (and stoped play
>> second prompt)
>>
>> 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



From mike at jerris.com  Fri May 22 20:04:31 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 22 May 2015 12:04:31 -0400
Subject: [Freeswitch-users] Video always muted in conference (fs_video2)
In-Reply-To: <CAFKjMAYjKz32rE-G_aA639J5b27eNwnRAME31La25W9KiWbagA@mail.gmail.com>
References: <CAFKjMAYjKz32rE-G_aA639J5b27eNwnRAME31La25W9KiWbagA@mail.gmail.com>
Message-ID: <7B131C27-7233-4E88-BB9E-E01BE588DE82@jerris.com>

I don't think its actually muted.  We are chasing down an issue that sounds just like this.  It might be related to ipv6 but we are still looking in to it.

> On May 22, 2015, at 8:40 AM, Carlos Gonz?lez Florido <carlosj.gf at gmail.com> wrote:
> 
> Hello,
> 
> I'm testing the impressive fs_video2 branch, but I have the following problems:
> 
> - some participants have their video always muted (and their camera is on and working for Hangouts, for example).
> - other participants start the same (video muted), but after 3-5 minutes the video is automatically unmuted.
> - if we do a layout change (using fs_cli), all participants (having video muted or unmuted at that moment) go to muted video for the rest of the conference.
> 
> What is the reason for fs to automatically mute the video? Is this expected or does it look like a bug? Can this behaviour be tuned or disconnected?
> 
> Thank you,
> Carlos Gonzalez



From giggsey at gmail.com  Fri May 22 20:27:00 2015
From: giggsey at gmail.com (Joshua Gigg)
Date: Fri, 22 May 2015 17:27:00 +0100
Subject: [Freeswitch-users] Update CLI on Freeswitch originated call
In-Reply-To: <79625001-D2C3-4A59-BCEA-0798C9F2F526@jerris.com>
References: <CADcg8iUL1EsMdedCm6793O83E+pK=UBPFBFqkgbVbJtjvdCmZw@mail.gmail.com>
	<CAKzWOxUut2N0-o+SKcqKkEpFp6g_9UyFHrnr7y2FobutycNRhw@mail.gmail.com>
	<CADcg8iUT=TybcX+9_X1+3kuHQde+Edj+2u+0=XcKf2Fccu5+sg@mail.gmail.com>
	<CADcg8iWyxxZjF1sesYOxT8di2L1N16=cODjjysAqNci=354+YQ@mail.gmail.com>
	<789BFCEB-3922-4952-BF04-4651B321CC0F@jerris.com>
	<CADcg8iVNCwUmhLBsBOVYe=r1tXwwPtwjbtxB7OyHRrKVjqOVzg@mail.gmail.com>
	<46E4639F-C60C-4A72-8134-CE977B2DDBD2@jerris.com>
	<CADcg8iVx9jdfNPXJ6zNpcumtgFPWetKz6uOz8=9uhnjrs+yqJA@mail.gmail.com>
	<79625001-D2C3-4A59-BCEA-0798C9F2F526@jerris.com>
Message-ID: <CADcg8iV+ApmG3HadsHYyYi+5aQkx7mwRpc4VTMFX9oob6PFt6g@mail.gmail.com>

Ahh, okay, I see now.

I'll try down the ESL route. We are sending messages via ESL anyway, but
just been executing commands or executing applications.


On 22 May 2015 at 17:02, Michael Jerris <mike at jerris.com> wrote:

> the command only supports sending name, it will use the destination_number
> from caller profile.  If you use esl or another way to generate the session
> message you can send both name and number.
>
> check out the code in mod_sofia.c at 1636 if you want to follow the logic
> how it decides what to send.  Also, we could take a patch to send_display
> to let you specify both if you can figure out a backwards compatible way to
> specify it.  See uuid_send_message for an example of how to set multiple
> args in that message.
>
>
> On May 22, 2015, at 4:10 AM, Joshua Gigg <giggsey at gmail.com> wrote:
>
> In mod_commands.c, it appears to split the argv by spaces, and pass that
> to the switch. mod_sofia will take the second argument as the phone number.
>
> When I try it from the API (uuid_display *callid* Name 1234), it sends
> the sip as:
>
>
> From: "Name 1234" *ThisCallToAddress*
>
> Instead of splitting it up.
>
> My snom does then display Name 1234 on the screen, but as a text name,
> instead of updating the number.
>
> This can work, but I'd rather be able to update the number too.
>
> On 21 May 2015 at 15:26, Michael Jerris <mike at jerris.com> wrote:
>
>> from mod_dptools:
>>
>>     SWITCH_ADD_APP(app_interface, "send_display", "Send session a new
>> display", "Send session a new display.", display_function, "<text>",
>>
>>                    SAF_SUPPORT_NOMEDIA);
>>
>>
>>
>>
>> there is a similar api command in mod_commands.  I think if you have
>> ignore_dispaly_updates it may not work, try it out to be sure.
>>
>> On May 21, 2015, at 4:05 AM, Joshua Gigg <giggsey at gmail.com> wrote:
>>
>> I do have ignore_dispaly_updates=true, but that's on outbound phones, my
>> displays were updating with the number being bridged out to.
>>
>> Also, I want to trigger it manually rather than just automatically when a
>> transfer takes place.
>>
>> On 20 May 2015 at 17:26, Michael Jerris <mike at jerris.com> wrote:
>>
>>> this should happen by default unless you have
>>> ignore_dispaly_updates=true.
>>>
>>>
>>> On May 20, 2015, at 12:18 PM, Joshua Gigg <giggsey at gmail.com> wrote:
>>>
>>> Digging up an old thread, but I've been looking into this a bit more.
>>>
>>> What I want to do is described in RFC 4916. What I want to try to use an
>>> application / api command to generate a SIP INFO/UPDATE packet that would
>>> update the Caller ID.
>>>
>>> On 29 January 2015 at 20:06, Joshua Gigg <giggsey at gmail.com> wrote:
>>>
>>>> When a transfer completes, FreeSWITCH will send an UPDATE message to
>>>> the SIP server updating the caller id.
>>>>
>>>> Is there a way of making FreeSWITCH send this message via a
>>>> dialplan/command?
>>>>
>>>> On 29 January 2015 at 17:58, Michael Collins <msc at freeswitch.org>
>>>> wrote:
>>>>
>>>>> Could you expound upon this question a bit? What does "update the CLI"
>>>>> mean?
>>>>> Thanks,
>>>>> MC
>>>>>
>>>>> On Thu, Jan 29, 2015 at 1:20 AM, Joshua Gigg <giggsey at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> Is it possible to update the CLI at will once a Freeswitch originated
>>>>>> call has been answered?
>>>>>>
>>>>>> I know it can update during a transfer, but I want to be able to
>>>>>> control it directly myself.
>>>>>>
>>>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Joshua Gigg
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Joshua Gigg
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From s.safarov at gmail.com  Fri May 22 21:24:29 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 22 May 2015 20:24:29 +0300
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <73BF366B-8CB1-4337-81B7-5237CD800147@jerris.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
	<CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
	<CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
	<CALXCt0r4RDeQQ2VSWuPqbm2Umoda50c0bg9NGEUYEbvGVi+ttA@mail.gmail.com>
	<CABsWbRbq9a48q+dzbsehYPM6-5QmFP=EttFBwEa1r7O6OjP3WA@mail.gmail.com>
	<1A821973-1FDC-41E4-8484-D89A4B3ABF4A@jerris.com>
	<CAHtxdDe8POGEPFs0auaUuGJwZbudkJ42dBs6qDozSGu_a8CLHA@mail.gmail.com>
	<73BF366B-8CB1-4337-81B7-5237CD800147@jerris.com>
Message-ID: <CAHtxdDfuv8U_UX1tvJmEq_vynGOeYMomDZPj77v99Pa3OLDAsA@mail.gmail.com>

Michael, thank very much. It is what I am searching.

On Fri, May 22, 2015 at 6:52 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On May 22, 2015, at 12:55 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> Michael, I think over how to organize the control of the call recording on
> OpenSips from FS dialplan (Scheme where FS is managing callcontrol,
> OpenSIPs RTP media processing)
> As an option I think to make it through
> 1) additional headers in the messages 100, 180, 183, 200 (start
> recording). Stop recording through RE-INVITE;
> 2) as a separate SIP INFO message (start, stop recording - more elegant
> solution).
>
> Tell me, please, as much as possible
> 1) send a custom SIP INFO message
>
>
> from api interface: uuid_send_info_function
> from dialplan: send_info
>
> 2) how custom headers can be added to 100, 180, 183, 200 messages
>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/Sofia+SIP+Stack#SofiaSIPStack-ChannelVariables
>
>
>
> On Thu, May 21, 2015 at 11:23 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Or its totally possible that I have asked for specifics multiple times
>> now and you have ignored me every single time.  YOU CAN DO THIS IN
>> FREESWITCH BUT I CAN NOT TELL YOU HOW UNLESS YOU TELL ME WHAT SIP MESSAGES
>> YOU ARE TRYING TO SEND IN WHAT SITUATIONS!!!!!!!!
>>
>>
>> On May 21, 2015, at 4:14 PM, Naveen Tamanam <naveen32india at gmail.com>
>> wrote:
>>
>> Giovanni,
>>
>> Thank you very much, my intention is to know the way  to
>> handle/manipulate (low level) sip packets with freeswitch if it possible.
>> As per your reply this is obvious way to use Kamailio or OpenSIPS
>> infront of FS.
>>
>> On Thu, May 21, 2015 at 4:31 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> Naveen,
>>>
>>> Please take note that FreeSWITCH is not intended to let the user to
>>> manipulate in arbitrary way SIP dialogs.
>>>
>>> You can do that for specific purposes and in.specific cases, within
>>> specific boundaries.
>>>
>>> If you are looking to interact with SIP directly and freely, you may
>>> want to look at Kamailio and OpenSIPS.
>>>
>>> You can put one of them in front of FreeSWITCH, and you can cross
>>> command them, for example via.lua scripting (both of them and FreeSWITCH
>>> can be scripted in.lua).
>>>
>>> sent from my mobile,
>>> Giovanni Maruzzelli
>>> cell: +39 347 266 56 18
>>> On May 20, 2015 3:40 PM, "Naveen Tamanam" <naveen32india at gmail.com>
>>> wrote:
>>>
>>>> I am aware of uuid_hangup. Indeed my intention to know the way to send
>>>> (low level)sip messages
>>>> from fs console for a selected user.
>>>>
>>>>
>>>> On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com>
>>>> wrote:
>>>>
>>>>>  I would like to reject the call when it ringing from the fs console.
>>>>>
>>>>>
>>>>> uuid_hangup
>>>>>
>>>>>
>>>>> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> I am trying to do the following,  I would like to reject the call
>>>>>> when it ringing from the fs console.
>>>>>> And second thing is I am pretty much eager to know the way to send
>>>>>> sip(signaling) message manually for a
>>>>>> selected channel from fs console.
>>>>>>
>>>>>>
>>>>>> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com>
>>>>>> wrote:
>>>>>>
>>>>>>> respond is one way, what exact message are you trying to send, and
>>>>>>> at what point in the call.  There are capabilites to trigger re-invites in
>>>>>>> some situations, transfer, send notify or info or message.  It depends on
>>>>>>> what exactly you are trying to do.
>>>>>>>
>>>>>>>
>>>>>>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> I am wondering how to send sip signaling message from the fs console
>>>>>>> for the particular user/caller.
>>>>>>> I found  respond dialplan application to send sip message back to
>>>>>>> the caller.
>>>>>>> Like
>>>>>>>
>>>>>>> <action application="respond" data="480 Try again later"/>
>>>>>>>
>>>>>>> ?Is there any way to send sip message back to the caller. One use
>>>>>>> case is call rejection or playing with SIP?
>>>>>>>
>>>>>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From s.safarov at gmail.com  Fri May 22 21:28:49 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 22 May 2015 20:28:49 +0300
Subject: [Freeswitch-users] Google Hangouts Integration
In-Reply-To: <CANBm6Sw0FNwo1Gw5sS-AmwcjBfM5x3S9-YtZ4n4GBvCgbL3paA@mail.gmail.com>
References: <CANBm6Sw0FNwo1Gw5sS-AmwcjBfM5x3S9-YtZ4n4GBvCgbL3paA@mail.gmail.com>
Message-ID: <CAHtxdDcUkGo+1euANa0ir+gpu4MDTgRYPEj9VENWD8EUSX8JeA@mail.gmail.com>

I think you can call to Hangouts user via mod_dingaling endpoint.

On Thu, May 21, 2015 at 7:10 PM, Mitchelle Johnson <mitchelle.bit at gmail.com>
wrote:

> Hi all,
>
> Like skype can be integrated with FreeSWITCH. Is there a way to integrate
> google Hangouts also?
>
> Thanks,
> Mitchelle
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From naveen32india at gmail.com  Fri May 22 21:57:38 2015
From: naveen32india at gmail.com (Naveen Tamanam)
Date: Fri, 22 May 2015 23:27:38 +0530
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <1A821973-1FDC-41E4-8484-D89A4B3ABF4A@jerris.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
	<CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
	<CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
	<CALXCt0r4RDeQQ2VSWuPqbm2Umoda50c0bg9NGEUYEbvGVi+ttA@mail.gmail.com>
	<CABsWbRbq9a48q+dzbsehYPM6-5QmFP=EttFBwEa1r7O6OjP3WA@mail.gmail.com>
	<1A821973-1FDC-41E4-8484-D89A4B3ABF4A@jerris.com>
Message-ID: <CABsWbRa-8=TD_jg21Ptf7kfB91mxaN=16P8AbeUYg9iBAU65NQ@mail.gmail.com>

Hi Michael,

Thanks for your concern. Many times I fell freeswitch commands are crafted
in such a way we don't need to deal with low level sip manipulation.

I am trying to know about,
Hang up the call with specific reason. Putting the sip header (Reason:
Q.850;cause=21;text="CALL_REJECTED")
Using custom user agent string when it is required.
Only putting commands we want in the sip allow header.




On Fri, May 22, 2015 at 1:53 AM, Michael Jerris <mike at jerris.com> wrote:

> Or its totally possible that I have asked for specifics multiple times now
> and you have ignored me every single time.  YOU CAN DO THIS IN FREESWITCH
> BUT I CAN NOT TELL YOU HOW UNLESS YOU TELL ME WHAT SIP MESSAGES YOU ARE
> TRYING TO SEND IN WHAT SITUATIONS!!!!!!!!
>
>
> On May 21, 2015, at 4:14 PM, Naveen Tamanam <naveen32india at gmail.com>
> wrote:
>
> Giovanni,
>
> Thank you very much, my intention is to know the way  to handle/manipulate
> (low level) sip packets with freeswitch if it possible. As per your reply
> this is obvious way to use Kamailio or OpenSIPS infront of FS.
>
> On Thu, May 21, 2015 at 4:31 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Naveen,
>>
>> Please take note that FreeSWITCH is not intended to let the user to
>> manipulate in arbitrary way SIP dialogs.
>>
>> You can do that for specific purposes and in.specific cases, within
>> specific boundaries.
>>
>> If you are looking to interact with SIP directly and freely, you may want
>> to look at Kamailio and OpenSIPS.
>>
>> You can put one of them in front of FreeSWITCH, and you can cross command
>> them, for example via.lua scripting (both of them and FreeSWITCH can be
>> scripted in.lua).
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On May 20, 2015 3:40 PM, "Naveen Tamanam" <naveen32india at gmail.com>
>> wrote:
>>
>>> I am aware of uuid_hangup. Indeed my intention to know the way to send
>>> (low level)sip messages
>>> from fs console for a selected user.
>>>
>>>
>>> On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com>
>>> wrote:
>>>
>>>>  I would like to reject the call when it ringing from the fs console.
>>>>
>>>>
>>>> uuid_hangup
>>>>
>>>>
>>>> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com>
>>>> wrote:
>>>>
>>>>> I am trying to do the following,  I would like to reject the call when
>>>>> it ringing from the fs console.
>>>>> And second thing is I am pretty much eager to know the way to send
>>>>> sip(signaling) message manually for a
>>>>> selected channel from fs console.
>>>>>
>>>>>
>>>>> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>>> respond is one way, what exact message are you trying to send, and at
>>>>>> what point in the call.  There are capabilites to trigger re-invites in
>>>>>> some situations, transfer, send notify or info or message.  It depends on
>>>>>> what exactly you are trying to do.
>>>>>>
>>>>>>
>>>>>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I am wondering how to send sip signaling message from the fs console
>>>>>> for the particular user/caller.
>>>>>> I found  respond dialplan application to send sip message back to the
>>>>>> caller.
>>>>>> Like
>>>>>>
>>>>>> <action application="respond" data="480 Try again later"/>
>>>>>>
>>>>>> ?Is there any way to send sip message back to the caller. One use
>>>>>> case is call rejection or playing with SIP?
>>>>>>
>>>>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regards,
Naveen Tamanam
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From mike at jerris.com  Sat May 23 00:19:24 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 22 May 2015 16:19:24 -0400
Subject: [Freeswitch-users] How to send SIP signaling message from fs
	console.
In-Reply-To: <CABsWbRa-8=TD_jg21Ptf7kfB91mxaN=16P8AbeUYg9iBAU65NQ@mail.gmail.com>
References: <CABsWbRY231efomAOHp-7TONEQNE4+FU=DjOR_aNd228WOaE_bA@mail.gmail.com>
	<DA91332F-FCAE-48E1-A943-86A64DE49947@jerris.com>
	<CABsWbRZYiyHq1Ux1uyOPSQq28rMPUbNNmagMRNNxkT3YHVZ97Q@mail.gmail.com>
	<CAFiqYun2PkZYFvFPjE87YJxM9pKo7JEJQsEGf4UhM2dPtgAE_g@mail.gmail.com>
	<CABsWbRatXgfH9pXRvBesH+on9aJm3Pv=W=HLJoDJoB0eAE7nTg@mail.gmail.com>
	<CALXCt0r4RDeQQ2VSWuPqbm2Umoda50c0bg9NGEUYEbvGVi+ttA@mail.gmail.com>
	<CABsWbRbq9a48q+dzbsehYPM6-5QmFP=EttFBwEa1r7O6OjP3WA@mail.gmail.com>
	<1A821973-1FDC-41E4-8484-D89A4B3ABF4A@jerris.com>
	<CABsWbRa-8=TD_jg21Ptf7kfB91mxaN=16P8AbeUYg9iBAU65NQ@mail.gmail.com>
Message-ID: <5BB4AA47-2998-4ABB-BF0A-260465575CCD@jerris.com>


> On May 22, 2015, at 1:57 PM, Naveen Tamanam <naveen32india at gmail.com> wrote:
> 
> Hi Michael, 
> 
> Thanks for your concern. Many times I fell freeswitch commands are crafted in such a way we don't need to deal with low level sip manipulation. 
> 
> I am trying to know about, 
> Hang up the call with specific reason. Putting the sip header (Reason: Q.850;cause=21;text="CALL_REJECTED")

hangup app or uuid_hangup api

> Using custom user agent string when it is required. 

sofia profile param user-agent-string

> Only putting commands we want in the sip allow header. 

The allow string will always include:

INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, NOTIFY

UPDATE, REGISTER, REFER, and PRACK are controlled by sofia profile params

I am not sure when you would want to control this further...

> 
> 
> 
> 
> On Fri, May 22, 2015 at 1:53 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Or its totally possible that I have asked for specifics multiple times now and you have ignored me every single time.  YOU CAN DO THIS IN FREESWITCH BUT I CAN NOT TELL YOU HOW UNLESS YOU TELL ME WHAT SIP MESSAGES YOU ARE TRYING TO SEND IN WHAT SITUATIONS!!!!!!!!
> 
> 
>> On May 21, 2015, at 4:14 PM, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> 
>> Giovanni, 
>> 
>> Thank you very much, my intention is to know the way  to handle/manipulate (low level) sip packets with freeswitch if it possible. As per your reply this is obvious way to use Kamailio or OpenSIPS infront of FS. 
>> 
>> On Thu, May 21, 2015 at 4:31 PM, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>> Naveen,
>> 
>> Please take note that FreeSWITCH is not intended to let the user to manipulate in arbitrary way SIP dialogs.
>> 
>> You can do that for specific purposes and in.specific cases, within specific boundaries.
>> 
>> If you are looking to interact with SIP directly and freely, you may want to look at Kamailio and OpenSIPS.
>> 
>> You can put one of them in front of FreeSWITCH, and you can cross command them, for example via.lua scripting (both of them and FreeSWITCH can be scripted in.lua).
>> 
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> 
>> On May 20, 2015 3:40 PM, "Naveen Tamanam" <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> I am aware of uuid_hangup. Indeed my intention to know the way to send (low level)sip messages 
>> from fs console for a selected user. 
>> 
>> 
>> On Tue, May 19, 2015 at 2:51 AM, Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
>>  I would like to reject the call when it ringing from the fs console. 
>> 
>> uuid_hangup
>>  
>> 
>> On 18 May 2015 at 21:59, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>> I am trying to do the following,  I would like to reject the call when it ringing from the fs console. 
>> And second thing is I am pretty much eager to know the way to send sip(signaling) message manually for a 
>> selected channel from fs console. 
>> 
>> 
>> On Tue, May 19, 2015 at 2:19 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> respond is one way, what exact message are you trying to send, and at what point in the call.  There are capabilites to trigger re-invites in some situations, transfer, send notify or info or message.  It depends on what exactly you are trying to do.
>> 
>> 
>>> On May 18, 2015, at 4:30 PM, Naveen Tamanam <naveen32india at gmail.com <mailto:naveen32india at gmail.com>> wrote:
>>> 
>>> Hi, 
>>> 
>>> I am wondering how to send sip signaling message from the fs console for the particular user/caller. 
>>> I found  respond dialplan application to send sip message back to the caller. 
>>> Like 
>>> <action application="respond" data="480 Try again later"/>
>>> ?Is there any way to send sip message back to the caller. One use case is call rejection or playing with SIP?
>>> 
> 
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> Thanks & Regards,
> Naveen Tamanam
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mitchelle.bit at gmail.com  Sat May 23 11:21:28 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Sat, 23 May 2015 12:51:28 +0530
Subject: [Freeswitch-users] XML CDR help
In-Reply-To: <CANBm6SwqcTQR4LDukb7rAdEZsGvDQHs1mCbiVG99tmQrf48-hA@mail.gmail.com>
References: <CANBm6SzMgrb8qNBrdU-X9WHpgoSStVg04NMtusfohGcvGsW-5g@mail.gmail.com>
	<CANBm6SwqcTQR4LDukb7rAdEZsGvDQHs1mCbiVG99tmQrf48-hA@mail.gmail.com>
Message-ID: <CANBm6SzvYx-=Z5fZCTW3r28Rj49m1EYpGR9agUfmXyNsvLJQwA@mail.gmail.com>

Hi all,
Please help me with my question. When processing XML CDRs sometimes(not
always) the callee_id_number tag is left blank. Please tell me why does
this happen? The pastebin line for the XML CDR is
http://pastebin.com/fAS7Gu2a

Thanks,
Mitchelle

On Fri, May 22, 2015 at 3:18 PM, Mitchelle Johnson <mitchelle.bit at gmail.com>
wrote:

> Please find the XML file on pastebin. The link is
> http://pastebin.com/fAS7Gu2a
>
> Thanks,
> Mitchelle
>
> On Fri, May 22, 2015 at 3:15 PM, Mitchelle Johnson <
> mitchelle.bit at gmail.com> wrote:
>
>> Hi,
>> When processing XML CDRs sometimes(not all) the callee_id_number tag is
>> left blank. Please tell me why does this happen?
>>
>> Thanks,
>> Mitchelle
>>
>
>
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From mitchelle.bit at gmail.com  Sat May 23 11:22:03 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Sat, 23 May 2015 12:52:03 +0530
Subject: [Freeswitch-users] Google Hangouts Integration
In-Reply-To: <CAHtxdDcUkGo+1euANa0ir+gpu4MDTgRYPEj9VENWD8EUSX8JeA@mail.gmail.com>
References: <CANBm6Sw0FNwo1Gw5sS-AmwcjBfM5x3S9-YtZ4n4GBvCgbL3paA@mail.gmail.com>
	<CAHtxdDcUkGo+1euANa0ir+gpu4MDTgRYPEj9VENWD8EUSX8JeA@mail.gmail.com>
Message-ID: <CANBm6SwpUHntcPtkzTzNo-yHTp8K=pMVvkroystjXM7edr0L5g@mail.gmail.com>

Thank you all for your answers.

Regards,
Mitchelle

On Fri, May 22, 2015 at 10:58 PM, Sergey Safarov <s.safarov at gmail.com>
wrote:

> I think you can call to Hangouts user via mod_dingaling endpoint.
>
> On Thu, May 21, 2015 at 7:10 PM, Mitchelle Johnson <
> mitchelle.bit at gmail.com> wrote:
>
>> Hi all,
>>
>> Like skype can be integrated with FreeSWITCH. Is there a way to integrate
>> google Hangouts also?
>>
>> Thanks,
>> Mitchelle
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From carlosj.gf at gmail.com  Sat May 23 23:39:09 2015
From: carlosj.gf at gmail.com (=?ISO-8859-1?Q?Carlos_Gonz=E1lez_Florido?=)
Date: Sat, 23 May 2015 21:39:09 +0200
Subject: [Freeswitch-users] Video always muted in conference (fs_video2)
In-Reply-To: <7B131C27-7233-4E88-BB9E-E01BE588DE82@jerris.com>
References: <CAFKjMAYjKz32rE-G_aA639J5b27eNwnRAME31La25W9KiWbagA@mail.gmail.com>
	<7B131C27-7233-4E88-BB9E-E01BE588DE82@jerris.com>
Message-ID: <CAFKjMAZ30sxNtAXU1iqksuNwM53gzQaqFQor7vhVHhg-3VBn5w@mail.gmail.com>

Ok, tell me if I can help with the testing.
Is there a way to disconnect ipv6?

On Fri, May 22, 2015 at 6:04 PM, Michael Jerris <mike at jerris.com> wrote:

> I don't think its actually muted.  We are chasing down an issue that
> sounds just like this.  It might be related to ipv6 but we are still
> looking in to it.
>
> > On May 22, 2015, at 8:40 AM, Carlos Gonz?lez Florido <
> carlosj.gf at gmail.com> wrote:
> >
> > Hello,
> >
> > I'm testing the impressive fs_video2 branch, but I have the following
> problems:
> >
> > - some participants have their video always muted (and their camera is
> on and working for Hangouts, for example).
> > - other participants start the same (video muted), but after 3-5 minutes
> the video is automatically unmuted.
> > - if we do a layout change (using fs_cli), all participants (having
> video muted or unmuted at that moment) go to muted video for the rest of
> the conference.
> >
> > What is the reason for fs to automatically mute the video? Is this
> expected or does it look like a bug? Can this behaviour be tuned or
> disconnected?
> >
> > Thank you,
> > Carlos Gonzalez
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dm at dwide.com  Sun May 24 00:22:58 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Sun, 24 May 2015 00:22:58 +0400
Subject: [Freeswitch-users] Difficult extension case
Message-ID: <5560E1A2.2050806@dwide.com>

Hello All!

I wish to implement one extension scenario, could help me?


- Play prompt (its easy)
- Play ringback music
- Bridge to one or multiple numbers
- On answer:
         - Play BEEP to incoming leg (stop music, play BEEP means call 
success, starting talks)
         - Play "IMPORTANT CALL!!!" to new outgoing leg
         - Connect incoming and outgoing legs together
- On fail (timeout):
         - Play "Please recall later"



I don't understand how can play on answer separately each leg

Thank you


From jonlederman at gmail.com  Sun May 24 00:57:35 2015
From: jonlederman at gmail.com (Jon Lederman)
Date: Sat, 23 May 2015 16:57:35 -0400
Subject: [Freeswitch-users] RTCP and WebRTC
Message-ID: <D00D93FE-03FC-444A-B5CD-1E9D9DA93866@gmail.com>

Hi,

I am trying to use WebRTC with Freeswitch.  WebRTC relies upon RTCP for bandwidth estimation.  I have tried to get forwarding of RTCP packets working with Freeswitch to make this work.  

However, a more fundamental issue seems to be occurring in that Freeswitch is not able to process the RTCP messages arriving from the WebRTC client.  I have traced the problem to the function process_rtcp_frame in switch_rtp.c.  In that function, the version of RTP appearing in the RTCP packets is 0, not 2, which is required and the packets are never processed (i.e., no fresh frames).  I am not sure what the cause of this is?  Could it be related to muxing of RTP/RTCP, which webrtc utilizes?  Any thoughts on resolving this would be greatly appreciated.

Thanks.

-Jon

From ssinyagin at gmail.com  Sun May 24 03:31:23 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 24 May 2015 01:31:23 +0200
Subject: [Freeswitch-users] Difficult extension case
In-Reply-To: <5560E1A2.2050806@dwide.com>
References: <5560E1A2.2050806@dwide.com>
Message-ID: <CACOYK=iukG6_nhhbEZD7t8CUcsTivESZQZL++dfzop2pYQaVMw@mail.gmail.com>

it's easy if you use an external program talking to FreeSWITCH via its API.

I actually implemented two prototypes which do approximately the same
as you described:

https://github.com/xlab1/freeswitch_secretary_bug
this one uses mod_perl and is executed within FreeSWITCH process, so
you need to watch out about CPU and memory usage and overall
stability. See the Jira ticket for details.


https://github.com/xlab1/go-fs-secretary-prototype
This is a prototype which implements exactly the same scenario, but is
controlled by an external process, so it's more stable and scalable.
It talks to FreeSWITCH via the Event Socket interface.








On Sat, May 23, 2015 at 10:22 PM, Dmitry Mordovin <dm at dwide.com> wrote:
> Hello All!
>
> I wish to implement one extension scenario, could help me?
>
>
> - Play prompt (its easy)
> - Play ringback music
> - Bridge to one or multiple numbers
> - On answer:
>          - Play BEEP to incoming leg (stop music, play BEEP means call
> success, starting talks)
>          - Play "IMPORTANT CALL!!!" to new outgoing leg
>          - Connect incoming and outgoing legs together
> - On fail (timeout):
>          - Play "Please recall later"
>
>
>
> I don't understand how can play on answer separately each leg
>
> 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


From mishehu at freeswitch.org  Sun May 24 03:40:46 2015
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Sat, 23 May 2015 18:40:46 -0500
Subject: [Freeswitch-users] XML CDR help
In-Reply-To: <CANBm6SzvYx-=Z5fZCTW3r28Rj49m1EYpGR9agUfmXyNsvLJQwA@mail.gmail.com>
References: <CANBm6SzMgrb8qNBrdU-X9WHpgoSStVg04NMtusfohGcvGsW-5g@mail.gmail.com>	<CANBm6SwqcTQR4LDukb7rAdEZsGvDQHs1mCbiVG99tmQrf48-hA@mail.gmail.com>
	<CANBm6SzvYx-=Z5fZCTW3r28Rj49m1EYpGR9agUfmXyNsvLJQwA@mail.gmail.com>
Message-ID: <55610FFE.2080908@freeswitch.org>

That example is a single-legged call. You are only ever interacting with 
one channel, so there's no callee party in the first place.  When you 
bridge channels you'll see records that contain callee.

-Yossi

On 05/23/2015 02:21 AM, Mitchelle Johnson wrote:
> Hi all,
> Please help me with my question. When processing XML CDRs 
> sometimes(not always) the callee_id_number tag is left blank. Please 
> tell me why does this happen? The pastebin line for the XML CDR is 
> http://pastebin.com/fAS7Gu2a
>
> Thanks,
> Mitchelle
>
> On Fri, May 22, 2015 at 3:18 PM, Mitchelle Johnson 
> <mitchelle.bit at gmail.com <mailto:mitchelle.bit at gmail.com>> wrote:
>
>     Please find the XML file on pastebin. The link is
>     http://pastebin.com/fAS7Gu2a
>
>     Thanks,
>     Mitchelle
>
>     On Fri, May 22, 2015 at 3:15 PM, Mitchelle Johnson
>     <mitchelle.bit at gmail.com <mailto:mitchelle.bit at gmail.com>> wrote:
>
>         Hi,
>         When processing XML CDRs sometimes(not all) the
>         callee_id_number tag is left blank. Please tell me why does
>         this happen?
>
>         Thanks,
>         Mitchelle
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Sun May 24 09:18:28 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 24 May 2015 00:18:28 -0500
Subject: [Freeswitch-users] RTCP and WebRTC
In-Reply-To: <D00D93FE-03FC-444A-B5CD-1E9D9DA93866@gmail.com>
References: <D00D93FE-03FC-444A-B5CD-1E9D9DA93866@gmail.com>
Message-ID: <CAKbxfG-SeHJb8Qz_xgM7B+WSvVLAQ5cWEx=sKaCsD23_ctCG=Q@mail.gmail.com>

Check out fs_video2 branch or wait for 1.6 merge later this week.  All of
that code is drasticly different now.

On Saturday, May 23, 2015, Jon Lederman <jonlederman at gmail.com> wrote:

> Hi,
>
> I am trying to use WebRTC with Freeswitch.  WebRTC relies upon RTCP for
> bandwidth estimation.  I have tried to get forwarding of RTCP packets
> working with Freeswitch to make this work.
>
> However, a more fundamental issue seems to be occurring in that Freeswitch
> is not able to process the RTCP messages arriving from the WebRTC client.
> I have traced the problem to the function process_rtcp_frame in
> switch_rtp.c.  In that function, the version of RTP appearing in the RTCP
> packets is 0, not 2, which is required and the packets are never processed
> (i.e., no fresh frames).  I am not sure what the cause of this is?  Could
> it be related to muxing of RTP/RTCP, which webrtc utilizes?  Any thoughts
> on resolving this would be greatly appreciated.
>
> Thanks.
>
> -Jon
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From dm at dwide.com  Sun May 24 18:33:31 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Sun, 24 May 2015 18:33:31 +0400
Subject: [Freeswitch-users] Difficult extension case
In-Reply-To: <CACOYK=iukG6_nhhbEZD7t8CUcsTivESZQZL++dfzop2pYQaVMw@mail.gmail.com>
References: <5560E1A2.2050806@dwide.com>
	<CACOYK=iukG6_nhhbEZD7t8CUcsTivESZQZL++dfzop2pYQaVMw@mail.gmail.com>
Message-ID: <5561E13B.3050103@dwide.com>

Hello

Problem with "play on answer" solved

<action application="bridge" data="{api_on_answer_1=''sleep 
1000'',api_on_answer_2=''uuid_broadcast ${uuid} ivr/ivr-call_from.wav 
aleg''}sofia/gateway/sipnet/1009 at sipnet"/>

Thanks all!


On 05/24/2015 03:31 AM, Stanislav Sinyagin wrote:
> it's easy if you use an external program talking to FreeSWITCH via its API.
>
> I actually implemented two prototypes which do approximately the same
> as you described:
>
> https://github.com/xlab1/freeswitch_secretary_bug
> this one uses mod_perl and is executed within FreeSWITCH process, so
> you need to watch out about CPU and memory usage and overall
> stability. See the Jira ticket for details.
>
>
> https://github.com/xlab1/go-fs-secretary-prototype
> This is a prototype which implements exactly the same scenario, but is
> controlled by an external process, so it's more stable and scalable.
> It talks to FreeSWITCH via the Event Socket interface.
>
>
>
>
>
>
>
>
> On Sat, May 23, 2015 at 10:22 PM, Dmitry Mordovin <dm at dwide.com> wrote:
>> Hello All!
>>
>> I wish to implement one extension scenario, could help me?
>>
>>
>> - Play prompt (its easy)
>> - Play ringback music
>> - Bridge to one or multiple numbers
>> - On answer:
>>           - Play BEEP to incoming leg (stop music, play BEEP means call
>> success, starting talks)
>>           - Play "IMPORTANT CALL!!!" to new outgoing leg
>>           - Connect incoming and outgoing legs together
>> - On fail (timeout):
>>           - Play "Please recall later"
>>
>>
>>
>> I don't understand how can play on answer separately each leg
>>
>> 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



From mike at jerris.com  Mon May 25 08:38:35 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 May 2015 00:38:35 -0400
Subject: [Freeswitch-users] Video always muted in conference (fs_video2)
In-Reply-To: <CAFKjMAZ30sxNtAXU1iqksuNwM53gzQaqFQor7vhVHhg-3VBn5w@mail.gmail.com>
References: <CAFKjMAYjKz32rE-G_aA639J5b27eNwnRAME31La25W9KiWbagA@mail.gmail.com>
	<7B131C27-7233-4E88-BB9E-E01BE588DE82@jerris.com>
	<CAFKjMAZ30sxNtAXU1iqksuNwM53gzQaqFQor7vhVHhg-3VBn5w@mail.gmail.com>
Message-ID: <CAEwTsAgsVXvE5Ym9TWvbSvd9i7gCgNWwmp-2adxYKug0RHHKBA@mail.gmail.com>

Some fixes were pushed for this on Friday.  Try out latest code and see if
that works any better for you?

On Saturday, May 23, 2015, Carlos Gonz?lez Florido <carlosj.gf at gmail.com>
wrote:

> Ok, tell me if I can help with the testing.
> Is there a way to disconnect ipv6?
>
> On Fri, May 22, 2015 at 6:04 PM, Michael Jerris <mike at jerris.com
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>
>> I don't think its actually muted.  We are chasing down an issue that
>> sounds just like this.  It might be related to ipv6 but we are still
>> looking in to it.
>>
>> > On May 22, 2015, at 8:40 AM, Carlos Gonz?lez Florido <
>> carlosj.gf at gmail.com
>> <javascript:_e(%7B%7D,'cvml','carlosj.gf at gmail.com');>> wrote:
>> >
>> > Hello,
>> >
>> > I'm testing the impressive fs_video2 branch, but I have the following
>> problems:
>> >
>> > - some participants have their video always muted (and their camera is
>> on and working for Hangouts, for example).
>> > - other participants start the same (video muted), but after 3-5
>> minutes the video is automatically unmuted.
>> > - if we do a layout change (using fs_cli), all participants (having
>> video muted or unmuted at that moment) go to muted video for the rest of
>> the conference.
>> >
>> > What is the reason for fs to automatically mute the video? Is this
>> expected or does it look like a bug? Can this behaviour be tuned or
>> disconnected?
>> >
>> > Thank you,
>> > Carlos Gonzalez
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From steveayre at gmail.com  Mon May 25 12:51:09 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 25 May 2015 09:51:09 +0100
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
Message-ID: <CAFiqYum4if-B8NGFu7PXjFpJZnPuFGy8XhWscM7p7q6BwZiqGg@mail.gmail.com>

The values to actual use are negotiated during the call invite, and your
trace shows it's using 2 minutes.
Session-Expires: 120;refresher=uac
Min-SE: 120

Just as an idea because you didn't send both invite and re-invite...
perhaps the SDP body is different in the reinvite without the version
number having changed. If so it may be
https://freeswitch.org/jira/browse/FS-7040

On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> But isn't that based on the session-timeout param which defaults to 30
> min?  My re-invites occur much sooner than 30 min into a call.  Or does
> session-timeout param only control sessions initiated by FS while incoming
> sessions use the minimum-session-expires param if it's not explicitly
> passed by the session initiator?
>
> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> session timer
>>
>>
>> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>
>>> Thanks Michael, I'll see if we can do that!
>>>
>>> So, is the re-INVITE legit and the problem is that JsSip does not
>>> respond to it?  Still, I am curious what is triggering the re-INVITE.
>>>
>>> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> I think the sip.js guys fixed this issue when they forked jssip.  I'd
>>>> suggest using that.
>>>>
>>>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>>> wrote:
>>>> >
>>>> > Hi guys,
>>>> >
>>>> > Several weeks ago I started getting an occasional problem where FS is
>>>> sending an INVITE to the other side in the middle of a call, the other side
>>>> does not respond and FS hangs up the leg.  Below is the relevant log.  The
>>>> user experience is that they keep talking and hearing each other up to the
>>>> very end.  I have a recording of that call, so can confirm.
>>>> >
>>>> > The call uses WebRTC and is originated by JsSip from Chrome.  Then
>>>> the user is put into a conference but I doubt it's relevant in this case
>>>> since the INVITE and disconnect are not happening from mod_conference
>>>> >
>>>> > I suspect it's a re-INVITE but what triggers FS to send it?  I
>>>> couldn't find anything in the logs that could shed light.
>>>> >
>>>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>>>> >
>>>> ------------------------------------------------------------------------
>>>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>>>> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>>>> >    Max-Forwards: 70
>>>> >    From:
>>>> <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>>>> >;tag=KQecUSr12rSQp
>>>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>>>> >    Call-ID: g8980rbrbk2t45oj5mru
>>>> >    CSeq: 75703945 INVITE
>>>> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@
>>>> ##.##.###.###:5080;transport=udp>
>>>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>>>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>> >    Supported: timer, path, replaces
>>>> >    Session-Expires: 120;refresher=uac
>>>> >    Min-SE: 120
>>>> >    Content-Type: application/sdp
>>>> >    Content-Disposition: session
>>>> >    Content-Length: 825
>>>> >
>>>> >    v=0
>>>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>>>> >    s=FreeSWITCH
>>>> >    c=IN IP4 ##.##.##.##
>>>> >    t=0 0
>>>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>> >    m=audio 22670 RTP/SAVPF 9 126 106
>>>> >    a=rtpmap:9 G722/8000
>>>> >    a=rtpmap:126 telephone-event/8000
>>>> >    a=rtpmap:106 CN/8000
>>>> >    a=ptime:20
>>>> >    a=fingerprint:sha-256
>>>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>>>> >    a=rtcp-mux
>>>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>>>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>>>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>>>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>>>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>>>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>>>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host
>>>> generation 0
>>>> >
>>>> ------------------------------------------------------------------------
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>> entering state [calling][0]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>> entering state [terminating][503]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
>>>> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [DEBUG] switch_channel.c:3242 Send signal
>>>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>> [DEBUG] switch_core_session.c:1396 Send signal
>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] mod_conference.c:5057 Channel leaving conference, cause:
>>>> NORMAL_TEMPORARY_FAILURE
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid
>>>> skip receive message [UNBRIDGE] (channel is hungup already)
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_media.c:7772
>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>> [REFER_EVENT] (channel is hungup already)
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
>>>> Restore previous codec G722:9.
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_session.c:2901
>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:535
>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to
>>>> sleep
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:472
>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>>>> CS_HANGUP
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:735
>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE
>>>> -> HANGUP
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:737
>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>> hanging up, cause: NORMAL_TEMPORARY_FAILURE
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:60
>>>> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause:
>>>> NORMAL_TEMPORARY_FAILURE
>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>> [DEBUG] switch_core_state_machine.c:737
>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to
>>>> sleep
>>>> >
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From dm at dwide.com  Mon May 25 17:59:50 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Mon, 25 May 2015 17:59:50 +0400
Subject: [Freeswitch-users] no ringing then calling local user
Message-ID: <55632AD6.50306@dwide.com>

Hello

My extension

<extension name="test10">
     <condition field="destination_number" expression="^2222$">
       <action application="answer"/>
       <action application="sleep" data="1500"/>
       <action application="set" data="hangup_after_bridge=true" />
       <action application="bridge" 
data="sofia/internal/1004%${domain_name}"/>
       <action application="hangup"/>
     </condition>
</extension>

 From fs_cli I originating call

originate 
{ignore_early_media=true}sofia/gateway/sipnet/12126248024\@sipnet 1004

Soft phone 1004 registered and ringing then incoming call, but 
12126248024 (aleg) is silence!

In debug looks good

2015-05-25 17:46:31.990988 [DEBUG] sofia.c:6627 Channel 
sofia/internal/1004 entering state [proceeding][180]
2015-05-25 17:46:31.990988 [NOTICE] sofia.c:6729 Ring-Ready 
sofia/internal/1004!
2015-05-25 17:46:31.990988 [DEBUG] switch_channel.c:3297 
(sofia/internal/1004) Callstate Change DOWN -> RINGING

May I forget something?





From dm at dwide.com  Mon May 25 18:02:40 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Mon, 25 May 2015 18:02:40 +0400
Subject: [Freeswitch-users] no ringing then calling local user
In-Reply-To: <55632AD6.50306@dwide.com>
References: <55632AD6.50306@dwide.com>
Message-ID: <55632B80.7070108@dwide.com>

Correction

originate
{ignore_early_media=true}sofia/gateway/sipnet/12126248024\@sipnet 2222



On 05/25/2015 05:59 PM, Dmitry Mordovin wrote:
> Hello
>
> My extension
>
> <extension name="test10">
>       <condition field="destination_number" expression="^2222$">
>         <action application="answer"/>
>         <action application="sleep" data="1500"/>
>         <action application="set" data="hangup_after_bridge=true" />
>         <action application="bridge"
> data="sofia/internal/1004%${domain_name}"/>
>         <action application="hangup"/>
>       </condition>
> </extension>
>
>   From fs_cli I originating call
>
> originate
> {ignore_early_media=true}sofia/gateway/sipnet/12126248024\@sipnet 1004
>
> Soft phone 1004 registered and ringing then incoming call, but
> 12126248024 (aleg) is silence!
>
> In debug looks good
>
> 2015-05-25 17:46:31.990988 [DEBUG] sofia.c:6627 Channel
> sofia/internal/1004 entering state [proceeding][180]
> 2015-05-25 17:46:31.990988 [NOTICE] sofia.c:6729 Ring-Ready
> sofia/internal/1004!
> 2015-05-25 17:46:31.990988 [DEBUG] switch_channel.c:3297
> (sofia/internal/1004) Callstate Change DOWN -> RINGING
>
> May I forget something?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 dm at dwide.com  Mon May 25 18:11:14 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Mon, 25 May 2015 18:11:14 +0400
Subject: [Freeswitch-users] no ringing then calling local user
In-Reply-To: <55632B80.7070108@dwide.com>
References: <55632AD6.50306@dwide.com> <55632B80.7070108@dwide.com>
Message-ID: <55632D82.8080508@dwide.com>

Excuse me, solved

On 05/25/2015 06:02 PM, Dmitry Mordovin wrote:
> Correction
>
> originate
> {ignore_early_media=true}sofia/gateway/sipnet/12126248024\@sipnet 2222
>
>
>
> On 05/25/2015 05:59 PM, Dmitry Mordovin wrote:
>> Hello
>>
>> My extension
>>
>> <extension name="test10">
>>        <condition field="destination_number" expression="^2222$">
>>          <action application="answer"/>
>>          <action application="sleep" data="1500"/>
>>          <action application="set" data="hangup_after_bridge=true" />
>>          <action application="bridge"
>> data="sofia/internal/1004%${domain_name}"/>
>>          <action application="hangup"/>
>>        </condition>
>> </extension>
>>
>>    From fs_cli I originating call
>>
>> originate
>> {ignore_early_media=true}sofia/gateway/sipnet/12126248024\@sipnet 1004
>>
>> Soft phone 1004 registered and ringing then incoming call, but
>> 12126248024 (aleg) is silence!
>>
>> In debug looks good
>>
>> 2015-05-25 17:46:31.990988 [DEBUG] sofia.c:6627 Channel
>> sofia/internal/1004 entering state [proceeding][180]
>> 2015-05-25 17:46:31.990988 [NOTICE] sofia.c:6729 Ring-Ready
>> sofia/internal/1004!
>> 2015-05-25 17:46:31.990988 [DEBUG] switch_channel.c:3297
>> (sofia/internal/1004) Callstate Change DOWN -> RINGING
>>
>> May I forget something?
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 olegstolyar at gmail.com  Mon May 25 18:36:53 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Mon, 25 May 2015 07:36:53 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAFiqYum4if-B8NGFu7PXjFpJZnPuFGy8XhWscM7p7q6BwZiqGg@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAFiqYum4if-B8NGFu7PXjFpJZnPuFGy8XhWscM7p7q6BwZiqGg@mail.gmail.com>
Message-ID: <CAC8JMO+-Wqu9_xQx6TzkUU3NLhVU=HTpDu6wO2p1TX8J1kxEKg@mail.gmail.com>

Thanks Steven!

It may be https://freeswitch.org/jira/browse/FS-7040.

As far as the 120 sec is concerned - Below are snippets from both the
invite and the 200 OK from FS.  I know that FS reads the Session-Expires
from the client because if I change it to a value less than 120, FS sends
back a "SIP/2.0 422 Session Interval Too Small".  I  even thought the
problem could be that the Session-Exipres format in the original INVITE is
incorrect since it does not contain ";refresher=uac" but when I added that
and made the line "Session-Expires: 300;refresher=uac" nothing changed.


INVITE sip:echo-test at anonymous.invalid SIP/2.0
Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029
Max-Forwards: 69
To: <sip:echo-test at anonymous.invalid>
From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
Call-ID: r65e48gp171p21rkppcu
CSeq: 6621 INVITE
Contact: <sip:mqthb9dj at pfcnm9rjv5en.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.26
Content-Length: 2754

SIP/2.0 200 OK
Via: SIP/2.0/WSS
pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200
From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
To: <sip:echo-test at anonymous.invalid>;tag=SgjX0X4arHUFg
Call-ID: r65e48gp171p21rkppcu
CSeq: 6621 INVITE
Contact: <sip:echo-test at 50.19.255.62:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Session-Expires: 120;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 809
X-Netflix: yes
Remote-Party-ID: "echo-test" <sip:echo-test at anonymous.invalid
>;party=calling;privacy=off;screen=no


On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre at gmail.com> wrote:

> The values to actual use are negotiated during the call invite, and your
> trace shows it's using 2 minutes.
> Session-Expires: 120;refresher=uac
> Min-SE: 120
>
> Just as an idea because you didn't send both invite and re-invite...
> perhaps the SDP body is different in the reinvite without the version
> number having changed. If so it may be
> https://freeswitch.org/jira/browse/FS-7040
>
> On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>> But isn't that based on the session-timeout param which defaults to 30
>> min?  My re-invites occur much sooner than 30 min into a call.  Or does
>> session-timeout param only control sessions initiated by FS while incoming
>> sessions use the minimum-session-expires param if it's not explicitly
>> passed by the session initiator?
>>
>> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> session timer
>>>
>>>
>>> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>>
>>>> Thanks Michael, I'll see if we can do that!
>>>>
>>>> So, is the re-INVITE legit and the problem is that JsSip does not
>>>> respond to it?  Still, I am curious what is triggering the re-INVITE.
>>>>
>>>> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>>
>>>>> I think the sip.js guys fixed this issue when they forked jssip.  I'd
>>>>> suggest using that.
>>>>>
>>>>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>>>> wrote:
>>>>> >
>>>>> > Hi guys,
>>>>> >
>>>>> > Several weeks ago I started getting an occasional problem where FS
>>>>> is sending an INVITE to the other side in the middle of a call, the other
>>>>> side does not respond and FS hangs up the leg.  Below is the relevant log.
>>>>> The user experience is that they keep talking and hearing each other up to
>>>>> the very end.  I have a recording of that call, so can confirm.
>>>>> >
>>>>> > The call uses WebRTC and is originated by JsSip from Chrome.  Then
>>>>> the user is put into a conference but I doubt it's relevant in this case
>>>>> since the INVITE and disconnect are not happening from mod_conference
>>>>> >
>>>>> > I suspect it's a re-INVITE but what triggers FS to send it?  I
>>>>> couldn't find anything in the logs that could shed light.
>>>>> >
>>>>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>>>>> >
>>>>> ------------------------------------------------------------------------
>>>>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>>>>> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>>>>> >    Max-Forwards: 70
>>>>> >    From:
>>>>> <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>>>>> >;tag=KQecUSr12rSQp
>>>>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>>>>> >    Call-ID: g8980rbrbk2t45oj5mru
>>>>> >    CSeq: 75703945 INVITE
>>>>> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@
>>>>> ##.##.###.###:5080;transport=udp>
>>>>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>>>>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>> >    Supported: timer, path, replaces
>>>>> >    Session-Expires: 120;refresher=uac
>>>>> >    Min-SE: 120
>>>>> >    Content-Type: application/sdp
>>>>> >    Content-Disposition: session
>>>>> >    Content-Length: 825
>>>>> >
>>>>> >    v=0
>>>>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>>>>> >    s=FreeSWITCH
>>>>> >    c=IN IP4 ##.##.##.##
>>>>> >    t=0 0
>>>>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>> >    m=audio 22670 RTP/SAVPF 9 126 106
>>>>> >    a=rtpmap:9 G722/8000
>>>>> >    a=rtpmap:126 telephone-event/8000
>>>>> >    a=rtpmap:106 CN/8000
>>>>> >    a=ptime:20
>>>>> >    a=fingerprint:sha-256
>>>>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>>>>> >    a=rtcp-mux
>>>>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>>>>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>>>>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>>>>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>>>>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>>>>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>>>>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host
>>>>> generation 0
>>>>> >
>>>>> ------------------------------------------------------------------------
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>> entering state [calling][0]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>> entering state [terminating][503]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
>>>>> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [DEBUG] switch_channel.c:3242 Send signal
>>>>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>> [DEBUG] switch_core_session.c:1396 Send signal
>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] mod_conference.c:5057 Channel leaving conference, cause:
>>>>> NORMAL_TEMPORARY_FAILURE
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid
>>>>> skip receive message [UNBRIDGE] (channel is hungup already)
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_media.c:7772
>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>> [REFER_EVENT] (channel is hungup already)
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
>>>>> Restore previous codec G722:9.
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_session.c:2901
>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:535
>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to
>>>>> sleep
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:472
>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>>>>> CS_HANGUP
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:735
>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change
>>>>> ACTIVE -> HANGUP
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:737
>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>> hanging up, cause: NORMAL_TEMPORARY_FAILURE
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:60
>>>>> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause:
>>>>> NORMAL_TEMPORARY_FAILURE
>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>> [DEBUG] switch_core_state_machine.c:737
>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to
>>>>> sleep
>>>>> >
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dm at dwide.com  Mon May 25 19:01:44 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Mon, 25 May 2015 19:01:44 +0400
Subject: [Freeswitch-users] ringback - multiple values
Message-ID: <55633958.3030002@dwide.com>

Hello

Is it possible to set multiple values in ringback?

<action application="set" 
data="ringback=file_string:///opt/freeswitch/sounds/en/us/callie/voicemail/8000/vm-goodbye.wav,${us-ring}"/>

I try to use " , ! ; " character for separate - Not works.

My idea is play wav file and then play us-ring while bridge is calling 
to destination

How can do it?


From cdgraff at gmail.com  Mon May 25 21:49:04 2015
From: cdgraff at gmail.com (Alejandro)
Date: Mon, 25 May 2015 14:49:04 -0300
Subject: [Freeswitch-users] CELL number providers in latam
Message-ID: <CALJtzxCgyGe09z8nng=TROdjfh5e9nc2fDu+VoxsgKPwUDbbAA@mail.gmail.com>

Hi all,

I'm searching for advices, about some company that provide Cell Numbers
locally in LATAM.

The idea is the customer can send a SMS to this number and we can receive
the message.

Thanks for any advice.
Ale
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From ssinyagin at gmail.com  Mon May 25 22:42:45 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 25 May 2015 20:42:45 +0200
Subject: [Freeswitch-users] CELL number providers in latam
In-Reply-To: <CALJtzxCgyGe09z8nng=TROdjfh5e9nc2fDu+VoxsgKPwUDbbAA@mail.gmail.com>
References: <CALJtzxCgyGe09z8nng=TROdjfh5e9nc2fDu+VoxsgKPwUDbbAA@mail.gmail.com>
Message-ID: <CACOYK=icTwLxOPg0mSE3aF19+wS4JcL6=_amgoNf1KSYy6GQhA@mail.gmail.com>

you can always set up a GSM gateway and receive SMS (and voice calls)
remotely.I used a Yeastar gateway for this, works quite fine.



On Mon, May 25, 2015 at 7:49 PM, Alejandro <cdgraff at gmail.com> wrote:
> Hi all,
>
> I'm searching for advices, about some company that provide Cell Numbers
> locally in LATAM.
>
> The idea is the customer can send a SMS to this number and we can receive
> the message.
>
> Thanks for any advice.
> Ale
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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.medina at cibersys.com  Mon May 25 23:41:49 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Mon, 25 May 2015 15:11:49 -0430
Subject: [Freeswitch-users] How to enable Screen sharing on Verto Conf?
Message-ID: <CACXsapOA9nG0uo3+artAT72DD17umf9Ck5dHCKjPXDzjd1+s_Q@mail.gmail.com>

Hi guys!

I been playing around with the verto conference demo on 1.6.

How do I enable the screen sharing feature?

-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From krice at freeswitch.org  Tue May 26 00:09:36 2015
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 25 May 2015 20:09:36 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) May
	16th-May 22nd
Message-ID: <55638180eb261_55a168b33440894@resque-worker-high.1.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1FbFRZL
FreeSWITCH Week in Review (Master Branch) May 16th-May 22nd
Hello, again. This passed week in the FreeSWITCH master branch we had 7 commits. We saw some neat bug fixes go in this week and I hope everyone partaking in today?s holiday enjoys their long weekend!

Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support.

Improvements in build system, cross platform support, and packaging:

FS-7488 [mod_managed] Fix a build error with Windows and removed duplicate files

The following bugs were squashed:

FS-7541 Fixed an issue with audio gaps in native audio recordings

FS-7562 [mod_sofia] Fixed an interop issue caused when using bypass media with t.38 passthru

FS-7567 Fixed a rare segfault on shutdown caused by a race condition

FS-7529 Fixed an error with call recording on G722 calls

?

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From victor.medina at cibersys.com  Tue May 26 01:12:03 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Mon, 25 May 2015 16:42:03 -0430
Subject: [Freeswitch-users] How to enable Screen sharing on Verto Conf?
In-Reply-To: <CACXsapOA9nG0uo3+artAT72DD17umf9Ck5dHCKjPXDzjd1+s_Q@mail.gmail.com>
References: <CACXsapOA9nG0uo3+artAT72DD17umf9Ck5dHCKjPXDzjd1+s_Q@mail.gmail.com>
Message-ID: <CACXsapOLEbJA9Dfbfc1YMOnSHa73eTMnm+NUG5qs=N0Xz3gzYQ@mail.gmail.com>

?Log..

Attempting Screen Capture....
SCREEN SHARE
Audio constraints false
Video constraints Object {mandatory: Object, optional: Array[0]}mandatory:
Objectoptional: Array[0]__proto__: Object
Stream Success
stream started
Offer SDP
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
offer RTCIceCandidate {}
ICE Complete
ICE SDP
offer v=0
o=- 3905402391138908374 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
m=audio 34382 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 186.14.135.35
a=rtcp:41117 IN IP4 186.14.135.35
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54692 typ host
generation 0
a=candidate:648569486 1 udp 2122129151 10.0.1.10 54693 typ host generation 0
a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54694 typ host
generation 0
a=candidate:648569486 2 udp 2122129150 10.0.1.10 54695 typ host generation 0
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
active generation 0
a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype active
generation 0
a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
active generation 0
a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype active
generation 0
a=candidate:1680186374 1 udp 1685921535 186.14.135.35 34382 typ srflx raddr
10.0.1.10 rport 54693 generation 0
a=candidate:1680186374 2 udp 1685921534 186.14.135.35 41117 typ srflx raddr
10.0.1.10 rport 54695 generation 0
a=ice-ufrag:qjfFF9BX74oS8gTh
a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
a=fingerprint:sha-256
52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
m=video 39129 RTP/SAVPF 100 116 117 96
c=IN IP4 186.14.135.35
a=rtcp:40989 IN IP4 186.14.135.35
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54696 typ host
generation 0
a=candidate:648569486 1 udp 2122129151 10.0.1.10 54697 typ host generation 0
a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54698 typ host
generation 0
a=candidate:648569486 2 udp 2122129150 10.0.1.10 54699 typ host generation 0
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
active generation 0
a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype active
generation 0
a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
active generation 0
a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype active
generation 0
a=candidate:1680186374 2 udp 1685921534 186.14.135.35 40989 typ srflx raddr
10.0.1.10 rport 54699 generation 0
a=candidate:1680186374 1 udp 1685921535 186.14.135.35 39129 typ srflx raddr
10.0.1.10 rport 54697 generation 0
a=ice-ufrag:qjfFF9BX74oS8gTh
a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
a=fingerprint:sha-256
52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=ssrc-group:FID 3962025390 1186212002
a=ssrc:3962025390 cname:Rvg58eAlJAmWf/sm
a=ssrc:3962025390 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
a45b2502-eb86-452c-b4cb-e69e5305938a
a=ssrc:3962025390 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
a=ssrc:3962025390 label:a45b2502-eb86-452c-b4cb-e69e5305938a
a=ssrc:1186212002 cname:Rvg58eAlJAmWf/sm
a=ssrc:1186212002 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
a45b2502-eb86-452c-b4cb-e69e5305938a
a=ssrc:1186212002 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
a=ssrc:1186212002 label:a45b2502-eb86-452c-b4cb-e69e5305938a

Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from new to
requesting
Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from requesting
to trying
Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from trying to
hangup
Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from hangup to
destroy
?
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From mike at jerris.com  Tue May 26 02:55:35 2015
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 May 2015 18:55:35 -0400
Subject: [Freeswitch-users] How to enable Screen sharing on Verto Conf?
In-Reply-To: <CACXsapOLEbJA9Dfbfc1YMOnSHa73eTMnm+NUG5qs=N0Xz3gzYQ@mail.gmail.com>
References: <CACXsapOA9nG0uo3+artAT72DD17umf9Ck5dHCKjPXDzjd1+s_Q@mail.gmail.com>
	<CACXsapOLEbJA9Dfbfc1YMOnSHa73eTMnm+NUG5qs=N0Xz3gzYQ@mail.gmail.com>
Message-ID: <CAEwTsAhpyeuBuCXB1bePAiOXPP8RtaYMgaSiiLjP6J+5aMVyFQ@mail.gmail.com>

for chrome to allow screen share it requires you to install a chrome plugin
or start chrome with a command line argument to allow screen share

On Monday, May 25, 2015, Victor Medina <victor.medina at cibersys.com> wrote:

> ?Log..
>
> Attempting Screen Capture....
> SCREEN SHARE
> Audio constraints false
> Video constraints Object {mandatory: Object, optional: Array[0]}mandatory:
> Objectoptional: Array[0]__proto__: Object
> Stream Success
> stream started
> Offer SDP
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> offer RTCIceCandidate {}
> ICE Complete
> ICE SDP
> offer v=0
> o=- 3905402391138908374 2 IN IP4 127.0.0.1
> s=-
> t=0 0
> a=group:BUNDLE audio video
> a=msid-semantic: WMS dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
> m=audio 34382 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
> c=IN IP4 186.14.135.35
> a=rtcp:41117 IN IP4 186.14.135.35
> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54692 typ host
> generation 0
> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54693 typ host generation
> 0
> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54694 typ host
> generation 0
> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54695 typ host generation
> 0
> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
> active generation 0
> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
> active generation 0
> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
> active generation 0
> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
> active generation 0
> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 34382 typ srflx
> raddr 10.0.1.10 rport 54693 generation 0
> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 41117 typ srflx
> raddr 10.0.1.10 rport 54695 generation 0
> a=ice-ufrag:qjfFF9BX74oS8gTh
> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
> a=fingerprint:sha-256
> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
> a=setup:actpass
> a=mid:audio
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=recvonly
> a=rtcp-mux
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10; useinbandfec=1
> a=rtpmap:103 ISAC/16000
> a=rtpmap:104 ISAC/32000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=maxptime:60
> m=video 39129 RTP/SAVPF 100 116 117 96
> c=IN IP4 186.14.135.35
> a=rtcp:40989 IN IP4 186.14.135.35
> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54696 typ host
> generation 0
> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54697 typ host generation
> 0
> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54698 typ host
> generation 0
> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54699 typ host generation
> 0
> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
> active generation 0
> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
> active generation 0
> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
> active generation 0
> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
> active generation 0
> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 40989 typ srflx
> raddr 10.0.1.10 rport 54699 generation 0
> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 39129 typ srflx
> raddr 10.0.1.10 rport 54697 generation 0
> a=ice-ufrag:qjfFF9BX74oS8gTh
> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
> a=fingerprint:sha-256
> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
> a=setup:actpass
> a=mid:video
> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=extmap:4 urn:3gpp:video-orientation
> a=sendrecv
> a=rtcp-mux
> a=rtpmap:100 VP8/90000
> a=rtcp-fb:100 ccm fir
> a=rtcp-fb:100 nack
> a=rtcp-fb:100 nack pli
> a=rtcp-fb:100 goog-remb
> a=rtpmap:116 red/90000
> a=rtpmap:117 ulpfec/90000
> a=rtpmap:96 rtx/90000
> a=fmtp:96 apt=100
> a=ssrc-group:FID 3962025390 1186212002
> a=ssrc:3962025390 cname:Rvg58eAlJAmWf/sm
> a=ssrc:3962025390 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
> a45b2502-eb86-452c-b4cb-e69e5305938a
> a=ssrc:3962025390 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
> a=ssrc:3962025390 label:a45b2502-eb86-452c-b4cb-e69e5305938a
> a=ssrc:1186212002 cname:Rvg58eAlJAmWf/sm
> a=ssrc:1186212002 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
> a45b2502-eb86-452c-b4cb-e69e5305938a
> a=ssrc:1186212002 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
> a=ssrc:1186212002 label:a45b2502-eb86-452c-b4cb-e69e5305938a
>
> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from new to
> requesting
> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from requesting
> to trying
> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from trying to
> hangup
> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from hangup to
> destroy
> ?
>
>
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From yadenis at seznam.cz  Tue May 26 11:40:02 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 26 May 2015 09:40:02 +0200
Subject: [Freeswitch-users] Problem with MP4 file record
Message-ID: <43162995.20150526094002@seznam.cz>

Hi all !

I have one problem with the MP4 file recording with sound. The video is great. But there is no sound.

Adding a log. files in place, but for some reason does not work. I use Debian and the latest version 1.6

EXECUTE sofia/internal/1004 at 192.168.242.132 record(/usr/local/freeswitch/recordings/testrecord.mp4)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}  write video

  

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From s.safarov at gmail.com  Tue May 26 13:02:02 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 26 May 2015 12:02:02 +0300
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <43162995.20150526094002@seznam.cz>
References: <43162995.20150526094002@seznam.cz>
Message-ID: <CAHtxdDe-jcRTpRtHQauTQOJCxop4Qz2f8-SZaE4LbMFpAC34+g@mail.gmail.com>

make install vlc-plugin-jack.x86_64

On Tue, May 26, 2015 at 10:40 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi all !
>
> I have one problem with the MP4 file recording with sound. The video is
> great. But there is no sound.
>
> Adding a log. files in place, but for some reason does not work. I use
> Debian and the latest version 1.6
>
> EXECUTE sofia/internal/1004 at 192.168.242.132
> record(/usr/local/freeswitch/recordings/testrecord.mp4)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
> open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open
> #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}
>  write video
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue May 26 13:20:11 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 26 May 2015 11:20:11 +0200
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <CAHtxdDe-jcRTpRtHQauTQOJCxop4Qz2f8-SZaE4LbMFpAC34+g@mail.gmail.com>
References: <43162995.20150526094002@seznam.cz> 
	<CAHtxdDe-jcRTpRtHQauTQOJCxop4Qz2f8-SZaE4LbMFpAC34+g@mail.gmail.com>
Message-ID: <14610348138.20150526112011@seznam.cz>

Dobr? den,

Unknown target vlc-plugin-jack.x86_64
Makefile:666: recipe for target 'vlc-plugin-jack.x86_64-all' failed
make[2]: *** [vlc-plugin-jack.x86_64-all] Error 1
make[2]: Leaving directory '/usr/src/freeswitch.git/src/mod'
Makefile:666: recipe for target 'vlc-plugin-jack.x86_64' failed
make[1]: *** [vlc-plugin-jack.x86_64] Error 2
make[1]: Leaving directory '/usr/src/freeswitch.git/src/mod'
Makefile:3087: recipe for target 'vlc-plugin-jack.x86_64' failed
make: *** [vlc-plugin-jack.x86_64] Error 2


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

?ter? 26. kv?tna 2015, 11:02:02, napsal jste:


make install vlc-plugin-jack.x86_64

On Tue, May 26, 2015 at 10:40 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi all !

I have one problem with the MP4 file recording with sound. The video is great. But there is no sound.

Adding a log. files in place, but for some reason does not work. I use Debian and the latest version 1.6

EXECUTE sofia/internal/1004 at 192.168.242.132 record(/usr/local/freeswitch/recordings/testrecord.mp4)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}  write video

 

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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From mike at jerris.com  Tue May 26 13:22:35 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 26 May 2015 05:22:35 -0400
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <43162995.20150526094002@seznam.cz>
References: <43162995.20150526094002@seznam.cz>
Message-ID: <CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>

try using mod av instead.  You'll need to install the -extra package for
libav

On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi all !
>
> I have one problem with the MP4 file recording with sound. The video is
> great. But there is no sound.
>
> Adding a log. files in place, but for some reason does not work. I use
> Debian and the latest version 1.6
>
> EXECUTE sofia/internal/1004 at 192.168.242.132
> <javascript:_e(%7B%7D,'cvml','1004 at 192.168.242.132');>
> record(/usr/local/freeswitch/recordings/testrecord.mp4)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
> open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open
> #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}
>  write video
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382*
>
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From s.safarov at gmail.com  Tue May 26 13:37:36 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 26 May 2015 12:37:36 +0300
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
References: <43162995.20150526094002@seznam.cz>
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
Message-ID: <CAHtxdDdScw0gV4_LhgtEp5H5rKUeYvqwFcLWf1QMp_gt=6OYiw@mail.gmail.com>

This case FS-7553 related
https://freeswitch.org/jira/browse/FS-7553

Denis execute
yum install lc-plugin-jack.x86_64

Prevesion message has typo

On Tue, May 26, 2015 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:

> try using mod av instead.  You'll need to install the -extra package for
> libav
>
>
> On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>>  Hi all !
>>
>> I have one problem with the MP4 file recording with sound. The video is
>> great. But there is no sound.
>>
>> Adding a log. files in place, but for some reason does not work. I use
>> Debian and the latest version 1.6
>>
>> EXECUTE sofia/internal/1004 at 192.168.242.132
>> record(/usr/local/freeswitch/recordings/testrecord.mp4)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
>> open shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
>> (libvpx.so.1: cannot open shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so'
>> (libvpx.so.1: cannot open shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open
>> #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}
>>  write video
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382*
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue May 26 13:49:46 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 26 May 2015 11:49:46 +0200
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <CAHtxdDdScw0gV4_LhgtEp5H5rKUeYvqwFcLWf1QMp_gt=6OYiw@mail.gmail.com>
References: <43162995.20150526094002@seznam.cz> 
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
	<CAHtxdDdScw0gV4_LhgtEp5H5rKUeYvqwFcLWf1QMp_gt=6OYiw@mail.gmail.com>
Message-ID: <982900675.20150526114946@seznam.cz>

Dobr? den,

I try this in Debian. ?ut of course also try to ?entos

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

?ter? 26. kv?tna 2015, 11:37:36, napsal jste:


This case FS-7553 related
https://freeswitch.org/jira/browse/FS-7553

Denis execute
yum install lc-plugin-jack.x86_64

Prevesion message has typo

On Tue, May 26, 2015 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:
try using mod av instead.  You'll need to install the -extra package for libav 


On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi all !

I have one problem with the MP4 file recording with sound. The video is great. But there is no sound.

Adding a log. files in place, but for some reason does not work. I use Debian and the latest version 1.6

EXECUTE sofia/internal/1004 at 192.168.242.132 record(/usr/local/freeswitch/recordings/testrecord.mp4)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}  write video

 

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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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  Tue May 26 13:54:27 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 26 May 2015 11:54:27 +0200
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <CAHtxdDdScw0gV4_LhgtEp5H5rKUeYvqwFcLWf1QMp_gt=6OYiw@mail.gmail.com>
References: <43162995.20150526094002@seznam.cz> 
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
	<CAHtxdDdScw0gV4_LhgtEp5H5rKUeYvqwFcLWf1QMp_gt=6OYiw@mail.gmail.com>
Message-ID: <10210685056.20150526115427@seznam.cz>

Dobr? den,

I do not understand what I'm doing wrong

yum install lc-plugin-jack.x86_64
Loaded plugins: fastestmirror, langpacks
adobe-linux-x86_64                                       |  951 B     00:00
base                                                     | 3.6 kB     00:00
epel/x86_64/metalink                                     |  25 kB     00:00
extras                                                   | 3.4 kB     00:00
fs                                                       | 2.9 kB     00:00
rpmforge                                                 | 1.9 kB     00:00
updates                                                  | 3.4 kB     00:00
fs/primary_db                                              |  61 kB   00:00
Loading mirror speeds from cached hostfile
 * base: mirror.hosting90.cz
 * epel: mirror.hosting90.cz
 * extras: mirror.hosting90.cz
 * rpmforge: mirror.vutbr.cz
 * updates: mirror.hosting90.cz
No package lc-plugin-jack.x86_64 available.
Error: Nothing to do

What do I need to connect the repository to make it work?


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

?ter? 26. kv?tna 2015, 11:37:36, napsal jste:


This case FS-7553 related
https://freeswitch.org/jira/browse/FS-7553

Denis execute
yum install lc-plugin-jack.x86_64

Prevesion message has typo

On Tue, May 26, 2015 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:
try using mod av instead.  You'll need to install the -extra package for libav 


On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi all !

I have one problem with the MP4 file recording with sound. The video is great. But there is no sound.

Adding a log. files in place, but for some reason does not work. I use Debian and the latest version 1.6

EXECUTE sofia/internal/1004 at 192.168.242.132 record(/usr/local/freeswitch/recordings/testrecord.mp4)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}  write video

 

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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 May 26 13:58:47 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 26 May 2015 12:58:47 +0300
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <10210685056.20150526115427@seznam.cz>
References: <43162995.20150526094002@seznam.cz>
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
	<CAHtxdDdScw0gV4_LhgtEp5H5rKUeYvqwFcLWf1QMp_gt=6OYiw@mail.gmail.com>
	<10210685056.20150526115427@seznam.cz>
Message-ID: <CAHtxdDe_gVzAFjjxA0gfm7k5ajobefZfEy+WTzaPie3cd3nrkA@mail.gmail.com>

yum install *v*lc-plugin-jack.x86_64
Typo

On Tue, May 26, 2015 at 12:54 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> I do not understand what I'm doing wrong
>
> yum install lc-plugin-jack.x86_64
> Loaded plugins: fastestmirror, langpacks
> adobe-linux-x86_64                                       |  951 B     00:00
> base                                                     | 3.6 kB     00:00
> epel/x86_64/metalink                                     |  25 kB     00:00
> extras                                                   | 3.4 kB     00:00
> fs                                                       | 2.9 kB     00:00
> rpmforge                                                 | 1.9 kB     00:00
> updates                                                  | 3.4 kB     00:00
> fs/primary_db                                              |  61 kB   00:00
> Loading mirror speeds from cached hostfile
> * base: mirror.hosting90.cz
> * epel: mirror.hosting90.cz
> * extras: mirror.hosting90.cz
> * rpmforge: mirror.vutbr.cz
> * updates: mirror.hosting90.cz
> No package lc-plugin-jack.x86_64 available.
> Error: Nothing to do
>
> What do I need to connect the repository to make it work?
>
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 ?ter? 26. kv?tna 2015, 11:37:36, napsal jste:
> *
>
>  This case FS-7553 related
> https://freeswitch.org/jira/browse/FS-7553
>
> Denis execute
> yum install lc-plugin-jack.x86_64
>
> Prevesion message has typo
>
> On Tue, May 26, 2015 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:
> try using mod av instead.  You'll need to install the -extra package for
> libav
>
>
> On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi all !
>
> I have one problem with the MP4 file recording with sound. The video is
> great. But there is no sound.
>
> Adding a log. files in place, but for some reason does not work. I use
> Debian and the latest version 1.6
>
> EXECUTE sofia/internal/1004 at 192.168.242.132
> record(/usr/local/freeswitch/recordings/testrecord.mp4)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
> open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open
> #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}
>  write video
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
> *. 775-415-382 *
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue May 26 14:07:10 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 26 May 2015 12:07:10 +0200
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
References: <43162995.20150526094002@seznam.cz> 
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
Message-ID: <16210177989.20150526120710@seznam.cz>

Dobr? den,

Where can I get the -extra package for libav?

I try from here https://freeswitch.org/stash/projects/SD/repos/libav/browse  It is compiled without problems. 

But I still have a error

2015-05-26 06:01:38.083643 [CRIT] switch_loadable_module.c:1520 Error Loading module /usr/local/freeswitch/mod/mod_av.so
**libvpx.so.1: cannot open shared object file: No such file or directory**




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

?ter? 26. kv?tna 2015, 11:22:35, napsal jste:


try using mod av instead.  You'll need to install the -extra package for libav 

On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi all !

I have one problem with the MP4 file recording with sound. The video is great. But there is no sound.

Adding a log. files in place, but for some reason does not work. I use Debian and the latest version 1.6

EXECUTE sofia/internal/1004 at 192.168.242.132 record(/usr/local/freeswitch/recordings/testrecord.mp4)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}  write video

 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From steveayre at gmail.com  Tue May 26 16:12:38 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 May 2015 13:12:38 +0100
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <43162995.20150526094002@seznam.cz>
References: <43162995.20150526094002@seznam.cz>
Message-ID: <CAFiqYumdqs_LnD2P9cqao3rDr2XqJE-SqisU+x3fY9XwZW_bww@mail.gmail.com>

You're missing the vlc plugins. Try installing the vlc-nox package (apt-get
install vlc-nox), it looks like that contains the files you're missing.

On 26 May 2015 at 08:40, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi all !
>
> I have one problem with the MP4 file recording with sound. The video is
> great. But there is no sound.
>
> Adding a log. files in place, but for some reason does not work. I use
> Debian and the latest version 1.6
>
> EXECUTE sofia/internal/1004 at 192.168.242.132
> record(/usr/local/freeswitch/recordings/testrecord.mp4)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
> open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open
> #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}
>  write video
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Tue May 26 16:13:52 2015
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 May 2015 13:13:52 +0100
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <CAFiqYumdqs_LnD2P9cqao3rDr2XqJE-SqisU+x3fY9XwZW_bww@mail.gmail.com>
References: <43162995.20150526094002@seznam.cz>
	<CAFiqYumdqs_LnD2P9cqao3rDr2XqJE-SqisU+x3fY9XwZW_bww@mail.gmail.com>
Message-ID: <CAFiqYun+kaBqXg5idYt_CfKA7+HZ2nNajkCW7502LLKCgedmng@mail.gmail.com>

Which is assuming you're on Debian as you originally said, and not CentOS
as your use of yum suggests.

On 26 May 2015 at 13:12, Steven Ayre <steveayre at gmail.com> wrote:

> You're missing the vlc plugins. Try installing the vlc-nox package
> (apt-get install vlc-nox), it looks like that contains the files you're
> missing.
>
> On 26 May 2015 at 08:40, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>>  Hi all !
>>
>> I have one problem with the MP4 file recording with sound. The video is
>> great. But there is no sound.
>>
>> Adding a log. files in place, but for some reason does not work. I use
>> Debian and the latest version 1.6
>>
>> EXECUTE sofia/internal/1004 at 192.168.242.132
>> record(/usr/local/freeswitch/recordings/testrecord.mp4)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
>> open shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
>> (libvpx.so.1: cannot open shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so'
>> (libvpx.so.1: cannot open shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open
>> #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}
>>  write video
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382*
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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  Tue May 26 16:35:25 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 26 May 2015 14:35:25 +0200
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <CAFiqYun+kaBqXg5idYt_CfKA7+HZ2nNajkCW7502LLKCgedmng@mail.gmail.com>
References: <43162995.20150526094002@seznam.cz> 
	<CAFiqYumdqs_LnD2P9cqao3rDr2XqJE-SqisU+x3fY9XwZW_bww@mail.gmail.com>
	<CAFiqYun+kaBqXg5idYt_CfKA7+HZ2nNajkCW7502LLKCgedmng@mail.gmail.com>
Message-ID: <1185692794.20150526143525@seznam.cz>

Dobr? den,

I'm trying to do it on Debian, as I wrote in the first message.

apt-get install vlc-nox
Reading package lists... Done
Building dependency tree       
Reading state information... Done
Some packages could not be installed. This may mean that you have
requested an impossible situation or if you are using the unstable
distribution that some required packages have not yet been created
or been moved out of Incoming.
The following information may help to resolve the situation:

The following packages have unmet dependencies:
 vlc-nox : Depends: libavcodec56 (>= 10:2.6.2) but it is not going to be installed
           Depends: libavformat56 (>= 10:2.6.2) but 6:11.3-3+b1 is to be installed
E: Unable to correct problems, you have held broken packages.


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

?ter? 26. kv?tna 2015, 14:13:52, napsal jste:


Which is assuming you're on Debian as you originally said, and not CentOS as your use of yum suggests.

On 26 May 2015 at 13:12, Steven Ayre <steveayre at gmail.com> wrote:
You're missing the vlc plugins. Try installing the vlc-nox package (apt-get install vlc-nox), it looks like that contains the files you're missing.

On 26 May 2015 at 08:40, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi all !

I have one problem with the MP4 file recording with sound. The video is great. But there is no sound.

Adding a log. files in place, but for some reason does not work. I use Debian and the latest version 1.6

EXECUTE sofia/internal/1004 at 192.168.242.132 record(/usr/local/freeswitch/recordings/testrecord.mp4)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}  write video

 

-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 May 26 16:57:11 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 26 May 2015 15:57:11 +0300
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <16210177989.20150526120710@seznam.cz>
References: <43162995.20150526094002@seznam.cz>
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
	<16210177989.20150526120710@seznam.cz>
Message-ID: <CAHtxdDd=vo6uwo-7imdo7MEGuRDC2PYbqk3HyLvvD1rdfkZxRw@mail.gmail.com>

execute as root "ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1"
And try again

On Tue, May 26, 2015 at 1:07 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> Where can I get the -extra package for libav?
>
> I try from here
> https://freeswitch.org/stash/projects/SD/repos/libav/browse  It is
> compiled without problems.
>
> But I still have a error
>
> 2015-05-26 06:01:38.083643 [CRIT] switch_loadable_module.c:1520 Error
> Loading module /usr/local/freeswitch/mod/mod_av.so
> **libvpx.so.1: cannot open shared object file: No such file or directory**
>
>
>
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 ?ter? 26. kv?tna 2015, 11:22:35, napsal jste:
> *
>
>  try using mod av instead.  You'll need to install the -extra package for
> libav
>
> On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi all !
>
> I have one problem with the MP4 file recording with sound. The video is
> great. But there is no sound.
>
> Adding a log. files in place, but for some reason does not work. I use
> Debian and the latest version 1.6
>
> EXECUTE sofia/internal/1004 at 192.168.242.132
> record(/usr/local/freeswitch/recordings/testrecord.mp4)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
> open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open
> #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}
>  write video
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel*.
> 775-415-382*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.medina at cibersys.com  Tue May 26 18:04:55 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Tue, 26 May 2015 09:34:55 -0430
Subject: [Freeswitch-users] How to enable Screen sharing on Verto Conf?
In-Reply-To: <CAEwTsAhpyeuBuCXB1bePAiOXPP8RtaYMgaSiiLjP6J+5aMVyFQ@mail.gmail.com>
References: <CACXsapOA9nG0uo3+artAT72DD17umf9Ck5dHCKjPXDzjd1+s_Q@mail.gmail.com>
	<CACXsapOLEbJA9Dfbfc1YMOnSHa73eTMnm+NUG5qs=N0Xz3gzYQ@mail.gmail.com>
	<CAEwTsAhpyeuBuCXB1bePAiOXPP8RtaYMgaSiiLjP6J+5aMVyFQ@mail.gmail.com>
Message-ID: <CACXsapM8YLag7fauLZpqJrApqiiModv72HAiaOAnc+HGJKV6dQ@mail.gmail.com>

I did both!
I installed the WebRTC Screen Capture extension and check all command line
option on chrome: --allow-http-screen-capture &
--auto-select-desktop-capture-source as of ...
http://peter.sh/experiments/chromium-command-line-switches/

I think only the extension is needed, right?

2015-05-25 18:25 GMT-04:30 Michael Jerris <mike at jerris.com>:

> for chrome to allow screen share it requires you to install a chrome
> plugin or start chrome with a command line argument to allow screen share
>
>
> On Monday, May 25, 2015, Victor Medina <victor.medina at cibersys.com> wrote:
>
>> ?Log..
>>
>> Attempting Screen Capture....
>> SCREEN SHARE
>> Audio constraints false
>> Video constraints Object {mandatory: Object, optional:
>> Array[0]}mandatory: Objectoptional: Array[0]__proto__: Object
>> Stream Success
>> stream started
>> Offer SDP
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> offer RTCIceCandidate {}
>> ICE Complete
>> ICE SDP
>> offer v=0
>> o=- 3905402391138908374 2 IN IP4 127.0.0.1
>> s=-
>> t=0 0
>> a=group:BUNDLE audio video
>> a=msid-semantic: WMS dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>> m=audio 34382 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>> c=IN IP4 186.14.135.35
>> a=rtcp:41117 IN IP4 186.14.135.35
>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54692 typ host
>> generation 0
>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54693 typ host
>> generation 0
>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54694 typ host
>> generation 0
>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54695 typ host
>> generation 0
>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
>> active generation 0
>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>> active generation 0
>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
>> active generation 0
>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>> active generation 0
>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 34382 typ srflx
>> raddr 10.0.1.10 rport 54693 generation 0
>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 41117 typ srflx
>> raddr 10.0.1.10 rport 54695 generation 0
>> a=ice-ufrag:qjfFF9BX74oS8gTh
>> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
>> a=fingerprint:sha-256
>> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
>> a=setup:actpass
>> a=mid:audio
>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>> a=recvonly
>> a=rtcp-mux
>> a=rtpmap:111 opus/48000/2
>> a=fmtp:111 minptime=10; useinbandfec=1
>> a=rtpmap:103 ISAC/16000
>> a=rtpmap:104 ISAC/32000
>> a=rtpmap:9 G722/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:106 CN/32000
>> a=rtpmap:105 CN/16000
>> a=rtpmap:13 CN/8000
>> a=rtpmap:126 telephone-event/8000
>> a=maxptime:60
>> m=video 39129 RTP/SAVPF 100 116 117 96
>> c=IN IP4 186.14.135.35
>> a=rtcp:40989 IN IP4 186.14.135.35
>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54696 typ host
>> generation 0
>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54697 typ host
>> generation 0
>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54698 typ host
>> generation 0
>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54699 typ host
>> generation 0
>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
>> active generation 0
>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>> active generation 0
>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
>> active generation 0
>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>> active generation 0
>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 40989 typ srflx
>> raddr 10.0.1.10 rport 54699 generation 0
>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 39129 typ srflx
>> raddr 10.0.1.10 rport 54697 generation 0
>> a=ice-ufrag:qjfFF9BX74oS8gTh
>> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
>> a=fingerprint:sha-256
>> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
>> a=setup:actpass
>> a=mid:video
>> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>> a=extmap:4 urn:3gpp:video-orientation
>> a=sendrecv
>> a=rtcp-mux
>> a=rtpmap:100 VP8/90000
>> a=rtcp-fb:100 ccm fir
>> a=rtcp-fb:100 nack
>> a=rtcp-fb:100 nack pli
>> a=rtcp-fb:100 goog-remb
>> a=rtpmap:116 red/90000
>> a=rtpmap:117 ulpfec/90000
>> a=rtpmap:96 rtx/90000
>> a=fmtp:96 apt=100
>> a=ssrc-group:FID 3962025390 1186212002
>> a=ssrc:3962025390 cname:Rvg58eAlJAmWf/sm
>> a=ssrc:3962025390 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>> a45b2502-eb86-452c-b4cb-e69e5305938a
>> a=ssrc:3962025390 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>> a=ssrc:3962025390 label:a45b2502-eb86-452c-b4cb-e69e5305938a
>> a=ssrc:1186212002 cname:Rvg58eAlJAmWf/sm
>> a=ssrc:1186212002 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>> a45b2502-eb86-452c-b4cb-e69e5305938a
>> a=ssrc:1186212002 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>> a=ssrc:1186212002 label:a45b2502-eb86-452c-b4cb-e69e5305938a
>>
>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from new to
>> requesting
>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from requesting
>> to trying
>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from trying to
>> hangup
>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from hangup to
>> destroy
>> ?
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From dujinfang at gmail.com  Tue May 26 18:34:28 2015
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 26 May 2015 22:34:28 +0800
Subject: [Freeswitch-users] XML_RPC connections limit
In-Reply-To: <B96DD723-6467-468D-9A8D-E28C0D020801@gmail.com>
References: <B96DD723-6467-468D-9A8D-E28C0D020801@gmail.com>
Message-ID: <CAJs+-5rjkCvqP4=7VvM2zKZ7MGZ1Y8s6wGm=P1QWmoLTU9Gngw@mail.gmail.com>

We found the lib performs bad on load. mod_verto has a non-documented http
interface should work better.


On Fri, May 15, 2015 at 10:22 PM, Leonid Nasedkin <leonidnasedkin at gmail.com>
wrote:

> Hi, All!
> Im trying to use freeswitch for automated outbound calls.
> Im init calls via xml_rpc interface with api call. But it seems like
> somewhere in freeswitch hardcoded connection limit to xml_rpc . I cant
> create more then 16 concurrent connections.
> How I can increase this value?
> Im using:
> FreeSWITCH version: 1.4.15~64bit ( 64bit)
> CentOS 6.5
> 2.6.32-504.16.2.el6.x86_64
>
> 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
>



-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn
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From mike at jerris.com  Tue May 26 18:54:44 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 26 May 2015 10:54:44 -0400
Subject: [Freeswitch-users] How to enable Screen sharing on Verto Conf?
In-Reply-To: <CACXsapM8YLag7fauLZpqJrApqiiModv72HAiaOAnc+HGJKV6dQ@mail.gmail.com>
References: <CACXsapOA9nG0uo3+artAT72DD17umf9Ck5dHCKjPXDzjd1+s_Q@mail.gmail.com>
	<CACXsapOLEbJA9Dfbfc1YMOnSHa73eTMnm+NUG5qs=N0Xz3gzYQ@mail.gmail.com>
	<CAEwTsAhpyeuBuCXB1bePAiOXPP8RtaYMgaSiiLjP6J+5aMVyFQ@mail.gmail.com>
	<CACXsapM8YLag7fauLZpqJrApqiiModv72HAiaOAnc+HGJKV6dQ@mail.gmail.com>
Message-ID: <CAEwTsAgGdi24bPQMRsdMKJiXgyrBLYi9ZV_CA4JpN-LDavJhmQ@mail.gmail.com>

yes.  is the call hitting your dial plan?

On Tuesday, May 26, 2015, Victor Medina <victor.medina at cibersys.com> wrote:

> I did both!
> I installed the WebRTC Screen Capture extension and check all command line
> option on chrome: --allow-http-screen-capture &
> --auto-select-desktop-capture-source as of ...
> http://peter.sh/experiments/chromium-command-line-switches/
>
> I think only the extension is needed, right?
>
> 2015-05-25 18:25 GMT-04:30 Michael Jerris <mike at jerris.com
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>>:
>
>> for chrome to allow screen share it requires you to install a chrome
>> plugin or start chrome with a command line argument to allow screen share
>>
>>
>> On Monday, May 25, 2015, Victor Medina <victor.medina at cibersys.com
>> <javascript:_e(%7B%7D,'cvml','victor.medina at cibersys.com');>> wrote:
>>
>>> ?Log..
>>>
>>> Attempting Screen Capture....
>>> SCREEN SHARE
>>> Audio constraints false
>>> Video constraints Object {mandatory: Object, optional:
>>> Array[0]}mandatory: Objectoptional: Array[0]__proto__: Object
>>> Stream Success
>>> stream started
>>> Offer SDP
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> offer RTCIceCandidate {}
>>> ICE Complete
>>> ICE SDP
>>> offer v=0
>>> o=- 3905402391138908374 2 IN IP4 127.0.0.1
>>> s=-
>>> t=0 0
>>> a=group:BUNDLE audio video
>>> a=msid-semantic: WMS dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>> m=audio 34382 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>> c=IN IP4 186.14.135.35
>>> a=rtcp:41117 IN IP4 186.14.135.35
>>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54692 typ host
>>> generation 0
>>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54693 typ host
>>> generation 0
>>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54694 typ host
>>> generation 0
>>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54695 typ host
>>> generation 0
>>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
>>> active generation 0
>>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>>> active generation 0
>>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
>>> active generation 0
>>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>>> active generation 0
>>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 34382 typ srflx
>>> raddr 10.0.1.10 rport 54693 generation 0
>>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 41117 typ srflx
>>> raddr 10.0.1.10 rport 54695 generation 0
>>> a=ice-ufrag:qjfFF9BX74oS8gTh
>>> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
>>> a=fingerprint:sha-256
>>> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
>>> a=setup:actpass
>>> a=mid:audio
>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>> a=recvonly
>>> a=rtcp-mux
>>> a=rtpmap:111 opus/48000/2
>>> a=fmtp:111 minptime=10; useinbandfec=1
>>> a=rtpmap:103 ISAC/16000
>>> a=rtpmap:104 ISAC/32000
>>> a=rtpmap:9 G722/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:106 CN/32000
>>> a=rtpmap:105 CN/16000
>>> a=rtpmap:13 CN/8000
>>> a=rtpmap:126 telephone-event/8000
>>> a=maxptime:60
>>> m=video 39129 RTP/SAVPF 100 116 117 96
>>> c=IN IP4 186.14.135.35
>>> a=rtcp:40989 IN IP4 186.14.135.35
>>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54696 typ host
>>> generation 0
>>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54697 typ host
>>> generation 0
>>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54698 typ host
>>> generation 0
>>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54699 typ host
>>> generation 0
>>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
>>> active generation 0
>>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>>> active generation 0
>>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
>>> active generation 0
>>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>>> active generation 0
>>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 40989 typ srflx
>>> raddr 10.0.1.10 rport 54699 generation 0
>>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 39129 typ srflx
>>> raddr 10.0.1.10 rport 54697 generation 0
>>> a=ice-ufrag:qjfFF9BX74oS8gTh
>>> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
>>> a=fingerprint:sha-256
>>> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
>>> a=setup:actpass
>>> a=mid:video
>>> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>> a=extmap:4 urn:3gpp:video-orientation
>>> a=sendrecv
>>> a=rtcp-mux
>>> a=rtpmap:100 VP8/90000
>>> a=rtcp-fb:100 ccm fir
>>> a=rtcp-fb:100 nack
>>> a=rtcp-fb:100 nack pli
>>> a=rtcp-fb:100 goog-remb
>>> a=rtpmap:116 red/90000
>>> a=rtpmap:117 ulpfec/90000
>>> a=rtpmap:96 rtx/90000
>>> a=fmtp:96 apt=100
>>> a=ssrc-group:FID 3962025390 1186212002
>>> a=ssrc:3962025390 cname:Rvg58eAlJAmWf/sm
>>> a=ssrc:3962025390 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>> a45b2502-eb86-452c-b4cb-e69e5305938a
>>> a=ssrc:3962025390 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>> a=ssrc:3962025390 label:a45b2502-eb86-452c-b4cb-e69e5305938a
>>> a=ssrc:1186212002 cname:Rvg58eAlJAmWf/sm
>>> a=ssrc:1186212002 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>> a45b2502-eb86-452c-b4cb-e69e5305938a
>>> a=ssrc:1186212002 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>> a=ssrc:1186212002 label:a45b2502-eb86-452c-b4cb-e69e5305938a
>>>
>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from new to
>>> requesting
>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from
>>> requesting to trying
>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from trying to
>>> hangup
>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from hangup to
>>> destroy
>>> ?
>>>
>>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>
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From yadenis at seznam.cz  Tue May 26 19:10:49 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Tue, 26 May 2015 17:10:49 +0200
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <CAHtxdDd=vo6uwo-7imdo7MEGuRDC2PYbqk3HyLvvD1rdfkZxRw@mail.gmail.com>
References: <43162995.20150526094002@seznam.cz> 
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
	<16210177989.20150526120710@seznam.cz>
	<CAHtxdDd=vo6uwo-7imdo7MEGuRDC2PYbqk3HyLvvD1rdfkZxRw@mail.gmail.com>
Message-ID: <1946256680.20150526171049@seznam.cz>

Dobr? den,

ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1
ln: failed to create symbolic link ?/usr/lib64/libvpx.so.1?: No such file or directory


I have libvpx.so in /usr/x_86_64_linux_gnu. But it steal not work. 

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

?ter? 26. kv?tna 2015, 14:57:11, napsal jste:


execute as root "ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1"
And try again

On Tue, May 26, 2015 at 1:07 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Dobr? den,

Where can I get the -extra package for libav?

I try from here https://freeswitch.org/stash/projects/SD/repos/libav/browse  It is compiled without problems. 

But I still have a error

2015-05-26 06:01:38.083643 [CRIT] switch_loadable_module.c:1520 Error Loading module /usr/local/freeswitch/mod/mod_av.so
**libvpx.so.1: cannot open shared object file: No such file or directory**




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

?ter? 26. kv?tna 2015, 11:22:35, napsal jste:


try using mod av instead.  You'll need to install the -extra package for libav 

On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi all !

I have one problem with the MP4 file recording with sound. The video is great. But there is no sound.

Adding a log. files in place, but for some reason does not work. I use Debian and the latest version 1.6

EXECUTE sofia/internal/1004 at 192.168.242.132 record(/usr/local/freeswitch/recordings/testrecord.mp4)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}  write video



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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From olegstolyar at gmail.com  Tue May 26 19:13:36 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Tue, 26 May 2015 08:13:36 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMO+-Wqu9_xQx6TzkUU3NLhVU=HTpDu6wO2p1TX8J1kxEKg@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAFiqYum4if-B8NGFu7PXjFpJZnPuFGy8XhWscM7p7q6BwZiqGg@mail.gmail.com>
	<CAC8JMO+-Wqu9_xQx6TzkUU3NLhVU=HTpDu6wO2p1TX8J1kxEKg@mail.gmail.com>
Message-ID: <CAC8JMO+YUezb5peCD0njovq83YwNYQMEx6ASee96RRq2-0Ocmg@mail.gmail.com>

Just FYI, the JsSip engineers replied to my questions with this:

The error happens because the incoming reINVITE has a=setup:active in
the SDP which is a bug in FreeSwitch (the RFC clearly states that the
SDP offer MUST have a=setup:actpass):

On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:

> Thanks Steven!
>
> It may be https://freeswitch.org/jira/browse/FS-7040.
>
> As far as the 120 sec is concerned - Below are snippets from both the
> invite and the 200 OK from FS.  I know that FS reads the Session-Expires
> from the client because if I change it to a value less than 120, FS sends
> back a "SIP/2.0 422 Session Interval Too Small".  I  even thought the
> problem could be that the Session-Exipres format in the original INVITE is
> incorrect since it does not contain ";refresher=uac" but when I added that
> and made the line "Session-Expires: 300;refresher=uac" nothing changed.
>
>
> INVITE sip:echo-test at anonymous.invalid SIP/2.0
> Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029
> Max-Forwards: 69
> To: <sip:echo-test at anonymous.invalid>
> From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
> Call-ID: r65e48gp171p21rkppcu
> CSeq: 6621 INVITE
> Contact: <sip:mqthb9dj at pfcnm9rjv5en.invalid;transport=ws;ob>
> Content-Type: application/sdp
> Session-Expires: 300
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
> Supported: timer,ice,outbound
> User-Agent: JsSIP 0.6.26
> Content-Length: 2754
>
> SIP/2.0 200 OK
> Via: SIP/2.0/WSS
> pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200
> From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
> To: <sip:echo-test at anonymous.invalid>;tag=SgjX0X4arHUFg
> Call-ID: r65e48gp171p21rkppcu
> CSeq: 6621 INVITE
> Contact: <sip:echo-test at 50.19.255.62:5080;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Require: timer
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
> line-seize, call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
> Session-Expires: 120;refresher=uac
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 809
> X-Netflix: yes
> Remote-Party-ID: "echo-test" <sip:echo-test at anonymous.invalid
> >;party=calling;privacy=off;screen=no
>
>
> On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> The values to actual use are negotiated during the call invite, and your
>> trace shows it's using 2 minutes.
>> Session-Expires: 120;refresher=uac
>> Min-SE: 120
>>
>> Just as an idea because you didn't send both invite and re-invite...
>> perhaps the SDP body is different in the reinvite without the version
>> number having changed. If so it may be
>> https://freeswitch.org/jira/browse/FS-7040
>>
>> On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>
>>> But isn't that based on the session-timeout param which defaults to 30
>>> min?  My re-invites occur much sooner than 30 min into a call.  Or does
>>> session-timeout param only control sessions initiated by FS while incoming
>>> sessions use the minimum-session-expires param if it's not explicitly
>>> passed by the session initiator?
>>>
>>> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>>
>>>> session timer
>>>>
>>>>
>>>> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>>>
>>>>> Thanks Michael, I'll see if we can do that!
>>>>>
>>>>> So, is the re-INVITE legit and the problem is that JsSip does not
>>>>> respond to it?  Still, I am curious what is triggering the re-INVITE.
>>>>>
>>>>> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>>> I think the sip.js guys fixed this issue when they forked jssip.  I'd
>>>>>> suggest using that.
>>>>>>
>>>>>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>>>>> wrote:
>>>>>> >
>>>>>> > Hi guys,
>>>>>> >
>>>>>> > Several weeks ago I started getting an occasional problem where FS
>>>>>> is sending an INVITE to the other side in the middle of a call, the other
>>>>>> side does not respond and FS hangs up the leg.  Below is the relevant log.
>>>>>> The user experience is that they keep talking and hearing each other up to
>>>>>> the very end.  I have a recording of that call, so can confirm.
>>>>>> >
>>>>>> > The call uses WebRTC and is originated by JsSip from Chrome.  Then
>>>>>> the user is put into a conference but I doubt it's relevant in this case
>>>>>> since the INVITE and disconnect are not happening from mod_conference
>>>>>> >
>>>>>> > I suspect it's a re-INVITE but what triggers FS to send it?  I
>>>>>> couldn't find anything in the logs that could shed light.
>>>>>> >
>>>>>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>>>>>> >
>>>>>> ------------------------------------------------------------------------
>>>>>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>>>>>> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>>>>>> >    Max-Forwards: 70
>>>>>> >    From:
>>>>>> <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>>>>>> >;tag=KQecUSr12rSQp
>>>>>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>>>>>> >    Call-ID: g8980rbrbk2t45oj5mru
>>>>>> >    CSeq: 75703945 INVITE
>>>>>> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@
>>>>>> ##.##.###.###:5080;transport=udp>
>>>>>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>>>>>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>>>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>> >    Supported: timer, path, replaces
>>>>>> >    Session-Expires: 120;refresher=uac
>>>>>> >    Min-SE: 120
>>>>>> >    Content-Type: application/sdp
>>>>>> >    Content-Disposition: session
>>>>>> >    Content-Length: 825
>>>>>> >
>>>>>> >    v=0
>>>>>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>>>>>> >    s=FreeSWITCH
>>>>>> >    c=IN IP4 ##.##.##.##
>>>>>> >    t=0 0
>>>>>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>>> >    m=audio 22670 RTP/SAVPF 9 126 106
>>>>>> >    a=rtpmap:9 G722/8000
>>>>>> >    a=rtpmap:126 telephone-event/8000
>>>>>> >    a=rtpmap:106 CN/8000
>>>>>> >    a=ptime:20
>>>>>> >    a=fingerprint:sha-256
>>>>>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>>>>>> >    a=rtcp-mux
>>>>>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>>>>>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>>>>>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>>>>>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>>>>>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>>>>>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>>>>>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ
>>>>>> host generation 0
>>>>>> >
>>>>>> ------------------------------------------------------------------------
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> entering state [calling][0]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> entering state [terminating][503]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [DEBUG] switch_channel.c:3242 Send signal
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>> [DEBUG] switch_core_session.c:1396 Send signal
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] mod_conference.c:5057 Channel leaving conference, cause:
>>>>>> NORMAL_TEMPORARY_FAILURE
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> skip receive message [UNBRIDGE] (channel is hungup already)
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_media.c:7772
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>> [REFER_EVENT] (channel is hungup already)
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> Restore previous codec G722:9.
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_session.c:2901
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:535
>>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going
>>>>>> to sleep
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:472
>>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>>>>>> CS_HANGUP
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:735
>>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change
>>>>>> ACTIVE -> HANGUP
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:737
>>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
>>>>>> hanging up, cause: NORMAL_TEMPORARY_FAILURE
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:60
>>>>>> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause:
>>>>>> NORMAL_TEMPORARY_FAILURE
>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>> [DEBUG] switch_core_state_machine.c:737
>>>>>> (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to
>>>>>> sleep
>>>>>> >
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.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.medina at cibersys.com  Tue May 26 19:19:45 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Tue, 26 May 2015 10:49:45 -0430
Subject: [Freeswitch-users] How to enable Screen sharing on Verto Conf?
In-Reply-To: <CAEwTsAgGdi24bPQMRsdMKJiXgyrBLYi9ZV_CA4JpN-LDavJhmQ@mail.gmail.com>
References: <CACXsapOA9nG0uo3+artAT72DD17umf9Ck5dHCKjPXDzjd1+s_Q@mail.gmail.com>
	<CACXsapOLEbJA9Dfbfc1YMOnSHa73eTMnm+NUG5qs=N0Xz3gzYQ@mail.gmail.com>
	<CAEwTsAhpyeuBuCXB1bePAiOXPP8RtaYMgaSiiLjP6J+5aMVyFQ@mail.gmail.com>
	<CACXsapM8YLag7fauLZpqJrApqiiModv72HAiaOAnc+HGJKV6dQ@mail.gmail.com>
	<CAEwTsAgGdi24bPQMRsdMKJiXgyrBLYi9ZV_CA4JpN-LDavJhmQ@mail.gmail.com>
Message-ID: <CACXsapPmFHUdq9QVvO6kxpBSmYZEii1EDS32wLHahYb1gUdhdA@mail.gmail.com>

Im seeing this while starting the screen sharing...

2015-05-26 11:15:36.690561 [NOTICE] switch_channel.c:1089 New Channel
verto.rtc/3500-screen [d4576cb3-e967-2bb1-be3c-a4831f8894fd]
2015-05-26 11:15:36.690561 [DEBUG] mod_verto.c:3407 Remote SDP
verto.rtc/3500-screen:
v=0
o=- 4034581275591452868 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
m=audio 33952 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 186.14.135.35
a=rtcp:33426 IN IP4 186.14.135.35
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 58898 typ host
generation 0
a=candidate:648569486 1 udp 2122129151 10.0.1.10 58899 typ host generation 0
a=candidate:2999745851 2 udp 2122194686 192.168.56.1 58900 typ host
generation 0
a=candidate:648569486 2 udp 2122129150 10.0.1.10 58901 typ host generation 0
a=candidate:1680186374 1 udp 1685921535 186.14.135.35 33952 typ srflx raddr
10.0.1.10 rport 58899 generation 0
a=candidate:1680186374 2 udp 1685921534 186.14.135.35 33426 typ srflx raddr
10.0.1.10 rport 58901 generation 0
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
active generation 0
a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype active
generation 0
a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
active generation 0
a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype active
generation 0
a=ice-ufrag:DXEOOI0UFrnM6fo8
a=ice-pwd:l3x/4ga4HunTyP6rGwExMzgv
a=fingerprint:sha-256
BF:00:AC:5F:C8:84:54:5F:EF:34:3C:EC:AF:1C:92:80:B6:16:45:EF:3A:39:A1:3C:9E:AF:88:B0:E3:06:BA:AC
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
m=video 42360 RTP/SAVPF 100 116 117 96
c=IN IP4 186.14.135.35
a=rtcp:44680 IN IP4 186.14.135.35
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 58902 typ host
generation 0
a=candidate:648569486 1 udp 2122129151 10.0.1.10 58903 typ host generation 0
a=candidate:2999745851 2 udp 2122194686 192.168.56.1 58904 typ host
generation 0
a=candidate:648569486 2 udp 2122129150 10.0.1.10 58905 typ host generation 0
a=candidate:1680186374 1 udp 1685921535 186.14.135.35 42360 typ srflx raddr
10.0.1.10 rport 58903 generation 0
a=candidate:1680186374 2 udp 1685921534 186.14.135.35 44680 typ srflx raddr
10.0.1.10 rport 58905 generation 0
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
active generation 0
a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype active
generation 0
a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
active generation 0
a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype active
generation 0
a=ice-ufrag:DXEOOI0UFrnM6fo8
a=ice-pwd:l3x/4ga4HunTyP6rGwExMzgv
a=fingerprint:sha-256
BF:00:AC:5F:C8:84:54:5F:EF:34:3C:EC:AF:1C:92:80:B6:16:45:EF:3A:39:A1:3C:9E:AF:88:B0:E3:06:BA:AC
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=ssrc-group:FID 2871497692 3867014262
a=ssrc:2871497692 cname:e4V6+GZYAdJCwcTf
a=ssrc:2871497692 msid:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
ef8fba5e-3643-4dbd-a038-5023f4521ba4
a=ssrc:2871497692 mslabel:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
a=ssrc:2871497692 label:ef8fba5e-3643-4dbd-a038-5023f4521ba4
a=ssrc:3867014262 cname:e4V6+GZYAdJCwcTf
a=ssrc:3867014262 msid:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
ef8fba5e-3643-4dbd-a038-5023f4521ba4
a=ssrc:3867014262 mslabel:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
a=ssrc:3867014262 label:ef8fba5e-3643-4dbd-a038-5023f4521ba4

2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
(verto.rtc/3500-screen) Running State Change CS_INIT
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:516
(verto.rtc/3500-screen) State INIT
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:40
verto.rtc/3500-screen Standard INIT
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:48
(verto.rtc/3500-screen) State Change CS_INIT -> CS_ROUTING
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:516
(verto.rtc/3500-screen) State INIT going to sleep
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
(verto.rtc/3500-screen) Running State Change CS_ROUTING
2015-05-26 11:15:36.690561 [DEBUG] switch_channel.c:2234
(verto.rtc/3500-screen) Callstate Change DOWN -> RINGING
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:532
(verto.rtc/3500-screen) State ROUTING
2015-05-26 11:15:36.690561 [DEBUG] mod_rtc.c:89 verto.rtc/3500-screen RTC
ROUTING
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:166
verto.rtc/3500-screen Standard ROUTING
2015-05-26 11:15:36.690561 [INFO] mod_dialplan_xml.c:636 Processing Victor
Medina (Screen) <1000 (screen)>->3500-screen in context default
Dialplan: verto.rtc/3500-screen parsing [default->unloop] continue=false
Dialplan: verto.rtc/3500-screen Regex (PASS) [unloop] ${unroll_loops}(true)
=~ /^true$/ break=on-false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [unloop] ${sip_looped_call}()
=~ /^true$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->tod_example] continue=true
Dialplan: verto.rtc/3500-screen Date/Time Match (PASS) [tod_example]
break=on-false
Dialplan: verto.rtc/3500-screen Action set(open=true)
Dialplan: verto.rtc/3500-screen parsing [default->holiday_example]
continue=true
Dialplan: verto.rtc/3500-screen Date/TimeMatch (FAIL) [holiday_example]
break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->global-intercept]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [global-intercept]
destination_number(3500-screen) =~ /^886$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->group-intercept]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [group-intercept]
destination_number(3500-screen) =~ /^\*8$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->intercept-ext]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [intercept-ext]
destination_number(3500-screen) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->redial] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [redial]
destination_number(3500-screen) =~ /^(redial|870)$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->global] continue=true
Dialplan: verto.rtc/3500-screen Regex (FAIL) [global] ${call_debug}(false)
=~ /^true$/ break=never
Dialplan: verto.rtc/3500-screen Regex (FAIL) [global]
${default_password}(p6rSp6rSqwerty) =~ /^1234$/ break=never
Dialplan: verto.rtc/3500-screen Regex (FAIL) [global] ${rtp_has_crypto}()
=~
/^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/
break=never
Dialplan: verto.rtc/3500-screen Regex (FAIL) [global]
${endpoint_disposition}() =~ /^(DELAYED NEGOTIATION)/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->snom-demo-2]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [snom-demo-2]
destination_number(3500-screen) =~ /^9001$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->snom-demo-1]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [snom-demo-1]
destination_number(3500-screen) =~ /^9000$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->eavesdrop] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [eavesdrop]
destination_number(3500-screen) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->eavesdrop] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [eavesdrop]
destination_number(3500-screen) =~ /^779$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->call_return]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [call_return]
destination_number(3500-screen) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->del-group] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [del-group]
destination_number(3500-screen) =~ /^80(\d{2})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->add-group] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [add-group]
destination_number(3500-screen) =~ /^81(\d{2})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->call-group-simo]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [call-group-simo]
destination_number(3500-screen) =~ /^82(\d{2})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->call-group-order]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [call-group-order]
destination_number(3500-screen) =~ /^83(\d{2})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->extension-intercom]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [extension-intercom]
destination_number(3500-screen) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->Local_Extension]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [Local_Extension]
destination_number(3500-screen) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->Local_Extension_Skinny]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [Local_Extension_Skinny]
destination_number(3500-screen) =~ /^(11[01][0-9])$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->group_dial_sales]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [group_dial_sales]
destination_number(3500-screen) =~ /^2000$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->group_dial_support]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [group_dial_support]
destination_number(3500-screen) =~ /^2001$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->group_dial_billing]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [group_dial_billing]
destination_number(3500-screen) =~ /^2002$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->operator] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [operator]
destination_number(3500-screen) =~ /^(operator|0)$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->vmain] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [vmain]
destination_number(3500-screen) =~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->sip_uri] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [sip_uri]
destination_number(3500-screen) =~ /^sip:(.*)$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->nb_conferences]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [nb_conferences]
destination_number(3500-screen) =~ /^(30\d{2})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->wb_conferences]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [wb_conferences]
destination_number(3500-screen) =~ /^(31\d{2})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->uwb_conferences]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [uwb_conferences]
destination_number(3500-screen) =~ /^(32\d{2})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->cdquality_conferences]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [cdquality_conferences]
destination_number(3500-screen) =~ /^(33\d{2})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->cdquality_conferences]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [cdquality_conferences]
destination_number(3500-screen) =~ /^(35\d{2})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing
[default->freeswitch_public_conf_via_sip] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL)
[freeswitch_public_conf_via_sip] destination_number(3500-screen) =~
/^9(888|8888|1616|3232)$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->mad_boss_intercom]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [mad_boss_intercom]
destination_number(3500-screen) =~ /^0911$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->mad_boss_intercom]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [mad_boss_intercom]
destination_number(3500-screen) =~ /^0912$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->mad_boss] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [mad_boss]
destination_number(3500-screen) =~ /^0913$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->ivr_demo] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [ivr_demo]
destination_number(3500-screen) =~ /^5000$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->dynamic_conference]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [dynamic_conference]
destination_number(3500-screen) =~ /^5001$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->rtp_multicast_page]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [rtp_multicast_page]
destination_number(3500-screen) =~ /^pagegroup$|^7243$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->park] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [park]
destination_number(3500-screen) =~ /^5900$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->unpark] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [unpark]
destination_number(3500-screen) =~ /^5901$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->valet_park] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [valet_park]
destination_number(3500-screen) =~ /^(6000)$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->valet_park] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [valet_park]
destination_number(3500-screen) =~ /^((?!6000)60\d{2})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->park] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [park] source(mod_verto) =~
/mod_sofia/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->unpark] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [unpark] source(mod_verto) =~
/mod_sofia/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->park] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [park] source(mod_verto) =~
/mod_sofia/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->unpark] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [unpark] source(mod_verto) =~
/mod_sofia/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->wait] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [wait]
destination_number(3500-screen) =~ /^wait$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->fax_receive]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [fax_receive]
destination_number(3500-screen) =~ /^9178$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->fax_transmit]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [fax_transmit]
destination_number(3500-screen) =~ /^9179$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->ringback_180]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_180]
destination_number(3500-screen) =~ /^9180$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->ringback_183_uk_ring]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_183_uk_ring]
destination_number(3500-screen) =~ /^9181$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->ringback_183_music_ring]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_183_music_ring]
destination_number(3500-screen) =~ /^9182$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing
[default->ringback_post_answer_uk_ring] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_post_answer_uk_ring]
destination_number(3500-screen) =~ /^9183$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing
[default->ringback_post_answer_music] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_post_answer_music]
destination_number(3500-screen) =~ /^9184$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->ClueCon] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [ClueCon]
destination_number(3500-screen) =~ /^9191$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->show_info] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [show_info]
destination_number(3500-screen) =~ /^9192$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->video_record]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [video_record]
destination_number(3500-screen) =~ /^9193$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->video_playback]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [video_playback]
destination_number(3500-screen) =~ /^9194$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->delay_echo] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [delay_echo]
destination_number(3500-screen) =~ /^9195$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->echo] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [echo]
destination_number(3500-screen) =~ /^9196$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->milliwatt] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [milliwatt]
destination_number(3500-screen) =~ /^9197$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->tone_stream]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [tone_stream]
destination_number(3500-screen) =~ /^9198$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->zrtp_enrollement]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [zrtp_enrollement]
destination_number(3500-screen) =~ /^9787$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->hold_music] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [hold_music]
destination_number(3500-screen) =~ /^9664$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->laugh break]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [laugh break]
destination_number(3500-screen) =~ /^9386$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->h] continue=true
Dialplan: verto.rtc/3500-screen Regex (FAIL) [h]
destination_number(3500-screen) =~ /^h264_(.*)$/ break=never
Dialplan: verto.rtc/3500-screen parsing [default->v] continue=true
Dialplan: verto.rtc/3500-screen Regex (FAIL) [v]
destination_number(3500-screen) =~ /^vp8_(.*)$/ break=never
Dialplan: verto.rtc/3500-screen parsing [default->h] continue=true
Dialplan: verto.rtc/3500-screen Regex (FAIL) [h]
destination_number(3500-screen) =~ /^hbr_(.*)$/ break=never
Dialplan: verto.rtc/3500-screen parsing [default->v] continue=true
Dialplan: verto.rtc/3500-screen Regex (FAIL) [v]
destination_number(3500-screen) =~ /^vbr_(.*)$/ break=never
Dialplan: verto.rtc/3500-screen parsing [default->bug] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [bug]
destination_number(3500-screen) =~ /^vbr_(.*)$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->bug] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [bug]
destination_number(3500-screen) =~ /^vid_(.*)$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->bug] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [bug]
destination_number(3500-screen) =~ /^decode$|^9952$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->101] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [101]
destination_number(3500-screen) =~ /^101$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->pizza_demo] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [pizza_demo]
destination_number(3500-screen) =~ /^(pizza|74992)$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->Talking Clock Time]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [Talking Clock Time]
destination_number(3500-screen) =~ /^9170$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->Talking Clock Date]
continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [Talking Clock Date]
destination_number(3500-screen) =~ /^9171$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->Talking Clock Date and
Time] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [Talking Clock Date and Time]
destination_number(3500-screen) =~ /^9172$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->local.example.com]
continue=false
Dialplan: verto.rtc/3500-screen Regex (PASS) [local.example.com]
${toll_allow}(domestic,international,local) =~ /local/ break=on-false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [local.example.com]
destination_number(3500-screen) =~ /^(\d{7})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->domestic.example.com]
continue=false
Dialplan: verto.rtc/3500-screen Regex (PASS) [domestic.example.com]
${toll_allow}(domestic,international,local) =~ /domestic/ break=on-false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [domestic.example.com]
destination_number(3500-screen) =~ /^(\d{11})$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->international.example.com]
continue=false
Dialplan: verto.rtc/3500-screen Regex (PASS) [international.example.com]
${toll_allow}(domestic,international,local) =~ /international/
break=on-false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [international.example.com]
destination_number(3500-screen) =~ /^(011\d+)$/ break=on-false
Dialplan: verto.rtc/3500-screen parsing [default->enum] continue=false
Dialplan: verto.rtc/3500-screen Regex (FAIL) [enum]
${module_exists(mod_enum)}(false) =~ /true/ break=on-false
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:216
(verto.rtc/3500-screen) State Change CS_ROUTING -> CS_EXECUTE
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:532
(verto.rtc/3500-screen) State ROUTING going to sleep
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
(verto.rtc/3500-screen) Running State Change CS_EXECUTE
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:539
(verto.rtc/3500-screen) State EXECUTE
2015-05-26 11:15:36.690561 [DEBUG] mod_rtc.c:120 verto.rtc/3500-screen RTC
EXECUTE
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:258
verto.rtc/3500-screen Standard EXECUTE
EXECUTE verto.rtc/3500-screen set(open=true)
2015-05-26 11:15:36.690561 [DEBUG] mod_dptools.c:1469 verto.rtc/3500-screen
SET [open]=[true]
2015-05-26 11:15:36.690561 [NOTICE] switch_core_state_machine.c:315
verto.rtc/3500-screen has executed the last dialplan instruction, hanging
up.
2015-05-26 11:15:36.690561 [NOTICE] switch_core_state_machine.c:317 Hangup
verto.rtc/3500-screen [CS_EXECUTE] [NORMAL_CLEARING]
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:539
(verto.rtc/3500-screen) State EXECUTE going to sleep
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
(verto.rtc/3500-screen) Running State Change CS_HANGUP
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:739
(verto.rtc/3500-screen) Callstate Change RINGING -> HANGUP
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:741
(verto.rtc/3500-screen) State HANGUP
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:60
verto.rtc/3500-screen Standard HANGUP, cause: NORMAL_CLEARING
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:741
(verto.rtc/3500-screen) State HANGUP going to sleep
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:508
(verto.rtc/3500-screen) State Change CS_HANGUP -> CS_REPORTING
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
(verto.rtc/3500-screen) Running State Change CS_REPORTING
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:827
(verto.rtc/3500-screen) State REPORTING
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:104
verto.rtc/3500-screen Standard REPORTING, cause: NORMAL_CLEARING
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:827
(verto.rtc/3500-screen) State REPORTING going to sleep
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:499
(verto.rtc/3500-screen) State Change CS_REPORTING -> CS_DESTROY
2015-05-26 11:15:36.690561 [DEBUG] switch_core_session.c:1638 Session 4
(verto.rtc/3500-screen) Locked, Waiting on external entities
2015-05-26 11:15:36.690561 [NOTICE] switch_core_session.c:1656 Session 4
(verto.rtc/3500-screen) Ended
2015-05-26 11:15:36.690561 [NOTICE] switch_core_session.c:1660 Close
Channel verto.rtc/3500-screen [CS_DESTROY]
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:630
(verto.rtc/3500-screen) Running State Change CS_DESTROY
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:640
(verto.rtc/3500-screen) State DESTROY
2015-05-26 11:15:36.690561 [DEBUG] mod_rtc.c:132 verto.rtc/3500-screen RTC
DESTROY
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:111
verto.rtc/3500-screen Standard DESTROY
2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:640
(verto.rtc/3500-screen) State DESTROY going to sleep
freeswitch at internal>


2015-05-26 10:24 GMT-04:30 Michael Jerris <mike at jerris.com>:

> yes.  is the call hitting your dial plan?
>
>
> On Tuesday, May 26, 2015, Victor Medina <victor.medina at cibersys.com>
> wrote:
>
>> I did both!
>> I installed the WebRTC Screen Capture extension and check all command
>> line option on chrome: --allow-http-screen-capture &
>> --auto-select-desktop-capture-source as of ...
>> http://peter.sh/experiments/chromium-command-line-switches/
>>
>> I think only the extension is needed, right?
>>
>> 2015-05-25 18:25 GMT-04:30 Michael Jerris <mike at jerris.com>:
>>
>>> for chrome to allow screen share it requires you to install a chrome
>>> plugin or start chrome with a command line argument to allow screen share
>>>
>>>
>>> On Monday, May 25, 2015, Victor Medina <victor.medina at cibersys.com>
>>> wrote:
>>>
>>>> ?Log..
>>>>
>>>> Attempting Screen Capture....
>>>> SCREEN SHARE
>>>> Audio constraints false
>>>> Video constraints Object {mandatory: Object, optional:
>>>> Array[0]}mandatory: Objectoptional: Array[0]__proto__: Object
>>>> Stream Success
>>>> stream started
>>>> Offer SDP
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> offer RTCIceCandidate {}
>>>> ICE Complete
>>>> ICE SDP
>>>> offer v=0
>>>> o=- 3905402391138908374 2 IN IP4 127.0.0.1
>>>> s=-
>>>> t=0 0
>>>> a=group:BUNDLE audio video
>>>> a=msid-semantic: WMS dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>> m=audio 34382 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>>> c=IN IP4 186.14.135.35
>>>> a=rtcp:41117 IN IP4 186.14.135.35
>>>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54692 typ host
>>>> generation 0
>>>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54693 typ host
>>>> generation 0
>>>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54694 typ host
>>>> generation 0
>>>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54695 typ host
>>>> generation 0
>>>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host
>>>> tcptype active generation 0
>>>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>>>> active generation 0
>>>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host
>>>> tcptype active generation 0
>>>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>>>> active generation 0
>>>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 34382 typ srflx
>>>> raddr 10.0.1.10 rport 54693 generation 0
>>>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 41117 typ srflx
>>>> raddr 10.0.1.10 rport 54695 generation 0
>>>> a=ice-ufrag:qjfFF9BX74oS8gTh
>>>> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
>>>> a=fingerprint:sha-256
>>>> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
>>>> a=setup:actpass
>>>> a=mid:audio
>>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>> a=recvonly
>>>> a=rtcp-mux
>>>> a=rtpmap:111 opus/48000/2
>>>> a=fmtp:111 minptime=10; useinbandfec=1
>>>> a=rtpmap:103 ISAC/16000
>>>> a=rtpmap:104 ISAC/32000
>>>> a=rtpmap:9 G722/8000
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:106 CN/32000
>>>> a=rtpmap:105 CN/16000
>>>> a=rtpmap:13 CN/8000
>>>> a=rtpmap:126 telephone-event/8000
>>>> a=maxptime:60
>>>> m=video 39129 RTP/SAVPF 100 116 117 96
>>>> c=IN IP4 186.14.135.35
>>>> a=rtcp:40989 IN IP4 186.14.135.35
>>>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54696 typ host
>>>> generation 0
>>>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54697 typ host
>>>> generation 0
>>>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54698 typ host
>>>> generation 0
>>>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54699 typ host
>>>> generation 0
>>>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host
>>>> tcptype active generation 0
>>>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>>>> active generation 0
>>>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host
>>>> tcptype active generation 0
>>>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>>>> active generation 0
>>>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 40989 typ srflx
>>>> raddr 10.0.1.10 rport 54699 generation 0
>>>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 39129 typ srflx
>>>> raddr 10.0.1.10 rport 54697 generation 0
>>>> a=ice-ufrag:qjfFF9BX74oS8gTh
>>>> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
>>>> a=fingerprint:sha-256
>>>> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
>>>> a=setup:actpass
>>>> a=mid:video
>>>> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>> a=extmap:4 urn:3gpp:video-orientation
>>>> a=sendrecv
>>>> a=rtcp-mux
>>>> a=rtpmap:100 VP8/90000
>>>> a=rtcp-fb:100 ccm fir
>>>> a=rtcp-fb:100 nack
>>>> a=rtcp-fb:100 nack pli
>>>> a=rtcp-fb:100 goog-remb
>>>> a=rtpmap:116 red/90000
>>>> a=rtpmap:117 ulpfec/90000
>>>> a=rtpmap:96 rtx/90000
>>>> a=fmtp:96 apt=100
>>>> a=ssrc-group:FID 3962025390 1186212002
>>>> a=ssrc:3962025390 cname:Rvg58eAlJAmWf/sm
>>>> a=ssrc:3962025390 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>> a45b2502-eb86-452c-b4cb-e69e5305938a
>>>> a=ssrc:3962025390 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>> a=ssrc:3962025390 label:a45b2502-eb86-452c-b4cb-e69e5305938a
>>>> a=ssrc:1186212002 cname:Rvg58eAlJAmWf/sm
>>>> a=ssrc:1186212002 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>> a45b2502-eb86-452c-b4cb-e69e5305938a
>>>> a=ssrc:1186212002 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>> a=ssrc:1186212002 label:a45b2502-eb86-452c-b4cb-e69e5305938a
>>>>
>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from new to
>>>> requesting
>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from
>>>> requesting to trying
>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from trying
>>>> to hangup
>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from hangup
>>>> to destroy
>>>> ?
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>>
>>
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561
>> BB #79A8AFA2
>> @VMCibersys
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From aqsyounas at gmail.com  Tue May 26 19:34:05 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 26 May 2015 08:34:05 -0700
Subject: [Freeswitch-users] user registration with sip.js
Message-ID: <CAGVOjVVL80T=V5qsT=802TyuvkkJhWbuXdJS3_rTwBVkuq+xaA@mail.gmail.com>

Hi, Users.
I am trying to register default freeswitch user "1003" through webrtc using
sip.js.
This is my "*index.html*"

<html>
  <head>
  </head>
  <body>
    <script src="sip-0.7.0.js"></script>
    <script src="example.js"></script>
  </body>
</html>

and *example.js*
var config = {

// Replace this IP address with your FreeSWITCH IP address
 uri: '1003 at 192.168.1.30',

  // Replace this IP address with your FreeSWITCH IP address
  // and replace the port with your FreeSWITCH port
  ws_servers: 'ws://192.168.1.30:5066',

  // FreeSWITCH Default Username
  authorizationUser: '1003',

  // FreeSWITCH Default Password
  password: '1234'
  register: true

};

var bob = new SIP.UA(config);

bob.start();

Both files are placed inside apache folder with name freeswitch_rtc.
When i run in my firefox.

*http://192.168.1.30/freeswitch_rtc/ <http://192.168.1.30/freeswitch_rtc/>*

I see nothing neither on fs_curl, nor tshark shows any register packet on
freeswitch ip.

Pardon me for my naive question, just learning how this amazing machine
(freeswitch) works.

Can someone please tell me what is wrong with this code. Or provide me some
simple example to make a call with webrtc or any tutorial.

Thanks.
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From victor.medina at cibersys.com  Tue May 26 19:49:41 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Tue, 26 May 2015 11:19:41 -0430
Subject: [Freeswitch-users] user registration with sip.js
In-Reply-To: <CAGVOjVVL80T=V5qsT=802TyuvkkJhWbuXdJS3_rTwBVkuq+xaA@mail.gmail.com>
References: <CAGVOjVVL80T=V5qsT=802TyuvkkJhWbuXdJS3_rTwBVkuq+xaA@mail.gmail.com>
Message-ID: <CACXsapMnCT9wTQQiGgYurOd=UQsS81iW25UxZAMs1yTH_8emKQ@mail.gmail.com>

Did you anable the webrtc endpoint in your profiles?

2015-05-26 11:04 GMT-04:30 Aqs Younas <aqsyounas at gmail.com>:

> Hi, Users.
> I am trying to register default freeswitch user "1003" through webrtc
> using sip.js.
> This is my "*index.html*"
>
> <html>
>   <head>
>   </head>
>   <body>
>     <script src="sip-0.7.0.js"></script>
>     <script src="example.js"></script>
>   </body>
> </html>
>
> and *example.js*
> var config = {
>
> // Replace this IP address with your FreeSWITCH IP address
>  uri: '1003 at 192.168.1.30',
>
>   // Replace this IP address with your FreeSWITCH IP address
>   // and replace the port with your FreeSWITCH port
>   ws_servers: 'ws://192.168.1.30:5066',
>
>   // FreeSWITCH Default Username
>   authorizationUser: '1003',
>
>   // FreeSWITCH Default Password
>   password: '1234'
>   register: true
>
> };
>
> var bob = new SIP.UA(config);
>
> bob.start();
>
> Both files are placed inside apache folder with name freeswitch_rtc.
> When i run in my firefox.
>
> *http://192.168.1.30/freeswitch_rtc/ <http://192.168.1.30/freeswitch_rtc/>*
>
> I see nothing neither on fs_curl, nor tshark shows any register packet on
> freeswitch ip.
>
> Pardon me for my naive question, just learning how this amazing machine
> (freeswitch) works.
>
> Can someone please tell me what is wrong with this code. Or provide me
> some simple example to make a call with webrtc or any tutorial.
>
> 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
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From victor.medina at cibersys.com  Tue May 26 19:51:35 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Tue, 26 May 2015 11:21:35 -0430
Subject: [Freeswitch-users] Video always muted in conference (fs_video2)
In-Reply-To: <CAEwTsAgsVXvE5Ym9TWvbSvd9i7gCgNWwmp-2adxYKug0RHHKBA@mail.gmail.com>
References: <CAFKjMAYjKz32rE-G_aA639J5b27eNwnRAME31La25W9KiWbagA@mail.gmail.com>
	<7B131C27-7233-4E88-BB9E-E01BE588DE82@jerris.com>
	<CAFKjMAZ30sxNtAXU1iqksuNwM53gzQaqFQor7vhVHhg-3VBn5w@mail.gmail.com>
	<CAEwTsAgsVXvE5Ym9TWvbSvd9i7gCgNWwmp-2adxYKug0RHHKBA@mail.gmail.com>
Message-ID: <CACXsapNJ_ErsOdZzc=_f4DJpzYe6NRGeNB8uRE1Uw1BSAaGOJg@mail.gmail.com>

Michael...

I just build FS with the latest sources and the Video Unmuted/Video Muted
ramains.
Just to confirm this.

If you guys need me to test anything I have a full blown server just for me
and fs tests.

2015-05-25 0:08 GMT-04:30 Michael Jerris <mike at jerris.com>:

> Some fixes were pushed for this on Friday.  Try out latest code and see if
> that works any better for you?
>
>
> On Saturday, May 23, 2015, Carlos Gonz?lez Florido <carlosj.gf at gmail.com>
> wrote:
>
>> Ok, tell me if I can help with the testing.
>> Is there a way to disconnect ipv6?
>>
>> On Fri, May 22, 2015 at 6:04 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> I don't think its actually muted.  We are chasing down an issue that
>>> sounds just like this.  It might be related to ipv6 but we are still
>>> looking in to it.
>>>
>>> > On May 22, 2015, at 8:40 AM, Carlos Gonz?lez Florido <
>>> carlosj.gf at gmail.com> wrote:
>>> >
>>> > Hello,
>>> >
>>> > I'm testing the impressive fs_video2 branch, but I have the following
>>> problems:
>>> >
>>> > - some participants have their video always muted (and their camera is
>>> on and working for Hangouts, for example).
>>> > - other participants start the same (video muted), but after 3-5
>>> minutes the video is automatically unmuted.
>>> > - if we do a layout change (using fs_cli), all participants (having
>>> video muted or unmuted at that moment) go to muted video for the rest of
>>> the conference.
>>> >
>>> > What is the reason for fs to automatically mute the video? Is this
>>> expected or does it look like a bug? Can this behaviour be tuned or
>>> disconnected?
>>> >
>>> > Thank you,
>>> > Carlos Gonzalez
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From anthony.minessale at gmail.com  Tue May 26 19:54:07 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 May 2015 10:54:07 -0500
Subject: [Freeswitch-users] How to enable Screen sharing on Verto Conf?
In-Reply-To: <CACXsapPmFHUdq9QVvO6kxpBSmYZEii1EDS32wLHahYb1gUdhdA@mail.gmail.com>
References: <CACXsapOA9nG0uo3+artAT72DD17umf9Ck5dHCKjPXDzjd1+s_Q@mail.gmail.com>
	<CACXsapOLEbJA9Dfbfc1YMOnSHa73eTMnm+NUG5qs=N0Xz3gzYQ@mail.gmail.com>
	<CAEwTsAhpyeuBuCXB1bePAiOXPP8RtaYMgaSiiLjP6J+5aMVyFQ@mail.gmail.com>
	<CACXsapM8YLag7fauLZpqJrApqiiModv72HAiaOAnc+HGJKV6dQ@mail.gmail.com>
	<CAEwTsAgGdi24bPQMRsdMKJiXgyrBLYi9ZV_CA4JpN-LDavJhmQ@mail.gmail.com>
	<CACXsapPmFHUdq9QVvO6kxpBSmYZEii1EDS32wLHahYb1gUdhdA@mail.gmail.com>
Message-ID: <CAKbxfG-WD03p7V=+myKHL9GypknQozzfqcJJtVK3Z+kL=M_JZw@mail.gmail.com>

The extension it calls is the same extension the original call is calling
only it also adds -screen to the end so you need your dialplan to catch
that as well and do whatever differently you may do such as not setting
banners etc.


On Tue, May 26, 2015 at 10:19 AM, Victor Medina <victor.medina at cibersys.com>
wrote:

> Im seeing this while starting the screen sharing...
>
> 2015-05-26 11:15:36.690561 [NOTICE] switch_channel.c:1089 New Channel
> verto.rtc/3500-screen [d4576cb3-e967-2bb1-be3c-a4831f8894fd]
> 2015-05-26 11:15:36.690561 [DEBUG] mod_verto.c:3407 Remote SDP
> verto.rtc/3500-screen:
> v=0
> o=- 4034581275591452868 2 IN IP4 127.0.0.1
> s=-
> t=0 0
> a=group:BUNDLE audio video
> a=msid-semantic: WMS fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
> m=audio 33952 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
> c=IN IP4 186.14.135.35
> a=rtcp:33426 IN IP4 186.14.135.35
> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 58898 typ host
> generation 0
> a=candidate:648569486 1 udp 2122129151 10.0.1.10 58899 typ host
> generation 0
> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 58900 typ host
> generation 0
> a=candidate:648569486 2 udp 2122129150 10.0.1.10 58901 typ host
> generation 0
> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 33952 typ srflx
> raddr 10.0.1.10 rport 58899 generation 0
> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 33426 typ srflx
> raddr 10.0.1.10 rport 58901 generation 0
> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
> active generation 0
> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
> active generation 0
> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
> active generation 0
> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
> active generation 0
> a=ice-ufrag:DXEOOI0UFrnM6fo8
> a=ice-pwd:l3x/4ga4HunTyP6rGwExMzgv
> a=fingerprint:sha-256
> BF:00:AC:5F:C8:84:54:5F:EF:34:3C:EC:AF:1C:92:80:B6:16:45:EF:3A:39:A1:3C:9E:AF:88:B0:E3:06:BA:AC
> a=setup:actpass
> a=mid:audio
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=recvonly
> a=rtcp-mux
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10; useinbandfec=1
> a=rtpmap:103 ISAC/16000
> a=rtpmap:104 ISAC/32000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:106 CN/32000
> a=rtpmap:105 CN/16000
> a=rtpmap:13 CN/8000
> a=rtpmap:126 telephone-event/8000
> a=maxptime:60
> m=video 42360 RTP/SAVPF 100 116 117 96
> c=IN IP4 186.14.135.35
> a=rtcp:44680 IN IP4 186.14.135.35
> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 58902 typ host
> generation 0
> a=candidate:648569486 1 udp 2122129151 10.0.1.10 58903 typ host
> generation 0
> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 58904 typ host
> generation 0
> a=candidate:648569486 2 udp 2122129150 10.0.1.10 58905 typ host
> generation 0
> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 42360 typ srflx
> raddr 10.0.1.10 rport 58903 generation 0
> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 44680 typ srflx
> raddr 10.0.1.10 rport 58905 generation 0
> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
> active generation 0
> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
> active generation 0
> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
> active generation 0
> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
> active generation 0
> a=ice-ufrag:DXEOOI0UFrnM6fo8
> a=ice-pwd:l3x/4ga4HunTyP6rGwExMzgv
> a=fingerprint:sha-256
> BF:00:AC:5F:C8:84:54:5F:EF:34:3C:EC:AF:1C:92:80:B6:16:45:EF:3A:39:A1:3C:9E:AF:88:B0:E3:06:BA:AC
> a=setup:actpass
> a=mid:video
> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=extmap:4 urn:3gpp:video-orientation
> a=sendrecv
> a=rtcp-mux
> a=rtpmap:100 VP8/90000
> a=rtcp-fb:100 ccm fir
> a=rtcp-fb:100 nack
> a=rtcp-fb:100 nack pli
> a=rtcp-fb:100 goog-remb
> a=rtpmap:116 red/90000
> a=rtpmap:117 ulpfec/90000
> a=rtpmap:96 rtx/90000
> a=fmtp:96 apt=100
> a=ssrc-group:FID 2871497692 3867014262
> a=ssrc:2871497692 cname:e4V6+GZYAdJCwcTf
> a=ssrc:2871497692 msid:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
> ef8fba5e-3643-4dbd-a038-5023f4521ba4
> a=ssrc:2871497692 mslabel:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
> a=ssrc:2871497692 label:ef8fba5e-3643-4dbd-a038-5023f4521ba4
> a=ssrc:3867014262 cname:e4V6+GZYAdJCwcTf
> a=ssrc:3867014262 msid:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
> ef8fba5e-3643-4dbd-a038-5023f4521ba4
> a=ssrc:3867014262 mslabel:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
> a=ssrc:3867014262 label:ef8fba5e-3643-4dbd-a038-5023f4521ba4
>
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
> (verto.rtc/3500-screen) Running State Change CS_INIT
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:516
> (verto.rtc/3500-screen) State INIT
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:40
> verto.rtc/3500-screen Standard INIT
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:48
> (verto.rtc/3500-screen) State Change CS_INIT -> CS_ROUTING
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:516
> (verto.rtc/3500-screen) State INIT going to sleep
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
> (verto.rtc/3500-screen) Running State Change CS_ROUTING
> 2015-05-26 11:15:36.690561 [DEBUG] switch_channel.c:2234
> (verto.rtc/3500-screen) Callstate Change DOWN -> RINGING
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:532
> (verto.rtc/3500-screen) State ROUTING
> 2015-05-26 11:15:36.690561 [DEBUG] mod_rtc.c:89 verto.rtc/3500-screen RTC
> ROUTING
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:166
> verto.rtc/3500-screen Standard ROUTING
> 2015-05-26 11:15:36.690561 [INFO] mod_dialplan_xml.c:636 Processing Victor
> Medina (Screen) <1000 (screen)>->3500-screen in context default
> Dialplan: verto.rtc/3500-screen parsing [default->unloop] continue=false
> Dialplan: verto.rtc/3500-screen Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [unloop] ${sip_looped_call}()
> =~ /^true$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->tod_example]
> continue=true
> Dialplan: verto.rtc/3500-screen Date/Time Match (PASS) [tod_example]
> break=on-false
> Dialplan: verto.rtc/3500-screen Action set(open=true)
> Dialplan: verto.rtc/3500-screen parsing [default->holiday_example]
> continue=true
> Dialplan: verto.rtc/3500-screen Date/TimeMatch (FAIL) [holiday_example]
> break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->global-intercept]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [global-intercept]
> destination_number(3500-screen) =~ /^886$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->group-intercept]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [group-intercept]
> destination_number(3500-screen) =~ /^\*8$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->intercept-ext]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [intercept-ext]
> destination_number(3500-screen) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->redial] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [redial]
> destination_number(3500-screen) =~ /^(redial|870)$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->global] continue=true
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [global] ${call_debug}(false)
> =~ /^true$/ break=never
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [global]
> ${default_password}(p6rSp6rSqwerty) =~ /^1234$/ break=never
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [global] ${rtp_has_crypto}()
> =~
> /^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/
> break=never
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [global]
> ${endpoint_disposition}() =~ /^(DELAYED NEGOTIATION)/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->snom-demo-2]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [snom-demo-2]
> destination_number(3500-screen) =~ /^9001$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->snom-demo-1]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [snom-demo-1]
> destination_number(3500-screen) =~ /^9000$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->eavesdrop] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [eavesdrop]
> destination_number(3500-screen) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->eavesdrop] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [eavesdrop]
> destination_number(3500-screen) =~ /^779$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->call_return]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [call_return]
> destination_number(3500-screen) =~ /^\*69$|^869$|^lcr$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->del-group] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [del-group]
> destination_number(3500-screen) =~ /^80(\d{2})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->add-group] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [add-group]
> destination_number(3500-screen) =~ /^81(\d{2})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->call-group-simo]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [call-group-simo]
> destination_number(3500-screen) =~ /^82(\d{2})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->call-group-order]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [call-group-order]
> destination_number(3500-screen) =~ /^83(\d{2})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->extension-intercom]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [extension-intercom]
> destination_number(3500-screen) =~ /^8(10[01][0-9])$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->Local_Extension]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [Local_Extension]
> destination_number(3500-screen) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->Local_Extension_Skinny]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [Local_Extension_Skinny]
> destination_number(3500-screen) =~ /^(11[01][0-9])$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->group_dial_sales]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [group_dial_sales]
> destination_number(3500-screen) =~ /^2000$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->group_dial_support]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [group_dial_support]
> destination_number(3500-screen) =~ /^2001$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->group_dial_billing]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [group_dial_billing]
> destination_number(3500-screen) =~ /^2002$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->operator] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [operator]
> destination_number(3500-screen) =~ /^(operator|0)$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->vmain] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [vmain]
> destination_number(3500-screen) =~ /^vmain$|^4000$|^\*98$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->sip_uri] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [sip_uri]
> destination_number(3500-screen) =~ /^sip:(.*)$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->nb_conferences]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [nb_conferences]
> destination_number(3500-screen) =~ /^(30\d{2})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->wb_conferences]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [wb_conferences]
> destination_number(3500-screen) =~ /^(31\d{2})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->uwb_conferences]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [uwb_conferences]
> destination_number(3500-screen) =~ /^(32\d{2})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->cdquality_conferences]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [cdquality_conferences]
> destination_number(3500-screen) =~ /^(33\d{2})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->cdquality_conferences]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [cdquality_conferences]
> destination_number(3500-screen) =~ /^(35\d{2})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing
> [default->freeswitch_public_conf_via_sip] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL)
> [freeswitch_public_conf_via_sip] destination_number(3500-screen) =~
> /^9(888|8888|1616|3232)$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->mad_boss_intercom]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [mad_boss_intercom]
> destination_number(3500-screen) =~ /^0911$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->mad_boss_intercom]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [mad_boss_intercom]
> destination_number(3500-screen) =~ /^0912$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->mad_boss] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [mad_boss]
> destination_number(3500-screen) =~ /^0913$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->ivr_demo] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ivr_demo]
> destination_number(3500-screen) =~ /^5000$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->dynamic_conference]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [dynamic_conference]
> destination_number(3500-screen) =~ /^5001$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->rtp_multicast_page]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [rtp_multicast_page]
> destination_number(3500-screen) =~ /^pagegroup$|^7243$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->park] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [park]
> destination_number(3500-screen) =~ /^5900$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->unpark] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [unpark]
> destination_number(3500-screen) =~ /^5901$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->valet_park]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [valet_park]
> destination_number(3500-screen) =~ /^(6000)$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->valet_park]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [valet_park]
> destination_number(3500-screen) =~ /^((?!6000)60\d{2})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->park] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [park] source(mod_verto) =~
> /mod_sofia/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->unpark] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [unpark] source(mod_verto) =~
> /mod_sofia/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->park] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [park] source(mod_verto) =~
> /mod_sofia/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->unpark] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [unpark] source(mod_verto) =~
> /mod_sofia/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->wait] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [wait]
> destination_number(3500-screen) =~ /^wait$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->fax_receive]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [fax_receive]
> destination_number(3500-screen) =~ /^9178$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->fax_transmit]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [fax_transmit]
> destination_number(3500-screen) =~ /^9179$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->ringback_180]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_180]
> destination_number(3500-screen) =~ /^9180$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->ringback_183_uk_ring]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_183_uk_ring]
> destination_number(3500-screen) =~ /^9181$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->ringback_183_music_ring]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_183_music_ring]
> destination_number(3500-screen) =~ /^9182$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing
> [default->ringback_post_answer_uk_ring] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL)
> [ringback_post_answer_uk_ring] destination_number(3500-screen) =~ /^9183$/
> break=on-false
> Dialplan: verto.rtc/3500-screen parsing
> [default->ringback_post_answer_music] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_post_answer_music]
> destination_number(3500-screen) =~ /^9184$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->ClueCon] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ClueCon]
> destination_number(3500-screen) =~ /^9191$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->show_info] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [show_info]
> destination_number(3500-screen) =~ /^9192$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->video_record]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [video_record]
> destination_number(3500-screen) =~ /^9193$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->video_playback]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [video_playback]
> destination_number(3500-screen) =~ /^9194$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->delay_echo]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [delay_echo]
> destination_number(3500-screen) =~ /^9195$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->echo] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [echo]
> destination_number(3500-screen) =~ /^9196$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->milliwatt] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [milliwatt]
> destination_number(3500-screen) =~ /^9197$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->tone_stream]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [tone_stream]
> destination_number(3500-screen) =~ /^9198$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->zrtp_enrollement]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [zrtp_enrollement]
> destination_number(3500-screen) =~ /^9787$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->hold_music]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [hold_music]
> destination_number(3500-screen) =~ /^9664$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->laugh break]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [laugh break]
> destination_number(3500-screen) =~ /^9386$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->h] continue=true
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [h]
> destination_number(3500-screen) =~ /^h264_(.*)$/ break=never
> Dialplan: verto.rtc/3500-screen parsing [default->v] continue=true
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [v]
> destination_number(3500-screen) =~ /^vp8_(.*)$/ break=never
> Dialplan: verto.rtc/3500-screen parsing [default->h] continue=true
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [h]
> destination_number(3500-screen) =~ /^hbr_(.*)$/ break=never
> Dialplan: verto.rtc/3500-screen parsing [default->v] continue=true
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [v]
> destination_number(3500-screen) =~ /^vbr_(.*)$/ break=never
> Dialplan: verto.rtc/3500-screen parsing [default->bug] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [bug]
> destination_number(3500-screen) =~ /^vbr_(.*)$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->bug] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [bug]
> destination_number(3500-screen) =~ /^vid_(.*)$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->bug] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [bug]
> destination_number(3500-screen) =~ /^decode$|^9952$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->101] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [101]
> destination_number(3500-screen) =~ /^101$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->pizza_demo]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [pizza_demo]
> destination_number(3500-screen) =~ /^(pizza|74992)$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->Talking Clock Time]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [Talking Clock Time]
> destination_number(3500-screen) =~ /^9170$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->Talking Clock Date]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [Talking Clock Date]
> destination_number(3500-screen) =~ /^9171$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->Talking Clock Date and
> Time] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [Talking Clock Date and Time]
> destination_number(3500-screen) =~ /^9172$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->local.example.com]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (PASS) [local.example.com]
> ${toll_allow}(domestic,international,local) =~ /local/ break=on-false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [local.example.com]
> destination_number(3500-screen) =~ /^(\d{7})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->domestic.example.com]
> continue=false
> Dialplan: verto.rtc/3500-screen Regex (PASS) [domestic.example.com]
> ${toll_allow}(domestic,international,local) =~ /domestic/ break=on-false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [domestic.example.com]
> destination_number(3500-screen) =~ /^(\d{11})$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->
> international.example.com] continue=false
> Dialplan: verto.rtc/3500-screen Regex (PASS) [international.example.com]
> ${toll_allow}(domestic,international,local) =~ /international/
> break=on-false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [international.example.com]
> destination_number(3500-screen) =~ /^(011\d+)$/ break=on-false
> Dialplan: verto.rtc/3500-screen parsing [default->enum] continue=false
> Dialplan: verto.rtc/3500-screen Regex (FAIL) [enum]
> ${module_exists(mod_enum)}(false) =~ /true/ break=on-false
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:216
> (verto.rtc/3500-screen) State Change CS_ROUTING -> CS_EXECUTE
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:532
> (verto.rtc/3500-screen) State ROUTING going to sleep
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
> (verto.rtc/3500-screen) Running State Change CS_EXECUTE
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:539
> (verto.rtc/3500-screen) State EXECUTE
> 2015-05-26 11:15:36.690561 [DEBUG] mod_rtc.c:120 verto.rtc/3500-screen RTC
> EXECUTE
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:258
> verto.rtc/3500-screen Standard EXECUTE
> EXECUTE verto.rtc/3500-screen set(open=true)
> 2015-05-26 11:15:36.690561 [DEBUG] mod_dptools.c:1469
> verto.rtc/3500-screen SET [open]=[true]
> 2015-05-26 11:15:36.690561 [NOTICE] switch_core_state_machine.c:315
> verto.rtc/3500-screen has executed the last dialplan instruction, hanging
> up.
> 2015-05-26 11:15:36.690561 [NOTICE] switch_core_state_machine.c:317 Hangup
> verto.rtc/3500-screen [CS_EXECUTE] [NORMAL_CLEARING]
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:539
> (verto.rtc/3500-screen) State EXECUTE going to sleep
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
> (verto.rtc/3500-screen) Running State Change CS_HANGUP
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:739
> (verto.rtc/3500-screen) Callstate Change RINGING -> HANGUP
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:741
> (verto.rtc/3500-screen) State HANGUP
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:60
> verto.rtc/3500-screen Standard HANGUP, cause: NORMAL_CLEARING
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:741
> (verto.rtc/3500-screen) State HANGUP going to sleep
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:508
> (verto.rtc/3500-screen) State Change CS_HANGUP -> CS_REPORTING
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
> (verto.rtc/3500-screen) Running State Change CS_REPORTING
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:827
> (verto.rtc/3500-screen) State REPORTING
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:104
> verto.rtc/3500-screen Standard REPORTING, cause: NORMAL_CLEARING
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:827
> (verto.rtc/3500-screen) State REPORTING going to sleep
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:499
> (verto.rtc/3500-screen) State Change CS_REPORTING -> CS_DESTROY
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_session.c:1638 Session 4
> (verto.rtc/3500-screen) Locked, Waiting on external entities
> 2015-05-26 11:15:36.690561 [NOTICE] switch_core_session.c:1656 Session 4
> (verto.rtc/3500-screen) Ended
> 2015-05-26 11:15:36.690561 [NOTICE] switch_core_session.c:1660 Close
> Channel verto.rtc/3500-screen [CS_DESTROY]
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:630
> (verto.rtc/3500-screen) Running State Change CS_DESTROY
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:640
> (verto.rtc/3500-screen) State DESTROY
> 2015-05-26 11:15:36.690561 [DEBUG] mod_rtc.c:132 verto.rtc/3500-screen RTC
> DESTROY
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:111
> verto.rtc/3500-screen Standard DESTROY
> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:640
> (verto.rtc/3500-screen) State DESTROY going to sleep
> freeswitch at internal>
>
>
> 2015-05-26 10:24 GMT-04:30 Michael Jerris <mike at jerris.com>:
>
> yes.  is the call hitting your dial plan?
>>
>>
>> On Tuesday, May 26, 2015, Victor Medina <victor.medina at cibersys.com>
>> wrote:
>>
>>> I did both!
>>> I installed the WebRTC Screen Capture extension and check all command
>>> line option on chrome: --allow-http-screen-capture &
>>> --auto-select-desktop-capture-source as of ...
>>> http://peter.sh/experiments/chromium-command-line-switches/
>>>
>>> I think only the extension is needed, right?
>>>
>>> 2015-05-25 18:25 GMT-04:30 Michael Jerris <mike at jerris.com>:
>>>
>>>> for chrome to allow screen share it requires you to install a chrome
>>>> plugin or start chrome with a command line argument to allow screen share
>>>>
>>>>
>>>> On Monday, May 25, 2015, Victor Medina <victor.medina at cibersys.com>
>>>> wrote:
>>>>
>>>>> ?Log..
>>>>>
>>>>> Attempting Screen Capture....
>>>>> SCREEN SHARE
>>>>> Audio constraints false
>>>>> Video constraints Object {mandatory: Object, optional:
>>>>> Array[0]}mandatory: Objectoptional: Array[0]__proto__: Object
>>>>> Stream Success
>>>>> stream started
>>>>> Offer SDP
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> offer RTCIceCandidate {}
>>>>> ICE Complete
>>>>> ICE SDP
>>>>> offer v=0
>>>>> o=- 3905402391138908374 2 IN IP4 127.0.0.1
>>>>> s=-
>>>>> t=0 0
>>>>> a=group:BUNDLE audio video
>>>>> a=msid-semantic: WMS dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>>> m=audio 34382 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>>>> c=IN IP4 186.14.135.35
>>>>> a=rtcp:41117 IN IP4 186.14.135.35
>>>>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54692 typ host
>>>>> generation 0
>>>>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54693 typ host
>>>>> generation 0
>>>>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54694 typ host
>>>>> generation 0
>>>>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54695 typ host
>>>>> generation 0
>>>>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host
>>>>> tcptype active generation 0
>>>>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>>>>> active generation 0
>>>>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host
>>>>> tcptype active generation 0
>>>>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>>>>> active generation 0
>>>>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 34382 typ srflx
>>>>> raddr 10.0.1.10 rport 54693 generation 0
>>>>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 41117 typ srflx
>>>>> raddr 10.0.1.10 rport 54695 generation 0
>>>>> a=ice-ufrag:qjfFF9BX74oS8gTh
>>>>> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
>>>>> a=fingerprint:sha-256
>>>>> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
>>>>> a=setup:actpass
>>>>> a=mid:audio
>>>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>>> a=recvonly
>>>>> a=rtcp-mux
>>>>> a=rtpmap:111 opus/48000/2
>>>>> a=fmtp:111 minptime=10; useinbandfec=1
>>>>> a=rtpmap:103 ISAC/16000
>>>>> a=rtpmap:104 ISAC/32000
>>>>> a=rtpmap:9 G722/8000
>>>>> a=rtpmap:0 PCMU/8000
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:106 CN/32000
>>>>> a=rtpmap:105 CN/16000
>>>>> a=rtpmap:13 CN/8000
>>>>> a=rtpmap:126 telephone-event/8000
>>>>> a=maxptime:60
>>>>> m=video 39129 RTP/SAVPF 100 116 117 96
>>>>> c=IN IP4 186.14.135.35
>>>>> a=rtcp:40989 IN IP4 186.14.135.35
>>>>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54696 typ host
>>>>> generation 0
>>>>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54697 typ host
>>>>> generation 0
>>>>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54698 typ host
>>>>> generation 0
>>>>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54699 typ host
>>>>> generation 0
>>>>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host
>>>>> tcptype active generation 0
>>>>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>>>>> active generation 0
>>>>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host
>>>>> tcptype active generation 0
>>>>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>>>>> active generation 0
>>>>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 40989 typ srflx
>>>>> raddr 10.0.1.10 rport 54699 generation 0
>>>>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 39129 typ srflx
>>>>> raddr 10.0.1.10 rport 54697 generation 0
>>>>> a=ice-ufrag:qjfFF9BX74oS8gTh
>>>>> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
>>>>> a=fingerprint:sha-256
>>>>> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
>>>>> a=setup:actpass
>>>>> a=mid:video
>>>>> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>>>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>>> a=extmap:4 urn:3gpp:video-orientation
>>>>> a=sendrecv
>>>>> a=rtcp-mux
>>>>> a=rtpmap:100 VP8/90000
>>>>> a=rtcp-fb:100 ccm fir
>>>>> a=rtcp-fb:100 nack
>>>>> a=rtcp-fb:100 nack pli
>>>>> a=rtcp-fb:100 goog-remb
>>>>> a=rtpmap:116 red/90000
>>>>> a=rtpmap:117 ulpfec/90000
>>>>> a=rtpmap:96 rtx/90000
>>>>> a=fmtp:96 apt=100
>>>>> a=ssrc-group:FID 3962025390 1186212002
>>>>> a=ssrc:3962025390 cname:Rvg58eAlJAmWf/sm
>>>>> a=ssrc:3962025390 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>>> a45b2502-eb86-452c-b4cb-e69e5305938a
>>>>> a=ssrc:3962025390 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>>> a=ssrc:3962025390 label:a45b2502-eb86-452c-b4cb-e69e5305938a
>>>>> a=ssrc:1186212002 cname:Rvg58eAlJAmWf/sm
>>>>> a=ssrc:1186212002 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>>> a45b2502-eb86-452c-b4cb-e69e5305938a
>>>>> a=ssrc:1186212002 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>>> a=ssrc:1186212002 label:a45b2502-eb86-452c-b4cb-e69e5305938a
>>>>>
>>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from new to
>>>>> requesting
>>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from
>>>>> requesting to trying
>>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from trying
>>>>> to hangup
>>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from hangup
>>>>> to destroy
>>>>> ?
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>> V?ctor E. Medina M.
>>> Platform Architect / Chief Infrastructure
>>> +58424 291 4561
>>> BB #79A8AFA2
>>> @VMCibersys
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mike at jerris.com  Tue May 26 20:07:09 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 26 May 2015 12:07:09 -0400
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <1946256680.20150526171049@seznam.cz>
References: <43162995.20150526094002@seznam.cz>
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
	<16210177989.20150526120710@seznam.cz>
	<CAHtxdDd=vo6uwo-7imdo7MEGuRDC2PYbqk3HyLvvD1rdfkZxRw@mail.gmail.com>
	<1946256680.20150526171049@seznam.cz>
Message-ID: <01CF76A0-3760-446E-9A5F-EC5025163004@jerris.com>

you should NOT be doing that.  That will 100% for sure crete a broken system,  Any other libs that link to freeswitch will need to be linked against the updated vpx lib.


> On May 26, 2015, at 11:10 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> 
> Dobr? den,
> 
> ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1
> ln: failed to create symbolic link ?/usr/lib64/libvpx.so.1?: No such file or directory
> 
> 
> I have libvpx.so in /usr/x_86_64_linux_gnu. But it steal not work. 
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel. 775-415-382
> 
> ?ter? 26. kv?tna 2015, 14:57:11, napsal jste:
> 
> 
> execute as root "ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1"
> And try again
> 
> On Tue, May 26, 2015 at 1:07 PM, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
> Dobr? den,
> 
> Where can I get the -extra package for libav?
> 
> I try from here https://freeswitch.org/stash/projects/SD/repos/libav/browse <https://freeswitch.org/stash/projects/SD/repos/libav/browse>  It is compiled without problems. 
> 
> But I still have a error
> 
> 2015-05-26 06:01:38.083643 [CRIT] switch_loadable_module.c:1520 Error Loading module /usr/local/freeswitch/mod/mod_av.so
> **libvpx.so.1: cannot open shared object file: No such file or directory**
> 
> 
> 
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel <http://mob.tel/>. 775-415-382
> 
> ?ter? 26. kv?tna 2015, 11:22:35, napsal jste:
> 
> 
> try using mod av instead.  You'll need to install the -extra package for libav 
> 
> On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
> Hi all !
> 
> I have one problem with the MP4 file recording with sound. The video is great. But there is no sound.
> 
> Adding a log. files in place, but for some reason does not work. I use Debian and the latest version 1.6
> 
> EXECUTE sofia/internal/1004 at 192.168.242.132 record(/usr/local/freeswitch/recordings/testrecord.mp4)
> [00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}  write video
> 
> 
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel <http://mob.tel/>. 775-415-382
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue May 26 20:07:53 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 26 May 2015 12:07:53 -0400
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMO+YUezb5peCD0njovq83YwNYQMEx6ASee96RRq2-0Ocmg@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAFiqYum4if-B8NGFu7PXjFpJZnPuFGy8XhWscM7p7q6BwZiqGg@mail.gmail.com>
	<CAC8JMO+-Wqu9_xQx6TzkUU3NLhVU=HTpDu6wO2p1TX8J1kxEKg@mail.gmail.com>
	<CAC8JMO+YUezb5peCD0njovq83YwNYQMEx6ASee96RRq2-0Ocmg@mail.gmail.com>
Message-ID: <DC7CE9BD-1271-4EBB-A7D1-4E3D0CCEBB46@jerris.com>

Please file a jira for us on that.  Even better with a pull request to fix it.

> On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> Just FYI, the JsSip engineers replied to my questions with this:
> 
> The error happens because the incoming reINVITE has a=setup:active in
> the SDP which is a bug in FreeSwitch (the RFC clearly states that the
> SDP offer MUST have a=setup:actpass):
> 
> On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar at gmail.com <mailto:olegstolyar at gmail.com>> wrote:
> Thanks Steven!
> 
> It may be https://freeswitch.org/jira/browse/FS-7040 <https://freeswitch.org/jira/browse/FS-7040>.
> 
> As far as the 120 sec is concerned - Below are snippets from both the invite and the 200 OK from FS.  I know that FS reads the Session-Expires from the client because if I change it to a value less than 120, FS sends back a "SIP/2.0 422 Session Interval Too Small".  I  even thought the problem could be that the Session-Exipres format in the original INVITE is incorrect since it does not contain ";refresher=uac" but when I added that and made the line "Session-Expires: 300;refresher=uac" nothing changed.
> 
> 
> INVITE sip:echo-test at anonymous.invalid SIP/2.0
> Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029
> Max-Forwards: 69
> To: <sip:echo-test at anonymous.invalid>
> From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
> Call-ID: r65e48gp171p21rkppcu
> CSeq: 6621 INVITE
> Contact: <sip:mqthb9dj at pfcnm9rjv5en.invalid;transport=ws;ob>
> Content-Type: application/sdp
> Session-Expires: 300
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
> Supported: timer,ice,outbound
> User-Agent: JsSIP 0.6.26
> Content-Length: 2754
> 
> SIP/2.0 200 OK
> Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200
> From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
> To: <sip:echo-test at anonymous.invalid>;tag=SgjX0X4arHUFg
> Call-ID: r65e48gp171p21rkppcu
> CSeq: 6621 INVITE
> Contact: <sip:echo-test at 50.19.255.62:5080;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
> Accept: application/sdp
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Require: timer
> Supported: timer, path, replaces
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
> Session-Expires: 120;refresher=uac
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 809
> X-Netflix: yes
> Remote-Party-ID: "echo-test" <sip:echo-test at anonymous.invalid>;party=calling;privacy=off;screen=no
> 
> 
> On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
> The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes.
> Session-Expires: 120;refresher=uac
> Min-SE: 120
> 
> Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040 <https://freeswitch.org/jira/browse/FS-7040>
> 
> On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar at gmail.com <mailto:olegstolyar at gmail.com>> wrote:
> But isn't that based on the session-timeout param which defaults to 30 min?  My re-invites occur much sooner than 30 min into a call.  Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?
> 
> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> session timer
> 
> 
> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com <mailto:olegstolyar at gmail.com>> wrote:
> Thanks Michael, I'll see if we can do that!
> 
> So, is the re-INVITE legit and the problem is that JsSip does not respond to it?  Still, I am curious what is triggering the re-INVITE.
> 
> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com <>> wrote:
> I think the sip.js guys fixed this issue when they forked jssip.  I'd suggest using that.
> 
> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com <>> wrote:
> >
> > Hi guys,
> >
> > Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg.  Below is the relevant log.  The user experience is that they keep talking and hearing each other up to the very end.  I have a recording of that call, so can confirm.
> >
> > The call uses WebRTC and is originated by JsSip from Chrome.  Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
> >
> > I suspect it's a re-INVITE but what triggers FS to send it?  I couldn't find anything in the logs that could shed light.
> >
> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
> >    ------------------------------------------------------------------------
> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
> >    Max-Forwards: 70
> >    From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid>;tag=KQecUSr12rSQp
> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
> >    Call-ID: g8980rbrbk2t45oj5mru
> >    CSeq: 75703945 INVITE
> >    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp>
> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >    Supported: timer, path, replaces
> >    Session-Expires: 120;refresher=uac
> >    Min-SE: 120
> >    Content-Type: application/sdp
> >    Content-Disposition: session
> >    Content-Length: 825
> >
> >    v=0
> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
> >    s=FreeSWITCH
> >    c=IN IP4 ##.##.##.##
> >    t=0 0
> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
> >    m=audio 22670 RTP/SAVPF 9 126 106
> >    a=rtpmap:9 G722/8000
> >    a=rtpmap:126 telephone-event/8000
> >    a=rtpmap:106 CN/8000
> >    a=ptime:20
> >    a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
> >    a=rtcp-mux
> >    a=rtcp:22670 IN IP4 ##.##.##.##
> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
> >    ------------------------------------------------------------------------
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid entering state [calling][0]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid entering state [terminating][503]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous at anonymous.invalid [KILL]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid skip receive message [UNBRIDGE] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous at anonymous.invalid skip receive message [REFER_EVENT] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid Restore previous codec G722:9.
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous at anonymous.invalid skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to sleep
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous at anonymous.invalid) Running State Change CS_HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE -> HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid hanging up, cause: NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to sleep
> >
> 
> 
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From mike at jerris.com  Tue May 26 20:08:34 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 26 May 2015 12:08:34 -0400
Subject: [Freeswitch-users] Video always muted in conference (fs_video2)
In-Reply-To: <CACXsapNJ_ErsOdZzc=_f4DJpzYe6NRGeNB8uRE1Uw1BSAaGOJg@mail.gmail.com>
References: <CAFKjMAYjKz32rE-G_aA639J5b27eNwnRAME31La25W9KiWbagA@mail.gmail.com>
	<7B131C27-7233-4E88-BB9E-E01BE588DE82@jerris.com>
	<CAFKjMAZ30sxNtAXU1iqksuNwM53gzQaqFQor7vhVHhg-3VBn5w@mail.gmail.com>
	<CAEwTsAgsVXvE5Ym9TWvbSvd9i7gCgNWwmp-2adxYKug0RHHKBA@mail.gmail.com>
	<CACXsapNJ_ErsOdZzc=_f4DJpzYe6NRGeNB8uRE1Uw1BSAaGOJg@mail.gmail.com>
Message-ID: <C6A42EE2-E59D-4F1A-8AB6-B3C61962D41A@jerris.com>

Please file a jira with details on how to reproduce and debug logs.

> On May 26, 2015, at 11:51 AM, Victor Medina <victor.medina at cibersys.com> wrote:
> 
> Michael...
> 
> I just build FS with the latest sources and the Video Unmuted/Video Muted ramains. 
> Just to confirm this.
> 
> If you guys need me to test anything I have a full blown server just for me and fs tests.
> 
> 2015-05-25 0:08 GMT-04:30 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> Some fixes were pushed for this on Friday.  Try out latest code and see if that works any better for you?
> 
> 
> On Saturday, May 23, 2015, Carlos Gonz?lez Florido <carlosj.gf at gmail.com <mailto:carlosj.gf at gmail.com>> wrote:
> Ok, tell me if I can help with the testing.
> Is there a way to disconnect ipv6?
> 
> On Fri, May 22, 2015 at 6:04 PM, Michael Jerris <mike at jerris.com <>> wrote:
> I don't think its actually muted.  We are chasing down an issue that sounds just like this.  It might be related to ipv6 but we are still looking in to it.
> 
> > On May 22, 2015, at 8:40 AM, Carlos Gonz?lez Florido <carlosj.gf at gmail.com <>> wrote:
> >
> > Hello,
> >
> > I'm testing the impressive fs_video2 branch, but I have the following problems:
> >
> > - some participants have their video always muted (and their camera is on and working for Hangouts, for example).
> > - other participants start the same (video muted), but after 3-5 minutes the video is automatically unmuted.
> > - if we do a layout change (using fs_cli), all participants (having video muted or unmuted at that moment) go to muted video for the rest of the conference.
> >
> > What is the reason for fs to automatically mute the video? Is this expected or does it look like a bug? Can this behaviour be tuned or disconnected?
> >
> > Thank you,
> > Carlos Gonzalez
> 
> 
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> 
> 
> -- 
> 
> 
> 
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561 <>
> BB #79A8AFA2
> @VMCibersys
> 
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue May 26 20:13:09 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 26 May 2015 11:13:09 -0500
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <01CF76A0-3760-446E-9A5F-EC5025163004@jerris.com>
References: <43162995.20150526094002@seznam.cz>
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
	<16210177989.20150526120710@seznam.cz>
	<CAHtxdDd=vo6uwo-7imdo7MEGuRDC2PYbqk3HyLvvD1rdfkZxRw@mail.gmail.com>
	<1946256680.20150526171049@seznam.cz>
	<01CF76A0-3760-446E-9A5F-EC5025163004@jerris.com>
Message-ID: <CAKbxfG-5kVqXWSqGj1T_uVrwSK2myWdZKV-sF8qygu3OgUTq+w@mail.gmail.com>

Really this stuff has not been released yet so we cannot spend much time
support it it yet.  Especially on platforms that do not package any of the
depends.
If you want to play with it early and have any hope for any assistance use
debian jessie and our repos.  Otherwise you will have to wait quite a long
time.



On Tue, May 26, 2015 at 11:07 AM, Michael Jerris <mike at jerris.com> wrote:

> you should NOT be doing that.  That will 100% for sure crete a broken
> system,  Any other libs that link to freeswitch will need to be linked
> against the updated vpx lib.
>
>
> On May 26, 2015, at 11:10 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>  Dobr? den,
>
> ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1
> ln: failed to create symbolic link ?/usr/lib64/libvpx.so.1?: No such file
> or directory
>
>
> I have libvpx.so in /usr/x_86_64_linux_gnu. But it steal not work.
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 ?ter? 26. kv?tna 2015, 14:57:11, napsal jste:
> *
>  execute as root "ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1"
> And try again
>
> On Tue, May 26, 2015 at 1:07 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Dobr? den,
>
> Where can I get the -extra package for libav?
>
> I try from here
> https://freeswitch.org/stash/projects/SD/repos/libav/browse  It is
> compiled without problems.
>
> But I still have a error
>
> 2015-05-26 06:01:38.083643 [CRIT] switch_loadable_module.c:1520 Error
> Loading module /usr/local/freeswitch/mod/mod_av.so
> **libvpx.so.1: cannot open shared object file: No such file or directory**
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
>
> *. 775-415-382 ?ter? 26. kv?tna 2015, 11:22:35, napsal jste: *
>  try using mod av instead.  You'll need to install the -extra package for
> libav
>
> On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi all !
>
> I have one problem with the MP4 file recording with sound. The video is
> great. But there is no sound.
>
> Adding a log. files in place, but for some reason does not work. I use
> Debian and the latest version 1.6
>
> EXECUTE sofia/internal/1004 at 192.168.242.132
> record(/usr/local/freeswitch/recordings/testrecord.mp4)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
> open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open
> #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}
>  write video
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel*.
> 775-415-382*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 s.safarov at gmail.com  Tue May 26 20:23:29 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 26 May 2015 19:23:29 +0300
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <17898630.20150512135025@seznam.cz>
References: <17898630.20150512135025@seznam.cz>
Message-ID: <CAHtxdDe09+FFaiYBNyPTcoaJAxX30cLKn6vn1m=yiN39esrwZg@mail.gmail.com>

First iteration for FS rpms is made.
You try can install FS 1.6 from rpm ecxecuting command below
1) wget -P /etc/yum.repos.d http://fs-repo.network-engineer.ru/fs.repo
2) yum -y install freeswitch-config-vanilla freeswitch-format-vlc
3) /usr/bin/freeswitch -nc
4) fs_cli

TODO:
1) Declare rpm dependences for new FS modules
2) mock FS source rpm
3) fix header file in SD/libilbc
4) minimize vlc dependency
5) compile FS with libvpx 2.0.0


On Tue, May 12, 2015 at 2:50 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Hi all,
>
> Is there a chance to put on CentOS the new version 1.6?
>
> I've tried. But I have a problem on ./configure -C
> checking for libyuv >= 0.0.1280... configure: error: You need to install
> libyuv-dev. Required library
>
> Where can I find this library? I tried to put it from here
> https://code.google.com/p/libyuv/ But FreeSwitch does not see it.
>
> Is there a possibility to find this library to CentOS?
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 26 20:27:23 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 26 May 2015 19:27:23 +0300
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <1946256680.20150526171049@seznam.cz>
References: <43162995.20150526094002@seznam.cz>
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
	<16210177989.20150526120710@seznam.cz>
	<CAHtxdDd=vo6uwo-7imdo7MEGuRDC2PYbqk3HyLvvD1rdfkZxRw@mail.gmail.com>
	<1946256680.20150526171049@seznam.cz>
Message-ID: <CAHtxdDe=AAK3vkozyB-p2ywFQ=F66rxXvD66QV9B_XCx6Vw71w@mail.gmail.com>

Denis, FS 1.6 compiled packages has been placed in repo
Recreate server and install FS from new repo. And I will help solve this
issue.
More info at
http://lists.freeswitch.org/pipermail/freeswitch-users/2015-May/113351.html

On Tue, May 26, 2015 at 6:10 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:

>  Dobr? den,
>
> ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1
> ln: failed to create symbolic link ?/usr/lib64/libvpx.so.1?: No such file
> or directory
>
>
> I have libvpx.so in /usr/x_86_64_linux_gnu. But it steal not work.
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 ?ter? 26. kv?tna 2015, 14:57:11, napsal jste:
> *
>
>  execute as root "ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1"
> And try again
>
> On Tue, May 26, 2015 at 1:07 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Dobr? den,
>
> Where can I get the -extra package for libav?
>
> I try from here
> https://freeswitch.org/stash/projects/SD/repos/libav/browse  It is
> compiled without problems.
>
> But I still have a error
>
> 2015-05-26 06:01:38.083643 [CRIT] switch_loadable_module.c:1520 Error
> Loading module /usr/local/freeswitch/mod/mod_av.so
> **libvpx.so.1: cannot open shared object file: No such file or directory**
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
>
> *. 775-415-382 ?ter? 26. kv?tna 2015, 11:22:35, napsal jste: *
>  try using mod av instead.  You'll need to install the -extra package for
> libav
>
> On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi all !
>
> I have one problem with the MP4 file recording with sound. The video is
> great. But there is no sound.
>
> Adding a log. files in place, but for some reason does not work. I use
> Debian and the latest version 1.6
>
> EXECUTE sofia/internal/1004 at 192.168.242.132
> record(/usr/local/freeswitch/recordings/testrecord.mp4)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
> open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open
> #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}
>  write video
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel*.
> 775-415-382*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Tue May 26 20:33:27 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 26 May 2015 12:33:27 -0400
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <CAHtxdDe09+FFaiYBNyPTcoaJAxX30cLKn6vn1m=yiN39esrwZg@mail.gmail.com>
References: <17898630.20150512135025@seznam.cz>
	<CAHtxdDe09+FFaiYBNyPTcoaJAxX30cLKn6vn1m=yiN39esrwZg@mail.gmail.com>
Message-ID: <12BAB240-50D1-4779-AC55-8B0B1D91C4B0@jerris.com>

Please do NOT encourage people to use outside repos.  Please work with us to create our own repos.  To all, use any repos not on the freeswitch domain at your own risk.

> On May 26, 2015, at 12:23 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> First iteration for FS rpms is made.
> You try can install FS 1.6 from rpm ecxecuting command below
> 1) wget -P /etc/yum.repos.d http://fs-repo.network-engineer.ru/fs.repo <http://fs-repo.network-engineer.ru/fs.repo>
> 2) yum -y install freeswitch-config-vanilla freeswitch-format-vlc
> 3) /usr/bin/freeswitch -nc
> 4) fs_cli
> 
> TODO: 
> 1) Declare rpm dependences for new FS modules 
> 2) mock FS source rpm 
> 3) fix header file in SD/libilbc 
> 4) minimize vlc dependency
> 5) compile FS with libvpx 2.0.0 
> 
> 
> On Tue, May 12, 2015 at 2:50 PM, Denis Jakovlev <yadenis at seznam.cz <mailto:yadenis at seznam.cz>> wrote:
> Hi all,
> 
> Is there a chance to put on CentOS the new version 1.6?
> 
> I've tried. But I have a problem on ./configure -C 
> checking for libyuv >= 0.0.1280... configure: error: You need to install libyuv-dev. Required library
> 
> Where can I find this library? I tried to put it from here https://code.google.com/p/libyuv/ <https://code.google.com/p/libyuv/> But FreeSwitch does not see it.
> 
> Is there a possibility to find this library to CentOS?
> 
> -- 
> S pozdravem,
> Ing.Denis Jakovlev                           
> mob.tel <http://mob.tel/>. 775-415-382
> 

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From s.safarov at gmail.com  Tue May 26 20:34:44 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 26 May 2015 19:34:44 +0300
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <01CF76A0-3760-446E-9A5F-EC5025163004@jerris.com>
References: <43162995.20150526094002@seznam.cz>
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
	<16210177989.20150526120710@seznam.cz>
	<CAHtxdDd=vo6uwo-7imdo7MEGuRDC2PYbqk3HyLvvD1rdfkZxRw@mail.gmail.com>
	<1946256680.20150526171049@seznam.cz>
	<01CF76A0-3760-446E-9A5F-EC5025163004@jerris.com>
Message-ID: <CAHtxdDfiiW0yE+QdZXZ=YZc4p355mdTJ-BF8TpKFBCB8uL6nQQ@mail.gmail.com>

For Michael
Denis has installed server from my repo (FS-7553
<https://freeswitch.org/jira/browse/FS-7553>) and libvpx 1.4.0 has not
correct libnames (more info below). Prevision recommendation taken to
workaround it.

[root at localhost ~]# rpm -qa | grep libvpx
libvpx-devel-1.4.0-6.el7.centos.x86_64
libvpx-1.4.0-6.el7.centos.x86_64

[root at localhost ~]# ls -l /usr/lib64/libvpx.so*
lrwxrwxrwx. 1 root root      15 May 16 01:48 /usr/lib64/libvpx.so ->
libvpx.so.2.0.0
lrwxrwxrwx. 1 root root      15 May 16 01:48 /usr/lib64/libvpx.so.2 ->
libvpx.so.2.0.0
lrwxrwxrwx. 1 root root      15 May 16 01:48 /usr/lib64/libvpx.so.2.0 ->
libvpx.so.2.0.0
-rwxr-xr-x. 1 root root 1548944 May 16 01:43 /usr/lib64/libvpx.so.2.0.0



On Tue, May 26, 2015 at 7:07 PM, Michael Jerris <mike at jerris.com> wrote:

> you should NOT be doing that.  That will 100% for sure crete a broken
> system,  Any other libs that link to freeswitch will need to be linked
> against the updated vpx lib.
>
>
> On May 26, 2015, at 11:10 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>  Dobr? den,
>
> ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1
> ln: failed to create symbolic link ?/usr/lib64/libvpx.so.1?: No such file
> or directory
>
>
> I have libvpx.so in /usr/x_86_64_linux_gnu. But it steal not work.
>
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
> <http://mob.tel>. 775-415-382 ?ter? 26. kv?tna 2015, 14:57:11, napsal jste:
> *
>  execute as root "ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1"
> And try again
>
> On Tue, May 26, 2015 at 1:07 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Dobr? den,
>
> Where can I get the -extra package for libav?
>
> I try from here
> https://freeswitch.org/stash/projects/SD/repos/libav/browse  It is
> compiled without problems.
>
> But I still have a error
>
> 2015-05-26 06:01:38.083643 [CRIT] switch_loadable_module.c:1520 Error
> Loading module /usr/local/freeswitch/mod/mod_av.so
> **libvpx.so.1: cannot open shared object file: No such file or directory**
>
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>
>
>
> *. 775-415-382 ?ter? 26. kv?tna 2015, 11:22:35, napsal jste: *
>  try using mod av instead.  You'll need to install the -extra package for
> libav
>
> On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
> Hi all !
>
> I have one problem with the MP4 file recording with sound. The video is
> great. But there is no sound.
>
> Adding a log. files in place, but for some reason does not work. I use
> Debian and the latest version 1.6
>
> EXECUTE sofia/internal/1004 at 192.168.242.132
> record(/usr/local/freeswitch/recordings/testrecord.mp4)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
> open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so'
> (libvpx.so.1: cannot open shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> [00007f16f00c7fc8] core libvlc warning: cannot load module
> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
> shared object file: No such file or directory)
> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open
> #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}
>  write video
>
>
>
>
>
>
> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel*.
> 775-415-382*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aqsyounas at gmail.com  Tue May 26 20:35:35 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 26 May 2015 09:35:35 -0700
Subject: [Freeswitch-users] user registration with sip.js
In-Reply-To: <CACXsapMnCT9wTQQiGgYurOd=UQsS81iW25UxZAMs1yTH_8emKQ@mail.gmail.com>
References: <CAGVOjVVL80T=V5qsT=802TyuvkkJhWbuXdJS3_rTwBVkuq+xaA@mail.gmail.com>
	<CACXsapMnCT9wTQQiGgYurOd=UQsS81iW25UxZAMs1yTH_8emKQ@mail.gmail.com>
Message-ID: <CAGVOjVVDJd6FzX1OVrZ2w78CUv=af73QEfut91Bod3Wkp+scXQ@mail.gmail.com>

Thanks for your reply. Yes, i have enable the webrtc in my internal.xml

tcp        0      0 192.168.1.30:5066       0.0.0.0:*
LISTEN      11078/freeswitch
tcp        0      0 192.168.1.30:7443       0.0.0.0:*
LISTEN      11078/freeswitch
tcp        0      0 127.0.0.1:8021          0.0.0.0:*
LISTEN      11078/freeswitch
tcp        0      0 192.168.1.30:5080       0.0.0.0:*
LISTEN      11078/freeswitch
tcp        0      0 192.168.1.30:5060       0.0.0.0:*
LISTEN      11078/freeswitch
tcp6       0      0 ::1:5080                :::*
LISTEN      11078/freeswitch
tcp6       0      0 ::1:5060                :::*
LISTEN      11078/freeswitch
udp        0      0 192.168.1.30:5080       0.0.0.0:*
11078/freeswitch
udp        0      0 239.255.255.250:1900    0.0.0.0:*
11078/freeswitch
udp        0      0 192.168.1.30:5060       0.0.0.0:*
11078/freeswitch
udp6       0      0 ::1:5080
:::*                                11078/freeswitch
udp6       0      0 ::1:5060
:::*                                11078/freeswitch

May be I am missing some packages, related to js.

Thanks


On 26 May 2015 at 08:49, Victor Medina <victor.medina at cibersys.com> wrote:

> Did you anable the webrtc endpoint in your profiles?
>
> 2015-05-26 11:04 GMT-04:30 Aqs Younas <aqsyounas at gmail.com>:
>
>> Hi, Users.
>> I am trying to register default freeswitch user "1003" through webrtc
>> using sip.js.
>> This is my "*index.html*"
>>
>> <html>
>>   <head>
>>   </head>
>>   <body>
>>     <script src="sip-0.7.0.js"></script>
>>     <script src="example.js"></script>
>>   </body>
>> </html>
>>
>> and *example.js*
>> var config = {
>>
>> // Replace this IP address with your FreeSWITCH IP address
>>  uri: '1003 at 192.168.1.30',
>>
>>   // Replace this IP address with your FreeSWITCH IP address
>>   // and replace the port with your FreeSWITCH port
>>   ws_servers: 'ws://192.168.1.30:5066',
>>
>>   // FreeSWITCH Default Username
>>   authorizationUser: '1003',
>>
>>   // FreeSWITCH Default Password
>>   password: '1234'
>>   register: true
>>
>> };
>>
>> var bob = new SIP.UA(config);
>>
>> bob.start();
>>
>> Both files are placed inside apache folder with name freeswitch_rtc.
>> When i run in my firefox.
>>
>> *http://192.168.1.30/freeswitch_rtc/
>> <http://192.168.1.30/freeswitch_rtc/>*
>>
>> I see nothing neither on fs_curl, nor tshark shows any register packet on
>> freeswitch ip.
>>
>> Pardon me for my naive question, just learning how this amazing machine
>> (freeswitch) works.
>>
>> Can someone please tell me what is wrong with this code. Or provide me
>> some simple example to make a call with webrtc or any tutorial.
>>
>> 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
>>
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Tue May 26 20:41:11 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 26 May 2015 12:41:11 -0400
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <CAHtxdDfiiW0yE+QdZXZ=YZc4p355mdTJ-BF8TpKFBCB8uL6nQQ@mail.gmail.com>
References: <43162995.20150526094002@seznam.cz>
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
	<16210177989.20150526120710@seznam.cz>
	<CAHtxdDd=vo6uwo-7imdo7MEGuRDC2PYbqk3HyLvvD1rdfkZxRw@mail.gmail.com>
	<1946256680.20150526171049@seznam.cz>
	<01CF76A0-3760-446E-9A5F-EC5025163004@jerris.com>
	<CAHtxdDfiiW0yE+QdZXZ=YZc4p355mdTJ-BF8TpKFBCB8uL6nQQ@mail.gmail.com>
Message-ID: <CAEwTsAicBL-TmZWLoTTyPJtttPCrXQ7JafYSPbhX_QRE8fr7mA@mail.gmail.com>

As I have said before.  Do NOT do that.  It will lead to a broken system.

On Tuesday, May 26, 2015, Sergey Safarov <s.safarov at gmail.com> wrote:

> For Michael
> Denis has installed server from my repo (FS-7553
> <https://freeswitch.org/jira/browse/FS-7553>) and libvpx 1.4.0 has not
> correct libnames (more info below). Prevision recommendation taken to
> workaround it.
>
> [root at localhost ~]# rpm -qa | grep libvpx
> libvpx-devel-1.4.0-6.el7.centos.x86_64
> libvpx-1.4.0-6.el7.centos.x86_64
>
> [root at localhost ~]# ls -l /usr/lib64/libvpx.so*
> lrwxrwxrwx. 1 root root      15 May 16 01:48 /usr/lib64/libvpx.so ->
> libvpx.so.2.0.0
> lrwxrwxrwx. 1 root root      15 May 16 01:48 /usr/lib64/libvpx.so.2 ->
> libvpx.so.2.0.0
> lrwxrwxrwx. 1 root root      15 May 16 01:48 /usr/lib64/libvpx.so.2.0 ->
> libvpx.so.2.0.0
> -rwxr-xr-x. 1 root root 1548944 May 16 01:43 /usr/lib64/libvpx.so.2.0.0
>
>
>
> On Tue, May 26, 2015 at 7:07 PM, Michael Jerris <mike at jerris.com
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>
>> you should NOT be doing that.  That will 100% for sure crete a broken
>> system,  Any other libs that link to freeswitch will need to be linked
>> against the updated vpx lib.
>>
>>
>> On May 26, 2015, at 11:10 AM, Denis Jakovlev <yadenis at seznam.cz
>> <javascript:_e(%7B%7D,'cvml','yadenis at seznam.cz');>> wrote:
>>
>>  Dobr? den,
>>
>> ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1
>> ln: failed to create symbolic link ?/usr/lib64/libvpx.so.1?: No such file
>> or directory
>>
>>
>> I have libvpx.so in /usr/x_86_64_linux_gnu. But it steal not work.
>>
>>
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel>. 775-415-382 ?ter? 26. kv?tna 2015, 14:57:11, napsal jste:
>> *
>>  execute as root "ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1"
>> And try again
>>
>> On Tue, May 26, 2015 at 1:07 PM, Denis Jakovlev <yadenis at seznam.cz
>> <javascript:_e(%7B%7D,'cvml','yadenis at seznam.cz');>> wrote:
>> Dobr? den,
>>
>> Where can I get the -extra package for libav?
>>
>> I try from here
>> https://freeswitch.org/stash/projects/SD/repos/libav/browse  It is
>> compiled without problems.
>>
>> But I still have a error
>>
>> 2015-05-26 06:01:38.083643 [CRIT] switch_loadable_module.c:1520 Error
>> Loading module /usr/local/freeswitch/mod/mod_av.so
>> **libvpx.so.1: cannot open shared object file: No such file or directory**
>>
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel
>>
>>
>>
>> *. 775-415-382 ?ter? 26. kv?tna 2015, 11:22:35, napsal jste: *
>>  try using mod av instead.  You'll need to install the -extra package
>> for libav
>>
>> On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz
>> <javascript:_e(%7B%7D,'cvml','yadenis at seznam.cz');>> wrote:
>> Hi all !
>>
>> I have one problem with the MP4 file recording with sound. The video is
>> great. But there is no sound.
>>
>> Adding a log. files in place, but for some reason does not work. I use
>> Debian and the latest version 1.6
>>
>> EXECUTE sofia/internal/1004 at 192.168.242.132
>> <javascript:_e(%7B%7D,'cvml','sofia/internal/1004 at 192.168.242.132');>
>> record(/usr/local/freeswitch/recordings/testrecord.mp4)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot
>> open shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so'
>> (libvpx.so.1: cannot open shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so'
>> (libvpx.so.1: cannot open shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>> [00007f16f00c7fc8] core libvlc warning: cannot load module
>> `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open
>> shared object file: No such file or directory)
>> 2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open
>> #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}
>>  write video
>>
>>
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           *mob.tel*.
>> 775-415-382*
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From blasterjr at gmail.com  Tue May 26 20:53:42 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Tue, 26 May 2015 10:53:42 -0600
Subject: [Freeswitch-users] user registration with sip.js
In-Reply-To: <CAGVOjVVDJd6FzX1OVrZ2w78CUv=af73QEfut91Bod3Wkp+scXQ@mail.gmail.com>
References: <CAGVOjVVL80T=V5qsT=802TyuvkkJhWbuXdJS3_rTwBVkuq+xaA@mail.gmail.com>
	<CACXsapMnCT9wTQQiGgYurOd=UQsS81iW25UxZAMs1yTH_8emKQ@mail.gmail.com>
	<CAGVOjVVDJd6FzX1OVrZ2w78CUv=af73QEfut91Bod3Wkp+scXQ@mail.gmail.com>
Message-ID: <CAO3hyzEmA1mu741Y6mgG95LzsJyHwaNcCzDraXjP7iDRekbsVw@mail.gmail.com>

Can you try using wsServers instead of ws_servers as noted here:
http://sipjs.com/api/0.7.0/ua_configuration_parameters/#wsservers

On Tue, May 26, 2015 at 10:35 AM, Aqs Younas <aqsyounas at gmail.com> wrote:

> Thanks for your reply. Yes, i have enable the webrtc in my internal.xml
>
> tcp        0      0 192.168.1.30:5066       0.0.0.0:*
> LISTEN      11078/freeswitch
> tcp        0      0 192.168.1.30:7443       0.0.0.0:*
> LISTEN      11078/freeswitch
> tcp        0      0 127.0.0.1:8021          0.0.0.0:*
> LISTEN      11078/freeswitch
> tcp        0      0 192.168.1.30:5080       0.0.0.0:*
> LISTEN      11078/freeswitch
> tcp        0      0 192.168.1.30:5060       0.0.0.0:*
> LISTEN      11078/freeswitch
> tcp6       0      0 ::1:5080                :::*
> LISTEN      11078/freeswitch
> tcp6       0      0 ::1:5060                :::*
> LISTEN      11078/freeswitch
> udp        0      0 192.168.1.30:5080       0.0.0.0:*
> 11078/freeswitch
> udp        0      0 239.255.255.250:1900    0.0.0.0:*
> 11078/freeswitch
> udp        0      0 192.168.1.30:5060       0.0.0.0:*
> 11078/freeswitch
> udp6       0      0 ::1:5080
> :::*                                11078/freeswitch
> udp6       0      0 ::1:5060
> :::*                                11078/freeswitch
>
> May be I am missing some packages, related to js.
>
> Thanks
>
>
> On 26 May 2015 at 08:49, Victor Medina <victor.medina at cibersys.com> wrote:
>
>> Did you anable the webrtc endpoint in your profiles?
>>
>> 2015-05-26 11:04 GMT-04:30 Aqs Younas <aqsyounas at gmail.com>:
>>
>>> Hi, Users.
>>> I am trying to register default freeswitch user "1003" through webrtc
>>> using sip.js.
>>> This is my "*index.html*"
>>>
>>> <html>
>>>   <head>
>>>   </head>
>>>   <body>
>>>     <script src="sip-0.7.0.js"></script>
>>>     <script src="example.js"></script>
>>>   </body>
>>> </html>
>>>
>>> and *example.js*
>>> var config = {
>>>
>>> // Replace this IP address with your FreeSWITCH IP address
>>>  uri: '1003 at 192.168.1.30',
>>>
>>>   // Replace this IP address with your FreeSWITCH IP address
>>>   // and replace the port with your FreeSWITCH port
>>>   ws_servers: 'ws://192.168.1.30:5066',
>>>
>>>   // FreeSWITCH Default Username
>>>   authorizationUser: '1003',
>>>
>>>   // FreeSWITCH Default Password
>>>   password: '1234'
>>>   register: true
>>>
>>> };
>>>
>>> var bob = new SIP.UA(config);
>>>
>>> bob.start();
>>>
>>> Both files are placed inside apache folder with name freeswitch_rtc.
>>> When i run in my firefox.
>>>
>>> *http://192.168.1.30/freeswitch_rtc/
>>> <http://192.168.1.30/freeswitch_rtc/>*
>>>
>>> I see nothing neither on fs_curl, nor tshark shows any register packet
>>> on freeswitch ip.
>>>
>>> Pardon me for my naive question, just learning how this amazing machine
>>> (freeswitch) works.
>>>
>>> Can someone please tell me what is wrong with this code. Or provide me
>>> some simple example to make a call with webrtc or any tutorial.
>>>
>>> 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
>>>
>>
>>
>>
>> --
>>
>>
>>
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561
>> BB #79A8AFA2
>> @VMCibersys
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From blasterjr at gmail.com  Tue May 26 20:54:37 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Tue, 26 May 2015 10:54:37 -0600
Subject: [Freeswitch-users] user registration with sip.js
In-Reply-To: <CAO3hyzEmA1mu741Y6mgG95LzsJyHwaNcCzDraXjP7iDRekbsVw@mail.gmail.com>
References: <CAGVOjVVL80T=V5qsT=802TyuvkkJhWbuXdJS3_rTwBVkuq+xaA@mail.gmail.com>
	<CACXsapMnCT9wTQQiGgYurOd=UQsS81iW25UxZAMs1yTH_8emKQ@mail.gmail.com>
	<CAGVOjVVDJd6FzX1OVrZ2w78CUv=af73QEfut91Bod3Wkp+scXQ@mail.gmail.com>
	<CAO3hyzEmA1mu741Y6mgG95LzsJyHwaNcCzDraXjP7iDRekbsVw@mail.gmail.com>
Message-ID: <CAO3hyzG-3eUV=CX-zwnKmzH=Wx7FpQsdELdP2mTwdEaLw3cZxg@mail.gmail.com>

Also with the newer versions of sip.js, you do not need to call ua.start()
as it should be set to autostart.

On Tue, May 26, 2015 at 10:53 AM, Chris Tunbridge <blasterjr at gmail.com>
wrote:

> Can you try using wsServers instead of ws_servers as noted here:
> http://sipjs.com/api/0.7.0/ua_configuration_parameters/#wsservers
>
> On Tue, May 26, 2015 at 10:35 AM, Aqs Younas <aqsyounas at gmail.com> wrote:
>
>> Thanks for your reply. Yes, i have enable the webrtc in my internal.xml
>>
>> tcp        0      0 192.168.1.30:5066       0.0.0.0:*
>> LISTEN      11078/freeswitch
>> tcp        0      0 192.168.1.30:7443       0.0.0.0:*
>> LISTEN      11078/freeswitch
>> tcp        0      0 127.0.0.1:8021          0.0.0.0:*
>> LISTEN      11078/freeswitch
>> tcp        0      0 192.168.1.30:5080       0.0.0.0:*
>> LISTEN      11078/freeswitch
>> tcp        0      0 192.168.1.30:5060       0.0.0.0:*
>> LISTEN      11078/freeswitch
>> tcp6       0      0 ::1:5080                :::*
>> LISTEN      11078/freeswitch
>> tcp6       0      0 ::1:5060                :::*
>> LISTEN      11078/freeswitch
>> udp        0      0 192.168.1.30:5080       0.0.0.0:*
>> 11078/freeswitch
>> udp        0      0 239.255.255.250:1900    0.0.0.0:*
>> 11078/freeswitch
>> udp        0      0 192.168.1.30:5060       0.0.0.0:*
>> 11078/freeswitch
>> udp6       0      0 ::1:5080
>> :::*                                11078/freeswitch
>> udp6       0      0 ::1:5060
>> :::*                                11078/freeswitch
>>
>> May be I am missing some packages, related to js.
>>
>> Thanks
>>
>>
>> On 26 May 2015 at 08:49, Victor Medina <victor.medina at cibersys.com>
>> wrote:
>>
>>> Did you anable the webrtc endpoint in your profiles?
>>>
>>> 2015-05-26 11:04 GMT-04:30 Aqs Younas <aqsyounas at gmail.com>:
>>>
>>>> Hi, Users.
>>>> I am trying to register default freeswitch user "1003" through webrtc
>>>> using sip.js.
>>>> This is my "*index.html*"
>>>>
>>>> <html>
>>>>   <head>
>>>>   </head>
>>>>   <body>
>>>>     <script src="sip-0.7.0.js"></script>
>>>>     <script src="example.js"></script>
>>>>   </body>
>>>> </html>
>>>>
>>>> and *example.js*
>>>> var config = {
>>>>
>>>> // Replace this IP address with your FreeSWITCH IP address
>>>>  uri: '1003 at 192.168.1.30',
>>>>
>>>>   // Replace this IP address with your FreeSWITCH IP address
>>>>   // and replace the port with your FreeSWITCH port
>>>>   ws_servers: 'ws://192.168.1.30:5066',
>>>>
>>>>   // FreeSWITCH Default Username
>>>>   authorizationUser: '1003',
>>>>
>>>>   // FreeSWITCH Default Password
>>>>   password: '1234'
>>>>   register: true
>>>>
>>>> };
>>>>
>>>> var bob = new SIP.UA(config);
>>>>
>>>> bob.start();
>>>>
>>>> Both files are placed inside apache folder with name freeswitch_rtc.
>>>> When i run in my firefox.
>>>>
>>>> *http://192.168.1.30/freeswitch_rtc/
>>>> <http://192.168.1.30/freeswitch_rtc/>*
>>>>
>>>> I see nothing neither on fs_curl, nor tshark shows any register packet
>>>> on freeswitch ip.
>>>>
>>>> Pardon me for my naive question, just learning how this amazing machine
>>>> (freeswitch) works.
>>>>
>>>> Can someone please tell me what is wrong with this code. Or provide me
>>>> some simple example to make a call with webrtc or any tutorial.
>>>>
>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>> V?ctor E. Medina M.
>>> Platform Architect / Chief Infrastructure
>>> +58424 291 4561
>>> BB #79A8AFA2
>>> @VMCibersys
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From blasterjr at gmail.com  Tue May 26 20:56:15 2015
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Tue, 26 May 2015 10:56:15 -0600
Subject: [Freeswitch-users] user registration with sip.js
In-Reply-To: <CAO3hyzG-3eUV=CX-zwnKmzH=Wx7FpQsdELdP2mTwdEaLw3cZxg@mail.gmail.com>
References: <CAGVOjVVL80T=V5qsT=802TyuvkkJhWbuXdJS3_rTwBVkuq+xaA@mail.gmail.com>
	<CACXsapMnCT9wTQQiGgYurOd=UQsS81iW25UxZAMs1yTH_8emKQ@mail.gmail.com>
	<CAGVOjVVDJd6FzX1OVrZ2w78CUv=af73QEfut91Bod3Wkp+scXQ@mail.gmail.com>
	<CAO3hyzEmA1mu741Y6mgG95LzsJyHwaNcCzDraXjP7iDRekbsVw@mail.gmail.com>
	<CAO3hyzG-3eUV=CX-zwnKmzH=Wx7FpQsdELdP2mTwdEaLw3cZxg@mail.gmail.com>
Message-ID: <CAO3hyzEH6Xo3=E0PDptChuWvCNDUZPiQjHO7s5UiHKq_fPKmMg@mail.gmail.com>

the uri field should also be in format of sip:<ext>@<domain>

On Tue, May 26, 2015 at 10:54 AM, Chris Tunbridge <blasterjr at gmail.com>
wrote:

> Also with the newer versions of sip.js, you do not need to call ua.start()
> as it should be set to autostart.
>
> On Tue, May 26, 2015 at 10:53 AM, Chris Tunbridge <blasterjr at gmail.com>
> wrote:
>
>> Can you try using wsServers instead of ws_servers as noted here:
>> http://sipjs.com/api/0.7.0/ua_configuration_parameters/#wsservers
>>
>> On Tue, May 26, 2015 at 10:35 AM, Aqs Younas <aqsyounas at gmail.com> wrote:
>>
>>> Thanks for your reply. Yes, i have enable the webrtc in my internal.xml
>>>
>>> tcp        0      0 192.168.1.30:5066       0.0.0.0:*
>>> LISTEN      11078/freeswitch
>>> tcp        0      0 192.168.1.30:7443       0.0.0.0:*
>>> LISTEN      11078/freeswitch
>>> tcp        0      0 127.0.0.1:8021          0.0.0.0:*
>>> LISTEN      11078/freeswitch
>>> tcp        0      0 192.168.1.30:5080       0.0.0.0:*
>>> LISTEN      11078/freeswitch
>>> tcp        0      0 192.168.1.30:5060       0.0.0.0:*
>>> LISTEN      11078/freeswitch
>>> tcp6       0      0 ::1:5080                :::*
>>> LISTEN      11078/freeswitch
>>> tcp6       0      0 ::1:5060                :::*
>>> LISTEN      11078/freeswitch
>>> udp        0      0 192.168.1.30:5080       0.0.0.0:*
>>> 11078/freeswitch
>>> udp        0      0 239.255.255.250:1900    0.0.0.0:*
>>> 11078/freeswitch
>>> udp        0      0 192.168.1.30:5060       0.0.0.0:*
>>> 11078/freeswitch
>>> udp6       0      0 ::1:5080
>>> :::*                                11078/freeswitch
>>> udp6       0      0 ::1:5060
>>> :::*                                11078/freeswitch
>>>
>>> May be I am missing some packages, related to js.
>>>
>>> Thanks
>>>
>>>
>>> On 26 May 2015 at 08:49, Victor Medina <victor.medina at cibersys.com>
>>> wrote:
>>>
>>>> Did you anable the webrtc endpoint in your profiles?
>>>>
>>>> 2015-05-26 11:04 GMT-04:30 Aqs Younas <aqsyounas at gmail.com>:
>>>>
>>>>> Hi, Users.
>>>>> I am trying to register default freeswitch user "1003" through webrtc
>>>>> using sip.js.
>>>>> This is my "*index.html*"
>>>>>
>>>>> <html>
>>>>>   <head>
>>>>>   </head>
>>>>>   <body>
>>>>>     <script src="sip-0.7.0.js"></script>
>>>>>     <script src="example.js"></script>
>>>>>   </body>
>>>>> </html>
>>>>>
>>>>> and *example.js*
>>>>> var config = {
>>>>>
>>>>> // Replace this IP address with your FreeSWITCH IP address
>>>>>  uri: '1003 at 192.168.1.30',
>>>>>
>>>>>   // Replace this IP address with your FreeSWITCH IP address
>>>>>   // and replace the port with your FreeSWITCH port
>>>>>   ws_servers: 'ws://192.168.1.30:5066',
>>>>>
>>>>>   // FreeSWITCH Default Username
>>>>>   authorizationUser: '1003',
>>>>>
>>>>>   // FreeSWITCH Default Password
>>>>>   password: '1234'
>>>>>   register: true
>>>>>
>>>>> };
>>>>>
>>>>> var bob = new SIP.UA(config);
>>>>>
>>>>> bob.start();
>>>>>
>>>>> Both files are placed inside apache folder with name freeswitch_rtc.
>>>>> When i run in my firefox.
>>>>>
>>>>> *http://192.168.1.30/freeswitch_rtc/
>>>>> <http://192.168.1.30/freeswitch_rtc/>*
>>>>>
>>>>> I see nothing neither on fs_curl, nor tshark shows any register packet
>>>>> on freeswitch ip.
>>>>>
>>>>> Pardon me for my naive question, just learning how this amazing
>>>>> machine (freeswitch) works.
>>>>>
>>>>> Can someone please tell me what is wrong with this code. Or provide me
>>>>> some simple example to make a call with webrtc or any tutorial.
>>>>>
>>>>> 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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>>
>>>>
>>>> V?ctor E. Medina M.
>>>> Platform Architect / Chief Infrastructure
>>>> +58424 291 4561
>>>> BB #79A8AFA2
>>>> @VMCibersys
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.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 May 26 21:40:09 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 26 May 2015 17:40:09 +0000
Subject: [Freeswitch-users] 1.6 install to Centos
In-Reply-To: <12BAB240-50D1-4779-AC55-8B0B1D91C4B0@jerris.com>
References: <17898630.20150512135025@seznam.cz>
	<CAHtxdDe09+FFaiYBNyPTcoaJAxX30cLKn6vn1m=yiN39esrwZg@mail.gmail.com>
	<12BAB240-50D1-4779-AC55-8B0B1D91C4B0@jerris.com>
Message-ID: <CAHtxdDeKZPUaUJHbKwVz_ixGXU6NBxm37B3QmJmre-TJbb_WDQ@mail.gmail.com>

Michael sorry for the haste. Repo removed from public access.

On Tue, May 26, 2015, 19:34 Michael Jerris <mike at jerris.com> wrote:

> Please do NOT encourage people to use outside repos.  Please work with us
> to create our own repos.  To all, use any repos not on the freeswitch
> domain at your own risk.
>
> On May 26, 2015, at 12:23 PM, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> First iteration for FS rpms is made.
> You try can install FS 1.6 from rpm ecxecuting command below
> 1) wget -P /etc/yum.repos.d http://fs-repo.network-engineer.ru/fs.repo
> 2) yum -y install freeswitch-config-vanilla freeswitch-format-vlc
> 3) /usr/bin/freeswitch -nc
> 4) fs_cli
>
> TODO:
> 1) Declare rpm dependences for new FS modules
> 2) mock FS source rpm
> 3) fix header file in SD/libilbc
> 4) minimize vlc dependency
> 5) compile FS with libvpx 2.0.0
>
>
> On Tue, May 12, 2015 at 2:50 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
>
>>  Hi all,
>>
>> Is there a chance to put on CentOS the new version 1.6?
>>
>> I've tried. But I have a problem on ./configure -C
>> checking for libyuv >= 0.0.1280... configure: error: You need to install
>> libyuv-dev. Required library
>>
>> Where can I find this library? I tried to put it from here
>> https://code.google.com/p/libyuv/ But FreeSwitch does not see it.
>>
>> Is there a possibility to find this library to CentOS?
>>
>>
>>
>>
>> *-- S pozdravem, Ing.Denis Jakovlev                           mob.tel
>> <http://mob.tel/>. 775-415-382*
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 26 21:59:16 2015
From: brian at freeswitch.org (Brian West)
Date: Tue, 26 May 2015 12:59:16 -0500
Subject: [Freeswitch-users] Join the Discussion at ClueCon 2015,
	Register before June 1st!
Message-ID: <CAEJMVkB+rctwOdYcLgT8AcuFpL=Z10J2tv9-Zk9zO5LK3DShXA@mail.gmail.com>

[image: Description: HDD:Users:anthm:Downloads:ccxx.jpg]

*2015 Update*

August 3rd ? August 6th 2015

877-742-2583  ?  marketing at cluecon.com

Register NOW!

?   $899 Staying at the Hotel

?   $1199 Staying Elsewhere

?   First 100 Gino?s PIZZA PARTY


*IMPORTANT DATES*

*Before June 1st*

15 Raffle Tickets Per Day

2 Gigabit Drink Coupons


*Before June 15th*

10 Raffle Tickets Per Day

2 Gigabit Drink Coupons


*Before July 1st*

5 Raffle Tickets Per Day

1 Gigabit Drink Coupons


*Contact Us*

https://cluecon.com

marketing at cluecon.com



*Speaker Submission Deadline*

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*Suggested Speaking Topics*

WebRTC Applications

Interesting Open Source Apps

Troubleshooting Networks

VoIP Database Integration

Clustering and H/A

Tech and Data Security

Standards

Billing, Routing and Rating

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SIP

TDM

IVR

Open Source Project Status

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Hosted API

Advanced Configurations

Your Latest and Greatest



*ORTC, Security and FreeSWITCH Roundtables, Coder Games and MUCH MORE at
ClueCon 2015!*



*Roundtable Schedule:*

August 4th - Object RTC (ORTC)

August 5th - Security Discussion
August 6th - FreeSWITCH Platform Discussion

Join ClueCon Weekly Tomorrow *May 27th *at* 1PM Eastern*, Robin Raymond
will be discussing the Object RTC (ORTC) project and working group.


Not only will it give you piece of mind, the sooner you register, the more
opportunities you will get to win prizes! You?ll also get more drink
coupons for the Gigabit Reception Tuesday Night!



[image: Description: mac1]The grand prize is a laser engraved commemorative
FreeSWITCH 1.6 Edition dual-core 13" Retina MacBook Pro!  See the Important
Dates Section for Registration details!








 Why I Think You MUST COME To ClueCon!



[image: Description: kk]Hi, I?m Kathleen. I?m the FreeSWITCH and ClueCon
Social Media Correspondent.  I?ve been working hard all year keeping you
all up to date on what?s going on with FreeSWITCH. Today I?m here to let
you know more about the upcoming ClueCon 2015 Conference!



This year we are adding an optional day on Monday with an all-day
Hack-A-Thon with great coding contests, game shows and kick-off fun!



If you are interested in WebRTC, Voice over IP or Open Source projects like
FreeSWITCH, ClueCon is the greatest opportunity you have to gain exposure
to the most knowledge and technology in one place.  Also, it?s the most fun
you can possibly have while still getting a ton of work done!  I really
look forward to seeing you all there and enjoying the amazing talks, the
Epic Annual Kick-Off Pizza Party, The Gigabit Reception and so much more.
Make sure you register today so you can reserve your place among the
attendees!


Be sure to follow us on Facebook and Twitter to get my latest updates in
info!


[image: Inline image 1] <https://facebook.com/cluecon> [image: Inline image
2] <https://twitter.com/cluecon>



 It?s Not Too Late to Sign Up as a Speaker



Do you have a great idea or an awesome project you have been working on? We
would love to hear about it! Show off what you know!

[image: Description: fss]Sign up to speak at ClueCon this year and let us
know what you?ve been doing. The call for speakers page for ClueCon 2015 is
now open and you can visit https://cluecon.com/speaking-proposals to submit
your speaking proposal right now!





[image: Inline image 3] <https://www.youtube.com/watch?v=NLaDpGQuZDA>


 [image: Inline image 4] <https://www.youtube.com/watch?v=9XXgW34t40s>
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From victor.medina at cibersys.com  Tue May 26 23:01:05 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Tue, 26 May 2015 14:31:05 -0430
Subject: [Freeswitch-users] Video always muted in conference (fs_video2)
In-Reply-To: <C6A42EE2-E59D-4F1A-8AB6-B3C61962D41A@jerris.com>
References: <CAFKjMAYjKz32rE-G_aA639J5b27eNwnRAME31La25W9KiWbagA@mail.gmail.com>
	<7B131C27-7233-4E88-BB9E-E01BE588DE82@jerris.com>
	<CAFKjMAZ30sxNtAXU1iqksuNwM53gzQaqFQor7vhVHhg-3VBn5w@mail.gmail.com>
	<CAEwTsAgsVXvE5Ym9TWvbSvd9i7gCgNWwmp-2adxYKug0RHHKBA@mail.gmail.com>
	<CACXsapNJ_ErsOdZzc=_f4DJpzYe6NRGeNB8uRE1Uw1BSAaGOJg@mail.gmail.com>
	<C6A42EE2-E59D-4F1A-8AB6-B3C61962D41A@jerris.com>
Message-ID: <CACXsapNBG_uGDKnuQ=3J2MzzZ00uquOQDeqwt6KiAcR_yRM=ag@mail.gmail.com>

did it but not added the debug logs... wait...

2015-05-26 11:38 GMT-04:30 Michael Jerris <mike at jerris.com>:

> Please file a jira with details on how to reproduce and debug logs.
>
> On May 26, 2015, at 11:51 AM, Victor Medina <victor.medina at cibersys.com>
> wrote:
>
> Michael...
>
> I just build FS with the latest sources and the Video Unmuted/Video Muted
> ramains.
> Just to confirm this.
>
> If you guys need me to test anything I have a full blown server just for
> me and fs tests.
>
> 2015-05-25 0:08 GMT-04:30 Michael Jerris <mike at jerris.com>:
>
>> Some fixes were pushed for this on Friday.  Try out latest code and see
>> if that works any better for you?
>>
>>
>> On Saturday, May 23, 2015, Carlos Gonz?lez Florido <carlosj.gf at gmail.com>
>> wrote:
>>
>>> Ok, tell me if I can help with the testing.
>>> Is there a way to disconnect ipv6?
>>>
>>> On Fri, May 22, 2015 at 6:04 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> I don't think its actually muted.  We are chasing down an issue that
>>>> sounds just like this.  It might be related to ipv6 but we are still
>>>> looking in to it.
>>>>
>>>> > On May 22, 2015, at 8:40 AM, Carlos Gonz?lez Florido <
>>>> carlosj.gf at gmail.com> wrote:
>>>> >
>>>> > Hello,
>>>> >
>>>> > I'm testing the impressive fs_video2 branch, but I have the following
>>>> problems:
>>>> >
>>>> > - some participants have their video always muted (and their camera
>>>> is on and working for Hangouts, for example).
>>>> > - other participants start the same (video muted), but after 3-5
>>>> minutes the video is automatically unmuted.
>>>> > - if we do a layout change (using fs_cli), all participants (having
>>>> video muted or unmuted at that moment) go to muted video for the rest of
>>>> the conference.
>>>> >
>>>> > What is the reason for fs to automatically mute the video? Is this
>>>> expected or does it look like a bug? Can this behaviour be tuned or
>>>> disconnected?
>>>> >
>>>> > Thank you,
>>>> > Carlos Gonzalez
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From aqsyounas at gmail.com  Tue May 26 23:10:41 2015
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 26 May 2015 12:10:41 -0700
Subject: [Freeswitch-users] user registration with sip.js
In-Reply-To: <CAO3hyzEH6Xo3=E0PDptChuWvCNDUZPiQjHO7s5UiHKq_fPKmMg@mail.gmail.com>
References: <CAGVOjVVL80T=V5qsT=802TyuvkkJhWbuXdJS3_rTwBVkuq+xaA@mail.gmail.com>
	<CACXsapMnCT9wTQQiGgYurOd=UQsS81iW25UxZAMs1yTH_8emKQ@mail.gmail.com>
	<CAGVOjVVDJd6FzX1OVrZ2w78CUv=af73QEfut91Bod3Wkp+scXQ@mail.gmail.com>
	<CAO3hyzEmA1mu741Y6mgG95LzsJyHwaNcCzDraXjP7iDRekbsVw@mail.gmail.com>
	<CAO3hyzG-3eUV=CX-zwnKmzH=Wx7FpQsdELdP2mTwdEaLw3cZxg@mail.gmail.com>
	<CAO3hyzEH6Xo3=E0PDptChuWvCNDUZPiQjHO7s5UiHKq_fPKmMg@mail.gmail.com>
Message-ID: <CAGVOjVW0AQbB9xXXNrMXXgobT73hRCxi=Bbc502UGHLQ0cukGQ@mail.gmail.com>

Thanks for you reply. Done changes according to your recommendations.

var config = {
  uri: 'sip:1003 at 192.168.1.30',
  wsServers: 'ws://192.168.1.30:5066',
  authorizationUser: '1003',
  password: '1234',
  register: true
};
var bob = new SIP.UA(config);

But Still no error on firefox console, no log on fs_curl, tshark neither
shows any SIP packet on the network.

Am i executing it correctely? (*http://192.168.1.30/freeswitch_rtc/
<http://192.168.1.30/freeswitch_rtc/>)*

Thanks.

On 26 May 2015 at 09:56, Chris Tunbridge <blasterjr at gmail.com> wrote:

> the uri field should also be in format of sip:<ext>@<domain>
>
> On Tue, May 26, 2015 at 10:54 AM, Chris Tunbridge <blasterjr at gmail.com>
> wrote:
>
>> Also with the newer versions of sip.js, you do not need to call
>> ua.start() as it should be set to autostart.
>>
>> On Tue, May 26, 2015 at 10:53 AM, Chris Tunbridge <blasterjr at gmail.com>
>> wrote:
>>
>>> Can you try using wsServers instead of ws_servers as noted here:
>>> http://sipjs.com/api/0.7.0/ua_configuration_parameters/#wsservers
>>>
>>> On Tue, May 26, 2015 at 10:35 AM, Aqs Younas <aqsyounas at gmail.com>
>>> wrote:
>>>
>>>> Thanks for your reply. Yes, i have enable the webrtc in my internal.xml
>>>>
>>>> tcp        0      0 192.168.1.30:5066       0.0.0.0:*
>>>> LISTEN      11078/freeswitch
>>>> tcp        0      0 192.168.1.30:7443       0.0.0.0:*
>>>> LISTEN      11078/freeswitch
>>>> tcp        0      0 127.0.0.1:8021          0.0.0.0:*
>>>> LISTEN      11078/freeswitch
>>>> tcp        0      0 192.168.1.30:5080       0.0.0.0:*
>>>> LISTEN      11078/freeswitch
>>>> tcp        0      0 192.168.1.30:5060       0.0.0.0:*
>>>> LISTEN      11078/freeswitch
>>>> tcp6       0      0 ::1:5080                :::*
>>>> LISTEN      11078/freeswitch
>>>> tcp6       0      0 ::1:5060                :::*
>>>> LISTEN      11078/freeswitch
>>>> udp        0      0 192.168.1.30:5080       0.0.0.0:*
>>>> 11078/freeswitch
>>>> udp        0      0 239.255.255.250:1900    0.0.0.0:*
>>>> 11078/freeswitch
>>>> udp        0      0 192.168.1.30:5060       0.0.0.0:*
>>>> 11078/freeswitch
>>>> udp6       0      0 ::1:5080
>>>> :::*                                11078/freeswitch
>>>> udp6       0      0 ::1:5060
>>>> :::*                                11078/freeswitch
>>>>
>>>> May be I am missing some packages, related to js.
>>>>
>>>> Thanks
>>>>
>>>>
>>>> On 26 May 2015 at 08:49, Victor Medina <victor.medina at cibersys.com>
>>>> wrote:
>>>>
>>>>> Did you anable the webrtc endpoint in your profiles?
>>>>>
>>>>> 2015-05-26 11:04 GMT-04:30 Aqs Younas <aqsyounas at gmail.com>:
>>>>>
>>>>>> Hi, Users.
>>>>>> I am trying to register default freeswitch user "1003" through webrtc
>>>>>> using sip.js.
>>>>>> This is my "*index.html*"
>>>>>>
>>>>>> <html>
>>>>>>   <head>
>>>>>>   </head>
>>>>>>   <body>
>>>>>>     <script src="sip-0.7.0.js"></script>
>>>>>>     <script src="example.js"></script>
>>>>>>   </body>
>>>>>> </html>
>>>>>>
>>>>>> and *example.js*
>>>>>> var config = {
>>>>>>
>>>>>> // Replace this IP address with your FreeSWITCH IP address
>>>>>>  uri: '1003 at 192.168.1.30',
>>>>>>
>>>>>>   // Replace this IP address with your FreeSWITCH IP address
>>>>>>   // and replace the port with your FreeSWITCH port
>>>>>>   ws_servers: 'ws://192.168.1.30:5066',
>>>>>>
>>>>>>   // FreeSWITCH Default Username
>>>>>>   authorizationUser: '1003',
>>>>>>
>>>>>>   // FreeSWITCH Default Password
>>>>>>   password: '1234'
>>>>>>   register: true
>>>>>>
>>>>>> };
>>>>>>
>>>>>> var bob = new SIP.UA(config);
>>>>>>
>>>>>> bob.start();
>>>>>>
>>>>>> Both files are placed inside apache folder with name freeswitch_rtc.
>>>>>> When i run in my firefox.
>>>>>>
>>>>>> *http://192.168.1.30/freeswitch_rtc/
>>>>>> <http://192.168.1.30/freeswitch_rtc/>*
>>>>>>
>>>>>> I see nothing neither on fs_curl, nor tshark shows any register
>>>>>> packet on freeswitch ip.
>>>>>>
>>>>>> Pardon me for my naive question, just learning how this amazing
>>>>>> machine (freeswitch) works.
>>>>>>
>>>>>> Can someone please tell me what is wrong with this code. Or provide
>>>>>> me some simple example to make a call with webrtc or any tutorial.
>>>>>>
>>>>>> 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
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>>
>>>>>
>>>>> V?ctor E. Medina M.
>>>>> Platform Architect / Chief Infrastructure
>>>>> +58424 291 4561
>>>>> BB #79A8AFA2
>>>>> @VMCibersys
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.medina at cibersys.com  Tue May 26 23:29:58 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Tue, 26 May 2015 14:59:58 -0430
Subject: [Freeswitch-users] How to enable Screen sharing on Verto Conf?
In-Reply-To: <CAKbxfG-WD03p7V=+myKHL9GypknQozzfqcJJtVK3Z+kL=M_JZw@mail.gmail.com>
References: <CACXsapOA9nG0uo3+artAT72DD17umf9Ck5dHCKjPXDzjd1+s_Q@mail.gmail.com>
	<CACXsapOLEbJA9Dfbfc1YMOnSHa73eTMnm+NUG5qs=N0Xz3gzYQ@mail.gmail.com>
	<CAEwTsAhpyeuBuCXB1bePAiOXPP8RtaYMgaSiiLjP6J+5aMVyFQ@mail.gmail.com>
	<CACXsapM8YLag7fauLZpqJrApqiiModv72HAiaOAnc+HGJKV6dQ@mail.gmail.com>
	<CAEwTsAgGdi24bPQMRsdMKJiXgyrBLYi9ZV_CA4JpN-LDavJhmQ@mail.gmail.com>
	<CACXsapPmFHUdq9QVvO6kxpBSmYZEii1EDS32wLHahYb1gUdhdA@mail.gmail.com>
	<CAKbxfG-WD03p7V=+myKHL9GypknQozzfqcJJtVK3Z+kL=M_JZw@mail.gmail.com>
Message-ID: <CACXsapP4o5S1H=_c0-y6O9V5Q3CLDyGmDqLcyCqJUuG45y05hg@mail.gmail.com>

Um....

But I did not add that!
Im currently using, following, the conference demo setup as it is. No
modifications so far to the dialplan.

384     <!-- STEREO 48kHz conferences / Video MCU -->
385     <extension name="cdquality_conferences">
386       <condition field="destination_number" expression="^(35\d{2})$">
387         <action application="answer"/>
388         <action application="conference"
data="$1-${domain_name}@video-mcu-stereo"/>
389       </condition>
390     </extension>




2015-05-26 11:24 GMT-04:30 Anthony Minessale <anthony.minessale at gmail.com>:

> The extension it calls is the same extension the original call is calling
> only it also adds -screen to the end so you need your dialplan to catch
> that as well and do whatever differently you may do such as not setting
> banners etc.
>
>
> On Tue, May 26, 2015 at 10:19 AM, Victor Medina <
> victor.medina at cibersys.com> wrote:
>
>> Im seeing this while starting the screen sharing...
>>
>> 2015-05-26 11:15:36.690561 [NOTICE] switch_channel.c:1089 New Channel
>> verto.rtc/3500-screen [d4576cb3-e967-2bb1-be3c-a4831f8894fd]
>> 2015-05-26 11:15:36.690561 [DEBUG] mod_verto.c:3407 Remote SDP
>> verto.rtc/3500-screen:
>> v=0
>> o=- 4034581275591452868 2 IN IP4 127.0.0.1
>> s=-
>> t=0 0
>> a=group:BUNDLE audio video
>> a=msid-semantic: WMS fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
>> m=audio 33952 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>> c=IN IP4 186.14.135.35
>> a=rtcp:33426 IN IP4 186.14.135.35
>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 58898 typ host
>> generation 0
>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 58899 typ host
>> generation 0
>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 58900 typ host
>> generation 0
>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 58901 typ host
>> generation 0
>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 33952 typ srflx
>> raddr 10.0.1.10 rport 58899 generation 0
>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 33426 typ srflx
>> raddr 10.0.1.10 rport 58901 generation 0
>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
>> active generation 0
>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>> active generation 0
>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
>> active generation 0
>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>> active generation 0
>> a=ice-ufrag:DXEOOI0UFrnM6fo8
>> a=ice-pwd:l3x/4ga4HunTyP6rGwExMzgv
>> a=fingerprint:sha-256
>> BF:00:AC:5F:C8:84:54:5F:EF:34:3C:EC:AF:1C:92:80:B6:16:45:EF:3A:39:A1:3C:9E:AF:88:B0:E3:06:BA:AC
>> a=setup:actpass
>> a=mid:audio
>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>> a=recvonly
>> a=rtcp-mux
>> a=rtpmap:111 opus/48000/2
>> a=fmtp:111 minptime=10; useinbandfec=1
>> a=rtpmap:103 ISAC/16000
>> a=rtpmap:104 ISAC/32000
>> a=rtpmap:9 G722/8000
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:106 CN/32000
>> a=rtpmap:105 CN/16000
>> a=rtpmap:13 CN/8000
>> a=rtpmap:126 telephone-event/8000
>> a=maxptime:60
>> m=video 42360 RTP/SAVPF 100 116 117 96
>> c=IN IP4 186.14.135.35
>> a=rtcp:44680 IN IP4 186.14.135.35
>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 58902 typ host
>> generation 0
>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 58903 typ host
>> generation 0
>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 58904 typ host
>> generation 0
>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 58905 typ host
>> generation 0
>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 42360 typ srflx
>> raddr 10.0.1.10 rport 58903 generation 0
>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 44680 typ srflx
>> raddr 10.0.1.10 rport 58905 generation 0
>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host tcptype
>> active generation 0
>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>> active generation 0
>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host tcptype
>> active generation 0
>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>> active generation 0
>> a=ice-ufrag:DXEOOI0UFrnM6fo8
>> a=ice-pwd:l3x/4ga4HunTyP6rGwExMzgv
>> a=fingerprint:sha-256
>> BF:00:AC:5F:C8:84:54:5F:EF:34:3C:EC:AF:1C:92:80:B6:16:45:EF:3A:39:A1:3C:9E:AF:88:B0:E3:06:BA:AC
>> a=setup:actpass
>> a=mid:video
>> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>> a=extmap:4 urn:3gpp:video-orientation
>> a=sendrecv
>> a=rtcp-mux
>> a=rtpmap:100 VP8/90000
>> a=rtcp-fb:100 ccm fir
>> a=rtcp-fb:100 nack
>> a=rtcp-fb:100 nack pli
>> a=rtcp-fb:100 goog-remb
>> a=rtpmap:116 red/90000
>> a=rtpmap:117 ulpfec/90000
>> a=rtpmap:96 rtx/90000
>> a=fmtp:96 apt=100
>> a=ssrc-group:FID 2871497692 3867014262
>> a=ssrc:2871497692 cname:e4V6+GZYAdJCwcTf
>> a=ssrc:2871497692 msid:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
>> ef8fba5e-3643-4dbd-a038-5023f4521ba4
>> a=ssrc:2871497692 mslabel:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
>> a=ssrc:2871497692 label:ef8fba5e-3643-4dbd-a038-5023f4521ba4
>> a=ssrc:3867014262 cname:e4V6+GZYAdJCwcTf
>> a=ssrc:3867014262 msid:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
>> ef8fba5e-3643-4dbd-a038-5023f4521ba4
>> a=ssrc:3867014262 mslabel:fHwWD5qx8NCmxjq8teycgmPFEEjdsnwq6IKh
>> a=ssrc:3867014262 label:ef8fba5e-3643-4dbd-a038-5023f4521ba4
>>
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
>> (verto.rtc/3500-screen) Running State Change CS_INIT
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:516
>> (verto.rtc/3500-screen) State INIT
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:40
>> verto.rtc/3500-screen Standard INIT
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:48
>> (verto.rtc/3500-screen) State Change CS_INIT -> CS_ROUTING
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:516
>> (verto.rtc/3500-screen) State INIT going to sleep
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
>> (verto.rtc/3500-screen) Running State Change CS_ROUTING
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_channel.c:2234
>> (verto.rtc/3500-screen) Callstate Change DOWN -> RINGING
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:532
>> (verto.rtc/3500-screen) State ROUTING
>> 2015-05-26 11:15:36.690561 [DEBUG] mod_rtc.c:89 verto.rtc/3500-screen RTC
>> ROUTING
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:166
>> verto.rtc/3500-screen Standard ROUTING
>> 2015-05-26 11:15:36.690561 [INFO] mod_dialplan_xml.c:636 Processing
>> Victor Medina (Screen) <1000 (screen)>->3500-screen in context default
>> Dialplan: verto.rtc/3500-screen parsing [default->unloop] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [unloop]
>> ${sip_looped_call}() =~ /^true$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->tod_example]
>> continue=true
>> Dialplan: verto.rtc/3500-screen Date/Time Match (PASS) [tod_example]
>> break=on-false
>> Dialplan: verto.rtc/3500-screen Action set(open=true)
>> Dialplan: verto.rtc/3500-screen parsing [default->holiday_example]
>> continue=true
>> Dialplan: verto.rtc/3500-screen Date/TimeMatch (FAIL) [holiday_example]
>> break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->global-intercept]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [global-intercept]
>> destination_number(3500-screen) =~ /^886$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->group-intercept]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [group-intercept]
>> destination_number(3500-screen) =~ /^\*8$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->intercept-ext]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [intercept-ext]
>> destination_number(3500-screen) =~ /^\*\*(\d+)$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->redial] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [redial]
>> destination_number(3500-screen) =~ /^(redial|870)$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->global] continue=true
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [global]
>> ${call_debug}(false) =~ /^true$/ break=never
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [global]
>> ${default_password}(p6rSp6rSqwerty) =~ /^1234$/ break=never
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [global] ${rtp_has_crypto}()
>> =~
>> /^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/
>> break=never
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [global]
>> ${endpoint_disposition}() =~ /^(DELAYED NEGOTIATION)/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->snom-demo-2]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [snom-demo-2]
>> destination_number(3500-screen) =~ /^9001$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->snom-demo-1]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [snom-demo-1]
>> destination_number(3500-screen) =~ /^9000$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->eavesdrop]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [eavesdrop]
>> destination_number(3500-screen) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->eavesdrop]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [eavesdrop]
>> destination_number(3500-screen) =~ /^779$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->call_return]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [call_return]
>> destination_number(3500-screen) =~ /^\*69$|^869$|^lcr$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->del-group]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [del-group]
>> destination_number(3500-screen) =~ /^80(\d{2})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->add-group]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [add-group]
>> destination_number(3500-screen) =~ /^81(\d{2})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->call-group-simo]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [call-group-simo]
>> destination_number(3500-screen) =~ /^82(\d{2})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->call-group-order]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [call-group-order]
>> destination_number(3500-screen) =~ /^83(\d{2})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->extension-intercom]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [extension-intercom]
>> destination_number(3500-screen) =~ /^8(10[01][0-9])$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->Local_Extension]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [Local_Extension]
>> destination_number(3500-screen) =~ /^(10[01][0-9])$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->Local_Extension_Skinny]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [Local_Extension_Skinny]
>> destination_number(3500-screen) =~ /^(11[01][0-9])$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->group_dial_sales]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [group_dial_sales]
>> destination_number(3500-screen) =~ /^2000$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->group_dial_support]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [group_dial_support]
>> destination_number(3500-screen) =~ /^2001$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->group_dial_billing]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [group_dial_billing]
>> destination_number(3500-screen) =~ /^2002$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->operator] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [operator]
>> destination_number(3500-screen) =~ /^(operator|0)$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->vmain] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [vmain]
>> destination_number(3500-screen) =~ /^vmain$|^4000$|^\*98$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->sip_uri] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [sip_uri]
>> destination_number(3500-screen) =~ /^sip:(.*)$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->nb_conferences]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [nb_conferences]
>> destination_number(3500-screen) =~ /^(30\d{2})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->wb_conferences]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [wb_conferences]
>> destination_number(3500-screen) =~ /^(31\d{2})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->uwb_conferences]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [uwb_conferences]
>> destination_number(3500-screen) =~ /^(32\d{2})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->cdquality_conferences]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [cdquality_conferences]
>> destination_number(3500-screen) =~ /^(33\d{2})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->cdquality_conferences]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [cdquality_conferences]
>> destination_number(3500-screen) =~ /^(35\d{2})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing
>> [default->freeswitch_public_conf_via_sip] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL)
>> [freeswitch_public_conf_via_sip] destination_number(3500-screen) =~
>> /^9(888|8888|1616|3232)$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->mad_boss_intercom]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [mad_boss_intercom]
>> destination_number(3500-screen) =~ /^0911$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->mad_boss_intercom]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [mad_boss_intercom]
>> destination_number(3500-screen) =~ /^0912$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->mad_boss] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [mad_boss]
>> destination_number(3500-screen) =~ /^0913$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->ivr_demo] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ivr_demo]
>> destination_number(3500-screen) =~ /^5000$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->dynamic_conference]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [dynamic_conference]
>> destination_number(3500-screen) =~ /^5001$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->rtp_multicast_page]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [rtp_multicast_page]
>> destination_number(3500-screen) =~ /^pagegroup$|^7243$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->park] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [park]
>> destination_number(3500-screen) =~ /^5900$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->unpark] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [unpark]
>> destination_number(3500-screen) =~ /^5901$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->valet_park]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [valet_park]
>> destination_number(3500-screen) =~ /^(6000)$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->valet_park]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [valet_park]
>> destination_number(3500-screen) =~ /^((?!6000)60\d{2})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->park] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [park] source(mod_verto) =~
>> /mod_sofia/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->unpark] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [unpark] source(mod_verto)
>> =~ /mod_sofia/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->park] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [park] source(mod_verto) =~
>> /mod_sofia/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->unpark] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [unpark] source(mod_verto)
>> =~ /mod_sofia/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->wait] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [wait]
>> destination_number(3500-screen) =~ /^wait$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->fax_receive]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [fax_receive]
>> destination_number(3500-screen) =~ /^9178$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->fax_transmit]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [fax_transmit]
>> destination_number(3500-screen) =~ /^9179$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->ringback_180]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_180]
>> destination_number(3500-screen) =~ /^9180$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->ringback_183_uk_ring]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_183_uk_ring]
>> destination_number(3500-screen) =~ /^9181$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing
>> [default->ringback_183_music_ring] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_183_music_ring]
>> destination_number(3500-screen) =~ /^9182$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing
>> [default->ringback_post_answer_uk_ring] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL)
>> [ringback_post_answer_uk_ring] destination_number(3500-screen) =~ /^9183$/
>> break=on-false
>> Dialplan: verto.rtc/3500-screen parsing
>> [default->ringback_post_answer_music] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ringback_post_answer_music]
>> destination_number(3500-screen) =~ /^9184$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->ClueCon] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [ClueCon]
>> destination_number(3500-screen) =~ /^9191$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->show_info]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [show_info]
>> destination_number(3500-screen) =~ /^9192$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->video_record]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [video_record]
>> destination_number(3500-screen) =~ /^9193$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->video_playback]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [video_playback]
>> destination_number(3500-screen) =~ /^9194$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->delay_echo]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [delay_echo]
>> destination_number(3500-screen) =~ /^9195$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->echo] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [echo]
>> destination_number(3500-screen) =~ /^9196$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->milliwatt]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [milliwatt]
>> destination_number(3500-screen) =~ /^9197$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->tone_stream]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [tone_stream]
>> destination_number(3500-screen) =~ /^9198$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->zrtp_enrollement]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [zrtp_enrollement]
>> destination_number(3500-screen) =~ /^9787$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->hold_music]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [hold_music]
>> destination_number(3500-screen) =~ /^9664$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->laugh break]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [laugh break]
>> destination_number(3500-screen) =~ /^9386$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->h] continue=true
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [h]
>> destination_number(3500-screen) =~ /^h264_(.*)$/ break=never
>> Dialplan: verto.rtc/3500-screen parsing [default->v] continue=true
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [v]
>> destination_number(3500-screen) =~ /^vp8_(.*)$/ break=never
>> Dialplan: verto.rtc/3500-screen parsing [default->h] continue=true
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [h]
>> destination_number(3500-screen) =~ /^hbr_(.*)$/ break=never
>> Dialplan: verto.rtc/3500-screen parsing [default->v] continue=true
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [v]
>> destination_number(3500-screen) =~ /^vbr_(.*)$/ break=never
>> Dialplan: verto.rtc/3500-screen parsing [default->bug] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [bug]
>> destination_number(3500-screen) =~ /^vbr_(.*)$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->bug] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [bug]
>> destination_number(3500-screen) =~ /^vid_(.*)$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->bug] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [bug]
>> destination_number(3500-screen) =~ /^decode$|^9952$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->101] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [101]
>> destination_number(3500-screen) =~ /^101$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->pizza_demo]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [pizza_demo]
>> destination_number(3500-screen) =~ /^(pizza|74992)$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->Talking Clock Time]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [Talking Clock Time]
>> destination_number(3500-screen) =~ /^9170$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->Talking Clock Date]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [Talking Clock Date]
>> destination_number(3500-screen) =~ /^9171$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->Talking Clock Date and
>> Time] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [Talking Clock Date and
>> Time] destination_number(3500-screen) =~ /^9172$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->local.example.com]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (PASS) [local.example.com]
>> ${toll_allow}(domestic,international,local) =~ /local/ break=on-false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [local.example.com]
>> destination_number(3500-screen) =~ /^(\d{7})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->domestic.example.com]
>> continue=false
>> Dialplan: verto.rtc/3500-screen Regex (PASS) [domestic.example.com]
>> ${toll_allow}(domestic,international,local) =~ /domestic/ break=on-false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [domestic.example.com]
>> destination_number(3500-screen) =~ /^(\d{11})$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->
>> international.example.com] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (PASS) [international.example.com]
>> ${toll_allow}(domestic,international,local) =~ /international/
>> break=on-false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [international.example.com]
>> destination_number(3500-screen) =~ /^(011\d+)$/ break=on-false
>> Dialplan: verto.rtc/3500-screen parsing [default->enum] continue=false
>> Dialplan: verto.rtc/3500-screen Regex (FAIL) [enum]
>> ${module_exists(mod_enum)}(false) =~ /true/ break=on-false
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:216
>> (verto.rtc/3500-screen) State Change CS_ROUTING -> CS_EXECUTE
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:532
>> (verto.rtc/3500-screen) State ROUTING going to sleep
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
>> (verto.rtc/3500-screen) Running State Change CS_EXECUTE
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:539
>> (verto.rtc/3500-screen) State EXECUTE
>> 2015-05-26 11:15:36.690561 [DEBUG] mod_rtc.c:120 verto.rtc/3500-screen
>> RTC EXECUTE
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:258
>> verto.rtc/3500-screen Standard EXECUTE
>> EXECUTE verto.rtc/3500-screen set(open=true)
>> 2015-05-26 11:15:36.690561 [DEBUG] mod_dptools.c:1469
>> verto.rtc/3500-screen SET [open]=[true]
>> 2015-05-26 11:15:36.690561 [NOTICE] switch_core_state_machine.c:315
>> verto.rtc/3500-screen has executed the last dialplan instruction, hanging
>> up.
>> 2015-05-26 11:15:36.690561 [NOTICE] switch_core_state_machine.c:317
>> Hangup verto.rtc/3500-screen [CS_EXECUTE] [NORMAL_CLEARING]
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:539
>> (verto.rtc/3500-screen) State EXECUTE going to sleep
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
>> (verto.rtc/3500-screen) Running State Change CS_HANGUP
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:739
>> (verto.rtc/3500-screen) Callstate Change RINGING -> HANGUP
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:741
>> (verto.rtc/3500-screen) State HANGUP
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:60
>> verto.rtc/3500-screen Standard HANGUP, cause: NORMAL_CLEARING
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:741
>> (verto.rtc/3500-screen) State HANGUP going to sleep
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:508
>> (verto.rtc/3500-screen) State Change CS_HANGUP -> CS_REPORTING
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:473
>> (verto.rtc/3500-screen) Running State Change CS_REPORTING
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:827
>> (verto.rtc/3500-screen) State REPORTING
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:104
>> verto.rtc/3500-screen Standard REPORTING, cause: NORMAL_CLEARING
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:827
>> (verto.rtc/3500-screen) State REPORTING going to sleep
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:499
>> (verto.rtc/3500-screen) State Change CS_REPORTING -> CS_DESTROY
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_session.c:1638 Session 4
>> (verto.rtc/3500-screen) Locked, Waiting on external entities
>> 2015-05-26 11:15:36.690561 [NOTICE] switch_core_session.c:1656 Session 4
>> (verto.rtc/3500-screen) Ended
>> 2015-05-26 11:15:36.690561 [NOTICE] switch_core_session.c:1660 Close
>> Channel verto.rtc/3500-screen [CS_DESTROY]
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:630
>> (verto.rtc/3500-screen) Running State Change CS_DESTROY
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:640
>> (verto.rtc/3500-screen) State DESTROY
>> 2015-05-26 11:15:36.690561 [DEBUG] mod_rtc.c:132 verto.rtc/3500-screen
>> RTC DESTROY
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:111
>> verto.rtc/3500-screen Standard DESTROY
>> 2015-05-26 11:15:36.690561 [DEBUG] switch_core_state_machine.c:640
>> (verto.rtc/3500-screen) State DESTROY going to sleep
>> freeswitch at internal>
>>
>>
>> 2015-05-26 10:24 GMT-04:30 Michael Jerris <mike at jerris.com>:
>>
>> yes.  is the call hitting your dial plan?
>>>
>>>
>>> On Tuesday, May 26, 2015, Victor Medina <victor.medina at cibersys.com>
>>> wrote:
>>>
>>>> I did both!
>>>> I installed the WebRTC Screen Capture extension and check all command
>>>> line option on chrome: --allow-http-screen-capture &
>>>> --auto-select-desktop-capture-source as of ...
>>>> http://peter.sh/experiments/chromium-command-line-switches/
>>>>
>>>> I think only the extension is needed, right?
>>>>
>>>> 2015-05-25 18:25 GMT-04:30 Michael Jerris <mike at jerris.com>:
>>>>
>>>>> for chrome to allow screen share it requires you to install a chrome
>>>>> plugin or start chrome with a command line argument to allow screen share
>>>>>
>>>>>
>>>>> On Monday, May 25, 2015, Victor Medina <victor.medina at cibersys.com>
>>>>> wrote:
>>>>>
>>>>>> ?Log..
>>>>>>
>>>>>> Attempting Screen Capture....
>>>>>> SCREEN SHARE
>>>>>> Audio constraints false
>>>>>> Video constraints Object {mandatory: Object, optional:
>>>>>> Array[0]}mandatory: Objectoptional: Array[0]__proto__: Object
>>>>>> Stream Success
>>>>>> stream started
>>>>>> Offer SDP
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> offer RTCIceCandidate {}
>>>>>> ICE Complete
>>>>>> ICE SDP
>>>>>> offer v=0
>>>>>> o=- 3905402391138908374 2 IN IP4 127.0.0.1
>>>>>> s=-
>>>>>> t=0 0
>>>>>> a=group:BUNDLE audio video
>>>>>> a=msid-semantic: WMS dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>>>> m=audio 34382 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
>>>>>> c=IN IP4 186.14.135.35
>>>>>> a=rtcp:41117 IN IP4 186.14.135.35
>>>>>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54692 typ host
>>>>>> generation 0
>>>>>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54693 typ host
>>>>>> generation 0
>>>>>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54694 typ host
>>>>>> generation 0
>>>>>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54695 typ host
>>>>>> generation 0
>>>>>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host
>>>>>> tcptype active generation 0
>>>>>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>>>>>> active generation 0
>>>>>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host
>>>>>> tcptype active generation 0
>>>>>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>>>>>> active generation 0
>>>>>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 34382 typ srflx
>>>>>> raddr 10.0.1.10 rport 54693 generation 0
>>>>>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 41117 typ srflx
>>>>>> raddr 10.0.1.10 rport 54695 generation 0
>>>>>> a=ice-ufrag:qjfFF9BX74oS8gTh
>>>>>> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
>>>>>> a=fingerprint:sha-256
>>>>>> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
>>>>>> a=setup:actpass
>>>>>> a=mid:audio
>>>>>> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
>>>>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>>>> a=recvonly
>>>>>> a=rtcp-mux
>>>>>> a=rtpmap:111 opus/48000/2
>>>>>> a=fmtp:111 minptime=10; useinbandfec=1
>>>>>> a=rtpmap:103 ISAC/16000
>>>>>> a=rtpmap:104 ISAC/32000
>>>>>> a=rtpmap:9 G722/8000
>>>>>> a=rtpmap:0 PCMU/8000
>>>>>> a=rtpmap:8 PCMA/8000
>>>>>> a=rtpmap:106 CN/32000
>>>>>> a=rtpmap:105 CN/16000
>>>>>> a=rtpmap:13 CN/8000
>>>>>> a=rtpmap:126 telephone-event/8000
>>>>>> a=maxptime:60
>>>>>> m=video 39129 RTP/SAVPF 100 116 117 96
>>>>>> c=IN IP4 186.14.135.35
>>>>>> a=rtcp:40989 IN IP4 186.14.135.35
>>>>>> a=candidate:2999745851 1 udp 2122194687 192.168.56.1 54696 typ host
>>>>>> generation 0
>>>>>> a=candidate:648569486 1 udp 2122129151 10.0.1.10 54697 typ host
>>>>>> generation 0
>>>>>> a=candidate:2999745851 2 udp 2122194686 192.168.56.1 54698 typ host
>>>>>> generation 0
>>>>>> a=candidate:648569486 2 udp 2122129150 10.0.1.10 54699 typ host
>>>>>> generation 0
>>>>>> a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 0 typ host
>>>>>> tcptype active generation 0
>>>>>> a=candidate:1747689086 1 tcp 1518149375 10.0.1.10 0 typ host tcptype
>>>>>> active generation 0
>>>>>> a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 0 typ host
>>>>>> tcptype active generation 0
>>>>>> a=candidate:1747689086 2 tcp 1518149374 10.0.1.10 0 typ host tcptype
>>>>>> active generation 0
>>>>>> a=candidate:1680186374 2 udp 1685921534 186.14.135.35 40989 typ srflx
>>>>>> raddr 10.0.1.10 rport 54699 generation 0
>>>>>> a=candidate:1680186374 1 udp 1685921535 186.14.135.35 39129 typ srflx
>>>>>> raddr 10.0.1.10 rport 54697 generation 0
>>>>>> a=ice-ufrag:qjfFF9BX74oS8gTh
>>>>>> a=ice-pwd:LCF7BftXclEb/p7tplMwidk5
>>>>>> a=fingerprint:sha-256
>>>>>> 52:E8:C6:72:FE:F2:83:67:5F:CA:92:48:91:FF:23:05:C4:9A:C4:B1:12:2B:3E:68:2F:CA:18:14:4E:C5:23:63
>>>>>> a=setup:actpass
>>>>>> a=mid:video
>>>>>> a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
>>>>>> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
>>>>>> a=extmap:4 urn:3gpp:video-orientation
>>>>>> a=sendrecv
>>>>>> a=rtcp-mux
>>>>>> a=rtpmap:100 VP8/90000
>>>>>> a=rtcp-fb:100 ccm fir
>>>>>> a=rtcp-fb:100 nack
>>>>>> a=rtcp-fb:100 nack pli
>>>>>> a=rtcp-fb:100 goog-remb
>>>>>> a=rtpmap:116 red/90000
>>>>>> a=rtpmap:117 ulpfec/90000
>>>>>> a=rtpmap:96 rtx/90000
>>>>>> a=fmtp:96 apt=100
>>>>>> a=ssrc-group:FID 3962025390 1186212002
>>>>>> a=ssrc:3962025390 cname:Rvg58eAlJAmWf/sm
>>>>>> a=ssrc:3962025390 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>>>> a45b2502-eb86-452c-b4cb-e69e5305938a
>>>>>> a=ssrc:3962025390 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>>>> a=ssrc:3962025390 label:a45b2502-eb86-452c-b4cb-e69e5305938a
>>>>>> a=ssrc:1186212002 cname:Rvg58eAlJAmWf/sm
>>>>>> a=ssrc:1186212002 msid:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>>>> a45b2502-eb86-452c-b4cb-e69e5305938a
>>>>>> a=ssrc:1186212002 mslabel:dPJVmJy18gxuWO9xmZrqIsVKQwSsNvUDrcu1
>>>>>> a=ssrc:1186212002 label:a45b2502-eb86-452c-b4cb-e69e5305938a
>>>>>>
>>>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from new to
>>>>>> requesting
>>>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from
>>>>>> requesting to trying
>>>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from trying
>>>>>> to hangup
>>>>>> Dialog 1da7651a-65e6-ecbb-f5b6-1fffb5245534: state change from hangup
>>>>>> to destroy
>>>>>> ?
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>>
>>>>
>>>> V?ctor E. Medina M.
>>>> Platform Architect / Chief Infrastructure
>>>> +58424 291 4561
>>>> BB #79A8AFA2
>>>> @VMCibersys
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>>
>>
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561
>> BB #79A8AFA2
>> @VMCibersys
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +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
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From carlosj.gf at gmail.com  Wed May 27 01:52:54 2015
From: carlosj.gf at gmail.com (=?ISO-8859-1?Q?Carlos_Gonz=E1lez_Florido?=)
Date: Tue, 26 May 2015 23:52:54 +0200
Subject: [Freeswitch-users] Video always muted in conference (fs_video2)
In-Reply-To: <CACXsapNBG_uGDKnuQ=3J2MzzZ00uquOQDeqwt6KiAcR_yRM=ag@mail.gmail.com>
References: <CAFKjMAYjKz32rE-G_aA639J5b27eNwnRAME31La25W9KiWbagA@mail.gmail.com>
	<7B131C27-7233-4E88-BB9E-E01BE588DE82@jerris.com>
	<CAFKjMAZ30sxNtAXU1iqksuNwM53gzQaqFQor7vhVHhg-3VBn5w@mail.gmail.com>
	<CAEwTsAgsVXvE5Ym9TWvbSvd9i7gCgNWwmp-2adxYKug0RHHKBA@mail.gmail.com>
	<CACXsapNJ_ErsOdZzc=_f4DJpzYe6NRGeNB8uRE1Uw1BSAaGOJg@mail.gmail.com>
	<C6A42EE2-E59D-4F1A-8AB6-B3C61962D41A@jerris.com>
	<CACXsapNBG_uGDKnuQ=3J2MzzZ00uquOQDeqwt6KiAcR_yRM=ag@mail.gmail.com>
Message-ID: <CAFKjMAYGxxMqtRs7bGq79kq+ehGnvXp8fnqjo-yZwmFOc6bVAw@mail.gmail.com>

I can confirm that the bug remains, but it seems to be less frequent.


On Tue, May 26, 2015 at 9:01 PM, Victor Medina <victor.medina at cibersys.com>
wrote:

> did it but not added the debug logs... wait...
>
> 2015-05-26 11:38 GMT-04:30 Michael Jerris <mike at jerris.com>:
>
> Please file a jira with details on how to reproduce and debug logs.
>>
>> On May 26, 2015, at 11:51 AM, Victor Medina <victor.medina at cibersys.com>
>> wrote:
>>
>> Michael...
>>
>> I just build FS with the latest sources and the Video Unmuted/Video Muted
>> ramains.
>> Just to confirm this.
>>
>> If you guys need me to test anything I have a full blown server just for
>> me and fs tests.
>>
>> 2015-05-25 0:08 GMT-04:30 Michael Jerris <mike at jerris.com>:
>>
>>> Some fixes were pushed for this on Friday.  Try out latest code and see
>>> if that works any better for you?
>>>
>>>
>>> On Saturday, May 23, 2015, Carlos Gonz?lez Florido <carlosj.gf at gmail.com>
>>> wrote:
>>>
>>>> Ok, tell me if I can help with the testing.
>>>> Is there a way to disconnect ipv6?
>>>>
>>>> On Fri, May 22, 2015 at 6:04 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>>
>>>>> I don't think its actually muted.  We are chasing down an issue that
>>>>> sounds just like this.  It might be related to ipv6 but we are still
>>>>> looking in to it.
>>>>>
>>>>> > On May 22, 2015, at 8:40 AM, Carlos Gonz?lez Florido <
>>>>> carlosj.gf at gmail.com> wrote:
>>>>> >
>>>>> > Hello,
>>>>> >
>>>>> > I'm testing the impressive fs_video2 branch, but I have the
>>>>> following problems:
>>>>> >
>>>>> > - some participants have their video always muted (and their camera
>>>>> is on and working for Hangouts, for example).
>>>>> > - other participants start the same (video muted), but after 3-5
>>>>> minutes the video is automatically unmuted.
>>>>> > - if we do a layout change (using fs_cli), all participants (having
>>>>> video muted or unmuted at that moment) go to muted video for the rest of
>>>>> the conference.
>>>>> >
>>>>> > What is the reason for fs to automatically mute the video? Is this
>>>>> expected or does it look like a bug? Can this behaviour be tuned or
>>>>> disconnected?
>>>>> >
>>>>> > Thank you,
>>>>> > Carlos Gonzalez
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>>
>>
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561
>> BB #79A8AFA2
>> @VMCibersys
>>
>>  _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
>
>
> V?ctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 27 04:00:14 2015
From: mike at jerris.com (Michael Jerris)
Date: Tue, 26 May 2015 20:00:14 -0400
Subject: [Freeswitch-users] Video always muted in conference (fs_video2)
In-Reply-To: <CAFKjMAYGxxMqtRs7bGq79kq+ehGnvXp8fnqjo-yZwmFOc6bVAw@mail.gmail.com>
References: <CAFKjMAYjKz32rE-G_aA639J5b27eNwnRAME31La25W9KiWbagA@mail.gmail.com>
	<7B131C27-7233-4E88-BB9E-E01BE588DE82@jerris.com>
	<CAFKjMAZ30sxNtAXU1iqksuNwM53gzQaqFQor7vhVHhg-3VBn5w@mail.gmail.com>
	<CAEwTsAgsVXvE5Ym9TWvbSvd9i7gCgNWwmp-2adxYKug0RHHKBA@mail.gmail.com>
	<CACXsapNJ_ErsOdZzc=_f4DJpzYe6NRGeNB8uRE1Uw1BSAaGOJg@mail.gmail.com>
	<C6A42EE2-E59D-4F1A-8AB6-B3C61962D41A@jerris.com>
	<CACXsapNBG_uGDKnuQ=3J2MzzZ00uquOQDeqwt6KiAcR_yRM=ag@mail.gmail.com>
	<CAFKjMAYGxxMqtRs7bGq79kq+ehGnvXp8fnqjo-yZwmFOc6bVAw@mail.gmail.com>
Message-ID: <CAEwTsAiuswQmGZ70yTe3bMbX5L=oEeL3yGi5xy8be5EpZN+HBQ@mail.gmail.com>

To clarify, when you are saying video is muted, do you actually see the
text "video muted" on the screen?

On Tuesday, May 26, 2015, Carlos Gonz?lez Florido <carlosj.gf at gmail.com>
wrote:

> I can confirm that the bug remains, but it seems to be less frequent.
>
>
> On Tue, May 26, 2015 at 9:01 PM, Victor Medina <victor.medina at cibersys.com
> <javascript:_e(%7B%7D,'cvml','victor.medina at cibersys.com');>> wrote:
>
>> did it but not added the debug logs... wait...
>>
>> 2015-05-26 11:38 GMT-04:30 Michael Jerris <mike at jerris.com
>> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>>:
>>
>> Please file a jira with details on how to reproduce and debug logs.
>>>
>>> On May 26, 2015, at 11:51 AM, Victor Medina <victor.medina at cibersys.com
>>> <javascript:_e(%7B%7D,'cvml','victor.medina at cibersys.com');>> wrote:
>>>
>>> Michael...
>>>
>>> I just build FS with the latest sources and the Video Unmuted/Video
>>> Muted ramains.
>>> Just to confirm this.
>>>
>>> If you guys need me to test anything I have a full blown server just for
>>> me and fs tests.
>>>
>>> 2015-05-25 0:08 GMT-04:30 Michael Jerris <mike at jerris.com
>>> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>>:
>>>
>>>> Some fixes were pushed for this on Friday.  Try out latest code and see
>>>> if that works any better for you?
>>>>
>>>>
>>>> On Saturday, May 23, 2015, Carlos Gonz?lez Florido <
>>>> carlosj.gf at gmail.com
>>>> <javascript:_e(%7B%7D,'cvml','carlosj.gf at gmail.com');>> wrote:
>>>>
>>>>> Ok, tell me if I can help with the testing.
>>>>> Is there a way to disconnect ipv6?
>>>>>
>>>>> On Fri, May 22, 2015 at 6:04 PM, Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>>> I don't think its actually muted.  We are chasing down an issue that
>>>>>> sounds just like this.  It might be related to ipv6 but we are still
>>>>>> looking in to it.
>>>>>>
>>>>>> > On May 22, 2015, at 8:40 AM, Carlos Gonz?lez Florido <
>>>>>> carlosj.gf at gmail.com> wrote:
>>>>>> >
>>>>>> > Hello,
>>>>>> >
>>>>>> > I'm testing the impressive fs_video2 branch, but I have the
>>>>>> following problems:
>>>>>> >
>>>>>> > - some participants have their video always muted (and their camera
>>>>>> is on and working for Hangouts, for example).
>>>>>> > - other participants start the same (video muted), but after 3-5
>>>>>> minutes the video is automatically unmuted.
>>>>>> > - if we do a layout change (using fs_cli), all participants (having
>>>>>> video muted or unmuted at that moment) go to muted video for the rest of
>>>>>> the conference.
>>>>>> >
>>>>>> > What is the reason for fs to automatically mute the video? Is this
>>>>>> expected or does it look like a bug? Can this behaviour be tuned or
>>>>>> disconnected?
>>>>>> >
>>>>>> > Thank you,
>>>>>> > Carlos Gonzalez
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>> V?ctor E. Medina M.
>>> Platform Architect / Chief Infrastructure
>>> +58424 291 4561
>>> BB #79A8AFA2
>>> @VMCibersys
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>>
>>
>> V?ctor E. Medina M.
>> Platform Architect / Chief Infrastructure
>> +58424 291 4561
>> BB #79A8AFA2
>> @VMCibersys
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From cdgraff at gmail.com  Wed May 27 05:03:38 2015
From: cdgraff at gmail.com (Alejandro)
Date: Tue, 26 May 2015 22:03:38 -0300
Subject: [Freeswitch-users] MONGODB ODBC Example?
Message-ID: <CALJtzxAOYqk4t1bOf5QF0jV7pLJ6vgO-=q6P==sZatq8JcVhHg@mail.gmail.com>

Hi guys,

I'm using voicemail with Mysql, anyone is using this but with MongoDB?

Some example to share about how to connect and use?

Thanks
Alejandro
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From ashwinrath at gmail.com  Wed May 27 08:22:29 2015
From: ashwinrath at gmail.com (Ashwin Rath)
Date: Wed, 27 May 2015 09:52:29 +0530
Subject: [Freeswitch-users] MONGODB ODBC Example?
In-Reply-To: <CALJtzxAOYqk4t1bOf5QF0jV7pLJ6vgO-=q6P==sZatq8JcVhHg@mail.gmail.com>
References: <CALJtzxAOYqk4t1bOf5QF0jV7pLJ6vgO-=q6P==sZatq8JcVhHg@mail.gmail.com>
Message-ID: <CAA_Nwu69KMho26P1L_dyKJLm_txwR_NroiYYZANxuAdT7phKRA@mail.gmail.com>

On a separate note, i tried using the ODBC for Cassandra with FS for
storing the core in Cassandra but it seemed to have lots of glitches.

Looking forward to tests results with Mongodb

On Wed, May 27, 2015 at 6:33 AM, Alejandro <cdgraff at gmail.com> wrote:

> Hi guys,
>
> I'm using voicemail with Mysql, anyone is using this but with MongoDB?
>
> Some example to share about how to connect and use?
>
> Thanks
> Alejandro
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ashwin Kumar Rath
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From abel at metrogsm.net  Wed May 27 08:12:24 2015
From: abel at metrogsm.net (Abel)
Date: Wed, 27 May 2015 00:12:24 -0400
Subject: [Freeswitch-users] Interface Listining Error
Message-ID: <016801d09833$5a156510$0e402f30$@metrogsm.net>

Hello,

 

 

I want from Freeswitch to be listening SIP registrations on two diferent ports: 5060, and 443, but the freeswitch log?s says there is an error:

 

2011-05-05 02:54:25.563289 [DEBUG] switch_nat.c:500 mapped public port 5060 protocol UDP to localport 5060

2011-05-05 02:54:25.563289 [DEBUG] sofia.c:1432 Created UDP nat mapping for sipinterface_1 port 5060

2011-05-05 02:54:25.848007 [DEBUG] switch_nat.c:500 mapped public port 443 protocol TCP to localport 443

2011-05-05 02:54:25.850024 [DEBUG] sofia.c:1435 Created TCP nat mapping for sipinterface_2 port 443

2011-05-05 02:54:25.852043 [ERR] sofia.c:1481 Error Creating SIP UA for profile: sipinterface_2

 

 

I make sure that that port is not being used for another services in the system:

 

freeswitch]# nestat -na | more

-bash: nestat: command not found

[root at bluebox freeswitch]# netstat -na | more

Active Internet connections (servers and established)

Proto Recv-Q Send-Q Local Address               Foreign Address             State

tcp        0      0 192.168.0.106:5060          0.0.0.0:*                   LISTEN

tcp        0      0 192.168.0.106:4453          0.0.0.0:*                   LISTEN

tcp        0      0 0.0.0.0:3306                0.0.0.0:*                   LISTEN

tcp        0      0 0.0.0.0:111                 0.0.0.0:*                   LISTEN

tcp        0      0 0.0.0.0:976                 0.0.0.0:*                   LISTEN

tcp        0      0 127.0.0.1:8021              0.0.0.0:*                   LISTEN

tcp        0      0 127.0.0.1:631               0.0.0.0:*                   LISTEN

tcp        0      0 127.0.0.1:25                0.0.0.0:*                   LISTEN

tcp        0      0 :::80                       :::*                        LISTEN

tcp        0      0 :::22                       :::*                        LISTEN

tcp        0      0 ::ffff:192.168.0.106:22     ::ffff:192.168.0.101:64845  ESTABLISHED

tcp        0     52 ::ffff:192.168.0.106:22     ::ffff:192.168.0.101:49783  ESTABLISHED

udp        0      0 192.168.0.106:49035         192.168.0.1:5351            ESTABLISHED

udp        0      0 192.168.0.106:5060          0.0.0.0:*

udp        0      0 0.0.0.0:68                  0.0.0.0:*

udp        0      0 0.0.0.0:970                 0.0.0.0:*

udp        0      0 0.0.0.0:973                 0.0.0.0:*

udp        0      0 192.168.0.106:4453          0.0.0.0:*

udp        0      0 0.0.0.0:39144               0.0.0.0:*

udp        0      0 0.0.0.0:5353                0.0.0.0:*

udp      864      0 239.255.255.250:1900        0.0.0.0:*

udp        0      0 0.0.0.0:111                 0.0.0.0:*

udp        0      0 0.0.0.0:631                 0.0.0.0:*

udp        0      0 :::44004                    :::*

udp        0      0 :::5353                     :::*

 

The freeswitch inself have this limitation with this port 443?

 

Thank You,



---
This email has been checked for viruses by Avast antivirus software.
http://www.avast.com
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From carlosj.gf at gmail.com  Wed May 27 09:42:31 2015
From: carlosj.gf at gmail.com (=?ISO-8859-1?Q?Carlos_Gonz=E1lez_Florido?=)
Date: Wed, 27 May 2015 07:42:31 +0200
Subject: [Freeswitch-users] Video always muted in conference (fs_video2)
In-Reply-To: <CAEwTsAiuswQmGZ70yTe3bMbX5L=oEeL3yGi5xy8be5EpZN+HBQ@mail.gmail.com>
References: <CAFKjMAYjKz32rE-G_aA639J5b27eNwnRAME31La25W9KiWbagA@mail.gmail.com>
	<7B131C27-7233-4E88-BB9E-E01BE588DE82@jerris.com>
	<CAFKjMAZ30sxNtAXU1iqksuNwM53gzQaqFQor7vhVHhg-3VBn5w@mail.gmail.com>
	<CAEwTsAgsVXvE5Ym9TWvbSvd9i7gCgNWwmp-2adxYKug0RHHKBA@mail.gmail.com>
	<CACXsapNJ_ErsOdZzc=_f4DJpzYe6NRGeNB8uRE1Uw1BSAaGOJg@mail.gmail.com>
	<C6A42EE2-E59D-4F1A-8AB6-B3C61962D41A@jerris.com>
	<CACXsapNBG_uGDKnuQ=3J2MzzZ00uquOQDeqwt6KiAcR_yRM=ag@mail.gmail.com>
	<CAFKjMAYGxxMqtRs7bGq79kq+ehGnvXp8fnqjo-yZwmFOc6bVAw@mail.gmail.com>
	<CAEwTsAiuswQmGZ70yTe3bMbX5L=oEeL3yGi5xy8be5EpZN+HBQ@mail.gmail.com>
Message-ID: <CAFKjMAatUS1yC9Lwir34LiHV4f5QLR1DsDNnZxOv-u9C5di1TA@mail.gmail.com>

Yes. We have several different (related) situations:
- Some clients start working (no video muted screen) but the "video muted"
screen appears randomly.
- Other clients always start with the video muted screen, and after a while
(maybe 1 or 10 minutes) it disappears and go on normally,
- Other clients start and stay in this situation (video muted screen) for
ever.
- In any case, after several video layout changes, all participants change
to the video muted screen and you must shutdown the conference to recover.

We have tested with different versions of Chrome on OS X and Linux, and we
couldn't find a clear pattern. A single participant tends to have the same
or similar behavior over time, but not always.


On Wed, May 27, 2015 at 2:00 AM, Michael Jerris <mike at jerris.com> wrote:

> To clarify, when you are saying video is muted, do you actually see the
> text "video muted" on the screen?
>
>
> On Tuesday, May 26, 2015, Carlos Gonz?lez Florido <carlosj.gf at gmail.com>
> wrote:
>
>> I can confirm that the bug remains, but it seems to be less frequent.
>>
>>
>> On Tue, May 26, 2015 at 9:01 PM, Victor Medina <
>> victor.medina at cibersys.com> wrote:
>>
>>> did it but not added the debug logs... wait...
>>>
>>> 2015-05-26 11:38 GMT-04:30 Michael Jerris <mike at jerris.com>:
>>>
>>> Please file a jira with details on how to reproduce and debug logs.
>>>>
>>>> On May 26, 2015, at 11:51 AM, Victor Medina <victor.medina at cibersys.com>
>>>> wrote:
>>>>
>>>> Michael...
>>>>
>>>> I just build FS with the latest sources and the Video Unmuted/Video
>>>> Muted ramains.
>>>> Just to confirm this.
>>>>
>>>> If you guys need me to test anything I have a full blown server just
>>>> for me and fs tests.
>>>>
>>>> 2015-05-25 0:08 GMT-04:30 Michael Jerris <mike at jerris.com>:
>>>>
>>>>> Some fixes were pushed for this on Friday.  Try out latest code and
>>>>> see if that works any better for you?
>>>>>
>>>>>
>>>>> On Saturday, May 23, 2015, Carlos Gonz?lez Florido <
>>>>> carlosj.gf at gmail.com> wrote:
>>>>>
>>>>>> Ok, tell me if I can help with the testing.
>>>>>> Is there a way to disconnect ipv6?
>>>>>>
>>>>>> On Fri, May 22, 2015 at 6:04 PM, Michael Jerris <mike at jerris.com>
>>>>>> wrote:
>>>>>>
>>>>>>> I don't think its actually muted.  We are chasing down an issue that
>>>>>>> sounds just like this.  It might be related to ipv6 but we are still
>>>>>>> looking in to it.
>>>>>>>
>>>>>>> > On May 22, 2015, at 8:40 AM, Carlos Gonz?lez Florido <
>>>>>>> carlosj.gf at gmail.com> wrote:
>>>>>>> >
>>>>>>> > Hello,
>>>>>>> >
>>>>>>> > I'm testing the impressive fs_video2 branch, but I have the
>>>>>>> following problems:
>>>>>>> >
>>>>>>> > - some participants have their video always muted (and their
>>>>>>> camera is on and working for Hangouts, for example).
>>>>>>> > - other participants start the same (video muted), but after 3-5
>>>>>>> minutes the video is automatically unmuted.
>>>>>>> > - if we do a layout change (using fs_cli), all participants
>>>>>>> (having video muted or unmuted at that moment) go to muted video for the
>>>>>>> rest of the conference.
>>>>>>> >
>>>>>>> > What is the reason for fs to automatically mute the video? Is this
>>>>>>> expected or does it look like a bug? Can this behaviour be tuned or
>>>>>>> disconnected?
>>>>>>> >
>>>>>>> > Thank you,
>>>>>>> > Carlos Gonzalez
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>>
>>>>
>>>> V?ctor E. Medina M.
>>>> Platform Architect / Chief Infrastructure
>>>> +58424 291 4561
>>>> BB #79A8AFA2
>>>> @VMCibersys
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>>
>>>
>>> V?ctor E. Medina M.
>>> Platform Architect / Chief Infrastructure
>>> +58424 291 4561
>>> BB #79A8AFA2
>>> @VMCibersys
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 27 11:03:58 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 27 May 2015 09:03:58 +0200
Subject: [Freeswitch-users] Interface Listining Error
In-Reply-To: <016801d09833$5a156510$0e402f30$@metrogsm.net>
References: <016801d09833$5a156510$0e402f30$@metrogsm.net>
Message-ID: <CACOYK=gXwFZsqZ0XxW27y8t-zpfsQBRvnGTQTCdzJDGTEDdx9Q@mail.gmail.com>

Maybe it's running as an unprivileged user? Only root can open sockets
listening on low ports.
On May 27, 2015 6:23 AM, "Abel" <abel at metrogsm.net> wrote:

> Hello,
>
>
>
>
>
> I want from Freeswitch to be listening SIP registrations on two diferent
> ports: 5060, and 443, but the freeswitch log?s says there is an error:
>
>
>
> 2011-05-05 02:54:25.563289 [DEBUG] switch_nat.c:500 mapped public port
> 5060 protocol UDP to localport 5060
>
> 2011-05-05 02:54:25.563289 [DEBUG] sofia.c:1432 Created UDP nat mapping
> for sipinterface_1 port 5060
>
> 2011-05-05 02:54:25.848007 [DEBUG] switch_nat.c:500 mapped public port 443
> protocol TCP to localport 443
>
> 2011-05-05 02:54:25.850024 [DEBUG] sofia.c:1435 Created TCP nat mapping
> for sipinterface_2 port 443
>
> 2011-05-05 02:54:25.852043 [ERR] sofia.c:1481 Error Creating SIP UA for
> profile: sipinterface_2
>
>
>
>
>
> I make sure that that port is not being used for another services in the
> system:
>
>
>
> freeswitch]# nestat -na | more
>
> -bash: nestat: command not found
>
> [root at bluebox freeswitch]# netstat -na | more
>
> Active Internet connections (servers and established)
>
> Proto Recv-Q Send-Q Local Address               Foreign
> Address             State
>
> tcp        0      0 192.168.0.106:5060          0.0.0.0:*
> LISTEN
>
> tcp        0      0 192.168.0.106:4453          0.0.0.0:*
> LISTEN
>
> tcp        0      0 0.0.0.0:3306                0.0.0.0:*
> LISTEN
>
> tcp        0      0 0.0.0.0:111                 0.0.0.0:*
> LISTEN
>
> tcp        0      0 0.0.0.0:976                 0.0.0.0:*
> LISTEN
>
> tcp        0      0 127.0.0.1:8021              0.0.0.0:*
> LISTEN
>
> tcp        0      0 127.0.0.1:631               0.0.0.0:*
> LISTEN
>
> tcp        0      0 127.0.0.1:25                0.0.0.0:*
> LISTEN
>
> tcp        0      0 :::80
> :::*                        LISTEN
>
> tcp        0      0 :::22
> :::*                        LISTEN
>
> tcp        0      0 ::ffff:192.168.0.106:22     ::ffff:192.168.0.101:64845
> ESTABLISHED
>
> tcp        0     52 ::ffff:192.168.0.106:22     ::ffff:192.168.0.101:49783
> ESTABLISHED
>
> udp        0      0 192.168.0.106:49035         192.168.0.1:5351
> ESTABLISHED
>
> udp        0      0 192.168.0.106:5060          0.0.0.0:*
>
> udp        0      0 0.0.0.0:68                  0.0.0.0:*
>
> udp        0      0 0.0.0.0:970                 0.0.0.0:*
>
> udp        0      0 0.0.0.0:973                 0.0.0.0:*
>
> udp        0      0 192.168.0.106:4453          0.0.0.0:*
>
> udp        0      0 0.0.0.0:39144               0.0.0.0:*
>
> udp        0      0 0.0.0.0:5353                0.0.0.0:*
>
> udp      864      0 239.255.255.250:1900        0.0.0.0:*
>
> udp        0      0 0.0.0.0:111                 0.0.0.0:*
>
> udp        0      0 0.0.0.0:631                 0.0.0.0:*
>
> udp        0      0 :::44004                    :::*
>
> udp        0      0 :::5353                     :::*
>
>
>
> The freeswitch inself have this limitation with this port 443?
>
>
>
> Thank You,
>
>
> ------------------------------
>   [image: Avast logo] <http://www.avast.com/>
>
> This email has been checked for viruses by Avast antivirus software.
> www.avast.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 s.safarov at gmail.com  Wed May 27 11:18:39 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 27 May 2015 10:18:39 +0300
Subject: [Freeswitch-users] Interface Listining Error
In-Reply-To: <CACOYK=gXwFZsqZ0XxW27y8t-zpfsQBRvnGTQTCdzJDGTEDdx9Q@mail.gmail.com>
References: <016801d09833$5a156510$0e402f30$@metrogsm.net>
	<CACOYK=gXwFZsqZ0XxW27y8t-zpfsQBRvnGTQTCdzJDGTEDdx9Q@mail.gmail.com>
Message-ID: <CAHtxdDdnkwpu1JQ5X-qG8Py0Fg_8gdjbpC4=WU8_wOf9zJF-7Q@mail.gmail.com>

You can try install nginx as reverse proxy or configure port forwarding on
iptables.
Runnign application as root is not recomended.

On Wed, May 27, 2015 at 10:03 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> Maybe it's running as an unprivileged user? Only root can open sockets
> listening on low ports.
> On May 27, 2015 6:23 AM, "Abel" <abel at metrogsm.net> wrote:
>
>> Hello,
>>
>>
>>
>>
>>
>> I want from Freeswitch to be listening SIP registrations on two diferent
>> ports: 5060, and 443, but the freeswitch log?s says there is an error:
>>
>>
>>
>> 2011-05-05 02:54:25.563289 [DEBUG] switch_nat.c:500 mapped public port
>> 5060 protocol UDP to localport 5060
>>
>> 2011-05-05 02:54:25.563289 [DEBUG] sofia.c:1432 Created UDP nat mapping
>> for sipinterface_1 port 5060
>>
>> 2011-05-05 02:54:25.848007 [DEBUG] switch_nat.c:500 mapped public port
>> 443 protocol TCP to localport 443
>>
>> 2011-05-05 02:54:25.850024 [DEBUG] sofia.c:1435 Created TCP nat mapping
>> for sipinterface_2 port 443
>>
>> 2011-05-05 02:54:25.852043 [ERR] sofia.c:1481 Error Creating SIP UA for
>> profile: sipinterface_2
>>
>>
>>
>>
>>
>> I make sure that that port is not being used for another services in the
>> system:
>>
>>
>>
>> freeswitch]# nestat -na | more
>>
>> -bash: nestat: command not found
>>
>> [root at bluebox freeswitch]# netstat -na | more
>>
>> Active Internet connections (servers and established)
>>
>> Proto Recv-Q Send-Q Local Address               Foreign
>> Address             State
>>
>> tcp        0      0 192.168.0.106:5060          0.0.0.0:*
>> LISTEN
>>
>> tcp        0      0 192.168.0.106:4453          0.0.0.0:*
>> LISTEN
>>
>> tcp        0      0 0.0.0.0:3306                0.0.0.0:*
>> LISTEN
>>
>> tcp        0      0 0.0.0.0:111                 0.0.0.0:*
>> LISTEN
>>
>> tcp        0      0 0.0.0.0:976                 0.0.0.0:*
>> LISTEN
>>
>> tcp        0      0 127.0.0.1:8021              0.0.0.0:*
>> LISTEN
>>
>> tcp        0      0 127.0.0.1:631               0.0.0.0:*
>> LISTEN
>>
>> tcp        0      0 127.0.0.1:25                0.0.0.0:*
>> LISTEN
>>
>> tcp        0      0 :::80
>> :::*                        LISTEN
>>
>> tcp        0      0 :::22
>> :::*                        LISTEN
>>
>> tcp        0      0 ::ffff:192.168.0.106:22     ::ffff:
>> 192.168.0.101:64845  ESTABLISHED
>>
>> tcp        0     52 ::ffff:192.168.0.106:22     ::ffff:
>> 192.168.0.101:49783  ESTABLISHED
>>
>> udp        0      0 192.168.0.106:49035         192.168.0.1:5351
>> ESTABLISHED
>>
>> udp        0      0 192.168.0.106:5060          0.0.0.0:*
>>
>> udp        0      0 0.0.0.0:68                  0.0.0.0:*
>>
>> udp        0      0 0.0.0.0:970                 0.0.0.0:*
>>
>> udp        0      0 0.0.0.0:973                 0.0.0.0:*
>>
>> udp        0      0 192.168.0.106:4453          0.0.0.0:*
>>
>> udp        0      0 0.0.0.0:39144               0.0.0.0:*
>>
>> udp        0      0 0.0.0.0:5353                0.0.0.0:*
>>
>> udp      864      0 239.255.255.250:1900        0.0.0.0:*
>>
>> udp        0      0 0.0.0.0:111                 0.0.0.0:*
>>
>> udp        0      0 0.0.0.0:631                 0.0.0.0:*
>>
>> udp        0      0 :::44004                    :::*
>>
>> udp        0      0 :::5353                     :::*
>>
>>
>>
>> The freeswitch inself have this limitation with this port 443?
>>
>>
>>
>> Thank You,
>>
>>
>> ------------------------------
>>   [image: Avast logo] <http://www.avast.com/>
>>
>> This email has been checked for viruses by Avast antivirus software.
>> www.avast.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 abel at metrogsm.net  Wed May 27 11:47:54 2015
From: abel at metrogsm.net (Abel)
Date: Wed, 27 May 2015 03:47:54 -0400
Subject: [Freeswitch-users] Interface Listining Error
In-Reply-To: <CAHtxdDdnkwpu1JQ5X-qG8Py0Fg_8gdjbpC4=WU8_wOf9zJF-7Q@mail.gmail.com>
References: <016801d09833$5a156510$0e402f30$@metrogsm.net>	<CACOYK=gXwFZsqZ0XxW27y8t-zpfsQBRvnGTQTCdzJDGTEDdx9Q@mail.gmail.com>
	<CAHtxdDdnkwpu1JQ5X-qG8Py0Fg_8gdjbpC4=WU8_wOf9zJF-7Q@mail.gmail.com>
Message-ID: <018b01d09851$7256abc0$57040340$@metrogsm.net>

Yes, all of you is right? Because Freeswitch is running on unprivileged user the system won?t allow it to be listening on that port, I made the test and is worked, but the best option is to configure port forwarding on the firewall since is not recommend.

 

 

Thank You again.

 

De: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Sergey Safarov
Enviado el: Wednesday, May 27, 2015 3:19 AM
Para: FreeSWITCH Users Help
Asunto: Re: [Freeswitch-users] Interface Listining Error

 

You can try install nginx as reverse proxy or configure port forwarding on iptables.

Runnign application as root is not recomended.

 

On Wed, May 27, 2015 at 10:03 AM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

Maybe it's running as an unprivileged user? Only root can open sockets listening on low ports.

On May 27, 2015 6:23 AM, "Abel" <abel at metrogsm.net> wrote:

Hello,

 

 

I want from Freeswitch to be listening SIP registrations on two diferent ports: 5060, and 443, but the freeswitch log?s says there is an error:

 

2011-05-05 02:54:25.563289 [DEBUG] switch_nat.c:500 mapped public port 5060 protocol UDP to localport 5060

2011-05-05 02:54:25.563289 [DEBUG] sofia.c:1432 Created UDP nat mapping for sipinterface_1 port 5060

2011-05-05 02:54:25.848007 [DEBUG] switch_nat.c:500 mapped public port 443 protocol TCP to localport 443

2011-05-05 02:54:25.850024 [DEBUG] sofia.c:1435 Created TCP nat mapping for sipinterface_2 port 443

2011-05-05 02:54:25.852043 [ERR] sofia.c:1481 Error Creating SIP UA for profile: sipinterface_2

 

 

I make sure that that port is not being used for another services in the system:

 

freeswitch]# nestat -na | more

-bash: nestat: command not found

[root at bluebox freeswitch]# netstat -na | more

Active Internet connections (servers and established)

Proto Recv-Q Send-Q Local Address               Foreign Address             State

tcp        0      0 192.168.0.106:5060          0.0.0.0:*                   LISTEN

tcp        0      0 192.168.0.106:4453          0.0.0.0:*                   LISTEN

tcp        0      0 0.0.0.0:3306                0.0.0.0:*                   LISTEN

tcp        0      0 0.0.0.0:111                 0.0.0.0:*                   LISTEN

tcp        0      0 0.0.0.0:976                 0.0.0.0:*                   LISTEN

tcp        0      0 127.0.0.1:8021              0.0.0.0:*                   LISTEN

tcp        0      0 127.0.0.1:631               0.0.0.0:*                   LISTEN

tcp        0      0 127.0.0.1:25                0.0.0.0:*                   LISTEN

tcp        0      0 :::80                       :::*                        LISTEN

tcp        0      0 :::22                       :::*                        LISTEN

tcp        0      0 ::ffff:192.168.0.106:22     ::ffff:192.168.0.101:64845  ESTABLISHED

tcp        0     52 ::ffff:192.168.0.106:22     ::ffff:192.168.0.101:49783  ESTABLISHED

udp        0      0 192.168.0.106:49035         192.168.0.1:5351            ESTABLISHED

udp        0      0 192.168.0.106:5060          0.0.0.0:*

udp        0      0 0.0.0.0:68                  0.0.0.0:*

udp        0      0 0.0.0.0:970                 0.0.0.0:*

udp        0      0 0.0.0.0:973                 0.0.0.0:*

udp        0      0 192.168.0.106:4453          0.0.0.0:*

udp        0      0 0.0.0.0:39144               0.0.0.0:*

udp        0      0 0.0.0.0:5353                0.0.0.0:*

udp      864      0 239.255.255.250:1900        0.0.0.0:*

udp        0      0 0.0.0.0:111                 0.0.0.0:*

udp        0      0 0.0.0.0:631                 0.0.0.0:*

udp        0      0 :::44004                    :::*

udp        0      0 :::5353                     :::*

 

The freeswitch inself have this limitation with this port 443?

 

Thank You,

 

  _____  


 <http://www.avast.com/> Avast logo

This email has been checked for viruses by Avast antivirus software. 
www.avast.com <http://www.avast.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 john.nash778 at gmail.com  Wed May 27 14:10:18 2015
From: john.nash778 at gmail.com (John Nash)
Date: Wed, 27 May 2015 15:40:18 +0530
Subject: [Freeswitch-users] Playing multiple files in lua script
In-Reply-To: <901298B3-9395-4AE4-A654-51CBD3D1CAFD@mgtech.com>
References: <CAA4rGPz=PveNCvUpbGfsHy8gargvnEdff0_k+tSF+66UZnsZJQ@mail.gmail.com>
	<CAFmZFiN8rynkr+F4F2dvAO3GCqHV+ErV_XEYHiZdCTEWtSJ+zw@mail.gmail.com>
	<CAA4rGPzV1oNbmT=cEQW==VN4wUQOaM7nGQu5LNn1udRqz9DwRg@mail.gmail.com>
	<CAFmZFiO1SBsjBfyGFk-rhNoMHY6vGhD28SyRKoE=r_v8E0S3Ww@mail.gmail.com>
	<901298B3-9395-4AE4-A654-51CBD3D1CAFD@mgtech.com>
Message-ID: <CAA4rGPyYH6WQR_YS-q3TSG4A850YLsfoezMos2wL1Cav1ZKK4g@mail.gmail.com>

Thank you,

I also noticed the volume of sound files a bit low. As i am using native
files (preencoded with audio codec) , if I understand correctly freeswitch
will just stream the contents of file to caller without doing any kind of
processing to sample. Is there a way to increase audio volume of sound
files?

On Wed, May 20, 2015 at 12:02 AM, Mario <mario_fs at mgtech.com> wrote:

> Here is how I construct a LUA variable to play multiple files in a
> ringback, syntax would be the same for your application. This is a complex
> one since it not only plays different files but sets a time limit for each:
>
>    ringback =
> "file_string://"..em1.."!{timeout="..moh1_timeout.."}local_stream://mohmv!file_string://"..em2.."!local_stream://mohmv?
>
> Also see https://wiki.freeswitch.org/wiki/Mod_local_stream, I added the
> timeout stuff at the bottom which shows syntax for a file.
> Hope this helps a little
> Mario G
>
>
>
> On May 18, 2015, at 4:15 PM, Abaci B <abaci64 at gmail.com> wrote:
>
> no idea about timer=soft issues on centos, I switched from centos to
> debian when the core devs switch as I like to be on the platform best
> supported.
> as far as usig streamFile over playback it gives you more control if you
> need handling of DTMF (or speech recognition) and iirc using playback will
> give you more logs
>
> On Mon, May 18, 2015 at 6:13 PM, John Nash <john.nash778 at gmail.com> wrote:
>
>> Actually streamfile and playing using "!" (file_string") also had same
>> issue. But this is solved when I used rtp-timer-name = soft . before I was
>> using rtp-timer-name = none. I read in some posts that timer = soft causes
>> performance issues in centos 6. Do you have any idea about this?
>>
>> also is Session:streamfile different from session:playback in some
>> way?...which should be preferred?
>>
>>
>>
>>
>> On Tue, May 19, 2015 at 3:07 AM, Abaci B <abaci64 at gmail.com> wrote:
>>
>>> is there anything in the logs when this happens?
>>> any reason not to play the files using
>>> file-string:///path/to/file1!/path/to/file2!/path/to/file3 ? not sure
>>> if it would mke a difference but you could try.
>>> did you also try if you see the same behavior when doing
>>> session:streamFile("/path/to/file1") instead of session:execute("playback",
>>> "/path/to/file1")
>>>
>>>
>>> On Mon, May 18, 2015 at 5:22 PM, John Nash <john.nash778 at gmail.com>
>>> wrote:
>>>
>>>> I am trying to play some native files (Preencoded) in lua script and
>>>> calling the lua script in dialpan like below..
>>>>
>>>> session:execute("playback", <path to file1>);
>>>> session:execute("playback", <path to file2>);
>>>> session:execute("playback", <path to file3>);
>>>>
>>>> When I make call I hear file1 clearly but after that for approx 10-15
>>>> seconds I do not hear anything (except some occasional noise like breaking
>>>> voice)
>>>>
>>>> I tried commenting file1 (to check if file1 is corrupt or something)
>>>> but then I hear file2 clearly and not file 3 for 10-15 seconds. I also
>>>> tried session:sleep(100) between playbacks but issues remains same.
>>>>
>>>> Am I doing something wrong?
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From abaci64 at gmail.com  Wed May 27 19:33:16 2015
From: abaci64 at gmail.com (Abaci B)
Date: Wed, 27 May 2015 11:33:16 -0400
Subject: [Freeswitch-users] Playing multiple files in lua script
In-Reply-To: <CAA4rGPyYH6WQR_YS-q3TSG4A850YLsfoezMos2wL1Cav1ZKK4g@mail.gmail.com>
References: <CAA4rGPz=PveNCvUpbGfsHy8gargvnEdff0_k+tSF+66UZnsZJQ@mail.gmail.com>
	<CAFmZFiN8rynkr+F4F2dvAO3GCqHV+ErV_XEYHiZdCTEWtSJ+zw@mail.gmail.com>
	<CAA4rGPzV1oNbmT=cEQW==VN4wUQOaM7nGQu5LNn1udRqz9DwRg@mail.gmail.com>
	<CAFmZFiO1SBsjBfyGFk-rhNoMHY6vGhD28SyRKoE=r_v8E0S3Ww@mail.gmail.com>
	<901298B3-9395-4AE4-A654-51CBD3D1CAFD@mgtech.com>
	<CAA4rGPyYH6WQR_YS-q3TSG4A850YLsfoezMos2wL1Cav1ZKK4g@mail.gmail.com>
Message-ID: <CAFmZFiOADc7yQVQuFHDaODu515Yf4pg35s7QsHbiX6Jp4C8QsA@mail.gmail.com>

You can either increase the volume of the file using a tool like sox, you
can also increase the volume of the entire call using the set_audio_level
tool (see
https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_audio_level) or
you can increase the volume while playing the file in lua using DTMF
callback, or you can use uuid_fileman (see
https://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman) whichever of
those methods suits you better.

On Wed, May 27, 2015 at 6:10 AM, John Nash <john.nash778 at gmail.com> wrote:

> Thank you,
>
> I also noticed the volume of sound files a bit low. As i am using native
> files (preencoded with audio codec) , if I understand correctly freeswitch
> will just stream the contents of file to caller without doing any kind of
> processing to sample. Is there a way to increase audio volume of sound
> files?
>
> On Wed, May 20, 2015 at 12:02 AM, Mario <mario_fs at mgtech.com> wrote:
>
>> Here is how I construct a LUA variable to play multiple files in a
>> ringback, syntax would be the same for your application. This is a complex
>> one since it not only plays different files but sets a time limit for each:
>>
>>    ringback =
>> "file_string://"..em1.."!{timeout="..moh1_timeout.."}local_stream://mohmv!file_string://"..em2.."!local_stream://mohmv?
>>
>> Also see https://wiki.freeswitch.org/wiki/Mod_local_stream, I added the
>> timeout stuff at the bottom which shows syntax for a file.
>> Hope this helps a little
>> Mario G
>>
>>
>>
>> On May 18, 2015, at 4:15 PM, Abaci B <abaci64 at gmail.com> wrote:
>>
>> no idea about timer=soft issues on centos, I switched from centos to
>> debian when the core devs switch as I like to be on the platform best
>> supported.
>> as far as usig streamFile over playback it gives you more control if you
>> need handling of DTMF (or speech recognition) and iirc using playback will
>> give you more logs
>>
>> On Mon, May 18, 2015 at 6:13 PM, John Nash <john.nash778 at gmail.com>
>> wrote:
>>
>>> Actually streamfile and playing using "!" (file_string") also had same
>>> issue. But this is solved when I used rtp-timer-name = soft . before I was
>>> using rtp-timer-name = none. I read in some posts that timer = soft causes
>>> performance issues in centos 6. Do you have any idea about this?
>>>
>>> also is Session:streamfile different from session:playback in some
>>> way?...which should be preferred?
>>>
>>>
>>>
>>>
>>> On Tue, May 19, 2015 at 3:07 AM, Abaci B <abaci64 at gmail.com> wrote:
>>>
>>>> is there anything in the logs when this happens?
>>>> any reason not to play the files using
>>>> file-string:///path/to/file1!/path/to/file2!/path/to/file3 ? not sure
>>>> if it would mke a difference but you could try.
>>>> did you also try if you see the same behavior when doing
>>>> session:streamFile("/path/to/file1") instead of session:execute("playback",
>>>> "/path/to/file1")
>>>>
>>>>
>>>> On Mon, May 18, 2015 at 5:22 PM, John Nash <john.nash778 at gmail.com>
>>>> wrote:
>>>>
>>>>> I am trying to play some native files (Preencoded) in lua script and
>>>>> calling the lua script in dialpan like below..
>>>>>
>>>>> session:execute("playback", <path to file1>);
>>>>> session:execute("playback", <path to file2>);
>>>>> session:execute("playback", <path to file3>);
>>>>>
>>>>> When I make call I hear file1 clearly but after that for approx 10-15
>>>>> seconds I do not hear anything (except some occasional noise like breaking
>>>>> voice)
>>>>>
>>>>> I tried commenting file1 (to check if file1 is corrupt or something)
>>>>> but then I hear file2 clearly and not file 3 for 10-15 seconds. I also
>>>>> tried session:sleep(100) between playbacks but issues remains same.
>>>>>
>>>>> Am I doing something wrong?
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dm at dwide.com  Wed May 27 20:10:04 2015
From: dm at dwide.com (Dmitry Mordovin)
Date: Wed, 27 May 2015 20:10:04 +0400
Subject: [Freeswitch-users] Routing question
Message-ID: <5565EC5C.3080707@dwide.com>

Hi all,

For sample:
- Have two voip gateways: provider_1 and provider_2
- Need originate multiple destinations at the same time, protected 
through each gateway

So, originate would be like:

originate 
(sofia/gateway/provider_1/919246461001|sofia/gateway/provider_2/919246461001),(sofia/gateway/provider_1/919246461002|sofia/gateway/provider_2/919246461002) 
5000

Explain.

Originate calls to 919246461001 and 919246461002 parallel through 
provider_1, if provider_1 failed, try provider_2

Does any one did it?


From olegstolyar at gmail.com  Wed May 27 20:13:36 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 27 May 2015 09:13:36 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <DC7CE9BD-1271-4EBB-A7D1-4E3D0CCEBB46@jerris.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAFiqYum4if-B8NGFu7PXjFpJZnPuFGy8XhWscM7p7q6BwZiqGg@mail.gmail.com>
	<CAC8JMO+-Wqu9_xQx6TzkUU3NLhVU=HTpDu6wO2p1TX8J1kxEKg@mail.gmail.com>
	<CAC8JMO+YUezb5peCD0njovq83YwNYQMEx6ASee96RRq2-0Ocmg@mail.gmail.com>
	<DC7CE9BD-1271-4EBB-A7D1-4E3D0CCEBB46@jerris.com>
Message-ID: <CAC8JMO+Taai2bM8A6kF8hEUJ6ZJ3LfG0_qohqx+CDenwuX5jiA@mail.gmail.com>

Jira: https://freeswitch.org/jira/browse/FS-7582

It turns out that FS actually does not send a=setup at all in reINVITES.  I
am not comfortable enough with the code to add an SDP attribute.  Also this
is now a low priority for me since there is a simple workaround of
disabling session timers.

On Tue, May 26, 2015 at 9:07 AM, Michael Jerris <mike at jerris.com> wrote:

> Please file a jira for us on that.  Even better with a pull request to fix
> it.
>
> On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
> Just FYI, the JsSip engineers replied to my questions with this:
>
> The error happens because the incoming reINVITE has a=setup:active in
> the SDP which is a bug in FreeSwitch (the RFC clearly states that the
> SDP offer MUST have a=setup:actpass):
>
> On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>> Thanks Steven!
>>
>> It may be https://freeswitch.org/jira/browse/FS-7040.
>>
>> As far as the 120 sec is concerned - Below are snippets from both the
>> invite and the 200 OK from FS.  I know that FS reads the Session-Expires
>> from the client because if I change it to a value less than 120, FS sends
>> back a "SIP/2.0 422 Session Interval Too Small".  I  even thought the
>> problem could be that the Session-Exipres format in the original INVITE is
>> incorrect since it does not contain ";refresher=uac" but when I added that
>> and made the line "Session-Expires: 300;refresher=uac" nothing changed.
>>
>>
>> INVITE sip:echo-test at anonymous.invalid SIP/2.0
>> Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029
>> Max-Forwards: 69
>> To: <sip:echo-test at anonymous.invalid>
>> From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
>> Call-ID: r65e48gp171p21rkppcu
>> CSeq: 6621 INVITE
>> Contact: <sip:mqthb9dj at pfcnm9rjv5en.invalid;transport=ws;ob>
>> Content-Type: application/sdp
>> Session-Expires: 300
>> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
>> Supported: timer,ice,outbound
>> User-Agent: JsSIP 0.6.26
>> Content-Length: 2754
>>
>> SIP/2.0 200 OK
>> Via: SIP/2.0/WSS
>> pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200
>> From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
>> To: <sip:echo-test at anonymous.invalid>;tag=SgjX0X4arHUFg
>> Call-ID: r65e48gp171p21rkppcu
>> CSeq: 6621 INVITE
>> Contact: <sip:echo-test at 50.19.255.62:5080;transport=udp>
>> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>> Accept: application/sdp
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Require: timer
>> Supported: timer, path, replaces
>> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
>> line-seize, call-info, sla, include-session-description, presence.winfo,
>> message-summary, refer
>> Session-Expires: 120;refresher=uac
>> Content-Type: application/sdp
>> Content-Disposition: session
>> Content-Length: 809
>> X-Netflix: yes
>> Remote-Party-ID: "echo-test" <sip:echo-test at anonymous.invalid
>> >;party=calling;privacy=off;screen=no
>>
>>
>> On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> The values to actual use are negotiated during the call invite, and your
>>> trace shows it's using 2 minutes.
>>> Session-Expires: 120;refresher=uac
>>> Min-SE: 120
>>>
>>> Just as an idea because you didn't send both invite and re-invite...
>>> perhaps the SDP body is different in the reinvite without the version
>>> number having changed. If so it may be
>>> https://freeswitch.org/jira/browse/FS-7040
>>>
>>> On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>>
>>>> But isn't that based on the session-timeout param which defaults to 30
>>>> min?  My re-invites occur much sooner than 30 min into a call.  Or does
>>>> session-timeout param only control sessions initiated by FS while incoming
>>>> sessions use the minimum-session-expires param if it's not explicitly
>>>> passed by the session initiator?
>>>>
>>>> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>>
>>>>> session timer
>>>>>
>>>>>
>>>>> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>>>>>
>>>>>> Thanks Michael, I'll see if we can do that!
>>>>>>
>>>>>> So, is the re-INVITE legit and the problem is that JsSip does not
>>>>>> respond to it?  Still, I am curious what is triggering the re-INVITE.
>>>>>>
>>>>>> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com>
>>>>>> wrote:
>>>>>>
>>>>>>> I think the sip.js guys fixed this issue when they forked jssip.
>>>>>>> I'd suggest using that.
>>>>>>>
>>>>>>> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
>>>>>>> wrote:
>>>>>>> >
>>>>>>> > Hi guys,
>>>>>>> >
>>>>>>> > Several weeks ago I started getting an occasional problem where FS
>>>>>>> is sending an INVITE to the other side in the middle of a call, the other
>>>>>>> side does not respond and FS hangs up the leg.  Below is the relevant log.
>>>>>>> The user experience is that they keep talking and hearing each other up to
>>>>>>> the very end.  I have a recording of that call, so can confirm.
>>>>>>> >
>>>>>>> > The call uses WebRTC and is originated by JsSip from Chrome.  Then
>>>>>>> the user is put into a conference but I doubt it's relevant in this case
>>>>>>> since the INVITE and disconnect are not happening from mod_conference
>>>>>>> >
>>>>>>> > I suspect it's a re-INVITE but what triggers FS to send it?  I
>>>>>>> couldn't find anything in the logs that could shed light.
>>>>>>> >
>>>>>>> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>>>>>>> >
>>>>>>> ------------------------------------------------------------------------
>>>>>>> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob
>>>>>>> SIP/2.0
>>>>>>> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>>>>>>> >    Max-Forwards: 70
>>>>>>> >    From: <
>>>>>>> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
>>>>>>> >;tag=KQecUSr12rSQp
>>>>>>> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>>>>>>> >    Call-ID: g8980rbrbk2t45oj5mru
>>>>>>> >    CSeq: 75703945 INVITE
>>>>>>> >    Contact: <
>>>>>>> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp
>>>>>>> >
>>>>>>> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>>>>>>> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
>>>>>>> UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>>>>> >    Supported: timer, path, replaces
>>>>>>> >    Session-Expires: 120;refresher=uac
>>>>>>> >    Min-SE: 120
>>>>>>> >    Content-Type: application/sdp
>>>>>>> >    Content-Disposition: session
>>>>>>> >    Content-Length: 825
>>>>>>> >
>>>>>>> >    v=0
>>>>>>> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>>>>>>> >    s=FreeSWITCH
>>>>>>> >    c=IN IP4 ##.##.##.##
>>>>>>> >    t=0 0
>>>>>>> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>>>> >    m=audio 22670 RTP/SAVPF 9 126 106
>>>>>>> >    a=rtpmap:9 G722/8000
>>>>>>> >    a=rtpmap:126 telephone-event/8000
>>>>>>> >    a=rtpmap:106 CN/8000
>>>>>>> >    a=ptime:20
>>>>>>> >    a=fingerprint:sha-256
>>>>>>> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>>>>>>> >    a=rtcp-mux
>>>>>>> >    a=rtcp:22670 IN IP4 ##.##.##.##
>>>>>>> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>>>>>>> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>>>>>>> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>>>>>>> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>>>>>>> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>>>>>>> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>>>>>>> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ
>>>>>>> host generation 0
>>>>>>> >
>>>>>>> ------------------------------------------------------------------------
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858
>>>>>>> [DEBUG] sofia.c:6627 Channel
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid entering state
>>>>>>> [calling][0]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [DEBUG] switch_core_session.c:1061 Send signal
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [DEBUG] sofia.c:6627 Channel
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid entering state
>>>>>>> [terminating][503]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [NOTICE] sofia.c:7543 Hangup
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [CS_EXECUTE]
>>>>>>> [NORMAL_TEMPORARY_FAILURE]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [DEBUG] switch_channel.c:3242 Send signal
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874
>>>>>>> [DEBUG] switch_core_session.c:1396 Send signal
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] mod_conference.c:5057 Channel leaving conference, cause:
>>>>>>> NORMAL_TEMPORARY_FAILURE
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] mod_conference.c:9650
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>>> [UNBRIDGE] (channel is hungup already)
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_media.c:7772
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>>> [REFER_EVENT] (channel is hungup already)
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_codec.c:246
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid Restore previous codec
>>>>>>> G722:9.
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_session.c:2901
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid skip receive message
>>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:535 (
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going
>>>>>>> to sleep
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:472 (
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
>>>>>>> CS_HANGUP
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:735 (
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change
>>>>>>> ACTIVE -> HANGUP
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:737 (
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] mod_sofia.c:413 Channel
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid hanging up, cause:
>>>>>>> NORMAL_TEMPORARY_FAILURE
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:60
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP,
>>>>>>> cause: NORMAL_TEMPORARY_FAILURE
>>>>>>> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900
>>>>>>> [DEBUG] switch_core_state_machine.c:737 (
>>>>>>> sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to
>>>>>>> sleep
>>>>>>> >
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 27 20:52:30 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 27 May 2015 11:52:30 -0500
Subject: [Freeswitch-users] Robin Raymond with ORTC Working Group today on
	ClueCon Weekly!
Message-ID: <D18B607E.9F9BA%krice@freeswitch.org>

Hey Guys,

Join us today at 1PM EST on ClueCon Weekly!

Today Robin Raymond of the ORTC Working Group will be speaking about ORTC,
what it is and where it is heading.

You can also watch on YouTube via https://youtu.be/4-LGMMJooOQ.

To join live go to https://conference.freeswitch.org/verto/ and call 888.
(it is recommended to have the latest Chrome browser with your headset and
webcam ready!)

Have a Great Week!

Ken 


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From Sharath.Kumar at meZocliq.com  Wed May 27 23:12:35 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Wed, 27 May 2015 19:12:35 +0000
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMO+Taai2bM8A6kF8hEUJ6ZJ3LfG0_qohqx+CDenwuX5jiA@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAFiqYum4if-B8NGFu7PXjFpJZnPuFGy8XhWscM7p7q6BwZiqGg@mail.gmail.com>
	<CAC8JMO+-Wqu9_xQx6TzkUU3NLhVU=HTpDu6wO2p1TX8J1kxEKg@mail.gmail.com>
	<CAC8JMO+YUezb5peCD0njovq83YwNYQMEx6ASee96RRq2-0Ocmg@mail.gmail.com>
	<DC7CE9BD-1271-4EBB-A7D1-4E3D0CCEBB46@jerris.com>
	<CAC8JMO+Taai2bM8A6kF8hEUJ6ZJ3LfG0_qohqx+CDenwuX5jiA@mail.gmail.com>
Message-ID: <CY1PR0501MB10987654E8A314E38D82DAA184CB0@CY1PR0501MB1098.namprd05.prod.outlook.com>

https://freeswitch.org/jira/browse/FS-7432

There is also this.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Oleg Stolyar
Sent: Wednesday, May 27, 2015 12:14 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] INVITE and hangup in the middle of a call

Jira: https://freeswitch.org/jira/browse/FS-7582

It turns out that FS actually does not send a=setup at all in reINVITES.  I am not comfortable enough with the code to add an SDP attribute.  Also this is now a low priority for me since there is a simple workaround of disabling session timers.

On Tue, May 26, 2015 at 9:07 AM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
Please file a jira for us on that.  Even better with a pull request to fix it.

On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar at gmail.com<mailto:olegstolyar at gmail.com>> wrote:

Just FYI, the JsSip engineers replied to my questions with this:

The error happens because the incoming reINVITE has a=setup:active in
the SDP which is a bug in FreeSwitch (the RFC clearly states that the
SDP offer MUST have a=setup:actpass):

On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar at gmail.com<mailto:olegstolyar at gmail.com>> wrote:
Thanks Steven!

It may be https://freeswitch.org/jira/browse/FS-7040.

As far as the 120 sec is concerned - Below are snippets from both the invite and the 200 OK from FS.  I know that FS reads the Session-Expires from the client because if I change it to a value less than 120, FS sends back a "SIP/2.0 422 Session Interval Too Small".  I  even thought the problem could be that the Session-Exipres format in the original INVITE is incorrect since it does not contain ";refresher=uac" but when I added that and made the line "Session-Expires: 300;refresher=uac" nothing changed.


INVITE sip:echo-test at anonymous.invalid SIP/2.0
Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029
Max-Forwards: 69
To: <sip:echo-test at anonymous.invalid>
From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
Call-ID: r65e48gp171p21rkppcu
CSeq: 6621 INVITE
Contact: <sip:mqthb9dj at pfcnm9rjv5en.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.26
Content-Length: 2754

SIP/2.0 200 OK
Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200
From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
To: <sip:echo-test at anonymous.invalid>;tag=SgjX0X4arHUFg
Call-ID: r65e48gp171p21rkppcu
CSeq: 6621 INVITE
Contact: <sip:echo-test at 50.19.255.62:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Session-Expires: 120;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 809
X-Netflix: yes
Remote-Party-ID: "echo-test" <sip:echo-test at anonymous.invalid>;party=calling;privacy=off;screen=no


On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes.
Session-Expires: 120;refresher=uac
Min-SE: 120

Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040

On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar at gmail.com<mailto:olegstolyar at gmail.com>> wrote:
But isn't that based on the session-timeout param which defaults to 30 min?  My re-invites occur much sooner than 30 min into a call.  Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?

On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
session timer


On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com<mailto:olegstolyar at gmail.com>> wrote:
Thanks Michael, I'll see if we can do that!

So, is the re-INVITE legit and the problem is that JsSip does not respond to it?  Still, I am curious what is triggering the re-INVITE.

On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
I think the sip.js guys fixed this issue when they forked jssip.  I'd suggest using that.

> On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com<mailto:olegstolyar at gmail.com>> wrote:
>
> Hi guys,
>
> Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg.  Below is the relevant log.  The user experience is that they keep talking and hearing each other up to the very end.  I have a recording of that call, so can confirm.
>
> The call uses WebRTC and is originated by JsSip from Chrome.  Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
>
> I suspect it's a re-INVITE but what triggers FS to send it?  I couldn't find anything in the logs that could shed light.
>
> send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>    ------------------------------------------------------------------------
>    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>    Max-Forwards: 70
>    From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid>;tag=KQecUSr12rSQp
>    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>    Call-ID: g8980rbrbk2t45oj5mru
>    CSeq: 75703945 INVITE
>    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Session-Expires: 120;refresher=uac
>    Min-SE: 120
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 825
>
>    v=0
>    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>    s=FreeSWITCH
>    c=IN IP4 ##.##.##.##
>    t=0 0
>    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>    m=audio 22670 RTP/SAVPF 9 126 106
>    a=rtpmap:9 G722/8000
>    a=rtpmap:126 telephone-event/8000
>    a=rtpmap:106 CN/8000
>    a=ptime:20
>    a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>    a=rtcp-mux
>    a=rtcp:22670 IN IP4 ##.##.##.##
>    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
>    ------------------------------------------------------------------------
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> entering state [calling][0]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> entering state [terminating][503]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [KILL]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> skip receive message [UNBRIDGE] (channel is hungup already)
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> skip receive message [REFER_EVENT] (channel is hungup already)
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> Restore previous codec G722:9.
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid>) State EXECUTE going to sleep
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid>) Running State Change CS_HANGUP
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid>) Callstate Change ACTIVE -> HANGUP
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid>) State HANGUP
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> hanging up, cause: NORMAL_TEMPORARY_FAILURE
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid>) State HANGUP going to sleep
>

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From craig at stevenson.com  Thu May 28 00:02:39 2015
From: craig at stevenson.com (Craig Stevenson)
Date: Wed, 27 May 2015 13:02:39 -0700
Subject: [Freeswitch-users] FreeSWITCH 1.6+Video ... TLS ports 5061, 5081,
	8082 issues
Message-ID: <CACLfcpvm1eJBUUjY2Gbt1hEPnv5T6fuePJJkP0tjrviw_4hLkA@mail.gmail.com>

I'm new at this and hoping someone can help me figure out what I'm
missing...

I've exactly followed the instructions at:

https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video#FreeSWITCH1.6Video-Introduction

But, the open_ssl test (test_tls.sh) is only working for 443 and failing
with connection refused (errno=111) for port 5061, port 5081 and port 8082.

Much thanks.
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From krice at freeswitch.org  Thu May 28 00:13:13 2015
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 27 May 2015 15:13:13 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.6+Video ... TLS ports 5061, 5081,
 8082 issues
In-Reply-To: <CACLfcpvm1eJBUUjY2Gbt1hEPnv5T6fuePJJkP0tjrviw_4hLkA@mail.gmail.com>
Message-ID: <D18B8F89.9F9D6%krice@freeswitch.org>

Sounds like FreeSWITCH isnt actually tunning its what listens on the high
ports, port 443 is apache in if following that doc


On 5/27/15, 3:02 PM, "Craig Stevenson" <craig at stevenson.com> wrote:

> I'm new at this and hoping someone can help me figure out what I'm missing...?
> 
> I've exactly followed the instructions at:
> ?https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video#Fre
> eSWITCH1.6Video-Introduction
> 
> But, the open_ssl test (test_tls.sh) is only working for 443 and failing with
> connection refused (errno=111) for port 5061, port 5081 and port 8082.
> 
> Much 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
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-- 
Ken
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irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From olegstolyar at gmail.com  Thu May 28 00:23:33 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 27 May 2015 13:23:33 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CY1PR0501MB10987654E8A314E38D82DAA184CB0@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAFiqYum4if-B8NGFu7PXjFpJZnPuFGy8XhWscM7p7q6BwZiqGg@mail.gmail.com>
	<CAC8JMO+-Wqu9_xQx6TzkUU3NLhVU=HTpDu6wO2p1TX8J1kxEKg@mail.gmail.com>
	<CAC8JMO+YUezb5peCD0njovq83YwNYQMEx6ASee96RRq2-0Ocmg@mail.gmail.com>
	<DC7CE9BD-1271-4EBB-A7D1-4E3D0CCEBB46@jerris.com>
	<CAC8JMO+Taai2bM8A6kF8hEUJ6ZJ3LfG0_qohqx+CDenwuX5jiA@mail.gmail.com>
	<CY1PR0501MB10987654E8A314E38D82DAA184CB0@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <CAC8JMO+0B855EkuTJFv_2faDGxNXdV4YWz2f2wwa9SB_9jxT8w@mail.gmail.com>

You are right Sharath!  I resolved mine as a duplicate.

On Wed, May 27, 2015 at 12:12 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
wrote:

>  https://freeswitch.org/jira/browse/FS-7432
>
>
>
> There is also this.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Oleg Stolyar
> *Sent:* Wednesday, May 27, 2015 12:14 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] INVITE and hangup in the middle of a
> call
>
>
>
> Jira: https://freeswitch.org/jira/browse/FS-7582
>
>
>
> It turns out that FS actually does not send a=setup at all in reINVITES.
> I am not comfortable enough with the code to add an SDP attribute.  Also
> this is now a low priority for me since there is a simple workaround of
> disabling session timers.
>
>
>
> On Tue, May 26, 2015 at 9:07 AM, Michael Jerris <mike at jerris.com> wrote:
>
>  Please file a jira for us on that.  Even better with a pull request to
> fix it.
>
>
>
>  On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>
>
> Just FYI, the JsSip engineers replied to my questions with this:
>
>
>
> The error happens because the incoming reINVITE has a=setup:active in
> the SDP which is a bug in FreeSwitch (the RFC clearly states that the
> SDP offer MUST have a=setup:actpass):
>
>
>
> On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>  Thanks Steven!
>
>
>
> It may be https://freeswitch.org/jira/browse/FS-7040.
>
>
>
> As far as the 120 sec is concerned - Below are snippets from both the
> invite and the 200 OK from FS.  I know that FS reads the Session-Expires
> from the client because if I change it to a value less than 120, FS sends
> back a "SIP/2.0 422 Session Interval Too Small".  I  even thought the
> problem could be that the Session-Exipres format in the original INVITE is
> incorrect since it does not contain ";refresher=uac" but when I added that
> and made the line "Session-Expires: 300;refresher=uac" nothing changed.
>
>
>
>
>
> INVITE sip:echo-test at anonymous.invalid SIP/2.0
>
> Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029
>
> Max-Forwards: 69
>
> To: <sip:echo-test at anonymous.invalid>
>
> From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
>
> Call-ID: r65e48gp171p21rkppcu
>
> CSeq: 6621 INVITE
>
> Contact: <sip:mqthb9dj at pfcnm9rjv5en.invalid;transport=ws;ob>
>
> Content-Type: application/sdp
>
> Session-Expires: 300
>
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
>
> Supported: timer,ice,outbound
>
> User-Agent: JsSIP 0.6.26
>
> Content-Length: 2754
>
>
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/WSS
> pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200
>
> From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
>
> To: <sip:echo-test at anonymous.invalid>;tag=SgjX0X4arHUFg
>
> Call-ID: r65e48gp171p21rkppcu
>
> CSeq: 6621 INVITE
>
> Contact: <sip:echo-test at 50.19.255.62:5080;transport=udp>
>
> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>
> Accept: application/sdp
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
> Require: timer
>
> Supported: timer, path, replaces
>
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
> line-seize, call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
>
> Session-Expires: 120;refresher=uac
>
> Content-Type: application/sdp
>
> Content-Disposition: session
>
> Content-Length: 809
>
> X-Netflix: yes
>
> Remote-Party-ID: "echo-test" <sip:echo-test at anonymous.invalid
> >;party=calling;privacy=off;screen=no
>
>
>
>
>
> On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>  The values to actual use are negotiated during the call invite, and your
> trace shows it's using 2 minutes.
>
> Session-Expires: 120;refresher=uac
> Min-SE: 120
>
> Just as an idea because you didn't send both invite and re-invite...
> perhaps the SDP body is different in the reinvite without the version
> number having changed. If so it may be
> https://freeswitch.org/jira/browse/FS-7040
>
>
>
> On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>  But isn't that based on the session-timeout param which defaults to 30
> min?  My re-invites occur much sooner than 30 min into a call.  Or does
> session-timeout param only control sessions initiated by FS while incoming
> sessions use the minimum-session-expires param if it's not explicitly
> passed by the session initiator?
>
>
>
> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com> wrote:
>
> session timer
>
>
>
> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>  Thanks Michael, I'll see if we can do that!
>
>
>
> So, is the re-INVITE legit and the problem is that JsSip does not respond
> to it?  Still, I am curious what is triggering the re-INVITE.
>
>
>
> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com> wrote:
>
> I think the sip.js guys fixed this issue when they forked jssip.  I'd
> suggest using that.
>
>
> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
> >
> > Hi guys,
> >
> > Several weeks ago I started getting an occasional problem where FS is
> sending an INVITE to the other side in the middle of a call, the other side
> does not respond and FS hangs up the leg.  Below is the relevant log.  The
> user experience is that they keep talking and hearing each other up to the
> very end.  I have a recording of that call, so can confirm.
> >
> > The call uses WebRTC and is originated by JsSip from Chrome.  Then the
> user is put into a conference but I doubt it's relevant in this case since
> the INVITE and disconnect are not happening from mod_conference
> >
> > I suspect it's a re-INVITE but what triggers FS to send it?  I couldn't
> find anything in the logs that could shed light.
> >
> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
> >
> ------------------------------------------------------------------------
> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
> >    Max-Forwards: 70
> >    From: <
> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
> >;tag=KQecUSr12rSQp
> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
> >    Call-ID: g8980rbrbk2t45oj5mru
> >    CSeq: 75703945 INVITE
> >    Contact: <
> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp
> >
> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >    Supported: timer, path, replaces
> >    Session-Expires: 120;refresher=uac
> >    Min-SE: 120
> >    Content-Type: application/sdp
> >    Content-Disposition: session
> >    Content-Length: 825
> >
> >    v=0
> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
> >    s=FreeSWITCH
> >    c=IN IP4 ##.##.##.##
> >    t=0 0
> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
> >    m=audio 22670 RTP/SAVPF 9 126 106
> >    a=rtpmap:9 G722/8000
> >    a=rtpmap:126 telephone-event/8000
> >    a=rtpmap:106 CN/8000
> >    a=ptime:20
> >    a=fingerprint:sha-256
> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
> >    a=rtcp-mux
> >    a=rtcp:22670 IN IP4 ##.##.##.##
> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host
> generation 0
> >
> ------------------------------------------------------------------------
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
> switch_core_session.c:1061 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
> sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
> entering state [calling][0]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_core_session.c:1061 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_core_session.c:1061 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
> entering state [terminating][503]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE]
> sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_channel.c:3242 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_core_session.c:1396 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> mod_conference.c:5057 Channel leaving conference, cause:
> NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid skip
> receive message [UNBRIDGE] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_media.c:7772 sofia/leia_agent/anonymous at anonymous.invalid
> skip receive message [REFER_EVENT] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
> Restore previous codec G722:9.
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_session.c:2901 sofia/leia_agent/anonymous at anonymous.invalid
> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:535 (
> sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to sleep
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:472 (
> sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
> CS_HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:735 (
> sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE ->
> HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:737 (
> sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
> hanging up, cause: NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:60
> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause:
> NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:737 (
> sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to sleep
> >
>
>
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>
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>
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From Sharath.Kumar at meZocliq.com  Thu May 28 00:47:50 2015
From: Sharath.Kumar at meZocliq.com (Sharath Kumar)
Date: Wed, 27 May 2015 20:47:50 +0000
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CAC8JMO+0B855EkuTJFv_2faDGxNXdV4YWz2f2wwa9SB_9jxT8w@mail.gmail.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAFiqYum4if-B8NGFu7PXjFpJZnPuFGy8XhWscM7p7q6BwZiqGg@mail.gmail.com>
	<CAC8JMO+-Wqu9_xQx6TzkUU3NLhVU=HTpDu6wO2p1TX8J1kxEKg@mail.gmail.com>
	<CAC8JMO+YUezb5peCD0njovq83YwNYQMEx6ASee96RRq2-0Ocmg@mail.gmail.com>
	<DC7CE9BD-1271-4EBB-A7D1-4E3D0CCEBB46@jerris.com>
	<CAC8JMO+Taai2bM8A6kF8hEUJ6ZJ3LfG0_qohqx+CDenwuX5jiA@mail.gmail.com>
	<CY1PR0501MB10987654E8A314E38D82DAA184CB0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAC8JMO+0B855EkuTJFv_2faDGxNXdV4YWz2f2wwa9SB_9jxT8w@mail.gmail.com>
Message-ID: <CY1PR0501MB1098547B9134A80987EF4B2C84CB0@CY1PR0501MB1098.namprd05.prod.outlook.com>

Thanks Oleg. Disabling session timers is okay to fix it for regular calls albeit without session timers. But there is also the case of hold/retrieve and other mid-call offer/answers, like say adding someone into a conference etc, where this problem arises. This is a complicated problem.  I am yet to test the fix. But chrome has some outstanding problems with role changes as well. They don?t fully support the RFC yet.
https://code.google.com/p/webrtc/issues/detail?id=2782

"#4 juberti at chromium.org<mailto:juberti at chromium.org>
Offers MUST always use actpass, but Chrome does not support role changes. So, if A calls B, A offers actpass, B responds with active. But then if B does a reoffer to A, B has to offer actpass, and A must respond with passive (since A was chosen as passive before by B's choice of active). Naturally, if A reoffers, A offers actpass and B must respond (again) with active."


Thank you,
Sharath



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Oleg Stolyar
Sent: Wednesday, May 27, 2015 4:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] INVITE and hangup in the middle of a call

You are right Sharath!  I resolved mine as a duplicate.

On Wed, May 27, 2015 at 12:12 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com<mailto:Sharath.Kumar at mezocliq.com>> wrote:
https://freeswitch.org/jira/browse/FS-7432

There is also this.

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 Oleg Stolyar
Sent: Wednesday, May 27, 2015 12:14 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] INVITE and hangup in the middle of a call

Jira: https://freeswitch.org/jira/browse/FS-7582

It turns out that FS actually does not send a=setup at all in reINVITES.  I am not comfortable enough with the code to add an SDP attribute.  Also this is now a low priority for me since there is a simple workaround of disabling session timers.

On Tue, May 26, 2015 at 9:07 AM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
Please file a jira for us on that.  Even better with a pull request to fix it.

On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar at gmail.com<mailto:olegstolyar at gmail.com>> wrote:

Just FYI, the JsSip engineers replied to my questions with this:

The error happens because the incoming reINVITE has a=setup:active in
the SDP which is a bug in FreeSwitch (the RFC clearly states that the
SDP offer MUST have a=setup:actpass):

On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar at gmail.com<mailto:olegstolyar at gmail.com>> wrote:
Thanks Steven!

It may be https://freeswitch.org/jira/browse/FS-7040.

As far as the 120 sec is concerned - Below are snippets from both the invite and the 200 OK from FS.  I know that FS reads the Session-Expires from the client because if I change it to a value less than 120, FS sends back a "SIP/2.0 422 Session Interval Too Small".  I  even thought the problem could be that the Session-Exipres format in the original INVITE is incorrect since it does not contain ";refresher=uac" but when I added that and made the line "Session-Expires: 300;refresher=uac" nothing changed.


INVITE sip:echo-test at anonymous.invalid SIP/2.0
Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029
Max-Forwards: 69
To: <sip:echo-test at anonymous.invalid>
From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
Call-ID: r65e48gp171p21rkppcu
CSeq: 6621 INVITE
Contact: <sip:mqthb9dj at pfcnm9rjv5en.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 300
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.26
Content-Length: 2754

SIP/2.0 200 OK
Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200
From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
To: <sip:echo-test at anonymous.invalid>;tag=SgjX0X4arHUFg
Call-ID: r65e48gp171p21rkppcu
CSeq: 6621 INVITE
Contact: <sip:echo-test at 50.19.255.62:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Session-Expires: 120;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 809
X-Netflix: yes
Remote-Party-ID: "echo-test" <sip:echo-test at anonymous.invalid>;party=calling;privacy=off;screen=no


On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre at gmail.com<mailto:steveayre at gmail.com>> wrote:
The values to actual use are negotiated during the call invite, and your trace shows it's using 2 minutes.
Session-Expires: 120;refresher=uac
Min-SE: 120

Just as an idea because you didn't send both invite and re-invite... perhaps the SDP body is different in the reinvite without the version number having changed. If so it may be https://freeswitch.org/jira/browse/FS-7040

On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar at gmail.com<mailto:olegstolyar at gmail.com>> wrote:
But isn't that based on the session-timeout param which defaults to 30 min?  My re-invites occur much sooner than 30 min into a call.  Or does session-timeout param only control sessions initiated by FS while incoming sessions use the minimum-session-expires param if it's not explicitly passed by the session initiator?

On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
session timer


On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com<mailto:olegstolyar at gmail.com>> wrote:
Thanks Michael, I'll see if we can do that!

So, is the re-INVITE legit and the problem is that JsSip does not respond to it?  Still, I am curious what is triggering the re-INVITE.

On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
I think the sip.js guys fixed this issue when they forked jssip.  I'd suggest using that.

> On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com<mailto:olegstolyar at gmail.com>> wrote:
>
> Hi guys,
>
> Several weeks ago I started getting an occasional problem where FS is sending an INVITE to the other side in the middle of a call, the other side does not respond and FS hangs up the leg.  Below is the relevant log.  The user experience is that they keep talking and hearing each other up to the very end.  I have a recording of that call, so can confirm.
>
> The call uses WebRTC and is originated by JsSip from Chrome.  Then the user is put into a conference but I doubt it's relevant in this case since the INVITE and disconnect are not happening from mod_conference
>
> I suspect it's a re-INVITE but what triggers FS to send it?  I couldn't find anything in the logs that could shed light.
>
> send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
>    ------------------------------------------------------------------------
>    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
>    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
>    Max-Forwards: 70
>    From: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid>;tag=KQecUSr12rSQp
>    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
>    Call-ID: g8980rbrbk2t45oj5mru
>    CSeq: 75703945 INVITE
>    Contact: <sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp>
>    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Session-Expires: 120;refresher=uac
>    Min-SE: 120
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 825
>
>    v=0
>    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
>    s=FreeSWITCH
>    c=IN IP4 ##.##.##.##
>    t=0 0
>    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>    m=audio 22670 RTP/SAVPF 9 126 106
>    a=rtpmap:9 G722/8000
>    a=rtpmap:126 telephone-event/8000
>    a=rtpmap:106 CN/8000
>    a=ptime:20
>    a=fingerprint:sha-256 E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
>    a=rtcp-mux
>    a=rtcp:22670 IN IP4 ##.##.##.##
>    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
>    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
>    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
>    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
>    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
>    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
>    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host generation 0
>    ------------------------------------------------------------------------
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> entering state [calling][0]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1061 Send signal sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> entering state [terminating][503]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE] sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_channel.c:3242 Send signal sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [KILL]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG] switch_core_session.c:1396 Send signal sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> [BREAK]
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:5057 Channel leaving conference, cause: NORMAL_TEMPORARY_FAILURE
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> skip receive message [UNBRIDGE] (channel is hungup already)
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_media.c:7772 sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> skip receive message [REFER_EVENT] (channel is hungup already)
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> Restore previous codec G722:9.
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_session.c:2901 sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:535 (sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid>) State EXECUTE going to sleep
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:472 (sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid>) Running State Change CS_HANGUP
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:735 (sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid>) Callstate Change ACTIVE -> HANGUP
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid>) State HANGUP
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> hanging up, cause: NORMAL_TEMPORARY_FAILURE
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:60 sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid> Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE
> fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG] switch_core_state_machine.c:737 (sofia/leia_agent/anonymous at anonymous.invalid<mailto:sofia/leia_agent/anonymous at anonymous.invalid>) State HANGUP going to sleep
>
_________________________________________________________________________
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://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 italorossib at gmail.com  Thu May 28 00:51:56 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 27 May 2015 17:51:56 -0300
Subject: [Freeswitch-users] FreeSWITCH 1.6+Video ... TLS ports 5061, 5081,
 8082 issues
In-Reply-To: <D18B8F89.9F9D6%krice@freeswitch.org>
References: <CACLfcpvm1eJBUUjY2Gbt1hEPnv5T6fuePJJkP0tjrviw_4hLkA@mail.gmail.com>
	<D18B8F89.9F9D6%krice@freeswitch.org>
Message-ID: <CAJR7f2H9T0H93NbGJRH0jhEbR1i1npJ-UMHU5-AMX0PMifCo=Q@mail.gmail.com>

Craig,

freeswitch at internal> global_setvar internal_ssl_enable=true
+OK
freeswitch at internal> global_setvar external_ssl_enable=true
+OK
freeswitch at internal> sofia profile external restart reloadxml
Reload XML [Success]
restarting: external
freeswitch at internal> sofia profile internal restart reloadxml
Reload XML [Success]
restarting: internal

I'll update the docs with these steps.


On Wed, May 27, 2015 at 5:13 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Sounds like FreeSWITCH isnt actually tunning its what listens on the
> high ports, port 443 is apache in if following that doc
>
>
>
> On 5/27/15, 3:02 PM, "Craig Stevenson" <craig at stevenson.com> wrote:
>
> I'm new at this and hoping someone can help me figure out what I'm
> missing...
>
> I've exactly followed the instructions at:
>
> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video#FreeSWITCH1.6Video-Introduction
>
> But, the open_ssl test (test_tls.sh) is only working for 443 and failing
> with connection refused (errno=111) for port 5061, port 5081 and port 8082.
>
> Much 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
>
>
> --
> Ken
>
>
>
> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From italorossib at gmail.com  Thu May 28 00:52:40 2015
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 27 May 2015 17:52:40 -0300
Subject: [Freeswitch-users] FreeSWITCH 1.6+Video ... TLS ports 5061, 5081,
 8082 issues
In-Reply-To: <CAJR7f2H9T0H93NbGJRH0jhEbR1i1npJ-UMHU5-AMX0PMifCo=Q@mail.gmail.com>
References: <CACLfcpvm1eJBUUjY2Gbt1hEPnv5T6fuePJJkP0tjrviw_4hLkA@mail.gmail.com>
	<D18B8F89.9F9D6%krice@freeswitch.org>
	<CAJR7f2H9T0H93NbGJRH0jhEbR1i1npJ-UMHU5-AMX0PMifCo=Q@mail.gmail.com>
Message-ID: <CAJR7f2GDAvfVFdto4XkCRq6dbrHQ6yJ-KqTC1Hi1_iPQoRPJAw@mail.gmail.com>

Also,

don't forget to set these values in your vars.xml

On Wed, May 27, 2015 at 5:51 PM, ?talo Rossi <italorossib at gmail.com> wrote:

> Craig,
>
> freeswitch at internal> global_setvar internal_ssl_enable=true
> +OK
> freeswitch at internal> global_setvar external_ssl_enable=true
> +OK
> freeswitch at internal> sofia profile external restart reloadxml
> Reload XML [Success]
> restarting: external
> freeswitch at internal> sofia profile internal restart reloadxml
> Reload XML [Success]
> restarting: internal
>
> I'll update the docs with these steps.
>
>
> On Wed, May 27, 2015 at 5:13 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>>  Sounds like FreeSWITCH isnt actually tunning its what listens on the
>> high ports, port 443 is apache in if following that doc
>>
>>
>>
>> On 5/27/15, 3:02 PM, "Craig Stevenson" <craig at stevenson.com> wrote:
>>
>> I'm new at this and hoping someone can help me figure out what I'm
>> missing...
>>
>> I've exactly followed the instructions at:
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video#FreeSWITCH1.6Video-Introduction
>>
>> But, the open_ssl test (test_tls.sh) is only working for 443 and failing
>> with connection refused (errno=111) for port 5061, port 5081 and port 8082.
>>
>> Much 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
>>
>>
>> --
>> Ken
>>
>>
>>
>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>> Twitter: @FreeSWITCH
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>



-- 
?talo Rossi
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From gmaruzz at gmail.com  Thu May 28 01:05:01 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 27 May 2015 23:05:01 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.6+Video ... TLS ports 5061, 5081,
 8082 issues
In-Reply-To: <CAJR7f2GDAvfVFdto4XkCRq6dbrHQ6yJ-KqTC1Hi1_iPQoRPJAw@mail.gmail.com>
References: <CACLfcpvm1eJBUUjY2Gbt1hEPnv5T6fuePJJkP0tjrviw_4hLkA@mail.gmail.com>
	<D18B8F89.9F9D6%krice@freeswitch.org>
	<CAJR7f2H9T0H93NbGJRH0jhEbR1i1npJ-UMHU5-AMX0PMifCo=Q@mail.gmail.com>
	<CAJR7f2GDAvfVFdto4XkCRq6dbrHQ6yJ-KqTC1Hi1_iPQoRPJAw@mail.gmail.com>
Message-ID: <CALXCt0qz13R-Pj_5vWRfr__h+7prtLNLAfdnkGErT9BkC5kgVQ@mail.gmail.com>

Those values for vars.xml are listed in the freeswitch1.6 page.

Just shutdown and restart freeswitch, but I believe restarting is listed
too.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 27, 2015 10:53 PM, "?talo Rossi" <italorossib at gmail.com> wrote:

> Also,
>
> don't forget to set these values in your vars.xml
>
> On Wed, May 27, 2015 at 5:51 PM, ?talo Rossi <italorossib at gmail.com>
> wrote:
>
>> Craig,
>>
>> freeswitch at internal> global_setvar internal_ssl_enable=true
>> +OK
>> freeswitch at internal> global_setvar external_ssl_enable=true
>> +OK
>> freeswitch at internal> sofia profile external restart reloadxml
>> Reload XML [Success]
>> restarting: external
>> freeswitch at internal> sofia profile internal restart reloadxml
>> Reload XML [Success]
>> restarting: internal
>>
>> I'll update the docs with these steps.
>>
>>
>> On Wed, May 27, 2015 at 5:13 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>>>  Sounds like FreeSWITCH isnt actually tunning its what listens on the
>>> high ports, port 443 is apache in if following that doc
>>>
>>>
>>>
>>> On 5/27/15, 3:02 PM, "Craig Stevenson" <craig at stevenson.com> wrote:
>>>
>>> I'm new at this and hoping someone can help me figure out what I'm
>>> missing...
>>>
>>> I've exactly followed the instructions at:
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video#FreeSWITCH1.6Video-Introduction
>>>
>>> But, the open_ssl test (test_tls.sh) is only working for 443 and failing
>>> with connection refused (errno=111) for port 5061, port 5081 and port 8082.
>>>
>>> Much 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
>>>
>>>
>>> --
>>> Ken
>>>
>>>
>>>
>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>> Twitter: @FreeSWITCH
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>
>
>
> --
> ?talo Rossi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From craig at stevenson.com  Thu May 28 01:11:54 2015
From: craig at stevenson.com (Craig Stevenson)
Date: Wed, 27 May 2015 14:11:54 -0700
Subject: [Freeswitch-users] FreeSWITCH 1.6+Video ... TLS ports 5061, 5081,
 8082 issues
In-Reply-To: <CALXCt0qz13R-Pj_5vWRfr__h+7prtLNLAfdnkGErT9BkC5kgVQ@mail.gmail.com>
References: <CACLfcpvm1eJBUUjY2Gbt1hEPnv5T6fuePJJkP0tjrviw_4hLkA@mail.gmail.com>
	<D18B8F89.9F9D6%krice@freeswitch.org>
	<CAJR7f2H9T0H93NbGJRH0jhEbR1i1npJ-UMHU5-AMX0PMifCo=Q@mail.gmail.com>
	<CAJR7f2GDAvfVFdto4XkCRq6dbrHQ6yJ-KqTC1Hi1_iPQoRPJAw@mail.gmail.com>
	<CALXCt0qz13R-Pj_5vWRfr__h+7prtLNLAfdnkGErT9BkC5kgVQ@mail.gmail.com>
Message-ID: <CACLfcpu=c_jY-ujSt4w7c5U+r8BGQ_AdvkcqtqJaZBLVfVeTtA@mail.gmail.com>

As a side note -- the current instructions do not say anything about
starting FreeSWITCH.  It shows restarting Apache and then running the
openssl tests.



On Wed, May 27, 2015 at 2:05 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Those values for vars.xml are listed in the freeswitch1.6 page.
>
> Just shutdown and restart freeswitch, but I believe restarting is listed
> too.
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On May 27, 2015 10:53 PM, "?talo Rossi" <italorossib at gmail.com> wrote:
>
>> Also,
>>
>> don't forget to set these values in your vars.xml
>>
>> On Wed, May 27, 2015 at 5:51 PM, ?talo Rossi <italorossib at gmail.com>
>> wrote:
>>
>>> Craig,
>>>
>>> freeswitch at internal> global_setvar internal_ssl_enable=true
>>> +OK
>>> freeswitch at internal> global_setvar external_ssl_enable=true
>>> +OK
>>> freeswitch at internal> sofia profile external restart reloadxml
>>> Reload XML [Success]
>>> restarting: external
>>> freeswitch at internal> sofia profile internal restart reloadxml
>>> Reload XML [Success]
>>> restarting: internal
>>>
>>> I'll update the docs with these steps.
>>>
>>>
>>> On Wed, May 27, 2015 at 5:13 PM, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>>>  Sounds like FreeSWITCH isnt actually tunning its what listens on the
>>>> high ports, port 443 is apache in if following that doc
>>>>
>>>>
>>>>
>>>> On 5/27/15, 3:02 PM, "Craig Stevenson" <craig at stevenson.com> wrote:
>>>>
>>>> I'm new at this and hoping someone can help me figure out what I'm
>>>> missing...
>>>>
>>>> I've exactly followed the instructions at:
>>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video#FreeSWITCH1.6Video-Introduction
>>>>
>>>> But, the open_ssl test (test_tls.sh) is only working for 443 and
>>>> failing with connection refused (errno=111) for port 5061, port 5081 and
>>>> port 8082.
>>>>
>>>> Much 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
>>>>
>>>>
>>>> --
>>>> Ken
>>>>
>>>>
>>>>
>>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>> Twitter: @FreeSWITCH
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>
>>
>>
>> --
>> ?talo Rossi
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From olegstolyar at gmail.com  Thu May 28 01:13:46 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Wed, 27 May 2015 14:13:46 -0700
Subject: [Freeswitch-users] INVITE and hangup in the middle of a call
In-Reply-To: <CY1PR0501MB1098547B9134A80987EF4B2C84CB0@CY1PR0501MB1098.namprd05.prod.outlook.com>
References: <CAC8JMOKtxO_UK3-tCvWZAeyxP=f3jj2u2wunpXnhwjCtv-RBOg@mail.gmail.com>
	<08EC14CB-FA17-4848-ABCC-7C68D4C3441C@jerris.com>
	<CAC8JMOKpqwUdRZg-i8qLpg2MihZkco8yNWnOz35AMig7E+5T7A@mail.gmail.com>
	<CAEwTsAgWktzwOWPxyarxwic8m5TC5R-ioPdCANd+eoV=qbynDA@mail.gmail.com>
	<CAC8JMOKASHJLAhhDSabrvR=NHjoJjQSJXkX0heQPKuj+zjiFwQ@mail.gmail.com>
	<CAFiqYum4if-B8NGFu7PXjFpJZnPuFGy8XhWscM7p7q6BwZiqGg@mail.gmail.com>
	<CAC8JMO+-Wqu9_xQx6TzkUU3NLhVU=HTpDu6wO2p1TX8J1kxEKg@mail.gmail.com>
	<CAC8JMO+YUezb5peCD0njovq83YwNYQMEx6ASee96RRq2-0Ocmg@mail.gmail.com>
	<DC7CE9BD-1271-4EBB-A7D1-4E3D0CCEBB46@jerris.com>
	<CAC8JMO+Taai2bM8A6kF8hEUJ6ZJ3LfG0_qohqx+CDenwuX5jiA@mail.gmail.com>
	<CY1PR0501MB10987654E8A314E38D82DAA184CB0@CY1PR0501MB1098.namprd05.prod.outlook.com>
	<CAC8JMO+0B855EkuTJFv_2faDGxNXdV4YWz2f2wwa9SB_9jxT8w@mail.gmail.com>
	<CY1PR0501MB1098547B9134A80987EF4B2C84CB0@CY1PR0501MB1098.namprd05.prod.outlook.com>
Message-ID: <CAC8JMO+CHh-fqEF2vt-4yCvHSqvJdiW6Y-W-RRCWo=+kRun-OA@mail.gmail.com>

Sure Sharath, I only meant that the workaround works for me since I don't
need the mid-call offers/answers.

On Wed, May 27, 2015 at 1:47 PM, Sharath Kumar <Sharath.Kumar at mezocliq.com>
wrote:

>  Thanks Oleg. Disabling session timers is okay to fix it for regular
> calls albeit without session timers. But there is also the case of
> hold/retrieve and other mid-call offer/answers, like say adding someone
> into a conference etc, where this problem arises. This is a complicated
> problem.  I am yet to test the fix. But chrome has some outstanding
> problems with role changes as well. They don?t fully support the RFC yet.
>
> https://code.google.com/p/webrtc/issues/detail?id=2782
>
>
>
> "#4 juberti at chromium.org
> Offers MUST always use actpass, but Chrome does not support role changes.
> So, if A calls B, A offers actpass, B responds with active. But then if B
> does a reoffer to A, B has to offer actpass, and A must respond with
> passive (since A was chosen as passive before by B's choice of active).
> Naturally, if A reoffers, A offers actpass and B must respond (again) with
> active."
>
>
>
>
>
> Thank you,
>
> Sharath
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Oleg Stolyar
> *Sent:* Wednesday, May 27, 2015 4:24 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] INVITE and hangup in the middle of a
> call
>
>
>
> You are right Sharath!  I resolved mine as a duplicate.
>
>
>
> On Wed, May 27, 2015 at 12:12 PM, Sharath Kumar <
> Sharath.Kumar at mezocliq.com> wrote:
>
>  https://freeswitch.org/jira/browse/FS-7432
>
>
>
> There is also this.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Oleg Stolyar
> *Sent:* Wednesday, May 27, 2015 12:14 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] INVITE and hangup in the middle of a
> call
>
>
>
> Jira: https://freeswitch.org/jira/browse/FS-7582
>
>
>
> It turns out that FS actually does not send a=setup at all in reINVITES.
> I am not comfortable enough with the code to add an SDP attribute.  Also
> this is now a low priority for me since there is a simple workaround of
> disabling session timers.
>
>
>
> On Tue, May 26, 2015 at 9:07 AM, Michael Jerris <mike at jerris.com> wrote:
>
>  Please file a jira for us on that.  Even better with a pull request to
> fix it.
>
>
>
>  On May 26, 2015, at 11:13 AM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>
>
> Just FYI, the JsSip engineers replied to my questions with this:
>
>
>
> The error happens because the incoming reINVITE has a=setup:active in
> the SDP which is a bug in FreeSwitch (the RFC clearly states that the
> SDP offer MUST have a=setup:actpass):
>
>
>
> On Mon, May 25, 2015 at 7:36 AM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
>
>  Thanks Steven!
>
>
>
> It may be https://freeswitch.org/jira/browse/FS-7040.
>
>
>
> As far as the 120 sec is concerned - Below are snippets from both the
> invite and the 200 OK from FS.  I know that FS reads the Session-Expires
> from the client because if I change it to a value less than 120, FS sends
> back a "SIP/2.0 422 Session Interval Too Small".  I  even thought the
> problem could be that the Session-Exipres format in the original INVITE is
> incorrect since it does not contain ";refresher=uac" but when I added that
> and made the line "Session-Expires: 300;refresher=uac" nothing changed.
>
>
>
>
>
> INVITE sip:echo-test at anonymous.invalid SIP/2.0
>
> Via: SIP/2.0/WSS pfcnm9rjv5en.invalid;branch=z9hG4bK8223029
>
> Max-Forwards: 69
>
> To: <sip:echo-test at anonymous.invalid>
>
> From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
>
> Call-ID: r65e48gp171p21rkppcu
>
> CSeq: 6621 INVITE
>
> Contact: <sip:mqthb9dj at pfcnm9rjv5en.invalid;transport=ws;ob>
>
> Content-Type: application/sdp
>
> Session-Expires: 300
>
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
>
> Supported: timer,ice,outbound
>
> User-Agent: JsSIP 0.6.26
>
> Content-Length: 2754
>
>
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/WSS
> pfcnm9rjv5en.invalid;branch=z9hG4bK8223029;received=69.53.236.236;rport=35200
>
> From: "b3N0b2x5YXI" <sip:anonymous at anonymous.invalid>;tag=h1059gcvb2
>
> To: <sip:echo-test at anonymous.invalid>;tag=SgjX0X4arHUFg
>
> Call-ID: r65e48gp171p21rkppcu
>
> CSeq: 6621 INVITE
>
> Contact: <sip:echo-test at 50.19.255.62:5080;transport=udp>
>
> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>
> Accept: application/sdp
>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>
> Require: timer
>
> Supported: timer, path, replaces
>
> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
> line-seize, call-info, sla, include-session-description, presence.winfo,
> message-summary, refer
>
> Session-Expires: 120;refresher=uac
>
> Content-Type: application/sdp
>
> Content-Disposition: session
>
> Content-Length: 809
>
> X-Netflix: yes
>
> Remote-Party-ID: "echo-test" <sip:echo-test at anonymous.invalid
> >;party=calling;privacy=off;screen=no
>
>
>
>
>
> On Mon, May 25, 2015 at 1:51 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>  The values to actual use are negotiated during the call invite, and your
> trace shows it's using 2 minutes.
>
> Session-Expires: 120;refresher=uac
> Min-SE: 120
>
> Just as an idea because you didn't send both invite and re-invite...
> perhaps the SDP body is different in the reinvite without the version
> number having changed. If so it may be
> https://freeswitch.org/jira/browse/FS-7040
>
>
>
> On 20 May 2015 at 15:40, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>  But isn't that based on the session-timeout param which defaults to 30
> min?  My re-invites occur much sooner than 30 min into a call.  Or does
> session-timeout param only control sessions initiated by FS while incoming
> sessions use the minimum-session-expires param if it's not explicitly
> passed by the session initiator?
>
>
>
> On Tue, May 19, 2015 at 11:40 PM, Michael Jerris <mike at jerris.com> wrote:
>
> session timer
>
>
>
> On Tuesday, May 19, 2015, Oleg Stolyar <olegstolyar at gmail.com> wrote:
>
>  Thanks Michael, I'll see if we can do that!
>
>
>
> So, is the re-INVITE legit and the problem is that JsSip does not respond
> to it?  Still, I am curious what is triggering the re-INVITE.
>
>
>
> On Tue, May 19, 2015 at 8:05 PM, Michael Jerris <mike at jerris.com> wrote:
>
> I think the sip.js guys fixed this issue when they forked jssip.  I'd
> suggest using that.
>
>
> > On May 19, 2015, at 10:01 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
> >
> > Hi guys,
> >
> > Several weeks ago I started getting an occasional problem where FS is
> sending an INVITE to the other side in the middle of a call, the other side
> does not respond and FS hangs up the leg.  Below is the relevant log.  The
> user experience is that they keep talking and hearing each other up to the
> very end.  I have a recording of that call, so can confirm.
> >
> > The call uses WebRTC and is originated by JsSip from Chrome.  Then the
> user is put into a conference but I doubt it's relevant in this case since
> the INVITE and disconnect are not happening from mod_conference
> >
> > I suspect it's a re-INVITE but what triggers FS to send it?  I couldn't
> find anything in the logs that could shed light.
> >
> > send 1625 bytes to wss/[##.##.##.##]:50292 at 00:00:19.702933:
> >
> ------------------------------------------------------------------------
> >    INVITE sip:dj012n41 at u40rf5qikah5.invalid;transport=ws;ob SIP/2.0
> >    Via: SIP/2.0/WSS 10.97.158.232:5067;branch=z9hG4bK7Xm4tjevU45Sr
> >    Max-Forwards: 70
> >    From: <
> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22 at anonymous.invalid
> >;tag=KQecUSr12rSQp
> >    To: "user1" <sip:anonymous at anonymous.invalid>;tag=v1rlqab64i
> >    Call-ID: g8980rbrbk2t45oj5mru
> >    CSeq: 75703945 INVITE
> >    Contact: <
> sip:answer-c4e14a20-fe82-11e4-95ed-22000b358f22@##.##.###.###:5080;transport=udp
> >
> >    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
> >    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> >    Supported: timer, path, replaces
> >    Session-Expires: 120;refresher=uac
> >    Min-SE: 120
> >    Content-Type: application/sdp
> >    Content-Disposition: session
> >    Content-Length: 825
> >
> >    v=0
> >    o=FreeSWITCH 1432057287 1432057288 IN IP4 ##.##.##.##
> >    s=FreeSWITCH
> >    c=IN IP4 ##.##.##.##
> >    t=0 0
> >    a=msid-semantic: WMS uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
> >    m=audio 22670 RTP/SAVPF 9 126 106
> >    a=rtpmap:9 G722/8000
> >    a=rtpmap:126 telephone-event/8000
> >    a=rtpmap:106 CN/8000
> >    a=ptime:20
> >    a=fingerprint:sha-256
> E4:E2:DD:6C:60:61:69:9D:FD:21:64:79:66:C0:14:58:DD:67:CE:29:35:35:58:65:2E:91:70:85:4C:6C:47:69
> >    a=rtcp-mux
> >    a=rtcp:22670 IN IP4 ##.##.##.##
> >    a=ssrc:1029894069 cname:VL2jTPmLiyFVIEaq
> >    a=ssrc:1029894069 msid:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW a0
> >    a=ssrc:1029894069 mslabel:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hW
> >    a=ssrc:1029894069 label:uZEfxiWc2Dm3Mj8pRAdC8BQDwtTzC6hWa0
> >    a=ice-ufrag:5dS3Fzx1Thrmdy9Z
> >    a=ice-pwd:a19UHlvPK1BjvBzrFilbII2s
> >    a=candidate:3876535948 1 udp 659136 107.20.175.160 22670 typ host
> generation 0
> >
> ------------------------------------------------------------------------
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
> switch_core_session.c:1061 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:19.376858 [DEBUG]
> sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
> entering state [calling][0]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_core_session.c:1061 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_core_session.c:1061 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> sofia.c:6627 Channel sofia/leia_agent/anonymous at anonymous.invalid
> entering state [terminating][503]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [NOTICE]
> sofia.c:7543 Hangup sofia/leia_agent/anonymous at anonymous.invalid
> [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_channel.c:3242 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [KILL]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.616874 [DEBUG]
> switch_core_session.c:1396 Send signal
> sofia/leia_agent/anonymous at anonymous.invalid [BREAK]
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> mod_conference.c:5057 Channel leaving conference, cause:
> NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> mod_conference.c:9650 sofia/leia_agent/anonymous at anonymous.invalid skip
> receive message [UNBRIDGE] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_media.c:7772 sofia/leia_agent/anonymous at anonymous.invalid
> skip receive message [REFER_EVENT] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_codec.c:246 sofia/leia_agent/anonymous at anonymous.invalid
> Restore previous codec G722:9.
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_session.c:2901 sofia/leia_agent/anonymous at anonymous.invalid
> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:535 (
> sofia/leia_agent/anonymous at anonymous.invalid) State EXECUTE going to sleep
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:472 (
> sofia/leia_agent/anonymous at anonymous.invalid) Running State Change
> CS_HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:735 (
> sofia/leia_agent/anonymous at anonymous.invalid) Callstate Change ACTIVE ->
> HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:737 (
> sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> mod_sofia.c:413 Channel sofia/leia_agent/anonymous at anonymous.invalid
> hanging up, cause: NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:60
> sofia/leia_agent/anonymous at anonymous.invalid Standard HANGUP, cause:
> NORMAL_TEMPORARY_FAILURE
> > fc1927a8-2257-400f-b4cb-67aff7297170 2015-05-20 00:00:22.636900 [DEBUG]
> switch_core_state_machine.c:737 (
> sofia/leia_agent/anonymous at anonymous.invalid) State HANGUP going to sleep
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 28 01:18:01 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 27 May 2015 23:18:01 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.6+Video ... TLS ports 5061, 5081,
 8082 issues
In-Reply-To: <CACLfcpu=c_jY-ujSt4w7c5U+r8BGQ_AdvkcqtqJaZBLVfVeTtA@mail.gmail.com>
References: <CACLfcpvm1eJBUUjY2Gbt1hEPnv5T6fuePJJkP0tjrviw_4hLkA@mail.gmail.com>
	<D18B8F89.9F9D6%krice@freeswitch.org>
	<CAJR7f2H9T0H93NbGJRH0jhEbR1i1npJ-UMHU5-AMX0PMifCo=Q@mail.gmail.com>
	<CAJR7f2GDAvfVFdto4XkCRq6dbrHQ6yJ-KqTC1Hi1_iPQoRPJAw@mail.gmail.com>
	<CALXCt0qz13R-Pj_5vWRfr__h+7prtLNLAfdnkGErT9BkC5kgVQ@mail.gmail.com>
	<CACLfcpu=c_jY-ujSt4w7c5U+r8BGQ_AdvkcqtqJaZBLVfVeTtA@mail.gmail.com>
Message-ID: <CALXCt0rqoAONm7Bu4E2FunbBzSUx0dbC5jekLeR812QrzF+wsg@mail.gmail.com>

Ok, I'll add tomorrow.

Anyway, any and each times you change vars.xml you must restart freeswitch

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 27, 2015 11:12 PM, "Craig Stevenson" <craig at stevenson.com> wrote:

>
> As a side note -- the current instructions do not say anything about
> starting FreeSWITCH.  It shows restarting Apache and then running the
> openssl tests.
>
>
>
> On Wed, May 27, 2015 at 2:05 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> Those values for vars.xml are listed in the freeswitch1.6 page.
>>
>> Just shutdown and restart freeswitch, but I believe restarting is listed
>> too.
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On May 27, 2015 10:53 PM, "?talo Rossi" <italorossib at gmail.com> wrote:
>>
>>> Also,
>>>
>>> don't forget to set these values in your vars.xml
>>>
>>> On Wed, May 27, 2015 at 5:51 PM, ?talo Rossi <italorossib at gmail.com>
>>> wrote:
>>>
>>>> Craig,
>>>>
>>>> freeswitch at internal> global_setvar internal_ssl_enable=true
>>>> +OK
>>>> freeswitch at internal> global_setvar external_ssl_enable=true
>>>> +OK
>>>> freeswitch at internal> sofia profile external restart reloadxml
>>>> Reload XML [Success]
>>>> restarting: external
>>>> freeswitch at internal> sofia profile internal restart reloadxml
>>>> Reload XML [Success]
>>>> restarting: internal
>>>>
>>>> I'll update the docs with these steps.
>>>>
>>>>
>>>> On Wed, May 27, 2015 at 5:13 PM, Ken Rice <krice at freeswitch.org> wrote:
>>>>
>>>>>  Sounds like FreeSWITCH isnt actually tunning its what listens on the
>>>>> high ports, port 443 is apache in if following that doc
>>>>>
>>>>>
>>>>>
>>>>> On 5/27/15, 3:02 PM, "Craig Stevenson" <craig at stevenson.com> wrote:
>>>>>
>>>>> I'm new at this and hoping someone can help me figure out what I'm
>>>>> missing...
>>>>>
>>>>> I've exactly followed the instructions at:
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video#FreeSWITCH1.6Video-Introduction
>>>>>
>>>>> But, the open_ssl test (test_tls.sh) is only working for 443 and
>>>>> failing with connection refused (errno=111) for port 5061, port 5081 and
>>>>> port 8082.
>>>>>
>>>>> Much 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
>>>>>
>>>>>
>>>>> --
>>>>> Ken
>>>>>
>>>>>
>>>>>
>>>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>>> Twitter: @FreeSWITCH
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>
>>>
>>>
>>> --
>>> ?talo Rossi
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From emplant2000 at gmail.com  Thu May 28 03:56:10 2015
From: emplant2000 at gmail.com (Masakazu Nakano)
Date: Thu, 28 May 2015 08:56:10 +0900
Subject: [Freeswitch-users] newbie: db setting for mysql
Message-ID: <CAPwL2MDuLWBcxFbLVyFeU9yJk_3Vy4KAmS8ssGSwx2OqvMB8=g@mail.gmail.com>

Dear sir.

is that good or not following like that ?
root at HOST:/opt/freeswitch/conf# grep -r odbc *
autoload_configs/directory.conf.xml:<param name="core-db-dsn"
value="odbc://freeswitch:user:pass"/>
autoload_configs/voicemail.conf.xml:<param name="odbc-dsn"
value="odbc://freeswitch:user:pass" />
autoload_configs/cidlookup.conf.xml:    <param name="odbc-dsn"
value="phone:phone:phone"/>
autoload_configs/db.conf.xml:<param name="core-db-dsn"
value="odbc://freeswitch:user:pass"/>
autoload_configs/callcenter.conf.xml:<param name="odbc-dsn"
value="odbc://freeswitch:user:pass" />
autoload_configs/nibblebill.conf.xml:    <param name="odbc-dsn" value="
bandwidth.com"/>
autoload_configs/switch.conf.xml: <param name="core-db-dsn"
value="odbc://freeswitch:user:pass" />
autoload_configs/easyroute.conf.xml:    <param name="odbc-retries"
value="120"/>
autoload_configs/lcr.conf.xml:<param name="odbc-dsn"
value="odbc://freeswitch:user:pass" />
jingle_profiles/server.xml:<param name="odbc-dsn"
value="odbc://freeswitch:user:pass" />
sip_profiles/internal-ipv6.xml:<param name="odbc-dsn"
value="odbc://freeswitch:user:pass" />
sip_profiles/internal.xml:<param name="odbc-dsn"
value="odbc://freeswitch:user:pass" />
skinny_profiles/internal.xml:    <param name="odbc-dsn" value=""/>
root at HOST:/opt/freeswitch/conf#


 I'm using fusionpbx.
How can I check the db connection is ok to fs_cli?

'freeswitch at internal> db count' says 0 and db exists to be !Err!

BR

mack

-- 
--- keep it bass,keep it drum.
ubuntu server 14.04 + nginx + mysql + ISPConfig3 + dovecot + postfix +
bingbluebutton + Freeswitch + FusionPBX + moodle ( + Blender + Cinelerra +
Synfig )
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From jonlederman at gmail.com  Thu May 28 03:57:39 2015
From: jonlederman at gmail.com (Jon Lederman)
Date: Wed, 27 May 2015 19:57:39 -0400
Subject: [Freeswitch-users] Mono To Stereo
Message-ID: <89F31D33-F1B9-45B3-B30A-9B0E776FC6A2@gmail.com>

Hi,

We need to set up calls in FS where a mono signal coming into Freeswitch is split into a stereo signal.  What is the easiest way to accomplish this?  

Thanks



Sent from my iPhone

From mike at jerris.com  Thu May 28 04:29:12 2015
From: mike at jerris.com (Michael Jerris)
Date: Wed, 27 May 2015 20:29:12 -0400
Subject: [Freeswitch-users] Mono To Stereo
In-Reply-To: <89F31D33-F1B9-45B3-B30A-9B0E776FC6A2@gmail.com>
References: <89F31D33-F1B9-45B3-B30A-9B0E776FC6A2@gmail.com>
Message-ID: <CAEwTsAj2iYPByo4u2CqtH=SeuX0nGdno0rWVdsqJqnzL10S-nw@mail.gmail.com>

Negotiate a call between a mono and stereo endpoint and it will do this.

On Wednesday, May 27, 2015, Jon Lederman <jonlederman at gmail.com> wrote:

> Hi,
>
> We need to set up calls in FS where a mono signal coming into Freeswitch
> is split into a stereo signal.  What is the easiest way to accomplish this?
>
> Thanks
>
>
>
> Sent from my iPhone
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From craig at stevenson.com  Thu May 28 05:05:01 2015
From: craig at stevenson.com (Craig Stevenson)
Date: Wed, 27 May 2015 18:05:01 -0700
Subject: [Freeswitch-users] Documentation for 1.6+Video build
Message-ID: <CACLfcpswTBxXGxxqKg_-U=kOrhSu4TBDVR55Ra-PcaYEir-rkg@mail.gmail.com>

As a novice trying to follow the documentation for creating FreeSWITCH
1.6+Video instance, I have a few questions and have noticed a few gaps in
the documentation.  For experienced users, these gaps are the "obvious"
steps... but not so obvious to those of us just getting started...


For consistency with the Wheezy 1.4 instructions, should the instructions
have users clone the code into /usr/local/src/freeswitch?  If so, modify
the install instructions to 'cd /usr/local/src' and clone the software into
'freeswitch' instead of 'freeswitch.git'

# then let's get the source

cd /usr/local/src

git clone https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch
cd freeswitch.git
git checkout fs-video2
./bootstrap.sh -j
./configure -C

I'm curious why there is a change in the recommended parameters for
./configure

Wheezy 1.4:  ./configure --enable-core-pgsql-support

Jessie 1.6: ./configure -C


Should the 1.6+Video install instructions page replicate (or at least
reference) create the freeswitch user?

Should the 1.6+Video install instructions page replicate (or at least
reference) how to setup FreeSWITCH for automatic start.

The 1.6+Video page has a section on setting up Bridging from WebRTC that
shows adding an extension to Dialplan.  For the novice user, it might help
to A) mention the exact file (e.g.
/usr/local/freeswitch/conf/dialplan/default.xml and B) mention if there is
a specific place in the file where that text should go (or is it OK to just
add it to the end of the file?

Again, I know some of my questions above are painfully obvious to most of
you.  But they are not obvious to the new person trying to get started.  I
mention these here in the hope of helping others who are also just getting
started.

Thanks,
Craig
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From avi at avimarcus.net  Thu May 28 07:57:46 2015
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 28 May 2015 03:57:46 +0000
Subject: [Freeswitch-users] Routing question
In-Reply-To: <5565EC5C.3080707@dwide.com>
References: <5565EC5C.3080707@dwide.com>
Message-ID: <0000014d98ab34c2-46f7bf92-720a-4257-adfa-629fc78e4284-000000@email.amazonses.com>

I believe you want enterprise originate :_:

${call1}:_:${call2} -- will call both 1 and 2 at the same time and the
first to answer takes the call.

In your example, with fail overs, it looks like:

originate sofia/gateway/provider_1/919246461001|sofia/gateway/
provider_2/919246461001:_:sofia/gateway/provider_1/
919246461002|sofia/gateway/provider_2/919246461002 5000

... using it in an originate command I don't know for sure if it will work,
I've only used it in a bridge command.


-Avi

On Wed, May 27, 2015 at 7:10 PM, Dmitry Mordovin <dm at dwide.com> wrote:

> Hi all,
>
> For sample:
> - Have two voip gateways: provider_1 and provider_2
> - Need originate multiple destinations at the same time, protected
> through each gateway
>
> So, originate would be like:
>
> originate
>
> (sofia/gateway/provider_1/919246461001|sofia/gateway/provider_2/919246461001),(sofia/gateway/provider_1/919246461002|sofia/gateway/provider_2/919246461002)
> 5000
>
> Explain.
>
> Originate calls to 919246461001 and 919246461002 parallel through
> provider_1, if provider_1 failed, try provider_2
>
> Does any one did 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
>
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From jack at livecall.com  Thu May 28 08:09:57 2015
From: jack at livecall.com (Jack)
Date: Wed, 27 May 2015 21:09:57 -0700
Subject: [Freeswitch-users] Verto on Windows?
Message-ID: <55669515.4010700@livecall.com>

Has anyone been successful at setting up the html5 verto page with 
windows IIS running on one server and Freeswitch running on a different 
windows server?





From covici at ccs.covici.com  Thu May 28 08:50:03 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 28 May 2015 00:50:03 -0400
Subject: [Freeswitch-users] problem sending to dtmf to some sites
Message-ID: <28414.1432788603@ccs.covici.com>

Hi.  I call lots of numbers which are conference lines and they require
various dtmf codes after you dial the line.  Now my problem is that some
of them will not work unless you send the dtmf with some pauses in
between each digit.  Is there any way freeswitch can do this --
i.e. mod_conference can outcall and send dtmf after the call is
answered.

Thanks in advance for any assistance.

-- 
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 yehavi.bourvine at gmail.com  Thu May 28 10:20:30 2015
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 28 May 2015 09:20:30 +0300
Subject: [Freeswitch-users] Some calls have one-way audio
Message-ID: <CAAtWKeHoqAq_tdEkyAw26Rsr+8Y_8R8H1mkD7SEQiU1+e_ffnQ@mail.gmail.com>

Hi,

  Before I dig into this issue and start collecting data in order to open a
Jira: Has anyone else noticed it? When I call or get a call sometimes there
is one way audio. Hanging up and calling again usually solves it. It
happens quite rarely, so I cannot reproduce it at will. However, it annoys
the users...

I am running FreeSwitch 1.4.18 and use various phones (Polycom, Yealink,
SNOM, etc.).

                  Thanks! __Yehavi:
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From gmaruzz at gmail.com  Thu May 28 10:30:09 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 May 2015 08:30:09 +0200
Subject: [Freeswitch-users] Some calls have one-way audio
In-Reply-To: <CAAtWKeHoqAq_tdEkyAw26Rsr+8Y_8R8H1mkD7SEQiU1+e_ffnQ@mail.gmail.com>
References: <CAAtWKeHoqAq_tdEkyAw26Rsr+8Y_8R8H1mkD7SEQiU1+e_ffnQ@mail.gmail.com>
Message-ID: <CALXCt0rqN_kp4JE_YokbxiNVYW4OGcPCnjzhrujB8OETTVVVLw@mail.gmail.com>

Hello,

is almost certainly a NAT problem.
Use something like pcapsipdump, take trace of all your calls, then when a
call with one-way audio happens, look into its pcap.

-giovanni


On Thu, May 28, 2015 at 8:20 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
wrote:

> Hi,
>
>   Before I dig into this issue and start collecting data in order to open
> a Jira: Has anyone else noticed it? When I call or get a call sometimes
> there is one way audio. Hanging up and calling again usually solves it. It
> happens quite rarely, so I cannot reproduce it at will. However, it annoys
> the users...
>
> I am running FreeSwitch 1.4.18 and use various phones (Polycom, Yealink,
> SNOM, etc.).
>
>                   Thanks! __Yehavi:
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 osblinnikov at gmail.com  Thu May 28 12:05:19 2015
From: osblinnikov at gmail.com (Oleg Blinnikov)
Date: Thu, 28 May 2015 10:05:19 +0200
Subject: [Freeswitch-users] compiling FreeSWITCH fs-video2 branch on
 debian jessie
In-Reply-To: <D156873B.9B0FD%krice@freeswitch.org>
References: <CAJwe+CBjCcXNssgntr3xSNdbmfURhjQYt7SCZkc3AW482rTcHQ@mail.gmail.com>
	<D156873B.9B0FD%krice@freeswitch.org>
Message-ID: <CA+M05dAD=HsD9UhhSBgatWTwAFOtqPhxucd=anV3cqycyVr_MQ@mail.gmail.com>

I also run into this problem, but I installed yuv with cmake install target:

  git clone http://git.chromium.org/external/libyuv.git
  cd libyuv
mkdir out cd out cmake -DCMAKE_INSTALL_PREFIX="/usr/lib"
-DCMAKE_BUILD_TYPE="Release" .. cmake --build . --config Release sudo cmake
--build . --target install --config Release

May be someone knows what do I do wrong?

On Fri, Apr 17, 2015 at 4:47 PM, Ken Rice <krice at freeswitch.org> wrote:

> We actually had to patch several things for this... We'll be making those
> things available shortly
>
>
> On 4/17/15, 9:42 AM, "E. Schmidbauer" <eschmidbauer at gmail.com> wrote:
>
> > Hi Sergey,
> > The make file (linux.mk) does not have an "install" action so that
> > will not work.
> > Thanks,
> > E
> >
> > On Fri, Apr 17, 2015 at 10:27 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
> >> try execute "make install" from libyuv folter. Then repeat FS compiling.
> >>
> >> Sergey
> >>
> >> On Fri, Apr 17, 2015 at 4:24 PM, E. Schmidbauer <eschmidbauer at gmail.com
> >
> >> wrote:
> >>>
> >>> Hello,
> >>> I'm trying to compile fs-video2 and getting the following error during
> >>> ./configure
> >>>
> >>> checking for libyuv >= 0.0.1280... configure: error: You need to
> >>> install libyuv-dev. Required library
> >>>
> >>>
> >>> I git cloned and compiled the libyuv library FS is looking for:
> >>>
> >>> git clone https://github.com/openpeer/libyuv.git
> >>> cd libyuv
> >>> make -j7 V=1 -f linux.mk
> >>>
> >>> (it compiles without errors)
> >>>
> >>> How can I tell FS where to find the library?
> >>>
> >>> Thanks,
> >>> 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
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Oleg Blinnikov
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From s.safarov at gmail.com  Thu May 28 14:02:11 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 28 May 2015 13:02:11 +0300
Subject: [Freeswitch-users] compiling FreeSWITCH fs-video2 branch on
 debian jessie
In-Reply-To: <CA+M05dAD=HsD9UhhSBgatWTwAFOtqPhxucd=anV3cqycyVr_MQ@mail.gmail.com>
References: <CAJwe+CBjCcXNssgntr3xSNdbmfURhjQYt7SCZkc3AW482rTcHQ@mail.gmail.com>
	<D156873B.9B0FD%krice@freeswitch.org>
	<CA+M05dAD=HsD9UhhSBgatWTwAFOtqPhxucd=anV3cqycyVr_MQ@mail.gmail.com>
Message-ID: <CAHtxdDcs-P+GJd-6AM5Nvch_1_8FN1B=YmpTPB=GCEWT2c-TMQ@mail.gmail.com>

When I create yum spec file I has diabled find jpeg library.

It requered because FS config file is not searching jpeg library

In SPEC file I added command
sed -i -e "s:STATIC:SHARED:" -e "s:\(include.FindJPEG.\):#\1:"
CMakeLists.txt

https://freeswitch.org/stash/projects/SD/repos/libyuv/browse/libyuv.spec


On Thu, May 28, 2015 at 11:05 AM, Oleg Blinnikov <osblinnikov at gmail.com>
wrote:

> I also run into this problem, but I installed yuv with cmake install
> target:
>
>   git clone http://git.chromium.org/external/libyuv.git
>   cd libyuv
> mkdir out cd out cmake -DCMAKE_INSTALL_PREFIX="/usr/lib"
> -DCMAKE_BUILD_TYPE="Release" .. cmake --build . --config Release sudo cmake
> --build . --target install --config Release
>
> May be someone knows what do I do wrong?
>
> On Fri, Apr 17, 2015 at 4:47 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> We actually had to patch several things for this... We'll be making those
>> things available shortly
>>
>>
>> On 4/17/15, 9:42 AM, "E. Schmidbauer" <eschmidbauer at gmail.com> wrote:
>>
>> > Hi Sergey,
>> > The make file (linux.mk) does not have an "install" action so that
>> > will not work.
>> > Thanks,
>> > E
>> >
>> > On Fri, Apr 17, 2015 at 10:27 AM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>> >> try execute "make install" from libyuv folter. Then repeat FS
>> compiling.
>> >>
>> >> Sergey
>> >>
>> >> On Fri, Apr 17, 2015 at 4:24 PM, E. Schmidbauer <
>> eschmidbauer at gmail.com>
>> >> wrote:
>> >>>
>> >>> Hello,
>> >>> I'm trying to compile fs-video2 and getting the following error during
>> >>> ./configure
>> >>>
>> >>> checking for libyuv >= 0.0.1280... configure: error: You need to
>> >>> install libyuv-dev. Required library
>> >>>
>> >>>
>> >>> I git cloned and compiled the libyuv library FS is looking for:
>> >>>
>> >>> git clone https://github.com/openpeer/libyuv.git
>> >>> cd libyuv
>> >>> make -j7 V=1 -f linux.mk
>> >>>
>> >>> (it compiles without errors)
>> >>>
>> >>> How can I tell FS where to find the library?
>> >>>
>> >>> Thanks,
>> >>> 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
>>
>> --
>> Ken
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> irc.freenode.net #freeswitch
>> Twitter: @FreeSWITCH
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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,
> Oleg Blinnikov
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 28 14:14:40 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 May 2015 12:14:40 +0200
Subject: [Freeswitch-users] compiling FreeSWITCH fs-video2 branch on
 debian jessie
In-Reply-To: <CAHtxdDcs-P+GJd-6AM5Nvch_1_8FN1B=YmpTPB=GCEWT2c-TMQ@mail.gmail.com>
References: <CAJwe+CBjCcXNssgntr3xSNdbmfURhjQYt7SCZkc3AW482rTcHQ@mail.gmail.com>
	<D156873B.9B0FD%krice@freeswitch.org>
	<CA+M05dAD=HsD9UhhSBgatWTwAFOtqPhxucd=anV3cqycyVr_MQ@mail.gmail.com>
	<CAHtxdDcs-P+GJd-6AM5Nvch_1_8FN1B=YmpTPB=GCEWT2c-TMQ@mail.gmail.com>
Message-ID: <CALXCt0oO84V7iice-D6FQ51T6mQ4sacgGjjPdppOhkaM-=_fuA@mail.gmail.com>

be patient, until merged into 1.6, fs_video2 is an experimental branch,
that goes through continous development that often breaks the building
process



On Thu, May 28, 2015 at 12:02 PM, Sergey Safarov <s.safarov at gmail.com>
wrote:

> When I create yum spec file I has diabled find jpeg library.
>
> It requered because FS config file is not searching jpeg library
>
> In SPEC file I added command
> sed -i -e "s:STATIC:SHARED:" -e "s:\(include.FindJPEG.\):#\1:"
> CMakeLists.txt
>
> https://freeswitch.org/stash/projects/SD/repos/libyuv/browse/libyuv.spec
>
>
> On Thu, May 28, 2015 at 11:05 AM, Oleg Blinnikov <osblinnikov at gmail.com>
> wrote:
>
>> I also run into this problem, but I installed yuv with cmake install
>> target:
>>
>>   git clone http://git.chromium.org/external/libyuv.git
>>   cd libyuv
>> mkdir out cd out cmake -DCMAKE_INSTALL_PREFIX="/usr/lib"
>> -DCMAKE_BUILD_TYPE="Release" .. cmake --build . --config Release sudo cmake
>> --build . --target install --config Release
>>
>> May be someone knows what do I do wrong?
>>
>> On Fri, Apr 17, 2015 at 4:47 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>>> We actually had to patch several things for this... We'll be making those
>>> things available shortly
>>>
>>>
>>> On 4/17/15, 9:42 AM, "E. Schmidbauer" <eschmidbauer at gmail.com> wrote:
>>>
>>> > Hi Sergey,
>>> > The make file (linux.mk) does not have an "install" action so that
>>> > will not work.
>>> > Thanks,
>>> > E
>>> >
>>> > On Fri, Apr 17, 2015 at 10:27 AM, Sergey Safarov <s.safarov at gmail.com>
>>> wrote:
>>> >> try execute "make install" from libyuv folter. Then repeat FS
>>> compiling.
>>> >>
>>> >> Sergey
>>> >>
>>> >> On Fri, Apr 17, 2015 at 4:24 PM, E. Schmidbauer <
>>> eschmidbauer at gmail.com>
>>> >> wrote:
>>> >>>
>>> >>> Hello,
>>> >>> I'm trying to compile fs-video2 and getting the following error
>>> during
>>> >>> ./configure
>>> >>>
>>> >>> checking for libyuv >= 0.0.1280... configure: error: You need to
>>> >>> install libyuv-dev. Required library
>>> >>>
>>> >>>
>>> >>> I git cloned and compiled the libyuv library FS is looking for:
>>> >>>
>>> >>> git clone https://github.com/openpeer/libyuv.git
>>> >>> cd libyuv
>>> >>> make -j7 V=1 -f linux.mk
>>> >>>
>>> >>> (it compiles without errors)
>>> >>>
>>> >>> How can I tell FS where to find the library?
>>> >>>
>>> >>> Thanks,
>>> >>> 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
>>>
>>> --
>>> Ken
>>> http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>> irc.freenode.net #freeswitch
>>> Twitter: @FreeSWITCH
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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,
>> Oleg Blinnikov
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mkvonarx at gmail.com  Thu May 28 16:08:54 2015
From: mkvonarx at gmail.com (Markus von Arx)
Date: Thu, 28 May 2015 14:08:54 +0200
Subject: [Freeswitch-users] SIP messages from ports 10000+
Message-ID: <CAHA5ALevvGuUskvVdJybt9n71CkcUpe=BwG9uWii4D2axHYoxg@mail.gmail.com>

Hallo

Our customers are complaining that their FreeSWITCH instances are sometimes
sending SIP messages from another port than the configured 5060 UDP port.
They observe some SIP messages coming from ports in the range 10000 - 10005.

I could verify this in Wireshark traces from our customers. The behavior
seems to be quite sporadic. In one instance, I see that the FreeSWITCH
sends an INVITE from port 10000, then resends the same INVITE 1 second
later (normal SIP resend/retry), but this time from port 5060. Strange.

Some logs from their firewall:
000066: May 26 2015 17:36:59.874 CEDT: %SEC-6-IPACCESSLOGP: list ACL-WAN-IN
denied udp 10.193.97.34(5060) -> 10.193.109.34(10001), 1 packet
?
000166: May 26 2015 17:46:18.115 CEDT: %SEC-6-IPACCESSLOGP: list ACL-WAN-IN
denied udp 10.193.97.34(5060) -> 10.193.109.34(10003), 1 packet
000167: May 26 2015 17:46:51.521 CEDT: %SEC-6-IPACCESSLOGP: list ACL-WAN-IN
denied udp 10.193.97.34(5060) -> 10.193.109.34(10000), 16 packets
000168: May 26 2015 17:47:51.520 CEDT: %SEC-6-IPACCESSLOGP: list ACL-WAN-IN
denied udp 10.193.97.34(5060) -> 10.193.109.34(10001), 17 packets
000169: May 26 2015 17:49:51.519 CEDT: %SEC-6-IPACCESSLOGP: list ACL-WAN-IN
denied udp 10.193.97.34(5060) -> 10.193.109.34(10002), 16 packets

=> Any idea what might cause this behavior? Maybe some configuration that
we haven't set correctly? Or maybe even a (known) FreeSWITCH bug?

Btw: using FreeSWITCH 1.4.15, 64bit, running on Windows Server.

Thanks for any help and best regards,
Markus
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From yehavi.bourvine at gmail.com  Thu May 28 16:25:42 2015
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 28 May 2015 15:25:42 +0300
Subject: [Freeswitch-users] Some calls have one-way audio
In-Reply-To: <CALXCt0rqN_kp4JE_YokbxiNVYW4OGcPCnjzhrujB8OETTVVVLw@mail.gmail.com>
References: <CAAtWKeHoqAq_tdEkyAw26Rsr+8Y_8R8H1mkD7SEQiU1+e_ffnQ@mail.gmail.com>
	<CALXCt0rqN_kp4JE_YokbxiNVYW4OGcPCnjzhrujB8OETTVVVLw@mail.gmail.com>
Message-ID: <CAAtWKeGEEOBbi4xOm5uEr3kg8GSptsG0iN=-cm9EA-A8xXbmQg@mail.gmail.com>

Hi,

  We are not using NAT. All the voice goes through the Freeswitch server,
and it relays the audio between the internal phones and outside.
The problem is that the outside leg is using SRTP, so I cannot hear the
voice there...

                 Thanks, __Yehavi:

2015-05-28 9:30 GMT+03:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:

> Hello,
>
> is almost certainly a NAT problem.
> Use something like pcapsipdump, take trace of all your calls, then when a
> call with one-way audio happens, look into its pcap.
>
> -giovanni
>
>
> On Thu, May 28, 2015 at 8:20 AM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>> Hi,
>>
>>   Before I dig into this issue and start collecting data in order to open
>> a Jira: Has anyone else noticed it? When I call or get a call sometimes
>> there is one way audio. Hanging up and calling again usually solves it. It
>> happens quite rarely, so I cannot reproduce it at will. However, it annoys
>> the users...
>>
>> I am running FreeSwitch 1.4.18 and use various phones (Polycom, Yealink,
>> SNOM, etc.).
>>
>>                   Thanks! __Yehavi:
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at gmail.com  Thu May 28 16:30:46 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 May 2015 14:30:46 +0200
Subject: [Freeswitch-users] Some calls have one-way audio
In-Reply-To: <CAAtWKeGEEOBbi4xOm5uEr3kg8GSptsG0iN=-cm9EA-A8xXbmQg@mail.gmail.com>
References: <CAAtWKeHoqAq_tdEkyAw26Rsr+8Y_8R8H1mkD7SEQiU1+e_ffnQ@mail.gmail.com>
	<CALXCt0rqN_kp4JE_YokbxiNVYW4OGcPCnjzhrujB8OETTVVVLw@mail.gmail.com>
	<CAAtWKeGEEOBbi4xOm5uEr3kg8GSptsG0iN=-cm9EA-A8xXbmQg@mail.gmail.com>
Message-ID: <CALXCt0rKtkqjE-UmuqPDkKVU=N0Bq_p_yhiciXGJ=E9SdhVfjg@mail.gmail.com>

Yehavi,

your description is not clear at all, at least to me.

Can you explain exactly what happens, what's your configuration, etc?

Also, I'm almost sure you will end up having to provide us with sip traces
of failed (one way) calls, so I would counseil you to begin using
pcapsipdump

-giovanni


On Thu, May 28, 2015 at 2:25 PM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
wrote:

> Hi,
>
>   We are not using NAT. All the voice goes through the Freeswitch server,
> and it relays the audio between the internal phones and outside.
> The problem is that the outside leg is using SRTP, so I cannot hear the
> voice there...
>
>                  Thanks, __Yehavi:
>
> 2015-05-28 9:30 GMT+03:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>
>> Hello,
>>
>> is almost certainly a NAT problem.
>> Use something like pcapsipdump, take trace of all your calls, then when a
>> call with one-way audio happens, look into its pcap.
>>
>> -giovanni
>>
>>
>> On Thu, May 28, 2015 at 8:20 AM, Yehavi Bourvine <
>> yehavi.bourvine at gmail.com> wrote:
>>
>>> Hi,
>>>
>>>   Before I dig into this issue and start collecting data in order to
>>> open a Jira: Has anyone else noticed it? When I call or get a call
>>> sometimes there is one way audio. Hanging up and calling again usually
>>> solves it. It happens quite rarely, so I cannot reproduce it at will.
>>> However, it annoys the users...
>>>
>>> I am running FreeSwitch 1.4.18 and use various phones (Polycom, Yealink,
>>> SNOM, etc.).
>>>
>>>                   Thanks! __Yehavi:
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From s.safarov at gmail.com  Thu May 28 16:54:03 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 28 May 2015 15:54:03 +0300
Subject: [Freeswitch-users] SIP messages from ports 10000+
In-Reply-To: <CAHA5ALevvGuUskvVdJybt9n71CkcUpe=BwG9uWii4D2axHYoxg@mail.gmail.com>
References: <CAHA5ALevvGuUskvVdJybt9n71CkcUpe=BwG9uWii4D2axHYoxg@mail.gmail.com>
Message-ID: <CAHtxdDeyf83ud+w2qdWYwRV8-dEitmm8JNnwbuhCoQUVvkLnbw@mail.gmail.com>

It may be result of wrong SIP messages send by client example in
https://freeswitch.org/jira/browse/FS-7530

On Thu, May 28, 2015 at 3:08 PM, Markus von Arx <mkvonarx at gmail.com> wrote:

> Hallo
>
> Our customers are complaining that their FreeSWITCH instances are
> sometimes sending SIP messages from another port than the configured 5060
> UDP port. They observe some SIP messages coming from ports in the range
> 10000 - 10005.
>
> I could verify this in Wireshark traces from our customers. The behavior
> seems to be quite sporadic. In one instance, I see that the FreeSWITCH
> sends an INVITE from port 10000, then resends the same INVITE 1 second
> later (normal SIP resend/retry), but this time from port 5060. Strange.
>
> Some logs from their firewall:
> 000066: May 26 2015 17:36:59.874 CEDT: %SEC-6-IPACCESSLOGP: list
> ACL-WAN-IN denied udp 10.193.97.34(5060) -> 10.193.109.34(10001), 1 packet
> ?
> 000166: May 26 2015 17:46:18.115 CEDT: %SEC-6-IPACCESSLOGP: list
> ACL-WAN-IN denied udp 10.193.97.34(5060) -> 10.193.109.34(10003), 1 packet
> 000167: May 26 2015 17:46:51.521 CEDT: %SEC-6-IPACCESSLOGP: list
> ACL-WAN-IN denied udp 10.193.97.34(5060) -> 10.193.109.34(10000), 16 packets
> 000168: May 26 2015 17:47:51.520 CEDT: %SEC-6-IPACCESSLOGP: list
> ACL-WAN-IN denied udp 10.193.97.34(5060) -> 10.193.109.34(10001), 17 packets
> 000169: May 26 2015 17:49:51.519 CEDT: %SEC-6-IPACCESSLOGP: list
> ACL-WAN-IN denied udp 10.193.97.34(5060) -> 10.193.109.34(10002), 16 packets
>
> => Any idea what might cause this behavior? Maybe some configuration that
> we haven't set correctly? Or maybe even a (known) FreeSWITCH bug?
>
> Btw: using FreeSWITCH 1.4.15, 64bit, running on Windows Server.
>
> Thanks for any help and best regards,
> Markus
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From s.safarov at gmail.com  Thu May 28 16:57:17 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 28 May 2015 15:57:17 +0300
Subject: [Freeswitch-users] Whar is mean "ending bridge by request from
	write function"
Message-ID: <CAHtxdDcT6PjM3-sP_Oqv51vqb-mxiemeD23XDxLykRK4eGbWfg@mail.gmail.com>

Some calls is droped during a conversation
What is means "[DEBUG] switch_ivr_bridge.c:579 sofia/internal/
1246 at client-1424.rcsnet.ru ending bridge by request from write function"?

In example message at 11:01:05
https://pastebin.freeswitch.org/24241
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From yadenis at seznam.cz  Thu May 28 17:40:56 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Thu, 28 May 2015 15:40:56 +0200
Subject: [Freeswitch-users] Problem with MP4 file record
In-Reply-To: <CAEwTsAicBL-TmZWLoTTyPJtttPCrXQ7JafYSPbhX_QRE8fr7mA@mail.gmail.com>
References: <43162995.20150526094002@seznam.cz> 
	<CAEwTsAhSMJX7sX3Jjf9fbF3G3sfFq7k7vDcH+UH4WAbfOTr7hA@mail.gmail.com>
	<16210177989.20150526120710@seznam.cz>
	<CAHtxdDd=vo6uwo-7imdo7MEGuRDC2PYbqk3HyLvvD1rdfkZxRw@mail.gmail.com>
	<1946256680.20150526171049@seznam.cz>
	<01CF76A0-3760-446E-9A5F-EC5025163004@jerris.com>
	<CAHtxdDfiiW0yE+QdZXZ=YZc4p355mdTJ-BF8TpKFBCB8uL6nQQ@mail.gmail.com>
	<CAEwTsAicBL-TmZWLoTTyPJtttPCrXQ7JafYSPbhX_QRE8fr7mA@mail.gmail.com>
Message-ID: <1425491988.20150528154056@seznam.cz>

Hi all,

Finally mod_av I have a system freeswitch. But still do not want to write MP4

4f98c0d7-5e8c-4e16-a8d4-47509d8e8f75 2015-05-28 09:37:26.693684 [DEBUG] switch_rtp.c:1778 rtcp_stats_init: audio ssrc[2716661911] base_seq[17278]
4f98c0d7-5e8c-4e16-a8d4-47509d8e8f75 EXECUTE sofia/internal/1004 at 192.168.242.132 record(/usr/local/freeswitch/recordings/20150528_093729_1004_9193.mp4)
2015-05-28 09:37:29.574626 [INFO] avformat.c:1363 init timer
2015-05-28 09:37:29.574626 [DEBUG] avformat.c:1380 sample rate: 48000, channels: 1
2015-05-28 09:37:29.574626 [INFO] avformat.c:1481 use video codec: [28] h264 (H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10)
2015-05-28 09:37:29.574626 [ERR] avformat.c:384 Could not open audio codec: Experimental feature
4f98c0d7-5e8c-4e16-a8d4-47509d8e8f75 2015-05-28 09:37:29.574626 [NOTICE] switch_ivr_play_say.c:544 Hangup sofia/internal/1004 at 192.168.242.132 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]


what am I doing wrong?

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

?ter? 26. kv?tna 2015, 18:41:11, napsal jste:


As I have said before.  Do NOT do that.  It will lead to a broken system.

On Tuesday, May 26, 2015, Sergey Safarov <s.safarov at gmail.com> wrote:
For Michael
Denis has installed server from my repo (FS-7553) and libvpx 1.4.0 has not correct libnames (more info below). Prevision recommendation taken to workaround it.

[root at localhost ~]# rpm -qa | grep libvpx
libvpx-devel-1.4.0-6.el7.centos.x86_64
libvpx-1.4.0-6.el7.centos.x86_64

[root at localhost ~]# ls -l /usr/lib64/libvpx.so*
lrwxrwxrwx. 1 root root      15 May 16 01:48 /usr/lib64/libvpx.so -> libvpx.so.2.0.0
lrwxrwxrwx. 1 root root      15 May 16 01:48 /usr/lib64/libvpx.so.2 -> libvpx.so.2.0.0
lrwxrwxrwx. 1 root root      15 May 16 01:48 /usr/lib64/libvpx.so.2.0 -> libvpx.so.2.0.0
-rwxr-xr-x. 1 root root 1548944 May 16 01:43 /usr/lib64/libvpx.so.2.0.0



On Tue, May 26, 2015 at 7:07 PM, Michael Jerris <mike at jerris.com> wrote:
you should NOT be doing that.  That will 100% for sure crete a broken system,  Any other libs that link to freeswitch will need to be linked against the updated vpx lib.


On May 26, 2015, at 11:10 AM, Denis Jakovlev <yadenis at seznam.cz> wrote:

Dobr? den,

ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1
ln: failed to create symbolic link ?/usr/lib64/libvpx.so.1?: No such file or directory


I have libvpx.so in /usr/x_86_64_linux_gnu. But it steal not work. 

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

?ter? 26. kv?tna 2015, 14:57:11, napsal jste:


execute as root "ln -s libvpx.so.2.0.0 /usr/lib64/libvpx.so.1"
And try again

On Tue, May 26, 2015 at 1:07 PM, Denis Jakovlev <yadenis at seznam.cz> wrote:
Dobr? den,

Where can I get the -extra package for libav?

I try from here https://freeswitch.org/stash/projects/SD/repos/libav/browse  It is compiled without problems. 

But I still have a error

2015-05-26 06:01:38.083643 [CRIT] switch_loadable_module.c:1520 Error Loading module /usr/local/freeswitch/mod/mod_av.so
**libvpx.so.1: cannot open shared object file: No such file or directory**




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

?ter? 26. kv?tna 2015, 11:22:35, napsal jste:


try using mod av instead.  You'll need to install the -extra package for libav 

On Tuesday, May 26, 2015, Denis Jakovlev <yadenis at seznam.cz> wrote:
Hi all !

I have one problem with the MP4 file recording with sound. The video is great. But there is no sound.

Adding a log. files in place, but for some reason does not work. I use Debian and the latest version 1.6

EXECUTE sofia/internal/1004 at 192.168.242.132 record(/usr/local/freeswitch/recordings/testrecord.mp4)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/access/libavio_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/demux/libavformat_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/stream_out/libstream_out_chromaprint_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/packetizer/libpacketizer_avparser_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libavcodec_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
[00007f16f00c7fc8] core libvlc warning: cannot load module `/usr/lib/vlc/plugins/codec/libhwdummy_plugin.so' (libvpx.so.1: cannot open shared object file: No such file or directory)
2015-05-26 03:24:54.095101 [DEBUG] mod_vlc.c:832 VLC attempt to open #transcode{vcodec=h264,acodec=aac}:std{access=file,mux=mp4,dst=/usr/local/freeswitch/recordings/testrecord.mp4}  write video



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


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch.org
http://www.freeswitchsolutions.com

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

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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Thu May 28 18:44:20 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 May 2015 09:44:20 -0500
Subject: [Freeswitch-users] Whar is mean "ending bridge by request from
 write function"
In-Reply-To: <CAHtxdDcT6PjM3-sP_Oqv51vqb-mxiemeD23XDxLykRK4eGbWfg@mail.gmail.com>
References: <CAHtxdDcT6PjM3-sP_Oqv51vqb-mxiemeD23XDxLykRK4eGbWfg@mail.gmail.com>
Message-ID: <CAKbxfG_rq2o8uYrF94z8H1LzBbm_K7uezK8wo+Bee4Eq_Q5FLQ@mail.gmail.com>

Not enough info really.

If you had the siptrace and the rest of the logs maybe it would be easier
to tell.
Basically it means when it tried to write the frame it failed.  Most likely
either someone was bind transferring it or FSAPI of uui_transfer was called
on it.


On Thu, May 28, 2015 at 7:57 AM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Some calls is droped during a conversation
> What is means "[DEBUG] switch_ivr_bridge.c:579 sofia/internal/
> 1246 at client-1424.rcsnet.ru ending bridge by request from write function"?
>
> In example message at 11:01:05
> https://pastebin.freeswitch.org/24241
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 switcherfree at gmail.com  Thu May 28 19:13:49 2015
From: switcherfree at gmail.com (Free Switcher)
Date: Thu, 28 May 2015 08:13:49 -0700
Subject: [Freeswitch-users] Updated javascript V8 engine
In-Reply-To: <CAM6Un23UCrRh4RuX-8qLToKNq9kXBnpiqxau=iJjGOQrg-ES+g@mail.gmail.com>
References: <CAM6Un23UCrRh4RuX-8qLToKNq9kXBnpiqxau=iJjGOQrg-ES+g@mail.gmail.com>
Message-ID: <CAM6Un22qWBz61gDaX8kk_UPBGeoGJe7B33C_aRs3_x0tMduuUw@mail.gmail.com>

Does anyone have any thoughts/pointers on using a more recent
JavaScript engine with FS?

On Thu, May 21, 2015 at 12:03 AM, Free Switcher <switcherfree at gmail.com> wrote:
> Hi Guys,
> The version of V8 javascript engine currently packaged with FreeSwitch
> is quite old. Is it possible to build with a more recent version? Any
> pointers appreciated.
>
> Thanks,
> Andy


From mike at jerris.com  Thu May 28 19:15:35 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 28 May 2015 11:15:35 -0400
Subject: [Freeswitch-users] Verto on Windows?
In-Reply-To: <55669515.4010700@livecall.com>
References: <55669515.4010700@livecall.com>
Message-ID: <A7D709E7-3421-4C26-ADF2-E2A56C553122@jerris.com>

The issue setting up verto on 2 different servers comes down to issues with certificates.  I would first try it out on a single linux server to confirm you have all the freeswitch side stuff setup properly... then move to a windows box still with apache, so we can verify windows mod_verto is working, then work on changing to IIS and splitting to multiple servers.  I dont' think you'll hit any major bumps, buf figuring out the right config and how to get the certificates right may take some fiddling.

> On May 28, 2015, at 12:09 AM, Jack <jack at livecall.com> wrote:
> 
> Has anyone been successful at setting up the html5 verto page with 
> windows IIS running on one server and Freeswitch running on a different 
> windows server?
> 
> 



From mike at jerris.com  Thu May 28 19:18:58 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 28 May 2015 11:18:58 -0400
Subject: [Freeswitch-users] problem sending to dtmf to some sites
In-Reply-To: <28414.1432788603@ccs.covici.com>
References: <28414.1432788603@ccs.covici.com>
Message-ID: <DE056E7B-B3BD-475B-8071-A1AC21AAF192@jerris.com>

try using 'w' and 'W' in the DTMF string for a 1/2 and full second pause.

> On May 28, 2015, at 12:50 AM, covici at ccs.covici.com wrote:
> 
> Hi.  I call lots of numbers which are conference lines and they require
> various dtmf codes after you dial the line.  Now my problem is that some
> of them will not work unless you send the dtmf with some pauses in
> between each digit.  Is there any way freeswitch can do this --
> i.e. mod_conference can outcall and send dtmf after the call is
> answered.
> 
> Thanks in advance for any assistance.
> 
> -- 
> 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From s.safarov at gmail.com  Thu May 28 19:19:05 2015
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 28 May 2015 18:19:05 +0300
Subject: [Freeswitch-users] Whar is mean "ending bridge by request from
 write function"
In-Reply-To: <CAKbxfG_rq2o8uYrF94z8H1LzBbm_K7uezK8wo+Bee4Eq_Q5FLQ@mail.gmail.com>
References: <CAHtxdDcT6PjM3-sP_Oqv51vqb-mxiemeD23XDxLykRK4eGbWfg@mail.gmail.com>
	<CAKbxfG_rq2o8uYrF94z8H1LzBbm_K7uezK8wo+Bee4Eq_Q5FLQ@mail.gmail.com>
Message-ID: <CAHtxdDcSzgUZugSwY8f7c+b9cbsVnjaR5EA9nM+K4sBOennSgg@mail.gmail.com>

Thank you Anthony for reply.
On production server sip trace will not help (I will get lot of debug
messages) that will be difficult to analyze.
I has seen source file where message is generated and has analyse FS call
log with sip messages captured by tcpdump. I think it happened when
endpoint received BYE message, close RTP port and send ICMP message for new
received RTP packets. When FS receive ICMP then is close channel with
message in example.
I will be  verify it version and send additional info.


On Thu, May 28, 2015 at 5:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Not enough info really.
>
> If you had the siptrace and the rest of the logs maybe it would be easier
> to tell.
> Basically it means when it tried to write the frame it failed.  Most
> likely either someone was bind transferring it or FSAPI of uui_transfer was
> called on it.
>
>
> On Thu, May 28, 2015 at 7:57 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Some calls is droped during a conversation
>> What is means "[DEBUG] switch_ivr_bridge.c:579 sofia/internal/
>> 1246 at client-1424.rcsnet.ru ending bridge by request from write function"?
>>
>> In example message at 11:01:05
>> https://pastebin.freeswitch.org/24241
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Thu May 28 19:22:14 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 28 May 2015 11:22:14 -0400
Subject: [Freeswitch-users] compiling FreeSWITCH fs-video2 branch on
	debian jessie
In-Reply-To: <CAHtxdDcs-P+GJd-6AM5Nvch_1_8FN1B=YmpTPB=GCEWT2c-TMQ@mail.gmail.com>
References: <CAJwe+CBjCcXNssgntr3xSNdbmfURhjQYt7SCZkc3AW482rTcHQ@mail.gmail.com>
	<D156873B.9B0FD%krice@freeswitch.org>
	<CA+M05dAD=HsD9UhhSBgatWTwAFOtqPhxucd=anV3cqycyVr_MQ@mail.gmail.com>
	<CAHtxdDcs-P+GJd-6AM5Nvch_1_8FN1B=YmpTPB=GCEWT2c-TMQ@mail.gmail.com>
Message-ID: <2E644986-D1A1-4642-9D94-5693C8C53722@jerris.com>

for yuv, try to just use make instead of cmake.

> On May 28, 2015, at 6:02 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> When I create yum spec file I has diabled find jpeg library.
> 
> It requered because FS config file is not searching jpeg library
> 
> In SPEC file I added command 
> sed -i -e "s:STATIC:SHARED:" -e "s:\(include.FindJPEG.\):#\1:" CMakeLists.txt
> 
> https://freeswitch.org/stash/projects/SD/repos/libyuv/browse/libyuv.spec <https://freeswitch.org/stash/projects/SD/repos/libyuv/browse/libyuv.spec>
> 
> 
> On Thu, May 28, 2015 at 11:05 AM, Oleg Blinnikov <osblinnikov at gmail.com <mailto:osblinnikov at gmail.com>> wrote:
> I also run into this problem, but I installed yuv with cmake install target:
> 
>   git clone http://git.chromium.org/external/libyuv.git <http://git.chromium.org/external/libyuv.git>
>   cd libyuv
>  mkdir out
>  cd out
>  cmake -DCMAKE_INSTALL_PREFIX="/usr/lib" -DCMAKE_BUILD_TYPE="Release" ..
>  cmake --build . --config Release
>  sudo cmake --build . --target install --config Release
> 
> May be someone knows what do I do wrong?
> 
> On Fri, Apr 17, 2015 at 4:47 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
> We actually had to patch several things for this... We'll be making those
> things available shortly
> 
> 
> On 4/17/15, 9:42 AM, "E. Schmidbauer" <eschmidbauer at gmail.com <mailto:eschmidbauer at gmail.com>> wrote:
> 
> > Hi Sergey,
> > The make file (linux.mk <http://linux.mk/>) does not have an "install" action so that
> > will not work.
> > Thanks,
> > E
> >
> > On Fri, Apr 17, 2015 at 10:27 AM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
> >> try execute "make install" from libyuv folter. Then repeat FS compiling.
> >>
> >> Sergey
> >>
> >> On Fri, Apr 17, 2015 at 4:24 PM, E. Schmidbauer <eschmidbauer at gmail.com <mailto:eschmidbauer at gmail.com>>
> >> wrote:
> >>>
> >>> Hello,
> >>> I'm trying to compile fs-video2 and getting the following error during
> >>> ./configure
> >>>
> >>> checking for libyuv >= 0.0.1280... configure: error: You need to
> >>> install libyuv-dev. Required library
> >>>
> >>>
> >>> I git cloned and compiled the libyuv library FS is looking for:
> >>>
> >>> git clone https://github.com/openpeer/libyuv.git <https://github.com/openpeer/libyuv.git>
> >>> cd libyuv
> >>> make -j7 V=1 -f linux.mk <http://linux.mk/>
> >>>
> >>> (it compiles without errors)
> >>>
> >>> How can I tell FS where to find the library?
> >>>
> >>> Thanks,
> >>> E
> >>>
> >>> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org <http://www.freeswitch.org/>
> >>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> >>> http://www.cluecon.com <http://www.cluecon.com/>
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> >>> http://www.freeswitch.org <http://www.freeswitch.org/>
> >>
> >>
> >>
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> >> http://www.cluecon.com <http://www.cluecon.com/>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >
> > _________________________________________________________________________
> > 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/>
> 
> --
> Ken
> http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <http://www.ostag.org/>
> irc.freenode.net <http://irc.freenode.net/> #freeswitch
> Twitter: @FreeSWITCH
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Sincerely,
> Oleg Blinnikov
> 
> _________________________________________________________________________
> 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  Thu May 28 19:23:22 2015
From: chad at apartmentlines.com (Chad Phillips)
Date: Thu, 28 May 2015 10:23:22 -0500
Subject: [Freeswitch-users] Documentation for 1.6+Video build
In-Reply-To: <CACLfcpswTBxXGxxqKg_-U=kOrhSu4TBDVR55Ra-PcaYEir-rkg@mail.gmail.com>
References: <CACLfcpswTBxXGxxqKg_-U=kOrhSu4TBDVR55Ra-PcaYEir-rkg@mail.gmail.com>
Message-ID: <CAJ0sghQ_6dJNiHpp+2rsps=__7HKJUFf5uAvFhhiTPfJp49gmg@mail.gmail.com>

Craig,

I have a completely automated setup script for the video branch, which I
use to roll both local dev servers (via Vagrant) and production (via
Linode) ? handles most or all of the niggly details you discuss above.

My intention is to remove my customizations and make it available to the
community at some point in the near future. It?s seriously crazy that
people hand build a server ? this is 2015 for god?s sake? :P

Chad

On Wed, May 27, 2015 at 8:05 PM, Craig Stevenson <craig at stevenson.com>
wrote:

> As a novice trying to follow the documentation for creating FreeSWITCH
> 1.6+Video instance, I have a few questions and have noticed a few gaps in
> the documentation.  For experienced users, these gaps are the "obvious"
> steps... but not so obvious to those of us just getting started...
>
>
> For consistency with the Wheezy 1.4 instructions, should the instructions
> have users clone the code into /usr/local/src/freeswitch?  If so, modify
> the install instructions to 'cd /usr/local/src' and clone the software into
> 'freeswitch' instead of 'freeswitch.git'
>
> # then let's get the source
>
> cd /usr/local/src
>
> git clone https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch
> cd freeswitch.git
> git checkout fs-video2
> ./bootstrap.sh -j
> ./configure -C
>
> I'm curious why there is a change in the recommended parameters for
> ./configure
>
> Wheezy 1.4:  ./configure --enable-core-pgsql-support
>
> Jessie 1.6: ./configure -C
>
>
> Should the 1.6+Video install instructions page replicate (or at least
> reference) create the freeswitch user?
>
> Should the 1.6+Video install instructions page replicate (or at least
> reference) how to setup FreeSWITCH for automatic start.
>
> The 1.6+Video page has a section on setting up Bridging from WebRTC that
> shows adding an extension to Dialplan.  For the novice user, it might help
> to A) mention the exact file (e.g.
> /usr/local/freeswitch/conf/dialplan/default.xml and B) mention if there is
> a specific place in the file where that text should go (or is it OK to just
> add it to the end of the file?
>
> Again, I know some of my questions above are painfully obvious to most of
> you.  But they are not obvious to the new person trying to get started.  I
> mention these here in the hope of helping others who are also just getting
> started.
>
> Thanks,
> Craig
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tru083 at yahoo.com  Thu May 28 19:26:01 2015
From: tru083 at yahoo.com (D D)
Date: Thu, 28 May 2015 15:26:01 +0000 (UTC)
Subject: [Freeswitch-users] Guidelines for adding a custom SIP INFO package?
Message-ID: <1331892441.522840.1432826761612.JavaMail.yahoo@mail.yahoo.com>

Hi,
I would like to add a custom SIP INFO package, controlled by an event socket application.
I know I can use sip_h variables to set header fields, but is there a better way to identify to sofiathat I am adding a custom SIP INFO package?

Also, do you have any tips on processing INFO events and sending INFO messages?
Thanks!David

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From jack at livecall.com  Thu May 28 19:44:43 2015
From: jack at livecall.com (Jack)
Date: Thu, 28 May 2015 08:44:43 -0700
Subject: [Freeswitch-users] Verto on Windows?
In-Reply-To: <A7D709E7-3421-4C26-ADF2-E2A56C553122@jerris.com>
References: <55669515.4010700@livecall.com>
	<A7D709E7-3421-4C26-ADF2-E2A56C553122@jerris.com>
Message-ID: <556737EB.9040004@livecall.com>

Thanks...

On 5/28/2015 8:15 AM, Michael Jerris wrote:
> The issue setting up verto on 2 different servers comes down to issues with certificates.  I would first try it out on a single linux server to confirm you have all the freeswitch side stuff setup properly... then move to a windows box still with apache, so we can verify windows mod_verto is working, then work on changing to IIS and splitting to multiple servers.  I dont' think you'll hit any major bumps, buf figuring out the right config and how to get the certificates right may take some fiddling.
>
>> On May 28, 2015, at 12:09 AM, Jack <jack at livecall.com> wrote:
>>
>> Has anyone been successful at setting up the html5 verto page with
>> windows IIS running on one server and Freeswitch running on a different
>> windows server?
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2014.0.4800 / Virus Database: 4311/9888 - Release Date: 05/28/15
>
>




From anthony.minessale at gmail.com  Thu May 28 19:53:27 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 May 2015 10:53:27 -0500
Subject: [Freeswitch-users] Documentation for 1.6+Video build
In-Reply-To: <CAJ0sghQ_6dJNiHpp+2rsps=__7HKJUFf5uAvFhhiTPfJp49gmg@mail.gmail.com>
References: <CACLfcpswTBxXGxxqKg_-U=kOrhSu4TBDVR55Ra-PcaYEir-rkg@mail.gmail.com>
	<CAJ0sghQ_6dJNiHpp+2rsps=__7HKJUFf5uAvFhhiTPfJp49gmg@mail.gmail.com>
Message-ID: <CAKbxfG8hU58EGvCVHU0oAisuf+HfpkAvad0XdqbmOz4tabxAdw@mail.gmail.com>

When we are done we will have full packages of FS 1.6 for Debian Jessie and
any other platform where serious volunteering is done.
The docs for building it yourself will also mature but first things first
we are working on finalizing 1.4 to its own branch and moving the video
branch into master.


On Thu, May 28, 2015 at 10:23 AM, Chad Phillips <chad at apartmentlines.com>
wrote:

> Craig,
>
> I have a completely automated setup script for the video branch, which I
> use to roll both local dev servers (via Vagrant) and production (via
> Linode) ? handles most or all of the niggly details you discuss above.
>
> My intention is to remove my customizations and make it available to the
> community at some point in the near future. It?s seriously crazy that
> people hand build a server ? this is 2015 for god?s sake? :P
>
> Chad
>
> On Wed, May 27, 2015 at 8:05 PM, Craig Stevenson <craig at stevenson.com>
> wrote:
>
>> As a novice trying to follow the documentation for creating FreeSWITCH
>> 1.6+Video instance, I have a few questions and have noticed a few gaps in
>> the documentation.  For experienced users, these gaps are the "obvious"
>> steps... but not so obvious to those of us just getting started...
>>
>>
>> For consistency with the Wheezy 1.4 instructions, should the instructions
>> have users clone the code into /usr/local/src/freeswitch?  If so, modify
>> the install instructions to 'cd /usr/local/src' and clone the software into
>> 'freeswitch' instead of 'freeswitch.git'
>>
>> # then let's get the source
>>
>> cd /usr/local/src
>>
>> git clone https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch
>> cd freeswitch.git
>> git checkout fs-video2
>> ./bootstrap.sh -j
>> ./configure -C
>>
>> I'm curious why there is a change in the recommended parameters for
>> ./configure
>>
>> Wheezy 1.4:  ./configure --enable-core-pgsql-support
>>
>> Jessie 1.6: ./configure -C
>>
>>
>> Should the 1.6+Video install instructions page replicate (or at least
>> reference) create the freeswitch user?
>>
>> Should the 1.6+Video install instructions page replicate (or at least
>> reference) how to setup FreeSWITCH for automatic start.
>>
>> The 1.6+Video page has a section on setting up Bridging from WebRTC that
>> shows adding an extension to Dialplan.  For the novice user, it might help
>> to A) mention the exact file (e.g.
>> /usr/local/freeswitch/conf/dialplan/default.xml and B) mention if there is
>> a specific place in the file where that text should go (or is it OK to just
>> add it to the end of the file?
>>
>> Again, I know some of my questions above are painfully obvious to most of
>> you.  But they are not obvious to the new person trying to get started.  I
>> mention these here in the hope of helping others who are also just getting
>> started.
>>
>> Thanks,
>> Craig
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mike at jerris.com  Thu May 28 20:15:18 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 28 May 2015 12:15:18 -0400
Subject: [Freeswitch-users] Guidelines for adding a custom SIP INFO
	package?
In-Reply-To: <1331892441.522840.1432826761612.JavaMail.yahoo@mail.yahoo.com>
References: <1331892441.522840.1432826761612.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <0ECDBFD2-CE60-453F-954D-477B191E99A6@jerris.com>

What do you mean by "package"?  You can use esl and events to send custom info.

> On May 28, 2015, at 11:26 AM, D D <tru083 at yahoo.com> wrote:
> 
> Hi,
> 
> I would like to add a custom SIP INFO package, controlled by an event socket application.
> 
> I know I can use sip_h variables to set header fields, but is there a better way to identify to sofia
> that I am adding a custom SIP INFO package?
> 
> Also, do you have any tips on processing INFO events and sending INFO messages?
> 
> Thanks!
> David

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From olegstolyar at gmail.com  Thu May 28 20:28:19 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Thu, 28 May 2015 09:28:19 -0700
Subject: [Freeswitch-users] Receiving and catching SIP INFO messages
Message-ID: <CAC8JMOLA2VnnfWHYSsmLp5HBQUaTZ4apPwYRZQ9-O4adVX4ojA@mail.gmail.com>

Hi guys,

Is there a way to receive SIP INFO messages in FS?  Ideally without events
but if there is no way, then with events may be OK.

For reference, I currently do it by having the other side send a DTMF tone
 via INFO, catching it with bind_digit_action and then checking the
sip_info_h headers for the payload.

But I was wondering if there is a way to do it with a straight INFO
messages rather than a DTMF.
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From mike at jerris.com  Thu May 28 20:38:14 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 28 May 2015 12:38:14 -0400
Subject: [Freeswitch-users] Receiving and catching SIP INFO messages
In-Reply-To: <CAC8JMOLA2VnnfWHYSsmLp5HBQUaTZ4apPwYRZQ9-O4adVX4ojA@mail.gmail.com>
References: <CAC8JMOLA2VnnfWHYSsmLp5HBQUaTZ4apPwYRZQ9-O4adVX4ojA@mail.gmail.com>
Message-ID: <4EDC66B1-6754-461F-B1AD-76707EDF6462@jerris.com>

The only way to hook them currently is with events.

> On May 28, 2015, at 12:28 PM, Oleg Stolyar <olegstolyar at gmail.com> wrote:
> 
> Hi guys,
> 
> Is there a way to receive SIP INFO messages in FS?  Ideally without events but if there is no way, then with events may be OK.
> 
> For reference, I currently do it by having the other side send a DTMF tone  via INFO, catching it with bind_digit_action and then checking the sip_info_h headers for the payload.
> 
> But I was wondering if there is a way to do it with a straight INFO messages rather than a DTMF.



From olegstolyar at gmail.com  Thu May 28 20:54:55 2015
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Thu, 28 May 2015 09:54:55 -0700
Subject: [Freeswitch-users] Receiving and catching SIP INFO messages
In-Reply-To: <4EDC66B1-6754-461F-B1AD-76707EDF6462@jerris.com>
References: <CAC8JMOLA2VnnfWHYSsmLp5HBQUaTZ4apPwYRZQ9-O4adVX4ojA@mail.gmail.com>
	<4EDC66B1-6754-461F-B1AD-76707EDF6462@jerris.com>
Message-ID: <CAC8JMOK=826rncdZADTvciBgiqVSLc_wkAQzLf3D7djXmMUevQ@mail.gmail.com>

Thanks Michael!
On May 28, 2015 9:41 AM, "Michael Jerris" <mike at jerris.com> wrote:

> The only way to hook them currently is with events.
>
> > On May 28, 2015, at 12:28 PM, Oleg Stolyar <olegstolyar at gmail.com>
> wrote:
> >
> > Hi guys,
> >
> > Is there a way to receive SIP INFO messages in FS?  Ideally without
> events but if there is no way, then with events may be OK.
> >
> > For reference, I currently do it by having the other side send a DTMF
> tone  via INFO, catching it with bind_digit_action and then checking the
> sip_info_h headers for the payload.
> >
> > But I was wondering if there is a way to do it with a straight INFO
> messages rather than a DTMF.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From craig at stevenson.com  Thu May 28 21:08:05 2015
From: craig at stevenson.com (Craig Stevenson)
Date: Thu, 28 May 2015 10:08:05 -0700
Subject: [Freeswitch-users] Documentation for 1.6+Video build
In-Reply-To: <CAKbxfG8hU58EGvCVHU0oAisuf+HfpkAvad0XdqbmOz4tabxAdw@mail.gmail.com>
References: <CACLfcpswTBxXGxxqKg_-U=kOrhSu4TBDVR55Ra-PcaYEir-rkg@mail.gmail.com>
	<CAJ0sghQ_6dJNiHpp+2rsps=__7HKJUFf5uAvFhhiTPfJp49gmg@mail.gmail.com>
	<CAKbxfG8hU58EGvCVHU0oAisuf+HfpkAvad0XdqbmOz4tabxAdw@mail.gmail.com>
Message-ID: <CACLfcpvsyaTtWnruqU2t=UtUB0yoEe0HyMa6V_RcVoLChdV3Bg@mail.gmail.com>

Great.  And, happy to help with the documentation as you are ready for
volunteers.

On Thu, May 28, 2015 at 8:53 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> When we are done we will have full packages of FS 1.6 for Debian Jessie
> and any other platform where serious volunteering is done.
> The docs for building it yourself will also mature but first things first
> we are working on finalizing 1.4 to its own branch and moving the video
> branch into master.
>
>
> On Thu, May 28, 2015 at 10:23 AM, Chad Phillips <chad at apartmentlines.com>
> wrote:
>
>> Craig,
>>
>> I have a completely automated setup script for the video branch, which I
>> use to roll both local dev servers (via Vagrant) and production (via
>> Linode) ? handles most or all of the niggly details you discuss above.
>>
>> My intention is to remove my customizations and make it available to the
>> community at some point in the near future. It?s seriously crazy that
>> people hand build a server ? this is 2015 for god?s sake? :P
>>
>> Chad
>>
>> On Wed, May 27, 2015 at 8:05 PM, Craig Stevenson <craig at stevenson.com>
>> wrote:
>>
>>> As a novice trying to follow the documentation for creating FreeSWITCH
>>> 1.6+Video instance, I have a few questions and have noticed a few gaps in
>>> the documentation.  For experienced users, these gaps are the "obvious"
>>> steps... but not so obvious to those of us just getting started...
>>>
>>>
>>> For consistency with the Wheezy 1.4 instructions, should the
>>> instructions have users clone the code into /usr/local/src/freeswitch?  If
>>> so, modify the install instructions to 'cd /usr/local/src' and clone the
>>> software into 'freeswitch' instead of 'freeswitch.git'
>>>
>>> # then let's get the source
>>>
>>> cd /usr/local/src
>>>
>>> git clone https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch
>>> cd freeswitch.git
>>> git checkout fs-video2
>>> ./bootstrap.sh -j
>>> ./configure -C
>>>
>>> I'm curious why there is a change in the recommended parameters for
>>> ./configure
>>>
>>> Wheezy 1.4:  ./configure --enable-core-pgsql-support
>>>
>>> Jessie 1.6: ./configure -C
>>>
>>>
>>> Should the 1.6+Video install instructions page replicate (or at least
>>> reference) create the freeswitch user?
>>>
>>> Should the 1.6+Video install instructions page replicate (or at least
>>> reference) how to setup FreeSWITCH for automatic start.
>>>
>>> The 1.6+Video page has a section on setting up Bridging from WebRTC that
>>> shows adding an extension to Dialplan.  For the novice user, it might help
>>> to A) mention the exact file (e.g.
>>> /usr/local/freeswitch/conf/dialplan/default.xml and B) mention if there is
>>> a specific place in the file where that text should go (or is it OK to just
>>> add it to the end of the file?
>>>
>>> Again, I know some of my questions above are painfully obvious to most
>>> of you.  But they are not obvious to the new person trying to get started.
>>> I mention these here in the hope of helping others who are also just
>>> getting started.
>>>
>>> Thanks,
>>> Craig
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Thu May 28 22:10:34 2015
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 May 2015 18:10:34 +0000
Subject: [Freeswitch-users] The FreeSWITCH 1.4.19 release is here!
Message-ID: <55675a1a3ae56_27a8112533024660@resque-worker-high.1.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1JXt4zH
The FreeSWITCH 1.4.19 release is here!
The FreeSWITCH 1.4.19 release is here!

This is routine maintenance release and the source tarballs can be found: http://ift.tt/1Fj7gsO

The features for this release include:

FS-7470 [mod_verto] Add a force-register-domain param

FS-7526 [mod_amqp] Add enable_fallback_format_fields for mod_amqp producer profiles if the profile param is set and create the amqp exchange on the first startup of a clean platform.

FS-7557 [mod_mongo] Add limit backend

Improvements in build system, cross platform support, and packaging:

FS-7488 [mod_managed] Fixed a build error with Windows and removed duplicate files

FS-7574 [build] Update to handle new download locations for sounds

FS-7574 [build] Fixed it so curl has the proper flag to follow 3XX redirects for getg729 and getsounds

The following bugs were squashed:

FS-7456 Fixed a bug in SDP parser, to allow it to handle sdp with m lines in any order

FS-7460 [mod_sofia] Don?t force ICE in 3pcc-mode=proxy

FS-7465 Fixed a crash in stereo file writing

FS-7425 Fixed an audio Handshake failure 1 when using webrtc with PFS-only client (Firefox &gt;=38)

FS-7472 [mod_sofia] Fix for a bug where the rtp-digit-delay profile param was being ignored

FS-7488 [mod_managed] Fixed a build error

FS-7490 [mod_rayo] Fixed a bug with the format of mod_rayo generated regex not working with newer libpcre

FS-7491 [mod_graylog2] Send timestamp with millisecond precision instead of microsecond as required by GELF

FS-7466 Fixed a bug causing audio issues by repeated log lines printing when rtp_manual_rtp_bugs is set to ALWAYS_AUTO_ADJUST

FS-7496 [mod_http_cache] Fixed an issue with the URL args being included in the cache file name and causing problems opening the files later

FS-7425 Fixed a bug when using a cert with missing dhparams resulting in a segfault.

FS-7523 [mod_json_cdr] Fixed a segfault caused by a missing config file.

FS-7357 FAX now tolerates EOP and PPS messages being incorrectly echoed.

FS-7552 [mod_amqp] Fixed a segfault on unload and when no connections were valid

FS-7463 [mod_sofia] Conditionally allow intercept of replaced call-id when processing replaces header

FS-7557 [mod_mongo] Fixed a crash when doing ?limit_usage mongo foo bar?

FS-7545 [mod_opus] Fixed RTP timestamps to prevent unneeded resampling when transcoding

FS-7184 [mod_spandsp] Fixed a fax buffer overflow in t38 on failure condition with some fax machines

FS-7546 [mod_spandsp] Fixed a crash when sending a fax when built using clang compiler

FS-7541 Fixed an issue with audio gaps in native audio recordings

FS-7562 [mod_sofia] Fixed an interop issue caused when using bypass media with t.38 passthru

FS-7567 Fixed a rare segfault on shutdown caused by a race condition

FS-7529 Fixed an error with call recording on G722 calls

FS-7584 Fixed rtcp-mux interop issue with chrome canary causing video transport failure

FS-7582 Fixed a bug with reINVITES not including a=setup:actpass in the SDP

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From covici at ccs.covici.com  Thu May 28 22:50:55 2015
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 28 May 2015 14:50:55 -0400
Subject: [Freeswitch-users] problem sending to dtmf to some sites
In-Reply-To: <DE056E7B-B3BD-475B-8071-A1AC21AAF192@jerris.com>
References: <28414.1432788603@ccs.covici.com>
	<DE056E7B-B3BD-475B-8071-A1AC21AAF192@jerris.com>
Message-ID: <2342.1432839055@ccs.covici.com>

OK, thanks so much.

Michael Jerris <mike at jerris.com> wrote:

> try using 'w' and 'W' in the DTMF string for a 1/2 and full second pause.
> 
> > On May 28, 2015, at 12:50 AM, covici at ccs.covici.com wrote:
> > 
> > Hi.  I call lots of numbers which are conference lines and they require
> > various dtmf codes after you dial the line.  Now my problem is that some
> > of them will not work unless you send the dtmf with some pauses in
> > between each digit.  Is there any way freeswitch can do this --
> > i.e. mod_conference can outcall and send dtmf after the call is
> > answered.
> > 
> > Thanks in advance for any assistance.
> > 
> > -- 
> > 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
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services: 
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 krice at freeswitch.org  Thu May 28 23:55:21 2015
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 May 2015 19:55:21 +0000
Subject: [Freeswitch-users] The 1.6 video branch has been merged into master!
Message-ID: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>

New Post on freeswitch.org from Kathleen
check it out at http://ift.tt/1RrHZnX
The 1.6 video branch has been merged into master!
Great news for our fellow FreeSWITCH users: in preparation for the 1.6 beta release, a lot of new functionality is being merged into the master branch. Some of these new features have new build requirements and dependencies, so please be sure to check the Confluence link here: http://ift.tt/1d2d4iX for platform specific instructions. Building and running FreeSWITCH will be easier if you are using Debian 8(Jessie). If you are not using Debian you can find some of the supporting dependencies tar.gz files here: http://ift.tt/1qAFKnH

Additionally, if you are using 1.4 in production, you need to be sure to switch to tracking the v1.4 branch, as master will be preparing for the 1.6 beta release.

Some of the new features and work that have gone into this release include:FS-7499 core RTCP improvementsFS-7500 core video transcoding supportFS-7501 core video jitterbufferFS-7502 core video media bugsFS-7503 core file interface video supportFS-7504 codecs let you choose which codec module to useFS-7505 file interface to let you specify which format module to use when multiple types are supportedFS-7506 core text renderingFS-7507 added new global directory variables and configure directory behavior changesFS-7508 mod_vpx transcoding vp8/vp9 and replace mod_v8FS-7509 mod_verto improvements allow for desktop share with the installation of this chrome extension http://ift.tt/1kQP2pS and improved bandwith and resolution handling.FS-7512 mod_png allows for image overlays for logos and images for video muteFS-7513 mod_conference MCU feature and avatar supportFS-7514 mod_vlc video support allows you to live stream, record calls to a video file, and playback videos into a call.FS-7515 mod_cv is a video media bug module that uses video recognition and facial recognition technology to allow you to modify a video stream by adding overlapping images and text or to silently detect and fire eventsFS-7516 mod_imagick allows for PDF and GIF rendered as videoFS-7517 mod_openh264 h264 codec moduleFS-7519 mod_av a file format and codec module that uses libav or ffmpegFS-7494 default avatar and mute images for video MCUFS-7471 improved configs for videoFS-7338 removed external library dependenciesFS-7585 added video support to mod_rtmp

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From gmaruzz at gmail.com  Fri May 29 00:45:44 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 May 2015 22:45:44 +0200
Subject: [Freeswitch-users] The 1.6 video branch has been merged into
	master!
In-Reply-To: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>
References: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>
Message-ID: <CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>

Yay!!!

All hail our FreeSWITCH developers and core team !

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 28, 2015 9:56 PM, "Ken Rice" <krice at freeswitch.org> wrote:

> New Post on freeswitch.org from Kathleen
> check it out at http://ift.tt/1RrHZnX
> The 1.6 video branch has been merged into master!
>
> Great news for our fellow FreeSWITCH users: in preparation for the 1.6
> beta release, a lot of new functionality is being merged into the master
> branch. Some of these new features have new build requirements and
> dependencies, so please be sure to check the Confluence link here:
> http://ift.tt/1d2d4iX for platform specific instructions. Building and
> running FreeSWITCH will be easier if you are using Debian 8(Jessie). If you
> are not using Debian you can find some of the supporting dependencies
> tar.gz files here: http://ift.tt/1qAFKnH
>
> Additionally, if you are using 1.4 in production, you need to be sure to
> switch to tracking the v1.4 branch, as master will be preparing for the 1.6
> beta release.
>
> Some of the new features and work that have gone into this release include:
> FS-7499 <http://ift.tt/1d2d2HU> core RTCP improvements
> FS-7500 <http://ift.tt/1PQJHBY> core video transcoding support
> FS-7501 <http://ift.tt/1d2d2HW> core video jitterbuffer
> FS-7502 <http://ift.tt/1PQJFdn> core video media bugs
> FS-7503 <http://ift.tt/1d2d4iZ> core file interface video support
> FS-7504 <http://ift.tt/1PQJHC2> codecs let you choose which codec module
> to use
> FS-7505 <http://ift.tt/1d2d4j1> file interface to let you specify which
> format module to use when multiple types are supported
> FS-7506 <http://ift.tt/1PQJHC4> core text rendering
> FS-7507 <http://ift.tt/1d2d4j3> added new global directory variables and
> configure directory behavior changes
> FS-7508 <http://ift.tt/1d2d2HY> mod_vpx transcoding vp8/vp9 and replace
> mod_v8
> FS-7509 <http://ift.tt/1PQJHC6> mod_verto improvements allow for desktop
> share with the installation of this chrome extension http://ift.tt/1kQP2pS
> and improved bandwith and resolution handling.
> FS-7512 <http://ift.tt/1PQJHCa> mod_png allows for image overlays for
> logos and images for video mute
> FS-7513 <http://ift.tt/1d2d4j5> mod_conference MCU feature and avatar
> support
> FS-7514 <http://ift.tt/1PQJI8Z> mod_vlc video support allows you to live
> stream, record calls to a video file, and playback videos into a call.
> FS-7515 <http://ift.tt/1d2d2Yc> mod_cv is a video media bug module that
> uses video recognition and facial recognition technology to allow you to
> modify a video stream by adding overlapping images and text or to silently
> detect and fire events
> FS-7516 <http://ift.tt/1PQJHSo> mod_imagick allows for PDF and GIF
> rendered as video
> FS-7517 <http://ift.tt/1d2d2Ye> mod_openh264 h264 codec module
> FS-7519 <http://ift.tt/1d2d4zl> mod_av a file format and codec module
> that uses libav or ffmpeg
> FS-7494 <http://ift.tt/1PQJHSs> default avatar and mute images for video
> MCU
> FS-7471 <http://ift.tt/1d2d4zn> improved configs for video
> FS-7338 <http://ift.tt/1PQJI97> removed external library dependencies
> FS-7585 <http://ift.tt/1d2d4zr> added video support to mod_rtmp
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bpriddy at bryantschools.org  Fri May 29 00:48:55 2015
From: bpriddy at bryantschools.org (Blake Priddy)
Date: Thu, 28 May 2015 15:48:55 -0500
Subject: [Freeswitch-users] The 1.6 video branch has been merged into
	master!
In-Reply-To: <CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>
References: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>
	<CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>
Message-ID: <CADK-BjuyMD74Epn=P5zviUGOKkKC+4ddNP_HXkRCOrpmFvy0nQ@mail.gmail.com>

Great work guys!

On Thu, May 28, 2015 at 3:45 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Yay!!!
>
> All hail our FreeSWITCH developers and core team !
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On May 28, 2015 9:56 PM, "Ken Rice" <krice at freeswitch.org> wrote:
>
>> New Post on freeswitch.org from Kathleen
>> check it out at http://ift.tt/1RrHZnX
>> The 1.6 video branch has been merged into master!
>>
>> Great news for our fellow FreeSWITCH users: in preparation for the 1.6
>> beta release, a lot of new functionality is being merged into the master
>> branch. Some of these new features have new build requirements and
>> dependencies, so please be sure to check the Confluence link here:
>> http://ift.tt/1d2d4iX for platform specific instructions. Building and
>> running FreeSWITCH will be easier if you are using Debian 8(Jessie). If you
>> are not using Debian you can find some of the supporting dependencies
>> tar.gz files here: http://ift.tt/1qAFKnH
>>
>> Additionally, if you are using 1.4 in production, you need to be sure to
>> switch to tracking the v1.4 branch, as master will be preparing for the 1.6
>> beta release.
>>
>> Some of the new features and work that have gone into this release
>> include:
>> FS-7499 <http://ift.tt/1d2d2HU> core RTCP improvements
>> FS-7500 <http://ift.tt/1PQJHBY> core video transcoding support
>> FS-7501 <http://ift.tt/1d2d2HW> core video jitterbuffer
>> FS-7502 <http://ift.tt/1PQJFdn> core video media bugs
>> FS-7503 <http://ift.tt/1d2d4iZ> core file interface video support
>> FS-7504 <http://ift.tt/1PQJHC2> codecs let you choose which codec module
>> to use
>> FS-7505 <http://ift.tt/1d2d4j1> file interface to let you specify which
>> format module to use when multiple types are supported
>> FS-7506 <http://ift.tt/1PQJHC4> core text rendering
>> FS-7507 <http://ift.tt/1d2d4j3> added new global directory variables and
>> configure directory behavior changes
>> FS-7508 <http://ift.tt/1d2d2HY> mod_vpx transcoding vp8/vp9 and replace
>> mod_v8
>> FS-7509 <http://ift.tt/1PQJHC6> mod_verto improvements allow for desktop
>> share with the installation of this chrome extension
>> http://ift.tt/1kQP2pS and improved bandwith and resolution handling.
>> FS-7512 <http://ift.tt/1PQJHCa> mod_png allows for image overlays for
>> logos and images for video mute
>> FS-7513 <http://ift.tt/1d2d4j5> mod_conference MCU feature and avatar
>> support
>> FS-7514 <http://ift.tt/1PQJI8Z> mod_vlc video support allows you to live
>> stream, record calls to a video file, and playback videos into a call.
>> FS-7515 <http://ift.tt/1d2d2Yc> mod_cv is a video media bug module that
>> uses video recognition and facial recognition technology to allow you to
>> modify a video stream by adding overlapping images and text or to silently
>> detect and fire events
>> FS-7516 <http://ift.tt/1PQJHSo> mod_imagick allows for PDF and GIF
>> rendered as video
>> FS-7517 <http://ift.tt/1d2d2Ye> mod_openh264 h264 codec module
>> FS-7519 <http://ift.tt/1d2d4zl> mod_av a file format and codec module
>> that uses libav or ffmpeg
>> FS-7494 <http://ift.tt/1PQJHSs> default avatar and mute images for video
>> MCU
>> FS-7471 <http://ift.tt/1d2d4zn> improved configs for video
>> FS-7338 <http://ift.tt/1PQJI97> removed external library dependencies
>> FS-7585 <http://ift.tt/1d2d4zr> added video support to mod_rtmp
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 


*Blakelund Priddy*
Network & Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
http://www.bryantschools.org
p 501-653-5038
f 501-847-5656
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From jungleboogie0 at gmail.com  Fri May 29 01:35:16 2015
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Thu, 28 May 2015 14:35:16 -0700
Subject: [Freeswitch-users] Install libyuv-dev needed
Message-ID: <CAKE2PDtoff2EVEoKGXOtf7JEHHYuMJjVBMb3DjTuAVniyT5Ofg@mail.gmail.com>

Hello All,

I'm updating master to latest code and my make is failing:
checking for libyuv >= 0.0.1280... configure: error: You need to
install libyuv-dev. Required library
Makefile:936: recipe for target 'config.status' failed
make: *** [config.status] Error 1

Where can I get this required library file?

I'm running raspberry pi 2 raspbian flavor.

Thanks!


-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From krice at freeswitch.org  Fri May 29 01:41:55 2015
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 May 2015 16:41:55 -0500
Subject: [Freeswitch-users] Install libyuv-dev needed
In-Reply-To: <CAKE2PDtoff2EVEoKGXOtf7JEHHYuMJjVBMb3DjTuAVniyT5Ofg@mail.gmail.com>
Message-ID: <D18CF5D3.9FA77%krice@freeswitch.org>

https://freeswitch.org/the-1-6-video-branch-has-been-merged-into-master/


On 5/28/15, 4:35 PM, "jungle Boogie" <jungleboogie0 at gmail.com> wrote:

> Hello All,
> 
> I'm updating master to latest code and my make is failing:
> checking for libyuv >= 0.0.1280... configure: error: You need to
> install libyuv-dev. Required library
> Makefile:936: recipe for target 'config.status' failed
> make: *** [config.status] Error 1
> 
> Where can I get this required library file?
> 
> I'm running raspberry pi 2 raspbian flavor.
> 
> Thanks!
> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH





From mike at jerris.com  Fri May 29 01:46:02 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 28 May 2015 17:46:02 -0400
Subject: [Freeswitch-users] Install libyuv-dev needed
In-Reply-To: <D18CF5D3.9FA77%krice@freeswitch.org>
References: <CAKE2PDtoff2EVEoKGXOtf7JEHHYuMJjVBMb3DjTuAVniyT5Ofg@mail.gmail.com>
	<D18CF5D3.9FA77%krice@freeswitch.org>
Message-ID: <CAEwTsAjmiS_YoewWwgtpm_QhcC+qLeh=o3ewgd9At_GuZafC8A@mail.gmail.com>

We are very interested in getting a raspian cross build working for 1.6.
If you can assist in getting all our dependency packages building for
raspian we would love to merge that in and add a repo for that.  If
interested please reach out to Ken or myself to coordinate.

On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org> wrote:

> https://freeswitch.org/the-1-6-video-branch-has-been-merged-into-master/
>
>
> On 5/28/15, 4:35 PM, "jungle Boogie" <jungleboogie0 at gmail.com
> <javascript:;>> wrote:
>
> > Hello All,
> >
> > I'm updating master to latest code and my make is failing:
> > checking for libyuv >= 0.0.1280... configure: error: You need to
> > install libyuv-dev. Required library
> > Makefile:936: recipe for target 'config.status' failed
> > make: *** [config.status] Error 1
> >
> > Where can I get this required library file?
> >
> > I'm running raspberry pi 2 raspbian flavor.
> >
> > Thanks!
> >
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mandra at gmail.com  Fri May 29 02:20:35 2015
From: mandra at gmail.com (Chris Mandra)
Date: Thu, 28 May 2015 18:20:35 -0400
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
	having no luck with fs_video
Message-ID: <CAJdRzN5D40JKuN3MN82P5gij-=XGsufc4cktVKkO4cFwbASLKg@mail.gmail.com>

Hi guys. I had been able to use webrtc via chrome with FS 1.4 for
connections and confenerce calls without incident (although the bandwidth
and video quality seemed rather low as compared to the apprtc.appspot.com
demo.)
I have followed the instructions as well I am able on
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
but so far I can't make an actual connection; the certs pass the cert test
without issue, but I have no joy using webrtc and fs_video

a few questions:
1. Is there any known problem using wildcard certs from godaddy?
2. is there other configuration beyond what's on the web page needed to
make webrtc work? (I have the apropos ports open etc)
3. Are there any known issues with fs_video and sip.js?
4. I feel like it's getting hung op on dtls, as it never passes
"2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS
state from OFF to HANDSHAKE"
(it repeats that twice with:
"2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
packet 0 bytes (ice not ready @ line 4241!)" not long after - does that
speak to something obvious to you guys?
5. is there a working webrtc / fs_video demo I can look at?

I realize this may all be solved by the release of 1.6, but I need this
working now. I've been looking forward to this bc I know there have been
some issues with webrtc, rtcp, freeswitch and solid BWE.

Thanks for all you do,
chris
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From bra_34 at yahoo.com  Fri May 29 01:02:35 2015
From: bra_34 at yahoo.com (Luciano de Oliveira)
Date: Thu, 28 May 2015 21:02:35 +0000 (UTC)
Subject: [Freeswitch-users] change reponse 503 on max-sessions
Message-ID: <1687101446.831910.1432846956004.JavaMail.yahoo@mail.yahoo.com>

Hi gentleman,
How do i change the?SIP/2.0 503 Maximum Calls In Progress, response when max-sessions parameter is reached.
I?m having some problems with the provider.
Best regards,Luciano.
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From krice at freeswitch.org  Fri May 29 02:45:41 2015
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 May 2015 17:45:41 -0500
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
 having no luck with fs_video
In-Reply-To: <CAJdRzN5D40JKuN3MN82P5gij-=XGsufc4cktVKkO4cFwbASLKg@mail.gmail.com>
Message-ID: <D18D04C5.9FA83%krice@freeswitch.org>

Hi Chris,

There is always https://conference.freeswitch.org/verto/ where we have been
demoing this for quite a while

If you need this working like now I would suggest emailing
consulting at freeswitch.org where you can get paid realtime assistance from
one of the Core Dev Team

See you @ ClueCon!
Ken
http://www.ClueCon.com
http://www.freeswitch.org
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH
         @ClueCon


On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com> wrote:

> Hi guys. I had been able to use webrtc via chrome with FS 1.4 for connections
> and confenerce calls without incident (although the bandwidth and video
> quality seemed rather low as compared to the apprtc.appspot.com
> <http://apprtc.appspot.com>  demo.)
> I have followed the instructions as well I am able
> on?https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
> but so far I can't make an actual connection; the certs pass the cert test
> without issue, but I have no joy using webrtc and fs_video
> 
> a few questions:?
> 1. Is there any known problem using wildcard certs from godaddy?
> 2. is there other configuration beyond what's on the web page needed to make
> webrtc work? (I have the apropos ports open etc)?
> 3. Are there any known issues with fs_video and sip.js?
> 4. I feel like it's getting hung op on dtls, as it never passes?
> "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS state
> from OFF to HANDSHAKE"
> (it repeats that twice with:
> "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
> packet 0 bytes (ice not ready @ line 4241!)" not long after - does that speak
> to something obvious to you guys?
> 5. is there a working webrtc / fs_video demo I can look at?
> 
> I realize this may all be solved by the release of 1.6, but I need this
> working now. I've been looking forward to this bc I know there have been some
> issues with webrtc, rtcp, freeswitch and solid BWE.?
> 
> Thanks for all you do,
> chris
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From ssinyagin at gmail.com  Fri May 29 05:02:37 2015
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 29 May 2015 03:02:37 +0200
Subject: [Freeswitch-users] Guidelines for adding a custom SIP INFO
	package?
In-Reply-To: <0ECDBFD2-CE60-453F-954D-477B191E99A6@jerris.com>
References: <1331892441.522840.1432826761612.JavaMail.yahoo@mail.yahoo.com>
	<0ECDBFD2-CE60-453F-954D-477B191E99A6@jerris.com>
Message-ID: <CACOYK=jpqbRUTMXcxX_0MjPY4-yHfPLtV6v1CaALXNsJwkspxw@mail.gmail.com>

"Package" is a typical German speaker's mistake for "packet" ;-)
On May 28, 2015 6:16 PM, "Michael Jerris" <mike at jerris.com> wrote:

> What do you mean by "package"?  You can use esl and events to send custom
> info.
>
> On May 28, 2015, at 11:26 AM, D D <tru083 at yahoo.com> wrote:
>
> Hi,
>
> I would like to add a custom SIP INFO package, controlled by an event
> socket application.
>
> I know I can use sip_h variables to set header fields, but is there a
> better way to identify to sofia
> that I am adding a custom SIP INFO package?
>
> Also, do you have any tips on processing INFO events and sending INFO
> messages?
>
> Thanks!
> David
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mandra at gmail.com  Fri May 29 05:42:03 2015
From: mandra at gmail.com (Chris Mandra)
Date: Thu, 28 May 2015 21:42:03 -0400
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
	having no luck with fs_video
In-Reply-To: <D18D04C5.9FA83%krice@freeswitch.org>
References: <CAJdRzN5D40JKuN3MN82P5gij-=XGsufc4cktVKkO4cFwbASLKg@mail.gmail.com>
	<D18D04C5.9FA83%krice@freeswitch.org>
Message-ID: <CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>

Hi Ken, thanks for the reply.
Does https://conference.freeswitch.org/verto/ run fs_video? I kind of
thought it was running an older version.

I'm hoping to get some help from the community bc, well, I'm probably not
the only person with these problems and, like many people, I can't really
afford paid assistance.

I have a few more questions:

I'm getting the followingf messages I could use some clarification on:
1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by
candidate to set remote rtcp audio addr to 205.215.241.76:59980 but this is
rtcp-mux so no thanks (is this affecting anything?)

2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video DTLS
state from OFF to HANDSHAKE
- it never gets beyond handshake - any ideas

3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
packet 0 bytes (ice not ready @ line 4241!)
- ice not ready - what's causing this?

2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for
zrtp-hash
2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding whether
to pass zrtp-hash between legs
2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361
CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash

Should we be using ZRTP? if it weren't enabled at all would I see these
messages?

Lastly, it looks like we're not getting any sip responses from FS - Any
ideas why?

All answers/help/hints are appreciated.

chris




On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org> wrote:

>  Hi Chris,
>
> There is always https://conference.freeswitch.org/verto/ where we have
> been demoing this for quite a while
>
> If you need this working like now I would suggest emailing
> consulting at freeswitch.org where you can get paid realtime assistance from
> one of the Core Dev Team
>
> See you @ ClueCon!
> Ken
>
>
>
> *http://www.ClueCon.com <http://www.ClueCon.com> http://www.freeswitch.org
> <http://www.freeswitch.org> http://www.OSTAG.org <http://www.OSTAG.org> *
> irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>          @ClueCon
>
>
> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com> wrote:
>
> Hi guys. I had been able to use webrtc via chrome with FS 1.4 for
> connections and confenerce calls without incident (although the bandwidth
> and video quality seemed rather low as compared to the apprtc.appspot.com
> <http://apprtc.appspot.com>  demo.)
> I have followed the instructions as well I am able on
> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
> but so far I can't make an actual connection; the certs pass the cert test
> without issue, but I have no joy using webrtc and fs_video
>
> a few questions:
> 1. Is there any known problem using wildcard certs from godaddy?
> 2. is there other configuration beyond what's on the web page needed to
> make webrtc work? (I have the apropos ports open etc)
> 3. Are there any known issues with fs_video and sip.js?
> 4. I feel like it's getting hung op on dtls, as it never passes
> "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS
> state from OFF to HANDSHAKE"
> (it repeats that twice with:
> "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
> packet 0 bytes (ice not ready @ line 4241!)" not long after - does that
> speak to something obvious to you guys?
> 5. is there a working webrtc / fs_video demo I can look at?
>
> I realize this may all be solved by the release of 1.6, but I need this
> working now. I've been looking forward to this bc I know there have been
> some issues with webrtc, rtcp, freeswitch and solid BWE.
>
> Thanks for all you do,
> chris
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
>
>
>
> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>
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From gmaruzz at gmail.com  Fri May 29 05:47:02 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 29 May 2015 03:47:02 +0200
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
 having no luck with fs_video
In-Reply-To: <CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>
References: <CAJdRzN5D40JKuN3MN82P5gij-=XGsufc4cktVKkO4cFwbASLKg@mail.gmail.com>
	<D18D04C5.9FA83%krice@freeswitch.org>
	<CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>
Message-ID: <CALXCt0rK5S22T=CD7Rw=KbfcHTU8oQJyvvYh70zSqRqW1yrwAw@mail.gmail.com>

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

On Fri, May 29, 2015 at 3:42 AM, Chris Mandra <mandra at gmail.com> wrote:

> Hi Ken, thanks for the reply.
> Does https://conference.freeswitch.org/verto/ run fs_video? I kind of
> thought it was running an older version.
>
> I'm hoping to get some help from the community bc, well, I'm probably not
> the only person with these problems and, like many people, I can't really
> afford paid assistance.
>
> I have a few more questions:
>
> I'm getting the followingf messages I could use some clarification on:
> 1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by
> candidate to set remote rtcp audio addr to 205.215.241.76:59980 but this
> is rtcp-mux so no thanks (is this affecting anything?)
>
> 2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video DTLS
> state from OFF to HANDSHAKE
> - it never gets beyond handshake - any ideas
>
> 3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
> packet 0 bytes (ice not ready @ line 4241!)
> - ice not ready - what's causing this?
>
> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for
> zrtp-hash
> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding
> whether to pass zrtp-hash between legs
> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361
> CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
>
> Should we be using ZRTP? if it weren't enabled at all would I see these
> messages?
>
> Lastly, it looks like we're not getting any sip responses from FS - Any
> ideas why?
>
> All answers/help/hints are appreciated.
>
> chris
>
>
>
>
> On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org> wrote:
>
>>  Hi Chris,
>>
>> There is always https://conference.freeswitch.org/verto/ where we have
>> been demoing this for quite a while
>>
>> If you need this working like now I would suggest emailing
>> consulting at freeswitch.org where you can get paid realtime assistance
>> from one of the Core Dev Team
>>
>> See you @ ClueCon!
>> Ken
>>
>>
>>
>> *http://www.ClueCon.com <http://www.ClueCon.com>
>> http://www.freeswitch.org <http://www.freeswitch.org> http://www.OSTAG.org
>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>> Twitter: @FreeSWITCH
>>          @ClueCon
>>
>>
>> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com> wrote:
>>
>> Hi guys. I had been able to use webrtc via chrome with FS 1.4 for
>> connections and confenerce calls without incident (although the bandwidth
>> and video quality seemed rather low as compared to the apprtc.appspot.com
>> <http://apprtc.appspot.com>  demo.)
>> I have followed the instructions as well I am able on
>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>> but so far I can't make an actual connection; the certs pass the cert
>> test without issue, but I have no joy using webrtc and fs_video
>>
>> a few questions:
>> 1. Is there any known problem using wildcard certs from godaddy?
>> 2. is there other configuration beyond what's on the web page needed to
>> make webrtc work? (I have the apropos ports open etc)
>> 3. Are there any known issues with fs_video and sip.js?
>> 4. I feel like it's getting hung op on dtls, as it never passes
>> "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS
>> state from OFF to HANDSHAKE"
>> (it repeats that twice with:
>> "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
>> packet 0 bytes (ice not ready @ line 4241!)" not long after - does that
>> speak to something obvious to you guys?
>> 5. is there a working webrtc / fs_video demo I can look at?
>>
>> I realize this may all be solved by the release of 1.6, but I need this
>> working now. I've been looking forward to this bc I know there have been
>> some issues with webrtc, rtcp, freeswitch and solid BWE.
>>
>> Thanks for all you do,
>> chris
>>
>>
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> Ken
>>
>>
>>
>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>> Twitter: @FreeSWITCH
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From gmaruzz at gmail.com  Fri May 29 05:50:08 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 29 May 2015 03:50:08 +0200
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
 having no luck with fs_video
In-Reply-To: <CALXCt0rK5S22T=CD7Rw=KbfcHTU8oQJyvvYh70zSqRqW1yrwAw@mail.gmail.com>
References: <CAJdRzN5D40JKuN3MN82P5gij-=XGsufc4cktVKkO4cFwbASLKg@mail.gmail.com>
	<D18D04C5.9FA83%krice@freeswitch.org>
	<CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>
	<CALXCt0rK5S22T=CD7Rw=KbfcHTU8oQJyvvYh70zSqRqW1yrwAw@mail.gmail.com>
Message-ID: <CALXCt0ryDVDVMkoLZsYOZQRtBdDJzLTB=cX000QEgBD6e4wd0w@mail.gmail.com>

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

Please follow the instruction exactly, step by step, without any
difference. (eg: use Debian Jessie, and copy and paste from the page).

That will work.

>From there, you can begin doing modifications.


On Fri, May 29, 2015 at 3:47 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>
> On Fri, May 29, 2015 at 3:42 AM, Chris Mandra <mandra at gmail.com> wrote:
>
>> Hi Ken, thanks for the reply.
>> Does https://conference.freeswitch.org/verto/ run fs_video? I kind of
>> thought it was running an older version.
>>
>> I'm hoping to get some help from the community bc, well, I'm probably not
>> the only person with these problems and, like many people, I can't really
>> afford paid assistance.
>>
>> I have a few more questions:
>>
>> I'm getting the followingf messages I could use some clarification on:
>> 1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by
>> candidate to set remote rtcp audio addr to 205.215.241.76:59980 but this
>> is rtcp-mux so no thanks (is this affecting anything?)
>>
>> 2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video
>> DTLS state from OFF to HANDSHAKE
>> - it never gets beyond handshake - any ideas
>>
>> 3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending
>> video packet 0 bytes (ice not ready @ line 4241!)
>> - ice not ready - what's causing this?
>>
>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for
>> zrtp-hash
>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding
>> whether to pass zrtp-hash between legs
>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361
>> CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
>>
>> Should we be using ZRTP? if it weren't enabled at all would I see these
>> messages?
>>
>> Lastly, it looks like we're not getting any sip responses from FS - Any
>> ideas why?
>>
>> All answers/help/hints are appreciated.
>>
>> chris
>>
>>
>>
>>
>> On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org> wrote:
>>
>>>  Hi Chris,
>>>
>>> There is always https://conference.freeswitch.org/verto/ where we have
>>> been demoing this for quite a while
>>>
>>> If you need this working like now I would suggest emailing
>>> consulting at freeswitch.org where you can get paid realtime assistance
>>> from one of the Core Dev Team
>>>
>>> See you @ ClueCon!
>>> Ken
>>>
>>>
>>>
>>> *http://www.ClueCon.com <http://www.ClueCon.com>
>>> http://www.freeswitch.org <http://www.freeswitch.org> http://www.OSTAG.org
>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>> Twitter: @FreeSWITCH
>>>          @ClueCon
>>>
>>>
>>> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com> wrote:
>>>
>>> Hi guys. I had been able to use webrtc via chrome with FS 1.4 for
>>> connections and confenerce calls without incident (although the bandwidth
>>> and video quality seemed rather low as compared to the
>>> apprtc.appspot.com <http://apprtc.appspot.com>  demo.)
>>> I have followed the instructions as well I am able on
>>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>>> but so far I can't make an actual connection; the certs pass the cert
>>> test without issue, but I have no joy using webrtc and fs_video
>>>
>>> a few questions:
>>> 1. Is there any known problem using wildcard certs from godaddy?
>>> 2. is there other configuration beyond what's on the web page needed to
>>> make webrtc work? (I have the apropos ports open etc)
>>> 3. Are there any known issues with fs_video and sip.js?
>>> 4. I feel like it's getting hung op on dtls, as it never passes
>>> "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS
>>> state from OFF to HANDSHAKE"
>>> (it repeats that twice with:
>>> "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
>>> packet 0 bytes (ice not ready @ line 4241!)" not long after - does that
>>> speak to something obvious to you guys?
>>> 5. is there a working webrtc / fs_video demo I can look at?
>>>
>>> I realize this may all be solved by the release of 1.6, but I need this
>>> working now. I've been looking forward to this bc I know there have been
>>> some issues with webrtc, rtcp, freeswitch and solid BWE.
>>>
>>> Thanks for all you do,
>>> chris
>>>
>>>
>>>
>>> ------------------------------
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> Ken
>>>
>>>
>>>
>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>> Twitter: @FreeSWITCH
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From mandra at gmail.com  Fri May 29 06:02:54 2015
From: mandra at gmail.com (Chris Mandra)
Date: Thu, 28 May 2015 22:02:54 -0400
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
 having no luck with fs_video
In-Reply-To: <CALXCt0rK5S22T=CD7Rw=KbfcHTU8oQJyvvYh70zSqRqW1yrwAw@mail.gmail.com>
References: <CAJdRzN5D40JKuN3MN82P5gij-=XGsufc4cktVKkO4cFwbASLKg@mail.gmail.com>
	<D18D04C5.9FA83%krice@freeswitch.org>
	<CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>
	<CALXCt0rK5S22T=CD7Rw=KbfcHTU8oQJyvvYh70zSqRqW1yrwAw@mail.gmail.com>
Message-ID: <CAJdRzN5MoNqcZDVdZV9Vp94K4A7LpZ7hTJTiCAAgF+ZJsHAzVQ@mail.gmail.com>

Thanks Giovanni - I followed those instructions this afternoon. I feel like
the main issues are
1. never getting beyond dtls handhsake:

2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS
state from OFF to HANDSHAKE

2. not getting sip messages back from freeswitch ( we're using sip.js and
not verto. This was all working (more or less) with 1.4)

thanks,

chris




On Thu, May 28, 2015 at 9:47 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>
> On Fri, May 29, 2015 at 3:42 AM, Chris Mandra <mandra at gmail.com> wrote:
>
>> Hi Ken, thanks for the reply.
>> Does https://conference.freeswitch.org/verto/ run fs_video? I kind of
>> thought it was running an older version.
>>
>> I'm hoping to get some help from the community bc, well, I'm probably not
>> the only person with these problems and, like many people, I can't really
>> afford paid assistance.
>>
>> I have a few more questions:
>>
>> I'm getting the followingf messages I could use some clarification on:
>> 1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by
>> candidate to set remote rtcp audio addr to 205.215.241.76:59980 but this
>> is rtcp-mux so no thanks (is this affecting anything?)
>>
>> 2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video
>> DTLS state from OFF to HANDSHAKE
>> - it never gets beyond handshake - any ideas
>>
>> 3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending
>> video packet 0 bytes (ice not ready @ line 4241!)
>> - ice not ready - what's causing this?
>>
>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for
>> zrtp-hash
>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding
>> whether to pass zrtp-hash between legs
>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361
>> CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
>>
>> Should we be using ZRTP? if it weren't enabled at all would I see these
>> messages?
>>
>> Lastly, it looks like we're not getting any sip responses from FS - Any
>> ideas why?
>>
>> All answers/help/hints are appreciated.
>>
>> chris
>>
>>
>>
>>
>> On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org> wrote:
>>
>>>  Hi Chris,
>>>
>>> There is always https://conference.freeswitch.org/verto/ where we have
>>> been demoing this for quite a while
>>>
>>> If you need this working like now I would suggest emailing
>>> consulting at freeswitch.org where you can get paid realtime assistance
>>> from one of the Core Dev Team
>>>
>>> See you @ ClueCon!
>>> Ken
>>>
>>>
>>>
>>> *http://www.ClueCon.com <http://www.ClueCon.com>
>>> http://www.freeswitch.org <http://www.freeswitch.org> http://www.OSTAG.org
>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>> Twitter: @FreeSWITCH
>>>          @ClueCon
>>>
>>>
>>> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com> wrote:
>>>
>>> Hi guys. I had been able to use webrtc via chrome with FS 1.4 for
>>> connections and confenerce calls without incident (although the bandwidth
>>> and video quality seemed rather low as compared to the
>>> apprtc.appspot.com <http://apprtc.appspot.com>  demo.)
>>> I have followed the instructions as well I am able on
>>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>>> but so far I can't make an actual connection; the certs pass the cert
>>> test without issue, but I have no joy using webrtc and fs_video
>>>
>>> a few questions:
>>> 1. Is there any known problem using wildcard certs from godaddy?
>>> 2. is there other configuration beyond what's on the web page needed to
>>> make webrtc work? (I have the apropos ports open etc)
>>> 3. Are there any known issues with fs_video and sip.js?
>>> 4. I feel like it's getting hung op on dtls, as it never passes
>>> "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS
>>> state from OFF to HANDSHAKE"
>>> (it repeats that twice with:
>>> "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
>>> packet 0 bytes (ice not ready @ line 4241!)" not long after - does that
>>> speak to something obvious to you guys?
>>> 5. is there a working webrtc / fs_video demo I can look at?
>>>
>>> I realize this may all be solved by the release of 1.6, but I need this
>>> working now. I've been looking forward to this bc I know there have been
>>> some issues with webrtc, rtcp, freeswitch and solid BWE.
>>>
>>> Thanks for all you do,
>>> chris
>>>
>>>
>>>
>>> ------------------------------
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> Ken
>>>
>>>
>>>
>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>> Twitter: @FreeSWITCH
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>



-- 
mandra
c:410.258.5281
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From gmaruzz at gmail.com  Fri May 29 06:08:19 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 29 May 2015 04:08:19 +0200
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
 having no luck with fs_video
In-Reply-To: <CAJdRzN5MoNqcZDVdZV9Vp94K4A7LpZ7hTJTiCAAgF+ZJsHAzVQ@mail.gmail.com>
References: <CAJdRzN5D40JKuN3MN82P5gij-=XGsufc4cktVKkO4cFwbASLKg@mail.gmail.com>
	<D18D04C5.9FA83%krice@freeswitch.org>
	<CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>
	<CALXCt0rK5S22T=CD7Rw=KbfcHTU8oQJyvvYh70zSqRqW1yrwAw@mail.gmail.com>
	<CAJdRzN5MoNqcZDVdZV9Vp94K4A7LpZ7hTJTiCAAgF+ZJsHAzVQ@mail.gmail.com>
Message-ID: <CALXCt0rnb-_2_iDt8q1Vw-X=ugYgn-osfGR7AHBNA+ovyb-K2g@mail.gmail.com>

On Fri, May 29, 2015 at 4:02 AM, Chris Mandra <mandra at gmail.com> wrote:

> Thanks Giovanni - I followed those instructions this afternoon. I feel
> like the main issues are
> 1. never getting beyond dtls handhsake:
>
> 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS
> state from OFF to HANDSHAKE
>
> 2. not getting sip messages back from freeswitch ( we're using sip.js and
> not verto. This was all working (more or less) with 1.4)
>

it works with verto? make it first work with verto, so you can be sure all
the infrastructure is there, then go for sip.js


> thanks,
>
> chris
>
>
>
>
> On Thu, May 28, 2015 at 9:47 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>>
>> On Fri, May 29, 2015 at 3:42 AM, Chris Mandra <mandra at gmail.com> wrote:
>>
>>> Hi Ken, thanks for the reply.
>>> Does https://conference.freeswitch.org/verto/ run fs_video? I kind of
>>> thought it was running an older version.
>>>
>>> I'm hoping to get some help from the community bc, well, I'm probably
>>> not the only person with these problems and, like many people, I can't
>>> really afford paid assistance.
>>>
>>> I have a few more questions:
>>>
>>> I'm getting the followingf messages I could use some clarification on:
>>> 1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by
>>> candidate to set remote rtcp audio addr to 205.215.241.76:59980 but
>>> this is rtcp-mux so no thanks (is this affecting anything?)
>>>
>>> 2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video
>>> DTLS state from OFF to HANDSHAKE
>>> - it never gets beyond handshake - any ideas
>>>
>>> 3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending
>>> video packet 0 bytes (ice not ready @ line 4241!)
>>> - ice not ready - what's causing this?
>>>
>>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for
>>> zrtp-hash
>>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding
>>> whether to pass zrtp-hash between legs
>>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361
>>> CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
>>>
>>> Should we be using ZRTP? if it weren't enabled at all would I see these
>>> messages?
>>>
>>> Lastly, it looks like we're not getting any sip responses from FS - Any
>>> ideas why?
>>>
>>> All answers/help/hints are appreciated.
>>>
>>> chris
>>>
>>>
>>>
>>>
>>> On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>>>  Hi Chris,
>>>>
>>>> There is always https://conference.freeswitch.org/verto/ where we have
>>>> been demoing this for quite a while
>>>>
>>>> If you need this working like now I would suggest emailing
>>>> consulting at freeswitch.org where you can get paid realtime assistance
>>>> from one of the Core Dev Team
>>>>
>>>> See you @ ClueCon!
>>>> Ken
>>>>
>>>>
>>>>
>>>> *http://www.ClueCon.com <http://www.ClueCon.com>
>>>> http://www.freeswitch.org <http://www.freeswitch.org> http://www.OSTAG.org
>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>> Twitter: @FreeSWITCH
>>>>          @ClueCon
>>>>
>>>>
>>>> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com> wrote:
>>>>
>>>> Hi guys. I had been able to use webrtc via chrome with FS 1.4 for
>>>> connections and confenerce calls without incident (although the bandwidth
>>>> and video quality seemed rather low as compared to the
>>>> apprtc.appspot.com <http://apprtc.appspot.com>  demo.)
>>>> I have followed the instructions as well I am able on
>>>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>>>> but so far I can't make an actual connection; the certs pass the cert
>>>> test without issue, but I have no joy using webrtc and fs_video
>>>>
>>>> a few questions:
>>>> 1. Is there any known problem using wildcard certs from godaddy?
>>>> 2. is there other configuration beyond what's on the web page needed to
>>>> make webrtc work? (I have the apropos ports open etc)
>>>> 3. Are there any known issues with fs_video and sip.js?
>>>> 4. I feel like it's getting hung op on dtls, as it never passes
>>>> "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio
>>>> DTLS state from OFF to HANDSHAKE"
>>>> (it repeats that twice with:
>>>> "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending
>>>> video packet 0 bytes (ice not ready @ line 4241!)" not long after - does
>>>> that speak to something obvious to you guys?
>>>> 5. is there a working webrtc / fs_video demo I can look at?
>>>>
>>>> I realize this may all be solved by the release of 1.6, but I need this
>>>> working now. I've been looking forward to this bc I know there have been
>>>> some issues with webrtc, rtcp, freeswitch and solid BWE.
>>>>
>>>> Thanks for all you do,
>>>> chris
>>>>
>>>>
>>>>
>>>> ------------------------------
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> --
>>>> Ken
>>>>
>>>>
>>>>
>>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>> Twitter: @FreeSWITCH
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>
>
>
> --
> mandra
> c:410.258.5281
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From krice at freeswitch.org  Fri May 29 06:21:47 2015
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 May 2015 21:21:47 -0500
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
 having no luck with fs_video
In-Reply-To: <CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>
Message-ID: <D18D376B.9FA8C%krice@freeswitch.org>

Conference.freeswitch.org has been running this code for a while and its
used quite heavily. We use it every week for the past month or so for video
with ClueCon Weekly.

The point of paid assistance is you said you have to have this working now.
The best way to get it working right now is to get paid support.

Each of the following things could be any number of issues.
1. did you follow the setup documentation on Debian Jessie to the letter?
2. Are you using the Latest Chrome? (FireFox may work but  no guarentees, IE
and Safari don?t even support WebRTC)
3. theres not enough information to even start saying anything about FS not
responding to SIP messages... You?ll have to review the logs and configs to
figure out why 
4. As far as ZRTP goes do you need it? We cant answer that, only you can...
If you don?t need it you can disable it




On 5/28/15, 8:42 PM, "Chris Mandra" <mandra at gmail.com> wrote:

> Hi Ken, thanks for the reply.?
> Does?https://conference.freeswitch.org/verto/?run fs_video? I kind of thought
> it was running an older version.?
> 
> I'm hoping to get some help from the community bc, well, I'm probably not the
> only person with these problems and, like many people, I can't really afford
> paid assistance.?
> 
> I have a few more questions:?
> 
> I'm getting the followingf messages I could use some clarification on:
> 1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by
> candidate to set remote rtcp audio addr to 205.215.241.76:59980
> <http://205.215.241.76:59980>  but this is rtcp-mux so no thanks (is this
> affecting anything?)
> 
> 2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video DTLS
> state from OFF to HANDSHAKE?
> - it never gets beyond handshake - any ideas
> 3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
> packet 0 bytes (ice not ready @ line 4241!)
> - ice not ready - what's causing this?
> 
> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for
> zrtp-hash
> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding whether to
> pass zrtp-hash between legs
> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361
> CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
> 
> Should we be using ZRTP? if it weren't enabled at all would I see these
> messages?
> 
> Lastly, it looks like we're not getting any sip responses from FS - Any ideas
> why?
> 
> All answers/help/hints are appreciated.
> 
> chris
> 
> 
> 
> 
> 
> On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org> wrote:
>> Hi Chris,
>> 
>> There is always https://conference.freeswitch.org/verto/ where we have been
>> demoing this for quite a while
>> 
>> If you need this working like now I would suggest emailing
>> consulting at freeswitch.org <http://consulting at freeswitch.org>  where you can
>> get paid realtime assistance from one of the Core Dev Team
>> 
>> See you @ ClueCon!
>> Ken
>> http://www.ClueCon.com
>> http://www.freeswitch.org
>> http://www.OSTAG.org
>> irc.freenode.net <http://irc.freenode.net>  #freeswitch
>> Twitter: @FreeSWITCH
>> ?????????@ClueCon
>> 
>> 
>> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com
>> <http://mandra at gmail.com> > wrote:
>> 
>>> Hi guys. I had been able to use webrtc via chrome with FS 1.4 for
>>> connections and confenerce calls without incident (although the bandwidth
>>> and video quality seemed rather low as compared to the apprtc.appspot.com
>>> <http://apprtc.appspot.com>  <http://apprtc.appspot.com> ?demo.)
>>> I have followed the instructions as well I am able
>>> on?https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>>> but so far I can't make an actual connection; the certs pass the cert test
>>> without issue, but I have no joy using webrtc and fs_video
>>> 
>>> a few questions:?
>>> 1. Is there any known problem using wildcard certs from godaddy?
>>> 2. is there other configuration beyond what's on the web page needed to make
>>> webrtc work? (I have the apropos ports open etc)?
>>> 3. Are there any known issues with fs_video and sip.js?
>>> 4. I feel like it's getting hung op on dtls, as it never passes?
>>> "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS
>>> state from OFF to HANDSHAKE"
>>> (it repeats that twice with:
>>> "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
>>> packet 0 bytes (ice not ready @ line 4241!)" not long after - does that
>>> speak to something obvious to you guys?
>>> 5. is there a working webrtc / fs_video demo I can look at?
>>> 
>>> I realize this may all be solved by the release of 1.6, but I need this
>>> working now. I've been looking forward to this bc I know there have been
>>> some issues with webrtc, rtcp, freeswitch and solid BWE.?
>>> 
>>> Thanks for all you do,
>>> chris
>>> 
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From mike at jerris.com  Fri May 29 06:29:00 2015
From: mike at jerris.com (Michael Jerris)
Date: Thu, 28 May 2015 22:29:00 -0400
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
 having no luck with fs_video
In-Reply-To: <CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>
References: <CAJdRzN5D40JKuN3MN82P5gij-=XGsufc4cktVKkO4cFwbASLKg@mail.gmail.com>
	<D18D04C5.9FA83%krice@freeswitch.org>
	<CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>
Message-ID: <CAEwTsAicYcbMRx+JTj6k6kC0a5DAAaQt2VEKmXhjKQ4bcizvFw@mail.gmail.com>

On Thursday, May 28, 2015, Chris Mandra <mandra at gmail.com> wrote:

> Hi Ken, thanks for the reply.
> Does https://conference.freeswitch.org/verto/ run fs_video? I kind of
> thought it was running an older version.
>

This is always running current code daily.  We always dogfood our bleeding
edge code.


> I'm hoping to get some help from the community bc, well, I'm probably not
> the only person with these problems and, like many people, I can't really
> afford paid assistance.
>
> I have a few more questions:
>
> I'm getting the followingf messages I could use some clarification on:
> 1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by
> candidate to set remote rtcp audio addr to 205.215.241.76:59980 but this
> is rtcp-mux so no thanks (is this affecting anything?)
>
>
Your using chrome canary I think.  This is a bug in canary.  Our latest
code addresses this chrome bug.


> 2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video DTLS
> state from OFF to HANDSHAKE
> - it never gets beyond handshake - any ideas
>

Look at packet captures, maybe firewall blocking things?

> 3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
> packet 0 bytes (ice not ready @ line 4241!)
> - ice not ready - what's causing this?
>

This is because you don't have a dtls handshake yet.

> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for
> zrtp-hash
> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding
> whether to pass zrtp-hash between legs
> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361
> CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
>
> Should we be using ZRTP? if it weren't enabled at all would I see these
> messages?
>

Is one leg of your call passing in zrtp stuff ?  You would not be running
zrtp on a webrtc call as no webrtc end points I know of support it.

> Lastly, it looks like we're not getting any sip responses from FS - Any
> ideas why?
>
>
> Maybe firewall, maybe misconfigured ip or acl?  If you are using verto,
where does sip come in to play?




> All answers/help/hints are appreciated.
>
> chris
>
>
>
>
> On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>> wrote:
>
>>  Hi Chris,
>>
>> There is always https://conference.freeswitch.org/verto/ where we have
>> been demoing this for quite a while
>>
>> If you need this working like now I would suggest emailing
>> consulting at freeswitch.org where you can get paid realtime assistance
>> from one of the Core Dev Team
>>
>> See you @ ClueCon!
>> Ken
>>
>>
>>
>> *http://www.ClueCon.com <http://www.ClueCon.com>
>> http://www.freeswitch.org <http://www.freeswitch.org> http://www.OSTAG.org
>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>> Twitter: @FreeSWITCH
>>          @ClueCon
>>
>>
>> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com> wrote:
>>
>> Hi guys. I had been able to use webrtc via chrome with FS 1.4 for
>> connections and confenerce calls without incident (although the bandwidth
>> and video quality seemed rather low as compared to the apprtc.appspot.com
>> <http://apprtc.appspot.com>  demo.)
>> I have followed the instructions as well I am able on
>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>> but so far I can't make an actual connection; the certs pass the cert
>> test without issue, but I have no joy using webrtc and fs_video
>>
>> a few questions:
>> 1. Is there any known problem using wildcard certs from godaddy?
>> 2. is there other configuration beyond what's on the web page needed to
>> make webrtc work? (I have the apropos ports open etc)
>> 3. Are there any known issues with fs_video and sip.js?
>> 4. I feel like it's getting hung op on dtls, as it never passes
>> "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS
>> state from OFF to HANDSHAKE"
>> (it repeats that twice with:
>> "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
>> packet 0 bytes (ice not ready @ line 4241!)" not long after - does that
>> speak to something obvious to you guys?
>> 5. is there a working webrtc / fs_video demo I can look at?
>>
>> I realize this may all be solved by the release of 1.6, but I need this
>> working now. I've been looking forward to this bc I know there have been
>> some issues with webrtc, rtcp, freeswitch and solid BWE.
>>
>> Thanks for all you do,
>> chris
>>
>>
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> Ken
>>
>>
>>
>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>> Twitter: @FreeSWITCH
>>
>>
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From yehavi.bourvine at gmail.com  Fri May 29 08:02:29 2015
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 29 May 2015 07:02:29 +0300
Subject: [Freeswitch-users] Some calls have one-way audio
In-Reply-To: <CALXCt0rKtkqjE-UmuqPDkKVU=N0Bq_p_yhiciXGJ=E9SdhVfjg@mail.gmail.com>
References: <CAAtWKeHoqAq_tdEkyAw26Rsr+8Y_8R8H1mkD7SEQiU1+e_ffnQ@mail.gmail.com>
	<CALXCt0rqN_kp4JE_YokbxiNVYW4OGcPCnjzhrujB8OETTVVVLw@mail.gmail.com>
	<CAAtWKeGEEOBbi4xOm5uEr3kg8GSptsG0iN=-cm9EA-A8xXbmQg@mail.gmail.com>
	<CALXCt0rKtkqjE-UmuqPDkKVU=N0Bq_p_yhiciXGJ=E9SdhVfjg@mail.gmail.com>
Message-ID: <CAAtWKeEbCGs3BhQFz8019zE7BrT4GMXV-R2iMF0L8OZCLF8MJA@mail.gmail.com>

Hello all,

  I didn't mean to ask help debugging this over the list, as this is not
the place to do it. I just wanted to know whether this is a known
phenomenon; I see it is not, so I'll invest the time gathering data in
order to open a JIRA.

I have a side question: When openning a JIRA I have to state that this
happens with the latest GIT. However, I am running the production release
1.4 version (updating today to 1.4.19). Since this is a production system,
and since I cannot replicate it on a test system, how do I get out from
this deadlock? (deadlock = I want to run a production version, but need a
GIT HEAD to report the problem...).

           Thanks! __Yehavi:

2015-05-28 15:30 GMT+03:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:

> Yehavi,
>
> your description is not clear at all, at least to me.
>
> Can you explain exactly what happens, what's your configuration, etc?
>
> Also, I'm almost sure you will end up having to provide us with sip traces
> of failed (one way) calls, so I would counseil you to begin using
> pcapsipdump
>
> -giovanni
>
>
> On Thu, May 28, 2015 at 2:25 PM, Yehavi Bourvine <
> yehavi.bourvine at gmail.com> wrote:
>
>> Hi,
>>
>>   We are not using NAT. All the voice goes through the Freeswitch server,
>> and it relays the audio between the internal phones and outside.
>> The problem is that the outside leg is using SRTP, so I cannot hear the
>> voice there...
>>
>>                  Thanks, __Yehavi:
>>
>> 2015-05-28 9:30 GMT+03:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>>
>>> Hello,
>>>
>>> is almost certainly a NAT problem.
>>> Use something like pcapsipdump, take trace of all your calls, then when
>>> a call with one-way audio happens, look into its pcap.
>>>
>>> -giovanni
>>>
>>>
>>> On Thu, May 28, 2015 at 8:20 AM, Yehavi Bourvine <
>>> yehavi.bourvine at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>>   Before I dig into this issue and start collecting data in order to
>>>> open a Jira: Has anyone else noticed it? When I call or get a call
>>>> sometimes there is one way audio. Hanging up and calling again usually
>>>> solves it. It happens quite rarely, so I cannot reproduce it at will.
>>>> However, it annoys the users...
>>>>
>>>> I am running FreeSwitch 1.4.18 and use various phones (Polycom,
>>>> Yealink, SNOM, etc.).
>>>>
>>>>                   Thanks! __Yehavi:
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 May 29 08:11:35 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 29 May 2015 06:11:35 +0200
Subject: [Freeswitch-users] Some calls have one-way audio
In-Reply-To: <CAAtWKeEbCGs3BhQFz8019zE7BrT4GMXV-R2iMF0L8OZCLF8MJA@mail.gmail.com>
References: <CAAtWKeHoqAq_tdEkyAw26Rsr+8Y_8R8H1mkD7SEQiU1+e_ffnQ@mail.gmail.com>
	<CALXCt0rqN_kp4JE_YokbxiNVYW4OGcPCnjzhrujB8OETTVVVLw@mail.gmail.com>
	<CAAtWKeGEEOBbi4xOm5uEr3kg8GSptsG0iN=-cm9EA-A8xXbmQg@mail.gmail.com>
	<CALXCt0rKtkqjE-UmuqPDkKVU=N0Bq_p_yhiciXGJ=E9SdhVfjg@mail.gmail.com>
	<CAAtWKeEbCGs3BhQFz8019zE7BrT4GMXV-R2iMF0L8OZCLF8MJA@mail.gmail.com>
Message-ID: <CALXCt0o02qGJWip5FZWcEdfDMoN0fR5iQ4bNZ8bcAZjiM8tAeg@mail.gmail.com>

On Fri, May 29, 2015 at 6:02 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
wrote:

>
> I have a side question: When openning a JIRA I have to state that this
> happens with the latest GIT. However, I am running the production release
> 1.4 version (updating today to 1.4.19). Since this is a production system,
> and since I cannot replicate it on a test system, how do I get out from
> this deadlock? (deadlock = I want to run a production version, but need a
> GIT HEAD to report the problem...).
>

 If you report for latest git in 1.4 branch (that's in this moment is
1.4.19) is ok

-giovanni
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From christian.hoene at symonics.com  Fri May 29 08:07:59 2015
From: christian.hoene at symonics.com (Christian Hoene)
Date: Fri, 29 May 2015 06:07:59 +0200
Subject: [Freeswitch-users] The 1.6 video branch has been merged
	into	master!
In-Reply-To: <CADK-BjuyMD74Epn=P5zviUGOKkKC+4ddNP_HXkRCOrpmFvy0nQ@mail.gmail.com>
References: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>	<CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>
	<CADK-BjuyMD74Epn=P5zviUGOKkKC+4ddNP_HXkRCOrpmFvy0nQ@mail.gmail.com>
Message-ID: <01d301d099c5$0d330980$27991c80$@symonics.com>

Congratulations!

Good work!

 

Christian Hoene

 

 

Von: Blake Priddy [mailto:bpriddy at bryantschools.org] 
Gesendet: Donnerstag, 28. Mai 2015 22:49
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] The 1.6 video branch has been merged into master!

 

Great work guys!

 

On Thu, May 28, 2015 at 3:45 PM, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com> > wrote:

Yay!!!

All hail our FreeSWITCH developers and core team !

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18

On May 28, 2015 9:56 PM, "Ken Rice" <krice at freeswitch.org <mailto:krice at freeswitch.org> > wrote:

New Post on freeswitch.org <http://freeswitch.org>  from Kathleen
check it out at http://ift.tt/1RrHZnX
The 1.6 video branch has been merged into master!

Great news for our fellow FreeSWITCH users: in preparation for the 1.6 beta release, a lot of new functionality is being merged into the master branch. Some of these new features have new build requirements and dependencies, so please be sure to check the Confluence link here: http://ift.tt/1d2d4iX for platform specific instructions. Building and running FreeSWITCH will be easier if you are using Debian 8(Jessie). If you are not using Debian you can find some of the supporting dependencies tar.gz files here: http://ift.tt/1qAFKnH

Additionally, if you are using 1.4 in production, you need to be sure to switch to tracking the v1.4 branch, as master will be preparing for the 1.6 beta release.

Some of the new features and work that have gone into this release include:
FS-7499 <http://ift.tt/1d2d2HU>  core RTCP improvements
FS-7500 <http://ift.tt/1PQJHBY>  core video transcoding support
FS-7501 <http://ift.tt/1d2d2HW>  core video jitterbuffer
FS-7502 <http://ift.tt/1PQJFdn>  core video media bugs
FS-7503 <http://ift.tt/1d2d4iZ>  core file interface video support
FS-7504 <http://ift.tt/1PQJHC2>  codecs let you choose which codec module to use
FS-7505 <http://ift.tt/1d2d4j1>  file interface to let you specify which format module to use when multiple types are supported
FS-7506 <http://ift.tt/1PQJHC4>  core text rendering
FS-7507 <http://ift.tt/1d2d4j3>  added new global directory variables and configure directory behavior changes
FS-7508 <http://ift.tt/1d2d2HY>  mod_vpx transcoding vp8/vp9 and replace mod_v8
FS-7509 <http://ift.tt/1PQJHC6>  mod_verto improvements allow for desktop share with the installation of this chrome extension http://ift.tt/1kQP2pS and improved bandwith and resolution handling.
FS-7512 <http://ift.tt/1PQJHCa>  mod_png allows for image overlays for logos and images for video mute
FS-7513 <http://ift.tt/1d2d4j5>  mod_conference MCU feature and avatar support
FS-7514 <http://ift.tt/1PQJI8Z>  mod_vlc video support allows you to live stream, record calls to a video file, and playback videos into a call.
FS-7515 <http://ift.tt/1d2d2Yc>  mod_cv is a video media bug module that uses video recognition and facial recognition technology to allow you to modify a video stream by adding overlapping images and text or to silently detect and fire events
FS-7516 <http://ift.tt/1PQJHSo>  mod_imagick allows for PDF and GIF rendered as video
FS-7517 <http://ift.tt/1d2d2Ye>  mod_openh264 h264 codec module
FS-7519 <http://ift.tt/1d2d4zl>  mod_av a file format and codec module that uses libav or ffmpeg
FS-7494 <http://ift.tt/1PQJHSs>  default avatar and mute images for video MCU
FS-7471 <http://ift.tt/1d2d4zn>  improved configs for video
FS-7338 <http://ift.tt/1PQJI97>  removed external library dependencies
FS-7585 <http://ift.tt/1d2d4zr>  added video support to mod_rtmp

 

_________________________________________________________________________
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





 

-- 

 

 

Blakelund Priddy

Network & Systems Engineer
Bryant Public School District
Bryant, Arkansas 72022
 <http://www.bryantschools.org/> http://www.bryantschools.org

p 501-653-5038
f 501-847-5656

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From osblinnikov at gmail.com  Fri May 29 11:57:51 2015
From: osblinnikov at gmail.com (Oleg Blinnikov)
Date: Fri, 29 May 2015 09:57:51 +0200
Subject: [Freeswitch-users] The 1.6 video branch has been merged into
	master!
In-Reply-To: <01d301d099c5$0d330980$27991c80$@symonics.com>
References: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>
	<CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>
	<CADK-BjuyMD74Epn=P5zviUGOKkKC+4ddNP_HXkRCOrpmFvy0nQ@mail.gmail.com>
	<01d301d099c5$0d330980$27991c80$@symonics.com>
Message-ID: <CA+M05dCkyO-qR3j+OvgU9k9Zu7xxD77nRrqVFvHa1SkGRr5Kuw@mail.gmail.com>

Impressive improvements, congrats! I gonna try it right away!

On Fri, May 29, 2015 at 6:07 AM, Christian Hoene <
christian.hoene at symonics.com> wrote:

> Congratulations!
>
> Good work!
>
>
>
> Christian Hoene
>
>
>
>
>
> *Von:* Blake Priddy [mailto:bpriddy at bryantschools.org]
> *Gesendet:* Donnerstag, 28. Mai 2015 22:49
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] The 1.6 video branch has been merged
> into master!
>
>
>
> Great work guys!
>
>
>
> On Thu, May 28, 2015 at 3:45 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
> Yay!!!
>
> All hail our FreeSWITCH developers and core team !
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
>
> On May 28, 2015 9:56 PM, "Ken Rice" <krice at freeswitch.org> wrote:
>
> New Post on freeswitch.org from Kathleen
> check it out at http://ift.tt/1RrHZnX
> The 1.6 video branch has been merged into master!
>
> Great news for our fellow FreeSWITCH users: in preparation for the 1.6
> beta release, a lot of new functionality is being merged into the master
> branch. Some of these new features have new build requirements and
> dependencies, so please be sure to check the Confluence link here:
> http://ift.tt/1d2d4iX for platform specific instructions. Building and
> running FreeSWITCH will be easier if you are using Debian 8(Jessie). If you
> are not using Debian you can find some of the supporting dependencies
> tar.gz files here: http://ift.tt/1qAFKnH
>
> Additionally, if you are using 1.4 in production, you need to be sure to
> switch to tracking the v1.4 branch, as master will be preparing for the 1.6
> beta release.
>
> Some of the new features and work that have gone into this release include:
> FS-7499 <http://ift.tt/1d2d2HU> core RTCP improvements
> FS-7500 <http://ift.tt/1PQJHBY> core video transcoding support
> FS-7501 <http://ift.tt/1d2d2HW> core video jitterbuffer
> FS-7502 <http://ift.tt/1PQJFdn> core video media bugs
> FS-7503 <http://ift.tt/1d2d4iZ> core file interface video support
> FS-7504 <http://ift.tt/1PQJHC2> codecs let you choose which codec module
> to use
> FS-7505 <http://ift.tt/1d2d4j1> file interface to let you specify which
> format module to use when multiple types are supported
> FS-7506 <http://ift.tt/1PQJHC4> core text rendering
> FS-7507 <http://ift.tt/1d2d4j3> added new global directory variables and
> configure directory behavior changes
> FS-7508 <http://ift.tt/1d2d2HY> mod_vpx transcoding vp8/vp9 and replace
> mod_v8
> FS-7509 <http://ift.tt/1PQJHC6> mod_verto improvements allow for desktop
> share with the installation of this chrome extension http://ift.tt/1kQP2pS
> and improved bandwith and resolution handling.
> FS-7512 <http://ift.tt/1PQJHCa> mod_png allows for image overlays for
> logos and images for video mute
> FS-7513 <http://ift.tt/1d2d4j5> mod_conference MCU feature and avatar
> support
> FS-7514 <http://ift.tt/1PQJI8Z> mod_vlc video support allows you to live
> stream, record calls to a video file, and playback videos into a call.
> FS-7515 <http://ift.tt/1d2d2Yc> mod_cv is a video media bug module that
> uses video recognition and facial recognition technology to allow you to
> modify a video stream by adding overlapping images and text or to silently
> detect and fire events
> FS-7516 <http://ift.tt/1PQJHSo> mod_imagick allows for PDF and GIF
> rendered as video
> FS-7517 <http://ift.tt/1d2d2Ye> mod_openh264 h264 codec module
> FS-7519 <http://ift.tt/1d2d4zl> mod_av a file format and codec module
> that uses libav or ffmpeg
> FS-7494 <http://ift.tt/1PQJHSs> default avatar and mute images for video
> MCU
> FS-7471 <http://ift.tt/1d2d4zn> improved configs for video
> FS-7338 <http://ift.tt/1PQJI97> removed external library dependencies
> FS-7585 <http://ift.tt/1d2d4zr> added video support to mod_rtmp
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> [image: Das Bild wurde vom Absender entfernt.] [image: Das Bild wurde vom
> Absender entfernt.]
>
>
>
> *Blakelund Priddy*
>
> Network & Systems Engineer
> Bryant Public School District
> Bryant, Arkansas 72022
> http://www.bryantschools.org
>
> p 501-653-5038
> f 501-847-5656
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Oleg Blinnikov
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From yadenis at seznam.cz  Fri May 29 16:30:43 2015
From: yadenis at seznam.cz (Denis Jakovlev)
Date: Fri, 29 May 2015 14:30:43 +0200
Subject: [Freeswitch-users] CentOS instalation problem
In-Reply-To: <01d301d099c5$0d330980$27991c80$@symonics.com>
References: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail> 
	<CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>
	<CADK-BjuyMD74Epn=P5zviUGOKkKC+4ddNP_HXkRCOrpmFvy0nQ@mail.gmail.com>
	<01d301d099c5$0d330980$27991c80$@symonics.com>
Message-ID: <972336332.20150529143043@seznam.cz>

Hi all,

I'm trying to do a clean install of this manual. 

https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7  

and after "make -f Makefile.centos7" I have an error.

checking whether to include odbc... no
checking for uuid >= 1.41.2... checking for uuid... configure: WARNING: libuuid development package highly recommended!
  setting PLATFORM_CORE_LIBS to "-ldl -lcrypt -lrt"
checking for inflateReset in -lz... yes
  adding "-lz" to PLATFORM_CORE_LIBS
checking for libyuv >= 0.0.1280... configure: error: You need to install libyuv-dev. Required library
make: *** [freeswitch.git/Makefile] Error 1


What am I doing wrong?


-- 
S pozdravem,
Ing.Denis Jakovlev                           
mob.tel. 775-415-382
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From mbrancaleoni at voismart.it  Fri May 29 16:53:40 2015
From: mbrancaleoni at voismart.it (Matteo)
Date: Fri, 29 May 2015 14:53:40 +0200 (CEST)
Subject: [Freeswitch-users] CentOS instalation problem
In-Reply-To: <972336332.20150529143043@seznam.cz>
References: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>
	<CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>
	<CADK-BjuyMD74Epn=P5zviUGOKkKC+4ddNP_HXkRCOrpmFvy0nQ@mail.gmail.com>
	<01d301d099c5$0d330980$27991c80$@symonics.com>
	<972336332.20150529143043@seznam.cz>
Message-ID: <265414108.21345.1432904020795.JavaMail.zimbra@voismart.it>

Hi,

----- Il 29-mag-15, alle 14:30, Denis Jakovlev yadenis at seznam.cz ha scritto:

> CentOS instalation problem Hi all,
> 
> I'm trying to do a clean install of this manual.
> 
> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7
> 
> and after "make -f Makefile.centos7" I have an error.
> 
> checking whether to include odbc... no
> checking for uuid >= 1.41.2... checking for uuid... configure: WARNING: libuuid
> development package highly recommended!
> setting PLATFORM_CORE_LIBS to "-ldl -lcrypt -lrt"
> checking for inflateReset in -lz... yes
> adding "-lz" to PLATFORM_CORE_LIBS
> checking for libyuv >= 0.0.1280... configure: error: You need to install
> libyuv-dev. Required library
> make: *** [freeswitch.git/Makefile] Error 1


Nothing,

simply there's something to be addressed when building in non-debian distros.

Right now the only answer is to build & compile libyuv (and others) by yourself,
and then build FS.

Check https://freeswitch.org/jira/browse/FS-7553 for what is going on.

Pay attention that some libs may conflict with installed ones, like libvpx .

You may check also
https://freeswitch.org/stash/projects/SD

to checkout the specs, create the needed rpms and try.

I would say as today building FS (with video) outside debian is not for the faint of heart :)

Hopefully the FS team will build everything for us, but needs some time I guess.

Personally I'm trying to figure out why not including the libs inside FS
like in the past... at least the most problematic ones.

After all (like libvpx) libs have moved out of tree, but they conflict,
so gets renamed and rebuilt as a specific packages...

By not leveraging on distro libs, is not simpler to keep them inside?

Just my 2c :)

Reagards,
Matteo


From krice at freeswitch.org  Fri May 29 18:01:01 2015
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 29 May 2015 14:01:01 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <5568711d9e28a_d20082f320781d6@resque-worker-high.3.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From royj at yandex.ru  Fri May 29 18:10:01 2015
From: royj at yandex.ru (royj at yandex.ru)
Date: Fri, 29 May 2015 17:10:01 +0300
Subject: [Freeswitch-users] Displaying number of incoming calls
Message-ID: <4946501432908601@web6h.yandex.ru>

Hi, all

There is an application 'bridge' with data '[leg_timeout=15]user/100|user/100,sofia/gateway/gateway_name/2000000'.
The idea is to call '100' within 15 seconds and then, if no answer, to call '100' and '2000000' simultaneously. The application above works fine, but 100's phone displays two incoming call. This is clear, because after '[leg_timeout=15]user/100' FreeSWITCH sends CANCEL and then again INVITE to 100.
Is there any way/practice to perform the idea with one incoming call displayed on 100's phone?


From mandra at gmail.com  Fri May 29 18:19:09 2015
From: mandra at gmail.com (Chris Mandra)
Date: Fri, 29 May 2015 10:19:09 -0400
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
 having no luck with fs_video
In-Reply-To: <D18D376B.9FA8C%krice@freeswitch.org>
References: <CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>
	<D18D376B.9FA8C%krice@freeswitch.org>
Message-ID: <CAJdRzN6NYM8jsMWVkbQ_a9pRef1Q-bLOorfnxHLOWP31kEWXfw@mail.gmail.com>

Thanks for the thoughtful response Ken. Answers inline.

On Thu, May 28, 2015 at 10:21 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Conference.freeswitch.org has been running this code for a while and its
> used quite heavily. We use it every week for the past month or so for video
> with ClueCon Weekly.
>
> The point of paid assistance is you said you have to have this working
> now. The best way to get it working right now is to get paid support.
>
> Each of the following things could be any number of issues.
>
>    1. did you follow the setup documentation on Debian Jessie to the
>    letter?
>
> yup


>
>    1.
>    2. Are you using the Latest Chrome? (FireFox may work but  no
>    guarentees, IE and Safari don?t even support WebRTC)
>
> yup

>
>    1.
>    2. theres not enough information to even start saying anything about
>    FS not responding to SIP messages... You?ll have to review the logs and
>    configs to figure out why
>
> ok - I'm going to try using the FS verto demo to troubleshoot and see if
my problems are with sip.js


>
>    1.
>    2. As far as ZRTP goes do you need it? We cant answer that, only you
>    can... If you don?t need it you can disable it
>
> Prolly not. It has to be explicitly compiled into the build, right?


>
>
>
> On 5/28/15, 8:42 PM, "Chris Mandra" <mandra at gmail.com> wrote:
>
> Hi Ken, thanks for the reply.
> Does https://conference.freeswitch.org/verto/ run fs_video? I kind of
> thought it was running an older version.
>
> I'm hoping to get some help from the community bc, well, I'm probably not
> the only person with these problems and, like many people, I can't really
> afford paid assistance.
>
> I have a few more questions:
>
> I'm getting the followingf messages I could use some clarification on:
> 1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by
> candidate to set remote rtcp audio addr to 205.215.241.76:59980 <
> http://205.215.241.76:59980>  but this is rtcp-mux so no thanks (is this
> affecting anything?)
>
> 2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video DTLS
> state from OFF to HANDSHAKE
> - it never gets beyond handshake - any ideas
> 3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
> packet 0 bytes (ice not ready @ line 4241!)
> - ice not ready - what's causing this?
>
> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for
> zrtp-hash
> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding
> whether to pass zrtp-hash between legs
> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361
> CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
>
> Should we be using ZRTP? if it weren't enabled at all would I see these
> messages?
>
> Lastly, it looks like we're not getting any sip responses from FS - Any
> ideas why?
>
> All answers/help/hints are appreciated.
>
> chris
>
>
>
>
>
> On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org> wrote:
>
> Hi Chris,
>
> There is always https://conference.freeswitch.org/verto/ where we have
> been demoing this for quite a while
>
> If you need this working like now I would suggest emailing
> consulting at freeswitch.org <http://consulting at freeswitch.org>  where you
> can get paid realtime assistance from one of the Core Dev Team
>
> See you @ ClueCon!
> Ken
>
>
>
> *http://www.ClueCon.com <http://www.ClueCon.com> http://www.freeswitch.org
> <http://www.freeswitch.org> http://www.OSTAG.org <http://www.OSTAG.org> *
> irc.freenode.net <http://irc.freenode.net>  #freeswitch
> Twitter: @FreeSWITCH
>          @ClueCon
>
>
> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com <
> http://mandra at gmail.com> > wrote:
>
> Hi guys. I had been able to use webrtc via chrome with FS 1.4 for
> connections and confenerce calls without incident (although the bandwidth
> and video quality seemed rather low as compared to the apprtc.appspot.com
> <http://apprtc.appspot.com>  <http://apprtc.appspot.com>  demo.)
> I have followed the instructions as well I am able on
> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
> but so far I can't make an actual connection; the certs pass the cert test
> without issue, but I have no joy using webrtc and fs_video
>
> a few questions:
> 1. Is there any known problem using wildcard certs from godaddy?
> 2. is there other configuration beyond what's on the web page needed to
> make webrtc work? (I have the apropos ports open etc)
> 3. Are there any known issues with fs_video and sip.js?
> 4. I feel like it's getting hung op on dtls, as it never passes
> "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS
> state from OFF to HANDSHAKE"
> (it repeats that twice with:
> "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
> packet 0 bytes (ice not ready @ line 4241!)" not long after - does that
> speak to something obvious to you guys?
> 5. is there a working webrtc / fs_video demo I can look at?
>
> I realize this may all be solved by the release of 1.6, but I need this
> working now. I've been looking forward to this bc I know there have been
> some issues with webrtc, rtcp, freeswitch and solid BWE.
>
> Thanks for all you do,
> chris
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <http://consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <
> http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
>
>
>
> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
mandra
c:410.258.5281
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From mandra at gmail.com  Fri May 29 18:22:50 2015
From: mandra at gmail.com (Chris Mandra)
Date: Fri, 29 May 2015 10:22:50 -0400
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
 having no luck with fs_video
In-Reply-To: <CAEwTsAicYcbMRx+JTj6k6kC0a5DAAaQt2VEKmXhjKQ4bcizvFw@mail.gmail.com>
References: <CAJdRzN5D40JKuN3MN82P5gij-=XGsufc4cktVKkO4cFwbASLKg@mail.gmail.com>
	<D18D04C5.9FA83%krice@freeswitch.org>
	<CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>
	<CAEwTsAicYcbMRx+JTj6k6kC0a5DAAaQt2VEKmXhjKQ4bcizvFw@mail.gmail.com>
Message-ID: <CAJdRzN7ODcqKcH5LOa3GDVefQ5QMkw-qcSy1kArVCKe3rRncFg@mail.gmail.com>

Thanks for this Mike. I'm going to test with verto (using sip.js currently.)
other responses inline.

On Thu, May 28, 2015 at 10:29 PM, Michael Jerris <mike at jerris.com> wrote:

>
>
> On Thursday, May 28, 2015, Chris Mandra <mandra at gmail.com> wrote:
>
>> Hi Ken, thanks for the reply.
>> Does https://conference.freeswitch.org/verto/ run fs_video? I kind of
>> thought it was running an older version.
>>
>
> This is always running current code daily.  We always dogfood our bleeding
> edge code.
>
> good to know. thanks!

>
>> I'm hoping to get some help from the community bc, well, I'm probably not
>> the only person with these problems and, like many people, I can't really
>> afford paid assistance.
>>
>> I have a few more questions:
>>
>> I'm getting the followingf messages I could use some clarification on:
>> 1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by
>> candidate to set remote rtcp audio addr to 205.215.241.76:59980 but this
>> is rtcp-mux so no thanks (is this affecting anything?)
>>
>>
> Your using chrome canary I think.  This is a bug in canary.  Our latest
> code addresses this chrome bug.
>

great.

>
>
>> 2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video
>> DTLS state from OFF to HANDSHAKE
>> - it never gets beyond handshake - any ideas
>>
>
> Look at packet captures, maybe firewall blocking things?
>
I'll check.

> 3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
>> packet 0 bytes (ice not ready @ line 4241!)
>> - ice not ready - what's causing this?
>>
>
> This is because you don't have a dtls handshake yet.
>
makes sense

> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for
>> zrtp-hash
>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding
>> whether to pass zrtp-hash between legs
>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361
>> CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
>>
>> Should we be using ZRTP? if it weren't enabled at all would I see these
>> messages?
>>
>
> Is one leg of your call passing in zrtp stuff ?  You would not be running
> zrtp on a webrtc call as no webrtc end points I know of support it.
>
I don't think so, but I see messages abut it. I've seen docs mention it has
to be turned on in the build - I haven't done that. Does it default to off
still?

> Lastly, it looks like we're not getting any sip responses from FS - Any
>> ideas why?
>>
>>
>> Maybe firewall, maybe misconfigured ip or acl?  If you are using verto,
> where does sip come in to play?
>
> using sip.js and not verto. will try the fs demo with verto.

>
>
>
>> All answers/help/hints are appreciated.
>>
>> chris
>>
>>
>>
>>
>> On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org> wrote:
>>
>>>  Hi Chris,
>>>
>>> There is always https://conference.freeswitch.org/verto/ where we have
>>> been demoing this for quite a while
>>>
>>> If you need this working like now I would suggest emailing
>>> consulting at freeswitch.org where you can get paid realtime assistance
>>> from one of the Core Dev Team
>>>
>>> See you @ ClueCon!
>>> Ken
>>>
>>>
>>>
>>> *http://www.ClueCon.com <http://www.ClueCon.com>
>>> http://www.freeswitch.org <http://www.freeswitch.org> http://www.OSTAG.org
>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>> Twitter: @FreeSWITCH
>>>          @ClueCon
>>>
>>>
>>> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com> wrote:
>>>
>>> Hi guys. I had been able to use webrtc via chrome with FS 1.4 for
>>> connections and confenerce calls without incident (although the bandwidth
>>> and video quality seemed rather low as compared to the
>>> apprtc.appspot.com <http://apprtc.appspot.com>  demo.)
>>> I have followed the instructions as well I am able on
>>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>>> but so far I can't make an actual connection; the certs pass the cert
>>> test without issue, but I have no joy using webrtc and fs_video
>>>
>>> a few questions:
>>> 1. Is there any known problem using wildcard certs from godaddy?
>>> 2. is there other configuration beyond what's on the web page needed to
>>> make webrtc work? (I have the apropos ports open etc)
>>> 3. Are there any known issues with fs_video and sip.js?
>>> 4. I feel like it's getting hung op on dtls, as it never passes
>>> "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio DTLS
>>> state from OFF to HANDSHAKE"
>>> (it repeats that twice with:
>>> "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending video
>>> packet 0 bytes (ice not ready @ line 4241!)" not long after - does that
>>> speak to something obvious to you guys?
>>> 5. is there a working webrtc / fs_video demo I can look at?
>>>
>>> I realize this may all be solved by the release of 1.6, but I need this
>>> working now. I've been looking forward to this bc I know there have been
>>> some issues with webrtc, rtcp, freeswitch and solid BWE.
>>>
>>> Thanks for all you do,
>>> chris
>>>
>>>
>>>
>>> ------------------------------
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> Ken
>>>
>>>
>>>
>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>> Twitter: @FreeSWITCH
>>>
>>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
mandra
c:410.258.5281
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From mike at jerris.com  Fri May 29 18:31:30 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 29 May 2015 10:31:30 -0400
Subject: [Freeswitch-users] Displaying number of incoming calls
In-Reply-To: <4946501432908601@web6h.yandex.ru>
References: <4946501432908601@web6h.yandex.ru>
Message-ID: <CAEwTsAjXOrqk-DG5vLYJZ0zoL-Ss9gjRnH66440fUt=rOcdWbQ@mail.gmail.com>

there is another var either leg_delay or leg_delay_start that will make it
not start calling a leg until that timeout

On Friday, May 29, 2015, royj at yandex.ru <royj at yandex.ru> wrote:

> Hi, all
>
> There is an application 'bridge' with data
> '[leg_timeout=15]user/100|user/100,sofia/gateway/gateway_name/2000000'.
> The idea is to call '100' within 15 seconds and then, if no answer, to
> call '100' and '2000000' simultaneously. The application above works fine,
> but 100's phone displays two incoming call. This is clear, because after
> '[leg_timeout=15]user/100' FreeSWITCH sends CANCEL and then again INVITE to
> 100.
> Is there any way/practice to perform the idea with one incoming call
> displayed on 100's phone?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri May 29 18:32:55 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 29 May 2015 10:32:55 -0400
Subject: [Freeswitch-users] WEBRTC was working with 1.4,
 having no luck with fs_video
In-Reply-To: <CAJdRzN7ODcqKcH5LOa3GDVefQ5QMkw-qcSy1kArVCKe3rRncFg@mail.gmail.com>
References: <CAJdRzN5D40JKuN3MN82P5gij-=XGsufc4cktVKkO4cFwbASLKg@mail.gmail.com>
	<D18D04C5.9FA83%krice@freeswitch.org>
	<CAJdRzN6QqLFmiq3DSvKxnw_+wHWQtmKQvSgDZ3_P6zbsYOp2mw@mail.gmail.com>
	<CAEwTsAicYcbMRx+JTj6k6kC0a5DAAaQt2VEKmXhjKQ4bcizvFw@mail.gmail.com>
	<CAJdRzN7ODcqKcH5LOa3GDVefQ5QMkw-qcSy1kArVCKe3rRncFg@mail.gmail.com>
Message-ID: <CAEwTsAgneqMM1p-=mP3TqP_EBfcudacs0GzyrHF3-eq7Sd2fhA@mail.gmail.com>

yes still defaults to off, but I think we still do pass through even when
it's built without it

On Friday, May 29, 2015, Chris Mandra <mandra at gmail.com> wrote:

> Thanks for this Mike. I'm going to test with verto (using sip.js
> currently.)
> other responses inline.
>
> On Thu, May 28, 2015 at 10:29 PM, Michael Jerris <mike at jerris.com
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>
>>
>>
>> On Thursday, May 28, 2015, Chris Mandra <mandra at gmail.com
>> <javascript:_e(%7B%7D,'cvml','mandra at gmail.com');>> wrote:
>>
>>> Hi Ken, thanks for the reply.
>>> Does https://conference.freeswitch.org/verto/ run fs_video? I kind of
>>> thought it was running an older version.
>>>
>>
>> This is always running current code daily.  We always dogfood our
>> bleeding edge code.
>>
>> good to know. thanks!
>
>>
>>> I'm hoping to get some help from the community bc, well, I'm probably
>>> not the only person with these problems and, like many people, I can't
>>> really afford paid assistance.
>>>
>>> I have a few more questions:
>>>
>>> I'm getting the followingf messages I could use some clarification on:
>>> 1. 2015-05-28 21:28:19.228386 [NOTICE] switch_core_media.c:3315 Asked by
>>> candidate to set remote rtcp audio addr to 205.215.241.76:59980 but
>>> this is rtcp-mux so no thanks (is this affecting anything?)
>>>
>>>
>> Your using chrome canary I think.  This is a bug in canary.  Our latest
>> code addresses this chrome bug.
>>
>
> great.
>
>>
>>
>>> 2. 2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing video
>>> DTLS state from OFF to HANDSHAKE
>>> - it never gets beyond handshake - any ideas
>>>
>>
>> Look at packet captures, maybe firewall blocking things?
>>
> I'll check.
>
>> 3. 2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending
>>> video packet 0 bytes (ice not ready @ line 4241!)
>>> - ice not ready - what's causing this?
>>>
>>
>> This is because you don't have a dtls handshake yet.
>>
> makes sense
>
>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:404 Looking for
>>> zrtp-hash
>>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:359 Deciding
>>> whether to pass zrtp-hash between legs
>>> 2015-05-29 00:15:43.708406 [ALERT] switch_core_media.c:361
>>> CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash
>>>
>>> Should we be using ZRTP? if it weren't enabled at all would I see these
>>> messages?
>>>
>>
>> Is one leg of your call passing in zrtp stuff ?  You would not be running
>> zrtp on a webrtc call as no webrtc end points I know of support it.
>>
> I don't think so, but I see messages abut it. I've seen docs mention it
> has to be turned on in the build - I haven't done that. Does it default to
> off still?
>
>> Lastly, it looks like we're not getting any sip responses from FS - Any
>>> ideas why?
>>>
>>>
>>> Maybe firewall, maybe misconfigured ip or acl?  If you are using verto,
>> where does sip come in to play?
>>
>> using sip.js and not verto. will try the fs demo with verto.
>
>>
>>
>>
>>> All answers/help/hints are appreciated.
>>>
>>> chris
>>>
>>>
>>>
>>>
>>> On Thursday, May 28, 2015, Ken Rice <krice at freeswitch.org> wrote:
>>>
>>>>  Hi Chris,
>>>>
>>>> There is always https://conference.freeswitch.org/verto/ where we have
>>>> been demoing this for quite a while
>>>>
>>>> If you need this working like now I would suggest emailing
>>>> consulting at freeswitch.org where you can get paid realtime assistance
>>>> from one of the Core Dev Team
>>>>
>>>> See you @ ClueCon!
>>>> Ken
>>>>
>>>>
>>>>
>>>> *http://www.ClueCon.com <http://www.ClueCon.com>
>>>> http://www.freeswitch.org <http://www.freeswitch.org> http://www.OSTAG.org
>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>> Twitter: @FreeSWITCH
>>>>          @ClueCon
>>>>
>>>>
>>>> On 5/28/15, 5:20 PM, "Chris Mandra" <mandra at gmail.com> wrote:
>>>>
>>>> Hi guys. I had been able to use webrtc via chrome with FS 1.4 for
>>>> connections and confenerce calls without incident (although the bandwidth
>>>> and video quality seemed rather low as compared to the
>>>> apprtc.appspot.com <http://apprtc.appspot.com>  demo.)
>>>> I have followed the instructions as well I am able on
>>>> https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video
>>>> but so far I can't make an actual connection; the certs pass the cert
>>>> test without issue, but I have no joy using webrtc and fs_video
>>>>
>>>> a few questions:
>>>> 1. Is there any known problem using wildcard certs from godaddy?
>>>> 2. is there other configuration beyond what's on the web page needed to
>>>> make webrtc work? (I have the apropos ports open etc)
>>>> 3. Are there any known issues with fs_video and sip.js?
>>>> 4. I feel like it's getting hung op on dtls, as it never passes
>>>> "2015-05-28 21:28:19.228386 [INFO] switch_rtp.c:3322 Changing audio
>>>> DTLS state from OFF to HANDSHAKE"
>>>> (it repeats that twice with:
>>>> "2015-05-28 21:28:19.228386 [ALERT] switch_rtp.c:6882 Skip sending
>>>> video packet 0 bytes (ice not ready @ line 4241!)" not long after - does
>>>> that speak to something obvious to you guys?
>>>> 5. is there a working webrtc / fs_video demo I can look at?
>>>>
>>>> I realize this may all be solved by the release of 1.6, but I need this
>>>> working now. I've been looking forward to this bc I know there have been
>>>> some issues with webrtc, rtcp, freeswitch and solid BWE.
>>>>
>>>> Thanks for all you do,
>>>> chris
>>>>
>>>>
>>>>
>>>> ------------------------------
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> --
>>>> Ken
>>>>
>>>>
>>>>
>>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>> Twitter: @FreeSWITCH
>>>>
>>>>
>> _________________________________________________________________________
>> 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
>>
>
>
>
> --
> mandra
> c:410.258.5281
>
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From royj at yandex.ru  Fri May 29 19:05:40 2015
From: royj at yandex.ru (royj at yandex.ru)
Date: Fri, 29 May 2015 18:05:40 +0300
Subject: [Freeswitch-users] Displaying number of incoming calls
In-Reply-To: <CAEwTsAjXOrqk-DG5vLYJZ0zoL-Ss9gjRnH66440fUt=rOcdWbQ@mail.gmail.com>
References: <4946501432908601@web6h.yandex.ru>
	<CAEwTsAjXOrqk-DG5vLYJZ0zoL-Ss9gjRnH66440fUt=rOcdWbQ@mail.gmail.com>
Message-ID: <5072131432911940@web27m.yandex.ru>

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From victor.medina at cibersys.com  Fri May 29 19:35:44 2015
From: victor.medina at cibersys.com (Victor Medina)
Date: Fri, 29 May 2015 11:05:44 -0430
Subject: [Freeswitch-users] The 1.6 video branch has been merged into
	master!
In-Reply-To: <CA+M05dCkyO-qR3j+OvgU9k9Zu7xxD77nRrqVFvHa1SkGRr5Kuw@mail.gmail.com>
References: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>
	<CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>
	<CADK-BjuyMD74Epn=P5zviUGOKkKC+4ddNP_HXkRCOrpmFvy0nQ@mail.gmail.com>
	<01d301d099c5$0d330980$27991c80$@symonics.com>
	<CA+M05dCkyO-qR3j+OvgU9k9Zu7xxD77nRrqVFvHa1SkGRr5Kuw@mail.gmail.com>
Message-ID: <CACXsapNjK9hg=TkUocUE0pY4kwfBBoKsioi+97XpEWdAdMuCZQ@mail.gmail.com>

YES!

2015-05-29 3:27 GMT-04:30 Oleg Blinnikov <osblinnikov at gmail.com>:

> Impressive improvements, congrats! I gonna try it right away!
>
> On Fri, May 29, 2015 at 6:07 AM, Christian Hoene <
> christian.hoene at symonics.com> wrote:
>
>> Congratulations!
>>
>> Good work!
>>
>>
>>
>> Christian Hoene
>>
>>
>>
>>
>>
>> *Von:* Blake Priddy [mailto:bpriddy at bryantschools.org]
>> *Gesendet:* Donnerstag, 28. Mai 2015 22:49
>> *An:* FreeSWITCH Users Help
>> *Betreff:* Re: [Freeswitch-users] The 1.6 video branch has been merged
>> into master!
>>
>>
>>
>> Great work guys!
>>
>>
>>
>> On Thu, May 28, 2015 at 3:45 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>> Yay!!!
>>
>> All hail our FreeSWITCH developers and core team !
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>>
>> On May 28, 2015 9:56 PM, "Ken Rice" <krice at freeswitch.org> wrote:
>>
>> New Post on freeswitch.org from Kathleen
>> check it out at http://ift.tt/1RrHZnX
>> The 1.6 video branch has been merged into master!
>>
>> Great news for our fellow FreeSWITCH users: in preparation for the 1.6
>> beta release, a lot of new functionality is being merged into the master
>> branch. Some of these new features have new build requirements and
>> dependencies, so please be sure to check the Confluence link here:
>> http://ift.tt/1d2d4iX for platform specific instructions. Building and
>> running FreeSWITCH will be easier if you are using Debian 8(Jessie). If you
>> are not using Debian you can find some of the supporting dependencies
>> tar.gz files here: http://ift.tt/1qAFKnH
>>
>> Additionally, if you are using 1.4 in production, you need to be sure to
>> switch to tracking the v1.4 branch, as master will be preparing for the 1.6
>> beta release.
>>
>> Some of the new features and work that have gone into this release
>> include:
>> FS-7499 <http://ift.tt/1d2d2HU> core RTCP improvements
>> FS-7500 <http://ift.tt/1PQJHBY> core video transcoding support
>> FS-7501 <http://ift.tt/1d2d2HW> core video jitterbuffer
>> FS-7502 <http://ift.tt/1PQJFdn> core video media bugs
>> FS-7503 <http://ift.tt/1d2d4iZ> core file interface video support
>> FS-7504 <http://ift.tt/1PQJHC2> codecs let you choose which codec module
>> to use
>> FS-7505 <http://ift.tt/1d2d4j1> file interface to let you specify which
>> format module to use when multiple types are supported
>> FS-7506 <http://ift.tt/1PQJHC4> core text rendering
>> FS-7507 <http://ift.tt/1d2d4j3> added new global directory variables and
>> configure directory behavior changes
>> FS-7508 <http://ift.tt/1d2d2HY> mod_vpx transcoding vp8/vp9 and replace
>> mod_v8
>> FS-7509 <http://ift.tt/1PQJHC6> mod_verto improvements allow for desktop
>> share with the installation of this chrome extension
>> http://ift.tt/1kQP2pS and improved bandwith and resolution handling.
>> FS-7512 <http://ift.tt/1PQJHCa> mod_png allows for image overlays for
>> logos and images for video mute
>> FS-7513 <http://ift.tt/1d2d4j5> mod_conference MCU feature and avatar
>> support
>> FS-7514 <http://ift.tt/1PQJI8Z> mod_vlc video support allows you to live
>> stream, record calls to a video file, and playback videos into a call.
>> FS-7515 <http://ift.tt/1d2d2Yc> mod_cv is a video media bug module that
>> uses video recognition and facial recognition technology to allow you to
>> modify a video stream by adding overlapping images and text or to silently
>> detect and fire events
>> FS-7516 <http://ift.tt/1PQJHSo> mod_imagick allows for PDF and GIF
>> rendered as video
>> FS-7517 <http://ift.tt/1d2d2Ye> mod_openh264 h264 codec module
>> FS-7519 <http://ift.tt/1d2d4zl> mod_av a file format and codec module
>> that uses libav or ffmpeg
>> FS-7494 <http://ift.tt/1PQJHSs> default avatar and mute images for video
>> MCU
>> FS-7471 <http://ift.tt/1d2d4zn> improved configs for video
>> FS-7338 <http://ift.tt/1PQJI97> removed external library dependencies
>> FS-7585 <http://ift.tt/1d2d4zr> added video support to mod_rtmp
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>>
>> --
>>
>> [image: Das Bild wurde vom Absender entfernt.] [image: Das Bild wurde
>> vom Absender entfernt.]
>>
>>
>>
>> *Blakelund Priddy*
>>
>> Network & Systems Engineer
>> Bryant Public School District
>> Bryant, Arkansas 72022
>> http://www.bryantschools.org
>>
>> p 501-653-5038
>> f 501-847-5656
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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,
> Oleg Blinnikov
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 



V?ctor E. Medina M.
Platform Architect / Chief Infrastructure
+58424 291 4561
BB #79A8AFA2
@VMCibersys
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From tru083 at yahoo.com  Fri May 29 20:16:33 2015
From: tru083 at yahoo.com (D D)
Date: Fri, 29 May 2015 16:16:33 +0000 (UTC)
Subject: [Freeswitch-users] Guidelines for adding a custom SIP INFO
 package?
Message-ID: <1539723496.1414577.1432916193161.JavaMail.yahoo@mail.yahoo.com>

By Package I mean a set of INFO capabilities, as described by RFC 6086(RFC 6086 - Session Initiation Protocol (SIP) INFO Method and Package Framework).

| ? |
| ? | ? | ? | ? | ? |
| RFC 6086 - Session Initiation Protocol (SIP) INFO Method and Package Framework[Docs] [txt|pdf] [draft-ietf-sipcor...] [Diff1] [Diff2] [Errata] PROPOSED STANDARD Errata Exist  |
|  |
| View on tools.ietf.org | Preview by Yahoo |
|  |
| ? |

Part of the RFC specifies the setting of header fields like Recv-Info and Supported.I was wondering if I can configure these values somewhere, or if I will need to set themin the esl program.? Also, I know I can set these headers in an originate request, but what if 
I receive an Invite with Recv-Info and I want the 200 OK to respond with the proposed Recv-Info?

Thanks!!David


> What do you mean by "package"?? You can use esl and events to send custom info.

> On May 28, 2015, at 11:26 AM, D D <tru083 at yahoo.com> wrote:
> 
> Hi,
> 
> I would like to add a custom SIP INFO package, controlled by an event socket application.
> 
> I know I can use sip_h variables to set header fields, but is there a better way to identify to sofia
> that I am adding a custom SIP INFO package?
> 
> Also, do you have any tips on processing INFO events and sending INFO messages?
> 
> Thanks!
> David

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From tru083 at yahoo.com  Fri May 29 21:06:17 2015
From: tru083 at yahoo.com (D D)
Date: Fri, 29 May 2015 17:06:17 +0000 (UTC)
Subject: [Freeswitch-users] Guidelines for adding a custom SIP
	INFO	package?
In-Reply-To: <CACOYK=jpqbRUTMXcxX_0MjPY4-yHfPLtV6v1CaALXNsJwkspxw@mail.gmail.com>
References: <1331892441.522840.1432826761612.JavaMail.yahoo@mail.yahoo.com>
	<0ECDBFD2-CE60-453F-954D-477B191E99A6@jerris.com>
	<CACOYK=jpqbRUTMXcxX_0MjPY4-yHfPLtV6v1CaALXNsJwkspxw@mail.gmail.com>
Message-ID: <123477423.1519014.1432919177415.JavaMail.yahoo@mail.yahoo.com>

By Package I mean a set of INFO capabilities, as described by RFC 6086 - Session Initiation Protocol (SIP) INFO Method and Package Framework.
Part of the RFC specifies the setting of header fields like Recv-Info and Supported.I was wondering if I can configure these values somewhere, or if I will need to set themin the esl program.? 

Also, I know I can set these headers in an originate request, but what if I receive an Invite with Recv-Info 
and I want the 200 OK to respond with the proposed Recv-Info?
Thanks!!David 



     On Thursday, May 28, 2015 8:03 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
   

 "Package" is a typical German speaker's mistake for "packet" ;-)On May 28, 2015 6:16 PM, "Michael Jerris" <mike at jerris.com> wrote:

What do you mean by "package"?? You can use esl and events to send custom info.

On May 28, 2015, at 11:26 AM, D D <tru083 at yahoo.com> wrote:
Hi,
I would like to add a custom SIP INFO package, controlled by an event socket application.
I know I can use sip_h variables to set header fields, but is there a better way to identify to sofiathat I am adding a custom SIP INFO package?

Also, do you have any tips on processing INFO events and sending INFO messages?
Thanks!David


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

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

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


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

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

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

  
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From james at onsip.com  Fri May 29 22:08:55 2015
From: james at onsip.com (James Criscuolo)
Date: Fri, 29 May 2015 14:08:55 -0400
Subject: [Freeswitch-users] Building FreeSWITCH master (> v1.7) on CentOS?
Message-ID: <CADz2+5URMbofJYKdHfvgcrOouz9wW4UYiS_0GXh4WvuPTVHkJQ@mail.gmail.com>

Hi,
I've run into a bit of an issue trying to get the latest HEAD compiled
after yesterday's merge and update to 1.7. The issue (at least one of them)
seems to lie with libyuv. The precompiled package at files.freeswitch.org only
has files for debian, and building libyuv from source also does not seem
feasible (it looks like Chromium (part of the build process) can only be
compiled on Ubuntu for some reason). I am running CentOS 6.2.

Has anybody figured out a way around this? It looks like libyuv is a
necessary piece of the build, but if there is a way to disable it that
would also work (although not preferable).

Thanks,
James
-- 
James Criscuolo
Software Engineer
OnSIP
www.onsip.com
p. 212.933.9190 x7047
SIP/Email: james at onsip.com
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From gabriel.peredo at clickfono.com  Fri May 29 18:30:10 2015
From: gabriel.peredo at clickfono.com (Gabriel Peredo)
Date: Fri, 29 May 2015 11:30:10 -0300
Subject: [Freeswitch-users] On call-retry intervals
Message-ID: <1FF6B954-6659-4071-B9BA-E259B6D40DD4@clickfono.com>

Hello all

I have a question regarding call retries. I see that there are two sets of variables relating to this:

Those relating to ?camping?:

campon_retries
campon_sleep

And two other ones that don?t appear under ?variables? in the wiki:

retry-seconds
expire-seconds (I include this one because I believe that expire-seconds/retry-seconds would be the number of retries?)

My question is: if I want to have the first leg on a bridge to be retried if it rejects the call, some minutes apart, which ones should I use?

Thanks in advance.

From mike at jerris.com  Fri May 29 23:08:28 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 29 May 2015 15:08:28 -0400
Subject: [Freeswitch-users] Building FreeSWITCH master (> v1.7) on
	CentOS?
In-Reply-To: <CADz2+5URMbofJYKdHfvgcrOouz9wW4UYiS_0GXh4WvuPTVHkJQ@mail.gmail.com>
References: <CADz2+5URMbofJYKdHfvgcrOouz9wW4UYiS_0GXh4WvuPTVHkJQ@mail.gmail.com>
Message-ID: <87394CCD-39F5-4FB4-A658-3FDF2A410076@jerris.com>

As detailed in the 1.6 merge release announcement:

https://freeswitch.org/the-1-6-video-branch-has-been-merged-into-master/

We have all the dependencies sorted out for you for Debian Jessie.  We have called for community members to help with the porting and packaging of dependencies that are required for other operating systems and distributions.  The work has begun on CentOS packages:

https://freeswitch.org/jira/browse/FS-7553

But this work is not yet complete, and will likely take quite some time for some platforms, depending on volunteer effort.  If you would like to begin testing 1.6 prior to the beta releases, we recommend using Debian Jessie as we have already resolved a number of fairly complex dependencies.  If you must use another operating system you will have to work out which of our custom dependencies are required:

https://freeswitch.org/stash/projects/SD

And how to get them built properly for your platform.  We welcome all community contributions to assist in this porting effort.  Please open pull requests against these libraries and coordinate with me or Ken Rice so we can get building packages for other platforms working on our build servers.

Thanks
Mike

> On May 29, 2015, at 2:08 PM, James Criscuolo <james at onsip.com> wrote:
> 
> 
> Hi,
> I've run into a bit of an issue trying to get the latest HEAD compiled after yesterday's merge and update to 1.7. The issue (at least one of them) seems to lie with libyuv. The precompiled package at files.freeswitch.org <http://files.freeswitch.org/> only has files for debian, and building libyuv from source also does not seem feasible (it looks like Chromium (part of the build process) can only be compiled on Ubuntu for some reason). I am running CentOS 6.2.
> 
> Has anybody figured out a way around this? It looks like libyuv is a necessary piece of the build, but if there is a way to disable it that would also work (although not preferable).
> 
> Thanks,
> James
> -- 
> James Criscuolo
> Software Engineer
> OnSIP
> www.onsip.com <http://www.onsip.com/>
> p. 212.933.9190 x7047 <>
> SIP/Email: james at onsip.com <mailto:james at onsip.com>

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From mike at jerris.com  Fri May 29 23:11:58 2015
From: mike at jerris.com (Michael Jerris)
Date: Fri, 29 May 2015 15:11:58 -0400
Subject: [Freeswitch-users] Guidelines for adding a custom SIP INFO
	package?
In-Reply-To: <123477423.1519014.1432919177415.JavaMail.yahoo@mail.yahoo.com>
References: <1331892441.522840.1432826761612.JavaMail.yahoo@mail.yahoo.com>
	<0ECDBFD2-CE60-453F-954D-477B191E99A6@jerris.com>
	<CACOYK=jpqbRUTMXcxX_0MjPY4-yHfPLtV6v1CaALXNsJwkspxw@mail.gmail.com>
	<123477423.1519014.1432919177415.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <9BD542D9-240E-4F9B-B230-5B11A0E778A9@jerris.com>

I belive if you send an in-dialog info, the appropriate headers will be copied from the invite, you will have to test this out to be sure.  For further customization, you probably need to use ESL to send the info request.

> On May 29, 2015, at 1:06 PM, D D <tru083 at yahoo.com> wrote:
> 
> By Package I mean a set of INFO capabilities, as described by RFC 6086 - Session Initiation Protocol (SIP) INFO Method and Package Framework.
> 
> Part of the RFC specifies the setting of header fields like Recv-Info and Supported.
> I was wondering if I can configure these values somewhere, or if I will need to set themin the esl program.  
> 
> Also, I know I can set these headers in an originate request, but what if I receive an Invite with Recv-Info 
> and I want the 200 OK to respond with the proposed Recv-Info?
> 
> Thanks!!
> David
> 
> 
> 
> 
> On Thursday, May 28, 2015 8:03 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> 
> 
> "Package" is a typical German speaker's mistake for "packet" ;-)
> On May 28, 2015 6:16 PM, "Michael Jerris" <mike at jerris.com <applewebdata://8048A5D4-0696-4F93-980E-3B59BE070435>> wrote:
> What do you mean by "package"?  You can use esl and events to send custom info.
> 
>> On May 28, 2015, at 11:26 AM, D D <tru083 at yahoo.com <applewebdata://8048A5D4-0696-4F93-980E-3B59BE070435>> wrote:
>> 
>> Hi,
>> 
>> I would like to add a custom SIP INFO package, controlled by an event socket application.
>> 
>> I know I can use sip_h variables to set header fields, but is there a better way to identify to sofia
>> that I am adding a custom SIP INFO package?
>> 
>> Also, do you have any tips on processing INFO events and sending INFO messages?
>> 
>> Thanks!
>> David
> 

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From william.king at quentustech.com  Sat May 30 05:34:59 2015
From: william.king at quentustech.com (William King)
Date: Fri, 29 May 2015 18:34:59 -0700
Subject: [Freeswitch-users] CentOS instalation problem
In-Reply-To: <265414108.21345.1432904020795.JavaMail.zimbra@voismart.it>
References: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>	<CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>	<CADK-BjuyMD74Epn=P5zviUGOKkKC+4ddNP_HXkRCOrpmFvy0nQ@mail.gmail.com>	<01d301d099c5$0d330980$27991c80$@symonics.com>	<972336332.20150529143043@seznam.cz>
	<265414108.21345.1432904020795.JavaMail.zimbra@voismart.it>
Message-ID: <556913C3.8030902@quentustech.com>

The one of the main reasons is so that we can get FreeSWITCH into the
official distribution repos, but they all require that software use the
system libraries instead of each application hauling along their own
versions of each library. A second major reason is that for many of the
libraries that FreeSWITCH uses in the modules have updates that the
distributions are able to keep up to date more quickly than we can.

We're all working hard to make(keep) building and installing FreeSWITCH
easy. Because of the move to system libs installing on different
platforms(Operating Systems, Distributions, CPU architectures, etc) the
install process has moved from a one process that 'should' work for
everyone to a process per platform:
https://freeswitch.org/confluence/display/FREESWITCH/Installation

To track the progress of FreeSWITCH becoming ready to include into
Debian you can follow this Jira:
https://freeswitch.org/jira/browse/FS-353

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 05/29/2015 05:53 AM, Matteo wrote:
> Hi,
> 
> ----- Il 29-mag-15, alle 14:30, Denis Jakovlev yadenis at seznam.cz ha scritto:
> 
>> CentOS instalation problem Hi all,
>>
>> I'm trying to do a clean install of this manual.
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7
>>
>> and after "make -f Makefile.centos7" I have an error.
>>
>> checking whether to include odbc... no
>> checking for uuid >= 1.41.2... checking for uuid... configure: WARNING: libuuid
>> development package highly recommended!
>> setting PLATFORM_CORE_LIBS to "-ldl -lcrypt -lrt"
>> checking for inflateReset in -lz... yes
>> adding "-lz" to PLATFORM_CORE_LIBS
>> checking for libyuv >= 0.0.1280... configure: error: You need to install
>> libyuv-dev. Required library
>> make: *** [freeswitch.git/Makefile] Error 1
> 
> 
> Nothing,
> 
> simply there's something to be addressed when building in non-debian distros.
> 
> Right now the only answer is to build & compile libyuv (and others) by yourself,
> and then build FS.
> 
> Check https://freeswitch.org/jira/browse/FS-7553 for what is going on.
> 
> Pay attention that some libs may conflict with installed ones, like libvpx .
> 
> You may check also
> https://freeswitch.org/stash/projects/SD
> 
> to checkout the specs, create the needed rpms and try.
> 
> I would say as today building FS (with video) outside debian is not for the faint of heart :)
> 
> Hopefully the FS team will build everything for us, but needs some time I guess.
> 
> Personally I'm trying to figure out why not including the libs inside FS
> like in the past... at least the most problematic ones.
> 
> After all (like libvpx) libs have moved out of tree, but they conflict,
> so gets renamed and rebuilt as a specific packages...
> 
> By not leveraging on distro libs, is not simpler to keep them inside?
> 
> Just my 2c :)
> 
> Reagards,
> Matteo
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 May 30 17:24:08 2015
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 30 May 2015 08:24:08 -0500
Subject: [Freeswitch-users] CentOS instalation problem
In-Reply-To: <556913C3.8030902@quentustech.com>
References: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>
	<CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>
	<CADK-BjuyMD74Epn=P5zviUGOKkKC+4ddNP_HXkRCOrpmFvy0nQ@mail.gmail.com>
	<01d301d099c5$0d330980$27991c80$@symonics.com>
	<972336332.20150529143043@seznam.cz>
	<265414108.21345.1432904020795.JavaMail.zimbra@voismart.it>
	<556913C3.8030902@quentustech.com>
Message-ID: <CAKbxfG__S_fG-6M_xWA471zhGXc48gwS4ySe9S3+wMg0koobvg@mail.gmail.com>

Otherwise you can just check out the 1.4 branch which has none of these
problems as its older and has no video depends.

Everyone who wants centos to work and has any skills with platform
admin needs to band together and help patch,  document and improve the
process for these other distros.

We are working with volunteers to make prebuilt packages for major distros
for those users who do not need to build their own binaries but our primary
focus is debian jessie.



On Friday, May 29, 2015, William King <william.king at quentustech.com> wrote:

> The one of the main reasons is so that we can get FreeSWITCH into the
> official distribution repos, but they all require that software use the
> system libraries instead of each application hauling along their own
> versions of each library. A second major reason is that for many of the
> libraries that FreeSWITCH uses in the modules have updates that the
> distributions are able to keep up to date more quickly than we can.
>
> We're all working hard to make(keep) building and installing FreeSWITCH
> easy. Because of the move to system libs installing on different
> platforms(Operating Systems, Distributions, CPU architectures, etc) the
> install process has moved from a one process that 'should' work for
> everyone to a process per platform:
> https://freeswitch.org/confluence/display/FREESWITCH/Installation
>
> To track the progress of FreeSWITCH becoming ready to include into
> Debian you can follow this Jira:
> https://freeswitch.org/jira/browse/FS-353
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com <javascript:;>
>
> On 05/29/2015 05:53 AM, Matteo wrote:
> > Hi,
> >
> > ----- Il 29-mag-15, alle 14:30, Denis Jakovlev yadenis at seznam.cz
> <javascript:;> ha scritto:
> >
> >> CentOS instalation problem Hi all,
> >>
> >> I'm trying to do a clean install of this manual.
> >>
> >> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7
> >>
> >> and after "make -f Makefile.centos7" I have an error.
> >>
> >> checking whether to include odbc... no
> >> checking for uuid >= 1.41.2... checking for uuid... configure: WARNING:
> libuuid
> >> development package highly recommended!
> >> setting PLATFORM_CORE_LIBS to "-ldl -lcrypt -lrt"
> >> checking for inflateReset in -lz... yes
> >> adding "-lz" to PLATFORM_CORE_LIBS
> >> checking for libyuv >= 0.0.1280... configure: error: You need to install
> >> libyuv-dev. Required library
> >> make: *** [freeswitch.git/Makefile] Error 1
> >
> >
> > Nothing,
> >
> > simply there's something to be addressed when building in non-debian
> distros.
> >
> > Right now the only answer is to build & compile libyuv (and others) by
> yourself,
> > and then build FS.
> >
> > Check https://freeswitch.org/jira/browse/FS-7553 for what is going on.
> >
> > Pay attention that some libs may conflict with installed ones, like
> libvpx .
> >
> > You may check also
> > https://freeswitch.org/stash/projects/SD
> >
> > to checkout the specs, create the needed rpms and try.
> >
> > I would say as today building FS (with video) outside debian is not for
> the faint of heart :)
> >
> > Hopefully the FS team will build everything for us, but needs some time
> I guess.
> >
> > Personally I'm trying to figure out why not including the libs inside FS
> > like in the past... at least the most problematic ones.
> >
> > After all (like libvpx) libs have moved out of tree, but they conflict,
> > so gets renamed and rebuilt as a specific packages...
> >
> > By not leveraging on distro libs, is not simpler to keep them inside?
> >
> > Just my 2c :)
> >
> > Reagards,
> > Matteo
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <javascript:;>
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From luis.daniel.lucio at gmail.com  Sat May 30 20:52:45 2015
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sat, 30 May 2015 12:52:45 -0400
Subject: [Freeswitch-users] CentOS instalation problem
In-Reply-To: <CAKbxfG__S_fG-6M_xWA471zhGXc48gwS4ySe9S3+wMg0koobvg@mail.gmail.com>
References: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>
	<CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>
	<CADK-BjuyMD74Epn=P5zviUGOKkKC+4ddNP_HXkRCOrpmFvy0nQ@mail.gmail.com>
	<01d301d099c5$0d330980$27991c80$@symonics.com>
	<972336332.20150529143043@seznam.cz>
	<265414108.21345.1432904020795.JavaMail.zimbra@voismart.it>
	<556913C3.8030902@quentustech.com>
	<CAKbxfG__S_fG-6M_xWA471zhGXc48gwS4ySe9S3+wMg0koobvg@mail.gmail.com>
Message-ID: <CAFLo2Qy9Kvsn4pwpyxnqhJ_Y0U8EYt7KAAcVRf_HJuN-EcMOoA@mail.gmail.com>

Where is the tarball? at least a beta tarball

Luis Daniel Lucio Quiroz
CISSP, CISM, CISA
Linux, VoIP and much more fun
www.okay.com.mx

Need LCR? Check out LCR for FusionPBX with FreeSWITCH
Need Billing? Check out Billing for FusionPBX with FreeSWITCH

2015-05-30 9:24 GMT-04:00 Anthony Minessale <anthony.minessale at gmail.com>:

> Otherwise you can just check out the 1.4 branch which has none of these
> problems as its older and has no video depends.
>
> Everyone who wants centos to work and has any skills with platform
> admin needs to band together and help patch,  document and improve the
> process for these other distros.
>
> We are working with volunteers to make prebuilt packages for major distros
> for those users who do not need to build their own binaries but our primary
> focus is debian jessie.
>
>
>
> On Friday, May 29, 2015, William King <william.king at quentustech.com>
> wrote:
>
>> The one of the main reasons is so that we can get FreeSWITCH into the
>> official distribution repos, but they all require that software use the
>> system libraries instead of each application hauling along their own
>> versions of each library. A second major reason is that for many of the
>> libraries that FreeSWITCH uses in the modules have updates that the
>> distributions are able to keep up to date more quickly than we can.
>>
>> We're all working hard to make(keep) building and installing FreeSWITCH
>> easy. Because of the move to system libs installing on different
>> platforms(Operating Systems, Distributions, CPU architectures, etc) the
>> install process has moved from a one process that 'should' work for
>> everyone to a process per platform:
>> https://freeswitch.org/confluence/display/FREESWITCH/Installation
>>
>> To track the progress of FreeSWITCH becoming ready to include into
>> Debian you can follow this Jira:
>> https://freeswitch.org/jira/browse/FS-353
>>
>> William King
>> Senior Engineer
>> Quentus Technologies, INC
>> 1037 NE 65th St Suite 273
>> Seattle, WA 98115
>> Main:   (877) 211-9337
>> Office: (206) 388-4772
>> Cell:   (253) 686-5518
>> william.king at quentustech.com
>>
>> On 05/29/2015 05:53 AM, Matteo wrote:
>> > Hi,
>> >
>> > ----- Il 29-mag-15, alle 14:30, Denis Jakovlev yadenis at seznam.cz ha
>> scritto:
>> >
>> >> CentOS instalation problem Hi all,
>> >>
>> >> I'm trying to do a clean install of this manual.
>> >>
>> >> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7
>> >>
>> >> and after "make -f Makefile.centos7" I have an error.
>> >>
>> >> checking whether to include odbc... no
>> >> checking for uuid >= 1.41.2... checking for uuid... configure:
>> WARNING: libuuid
>> >> development package highly recommended!
>> >> setting PLATFORM_CORE_LIBS to "-ldl -lcrypt -lrt"
>> >> checking for inflateReset in -lz... yes
>> >> adding "-lz" to PLATFORM_CORE_LIBS
>> >> checking for libyuv >= 0.0.1280... configure: error: You need to
>> install
>> >> libyuv-dev. Required library
>> >> make: *** [freeswitch.git/Makefile] Error 1
>> >
>> >
>> > Nothing,
>> >
>> > simply there's something to be addressed when building in non-debian
>> distros.
>> >
>> > Right now the only answer is to build & compile libyuv (and others) by
>> yourself,
>> > and then build FS.
>> >
>> > Check https://freeswitch.org/jira/browse/FS-7553 for what is going on.
>> >
>> > Pay attention that some libs may conflict with installed ones, like
>> libvpx .
>> >
>> > You may check also
>> > https://freeswitch.org/stash/projects/SD
>> >
>> > to checkout the specs, create the needed rpms and try.
>> >
>> > I would say as today building FS (with video) outside debian is not for
>> the faint of heart :)
>> >
>> > Hopefully the FS team will build everything for us, but needs some time
>> I guess.
>> >
>> > Personally I'm trying to figure out why not including the libs inside FS
>> > like in the past... at least the most problematic ones.
>> >
>> > After all (like libvpx) libs have moved out of tree, but they conflict,
>> > so gets renamed and rebuilt as a specific packages...
>> >
>> > By not leveraging on distro libs, is not simpler to keep them inside?
>> >
>> > Just my 2c :)
>> >
>> > Reagards,
>> > Matteo
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> [image: ?] sip:888 at conference.freeswitch.org  [image: ?] +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Sat May 30 22:19:14 2015
From: mike at jerris.com (Michael Jerris)
Date: Sat, 30 May 2015 14:19:14 -0400
Subject: [Freeswitch-users] CentOS instalation problem
In-Reply-To: <CAFLo2Qy9Kvsn4pwpyxnqhJ_Y0U8EYt7KAAcVRf_HJuN-EcMOoA@mail.gmail.com>
References: <556772a9d8e73_1afe3f32866091@resque-worker-high.2.mail>
	<CALXCt0qWzVMBJh-TgVwzPibgomps8LHKSMj5-3CSfUNB-DtH0A@mail.gmail.com>
	<CADK-BjuyMD74Epn=P5zviUGOKkKC+4ddNP_HXkRCOrpmFvy0nQ@mail.gmail.com>
	<01d301d099c5$0d330980$27991c80$@symonics.com>
	<972336332.20150529143043@seznam.cz>
	<265414108.21345.1432904020795.JavaMail.zimbra@voismart.it>
	<556913C3.8030902@quentustech.com>
	<CAKbxfG__S_fG-6M_xWA471zhGXc48gwS4ySe9S3+wMg0koobvg@mail.gmail.com>
	<CAFLo2Qy9Kvsn4pwpyxnqhJ_Y0U8EYt7KAAcVRf_HJuN-EcMOoA@mail.gmail.com>
Message-ID: <CAEwTsAj1+F-ykN41mQoP68XGY1q0quvA95QmJbMetVQ9BgwPuQ@mail.gmail.com>

See the 1.6 merge announcement

On Saturday, May 30, 2015, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> Where is the tarball? at least a beta tarball
>
> Luis Daniel Lucio Quiroz
> CISSP, CISM, CISA
> Linux, VoIP and much more fun
> www.okay.com.mx
>
> Need LCR? Check out LCR for FusionPBX with FreeSWITCH
> Need Billing? Check out Billing for FusionPBX with FreeSWITCH
>
> 2015-05-30 9:24 GMT-04:00 Anthony Minessale <anthony.minessale at gmail.com
> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>>:
>
>> Otherwise you can just check out the 1.4 branch which has none of these
>> problems as its older and has no video depends.
>>
>> Everyone who wants centos to work and has any skills with platform
>> admin needs to band together and help patch,  document and improve the
>> process for these other distros.
>>
>> We are working with volunteers to make prebuilt packages for major
>> distros for those users who do not need to build their own binaries but our
>> primary focus is debian jessie.
>>
>>
>>
>> On Friday, May 29, 2015, William King <william.king at quentustech.com
>> <javascript:_e(%7B%7D,'cvml','william.king at quentustech.com');>> wrote:
>>
>>> The one of the main reasons is so that we can get FreeSWITCH into the
>>> official distribution repos, but they all require that software use the
>>> system libraries instead of each application hauling along their own
>>> versions of each library. A second major reason is that for many of the
>>> libraries that FreeSWITCH uses in the modules have updates that the
>>> distributions are able to keep up to date more quickly than we can.
>>>
>>> We're all working hard to make(keep) building and installing FreeSWITCH
>>> easy. Because of the move to system libs installing on different
>>> platforms(Operating Systems, Distributions, CPU architectures, etc) the
>>> install process has moved from a one process that 'should' work for
>>> everyone to a process per platform:
>>> https://freeswitch.org/confluence/display/FREESWITCH/Installation
>>>
>>> To track the progress of FreeSWITCH becoming ready to include into
>>> Debian you can follow this Jira:
>>> https://freeswitch.org/jira/browse/FS-353
>>>
>>> William King
>>> Senior Engineer
>>> Quentus Technologies, INC
>>> 1037 NE 65th St Suite 273
>>> Seattle, WA 98115
>>> Main:   (877) 211-9337
>>> Office: (206) 388-4772
>>> Cell:   (253) 686-5518
>>> william.king at quentustech.com
>>>
>>> On 05/29/2015 05:53 AM, Matteo wrote:
>>> > Hi,
>>> >
>>> > ----- Il 29-mag-15, alle 14:30, Denis Jakovlev yadenis at seznam.cz ha
>>> scritto:
>>> >
>>> >> CentOS instalation problem Hi all,
>>> >>
>>> >> I'm trying to do a clean install of this manual.
>>> >>
>>> >> https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7
>>> >>
>>> >> and after "make -f Makefile.centos7" I have an error.
>>> >>
>>> >> checking whether to include odbc... no
>>> >> checking for uuid >= 1.41.2... checking for uuid... configure:
>>> WARNING: libuuid
>>> >> development package highly recommended!
>>> >> setting PLATFORM_CORE_LIBS to "-ldl -lcrypt -lrt"
>>> >> checking for inflateReset in -lz... yes
>>> >> adding "-lz" to PLATFORM_CORE_LIBS
>>> >> checking for libyuv >= 0.0.1280... configure: error: You need to
>>> install
>>> >> libyuv-dev. Required library
>>> >> make: *** [freeswitch.git/Makefile] Error 1
>>> >
>>> >
>>> > Nothing,
>>> >
>>> > simply there's something to be addressed when building in non-debian
>>> distros.
>>> >
>>> > Right now the only answer is to build & compile libyuv (and others) by
>>> yourself,
>>> > and then build FS.
>>> >
>>> > Check https://freeswitch.org/jira/browse/FS-7553 for what is going on.
>>> >
>>> > Pay attention that some libs may conflict with installed ones, like
>>> libvpx .
>>> >
>>> > You may check also
>>> > https://freeswitch.org/stash/projects/SD
>>> >
>>> > to checkout the specs, create the needed rpms and try.
>>> >
>>> > I would say as today building FS (with video) outside debian is not
>>> for the faint of heart :)
>>> >
>>> > Hopefully the FS team will build everything for us, but needs some
>>> time I guess.
>>> >
>>> > Personally I'm trying to figure out why not including the libs inside
>>> FS
>>> > like in the past... at least the most problematic ones.
>>> >
>>> > After all (like libvpx) libs have moved out of tree, but they conflict,
>>> > so gets renamed and rebuilt as a specific packages...
>>> >
>>> > By not leveraging on distro libs, is not simpler to keep them inside?
>>> >
>>> > Just my 2c :)
>>> >
>>> > Reagards,
>>> > Matteo
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>> [image: ?] sip:888 at conference.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','sip:888 at conference.freeswitch.org');>  [image:
>> ?] +19193869900
>>
>> ClueCon Testimonials <https://www.youtube.com/watch?v=9XXgW34t40s>
>> ClueCon This Summer <https://www.youtube.com/watch?v=NLaDpGQuZDA>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From nikotja at gmail.com  Sat May 30 21:30:38 2015
From: nikotja at gmail.com (Nikotja .)
Date: Sat, 30 May 2015 20:30:38 +0300
Subject: [Freeswitch-users] Error in Acrobits with use zrtp (MiTM)
Message-ID: <CAPXqz1RhD0_YeniS9BNYUNr2PgKWH7cyerFDZOf0r5nK7-V0UA@mail.gmail.com>

Hi,
I use Acrobits (ios and android) and Freeswitch with ZRTP (MiTM).
During call acrobits show an error: "The PBX is offering an enrollment, but
you are already enrolled. Don`t Trust PBX".

In freeswitch log:
[DEBUG] switch_rtp.c:1370 [zrtp protoco]: INFO! User have been already
registered - skip enrollment ritual. ID=1008
[DEBUG] switch_rtp.c:1327 User already enrolled!

Please help
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From jim at k4gvo.com  Sun May 31 00:21:22 2015
From: jim at k4gvo.com (Jim Lynch)
Date: Sat, 30 May 2015 16:21:22 -0400
Subject: [Freeswitch-users] Need help with lua called from dial plan - How
	to delimit arguments?
Message-ID: <556A1BC2.2030101@k4gvo.com>

I have a line in my dialplan that looks like this:
     <action application="lua" data="cid.lua  ${caller_id_name} 
${caller_id_number}" />

However when the program is executed and the caller_id_name has a space 
in it, the number gets lost.
I can't figure out a way to add delimiters that get passed to the 
shell.  None of these work:

     <action application="lua" data="cid.lua \"${caller_id_name}\" 
\"${caller_id_number}\"" />
or
     <action application="lua" data="cid.lua ''${caller_id_name}'' 
''${caller_id_number}''" />
or
     <action application="lua" data="cid.lua \"\"${caller_id_name}\"\" 
\"\"${caller_id_number}\"\"" />

None of those work.  The quotes are all stripped off when running this test

lua file:
tmp_file = "/tmp/cidtest.txt";
local file = assert(io.open(tmp_file, "a"));
local m_name=argv[1];
local m_number=argv[2];
file:write("j cid is ",m_name," number is ",m_number,"\n");

Any suggestions?

Thanks,
Jim.


From gmaruzz at gmail.com  Sun May 31 01:21:02 2015
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 30 May 2015 23:21:02 +0200
Subject: [Freeswitch-users] Need help with lua called from dial plan -
 How to delimit arguments?
In-Reply-To: <556A1BC2.2030101@k4gvo.com>
References: <556A1BC2.2030101@k4gvo.com>
Message-ID: <CALXCt0ovdiTgakup+HMD0-KBpwAbG0GyF_XpL9FOceTApHwzXA@mail.gmail.com>

Hi Jim,

get the variables from the session:

https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:getVariable



On Sat, May 30, 2015 at 10:21 PM, Jim Lynch <jim at k4gvo.com> wrote:

> I have a line in my dialplan that looks like this:
>      <action application="lua" data="cid.lua  ${caller_id_name}
> ${caller_id_number}" />
>
> However when the program is executed and the caller_id_name has a space
> in it, the number gets lost.
> I can't figure out a way to add delimiters that get passed to the
> shell.  None of these work:
>
>      <action application="lua" data="cid.lua \"${caller_id_name}\"
> \"${caller_id_number}\"" />
> or
>      <action application="lua" data="cid.lua ''${caller_id_name}''
> ''${caller_id_number}''" />
> or
>      <action application="lua" data="cid.lua \"\"${caller_id_name}\"\"
> \"\"${caller_id_number}\"\"" />
>
> None of those work.  The quotes are all stripped off when running this test
>
> lua file:
> tmp_file = "/tmp/cidtest.txt";
> local file = assert(io.open(tmp_file, "a"));
> local m_name=argv[1];
> local m_number=argv[2];
> file:write("j cid is ",m_name," number is ",m_number,"\n");
>
> Any suggestions?
>
> Thanks,
> Jim.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 dragic.dusan at gmail.com  Sun May 31 02:41:06 2015
From: dragic.dusan at gmail.com (=?UTF-8?B?RHXFoWFuIERyYWdpxIc=?=)
Date: Sun, 31 May 2015 00:41:06 +0200
Subject: [Freeswitch-users] Need help with lua called from dial plan -
 How to delimit arguments?
In-Reply-To: <556A1BC2.2030101@k4gvo.com>
References: <556A1BC2.2030101@k4gvo.com>
Message-ID: <CAPHNFcrGLyjjNtMVRrmLFJZz5_aftyeaKe42qRrNTp+u9q6=rg@mail.gmail.com>

Use single quotes (but Giovanni's suggestion is the way to go).

On 30 May 2015 at 22:21, Jim Lynch <jim at k4gvo.com> wrote:
> I have a line in my dialplan that looks like this:
>      <action application="lua" data="cid.lua  ${caller_id_name}
> ${caller_id_number}" />
>
> However when the program is executed and the caller_id_name has a space
> in it, the number gets lost.
> I can't figure out a way to add delimiters that get passed to the
> shell.  None of these work:
>
>      <action application="lua" data="cid.lua \"${caller_id_name}\"
> \"${caller_id_number}\"" />
> or
>      <action application="lua" data="cid.lua ''${caller_id_name}''
> ''${caller_id_number}''" />
> or
>      <action application="lua" data="cid.lua \"\"${caller_id_name}\"\"
> \"\"${caller_id_number}\"\"" />
>
> None of those work.  The quotes are all stripped off when running this test
>
> lua file:
> tmp_file = "/tmp/cidtest.txt";
> local file = assert(io.open(tmp_file, "a"));
> local m_name=argv[1];
> local m_number=argv[2];
> file:write("j cid is ",m_name," number is ",m_number,"\n");
>
> Any suggestions?
>
> Thanks,
> Jim.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Du?an Dragi?


From doug.vought at gmail.com  Sun May 31 06:41:50 2015
From: doug.vought at gmail.com (Doug Vought)
Date: Sat, 30 May 2015 22:41:50 -0400
Subject: [Freeswitch-users] LuaSQL doesn't work in Freeswitch
Message-ID: <CA+yzRF8cELqh_nYOA9-eg9Jgk0LH9i5ADhQTXWOcmv6NDW=qCQ@mail.gmail.com>

<https://freeswitch.org/confluence/display/FREESWITCH/Lua+with+Database>

On this confluence page, it indicates that one can use LuaSQL to connect to
a database.

FreeSWITCH Version 1.4.18-1~64bit (-1 64bit)
Lua 5.2.1
Linux freeswitch 3.2.0-4-amd64 #1 SMP Debian 3.2.65-1+deb7u2 x86_64

Utilizing the LuaSQL-sqlite3 database connector, I will attempt to INSERT,
but the value never gets inserted.

The reason I think it is freeswitch is because utilizing the same code
(without the freeswitch hooks), the INSERT works just fine.

Here's what I'm doing:


api = freeswitch.API()
> require"luasql.sqlite3"
> env = luasql.sqlite3()
> conn = env:connect("db.db")
> session:answer()
> query = [[INSERT INTO TEST_TABLE VALUES("0000000000")]]
> session:consoleLog("info", query .. "\n")
> conn:execute(query)
> conn:commit()
> conn:close()
> env:close()
> session:hangup()
>

The Lua interpreter is not throwing me any errors and the dialplan ends, so
I assume it's working. I go into the database to see and nothing has been
inserted.

Also note: I did try to make sure I was taking locking into account by
giving env:connect() the appropriate millisecond value, but this did not
change anything.

If I do this same thing, but without Freeswitch involved and ran by the Lua
interpreter:

require"luasql.sqlite3"
> env = luasql.sqlite3()
> conn = env:connect("db.db")

query = [[INSERT INTO TEST_TABLE VALUES("0000000000")]]
>
conn:execute(query)
> conn:commit()
> conn:close()
> env:close()
>

 The values are inserted.

----

I am putting this here because I am assuming this is a failure on my part
and I don't want to muck up Jira if that's the case.

BUT, it would be nice to see the LuaSQL recommendation removed if doesn't
work as expected with Freeswitch.

Can anyone try this and see if they are having the same problem? Maybe we
can save a few people some time by getting that info off of the confluence
page if it's found not to work with multiple people :).

Thanks,
Doug
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From doug.vought at gmail.com  Sun May 31 06:49:51 2015
From: doug.vought at gmail.com (Doug Vought)
Date: Sat, 30 May 2015 22:49:51 -0400
Subject: [Freeswitch-users] LuaSQL doesn't work in Freeswitch
In-Reply-To: <CA+yzRF8cELqh_nYOA9-eg9Jgk0LH9i5ADhQTXWOcmv6NDW=qCQ@mail.gmail.com>
References: <CA+yzRF8cELqh_nYOA9-eg9Jgk0LH9i5ADhQTXWOcmv6NDW=qCQ@mail.gmail.com>
Message-ID: <CA+yzRF_-2SDXFRYMxb2ix1wXMA2-Nbteio-CvwtwJ1cpRwFa1A@mail.gmail.com>

Hi, sorry. It looks like I messed up the code. I pasted in some easier to
copy code:

api = freeswitch.API()
require"luasql.sqlite3"
env = luasql.sqlite3()
conn = env:connect("db.db")
session:answer()
query = [[INSERT INTO TEST_TABLE VALUES("0000000000")]]
session:consoleLog("info", query .. "\n")
conn:execute(query)
conn:commit()
conn:close()
env:close()
session:hangup()

----

require"luasql.sqlite3"
env = luasql.sqlite3()
conn = env:connect("db.db")
query = [[INSERT INTO TEST_TABLE VALUES("0000000000")]]
conn:execute(query)
conn:commit()
conn:close()
env:close()


On Sat, May 30, 2015 at 10:41 PM, Doug Vought <doug.vought at gmail.com> wrote:

> <https://freeswitch.org/confluence/display/FREESWITCH/Lua+with+Database>
>
> On this confluence page, it indicates that one can use LuaSQL to connect
> to a database.
>
> FreeSWITCH Version 1.4.18-1~64bit (-1 64bit)
> Lua 5.2.1
> Linux freeswitch 3.2.0-4-amd64 #1 SMP Debian 3.2.65-1+deb7u2 x86_64
>
> Utilizing the LuaSQL-sqlite3 database connector, I will attempt to INSERT,
> but the value never gets inserted.
>
> The reason I think it is freeswitch is because utilizing the same code
> (without the freeswitch hooks), the INSERT works just fine.
>
> Here's what I'm doing:
>
>
> api = freeswitch.API()
>> require"luasql.sqlite3"
>> env = luasql.sqlite3()
>> conn = env:connect("db.db")
>> session:answer()
>> query = [[INSERT INTO TEST_TABLE VALUES("0000000000")]]
>> session:consoleLog("info", query .. "\n")
>> conn:execute(query)
>> conn:commit()
>> conn:close()
>> env:close()
>> session:hangup()
>>
>
> The Lua interpreter is not throwing me any errors and the dialplan ends,
> so I assume it's working. I go into the database to see and nothing has
> been inserted.
>
> Also note: I did try to make sure I was taking locking into account by
> giving env:connect() the appropriate millisecond value, but this did not
> change anything.
>
> If I do this same thing, but without Freeswitch involved and ran by the
> Lua interpreter:
>
> require"luasql.sqlite3"
>> env = luasql.sqlite3()
>> conn = env:connect("db.db")
>
> query = [[INSERT INTO TEST_TABLE VALUES("0000000000")]]
>>
> conn:execute(query)
>> conn:commit()
>> conn:close()
>> env:close()
>>
>
>  The values are inserted.
>
> ----
>
> I am putting this here because I am assuming this is a failure on my part
> and I don't want to muck up Jira if that's the case.
>
> BUT, it would be nice to see the LuaSQL recommendation removed if doesn't
> work as expected with Freeswitch.
>
> Can anyone try this and see if they are having the same problem? Maybe we
> can save a few people some time by getting that info off of the confluence
> page if it's found not to work with multiple people :).
>
> Thanks,
> Doug
>
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From lists at kavun.ch  Sun May 31 08:09:26 2015
From: lists at kavun.ch (Emrah)
Date: Sat, 30 May 2015 23:09:26 -0500
Subject: [Freeswitch-users] FS6947 - Tuning Opus bandwidth
Message-ID: <C518929C-3EB0-45E9-BAE7-A18D8693B44C@kavun.ch>

Hi list,

I re-read FS6947 and don't understand how this problematic was addressed and the issue fixed.
The scope is simple. There should be a setting in the dialplan that allows downsampling of Opus for applications that do not require the 48khz / 2 channels framework. I.e.: terminating to the PSTN with Opus to take advantage of low bandwidth and great PLC.
There seems to be a lot of confusion around bandwidth in general there. It doesn't matter if the internal clock of the device is always sampling at 48khz / 2ch. There are settings that can facilitate a lower bandwidth consumption for particular use cases, and it seems the reason it is not being implemented in FS is just a matter of being confused about the intent of the 48khz 2ch base.
Please revisit this issue. FS should allow tuning of Opus audio / network bandwidth in the dialplan. It would optimize greatly lots of use cases. 
If I'm calling the PSTN, I'd rather have my client downsample and stream at a lower bandwidth, even if my audio capture would still be at 48khz / 2ch as per the RFC, and save on bandwidth, than transcode the full 48khz spectrum into PCM on my FS and minimize processing power on the client's side.

Jira here: https://freeswitch.org/jira/browse/FS-6947

Emrah

From jonlederman at gmail.com  Sun May 31 16:56:00 2015
From: jonlederman at gmail.com (Jon Lederman)
Date: Sun, 31 May 2015 08:56:00 -0400
Subject: [Freeswitch-users] DTLS ICE and WebRTC using Freeswitch 1.6
Message-ID: <19ACF7D4-9617-4FC9-81CF-D704BDBA4768@gmail.com>

Hi,

We are having trouble getting WebRTC working with 1.6.  I have tracked the problem down to the do_dtls function in switch_rtp.c (shown partially below).  Two things we have noticed:

1) It appears that ice.ready and ice.ready are always 0.  
2) DTLS negotiation never completes the handshake.

Does ice have to be operative in order for dtls negotiation to proceed or the other way around?  What should I look for, which may prevent the ice setup and/or dtls handshake.  
We are running on google cloud.  Are there networking/NAT issues running on cloud servers that might interfere with either ICE or DTLS?

Thanks in advance.

-Jon


static int do_dtls(switch_rtp_t *rtp_session, switch_dtls_t *dtls)
{
	int r = 0, ret = 0, len;
	switch_size_t bytes;
	unsigned char buf[4096] = "";
	int ready = rtp_session->ice.ice_user ? (rtp_session->ice.rready && rtp_session->ice.ready) : 1;
	


	if (!dtls->bytes && !ready) {
		//switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_ALERT, "Ice not ready");
		return 0;
	}

?





From mitchelle.bit at gmail.com  Sun May 31 22:43:26 2015
From: mitchelle.bit at gmail.com (Mitchelle Johnson)
Date: Mon, 1 Jun 2015 00:13:26 +0530
Subject: [Freeswitch-users] Configure IPKALL DID number to FreeSWITCH
Message-ID: <CANBm6SzxBtmvvW1_Se5EjmcNmD_8BcTmGSo3uYGTm8iMhQeXyQ@mail.gmail.com>

HI,

I am trying to configure a DID number assigned by IPKALL which is a
washington state phone number. The details are given below.

DID number=2532432058

When I try to call i get line as busy on my phone.

The dialplan is pasted below.

<include>

  <extension name="IPKALL">
    <condition field="destination_number" expression="^1001$">
      <action application="set" data="domain_name=59.99.250.70"/>
      <action application="transfer" data="1001 XML default"/>
    </condition>
  </extension>
  <extension name="IPKALL1">
    <condition field="destination_number" expression="^2532432058$">
      <action application="set" data="domain_name=59.99.250.70"/>
      <action application="transfer" data="1001 XML default"/>
    </condition>
  </extension>
  <extension name="IPKALL2">
    <condition field="destination_number" expression="^\+?1?(2532432058)$">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="transfer" data="1001 XML default"/>
    </condition>
  </extension>

<extension name="IPKALL3">
    <condition field="destination_number" expression="^\+?1?(2532432058)$">
      <action application="set" data="domain_name=59.99.250.70"/>
      <action application="transfer" data="1001 XML default"/>
    </condition>
  </extension>


</include>


I have also enclosed the log in the following pastebin link. The link is :
http://pastebin.com/U70HUMHw

Please help.

Thanks,
Mitchelle
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From jchavanton at gmail.com  Sun May 31 14:09:48 2015
From: jchavanton at gmail.com (Julien Chavanton)
Date: Sun, 31 May 2015 12:09:48 +0200
Subject: [Freeswitch-users] FS6947 - Tuning Opus bandwidth
In-Reply-To: <C518929C-3EB0-45E9-BAE7-A18D8693B44C@kavun.ch>
References: <C518929C-3EB0-45E9-BAE7-A18D8693B44C@kavun.ch>
Message-ID: <CAKmcL2=dn8T-=5zTAFC-1EFAbZTxw3uYLgncDVc6tqYvh8Nd=Q@mail.gmail.com>

Hi Emrah,

The settings exist but they are not available from the dialplan, right now
they can only be set globally .
https://freeswitch.org/confluence/display/FREESWITCH/mod_opus

You can control the bandwidth using maxplaybackrate and maxplaybackrate
this will control the local encoder and also adds the corresponding FMTP
parameters to the SDP to be used by the remote encoder (if it does
implement the following draft, the draft is evolving but I think it as not
changed)

https://tools.ietf.org/html/draft-ietf-payload-rtp-opus-11

Maybe something like :

maxaveragebitrate 24000
maxplaybackrate 8000

The discussion was getting slightly more complicated when we where
discussing about unnecessary resampling this was not a problem but it was
just adding extra load on the server.

On Sun, May 31, 2015 at 6:09 AM, Emrah <lists at kavun.ch> wrote:

> Hi list,
>
> I re-read FS6947 and don't understand how this problematic was addressed
> and the issue fixed.
> The scope is simple. There should be a setting in the dialplan that allows
> downsampling of Opus for applications that do not require the 48khz / 2
> channels framework. I.e.: terminating to the PSTN with Opus to take
> advantage of low bandwidth and great PLC.
> There seems to be a lot of confusion around bandwidth in general there. It
> doesn't matter if the internal clock of the device is always sampling at
> 48khz / 2ch. There are settings that can facilitate a lower bandwidth
> consumption for particular use cases, and it seems the reason it is not
> being implemented in FS is just a matter of being confused about the intent
> of the 48khz 2ch base.
> Please revisit this issue. FS should allow tuning of Opus audio / network
> bandwidth in the dialplan. It would optimize greatly lots of use cases.
> If I'm calling the PSTN, I'd rather have my client downsample and stream
> at a lower bandwidth, even if my audio capture would still be at 48khz /
> 2ch as per the RFC, and save on bandwidth, than transcode the full 48khz
> spectrum into PCM on my FS and minimize processing power on the client's
> side.
>
> Jira here: https://freeswitch.org/jira/browse/FS-6947
>
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sun May 31 23:04:32 2015
From: mike at jerris.com (Michael Jerris)
Date: Sun, 31 May 2015 15:04:32 -0400
Subject: [Freeswitch-users] DTLS ICE and WebRTC using Freeswitch 1.6
In-Reply-To: <19ACF7D4-9617-4FC9-81CF-D704BDBA4768@gmail.com>
References: <19ACF7D4-9617-4FC9-81CF-D704BDBA4768@gmail.com>
Message-ID: <CAEwTsAj67NcM7V1Oyoqh2OuMyZ5ex1T_X3TMsVb9hfVcD5VZfg@mail.gmail.com>

Typically this happens when the traffic isn't making it back to FreeSWITCH.

On Sunday, May 31, 2015, Jon Lederman <jonlederman at gmail.com> wrote:

> Hi,
>
> We are having trouble getting WebRTC working with 1.6.  I have tracked the
> problem down to the do_dtls function in switch_rtp.c (shown partially
> below).  Two things we have noticed:
>
> 1) It appears that ice.ready and ice.ready are always 0.
> 2) DTLS negotiation never completes the handshake.
>
> Does ice have to be operative in order for dtls negotiation to proceed or
> the other way around?  What should I look for, which may prevent the ice
> setup and/or dtls handshake.
> We are running on google cloud.  Are there networking/NAT issues running
> on cloud servers that might interfere with either ICE or DTLS?
>
> Thanks in advance.
>
> -Jon
>
>
> static int do_dtls(switch_rtp_t *rtp_session, switch_dtls_t *dtls)
> {
>         int r = 0, ret = 0, len;
>         switch_size_t bytes;
>         unsigned char buf[4096] = "";
>         int ready = rtp_session->ice.ice_user ? (rtp_session->ice.rready
> && rtp_session->ice.ready) : 1;
>
>
>
>         if (!dtls->bytes && !ready) {
>
> //switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session),
> SWITCH_LOG_ALERT, "Ice not ready");
>                 return 0;
>         }
>
> ?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Sun May 31 23:13:03 2015
From: mike at jerris.com (Michael Jerris)
Date: Sun, 31 May 2015 15:13:03 -0400
Subject: [Freeswitch-users] FS6947 - Tuning Opus bandwidth
In-Reply-To: <CAKmcL2=dn8T-=5zTAFC-1EFAbZTxw3uYLgncDVc6tqYvh8Nd=Q@mail.gmail.com>
References: <C518929C-3EB0-45E9-BAE7-A18D8693B44C@kavun.ch>
	<CAKmcL2=dn8T-=5zTAFC-1EFAbZTxw3uYLgncDVc6tqYvh8Nd=Q@mail.gmail.com>
Message-ID: <CAEwTsAjyZM9jdQv08Jv6AT2zo8HgNa5vPV4YVCv8a9sT7W3qyw@mail.gmail.com>

Side note, opening at the different rate I believe just makes the opus
library do the re sampling instead of FreeSWITCH.

On Sunday, May 31, 2015, Julien Chavanton <jchavanton at gmail.com> wrote:

> Hi Emrah,
>
> The settings exist but they are not available from the dialplan, right
> now they can only be set globally .
> https://freeswitch.org/confluence/display/FREESWITCH/mod_opus
>
> You can control the bandwidth using maxplaybackrate and maxplaybackrate
> this will control the local encoder and also adds the corresponding FMTP
> parameters to the SDP to be used by the remote encoder (if it does
> implement the following draft, the draft is evolving but I think it as not
> changed)
>
> https://tools.ietf.org/html/draft-ietf-payload-rtp-opus-11
>
> Maybe something like :
>
> maxaveragebitrate 24000
> maxplaybackrate 8000
>
> The discussion was getting slightly more complicated when we where
> discussing about unnecessary resampling this was not a problem but it was
> just adding extra load on the server.
>
> On Sun, May 31, 2015 at 6:09 AM, Emrah <lists at kavun.ch
> <javascript:_e(%7B%7D,'cvml','lists at kavun.ch');>> wrote:
>
>> Hi list,
>>
>> I re-read FS6947 and don't understand how this problematic was addressed
>> and the issue fixed.
>> The scope is simple. There should be a setting in the dialplan that
>> allows downsampling of Opus for applications that do not require the 48khz
>> / 2 channels framework. I.e.: terminating to the PSTN with Opus to take
>> advantage of low bandwidth and great PLC.
>> There seems to be a lot of confusion around bandwidth in general there.
>> It doesn't matter if the internal clock of the device is always sampling at
>> 48khz / 2ch. There are settings that can facilitate a lower bandwidth
>> consumption for particular use cases, and it seems the reason it is not
>> being implemented in FS is just a matter of being confused about the intent
>> of the 48khz 2ch base.
>> Please revisit this issue. FS should allow tuning of Opus audio / network
>> bandwidth in the dialplan. It would optimize greatly lots of use cases.
>> If I'm calling the PSTN, I'd rather have my client downsample and stream
>> at a lower bandwidth, even if my audio capture would still be at 48khz /
>> 2ch as per the RFC, and save on bandwidth, than transcode the full 48khz
>> spectrum into PCM on my FS and minimize processing power on the client's
>> side.
>>
>> Jira here: https://freeswitch.org/jira/browse/FS-6947
>>
>> Emrah
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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