From vma at vallimamod.org  Fri Jun  1 09:20:44 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Fri, 1 Jun 2018 11:20:44 +0200
Subject: [Freeswitch-users] How to get working leg timeout
In-Reply-To: <CAHtxdDc4nDG+urR2OXEkDKpz7r-oPAb9Oy8yqrTRs3CTyLywTw@mail.gmail.com>
References: <CAHtxdDc4nDG+urR2OXEkDKpz7r-oPAb9Oy8yqrTRs3CTyLywTw@mail.gmail.com>
Message-ID: <0FC59999-3D73-4527-9AE3-09C418A10947@vallimamod.org>

Hi,

Have you tried with ignore_early_media=true ?

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sip.solutions <mailto:vma at sip.solutions>
linkedin.com/in/vallimamod <http://linkedin.com/in/vallimamod>
.


> On 1 Jun 2018, at 00:25, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
> 
> Hello i want hangup call if caller not responded for 20 sec.
> When bleg send RINGING message then leg_timeout works as expected.
> But if bleg send SESSION_PROGGRESS message then leg_timeout help.
> 
> Is exist other way to get this working?




-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180601/917b52cb/attachment.html>

From udy786 at gmail.com  Fri Jun  1 05:12:23 2018
From: udy786 at gmail.com (Uday kumar)
Date: Fri, 1 Jun 2018 10:42:23 +0530
Subject: [Freeswitch-users] Bookmark in Play recording using
	bind_digit_action on call
In-Reply-To: <CAOfqg5YSBwWj06t__n0E+m5fPGsQCEyQQ+eg+SxpYnyQNZpjDg@mail.gmail.com>
References: <CAOfqg5YSBwWj06t__n0E+m5fPGsQCEyQQ+eg+SxpYnyQNZpjDg@mail.gmail.com>
Message-ID: <CAOfqg5afBNaVqZh+qhgWzmKZm75Mibo+LK0Z-nS9FvDV3DA2xw@mail.gmail.com>

Is anyone can help / guide or give me hint?


Thanks
Uday.

On Thu, May 31, 2018 at 7:07 PM, Uday kumar <udy786 at gmail.com> wrote:

> Hello All,
>
> When any call is coming server, I am just playing a .wav file. Recording
> file length is 120min.
>
> Using *bind_digit_action*, user able to pause, seek, speed volume up and
> down.
>
> But no have time to listen 120min recording at a time. So If user will
> press a key, example 9 then save that playing location with caller phone
> number. If same caller calling from same number then start playing file
> from same where bookmark key was pressed.
>
> Please advise.
>
>
> --
> Thanks & Regard
> Uday.
> Mobile:- +91-9377579349
>



-- 
Thanks & Regard
Uday.
Mobile:- +91-9377579349
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180601/e6734798/attachment.html>

From pascal.emond at interact-iv.com  Fri Jun  1 06:53:27 2018
From: pascal.emond at interact-iv.com (Pascal Emond)
Date: Fri, 1 Jun 2018 08:53:27 +0200
Subject: [Freeswitch-users] How to get working leg timeout
In-Reply-To: <CAHtxdDc4nDG+urR2OXEkDKpz7r-oPAb9Oy8yqrTRs3CTyLywTw@mail.gmail.com>
References: <CAHtxdDc4nDG+urR2OXEkDKpz7r-oPAb9Oy8yqrTRs3CTyLywTw@mail.gmail.com>
Message-ID: <CADA78eafyygTri0W13A_JmteP5OAk_Jrs3YmDfdrikFCPUn9Cw@mail.gmail.com>

Hello,

with :  leg_progress_timeout=20    and leg_timeout=30
it will hangup after 20s if no ringing is received and after 30s if there
is no response.




Pascal Emond

Ingénieur Expert Infrastructures

+33 6 61 28 76 43



2018-06-01 0:25 GMT+02:00 Sergey Safarov <s.safarov at gmail.com>:

> Hello i want hangup call if caller not responded for 20 sec.
> When bleg send RINGING message then leg_timeout works as expected.
> But if bleg send SESSION_PROGGRESS message then leg_timeout help.
>
> Is exist other way to get this working?
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180601/90e294f0/attachment.html>

From kkothari157 at gmail.com  Fri Jun  1 11:12:31 2018
From: kkothari157 at gmail.com (Ketan Kothari)
Date: Fri, 1 Jun 2018 16:42:31 +0530
Subject: [Freeswitch-users] Freeswitch records only 15 seconds of call.
In-Reply-To: <CAGqqW9_RZMTZb7ZONQMTWbsrE9o6i4LDvQW5aoKbK+zZws=MNg@mail.gmail.com>
References: <CAGqqW98UGNiA3WpptyCQsvt7i1B4gc_AbmfR-DuzqSoxGZNG5Q@mail.gmail.com>
 <1958CFFC-111B-4E0C-8EA3-6F026BAFACB9@jerris.com>
 <CAGqqW9_RZMTZb7ZONQMTWbsrE9o6i4LDvQW5aoKbK+zZws=MNg@mail.gmail.com>
Message-ID: <CAGqqW9-niF7G8pW=wd45NJvQ-HL_-JT6LXqgE_ArdXSLW72Juw@mail.gmail.com>

Any suggestion to resolver this issue ?

On Thu, May 31, 2018 at 4:11 PM, Ketan Kothari <kkothari157 at gmail.com>
wrote:

> Hello Michael,
>
> Here is my server logs where i'm getting issue :
> https://pastebin.freeswitch.org/view/9538bc8f
>
> freeswitch at freeswitch> status
> UP 0 years, 4 days, 10 hours, 45 minutes, 38 seconds, 273 milliseconds,
> 968 microseconds
> FreeSWITCH (Version 1.6.17  64bit) is ready
> 617 session(s) since startup
> 0 session(s) - peak 20, last 5min 0
> 0 session(s) per Sec out of max 30, peak 6, last 5min 0
> 1000 session(s) max
> min idle cpu 0.00/99.93
> Current Stack Size/Max 240K/8192K
>
>
>
> On Wed, May 30, 2018 at 7:24 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I would suggest reviewing the logs and seeing if there is any indication
>> of what is happening
>>
>> > On May 30, 2018, at 7:20 AM, Ketan Kothari <kkothari157 at gmail.com>
>> wrote:
>> >
>> > Hi All,
>> > I'm getting issue on call recordings.
>> >
>> > Freeswitch can records maximum 15 seconds call but actual call was
>> around 5 minutes.
>> >
>> > Any suggestion to overcome this issue ?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180601/921cc698/attachment.html>

From joel at textplus.com  Fri Jun  1 01:06:35 2018
From: joel at textplus.com (Joel Serrano)
Date: Thu, 31 May 2018 18:06:35 -0700
Subject: [Freeswitch-users] Achieving TLS + SRTP for inbound calls
In-Reply-To: <CAE23trs_Uc2X7PsGKcdUPeR+Y+b82zJ4=dpu-BhCwsF4AF_qFQ@mail.gmail.com>
References: <CAE23trvirexj7LitkD-vg7YnHRaYM6GntaSC-zPemkUEmJQ5Cg@mail.gmail.com>
 <A90FDA4D-76D3-4506-9B90-015A9E72C0DA@jerris.com>
 <CAE23trtRG02bVRtDW81nX9MXPA29popseRs4bPMeV6T6rFqgRw@mail.gmail.com>
 <CALXCt0pwRJgrhmVs4mC09_we5-=Hz0UEOMAEDeRGhjiQ3kFEQA@mail.gmail.com>
 <CAE23trte7ag8djhnhtKj1HKUUZ4nyi=tMN8QEmZ6569fRRT=KQ@mail.gmail.com>
 <CALXCt0rV+1P2vMByMcgh03sh0nNJfEQQD8ueCZ-pThqW2R=i=A@mail.gmail.com>
 <CAE23truz1DhjnjcwODfqgchKVhN5dCFFFkYdHDtePWnAzy-m_A@mail.gmail.com>
 <CAE23trsULE1r8RXECAQoud8XnrhTTmzCmtunoRu9Rxmad6E5Cg@mail.gmail.com>
 <CAMtXxQk8Gs20meYJmkDSdDqdJDFgNzmkQ=nL0SucTF7aFzZu8A@mail.gmail.com>
 <CAE23trtzmtVawhNSZFqHyTg5Vc-q3gDTGMOJGMjSsd=_FOOSzQ@mail.gmail.com>
 <CAMtXxQkznbJPApaqvkD=eiUcfyTZNDyEm_57dt+p0yoLaU4PuA@mail.gmail.com>
 <CAE23trtC2j9abPjOt+6cEOJqCDz7shZaQhrfH9H22sXfQGzeWA@mail.gmail.com>
 <CAMtXxQmFXpUi3XumQA7ROhFx9pW6FSXQ_AG6ADWHoebj-vZBwg@mail.gmail.com>
 <CAE23trs_Uc2X7PsGKcdUPeR+Y+b82zJ4=dpu-BhCwsF4AF_qFQ@mail.gmail.com>
Message-ID: <CAMtXxQkok0pYGgFpO-BhkE-BnKKokAgpsEjzhvh72Urk0q6+Zg@mail.gmail.com>

Hi David,

The order to create the .pem file would be:

private-key
certificate
intermediate1
intermediate...X


On Thu, May 31, 2018 at 15:40 David P <davidswalkabout at gmail.com> wrote:

> Hi Joel,
>
> I'm on mobile now, but when I looked at the certbot install page for
> debian-jessie manual install, it links to backports page, and that page
> warns it's not prod-ready, IIRC.
>
> My EC2 is already in the state described by your steps. The question is
> how to create wss.pem (and dtls-srtp.pem?) from the pem's installed by
> certbot.
>
> Cheers,
> David
>
> On Thu, 31 May 2018, 9:55 am Joel Serrano, <joel at textplus.com> wrote:
>
>> Hi David,
>>
>> I don't understand your issues with goal 2 using let's encrypt, that
>> option certainly works and is widely used. What problems are you facing?
>> If you don't have enough confidence on backports you can always download
>> the latest stable release of certbot:
>> https://certbot.eff.org/lets-encrypt/pip-other
>>
>> *Install*
>> *Since it doesn't seem like your operating system has a packaged version
>> of Certbot, you should use our certbot-auto script to get a copy:*
>>
>> *wget https://dl.eff.org/certbot-auto <https://dl.eff.org/certbot-auto>*
>> *chmod a+x certbot-auto*
>>
>>
>> But, Certbot themselves are recommending installation on debian jessie
>> using the backports repository:
>>
>> *Install*
>> *Since Certbot is packaged for your system, all you'll need to do is
>> apt-get the following packages.*
>>
>> *First you'll have to follow the instructions here to enable the Jessie
>> backports repo, if you have not already done so. Then run:*
>>
>> *$ sudo apt-get install certbot -t jessie-backports*
>>
>>
>>
>> What are your concerns regarding using certbot installed from
>> jessie-backports?
>>
>>
>>
>> Going back to the topic, if you created /etc/freeswitch before installing
>> the packages, the installer will not deploy the vanilla config.
>>
>> I assume you installed from packages (as it's the recommended easy way),
>> if so, uninstall them, delete /etc/freeswitch, then install again:
>>
>> 1- Add signing key and repo (only done once, you should have already done
>> this):
>>
>> wget -O -
>> https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub |
>> apt-key add -
>> echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie
>> main" > /etc/apt/sources.list.d/freeswitch.list
>>
>> 2- Remove current installation:
>>
>> apt-get purge freeswitch*
>>
>> 3- Make sure /etc/freeswitch doesn't exist:
>>
>> rm -rf /etc/freeswitch
>>
>> 4- Install freeswitch:
>>
>> apt-get update && apt-get install -y freeswitch-meta-all
>>
>>
>> Done!
>>
>> You should have /etc/freeswitch deployed, and you can start doing your
>> updates in /etc/freeswitch/sip_profiles etc...
>>
>>
>>
>>
>> On Wed, May 30, 2018 at 4:33 PM, David P <davidswalkabout at gmail.com>
>> wrote:
>>
>>> Hi Joel and Branden,
>>>
>>> I have three goals:
>>> 1) To have an FS install that secures all WebRTC and SIP traffic to it
>>> 2) An install that doesn't require WebRTC users to manually fetch the
>>> certificate
>>> 3) An install that uses only production-ready software
>>>
>>> For goal 1, Mike and Giovanni have said a Debian Jessie minimal is the
>>> best or only choice.
>>>
>>> For goal 2, I'm avoiding gentls_cert and its self-signed certs. As a
>>> first attempt, I'm trying to get a free CA cert from LetsEncrypt via
>>> certbot. Unfortunately, doing this on debian jessie requires that I use
>>> backports that are described as "as-is", so I'm sacrificing goal 3 for the
>>> time being.
>>>
>>> In order to inform FS where it can find the private key, cert, and
>>> chain, I was planning to introduce soft links to the files that certbot put
>>> under /etc/letsencrypt/live/my.domain.com/
>>>
>>> I'm ready to do that, except that sip_profiles/internal.xml isn't where
>>> it normally would be, because I followed
>>> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#highlighter_549778
>>> and created /etc/freeswitch/ without knowing why I should do that. So
>>>  /usr/local/freeswitch/  does not exist, unfortunately. Also, echo
>>> ${prefix} is blank. So, I did a find from slash for internal.xml and found
>>> four matches:
>>>
>>> /usr/share/freeswitch/conf/insideout/sip_profiles/internal.xml
>>> /usr/share/freeswitch/conf/sbc/sbc_profiles/internal.xml
>>> /usr/share/freeswitch/conf/vanilla/sip_profiles/internal.xml
>>> /usr/share/freeswitch/conf/vanilla/skinny_profiles/internal.xml
>>>
>>> Which of these should I edit?
>>>
>>> Also, is it necessary to concatenate my private key, cert, and chain
>>> into a "wss.pem" as suggested at
>>> https://freeswitch.org/confluence/display/FREESWITCH/WebRTC#highlighter_647427
>>>
>>> Cheers,
>>> David
>>>
>>> On Tue, May 29, 2018 at 12:34 PM, Joel Serrano <joel at textplus.com>
>>> wrote:
>>>
>>>> Hi David,
>>>>
>>>> So it all depends.. Those docs are just introductions to get a setup
>>>> "up and running". For example, in the docs you generate self-signed
>>>> certificates that (although perfectly valid) can give you issues with
>>>> browsers because their CA is not trusted, etc. Regarding expiration, it all
>>>> depends, as this is something you choose.
>>>>
>>>> Going down to your specific problems:
>>>>
>>>> 1- ..${prefix}.. is just a variable, that will be replaced with a
>>>> value, normally /usr/local/freeswitch, but can be anything (depending on
>>>> where you installed FS).
>>>> 2- When it comes to the "path" that you specify in the config for the
>>>> certificates, it can also be anything, the important part is that you make
>>>> sure that the user you run FS with has access to reading those files. If
>>>> you don't like using ${prefix} you can directly set /path/to/your/certs,
>>>> just remember double checking the permissions.
>>>> 3- When you renew your certificate, you will have to make FS aware of
>>>> that, I'd have to check but I'm pretty sure that after updating the files a
>>>> sofia profile rescan should be enough.
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180531/825c9cc7/attachment-0001.html>

From hggh96.hg at gmail.com  Fri Jun  1 07:08:46 2018
From: hggh96.hg at gmail.com (Hossein Gholizadeh)
Date: Fri, 1 Jun 2018 11:38:46 +0430
Subject: [Freeswitch-users] Gateway
In-Reply-To: <CAHKqMb4XbFcc65pZYO6_rAH8ktn0FGSa=umSbP_sfEuCwHDMkg@mail.gmail.com>
References: <CAA7fESKvUbxAn1qhPSBrd_EEO7Or4jhFVGakpyQQEVYLouu8YA@mail.gmail.com>
 <CAHKqMb7O_kzucY6VzD5N99jyqrBzajADC471=R_-uTMUW15Y6Q@mail.gmail.com>
 <CAA7fESLnQbngpQQUFEGxeDnc_jJOgwd78kGrp6oTFU7B3pcMew@mail.gmail.com>
 <CAHKqMb4XbFcc65pZYO6_rAH8ktn0FGSa=umSbP_sfEuCwHDMkg@mail.gmail.com>
Message-ID: <CAA7fESKkXBix6uSOdd=+WuQ60S2gJ7OeS79koperYaifqo_eMQ@mail.gmail.com>

Thank you so much

Best regards

On Fri, Jun 1, 2018, 03:35 Nandy Dagondon <nandy1925 at gmail.com> wrote:

> Yes, the two "fax_enable_t38*" will signal to your gateway to enable T.38.
> Your gateway, of course, must be configured to T.38 mode when requested.
>
> Correction to my statement. mod_spandsp works with both ATA gateways and
> TDM cards.
>
>
>
> On Thu, May 31, 2018 at 4:10 AM, Hossein Gholizadeh <hggh96.hg at gmail.com>
> wrote:
>
>> You mean like this i have to enable T.38 ? may you explain it more about
>> configs?
>>
>> <action application="answer" />
>> <action application="playback" data="silence_stream://2000"/>
>> <action application="set" data="fax_enable_t38=true"/>
>> <action application="set" data="fax_enable_t38_request=true"/>
>> <action application="rxfax" data=" $${base_dir}/faxes/${uuid}.tif "/>
>> <action application="hangup"/>
>>
>>
>> On Thu, May 31, 2018 at 5:20 AM Nandy Dagondon <nandy1925 at gmail.com>
>> wrote:
>>
>>> If you're using an FXO gateway, use its T.38 feature.  In your dialplan,
>>> you're using spandsp (e.g. rxfax). They work only with FTDM analog cards
>>> e.g. Sangoma, Digium, etc. - I guess.
>>>
>>>
>>>
>>>
>>> On Tue, May 29, 2018 at 10:10 AM, Hossein Gholizadeh <
>>> hggh96.hg at gmail.com> wrote:
>>>
>>>> hi ... i have my gateway on 192.168.30.32 and a phone which connected
>>>> to gateway 984432250161.
>>>> I configured them like this :
>>>> 1.created a xml file in \conf\sip_profiles\external with
>>>> name cheap_tel.xml and the content is this :
>>>> <include>
>>>>    <gateway name="provider">
>>>>      <param name="proxy" value="192.168.30.32"/>
>>>>      <param name="register" value="false"/>
>>>>      <param name="caller-id-in-from" value="true"/> <!--Most gateways
>>>> seem to want this-->
>>>>      <param name="username" value="not-used"/>
>>>>      <param name="password" value="not-used"/>
>>>>    </gateway>
>>>> </include>
>>>>
>>>> 2.and in acl.conf added this node :
>>>>       <node type="allow" cidr="192.168.30.32/32"/>
>>>>
>>>> is my gateway config right ? because when i want to send a fax file by
>>>> loopback with this command i have no error but fax is not saved in
>>>> directory :
>>>> Command ----> originate sofia/gateway/provider/984432250161
>>>> &txfax(C:/test.tiff)
>>>>
>>>> And also my dialplan configs for fax is this :
>>>>  <extension name="fax_detect">
>>>>       <condition field="destination_number" expression="^984432250161$">
>>>>         <action application="answer" />
>>>>         <action application="set" data="transfer_ringback=${us-ring}" />
>>>> <action application="spandsp_start_fax_detect" data="transfer '9978
>>>>                                 XML default' 10 ced"/>
>>>>         <action application="bridge" data="user/$1@${domain_name}" />
>>>>       </condition>
>>>>     </extension>
>>>>
>>>>     <extension name="fax_receive">
>>>>   <condition field="destination_number" expression="^9978$">
>>>> <action application="answer" />
>>>> <action application="playback" data="silence_stream://2000"/>
>>>> <action application="rxfax" data="$${base_dir}/faxes/${uuid}.tif"/>
>>>> <action application="hangup"/>
>>>>   </condition>
>>>>     </extension>
>>>>
>>>> Can any one help me?
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180601/1c469e31/attachment.html>

From hggh96.hg at gmail.com  Sat Jun  2 04:13:59 2018
From: hggh96.hg at gmail.com (Hossein Gholizadeh)
Date: Sat, 2 Jun 2018 08:43:59 +0430
Subject: [Freeswitch-users] fax T.38
Message-ID: <CAA7fESLKMeReSq7_JxX6nfR9KzFf5P=Ubmb2zWU-3Zggx5x26w@mail.gmail.com>

Hi all

Can any one tell me what is the differences between these two configs?

#1
-------------------------------------------------------
*<extension name="t38_gateway">*
*      <condition field="destination_number" expression="^9108$">*
*        <action application="set" data="fax_enable_image_resizing=true" />*
* <action application="set" data="fax_enable_t38=true"/>*
*        <action application="answer" />*
* <action application="bridge" data="{sip_execute_on_image='t38_gateway
self
nocng',absolute_codec_string=PCMU}sofia/external/9107@${local_ip_v4}:5080"/>*
*      </condition>*
*    </extension>*

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

#2
-------------------------------------------------------
* <extension name="fax_detect">*
*      <condition field="destination_number" expression="^984432250161$">*
* <action application="set" data="fax_enable_t38_request=true"/>*
* <action application="set" data="fax_enable_t38=true"/>*
*        <action application="answer" />*
*        <action application="set" data="transfer_ringback=${us-ring}" />*
* <action application="spandsp_start_fax_detect" data="transfer '9978 XML
default' 10 ced"/>*
*       <action application="bridge" data="user/$1000@${domain_name}" />*
*      </condition>*
*    </extension>*
-------------------------------------------------------

are they both detecting the fax and send it to other extension ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180602/e58b1287/attachment.html>

From j4v28bsjtp43hnf865 at gmail.com  Sat Jun  2 10:58:19 2018
From: j4v28bsjtp43hnf865 at gmail.com (Xenia Obolenskaya)
Date: Sat, 2 Jun 2018 10:58:19 +0000
Subject: [Freeswitch-users] No ACK for 200 OK
Message-ID: <CAF7mOn5R6maeewMCkCqWqWx0tuV127L1OemrrRtMktfmPQM4ew@mail.gmail.com>

Hi, All!

Freeswitch-1 sends 200 OK to Freeswitch-2 via IPv6.
Freeswitch-2 received these packets and it can be seen in sip trace.
However Freeswitch-2 does not send ACK.

recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:35.903592:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F
   Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f
   CSeq: 123631351 INVITE
   Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   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: 339
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2]
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2
   s=FreeSWITCH
   c=IN IP6 2a02:7b43:1eb0:eb21::2
   t=0 0
   m=audio 32068 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=crypto:1 AEAD_AES_256_GCM_8
inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck
   ------------------------------------------------------------------------
recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:37.903708:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F
   Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f
   CSeq: 123631351 INVITE
   Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   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: 339
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2]
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2
   s=FreeSWITCH
   c=IN IP6 2a02:7b43:1eb0:eb21::2
   t=0 0
   m=audio 32068 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=crypto:1 AEAD_AES_256_GCM_8
inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck
   ------------------------------------------------------------------------
recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:41.903776:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F
   Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f
   CSeq: 123631351 INVITE
   Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   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: 339
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2]
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2
   s=FreeSWITCH
   c=IN IP6 2a02:7b43:1eb0:eb21::2
   t=0 0
   m=audio 32068 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=crypto:1 AEAD_AES_256_GCM_8
inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck
   ------------------------------------------------------------------------
recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:45.904373:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F
   Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f
   CSeq: 123631351 INVITE
   Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   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: 339
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2]
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2
   s=FreeSWITCH
   c=IN IP6 2a02:7b43:1eb0:eb21::2
   t=0 0
   m=audio 32068 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=crypto:1 AEAD_AES_256_GCM_8
inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck
   ------------------------------------------------------------------------
recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:49.904423:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F
   Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f
   CSeq: 123631351 INVITE
   Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   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: 339
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2]
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2
   s=FreeSWITCH
   c=IN IP6 2a02:7b43:1eb0:eb21::2
   t=0 0
   m=audio 32068 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=crypto:1 AEAD_AES_256_GCM_8
inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck
   ------------------------------------------------------------------------
recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:53.905075:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F
   Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f
   CSeq: 123631351 INVITE
   Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   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: 339
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2]
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2
   s=FreeSWITCH
   c=IN IP6 2a02:7b43:1eb0:eb21::2
   t=0 0
   m=audio 32068 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=crypto:1 AEAD_AES_256_GCM_8
inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck
   ------------------------------------------------------------------------
2018-06-02 13:20:55.780247 [DEBUG] sofia_reg.c:2435 Changing expire time to
897 by request of proxy sip:ipvs.sipnet.ru:5061
recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:20:57.905810:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F
   Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f
   CSeq: 123631351 INVITE
   Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   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: 339
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2]
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2
   s=FreeSWITCH
   c=IN IP6 2a02:7b43:1eb0:eb21::2
   t=0 0
   m=audio 32068 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=crypto:1 AEAD_AES_256_GCM_8
inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck
   ------------------------------------------------------------------------
recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:21:01.905406:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F
   Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f
   CSeq: 123631351 INVITE
   Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   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: 339
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2]
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2
   s=FreeSWITCH
   c=IN IP6 2a02:7b43:1eb0:eb21::2
   t=0 0
   m=audio 32068 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=crypto:1 AEAD_AES_256_GCM_8
inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck
   ------------------------------------------------------------------------
recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 13:21:05.905437:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKZ2359mFaFv98p;rport=55317
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=y912pHDgNp3QF
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=pNyNt1y311K1F
   Call-ID: 6b3983f9-e0f1-1236-0d9d-525400af4b7f
   CSeq: 123631351 INVITE
   Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=udp>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   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: 339
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2]
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1527902766 1527902767 IN IP6 2a02:7b43:1eb0:eb21::2
   s=FreeSWITCH
   c=IN IP6 2a02:7b43:1eb0:eb21::2
   t=0 0
   m=audio 32068 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=crypto:1 AEAD_AES_256_GCM_8
inline:LshmymiWylaGtFqTrEo3mLmx5sVN0Th32k7FSPcjicE4sDva+ZM4tlP2gck

So, Freeswitch-1 hangup:
2018-06-02 13:21:06.372001 [DEBUG] sofia.c:7084 Channel
sofia/external-ipv6/1000@[2a02:f680:e0:1210::2123] entering state
[terminating][0]
2018-06-02 13:21:06.372001 [NOTICE] sofia.c:8273 Hangup
sofia/external-ipv6/1000@[2a02:f680:e0:1210::2123] [CS_EXECUTE]
[NORMAL_UNSPECIFIED]
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180602/da41d71a/attachment-0001.html>

From j4v28bsjtp43hnf865 at gmail.com  Sun Jun  3 07:35:21 2018
From: j4v28bsjtp43hnf865 at gmail.com (Xenia Obolenskaya)
Date: Sun, 3 Jun 2018 07:35:21 +0000
Subject: [Freeswitch-users] No ACK for 200 OK
Message-ID: <CAF7mOn72EQFzMbrvG5LfA9jzqPYjmQbsbEKOoduMgKoCSzj04A@mail.gmail.com>

Hi, All!

Thank All!

I apologize for my negligence and  inadvertence.

Because a bare-bones gateway was setup for outbound calls with no
username/pass and without registration I omitted an important parameter
<param name="register-transport" value="tls"/>

Now, with this parameter, the situation has returned to normal.

A lot of thanks to All!

Xenia Obolenskaya
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180603/0fc50e18/attachment.html>

From pierre at couderc.eu  Sat Jun  2 13:44:40 2018
From: pierre at couderc.eu (Pierre Couderc)
Date: Sat, 2 Jun 2018 15:44:40 +0200
Subject: [Freeswitch-users] What is wrong with my originate ?
Message-ID: <3bf792d5-5171-bc31-397d-57a838d334fb@couderc.eu>

  originate  sofia/internal/98%free.couderc.eu  0033484253887 XML 
int.example.net

See my traces below. The strange is that if I call from my extension 98 
the same number all is fine.

I understand it may be question of CODEC, as when calling from 98 I see 
PCMA, PCMU in the traces. But how to correct?

Thanks to read me.

Pierre Couderc



......

2018-06-02 08:47:43.279349 [DEBUG] sofia_glue.c:1295 
sofia/external/0033484253887 sending invite version: 1.6.20 -37-987c9b9 
64bit
Local SDP:
v=0
o=FreeSWITCH 1527903329 1527903330 IN IP4 192.168.163.67
s=FreeSWITCH
c=IN IP4 192.168.163.67
t=0 0
m=audio 25934 RTP/AVP 9 101 13
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv

2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:40 
sofia/external/0033484253887 Standard INIT
2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:48 
(sofia/external/0033484253887) State Change CS_INIT -> CS_ROUTING
2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:627 
(sofia/external/0033484253887) State INIT going to sleep
2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:584 
(sofia/external/0033484253887) Running State Change CS_ROUTING (Cur 2 
Tot 201)
2018-06-02 08:47:43.279349 [DEBUG] sofia.c:7084 Channel 
sofia/external/0033484253887 entering state [calling][0]
2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:643 
(sofia/external/0033484253887) State ROUTING
2018-06-02 08:47:43.279349 [DEBUG] mod_sofia.c:143 
sofia/external/0033484253887 SOFIA ROUTING
2018-06-02 08:47:43.279349 [DEBUG] switch_ivr_originate.c:67 
(sofia/external/0033484253887) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:643 
(sofia/external/0033484253887) State ROUTING going to sleep
2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:584 
(sofia/external/0033484253887) Running State Change CS_CONSUME_MEDIA 
(Cur 2 Tot 2
01)
2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:662 
(sofia/external/0033484253887) State CONSUME_MEDIA
2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:662 
(sofia/external/0033484253887) State CONSUME_MEDIA going to sleep
2018-06-02 08:47:43.339353 [DEBUG] switch_rtp.c:7308 Correct audio 
ip/port confirmed.
2018-06-02 08:47:43.339353 [DEBUG] sofia.c:7084 Channel 
sofia/external/0033484253887 entering state [calling][0]
2018-06-02 08:47:43.399358 [DEBUG] sofia.c:7084 Channel 
sofia/external/0033484253887 entering state [terminated][488]
2018-06-02 08:47:43.399358 [NOTICE] sofia.c:8273 Hangup 
sofia/external/0033484253887 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
2

....



From vishalmpai at gmail.com  Sun Jun  3 08:02:15 2018
From: vishalmpai at gmail.com (Vishal Pai)
Date: Sun, 3 Jun 2018 13:32:15 +0530
Subject: [Freeswitch-users] mod_verto
Message-ID: <CACOmXtBSxNbtKgH961AP0D=5jEpgtPe3gb+cjR-w4GU3oZKUFQ@mail.gmail.com>

Hi All


Can we record the video conferencing using mod_verto. If yes how it is
possible and and what would it’s file format.


Thanks
Vishal Pai
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180603/056b93bd/attachment.html>

From tculjaga at gmail.com  Mon Jun  4 12:15:35 2018
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 4 Jun 2018 14:15:35 +0200
Subject: [Freeswitch-users] How to get working leg timeout
In-Reply-To: <CAGJLwFiza5iU1Apx-aYQT8+1xy86VmLk+k66D6m+chE_OmKZOw@mail.gmail.com>
References: <CAHtxdDc4nDG+urR2OXEkDKpz7r-oPAb9Oy8yqrTRs3CTyLywTw@mail.gmail.com>
 <CAGJLwFiza5iU1Apx-aYQT8+1xy86VmLk+k66D6m+chE_OmKZOw@mail.gmail.com>
Message-ID: <CAJoKKdWRu_LK2owAU0TB1uvW=qTOC5+cwS+DV4dUUEHhRt=dNg@mail.gmail.com>

you can also abuse sched_hangup as well.

set it to +20 when you place the call and reset it on answer.

On 1 June 2018 at 01:46, Alexey Sibyakin <alex at freeswitch.com> wrote:

> Hello,
>
> Have you tried leg_progress_timeout ?
> https://freeswitch.org/confluence/display/FREESWITCH/
> Variables+Master+List#VariablesMasterList-leg_progress_timeout
>
> Regards,
>
> Alex
>
> On Fri, Jun 1, 2018 at 7:25 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Hello i want hangup call if caller not responded for 20 sec.
>> When bleg send RINGING message then leg_timeout works as expected.
>> But if bleg send SESSION_PROGGRESS message then leg_timeout help.
>>
>> Is exist other way to get this working?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
>
>
> --
> Alex Sibyakin | Support Engineer
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
> Email: alex at freeswitch.com
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
> Need commercial support? Contact sales at freeswitch.com for details.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180604/eeccbdce/attachment.html>

From s.safarov at gmail.com  Sat Jun  2 05:06:20 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 2 Jun 2018 08:06:20 +0300
Subject: [Freeswitch-users] Freeswitch records only 15 seconds of call.
In-Reply-To: <CAGqqW9-niF7G8pW=wd45NJvQ-HL_-JT6LXqgE_ArdXSLW72Juw@mail.gmail.com>
References: <CAGqqW98UGNiA3WpptyCQsvt7i1B4gc_AbmfR-DuzqSoxGZNG5Q@mail.gmail.com>
 <1958CFFC-111B-4E0C-8EA3-6F026BAFACB9@jerris.com>
 <CAGqqW9_RZMTZb7ZONQMTWbsrE9o6i4LDvQW5aoKbK+zZws=MNg@mail.gmail.com>
 <CAGqqW9-niF7G8pW=wd45NJvQ-HL_-JT6LXqgE_ArdXSLW72Juw@mail.gmail.com>
Message-ID: <CAHtxdDeN9cJFJqJWEiNVWq+w4RUfnDqizFyOi26PGEk9X91iRg@mail.gmail.com>

you can
1) test recording on vanilla config. If all works as expected, then;
2) add dialplan elements step-by-step and need to identify whats is
elements in dialplan breaks recordings.

Sergey

пт, 1 июн. 2018 г. в 20:51, Ketan Kothari <kkothari157 at gmail.com>:

> Any suggestion to resolver this issue ?
>
> On Thu, May 31, 2018 at 4:11 PM, Ketan Kothari <kkothari157 at gmail.com>
> wrote:
>
>> Hello Michael,
>>
>> Here is my server logs where i'm getting issue :
>> https://pastebin.freeswitch.org/view/9538bc8f
>>
>> freeswitch at freeswitch> status
>> UP 0 years, 4 days, 10 hours, 45 minutes, 38 seconds, 273 milliseconds,
>> 968 microseconds
>> FreeSWITCH (Version 1.6.17  64bit) is ready
>> 617 session(s) since startup
>> 0 session(s) - peak 20, last 5min 0
>> 0 session(s) per Sec out of max 30, peak 6, last 5min 0
>> 1000 session(s) max
>> min idle cpu 0.00/99.93
>> Current Stack Size/Max 240K/8192K
>>
>>
>>
>> On Wed, May 30, 2018 at 7:24 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> I would suggest reviewing the logs and seeing if there is any indication
>>> of what is happening
>>>
>>> > On May 30, 2018, at 7:20 AM, Ketan Kothari <kkothari157 at gmail.com>
>>> wrote:
>>> >
>>> > Hi All,
>>> > I'm getting issue on call recordings.
>>> >
>>> > Freeswitch can records maximum 15 seconds call but actual call was
>>> around 5 minutes.
>>> >
>>> > Any suggestion to overcome this issue ?
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180602/9a0c9647/attachment-0001.html>

From s.safarov at gmail.com  Sat Jun  2 05:10:21 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 2 Jun 2018 08:10:21 +0300
Subject: [Freeswitch-users] Freeswitch records only 15 seconds of call.
In-Reply-To: <CAHtxdDeN9cJFJqJWEiNVWq+w4RUfnDqizFyOi26PGEk9X91iRg@mail.gmail.com>
References: <CAGqqW98UGNiA3WpptyCQsvt7i1B4gc_AbmfR-DuzqSoxGZNG5Q@mail.gmail.com>
 <1958CFFC-111B-4E0C-8EA3-6F026BAFACB9@jerris.com>
 <CAGqqW9_RZMTZb7ZONQMTWbsrE9o6i4LDvQW5aoKbK+zZws=MNg@mail.gmail.com>
 <CAGqqW9-niF7G8pW=wd45NJvQ-HL_-JT6LXqgE_ArdXSLW72Juw@mail.gmail.com>
 <CAHtxdDeN9cJFJqJWEiNVWq+w4RUfnDqizFyOi26PGEk9X91iRg@mail.gmail.com>
Message-ID: <CAHtxdDeKKJ0zt_=Wd+4t4zciOxj0QqBagV6XKb2=CxgUyEVghw@mail.gmail.com>

One of leg is disconected from call

   1. 2018-05-31 11:44:58.173630 [NOTICE] sofia.c:1012 Hangup sofia/default/
   7900757132 at 91.12.24.13 [CS_EXECUTE] [NORMAL_CLEARING]
   2. 2018-05-31 11:44:58.173630 [DEBUG] switch_ivr_bridge.c:787 BRIDGE
   THREAD DONE [sofia/default/00555140094888]
   3. 2018-05-31 11:44:58.173630 [NOTICE] switch_ivr_bridge.c:891 Hangup
   sofia/default/00555140094888 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
   4. 2018-05-31 11:44:58.173630 [DEBUG] switch_core_state_machine.c:653 (
   sofia/default/00555140094888) State EXCHANGE_MEDIA going to sleep
   5. 2018-05-31 11:44:58.173630 [DEBUG] switch_core_state_machine.c:584 (
   sofia/default/00555140094888) Running State Change CS_HANGUP (Cur 2 Tot
   617)
   6. 2018-05-31 11:44:58.173630 [DEBUG] switch_ivr_async.c:1316 Stop
   recording file
   /home/shared/recordings/f08dd287-60c1-407c-a012-77cdb134e3d0.wav

need same log with enabled siptrace
sofia global siptrace on


сб, 2 июн. 2018 г. в 8:06, Sergey Safarov <s.safarov at gmail.com>:

> you can
> 1) test recording on vanilla config. If all works as expected, then;
> 2) add dialplan elements step-by-step and need to identify whats is
> elements in dialplan breaks recordings.
>
> Sergey
>
> пт, 1 июн. 2018 г. в 20:51, Ketan Kothari <kkothari157 at gmail.com>:
>
>> Any suggestion to resolver this issue ?
>>
>> On Thu, May 31, 2018 at 4:11 PM, Ketan Kothari <kkothari157 at gmail.com>
>> wrote:
>>
>>> Hello Michael,
>>>
>>> Here is my server logs where i'm getting issue :
>>> https://pastebin.freeswitch.org/view/9538bc8f
>>>
>>> freeswitch at freeswitch> status
>>> UP 0 years, 4 days, 10 hours, 45 minutes, 38 seconds, 273 milliseconds,
>>> 968 microseconds
>>> FreeSWITCH (Version 1.6.17  64bit) is ready
>>> 617 session(s) since startup
>>> 0 session(s) - peak 20, last 5min 0
>>> 0 session(s) per Sec out of max 30, peak 6, last 5min 0
>>> 1000 session(s) max
>>> min idle cpu 0.00/99.93
>>> Current Stack Size/Max 240K/8192K
>>>
>>>
>>>
>>> On Wed, May 30, 2018 at 7:24 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> I would suggest reviewing the logs and seeing if there is any
>>>> indication of what is happening
>>>>
>>>> > On May 30, 2018, at 7:20 AM, Ketan Kothari <kkothari157 at gmail.com>
>>>> wrote:
>>>> >
>>>> > Hi All,
>>>> > I'm getting issue on call recordings.
>>>> >
>>>> > Freeswitch can records maximum 15 seconds call but actual call was
>>>> around 5 minutes.
>>>> >
>>>> > Any suggestion to overcome this issue ?
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180602/687fa1b9/attachment-0001.html>

From s.safarov at gmail.com  Sat Jun  2 05:13:57 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 2 Jun 2018 08:13:57 +0300
Subject: [Freeswitch-users] How to get working leg timeout
In-Reply-To: <0FC59999-3D73-4527-9AE3-09C418A10947@vallimamod.org>
References: <CAHtxdDc4nDG+urR2OXEkDKpz7r-oPAb9Oy8yqrTRs3CTyLywTw@mail.gmail.com>
 <0FC59999-3D73-4527-9AE3-09C418A10947@vallimamod.org>
Message-ID: <CAHtxdDckr6pUG37i67=h6okEvJT95yB4uj3O3cd-kia0dKvZBg@mail.gmail.com>

Yes, tried.
In this case leg_timeout is fired but Caller also cannot listen
announcement in session progress stare of b-leg. For me this critical.


пт, 1 июн. 2018 г. в 22:23, Vallimamod Abdullah <vma at vallimamod.org>:

> Hi,
>
> Have you tried with ignore_early_media=true ?
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sip.solutions
> linkedin.com/in/vallimamod
> .
>
>
> On 1 Jun 2018, at 00:25, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> Hello i want hangup call if caller not responded for 20 sec.
> When bleg send RINGING message then leg_timeout works as expected.
> But if bleg send SESSION_PROGGRESS message then leg_timeout help.
>
> Is exist other way to get this working?
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180602/b13339c6/attachment-0001.html>

From s.safarov at gmail.com  Sat Jun  2 05:15:52 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 2 Jun 2018 08:15:52 +0300
Subject: [Freeswitch-users] How to get working leg timeout
In-Reply-To: <CAGJLwFiza5iU1Apx-aYQT8+1xy86VmLk+k66D6m+chE_OmKZOw@mail.gmail.com>
References: <CAHtxdDc4nDG+urR2OXEkDKpz7r-oPAb9Oy8yqrTRs3CTyLywTw@mail.gmail.com>
 <CAGJLwFiza5iU1Apx-aYQT8+1xy86VmLk+k66D6m+chE_OmKZOw@mail.gmail.com>
Message-ID: <CAHtxdDfCo8z+hKgqFV=X-Om5vQ6jq9UVGW-APvFQ58qJm7yWxg@mail.gmail.com>

leg_progress_timeout is fired when not received 180 or 183 message.
In my case i receive 183 message and leg_progress_timeout not help.


пт, 1 июн. 2018 г. в 22:26, Alexey Sibyakin <alex at freeswitch.com>:

> Hello,
>
> Have you tried leg_progress_timeout ?
>
> https://freeswitch.org/confluence/display/FREESWITCH/Variables+Master+List#VariablesMasterList-leg_progress_timeout
>
> Regards,
>
> Alex
>
> On Fri, Jun 1, 2018 at 7:25 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Hello i want hangup call if caller not responded for 20 sec.
>> When bleg send RINGING message then leg_timeout works as expected.
>> But if bleg send SESSION_PROGGRESS message then leg_timeout help.
>>
>> Is exist other way to get this working?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
>
>
> --
> Alex Sibyakin | Support Engineer
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
> Email: alex at freeswitch.com
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
> Need commercial support? Contact sales at freeswitch.com for details.
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180602/744bd97d/attachment-0001.html>

From s.safarov at gmail.com  Sat Jun  2 13:26:45 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 2 Jun 2018 16:26:45 +0300
Subject: [Freeswitch-users] How to get working leg timeout
In-Reply-To: <0FC59999-3D73-4527-9AE3-09C418A10947@vallimamod.org>
References: <CAHtxdDc4nDG+urR2OXEkDKpz7r-oPAb9Oy8yqrTRs3CTyLywTw@mail.gmail.com>
 <0FC59999-3D73-4527-9AE3-09C418A10947@vallimamod.org>
Message-ID: <CAHtxdDd6A4m3q8iSBo7L725QCtrOzyB6WRrwQP9W4sWp+9RmcA@mail.gmail.com>

After looking FS source, i found that ignore_early_media may be set to
value consume.
Looks as this help me.

Sergey.

пт, 1 июн. 2018 г. в 22:23, Vallimamod Abdullah <vma at vallimamod.org>:

> Hi,
>
> Have you tried with ignore_early_media=true ?
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sip.solutions
> linkedin.com/in/vallimamod
> .
>
>
> On 1 Jun 2018, at 00:25, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> Hello i want hangup call if caller not responded for 20 sec.
> When bleg send RINGING message then leg_timeout works as expected.
> But if bleg send SESSION_PROGGRESS message then leg_timeout help.
>
> Is exist other way to get this working?
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180602/8a06476e/attachment.html>

From steveayre at gmail.com  Mon Jun  4 10:32:32 2018
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 4 Jun 2018 11:32:32 +0100
Subject: [Freeswitch-users] Freeswitch records only 15 seconds of call.
In-Reply-To: <CAGqqW9_RZMTZb7ZONQMTWbsrE9o6i4LDvQW5aoKbK+zZws=MNg@mail.gmail.com>
References: <CAGqqW98UGNiA3WpptyCQsvt7i1B4gc_AbmfR-DuzqSoxGZNG5Q@mail.gmail.com>
 <1958CFFC-111B-4E0C-8EA3-6F026BAFACB9@jerris.com>
 <CAGqqW9_RZMTZb7ZONQMTWbsrE9o6i4LDvQW5aoKbK+zZws=MNg@mail.gmail.com>
Message-ID: <CAFiqYum1gkw=jsHwEkCb=sSaccNnEKz3YXZO69u0O=OZy-8MKA@mail.gmail.com>

In that example you start to record at 11:44:33 and the call hangs up at
11:44:58, only 25 seconds. How long was that recording?

On 31 May 2018 at 11:41, Ketan Kothari <kkothari157 at gmail.com> wrote:

> Hello Michael,
>
> Here is my server logs where i'm getting issue :
> https://pastebin.freeswitch.org/view/9538bc8f
>
> freeswitch at freeswitch> status
> UP 0 years, 4 days, 10 hours, 45 minutes, 38 seconds, 273 milliseconds,
> 968 microseconds
> FreeSWITCH (Version 1.6.17  64bit) is ready
> 617 session(s) since startup
> 0 session(s) - peak 20, last 5min 0
> 0 session(s) per Sec out of max 30, peak 6, last 5min 0
> 1000 session(s) max
> min idle cpu 0.00/99.93
> Current Stack Size/Max 240K/8192K
>
>
>
> On Wed, May 30, 2018 at 7:24 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I would suggest reviewing the logs and seeing if there is any indication
>> of what is happening
>>
>> > On May 30, 2018, at 7:20 AM, Ketan Kothari <kkothari157 at gmail.com>
>> wrote:
>> >
>> > Hi All,
>> > I'm getting issue on call recordings.
>> >
>> > Freeswitch can records maximum 15 seconds call but actual call was
>> around 5 minutes.
>> >
>> > Any suggestion to overcome this issue ?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180604/b1aa0c18/attachment.html>

From vma at vallimamod.org  Mon Jun  4 15:26:25 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Mon, 4 Jun 2018 17:26:25 +0200
Subject: [Freeswitch-users] What is wrong with my originate ?
In-Reply-To: <3bf792d5-5171-bc31-397d-57a838d334fb@couderc.eu>
References: <3bf792d5-5171-bc31-397d-57a838d334fb@couderc.eu>
Message-ID: <54D1C9D5-80C2-41EC-88CC-8A373C52D1B2@vallimamod.org>

Hi,

You are getting INCOMPATIBLE_DESTINATION error as codecs on leg A and leg B don't match.

To correct it, either set your inbound-codec-prefs and outbound-codec-prefs params in your profile to PCMA,PCMU or set absolute_codec_string=PCMA,PCMU before the originate (in the dialplan or on your dial-string with the {} notation).

Hope this helps.


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

> On 2 Jun 2018, at 15:44, Pierre Couderc <pierre at couderc.eu> wrote:
> 
> originate  sofia/internal/98%free.couderc.eu  0033484253887 XML int.example.net
> 
> See my traces below. The strange is that if I call from my extension 98 the same number all is fine.
> 
> I understand it may be question of CODEC, as when calling from 98 I see PCMA, PCMU in the traces. But how to correct?
> 
> Thanks to read me.
> 
> Pierre Couderc
> 
> 
> 
> ......
> 
> 2018-06-02 08:47:43.279349 [DEBUG] sofia_glue.c:1295 sofia/external/0033484253887 sending invite version: 1.6.20 -37-987c9b9 64bit
> Local SDP:
> v=0
> o=FreeSWITCH 1527903329 1527903330 IN IP4 192.168.163.67
> s=FreeSWITCH
> c=IN IP4 192.168.163.67
> t=0 0
> m=audio 25934 RTP/AVP 9 101 13
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
> a=sendrecv
> 
> 2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:40 sofia/external/0033484253887 Standard INIT
> 2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:48 (sofia/external/0033484253887) State Change CS_INIT -> CS_ROUTING
> 2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:627 (sofia/external/0033484253887) State INIT going to sleep
> 2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:584 (sofia/external/0033484253887) Running State Change CS_ROUTING (Cur 2 Tot 201)
> 2018-06-02 08:47:43.279349 [DEBUG] sofia.c:7084 Channel sofia/external/0033484253887 entering state [calling][0]
> 2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:643 (sofia/external/0033484253887) State ROUTING
> 2018-06-02 08:47:43.279349 [DEBUG] mod_sofia.c:143 sofia/external/0033484253887 SOFIA ROUTING
> 2018-06-02 08:47:43.279349 [DEBUG] switch_ivr_originate.c:67 (sofia/external/0033484253887) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:643 (sofia/external/0033484253887) State ROUTING going to sleep
> 2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:584 (sofia/external/0033484253887) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 2
> 01)
> 2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:662 (sofia/external/0033484253887) State CONSUME_MEDIA
> 2018-06-02 08:47:43.279349 [DEBUG] switch_core_state_machine.c:662 (sofia/external/0033484253887) State CONSUME_MEDIA going to sleep
> 2018-06-02 08:47:43.339353 [DEBUG] switch_rtp.c:7308 Correct audio ip/port confirmed.
> 2018-06-02 08:47:43.339353 [DEBUG] sofia.c:7084 Channel sofia/external/0033484253887 entering state [calling][0]
> 2018-06-02 08:47:43.399358 [DEBUG] sofia.c:7084 Channel sofia/external/0033484253887 entering state [terminated][488]
> 2018-06-02 08:47:43.399358 [NOTICE] sofia.c:8273 Hangup sofia/external/0033484253887 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
> 2
> 
> ....
> 



From francesco at delagarda.com  Mon Jun  4 15:54:17 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Mon, 4 Jun 2018 17:54:17 +0200
Subject: [Freeswitch-users] How to get working leg timeout
In-Reply-To: <CAJoKKdWRu_LK2owAU0TB1uvW=qTOC5+cwS+DV4dUUEHhRt=dNg@mail.gmail.com>
References: <CAHtxdDc4nDG+urR2OXEkDKpz7r-oPAb9Oy8yqrTRs3CTyLywTw@mail.gmail.com>
 <CAGJLwFiza5iU1Apx-aYQT8+1xy86VmLk+k66D6m+chE_OmKZOw@mail.gmail.com>
 <CAJoKKdWRu_LK2owAU0TB1uvW=qTOC5+cwS+DV4dUUEHhRt=dNg@mail.gmail.com>
Message-ID: <017301d3fc1c$4d23ec00$e76bc400$@delagarda.com>

.. unless you have analogue lines that don’t communicate the “on answer” (or that, at least is my experience)…
PLEASE someone prove me wrong!

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Tihomir Culjaga
Sent: lunedì 4 giugno 2018 14:16
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] How to get working leg timeout

 

you can also abuse sched_hangup as well.

 

set it to +20 when you place the call and reset it on answer.

 

On 1 June 2018 at 01:46, Alexey Sibyakin <alex at freeswitch.com <mailto:alex at freeswitch.com> > wrote:

Hello,

 

Have you tried leg_progress_timeout ?

https://freeswitch.org/confluence/display/FREESWITCH/Variables+Master+List#VariablesMasterList-leg_progress_timeout

 

Regards,

 

Alex

 

On Fri, Jun 1, 2018 at 7:25 AM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com> > wrote:

Hello i want hangup call if caller not responded for 20 sec.
When bleg send RINGING message then leg_timeout works as expected.
But if bleg send SESSION_PROGGRESS message then leg_timeout help.

Is exist other way to get this working?

 

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

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

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





 

-- 

Alex Sibyakin | Support Engineer

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

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

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

Need commercial support? Contact  <mailto:sales at freeswitch.com> sales at freeswitch.com for details.


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

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

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

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180604/fe04fa14/attachment-0001.html>

From mike at jerris.com  Mon Jun  4 17:45:58 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Jun 2018 13:45:58 -0400
Subject: [Freeswitch-users] No ACK for 200 OK
In-Reply-To: <CAF7mOn72EQFzMbrvG5LfA9jzqPYjmQbsbEKOoduMgKoCSzj04A@mail.gmail.com>
References: <CAF7mOn72EQFzMbrvG5LfA9jzqPYjmQbsbEKOoduMgKoCSzj04A@mail.gmail.com>
Message-ID: <7237CF75-C165-457C-8BDF-B55E24153450@jerris.com>

I suspect if you enabled the tport_log sofia logging you would have seen it trying to send and failing for some reason

> On Jun 3, 2018, at 3:35 AM, Xenia Obolenskaya <j4v28bsjtp43hnf865 at gmail.com> wrote:
> 
> Hi, All!
> 
> Thank All!
> 
> I apologize for my negligence and  inadvertence.
> 
> Because a bare-bones gateway was setup for outbound calls with no username/pass and without registration I omitted an important parameter 
> <param name="register-transport" value="tls"/>
> 
> Now, with this parameter, the situation has returned to normal.
> 
> A lot of thanks to All!
> 
> Xenia Obolenskaya
> _________________________________________________________________________

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180604/46ddb816/attachment.html>

From mike at jerris.com  Mon Jun  4 17:46:36 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Jun 2018 13:46:36 -0400
Subject: [Freeswitch-users] mod_verto
In-Reply-To: <CACOmXtBSxNbtKgH961AP0D=5jEpgtPe3gb+cjR-w4GU3oZKUFQ@mail.gmail.com>
References: <CACOmXtBSxNbtKgH961AP0D=5jEpgtPe3gb+cjR-w4GU3oZKUFQ@mail.gmail.com>
Message-ID: <88C5D191-E7F2-4ACF-AE27-61816DF12384@jerris.com>

mod_av supports mp4 and others.


> On Jun 3, 2018, at 4:02 AM, Vishal Pai <vishalmpai at gmail.com> wrote:
> 
> Can we record the video conferencing using mod_verto. If yes how it is possible and and what would it’s file format.



From j4v28bsjtp43hnf865 at gmail.com  Tue Jun  5 09:30:30 2018
From: j4v28bsjtp43hnf865 at gmail.com (Xenia Obolenskaya)
Date: Tue, 5 Jun 2018 09:30:30 +0000
Subject: [Freeswitch-users] No ACK for 200 OK
Message-ID: <CAF7mOn6TfbSqJP5V9GXKW6RrsBV8Q12UNmBbd_UuzyjQOpZJzw@mail.gmail.com>

Dear Michael Jerris,

Thank you for your consideration!

sofia loglevel tport 9
Sofia log level for component [tport] has been set to [9]

Here is the log when the parameter in the gateway was commented out
<!--<param name="register-transport" value="tls"/> -->
In the received packet in the header Contact:
<sip:1000@[2a02:7b43:1eb0:eb21::2]:5060;transport=udp>
the selected transport is udp!!! Freeswitch does not sent ACK
As for tport log I do not see any fatal errors in this case, maybe you, as
a specialist, can see?

tport.c:3023 tport_deliver() tport_deliver(0x7f4e6c0ce360): msg
0x7f4e6c0e7520 (1184 bytes) from tls/[2a02:7b43:1eb0:eb21::2]:5061/sips
next=(nil)
tport.c:2296 tport_set_secondary_timer() tport(0x7f4e6c0ce360): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7f4e58096970): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7f4e58096970)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f4e58096970):
tls_read() returned 4
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f4e58096970) msg
0x7f4e58068390 from (tls/181.201.33.15:39478) has 4 bytes, veclen = 1
tport.c:2296 tport_set_secondary_timer() tport(0x7f4e58096970): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7f4e6c0ce360): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7f4e6c0ce360)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f4e6c0ce360):
tls_read() returned 845
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f4e6c0ce360) msg
0x7f4e6c0e7520 from (tls/[2a02:7b43:1eb0:eb21::2]:5061) has 845 bytes,
veclen = 1
tport.c:2296 tport_set_secondary_timer() tport(0x7f4e6c0ce360): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7f4e6c0ce360): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7f4e6c0ce360)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f4e6c0ce360):
tls_read() returned 339
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f4e6c0ce360) msg
0x7f4e6c0e7520 from (tls/[2a02:7b43:1eb0:eb21::2]:5061) has 339 bytes,
veclen = 1
recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 07:05:06.088128:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bK0UjgcDj2ae5Zr;rport=54911
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=1mKgNjj91mpSe
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=jmg806XU07Dee
   Call-ID: 6368d0a7-e318-1236-2fbf-525400af4b7f
   CSeq: 123749671 INVITE
   Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5060;transport=udp>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   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: 339
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2]
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1528150294 1528150295 IN IP6 2a02:7b43:1eb0:eb21::2
   s=FreeSWITCH
   c=IN IP6 2a02:7b43:1eb0:eb21::2
   t=0 0
   m=audio 21180 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=crypto:1 AEAD_AES_256_GCM_8
inline:eezgaWEZZUvKciFAkpVWpZhdv39wLjvIyzMZMbmk6FbkYfnrFAm9eziQq5M
   ------------------------------------------------------------------------
tport.c:3023 tport_deliver() tport_deliver(0x7f4e6c0ce360): msg
0x7f4e6c0e7520 (1184 bytes) from tls/[2a02:7b43:1eb0:eb21::2]:5061/sips
next=(nil)
tport.c:2296 tport_set_secondary_timer() tport(0x7f4e6c0ce360): reset timer
tport.c:2296 tport_set_secondary_timer() tport(0x7f4e6c0ce360): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7f4e58096970): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7f4e58096970)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f4e58096970):
tls_read() returned 565
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f4e58096970) msg
0x7f4e58068390 from (tls/181.201.33.15:39478) has 565 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7f4e58096970): msg
0x7f4e58068390 (573 bytes) from tls/181.201.33.15:39478/sips next=(nil)
tport.c:2296 tport_set_secondary_timer() tport(0x7f4e58096970): reset timer
tport.c:3257 tport_tsend() tport_tsend(0x7f4e58096970) tpn = TLS/
181.201.33.15:39478
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
0x7f4e58047300 0x7f4e58097d90 628 (628)
tport.c:3594 tport_vsend() tport_vsend(0x7f4e58096970): 628 bytes of 628 to
tls/181.201.33.15:39478
tport.c:3492 tport_send_msg() tport_vsend returned 628
tport.c:2296 tport_set_secondary_timer() tport(0x7f4e58096970): reset timer
tport.c:2296 tport_set_secondary_timer() tport(0x7f4e58096970): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7f4e58096970): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7f4e58096970)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f4e58096970):
tls_read() returned 562
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f4e58096970) msg
0x7f4e58068390 from (tls/181.201.33.15:39478) has 562 bytes, veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7f4e58096970): msg
0x7f4e58068390 (562 bytes) from tls/181.201.33.15:39478/sips next=(nil)
tport.c:2296 tport_set_secondary_timer() tport(0x7f4e58096970): reset timer


===========================================================================
And here is the log when the parameter in the gateway was uncommented
<param name="register-transport" value="tls"/>  and Freeswitch worked
smoothly


recv 1184 bytes from tls/[2a02:7b43:1eb0:eb21::2]:5061 at 07:10:20.917599:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKFKyaBgKr06NHH;rport=48779
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=FQ30U1X208Qje
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=prNB8j19mB7rc
   Call-ID: 31efea17-e319-1236-2fbf-525400af4b7f
   CSeq: 123749844 INVITE
   Contact: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=tls>
   User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
   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: 339
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:1000%@[2a02:7b43:1eb0:eb21::2]
   X-FS-Support: update_display,send_info

   v=0
   o=FreeSWITCH 1528143500 1528143501 IN IP6 2a02:7b43:1eb0:eb21::2
   s=FreeSWITCH
   c=IN IP6 2a02:7b43:1eb0:eb21::2
   t=0 0
   m=audio 28320 RTP/SAVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=crypto:1 AEAD_AES_256_GCM_8
inline:7/8wl/6LKSgPhTNg1czXmvO1xFiiRNmC4Ugk8MYVqOFe+SMuWdbCWdH6hCE
   ------------------------------------------------------------------------
tport.c:3023 tport_deliver() tport_deliver(0x7f4e740d82a0): msg
0x7f4e740b6be0 (1184 bytes) from tls/[2a02:7b43:1eb0:eb21::2]:5061/sips
next=(nil)
tport.c:4222 tport_release() tport_release(0x7f4e740d82a0): 0x7f4e740d7590
by 0x7f4e740d2e30 with 0x7f4e740b6be0
tport.c:2296 tport_set_secondary_timer() tport(0x7f4e740d82a0): reset timer
2018-06-05 07:10:20.914430 [INFO] sofia.c:1279 sofia/external-ipv6/1000
Update Callee ID to "Outbound Call" <sip:1000%@[2a02:7b43:1eb0:eb21::2]>
2018-06-05 07:10:20.914430 [DEBUG] sofia.c:7084 Channel
sofia/external-ipv6/1000 entering state [completing][200]
2018-06-05 07:10:20.914430 [DEBUG] sofia.c:7094 Remote SDP:
v=0
o=FreeSWITCH 1528143500 1528143501 IN IP6 2a02:7b43:1eb0:eb21::2
s=FreeSWITCH
c=IN IP6 2a02:7b43:1eb0:eb21::2
t=0 0
m=audio 28320 RTP/SAVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=crypto:1 AEAD_AES_256_GCM_8
inline:7/8wl/6LKSgPhTNg1czXmvO1xFiiRNmC4Ugk8MYVqOFe+SMuWdbCWdH6hCE

tport.c:4588 tport_by_name() tport(0x7f4e74083b20): found 0x7f4e740d82a0 by
name tls/[2a02:7b43:1eb0:eb21::2]:5061
tport.c:3257 tport_tsend() tport_tsend(0x7f4e740d82a0) tpn =
tls/[2a02:7b43:1eb0:eb21::2]:5061
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
0x7f4e74085160 0x7f4e740d8840 483 (483)
tport.c:3594 tport_vsend() tport_vsend(0x7f4e740d82a0): 483 bytes of 483 to
tls/[2a02:7b43:1eb0:eb21::2]:5061
tport.c:3492 tport_send_msg() tport_vsend returned 483
send 483 bytes to tls/[2a02:7b43:1eb0:eb21::2]:5061 at 07:10:20.921598:
   ------------------------------------------------------------------------
   ACK sip:1000@[2a02:7b43:1eb0:eb21::2]:5061;transport=tls SIP/2.0
   Via: SIP/2.0/TLS
[2a02:f680:e0:1210::2123]:5061;branch=z9hG4bKgvQ3cB4UXFc4c
   Max-Forwards: 70
   From: "Extension 1000" <sip:1000@
[2a02:f680:e0:1210::2123]>;tag=FQ30U1X208Qje
   To: <sip:1000@[2a02:7b43:1eb0:eb21::2]:5061>;tag=prNB8j19mB7rc
   Call-ID: 31efea17-e319-1236-2fbf-525400af4b7f
   CSeq: 123749844 ACK
   Contact: <sip:gw+lt_ipv6@
[2a02:f680:e0:1210::2123]:5061;tport=tcp;transport=tls;gw=lt_ipv6>
   Content-Length: 0

   ------------------------------------------------------------------------
tport.c:2296 tport_set_secondary_timer() tport(0x7f4e740d82a0): reset timer


Regards,
Xenia Obolenskaya
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180605/b173b033/attachment.html>

From ciprian.dosoftei at gmail.com  Tue Jun  5 19:49:31 2018
From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei)
Date: Tue, 5 Jun 2018 15:49:31 -0400
Subject: [Freeswitch-users] Energy level for record_session
Message-ID: <CAEKk1MCKfqP7uK1Dp8jTX9skQy_miC6ErrV1c-tYiAsNqUxwSQ@mail.gmail.com>

Hello everyone!

I have checked the Confluence documentation and I cannot seem to find
energy level settings for the record_session application (not record).

I do however see references to a RECORD_SILENCE_THRESHOLD variable in the
source code, if I am looking at the right file (
https://docs.freeswitch.org/switch__ivr__async_8c_source.html line 2688).

Could it be enforced through a channel variable setting prior to executing
the application?

My usage scenario is pretty straight forward: I am using record_session
(with RECORD_READ_ONLY set) to record each individual channel in a
conference separately, which works great, but it also captures much more
background noise than ideal, which for muxed mod_conference recordings is
not an issue as the energy level setting will naturally clean the recording
prior to muxing. I also want the recording to continue should there be
excessive silence from a channel (unlike record for which a silence hits
threshold is almost always desirable).

Thank you!

-- 
Best Regards,
Ciprian Dosoftei

The information transmitted is intended only for the addressee and may
contain privileged and/or confidential material. If you are not the
intended recipient, kindly contact the sender and delete the message.

Any disclosure, distribution or copying of this message is strictly
prohibited without the expressed permission of the sender.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180605/fc991f81/attachment-0001.html>

From mike at jerris.com  Wed Jun  6 17:11:33 2018
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Jun 2018 13:11:33 -0400
Subject: [Freeswitch-users] mod_verto
In-Reply-To: <CACOmXtBSxNbtKgH961AP0D=5jEpgtPe3gb+cjR-w4GU3oZKUFQ@mail.gmail.com>
References: <CACOmXtBSxNbtKgH961AP0D=5jEpgtPe3gb+cjR-w4GU3oZKUFQ@mail.gmail.com>
Message-ID: <C16D05FD-E04E-4519-A75F-50BADE4FCAE0@jerris.com>

Thank you for your question! We answered it live on ClueCon weekly today! You can watch it at https://www.youtube.com/watch?v=gIsEiddBbFo <https://www.youtube.com/watch?v=gIsEiddBbFo>

Mike

> On Jun 3, 2018, at 4:02 AM, Vishal Pai <vishalmpai at gmail.com> wrote:
> 
> Hi All
> 
> 
> Can we record the video conferencing using mod_verto. If yes how it is possible and and what would it’s file format.
> 
> 
> Thanks
> Vishal Pai
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180606/7527a6a1/attachment.html>

From ap at gen-ip.fr  Thu Jun  7 15:18:27 2018
From: ap at gen-ip.fr (Alexis)
Date: Thu, 7 Jun 2018 17:18:27 +0200
Subject: [Freeswitch-users] Eavesdrop - allow bleg whisper and conference
	but deny aleg whisper
Message-ID: <825064b1-ac21-9e9d-85db-c06ec8b416f9@gen-ip.fr>

Hi,

Is it possible to permit one-leg whispering and deny other-leg 
whispering using eavesdrop application ? I want to permit my users to 
whisper "company employee leg" or do 3 ways conference but not to 
whisper only "other leg peoples".

It seems that eavesdrop_whisper_aleg=false set which "eavesdrop mode" is 
set at the beginning of eavesdrop application but doesn't restrict any 
DTMF (i.e doesn't disable DTMF 2 [whisper with the other half]).

-- 
Alexis



From david.villasmil.work at gmail.com  Wed Jun  6 19:02:54 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 6 Jun 2018 21:02:54 +0200
Subject: [Freeswitch-users] wiki and org sites
Message-ID: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>

Hello Guys,

I can't connect to wiki.freeswitch.org, is the web down? I just get
redirected to freeswitch's (new?) commercial web page (freeswitch.com)

thanks!

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
ᐧ
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180606/40e6dee7/attachment.html>

From david.witham at netsip.com.au  Thu Jun  7 00:04:39 2018
From: david.witham at netsip.com.au (David Witham)
Date: Thu, 7 Jun 2018 10:04:39 +1000
Subject: [Freeswitch-users] FS skips an RTP timestamp after an RTCP Sender
	Report
Message-ID: <CAPY30xZnYbuhWnYmkw3kzOoF5kQhQV8svwjcTVQHG7094am6rg@mail.gmail.com>

Hi all,

Wondering if others have seen this. Running FS 1.6.20 on Debian 8 as B2BUA,
no proxy media. I'm not sure if this is a bug or whether its environmental
in some way. If we dig further and think it really is a bug we'll log a
proper bug tomorrow.

A call comes in interface A, does not have any RTCP packets. Call is
bridged to interface B which does contain RTCP packets. Same confg on both
sofia profiles. The RTP packet in the A->B stream immediately after the
RTCP packet sent by FS skips a timestamp, even though the sequence number
increments correctly.

e.g
Interface A packets
4.58 seconds RTP timestamp=855198950 seq=53269
4.60 seconds RTP timestamp=855199110 seq=53270
4.62 seconds RTP timestamp=855199270 seq=53271
4.64 seconds RTP timestamp=855199430 seq=53272

Interface B packets
4.77 seconds RTP timestamp=855198950 seq=20446
4.79 seconds RTP timestamp=855199110 seq=20447
4.80 seconds RTCP Sender Report
???
4.83 seconds RTP timestamp=855199430 seq=20448

In the above example, the packet with timestamp 855199270 does not appear
at 4.81 seconds as expected and the next packet contains the next
timestamp 855199430
but the sequence numbers keep incrementing normally. Its like FS has
dropped that packet internally. We can hear this as an audio artefact when
listening to the call.

In this case I don't think rewriting timestamps will help if the packet is
being dropped - we'll still hear the missing audio.

Secondly, in troubleshooting this issue, I've noticed that most of our
traffic does not contain RTCP packets, but some do and I can't seem to find
any rhyme or reason for this behaviour. Does anyone know why a call will or
won't contain RTCP? I had 2 calls in a minute go via 2 FS boxes running the
same config, same FS version, same A to B numbers and path through the
network and one had RTCP and one didn't. You'd think if the same CPE made
the same call to the same FS configuration (albeit a different physical box
on the same LAN segment), that you'd see the same behaviour.

thanks,
David
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/9fdf3a2e/attachment-0001.html>

From pierre at couderc.eu  Wed Jun  6 21:41:56 2018
From: pierre at couderc.eu (Pierre Couderc)
Date: Wed, 6 Jun 2018 23:41:56 +0200
Subject: [Freeswitch-users] What is wrong with my originate ?
In-Reply-To: <54D1C9D5-80C2-41EC-88CC-8A373C52D1B2@vallimamod.org>
References: <3bf792d5-5171-bc31-397d-57a838d334fb@couderc.eu>
 <54D1C9D5-80C2-41EC-88CC-8A373C52D1B2@vallimamod.org>
Message-ID: <158828e4-ec5f-7436-d954-0fc6470044ba@couderc.eu>

Thank you very much, Abdullah, it works fine!


On 06/04/2018 05:26 PM, Vallimamod Abdullah wrote:
> Hi,
>
> You are getting INCOMPATIBLE_DESTINATION error as codecs on leg A and leg B don't match.
>
> To correct it, either set your inbound-codec-prefs and outbound-codec-prefs params in your profile to PCMA,PCMU or set absolute_codec_string=PCMA,PCMU before the originate (in the dialplan or on your dial-string with the {} notation).
>
> Hope this helps.
>
>
> Best Regards,



From francesco at delagarda.com  Wed Jun  6 18:05:51 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Wed, 6 Jun 2018 20:05:51 +0200
Subject: [Freeswitch-users] warn and hangup after xx seconds
Message-ID: <000201d3fdc1$02be55a0$083b00e0$@delagarda.com>

I am scripting in js

I have both analogue and digital trunks out (analogue trunks out don't give
me the "answer" event)

Lets say we have a "totTime" variable for max length of call

I need to play a warning at totTime - 30

And hangup at totTime..

I have tried using:

 

 

sessOut.execute("set", "execute_on_answer_1=sched_hangup +" + totTime + "
alloted_timeout");

sessOut.execute("set", "execute_on_answer_2=sched_broadcast +" + (totTime -
30) + "layback::" + soundDir + "time_off.wav both");

 

But I can only get one or the other to work.. never both!

 

Also, how would I schedule there two events for analogue lines (where I
don't have "on_answer")

 

Thanks all

 

 

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180606/dc0c6a72/attachment.html>

From f.antonini at tiesse.com  Thu Jun  7 09:05:13 2018
From: f.antonini at tiesse.com (fabio)
Date: Thu, 7 Jun 2018 11:05:13 +0200
Subject: [Freeswitch-users] SIP over TLS configuration problem
Message-ID: <193f5f5a-bc26-c0d6-116f-97bc160cb654@tiesse.com>

Hi all


I'm a Freeswitch newbie and I'm trying to setup SIP over TLS in my FS 
version 1.5.15.

As first step I have configured a SIP Gateway that successfully 
registers to a dedicated SIP Registrar/Proxy (opensips) using SIP over 
UDP. With this configuration I can successfully place outbound and 
inbound calls without any problem. Everything works as a charm.

Further I have tried to switch to SIP over TLS and I followed the steps 
described in https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS.

I have installed the agent.pem and cafile.pem generated by opensips (my 
SIP Registrar) and I configured FS to use them. After restart the sofia 
gateway profile can successfully register to the SIP Registrar by SIP 
over TLS.

Further I can successfully place outbound call (from internal channel 
through the SIP gateway).  It sounds great!

Unfortunately FS fails to handle inbound calls (SIP INVITE from an 
external SIP UA registered to the same SIP Registrar to the SIP UA 
extension of the FS SIP gateway).

I have tried to trace all the logs I can. Here below some traces from 
the FS console when an inbound INVITE is received:


tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb7f28): events IN
tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0xb7f28): new 
secondary tport 0x1398c0
tport_type_tcp.c:203 tport_tcp_init_secondary() 
tport_tcp_init_secondary(0x1398c0): Setting TCP_KEEPIDLE to 30
tport_type_tcp.c:209 tport_tcp_init_secondary() 
tport_tcp_init_secondary(0x1398c0): Setting TCP_KEEPINTVL to 30
tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x1398c0): new 
connection from tls/10.3.10.110:38632/sips
tport_tls.c:919 tls_connect() tls_connect(0x1398c0): events NEGOTIATING
tport_tls.c:1008 tls_connect() tls_connect(0x1398c0): TLS setup failed 
(error:00000001:lib(0):func(0):reason(1))
tport.c:2090 tport_close() tport_close(0x1398c0): tls/10.3.10.110:38632/sips
tport.c:2263 tport_set_secondary_timer() tport(0x1398c0): set timer at 0 
ms because zap


In order to simplify the test I have also tried to connect to the 5061 
TLS port by a simple openssl command from a linux shell of the SIP 
Registrar box:


openssl  s_client -connect 10.11.4.103:5061 -tls1_2
CONNECTED(00000003)
3074304200:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert 
handshake failure:s3_pkt.c:1256:SSL alert number 40
3074304200:error:1409E0E5:SSL routines:SSL3_WRITE_BYTES:ssl handshake 
failure:s3_pkt.c:596:
---
no peer certificate available
---
No client certificate CA names sent
---
SSL handshake has read 7 bytes and written 0 bytes
---
New, (NONE), Cipher is (NONE)
Secure Renegotiation IS NOT supported
Compression: NONE
Expansion: NONE
SSL-Session:
     Protocol  : TLSv1.2
     Cipher    : 0000
     Session-ID:
     Session-ID-ctx:
     Master-Key:
     Key-Arg   : None
     PSK identity: None
     PSK identity hint: None
     SRP username: None
     Start Time: 1528361426
     Timeout   : 7200 (sec)
     Verify return code: 0 (ok)
---

In the FS console I read the same traces received in the previous test 
with the inbound call.


tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb7f28): events IN
tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0xb7f28): new 
secondary tport 0x248210
tport_type_tcp.c:203 tport_tcp_init_secondary() 
tport_tcp_init_secondary(0x248210): Setting TCP_KEEPIDLE to 30
tport_type_tcp.c:209 tport_tcp_init_secondary() 
tport_tcp_init_secondary(0x248210): Setting TCP_KEEPINTVL to 30
tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x248210): new 
connection from tls/10.11.4.103:33168/sips
tport_tls.c:919 tls_connect() tls_connect(0x248210): events NEGOTIATING
tport_tls.c:1008 tls_connect() tls_connect(0x248210): TLS setup failed 
(error:00000001:lib(0):func(0):reason(1))
tport.c:2090 tport_close() tport_close(0x248210): tls/10.11.4.103:33168/sips
tport.c:2263 tport_set_secondary_timer() tport(0x248210): set timer at 0 
ms because zap

I have attached also a wireshark capture of the inbound call. In this 
capture the SIP Registrar has IP 10.3.10.110. The FS device is 
10.11.4.103. The Client Hello is sent by the SIP Registrar, but the FS 
device replies with an "Alert: Level: fatal, Description: handshake 
failure (40).

I guess that there is some misconfiguration related to the TLS version 
or proposed ciphers  or any certifcates but I cannot understand what.


For comparison I have tried to run the same openssl command from FS to 
the external SIP Registrar (outbound).


openssl  s_client -connect 10.3.10.110:5061 -tls1_2
CONNECTED(00000003)
depth=1 CN = Your_NAME, ST = Your_STATE, C = CO, emailAddress = 
YOUR_EMAIL, O = YOUR_ORG_NAME
verify error:num=19:self signed certificate in certificate chain
verify return:0
---
Certificate chain
  0 s:/C=XY/ST=Some State/O=My Large Organization Name/OU=My Subunit of 
Large 
Organization/CN=somename.somewhere.com/emailAddress=root at somename.somewhere.com
i:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
  1 
s:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
i:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
---
Server certificate
-----BEGIN CERTIFICATE-----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-----END CERTIFICATE-----
subject=/C=XY/ST=Some State/O=My Large Organization Name/OU=My Subunit 
of Large 
Organization/CN=somename.somewhere.com/emailAddress=root at somename.somewhere.com
issuer=/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
---
No client certificate CA names sent
---
SSL handshake has read 1979 bytes and written 337 bytes
---
New, TLSv1/SSLv3, Cipher is AES256-GCM-SHA384
Server public key is 512 bit
Secure Renegotiation IS NOT supported
Compression: NONE
Expansion: NONE
SSL-Session:
     Protocol  : TLSv1.2
     Cipher    : AES256-GCM-SHA384
     Session-ID: 
EA8B17008E58F1D04CD1CEA53103CF477AA9DE0DC80A4FF4F0DD4814031E4C15
     Session-ID-ctx:
     Master-Key: 
D28ED5C21D288944D2277AF86FE82A9BF3BEDABAA14DBCD5AE32B190EF0A0CA6AB99719E751E6DD4FECAA9DD1307A3C0
     Key-Arg   : None
     PSK identity: None
     PSK identity hint: None
     SRP username: None
     TLS session ticket lifetime hint: 300 (seconds)
     TLS session ticket:
     0000 - 2f c5 82 ea bf 8b 66 49-bc bc ee 48 1a fb 8e 6c /.....fI...H...l
     0010 - de 42 d9 e0 6e 36 40 78-06 cc 68 c6 74 6d 6e aa .B..n6 at x..h.tmn.
     0020 - b6 53 8a ed b2 8d 5a c4-02 e1 88 8b d2 a9 56 5f .S....Z.......V_
     0030 - ee c6 b9 14 55 da 37 df-8f aa af 81 b4 22 4e be ....U.7......"N.
     0040 - 9c c5 87 d6 46 22 47 03-4a 88 dd 1e 9d 05 81 09 ....F"G.J.......
     0050 - c3 8b 9f 44 29 90 4d 93-c9 f5 41 e2 4d 72 1b de ...D).M...A.Mr..
     0060 - 8d c2 15 ab 49 ad da 26-0e 72 a9 01 02 3e 89 33 ....I..&.r...>.3
     0070 - 6e 6c 2f 20 1c 15 06 7a-8d c5 a6 6e ee 46 d2 76   nl/ 
...z...n.F.v
     0080 - 63 c1 89 1e 9b 3c a1 10-d0 78 31 9e e6 8e 86 ab c....<...x1.....
     0090 - ff bc 3a 4c ab 3d 33 8f-e9 56 c5 f1 45 46 73 41 ..:L.=3..V..EFsA

     Start Time: 1528361487
     Timeout   : 7200 (sec)
     Verify return code: 19 (self signed certificate in certificate chain)
---
closed


In this case the command seems to have been successfully executed. I 
remark that the outbound TLS transactions seems to be working fine also 
from FS (SIP Registrar, SIP INVITE in outbound don't have any problem).

If required I can provide also the FS configuration files (vars.xml, 
sofia.conf.xml,  etc etc).

Any help will be greatly appreciated.

Thanks in advance

Best regards


fabio

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/87fe4bb9/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: siptls-inbound.pcap
Type: application/vnd.tcpdump.pcap
Size: 1092 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/87fe4bb9/attachment-0001.pcap>

From davidswalkabout at gmail.com  Wed Jun  6 18:33:53 2018
From: davidswalkabout at gmail.com (David P)
Date: Wed, 6 Jun 2018 11:33:53 -0700
Subject: [Freeswitch-users] Achieving TLS + SRTP for inbound calls
In-Reply-To: <C8FC74F4-F84B-458B-BCF3-8E8D6829D4E9@jerris.com>
References: <CAE23truacwofg__v2ZGimOXn46D+ZMAT2mkkkUcwD6q0+KsdwA@mail.gmail.com>
 <C8FC74F4-F84B-458B-BCF3-8E8D6829D4E9@jerris.com>
Message-ID: <CAE23trsBgj_SSDkc2t88wwP+KTwEd9y69-jYYEVv+2O4PCGzEw@mail.gmail.com>

Hi Mike,

In today's weekly webinar, you answered a question about enabling video via
verto from secured debian jessie FS. I didn't catch all of your answer. I
think these webinars are archived, but I don't see an archive at
https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Conference+call

I searched your confluence, and just found someone else ask the same
question:
https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie?focusedCommentId=16354730#comment-16354730

I think you said we need to use "mod_av" (as well as mod_verto) but
entering that into your confluence searchbox didn't result in a hit.

Would you repeat your advice here?

Cheers,
David
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180606/fc99896a/attachment.html>

From mike at jerris.com  Thu Jun  7 17:24:54 2018
From: mike at jerris.com (Michael Jerris)
Date: Thu, 7 Jun 2018 13:24:54 -0400
Subject: [Freeswitch-users] wiki and org sites
In-Reply-To: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>
References: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>
Message-ID: <E785B568-3F86-4B2D-90E3-7771E417FCD3@jerris.com>

wiki.freeswitch.org <http://wiki.freeswitch.org/> has been deprecated for many years now.  The correct location for the wiki is https://freeswitch.org/confluence <https://freeswitch.org/confluence>

Mike

> On Jun 6, 2018, at 3:02 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> I can't connect to wiki.freeswitch.org <http://wiki.freeswitch.org/>, is the web down? I just get redirected to freeswitch's (new?) commercial web page (freeswitch.com <http://freeswitch.com/>)

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/5269e3e6/attachment.html>

From joel at textplus.com  Thu Jun  7 16:51:36 2018
From: joel at textplus.com (Joel Serrano)
Date: Thu, 7 Jun 2018 09:51:36 -0700
Subject: [Freeswitch-users] SIP over TLS configuration problem
In-Reply-To: <193f5f5a-bc26-c0d6-116f-97bc160cb654@tiesse.com>
References: <193f5f5a-bc26-c0d6-116f-97bc160cb654@tiesse.com>
Message-ID: <CAMtXxQ=FwqpKEUHOQZfrH3gHEKy1Dw4S0TZLJgXTjf6RDS-G-Q@mail.gmail.com>

Hi Fabio,

First thing you should do is try on latest 1.6.X version, 1.5 was never
even released AFAIK.



On Thu, Jun 7, 2018 at 2:05 AM, fabio <f.antonini at tiesse.com> wrote:

> Hi all
>
>
> I'm a Freeswitch newbie and I'm trying to setup SIP over TLS in my FS
> version 1.5.15.
>
> As first step I have configured a SIP Gateway that successfully registers
> to a dedicated SIP Registrar/Proxy (opensips) using SIP over UDP. With this
> configuration I can successfully place outbound and inbound calls without
> any problem. Everything works as a charm.
>
> Further I have tried to switch to SIP over TLS and I followed the steps
> described in https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS.
>
> I have installed the agent.pem and cafile.pem generated by opensips (my
> SIP Registrar) and I configured FS to use them. After restart the sofia
> gateway profile can successfully register to the SIP Registrar by SIP over
> TLS.
>
> Further I can successfully place outbound call (from internal channel
> through the SIP gateway).  It sounds great!
>
> Unfortunately FS fails to handle inbound calls (SIP INVITE from an
> external SIP UA registered to the same SIP Registrar to the SIP UA
> extension of the FS SIP gateway).
>
> I have tried to trace all the logs I can. Here below some traces from the
> FS console when an inbound INVITE is received:
>
>
> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb7f28): events IN
> tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0xb7f28): new
> secondary tport 0x1398c0
> tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(0x1398c0):
> Setting TCP_KEEPIDLE to 30
> tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(0x1398c0):
> Setting TCP_KEEPINTVL to 30
> tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x1398c0): new
> connection from tls/10.3.10.110:38632/sips
> tport_tls.c:919 tls_connect() tls_connect(0x1398c0): events NEGOTIATING
> tport_tls.c:1008 tls_connect() tls_connect(0x1398c0): TLS setup failed
> (error:00000001:lib(0):func(0):reason(1))
> tport.c:2090 tport_close() tport_close(0x1398c0): tls/
> 10.3.10.110:38632/sips
> tport.c:2263 tport_set_secondary_timer() tport(0x1398c0): set timer at 0
> ms because zap
>
>
> In order to simplify the test I have also tried to connect to the 5061 TLS
> port by a simple openssl command from a linux shell of the SIP Registrar
> box:
>
>
> openssl  s_client -connect 10.11.4.103:5061 -tls1_2
> CONNECTED(00000003)
> 3074304200:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert
> handshake failure:s3_pkt.c:1256:SSL alert number 40
> 3074304200:error:1409E0E5:SSL routines:SSL3_WRITE_BYTES:ssl handshake
> failure:s3_pkt.c:596:
> ---
> no peer certificate available
> ---
> No client certificate CA names sent
> ---
> SSL handshake has read 7 bytes and written 0 bytes
> ---
> New, (NONE), Cipher is (NONE)
> Secure Renegotiation IS NOT supported
> Compression: NONE
> Expansion: NONE
> SSL-Session:
>     Protocol  : TLSv1.2
>     Cipher    : 0000
>     Session-ID:
>     Session-ID-ctx:
>     Master-Key:
>     Key-Arg   : None
>     PSK identity: None
>     PSK identity hint: None
>     SRP username: None
>     Start Time: 1528361426
>     Timeout   : 7200 (sec)
>     Verify return code: 0 (ok)
> ---
>
> In the FS console I read the same traces received in the previous test
> with the inbound call.
>
>
> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb7f28): events IN
> tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0xb7f28): new
> secondary tport 0x248210
> tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(0x248210):
> Setting TCP_KEEPIDLE to 30
> tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(0x248210):
> Setting TCP_KEEPINTVL to 30
> tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x248210): new
> connection from tls/10.11.4.103:33168/sips
> tport_tls.c:919 tls_connect() tls_connect(0x248210): events NEGOTIATING
> tport_tls.c:1008 tls_connect() tls_connect(0x248210): TLS setup failed
> (error:00000001:lib(0):func(0):reason(1))
> tport.c:2090 tport_close() tport_close(0x248210): tls/
> 10.11.4.103:33168/sips
> tport.c:2263 tport_set_secondary_timer() tport(0x248210): set timer at 0
> ms because zap
>
> I have attached also a wireshark capture of the inbound call. In this
> capture the SIP Registrar has IP 10.3.10.110. The FS device is 10.11.4.103.
> The Client Hello is sent by the SIP Registrar, but the FS device replies
> with an "Alert: Level: fatal, Description:  handshake failure (40).
>
> I guess that there is some misconfiguration related to the TLS version or
> proposed ciphers  or any certifcates but I cannot understand what.
>
>
> For comparison I have tried to run the same openssl command from FS to the
> external SIP Registrar (outbound).
>
>
> openssl  s_client -connect 10.3.10.110:5061 -tls1_2
> CONNECTED(00000003)
> depth=1 CN = Your_NAME, ST = Your_STATE, C = CO, emailAddress =
> YOUR_EMAIL, O = YOUR_ORG_NAME
> verify error:num=19:self signed certificate in certificate chain
> verify return:0
> ---
> Certificate chain
>  0 s:/C=XY/ST=Some State/O=My Large Organization Name/OU=My Subunit of
> Large Organization/CN=somename.somewhere.com/emailAddress=
> root at somename.somewhere.com
>    i:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/
> O=YOUR_ORG_NAME
>  1 s:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/
> O=YOUR_ORG_NAME
>    i:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/
> O=YOUR_ORG_NAME
> ---
> Server certificate
> -----BEGIN CERTIFICATE-----
> MIIC6TCCAdGgAwIBAgIBATANBgkqhkiG9w0BAQUFADBpMRIwEAYDVQQDFAlZb3Vy
> X05BTUUxEzARBgNVBAgUCllvdXJfU1RBVEUxCzAJBgNVBAYTAkNPMRkwFwYJKoZI
> hvcNAQkBFgpZT1VSX0VNQUlMMRYwFAYDVQQKFA1ZT1VSX09SR19OQU1FMB4XDTE4
> MDUwODEyMzcyM1oXDTE5MDUwODEyMzcyM1owgb8xCzAJBgNVBAYTAlhZMRMwEQYD
> VQQIEwpTb21lIFN0YXRlMSMwIQYDVQQKExpNeSBMYXJnZSBPcmdhbml6YXRpb24g
> TmFtZTEpMCcGA1UECxMgTXkgU3VidW5pdCBvZiBMYXJnZSBPcmdhbml6YXRpb24x
> HzAdBgNVBAMTFnNvbWVuYW1lLnNvbWV3aGVyZS5jb20xKjAoBgkqhkiG9w0BCQEW
> G3Jvb3RAc29tZW5hbWUuc29tZXdoZXJlLmNvbTBcMA0GCSqGSIb3DQEBAQUAA0sA
> MEgCQQDL7uikSc1kVIvw5rhyQzk2dSJcmJ6EJ1LSmtAoafZH8bqfZ25cDQZQGi05
> YcuxGR0vSaW7xPnyhaWCLQlxQFx7AgMBAAGjDTALMAkGA1UdEwQCMAAwDQYJKoZI
> hvcNAQEFBQADggEBAHv4WzGdYhoEyHZmBQTVjdEKOVBMnNoOqum79uzWtSzSjG4E
> pP/9c331uT7fBZ/Z7XNhIV+PbDZXorLgUhwwT7zxYURNnV52Of2SWRmWtPBrgEX1
> +8S0IMtJFfJta8FAfTTaNqLpRDaiTQs3em1Maxls15cTyRQzMIjIJnY4eRrh5CNM
> YV/+kg/lpKAe0awiMu96cxpnMdz9h33g7RedBnh9wDi6k7pfYtvlC6o4snZO01AN
> 8qRiQf54OPvKcVeseJFBPWLhdYns6g+/SXhq1Lek2us93ZpuKgIaBtzkyDm2+SFa
> QXF9f0a+UuEdPvrtvMjAijcDwcaXq0r2f2MA++M=
> -----END CERTIFICATE-----
> subject=/C=XY/ST=Some State/O=My Large Organization Name/OU=My Subunit of
> Large Organization/CN=somename.somewhere.com/emailAddress=
> root at somename.somewhere.com
> issuer=/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_
> EMAIL/O=YOUR_ORG_NAME
> ---
> No client certificate CA names sent
> ---
> SSL handshake has read 1979 bytes and written 337 bytes
> ---
> New, TLSv1/SSLv3, Cipher is AES256-GCM-SHA384
> Server public key is 512 bit
> Secure Renegotiation IS NOT supported
> Compression: NONE
> Expansion: NONE
> SSL-Session:
>     Protocol  : TLSv1.2
>     Cipher    : AES256-GCM-SHA384
>     Session-ID: EA8B17008E58F1D04CD1CEA53103CF
> 477AA9DE0DC80A4FF4F0DD4814031E4C15
>     Session-ID-ctx:
>     Master-Key: D28ED5C21D288944D2277AF86FE82A
> 9BF3BEDABAA14DBCD5AE32B190EF0A0CA6AB99719E751E6DD4FECAA9DD1307A3C0
>     Key-Arg   : None
>     PSK identity: None
>     PSK identity hint: None
>     SRP username: None
>     TLS session ticket lifetime hint: 300 (seconds)
>     TLS session ticket:
>     0000 - 2f c5 82 ea bf 8b 66 49-bc bc ee 48 1a fb 8e 6c
> /.....fI...H...l
>     0010 - de 42 d9 e0 6e 36 40 78-06 cc 68 c6 74 6d 6e aa   .B..n6 at x.
> .h.tmn.
>     0020 - b6 53 8a ed b2 8d 5a c4-02 e1 88 8b d2 a9 56 5f
> .S....Z.......V_
>     0030 - ee c6 b9 14 55 da 37 df-8f aa af 81 b4 22 4e be
> ....U.7......"N.
>     0040 - 9c c5 87 d6 46 22 47 03-4a 88 dd 1e 9d 05 81 09
> ....F"G.J.......
>     0050 - c3 8b 9f 44 29 90 4d 93-c9 f5 41 e2 4d 72 1b de
> ...D).M...A.Mr..
>     0060 - 8d c2 15 ab 49 ad da 26-0e 72 a9 01 02 3e 89 33
> ....I..&.r...>.3
>     0070 - 6e 6c 2f 20 1c 15 06 7a-8d c5 a6 6e ee 46 d2 76   nl/
> ...z...n.F.v
>     0080 - 63 c1 89 1e 9b 3c a1 10-d0 78 31 9e e6 8e 86 ab
> c....<...x1.....
>     0090 - ff bc 3a 4c ab 3d 33 8f-e9 56 c5 f1 45 46 73 41
> ..:L.=3..V..EFsA
>
>     Start Time: 1528361487
>     Timeout   : 7200 (sec)
>     Verify return code: 19 (self signed certificate in certificate chain)
> ---
> closed
>
>
> In this case the command seems to have been successfully executed. I
> remark that the outbound TLS transactions seems to be working fine also
> from FS (SIP Registrar, SIP INVITE in outbound don't have any problem).
>
> If required I can provide also the FS configuration files (vars.xml,
> sofia.conf.xml,  etc etc).
>
> Any help will be greatly appreciated.
>
> Thanks in advance
>
> Best regards
>
>
> fabio
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/a958c932/attachment-0001.html>

From francesco at delagarda.com  Thu Jun  7 16:52:51 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Thu, 7 Jun 2018 18:52:51 +0200
Subject: [Freeswitch-users] warn and hangup after xx seconds
In-Reply-To: <000201d3fdc1$02be55a0$083b00e0$@delagarda.com>
References: <000201d3fdc1$02be55a0$083b00e0$@delagarda.com>
Message-ID: <018601d3fe7f$fa26e0f0$ee74a2d0$@delagarda.com>

Sometimes I have the IMMESURABLE pleasure of posting and sharing the
solution of a problem I had


 

The problem:

I wanted to be able to :

*	Broadcast a soundfile saying “30 seconds left” 
 to both parties
*	Terminating a call

After having calculated the max time allowed for a call

In both cases of:

*	Digital lines (that have the “on answer” event)
*	Analogue lines (that DON’T; therefore I use a “grace” period from
when dialing ends


All this in javascript..

 

So I did:

                var endSecs = secsMax + secsGrace;

                var warnSecs = secsMax +secsGrace - 30;             

where secsGrace is ZERO for digital lines 


 

then, for digital I use after the session out ready, so :

if (sessOut.ready())         {

                                               sessOut.execute("set",
"execute_on_answer_1=sched_broadcast +" + warnSecs + " playback::" +
soundDir + "time_off.wav both");

                                               sessOut.execute("set",
"execute_on_answer_2=sched_hangup +" + endSecs);

}

 

 

And for analogue, BEFORE thesessout ready

 

                               sessOut.execute("sched_hangup",  "+" +
endSecs + " alloted_timeout");

                               sessOut.execute("sched_broadcast", "+" +
warnSecs + " " + soundDir + "time_off.wav both");

                

Tried and tested..

 

I HOPE it helps someone

 

 

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On
Behalf Of Francesco Facco de Lagarda
Sent: mercoledì 6 giugno 2018 20:06
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] warn and hangup after xx seconds

 

I am scripting in js

I have both analogue and digital trunks out (analogue trunks out don’t give
me the “answer” event)

Lets say we have a “totTime” variable for max length of call

I need to play a warning at totTime – 30

And hangup at totTime..

I have tried using:

 

 

sessOut.execute("set", "execute_on_answer_1=sched_hangup +” + totTime + “
alloted_timeout");

sessOut.execute("set", "execute_on_answer_2=sched_broadcast +” + (totTime –
30) + “layback::" + soundDir + "time_off.wav both");

 

But I can only get one or the other to work.. never both!

 

Also, how would I schedule there two events for analogue lines (where I
don’t have “on_answer”)

 

Thanks all

 

 

 

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/aebc05fd/attachment.html>

From david.villasmil.work at gmail.com  Thu Jun  7 17:52:38 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Jun 2018 19:52:38 +0200
Subject: [Freeswitch-users] SIP over TLS configuration problem
In-Reply-To: <193f5f5a-bc26-c0d6-116f-97bc160cb654@tiesse.com>
References: <193f5f5a-bc26-c0d6-116f-97bc160cb654@tiesse.com>
Message-ID: <CAFGRPVr=X2XOhu8v2wqi4QjHK69sxqLgFAbU7r83QrukYkFmuw@mail.gmail.com>

If the certs are self-signed you're going to have many many problems. I
followed that tutorial with a valid cert and worked beautifully. Also i did
it with 1.6.

On Thu, Jun 7, 2018, 18:40 fabio <f.antonini at tiesse.com> wrote:

> Hi all
>
>
> I'm a Freeswitch newbie and I'm trying to setup SIP over TLS in my FS
> version 1.5.15.
>
> As first step I have configured a SIP Gateway that successfully registers
> to a dedicated SIP Registrar/Proxy (opensips) using SIP over UDP. With this
> configuration I can successfully place outbound and inbound calls without
> any problem. Everything works as a charm.
>
> Further I have tried to switch to SIP over TLS and I followed the steps
> described in https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS.
>
> I have installed the agent.pem and cafile.pem generated by opensips (my
> SIP Registrar) and I configured FS to use them. After restart the sofia
> gateway profile can successfully register to the SIP Registrar by SIP over
> TLS.
>
> Further I can successfully place outbound call (from internal channel
> through the SIP gateway).  It sounds great!
>
> Unfortunately FS fails to handle inbound calls (SIP INVITE from an
> external SIP UA registered to the same SIP Registrar to the SIP UA
> extension of the FS SIP gateway).
>
> I have tried to trace all the logs I can. Here below some traces from the
> FS console when an inbound INVITE is received:
>
>
> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb7f28): events IN
> tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0xb7f28): new
> secondary tport 0x1398c0
> tport_type_tcp.c:203 tport_tcp_init_secondary()
> tport_tcp_init_secondary(0x1398c0): Setting TCP_KEEPIDLE to 30
> tport_type_tcp.c:209 tport_tcp_init_secondary()
> tport_tcp_init_secondary(0x1398c0): Setting TCP_KEEPINTVL to 30
> tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x1398c0): new
> connection from tls/10.3.10.110:38632/sips
> tport_tls.c:919 tls_connect() tls_connect(0x1398c0): events NEGOTIATING
> tport_tls.c:1008 tls_connect() tls_connect(0x1398c0): TLS setup failed
> (error:00000001:lib(0):func(0):reason(1))
> tport.c:2090 tport_close() tport_close(0x1398c0): tls/
> 10.3.10.110:38632/sips
> tport.c:2263 tport_set_secondary_timer() tport(0x1398c0): set timer at 0
> ms because zap
>
>
> In order to simplify the test I have also tried to connect to the 5061 TLS
> port by a simple openssl command from a linux shell of the SIP Registrar
> box:
>
>
> openssl  s_client -connect 10.11.4.103:5061 -tls1_2
> CONNECTED(00000003)
> 3074304200:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert
> handshake failure:s3_pkt.c:1256:SSL alert number 40
> 3074304200:error:1409E0E5:SSL routines:SSL3_WRITE_BYTES:ssl handshake
> failure:s3_pkt.c:596:
> ---
> no peer certificate available
> ---
> No client certificate CA names sent
> ---
> SSL handshake has read 7 bytes and written 0 bytes
> ---
> New, (NONE), Cipher is (NONE)
> Secure Renegotiation IS NOT supported
> Compression: NONE
> Expansion: NONE
> SSL-Session:
>     Protocol  : TLSv1.2
>     Cipher    : 0000
>     Session-ID:
>     Session-ID-ctx:
>     Master-Key:
>     Key-Arg   : None
>     PSK identity: None
>     PSK identity hint: None
>     SRP username: None
>     Start Time: 1528361426
>     Timeout   : 7200 (sec)
>     Verify return code: 0 (ok)
> ---
>
> In the FS console I read the same traces received in the previous test
> with the inbound call.
>
>
> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb7f28): events IN
> tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0xb7f28): new
> secondary tport 0x248210
> tport_type_tcp.c:203 tport_tcp_init_secondary()
> tport_tcp_init_secondary(0x248210): Setting TCP_KEEPIDLE to 30
> tport_type_tcp.c:209 tport_tcp_init_secondary()
> tport_tcp_init_secondary(0x248210): Setting TCP_KEEPINTVL to 30
> tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x248210): new
> connection from tls/10.11.4.103:33168/sips
> tport_tls.c:919 tls_connect() tls_connect(0x248210): events NEGOTIATING
> tport_tls.c:1008 tls_connect() tls_connect(0x248210): TLS setup failed
> (error:00000001:lib(0):func(0):reason(1))
> tport.c:2090 tport_close() tport_close(0x248210): tls/
> 10.11.4.103:33168/sips
> tport.c:2263 tport_set_secondary_timer() tport(0x248210): set timer at 0
> ms because zap
>
> I have attached also a wireshark capture of the inbound call. In this
> capture the SIP Registrar has IP 10.3.10.110. The FS device is 10.11.4.103.
> The Client Hello is sent by the SIP Registrar, but the FS device replies
> with an "Alert: Level: fatal, Description:  handshake failure (40).
>
> I guess that there is some misconfiguration related to the TLS version or
> proposed ciphers  or any certifcates but I cannot understand what.
>
>
> For comparison I have tried to run the same openssl command from FS to the
> external SIP Registrar (outbound).
>
>
> openssl  s_client -connect 10.3.10.110:5061 -tls1_2
> CONNECTED(00000003)
> depth=1 CN = Your_NAME, ST = Your_STATE, C = CO, emailAddress =
> YOUR_EMAIL, O = YOUR_ORG_NAME
> verify error:num=19:self signed certificate in certificate chain
> verify return:0
> ---
> Certificate chain
>  0 s:/C=XY/ST=Some State/O=My Large Organization Name/OU=My Subunit of
> Large
> Organization/CN=somename.somewhere.com/emailAddress=root at somename.somewhere.com
>
> i:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
>  1
> s:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
>
> i:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
> ---
> Server certificate
> -----BEGIN CERTIFICATE-----
> MIIC6TCCAdGgAwIBAgIBATANBgkqhkiG9w0BAQUFADBpMRIwEAYDVQQDFAlZb3Vy
> X05BTUUxEzARBgNVBAgUCllvdXJfU1RBVEUxCzAJBgNVBAYTAkNPMRkwFwYJKoZI
> hvcNAQkBFgpZT1VSX0VNQUlMMRYwFAYDVQQKFA1ZT1VSX09SR19OQU1FMB4XDTE4
> MDUwODEyMzcyM1oXDTE5MDUwODEyMzcyM1owgb8xCzAJBgNVBAYTAlhZMRMwEQYD
> VQQIEwpTb21lIFN0YXRlMSMwIQYDVQQKExpNeSBMYXJnZSBPcmdhbml6YXRpb24g
> TmFtZTEpMCcGA1UECxMgTXkgU3VidW5pdCBvZiBMYXJnZSBPcmdhbml6YXRpb24x
> HzAdBgNVBAMTFnNvbWVuYW1lLnNvbWV3aGVyZS5jb20xKjAoBgkqhkiG9w0BCQEW
> G3Jvb3RAc29tZW5hbWUuc29tZXdoZXJlLmNvbTBcMA0GCSqGSIb3DQEBAQUAA0sA
> MEgCQQDL7uikSc1kVIvw5rhyQzk2dSJcmJ6EJ1LSmtAoafZH8bqfZ25cDQZQGi05
> YcuxGR0vSaW7xPnyhaWCLQlxQFx7AgMBAAGjDTALMAkGA1UdEwQCMAAwDQYJKoZI
> hvcNAQEFBQADggEBAHv4WzGdYhoEyHZmBQTVjdEKOVBMnNoOqum79uzWtSzSjG4E
> pP/9c331uT7fBZ/Z7XNhIV+PbDZXorLgUhwwT7zxYURNnV52Of2SWRmWtPBrgEX1
> +8S0IMtJFfJta8FAfTTaNqLpRDaiTQs3em1Maxls15cTyRQzMIjIJnY4eRrh5CNM
> YV/+kg/lpKAe0awiMu96cxpnMdz9h33g7RedBnh9wDi6k7pfYtvlC6o4snZO01AN
> 8qRiQf54OPvKcVeseJFBPWLhdYns6g+/SXhq1Lek2us93ZpuKgIaBtzkyDm2+SFa
> QXF9f0a+UuEdPvrtvMjAijcDwcaXq0r2f2MA++M=
> -----END CERTIFICATE-----
> subject=/C=XY/ST=Some State/O=My Large Organization Name/OU=My Subunit of
> Large
> Organization/CN=somename.somewhere.com/emailAddress=root at somename.somewhere.com
>
> issuer=/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
> ---
> No client certificate CA names sent
> ---
> SSL handshake has read 1979 bytes and written 337 bytes
> ---
> New, TLSv1/SSLv3, Cipher is AES256-GCM-SHA384
> Server public key is 512 bit
> Secure Renegotiation IS NOT supported
> Compression: NONE
> Expansion: NONE
> SSL-Session:
>     Protocol  : TLSv1.2
>     Cipher    : AES256-GCM-SHA384
>     Session-ID:
> EA8B17008E58F1D04CD1CEA53103CF477AA9DE0DC80A4FF4F0DD4814031E4C15
>     Session-ID-ctx:
>     Master-Key:
> D28ED5C21D288944D2277AF86FE82A9BF3BEDABAA14DBCD5AE32B190EF0A0CA6AB99719E751E6DD4FECAA9DD1307A3C0
>     Key-Arg   : None
>     PSK identity: None
>     PSK identity hint: None
>     SRP username: None
>     TLS session ticket lifetime hint: 300 (seconds)
>     TLS session ticket:
>     0000 - 2f c5 82 ea bf 8b 66 49-bc bc ee 48 1a fb 8e 6c
> /.....fI...H...l
>     0010 - de 42 d9 e0 6e 36 40 78-06 cc 68 c6 74 6d 6e aa   .B..n6 at x.
> .h.tmn.
>     0020 - b6 53 8a ed b2 8d 5a c4-02 e1 88 8b d2 a9 56 5f
> .S....Z.......V_
>     0030 - ee c6 b9 14 55 da 37 df-8f aa af 81 b4 22 4e be
> ....U.7......"N.
>     0040 - 9c c5 87 d6 46 22 47 03-4a 88 dd 1e 9d 05 81 09
> ....F"G.J.......
>     0050 - c3 8b 9f 44 29 90 4d 93-c9 f5 41 e2 4d 72 1b de
> ...D).M...A.Mr..
>     0060 - 8d c2 15 ab 49 ad da 26-0e 72 a9 01 02 3e 89 33
> ....I..&.r...>.3
>     0070 - 6e 6c 2f 20 1c 15 06 7a-8d c5 a6 6e ee 46 d2 76   nl/
> ...z...n.F.v
>     0080 - 63 c1 89 1e 9b 3c a1 10-d0 78 31 9e e6 8e 86 ab
> c....<...x1.....
>     0090 - ff bc 3a 4c ab 3d 33 8f-e9 56 c5 f1 45 46 73 41
> ..:L.=3..V..EFsA
>
>     Start Time: 1528361487
>     Timeout   : 7200 (sec)
>     Verify return code: 19 (self signed certificate in certificate chain)
> ---
> closed
>
>
> In this case the command seems to have been successfully executed. I
> remark that the outbound TLS transactions seems to be working fine also
> from FS (SIP Registrar, SIP INVITE in outbound don't have any problem).
>
> If required I can provide also the FS configuration files (vars.xml,
> sofia.conf.xml,  etc etc).
>
> Any help will be greatly appreciated.
>
> Thanks in advance
>
> Best regards
>
>
> fabio
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/32da3504/attachment-0001.html>

From david.villasmil.work at gmail.com  Thu Jun  7 18:26:20 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Jun 2018 20:26:20 +0200
Subject: [Freeswitch-users] wiki and org sites
In-Reply-To: <E785B568-3F86-4B2D-90E3-7771E417FCD3@jerris.com>
References: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>
 <E785B568-3F86-4B2D-90E3-7771E417FCD3@jerris.com>
Message-ID: <CAFGRPVpOxHeVtE95CCqb3VmK51_H1N875a9EKRqQF1kgWbVxVA@mail.gmail.com>

Yeah, I remember that afterwards... it's just that LOTS of things are
pointing there...

Regards,

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

ᐧ

On Thu, Jun 7, 2018 at 8:25 PM Michael Jerris <mike at jerris.com> wrote:

> wiki.freeswitch.org has been deprecated for many years now.  The correct
> location for the wiki is https://freeswitch.org/confluence
>
> Mike
>
> On Jun 6, 2018, at 3:02 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
> I can't connect to wiki.freeswitch.org, is the web down? I just get
> redirected to freeswitch's (new?) commercial web page (freeswitch.com)
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/a18f4b99/attachment.html>

From david.villasmil.work at gmail.com  Thu Jun  7 18:30:59 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Jun 2018 20:30:59 +0200
Subject: [Freeswitch-users] Can't get release cause on dialplan
Message-ID: <CAFGRPVr_SM83QKsiB8bDX5Wi4-21Wkaz876kw-85U4z0AGq90A@mail.gmail.com>

Hello guys,

I've got the following extension:

<include>
  <extension name="public_did0">
    <condition field="destination_number" expression="^9999999$">
      <action application="set" data="failure_causes=USER_BUSY,NO_ANSWER"/>
      <action application="bridge"
data="{continue_on_fail=true,hangup_after_bridge=false}sofia/external/
9999999 at 1.2.3.4:5080 <http://9999999 at 54.84.149.30:5080>"/>
      <action application="log" data="CRIT Hangup cause was:
${bridge_hangup_cause}"/>
    </condition>
  </extension>
</include>

But it just shows:

2018-06-07 18:29:42.639898 [DEBUG] switch_ivr_originate.c:3848 Originate
Resulted in Error Cause: 65 [BEARERCAPABILITY_NOTIMPL]
2018-06-07 18:29:42.639898 [INFO] mod_dptools.c:3436 Originate Failed.
Cause: BEARERCAPABILITY_NOTIMPL
2018-06-07 18:29:42.639898 [DEBUG] switch_channel.c:4758 Failure causes
[USER_BUSY,NO_ANSWER]:  Cause: BEARERCAPABILITY_NOTIMPL
EXECUTE sofia/external/david.villasmil at test-fs02 log(CRIT Hangup cause was:
)
2018-06-07 18:29:42.639898 [CRIT] mod_dptools.c:1742 Hangup cause was:
2018-06-07 18:29:42.639898 [NOTICE]


Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
ᐧ
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/978df1f5/attachment.html>

From vma at vallimamod.org  Thu Jun  7 20:32:21 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Thu, 7 Jun 2018 22:32:21 +0200
Subject: [Freeswitch-users] Can't get release cause on dialplan
In-Reply-To: <CAFGRPVr_SM83QKsiB8bDX5Wi4-21Wkaz876kw-85U4z0AGq90A@mail.gmail.com>
References: <CAFGRPVr_SM83QKsiB8bDX5Wi4-21Wkaz876kw-85U4z0AGq90A@mail.gmail.com>
Message-ID: <307B58D6-6783-4B25-8A66-8B68A93575A0@vallimamod.org>

Hi,

I generally use ${originate_disposition} to get the leg B hangup cause.  Can you try it with your dialplan?


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


> On 7 Jun 2018, at 20:30, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> Hello guys,
> 
> I've got the following extension:
> 
> <include>
>  <extension name="public_did0">
>    <condition field="destination_number" expression="^9999999$">
>      <action application="set" data="failure_causes=USER_BUSY,NO_ANSWER"/>
>      <action application="bridge" data="{continue_on_fail=true,hangup_after_bridge=false}sofia/external/9999999 at 1.2.3.4:5080"/>
>      <action application="log" data="CRIT Hangup cause was: ${bridge_hangup_cause}"/>
>    </condition>
>  </extension>
> </include>
> 
> But it just shows:
> 
> 2018-06-07 18:29:42.639898 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 65 [BEARERCAPABILITY_NOTIMPL]
> 2018-06-07 18:29:42.639898 [INFO] mod_dptools.c:3436 Originate Failed.  Cause: BEARERCAPABILITY_NOTIMPL
> 2018-06-07 18:29:42.639898 [DEBUG] switch_channel.c:4758 Failure causes [USER_BUSY,NO_ANSWER]:  Cause: BEARERCAPABILITY_NOTIMPL
> EXECUTE sofia/external/david.villasmil at test-fs02 log(CRIT Hangup cause was: )
> 2018-06-07 18:29:42.639898 [CRIT] mod_dptools.c:1742 Hangup cause was:
> 2018-06-07 18:29:42.639898 [NOTICE]
> 
> 
> Regards,
> 
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> ᐧ
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



Cordialement,
-- 
Vallimamod Abdullah
SIP Solutions
Conseil & Développement de plateformes et services VOIP
vma at sip.solutions
+33 6 62 60 68 97
linkedin.com/in/vallimamod
.



From mike at jerris.com  Thu Jun  7 19:44:12 2018
From: mike at jerris.com (Michael Jerris)
Date: Thu, 7 Jun 2018 15:44:12 -0400
Subject: [Freeswitch-users] wiki and org sites
In-Reply-To: <CAFGRPVpOxHeVtE95CCqb3VmK51_H1N875a9EKRqQF1kgWbVxVA@mail.gmail.com>
References: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>
 <E785B568-3F86-4B2D-90E3-7771E417FCD3@jerris.com>
 <CAFGRPVpOxHeVtE95CCqb3VmK51_H1N875a9EKRqQF1kgWbVxVA@mail.gmail.com>
Message-ID: <A7460BF3-DE88-4C9F-B5A8-3033DCB7A3BC@jerris.com>

If there is anything still pointing there it needs to be fixed.  

> On Jun 7, 2018, at 2:26 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> Yeah, I remember that afterwards... it's just that LOTS of things are pointing there...
> 
> Regards,
> 
> David Villasmil
> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
> 
> ᐧ
> 
> On Thu, Jun 7, 2018 at 8:25 PM Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> wiki.freeswitch.org <http://wiki.freeswitch.org/> has been deprecated for many years now.  The correct location for the wiki is https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> 
> Mike
> 
>> On Jun 6, 2018, at 3:02 PM, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>> 
>> I can't connect to wiki.freeswitch.org <http://wiki.freeswitch.org/>, is the web down? I just get redirected to freeswitch's (new?) commercial web page (freeswitch.com <http://freeswitch.com/>)
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/fe4edaf8/attachment.html>

From f.antonini at tiesse.com  Fri Jun  8 05:44:16 2018
From: f.antonini at tiesse.com (Fabio Antonini)
Date: Fri, 08 Jun 2018 07:44:16 +0200
Subject: [Freeswitch-users] SIP over TLS configuration problem
In-Reply-To: <CAMtXxQ=FwqpKEUHOQZfrH3gHEKy1Dw4S0TZLJgXTjf6RDS-G-Q@mail.gmail.com>
Message-ID: <77b6eecf-22a3-4d40-ab95-25959cd73feb@email.android.com>

An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180608/370167be/attachment-0001.html>

From f.antonini at tiesse.com  Fri Jun  8 06:16:27 2018
From: f.antonini at tiesse.com (fabio)
Date: Fri, 8 Jun 2018 08:16:27 +0200
Subject: [Freeswitch-users] SIP over TLS configuration problem
In-Reply-To: <CAFGRPVr=X2XOhu8v2wqi4QjHK69sxqLgFAbU7r83QrukYkFmuw@mail.gmail.com>
References: <193f5f5a-bc26-c0d6-116f-97bc160cb654@tiesse.com>
 <CAFGRPVr=X2XOhu8v2wqi4QjHK69sxqLgFAbU7r83QrukYkFmuw@mail.gmail.com>
Message-ID: <847ca2f0-5220-22c4-060c-1cf0ae3d12e4@tiesse.com>

Hi David

thanks a lot for your feedback.


I have investigated more in depth and I found out that the problem was 
caused by the format of the agent.pem certificate generated by opensips.

I followed the following short tutorial to create the agent.pem certificate


htps://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/docs/how_to_make_your_own_ca_correctly.txt 
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/docs/how_to_make_your_own_ca_correctly.txt>t 
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/docs/how_to_make_your_own_ca_correctly.txt>


and now I can manage also the inbound calls.
I'm waiting for the customer's official certificates. Anyway at the 
moment everything is working fine.
In the meantime I'll move to 1.6.
Thanks
fabio

On 07/06/2018 19:52, David Villasmil wrote:
>
> If the certs are self-signed you're going to have many many problems. 
> I followed that tutorial with a valid cert and worked beautifully. 
> Also i did it with 1.6.
>
>
> On Thu, Jun 7, 2018, 18:40 fabio <f.antonini at tiesse.com 
> <mailto:f.antonini at tiesse.com>> wrote:
>
>     Hi all
>
>
>     I'm a Freeswitch newbie and I'm trying to setup SIP over TLS in my
>     FS version 1.5.15.
>
>     As first step I have configured a SIP Gateway that successfully
>     registers to a dedicated SIP Registrar/Proxy (opensips) using SIP
>     over UDP. With this configuration I can successfully place
>     outbound and inbound calls without any problem. Everything works
>     as a charm.
>
>     Further I have tried to switch to SIP over TLS and I followed the
>     steps described in
>     https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS.
>
>     I have installed the agent.pem and cafile.pem generated by
>     opensips (my SIP Registrar) and I configured FS to use them. After
>     restart the sofia gateway profile can successfully register to the
>     SIP Registrar by SIP over TLS.
>
>     Further I can successfully place outbound call (from internal
>     channel through the SIP gateway).  It sounds great!
>
>     Unfortunately FS fails to handle inbound calls (SIP INVITE from an
>     external SIP UA registered to the same SIP Registrar to the SIP UA
>     extension of the FS SIP gateway).
>
>     I have tried to trace all the logs I can. Here below some traces
>     from the FS console when an inbound INVITE is received:
>
>
>     tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb7f28): events IN
>     tport.c:862 tport_alloc_secondary()
>     tport_alloc_secondary(0xb7f28): new secondary tport 0x1398c0
>     tport_type_tcp.c:203 tport_tcp_init_secondary()
>     tport_tcp_init_secondary(0x1398c0): Setting TCP_KEEPIDLE to 30
>     tport_type_tcp.c:209 tport_tcp_init_secondary()
>     tport_tcp_init_secondary(0x1398c0): Setting TCP_KEEPINTVL to 30
>     tport_type_tls.c:610 tport_tls_accept()
>     tport_tls_accept(0x1398c0): new connection from
>     tls/10.3.10.110:38632/sips <http://10.3.10.110:38632/sips>
>     tport_tls.c:919 tls_connect() tls_connect(0x1398c0): events
>     NEGOTIATING
>     tport_tls.c:1008 tls_connect() tls_connect(0x1398c0): TLS setup
>     failed (error:00000001:lib(0):func(0):reason(1))
>     tport.c:2090 tport_close() tport_close(0x1398c0):
>     tls/10.3.10.110:38632/sips <http://10.3.10.110:38632/sips>
>     tport.c:2263 tport_set_secondary_timer() tport(0x1398c0): set
>     timer at 0 ms because zap
>
>
>     In order to simplify the test I have also tried to connect to the
>     5061 TLS port by a simple openssl command from a linux shell of
>     the SIP Registrar box:
>
>
>     openssl s_client -connect 10.11.4.103:5061
>     <http://10.11.4.103:5061> -tls1_2
>     CONNECTED(00000003)
>     3074304200:error:14094410:SSL routines:SSL3_READ_BYTES:sslv3 alert
>     handshake failure:s3_pkt.c:1256:SSL alert number 40
>     3074304200:error:1409E0E5:SSL routines:SSL3_WRITE_BYTES:ssl
>     handshake failure:s3_pkt.c:596:
>     ---
>     no peer certificate available
>     ---
>     No client certificate CA names sent
>     ---
>     SSL handshake has read 7 bytes and written 0 bytes
>     ---
>     New, (NONE), Cipher is (NONE)
>     Secure Renegotiation IS NOT supported
>     Compression: NONE
>     Expansion: NONE
>     SSL-Session:
>         Protocol  : TLSv1.2
>         Cipher    : 0000
>         Session-ID:
>         Session-ID-ctx:
>         Master-Key:
>         Key-Arg   : None
>         PSK identity: None
>         PSK identity hint: None
>         SRP username: None
>         Start Time: 1528361426
>         Timeout   : 7200 (sec)
>         Verify return code: 0 (ok)
>     ---
>
>     In the FS console I read the same traces received in the previous
>     test with the inbound call.
>
>
>     tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb7f28): events IN
>     tport.c:862 tport_alloc_secondary()
>     tport_alloc_secondary(0xb7f28): new secondary tport 0x248210
>     tport_type_tcp.c:203 tport_tcp_init_secondary()
>     tport_tcp_init_secondary(0x248210): Setting TCP_KEEPIDLE to 30
>     tport_type_tcp.c:209 tport_tcp_init_secondary()
>     tport_tcp_init_secondary(0x248210): Setting TCP_KEEPINTVL to 30
>     tport_type_tls.c:610 tport_tls_accept()
>     tport_tls_accept(0x248210): new connection from
>     tls/10.11.4.103:33168/sips <http://10.11.4.103:33168/sips>
>     tport_tls.c:919 tls_connect() tls_connect(0x248210): events
>     NEGOTIATING
>     tport_tls.c:1008 tls_connect() tls_connect(0x248210): TLS setup
>     failed (error:00000001:lib(0):func(0):reason(1))
>     tport.c:2090 tport_close() tport_close(0x248210):
>     tls/10.11.4.103:33168/sips <http://10.11.4.103:33168/sips>
>     tport.c:2263 tport_set_secondary_timer() tport(0x248210): set
>     timer at 0 ms because zap
>
>     I have attached also a wireshark capture of the inbound call. In
>     this capture the SIP Registrar has IP 10.3.10.110. The FS device
>     is 10.11.4.103. The Client Hello is sent by the SIP Registrar, but
>     the FS device replies with an "Alert: Level: fatal, Description:
>     handshake failure (40).
>
>     I guess that there is some misconfiguration related to the TLS
>     version or proposed ciphers  or any certifcates but I cannot
>     understand what.
>
>
>     For comparison I have tried to run the same openssl command from
>     FS to the external SIP Registrar (outbound).
>
>
>     openssl s_client -connect 10.3.10.110:5061
>     <http://10.3.10.110:5061> -tls1_2
>     CONNECTED(00000003)
>     depth=1 CN = Your_NAME, ST = Your_STATE, C = CO, emailAddress =
>     YOUR_EMAIL, O = YOUR_ORG_NAME
>     verify error:num=19:self signed certificate in certificate chain
>     verify return:0
>     ---
>     Certificate chain
>      0 s:/C=XY/ST=Some State/O=My Large Organization Name/OU=My
>     Subunit of Large
>     Organization/CN=somename.somewhere.com/emailAddress=root at somename.somewhere.com
>     <mailto:Organization/CN=somename.somewhere.com/emailAddress=root at somename.somewhere.com>
>     i:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
>      1
>     s:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
>     i:/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
>     ---
>     Server certificate
>     -----BEGIN CERTIFICATE-----
>     MIIC6TCCAdGgAwIBAgIBATANBgkqhkiG9w0BAQUFADBpMRIwEAYDVQQDFAlZb3Vy
>     X05BTUUxEzARBgNVBAgUCllvdXJfU1RBVEUxCzAJBgNVBAYTAkNPMRkwFwYJKoZI
>     hvcNAQkBFgpZT1VSX0VNQUlMMRYwFAYDVQQKFA1ZT1VSX09SR19OQU1FMB4XDTE4
>     MDUwODEyMzcyM1oXDTE5MDUwODEyMzcyM1owgb8xCzAJBgNVBAYTAlhZMRMwEQYD
>     VQQIEwpTb21lIFN0YXRlMSMwIQYDVQQKExpNeSBMYXJnZSBPcmdhbml6YXRpb24g
>     TmFtZTEpMCcGA1UECxMgTXkgU3VidW5pdCBvZiBMYXJnZSBPcmdhbml6YXRpb24x
>     HzAdBgNVBAMTFnNvbWVuYW1lLnNvbWV3aGVyZS5jb20xKjAoBgkqhkiG9w0BCQEW
>     G3Jvb3RAc29tZW5hbWUuc29tZXdoZXJlLmNvbTBcMA0GCSqGSIb3DQEBAQUAA0sA
>     MEgCQQDL7uikSc1kVIvw5rhyQzk2dSJcmJ6EJ1LSmtAoafZH8bqfZ25cDQZQGi05
>     YcuxGR0vSaW7xPnyhaWCLQlxQFx7AgMBAAGjDTALMAkGA1UdEwQCMAAwDQYJKoZI
>     hvcNAQEFBQADggEBAHv4WzGdYhoEyHZmBQTVjdEKOVBMnNoOqum79uzWtSzSjG4E
>     pP/9c331uT7fBZ/Z7XNhIV+PbDZXorLgUhwwT7zxYURNnV52Of2SWRmWtPBrgEX1
>     +8S0IMtJFfJta8FAfTTaNqLpRDaiTQs3em1Maxls15cTyRQzMIjIJnY4eRrh5CNM
>     YV/+kg/lpKAe0awiMu96cxpnMdz9h33g7RedBnh9wDi6k7pfYtvlC6o4snZO01AN
>     8qRiQf54OPvKcVeseJFBPWLhdYns6g+/SXhq1Lek2us93ZpuKgIaBtzkyDm2+SFa
>     QXF9f0a+UuEdPvrtvMjAijcDwcaXq0r2f2MA++M=
>     -----END CERTIFICATE-----
>     subject=/C=XY/ST=Some State/O=My Large Organization Name/OU=My
>     Subunit of Large
>     Organization/CN=somename.somewhere.com/emailAddress=root at somename.somewhere.com
>     <mailto:Organization/CN=somename.somewhere.com/emailAddress=root at somename.somewhere.com>
>     issuer=/CN=Your_NAME/ST=Your_STATE/C=CO/emailAddress=YOUR_EMAIL/O=YOUR_ORG_NAME
>     ---
>     No client certificate CA names sent
>     ---
>     SSL handshake has read 1979 bytes and written 337 bytes
>     ---
>     New, TLSv1/SSLv3, Cipher is AES256-GCM-SHA384
>     Server public key is 512 bit
>     Secure Renegotiation IS NOT supported
>     Compression: NONE
>     Expansion: NONE
>     SSL-Session:
>         Protocol  : TLSv1.2
>         Cipher    : AES256-GCM-SHA384
>         Session-ID:
>     EA8B17008E58F1D04CD1CEA53103CF477AA9DE0DC80A4FF4F0DD4814031E4C15
>         Session-ID-ctx:
>         Master-Key:
>     D28ED5C21D288944D2277AF86FE82A9BF3BEDABAA14DBCD5AE32B190EF0A0CA6AB99719E751E6DD4FECAA9DD1307A3C0
>         Key-Arg   : None
>         PSK identity: None
>         PSK identity hint: None
>         SRP username: None
>         TLS session ticket lifetime hint: 300 (seconds)
>         TLS session ticket:
>         0000 - 2f c5 82 ea bf 8b 66 49-bc bc ee 48 1a fb 8e 6c  
>     /.....fI...H...l
>         0010 - de 42 d9 e0 6e 36 40 78-06 cc 68 c6 74 6d 6e aa  
>     .B..n6 at x..h.tmn.
>         0020 - b6 53 8a ed b2 8d 5a c4-02 e1 88 8b d2 a9 56 5f  
>     .S....Z.......V_
>         0030 - ee c6 b9 14 55 da 37 df-8f aa af 81 b4 22 4e be  
>     ....U.7......"N.
>         0040 - 9c c5 87 d6 46 22 47 03-4a 88 dd 1e 9d 05 81 09  
>     ....F"G.J.......
>         0050 - c3 8b 9f 44 29 90 4d 93-c9 f5 41 e2 4d 72 1b de  
>     ...D).M...A.Mr..
>         0060 - 8d c2 15 ab 49 ad da 26-0e 72 a9 01 02 3e 89 33  
>     ....I..&.r...>.3
>         0070 - 6e 6c 2f 20 1c 15 06 7a-8d c5 a6 6e ee 46 d2 76   nl/
>     ...z...n.F.v
>         0080 - 63 c1 89 1e 9b 3c a1 10-d0 78 31 9e e6 8e 86 ab  
>     c....<...x1.....
>         0090 - ff bc 3a 4c ab 3d 33 8f-e9 56 c5 f1 45 46 73 41  
>     ..:L.=3..V..EFsA
>
>         Start Time: 1528361487
>         Timeout   : 7200 (sec)
>         Verify return code: 19 (self signed certificate in certificate
>     chain)
>     ---
>     closed
>
>
>     In this case the command seems to have been successfully executed.
>     I remark that the outbound TLS transactions seems to be working
>     fine also from FS (SIP Registrar, SIP INVITE in outbound don't
>     have any problem).
>
>     If required I can provide also the FS configuration files
>     (vars.xml, sofia.conf.xml,  etc etc).
>
>     Any help will be greatly appreciated.
>
>     Thanks in advance
>
>     Best regards
>
>
>     fabio
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Services
>     sales at freeswitch.com <mailto:sales at freeswitch.com>
>     https://freeswitch.com
>
>     Official FreeSWITCH Sites
>     https://freeswitch.com/oss
>     https://freeswitch.org/confluence
>     https://cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     https://freeswitch.com
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 

Fabio Antonini

/Software Engineer (Ph.D)/
f.antonini at tiesse.com <mailto:f.antonini at tiesse.com>
*Tel* +39.0863.455830
*Mob* +39.393.9261941
*Fax* +39.0863.455830
Via Corradini 80
67051 Avezzano (AQ)

	

Logo Tiesse dal 1998 al 2018, vent'anni di Innovazione Made in Italy. 
Clicca per visitare il sito Tiesse <http://www.tiesse.com/>

*Tiesse S.p.A.* - www.tiesse.com <http://www.tiesse.com/>
Via Asti 4, 10015 Ivrea (TO)
Pagina Tiesse su Linkedin, clicca e visitaci 
<https://linkedin.com/company/tiesse-spa>

*Disclaimer:* il contenuto di questa email è riservato e non vincolante 
per Tiesse S.p.A.. Se lo avesse ricevuto per errore, la preghiamo di 
segnalarlo immediatamente al mittente, di non utilizzare e divulgare il 
contenuto e di distruggere ogni copia in suo possesso. Tiesse S.p.A. 
declina ogni responsabilità da qualsiasi conseguenza derivante da 
utilizzi non autorizzati, contraffazioni o manomissioni di email recanti 
riferimenti all'azienda.
*Rispetta l'ambiente. Non stampare questa mail se non è necessario.*

	

	

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180608/2d7be35f/attachment-0001.html>

From david.villasmil.work at gmail.com  Thu Jun  7 19:58:11 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Jun 2018 21:58:11 +0200
Subject: [Freeswitch-users] wiki and org sites
In-Reply-To: <CAFGRPVpOxHeVtE95CCqb3VmK51_H1N875a9EKRqQF1kgWbVxVA@mail.gmail.com>
References: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>
 <E785B568-3F86-4B2D-90E3-7771E417FCD3@jerris.com>
 <CAFGRPVpOxHeVtE95CCqb3VmK51_H1N875a9EKRqQF1kgWbVxVA@mail.gmail.com>
Message-ID: <CAFGRPVr8kVwMdC8wALCEY+KM6PQ6uUGxuguw2HcFCNWu-KUwXA@mail.gmail.com>

BTW, I'll be at ClueCon this year, so see you guys in August!

Regards,

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

ᐧ

On Thu, Jun 7, 2018 at 8:26 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Yeah, I remember that afterwards... it's just that LOTS of things are
> pointing there...
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
> ᐧ
>
> On Thu, Jun 7, 2018 at 8:25 PM Michael Jerris <mike at jerris.com> wrote:
>
>> wiki.freeswitch.org has been deprecated for many years now.  The correct
>> location for the wiki is https://freeswitch.org/confluence
>>
>> Mike
>>
>> On Jun 6, 2018, at 3:02 PM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>> I can't connect to wiki.freeswitch.org, is the web down? I just get
>> redirected to freeswitch's (new?) commercial web page (freeswitch.com)
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/5d7ffbd3/attachment-0001.html>

From david.villasmil.work at gmail.com  Thu Jun  7 20:56:31 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Jun 2018 22:56:31 +0200
Subject: [Freeswitch-users] Can't get release cause on dialplan
In-Reply-To: <CAFGRPVr_SM83QKsiB8bDX5Wi4-21Wkaz876kw-85U4z0AGq90A@mail.gmail.com>
References: <CAFGRPVr_SM83QKsiB8bDX5Wi4-21Wkaz876kw-85U4z0AGq90A@mail.gmail.com>
Message-ID: <CAFGRPVrTahL2j=M6E23b0uFkHNnXQkastuMeRUyYq_YXx+a12w@mail.gmail.com>

Ok, it seems setting the failure_causes was messing up my dialplan.

Regards,

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

ᐧ

On Thu, Jun 7, 2018 at 8:30 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello guys,
>
> I've got the following extension:
>
> <include>
>   <extension name="public_did0">
>     <condition field="destination_number" expression="^9999999$">
>       <action application="set" data="failure_causes=USER_BUSY,NO_ANSWER"/>
>       <action application="bridge"
> data="{continue_on_fail=true,hangup_after_bridge=false}sofia/external/
> 9999999 at 1.2.3.4:5080 <http://9999999 at 54.84.149.30:5080>"/>
>       <action application="log" data="CRIT Hangup cause was:
> ${bridge_hangup_cause}"/>
>     </condition>
>   </extension>
> </include>
>
> But it just shows:
>
> 2018-06-07 18:29:42.639898 [DEBUG] switch_ivr_originate.c:3848 Originate
> Resulted in Error Cause: 65 [BEARERCAPABILITY_NOTIMPL]
> 2018-06-07 18:29:42.639898 [INFO] mod_dptools.c:3436 Originate Failed.
> Cause: BEARERCAPABILITY_NOTIMPL
> 2018-06-07 18:29:42.639898 [DEBUG] switch_channel.c:4758 Failure causes
> [USER_BUSY,NO_ANSWER]:  Cause: BEARERCAPABILITY_NOTIMPL
> EXECUTE sofia/external/david.villasmil at test-fs02 log(CRIT Hangup cause
> was: )
> 2018-06-07 18:29:42.639898 [CRIT] mod_dptools.c:1742 Hangup cause was:
> 2018-06-07 18:29:42.639898 [NOTICE]
>
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> ᐧ
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/1e4b2cd3/attachment-0001.html>

From david.villasmil.work at gmail.com  Thu Jun  7 21:11:07 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 7 Jun 2018 23:11:07 +0200
Subject: [Freeswitch-users] Wrong release cause text, bug?
Message-ID: <CAFGRPVrJ1tw8V4Q9o7wmLjpnLp+Be01fW-sAue=4mHyHhf_WRg@mail.gmail.com>

Hello guys,

When hanging up a call with a specific release, FS sends the specific code,
i.e.:

if i release a call like so:

<action inline="true" application="set" data="release_code=483
EXCHANGE_ROUTING_ERROR"/>

The BYE message is as follows:

   SIP/2.0 483 EXCHANGE_ROUTING_ERROR
   Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bK0y1H2X8899veB
   Max-Forwards: 69
   From: "david.villasmil" <sip:david.villasmil at 1.2.3.4>;tag=t35r9gatZFcjj
   To: <sip:9999999 at 1.2.3.4:5080>;tag=pB7mFpUrc30gB
   Call-ID: e2da7a79-e537-1236-5691-02e1e654f872
   CSeq: 123866386 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
   Content-Length: 0
   X-FS-Display-Name: 9999999
   X-FS-Display-Number: sip:9999999 at 1.2.3.4
   Remote-Party-ID: "9999999" <sip:9999999 at 1.2.3.4
>;party=calling;privacy=off;screen=no

Another example:

<action inline="true" application="set" data="release_code=410
NUMBER_CHANGED"/>

The BYE message is SOMETIMES as follows:

   SIP/2.0 410 NUMBER_CHANGED
   Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKZN8r02Q5c16UF
   Max-Forwards: 69
   From: "david.villasmil" <sip:david.villasmil at 1.2.3.4>;tag=Q8Se4ZQF8m9SF
   To: <sip:9999999 at 1.2.3.4:5080>;tag=mSm3B0SHjHmBm
   Call-ID: dfbb5497-e537-1236-5691-02e1e654f872
   CSeq: 123866383 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Reason: Q.850;cause=22;text="NUMBER_CHANGED"
   Content-Length: 0
   X-FS-Display-Name: 9999999
   X-FS-Display-Number: sip:9999999 at 1.2.3.4
   Remote-Party-ID: "9999999" <sip:9999999 at 54.84.149.30
>;party=calling;privacy=off;screen=no

And other times as follows:

   SIP/2.0 410 Gone
   Via: SIP/2.0/UDP 1.2.3.4:32200
;branch=z9hG4bK-d8754z-86e4295fca2efa0a-1---d8754z-;rport=32200
   Max-Forwards: 69
   From: <sip:david.villasmil at 1.2.3.4;transport=UDP>;tag=42169b65
   To: <sip:9999999 at 1.2.3.4;transport=UDP>;tag=pZ0N246BBcK7K
   Call-ID: YmU5YjEwODk2ZmZhODJjNDgzNTNlM2FiZWVmMjI2MDg.
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Reason: Q.850;cause=16;text="NORMAL_CLEARING"
   Content-Length: 0
   Remote-Party-ID: "9999999" <sip:9999999 at 1.2.3.4
>;party=calling;privacy=off;screen=no


But I've noticed that when i do:

<action inline="true" application="set" data="release_code=488
BEARERCAPABILITY_NOTIMPL"/>

The BYE message is as follows:

   SIP/2.0 488 BEARERCAPABILITY_NOTIMPL
   Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bK3SDv7eUK14Z6D
   Max-Forwards: 69
   From: "david.villasmil" <sip:david.villasmil at 1.2.3.4>;tag=F3am0c2594H7a
   To: <sip:9999999 at 1.2.3.4:5080>;tag=6XDmXaZj5BQ8S
   Call-ID: aeb6624e-e539-1236-5691-02e1e654f872
   CSeq: 123866772 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, refer
   Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
   Content-Length: 0
   X-FS-Display-Name: 9999999
   X-FS-Display-Number: sip:9999999 at 1.2.3.4
   Remote-Party-ID: "9999999" <sip:9999999 at 54.84.149.30
>;party=calling;privacy=off;screen=no


So I guess, my question is why sometimes the "text" is set properly, and
sometimes not??


Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
ᐧ
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180607/d6040bf1/attachment-0001.html>

From gmina at connectfirst.com  Fri Jun  8 17:04:38 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Fri, 8 Jun 2018 11:04:38 -0600
Subject: [Freeswitch-users] Hung Sofia Profile
Message-ID: <CAKzROs3Um7Vae47zRQLGMYWLKyw7jL9fuHm+TSABf8X4bmc6VQ@mail.gmail.com>

Greetings,
Has anyone seen an instance where freeswitch gets hung with the following
stack trace?  I have about 400 threads in this state and eventually FS
(1.6.19) just stops responding.

I am running FS against MySQL 5.6.38 on CentOS using ODBC.   Guessing if
this is known, it probably isn't fixed in 1.6.20... but I can hope.  Any
input on what a possible solution would be greatly appreciated.  The MySQL
server and FS are running on the same host connecting via localhost
interface, so it's not a networking issue.

Thread 445 (Thread 0x7fa2df6db700 (LWP 9395)):
#0  0x00007fa3712da42d in __lll_lock_wait () from /usr/lib64/libpthread.so.0
#1  0x00007fa3712d5de6 in _L_lock_870 () from /usr/lib64/libpthread.so.0
#2  0x00007fa3712d5cdf in pthread_mutex_lock () from
/usr/lib64/libpthread.so.0
#3  0x00007fa35742468a in sofia_glue_execute_sql_callback () from
/usr/lib64/freeswitch/mod/mod_sofia.so
#4  0x00007fa3573df231 in select_from_profile () from
/usr/lib64/freeswitch/mod/mod_sofia.so
#5  0x00007fa3573e3b17 in sofia_contact_function () from
/usr/lib64/freeswitch/mod/mod_sofia.so
#6  0x00007fa3730594cc in switch_api_execute () from
/usr/lib64/libfreeswitch.so.1
#7  0x00007fa372fe69d6 in switch_channel_expand_variables_check () from
/usr/lib64/libfreeswitch.so.1
#8  0x00007fa3552d45e4 in user_outgoing_channel () from
/usr/lib64/freeswitch/mod/mod_dptools.so
#9  0x00007fa37300b42a in switch_core_session_outgoing_channel () from
/usr/lib64/libfreeswitch.so.1
#10 0x00007fa37308dc3b in switch_ivr_originate () from
/usr/lib64/libfreeswitch.so.1
#11 0x00007fa3552db0b5 in audio_bridge_function () from
/usr/lib64/freeswitch/mod/mod_dptools.so
#12 0x00007fa37300ebcb in switch_core_session_exec () from
/usr/lib64/libfreeswitch.so.1
#13 0x00007fa37300f179 in switch_core_session_execute_application_get_flags
() from /usr/lib64/libfreeswitch.so.1
#14 0x00007fa373012a34 in switch_core_session_run () from
/usr/lib64/libfreeswitch.so.1
#15 0x00007fa37300c33e in switch_core_session_thread () from
/usr/lib64/libfreeswitch.so.1
#16 0x00007fa373007d83 in switch_core_session_thread_pool_worker () from
/usr/lib64/libfreeswitch.so.1
#17 0x00007fa3732ce7c0 in dummy_worker () from /usr/lib64/libfreeswitch.so.1
#18 0x00007fa3712d3e25 in start_thread () from /usr/lib64/libpthread.so.0
#19 0x00007fa37092e34d in clone () from /usr/lib64/libc.so.6
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180608/af348857/attachment.html>

From mario_fs at mgtech.com  Fri Jun  8 16:01:13 2018
From: mario_fs at mgtech.com (Mario)
Date: Fri, 8 Jun 2018 09:01:13 -0700
Subject: [Freeswitch-users] When 1.6.21 ?
Message-ID: <A34EB9C4-2177-4757-B5A7-8B8C9ECD900E@mgtech.com>

Can anyone answer when 1.6.21 will be available? Just wondering since I haven’t been able to test FS production for macOS wiki since PCRE was updated (FS-11061). Also wanted to start testing on macOS Mojave to see how that goes. Thanks,
Mario G

From mike at jerris.com  Fri Jun  8 18:21:57 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 Jun 2018 14:21:57 -0400
Subject: [Freeswitch-users] Hung Sofia Profile
In-Reply-To: <CAKzROs3Um7Vae47zRQLGMYWLKyw7jL9fuHm+TSABf8X4bmc6VQ@mail.gmail.com>
References: <CAKzROs3Um7Vae47zRQLGMYWLKyw7jL9fuHm+TSABf8X4bmc6VQ@mail.gmail.com>
Message-ID: <3FAF7D44-1108-4833-8EF9-DAF4201288A4@jerris.com>

look for the other thread inside sofia_glue_execute_sql_callback that is NOT blocked on pthread_mutex_lock for your answer.  My guess is this is a deadlock inside the mysql drivers blocking us.  Mysql drivers have a long long history of thread safety issues.


> On Jun 8, 2018, at 1:04 PM, Geoff Mina <gmina at connectfirst.com> wrote:
> 
> Greetings,
> Has anyone seen an instance where freeswitch gets hung with the following stack trace?  I have about 400 threads in this state and eventually FS (1.6.19) just stops responding.  
> 
> I am running FS against MySQL 5.6.38 on CentOS using ODBC.   Guessing if this is known, it probably isn't fixed in 1.6.20... but I can hope.  Any input on what a possible solution would be greatly appreciated.  The MySQL server and FS are running on the same host connecting via localhost interface, so it's not a networking issue.
> 
> Thread 445 (Thread 0x7fa2df6db700 (LWP 9395)):
> #0  0x00007fa3712da42d in __lll_lock_wait () from /usr/lib64/libpthread.so.0
> #1  0x00007fa3712d5de6 in _L_lock_870 () from /usr/lib64/libpthread.so.0
> #2  0x00007fa3712d5cdf in pthread_mutex_lock () from /usr/lib64/libpthread.so.0
> #3  0x00007fa35742468a in sofia_glue_execute_sql_callback () from /usr/lib64/freeswitch/mod/mod_sofia.so
> #4  0x00007fa3573df231 in select_from_profile () from /usr/lib64/freeswitch/mod/mod_sofia.so
> #5  0x00007fa3573e3b17 in sofia_contact_function () from /usr/lib64/freeswitch/mod/mod_sofia.so
> #6  0x00007fa3730594cc in switch_api_execute () from /usr/lib64/libfreeswitch.so.1
> #7  0x00007fa372fe69d6 in switch_channel_expand_variables_check () from /usr/lib64/libfreeswitch.so.1
> #8  0x00007fa3552d45e4 in user_outgoing_channel () from /usr/lib64/freeswitch/mod/mod_dptools.so
> #9  0x00007fa37300b42a in switch_core_session_outgoing_channel () from /usr/lib64/libfreeswitch.so.1
> #10 0x00007fa37308dc3b in switch_ivr_originate () from /usr/lib64/libfreeswitch.so.1
> #11 0x00007fa3552db0b5 in audio_bridge_function () from /usr/lib64/freeswitch/mod/mod_dptools.so
> #12 0x00007fa37300ebcb in switch_core_session_exec () from /usr/lib64/libfreeswitch.so.1
> #13 0x00007fa37300f179 in switch_core_session_execute_application_get_flags () from /usr/lib64/libfreeswitch.so.1
> #14 0x00007fa373012a34 in switch_core_session_run () from /usr/lib64/libfreeswitch.so.1
> #15 0x00007fa37300c33e in switch_core_session_thread () from /usr/lib64/libfreeswitch.so.1
> #16 0x00007fa373007d83 in switch_core_session_thread_pool_worker () from /usr/lib64/libfreeswitch.so.1
> #17 0x00007fa3732ce7c0 in dummy_worker () from /usr/lib64/libfreeswitch.so.1
> #18 0x00007fa3712d3e25 in start_thread () from /usr/lib64/libpthread.so.0
> #19 0x00007fa37092e34d in clone () from /usr/lib64/libc.so.6
> 



From mike at jerris.com  Fri Jun  8 18:23:08 2018
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 Jun 2018 14:23:08 -0400
Subject: [Freeswitch-users] When 1.6.21 ?
In-Reply-To: <A34EB9C4-2177-4757-B5A7-8B8C9ECD900E@mgtech.com>
References: <A34EB9C4-2177-4757-B5A7-8B8C9ECD900E@mgtech.com>
Message-ID: <CFFFCB20-E42D-4525-A0E1-DC11C533BE88@jerris.com>

I plan to put a call for suggestions of what to back port into 1.6.21 in the next few weeks, then it will be just based on my availability to review those and see what is practical to put in.

> On Jun 8, 2018, at 12:01 PM, Mario <mario_fs at mgtech.com> wrote:
> 
> Can anyone answer when 1.6.21 will be available? Just wondering since I haven’t been able to test FS production for macOS wiki since PCRE was updated (FS-11061). Also wanted to start testing on macOS Mojave to see how that goes. Thanks,
> Mario G



From prestonh at gmail.com  Fri Jun  8 18:42:07 2018
From: prestonh at gmail.com (Preston Hagar)
Date: Fri, 8 Jun 2018 13:42:07 -0500
Subject: [Freeswitch-users] wiki and org sites
In-Reply-To: <A7460BF3-DE88-4C9F-B5A8-3033DCB7A3BC@jerris.com>
References: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>
 <E785B568-3F86-4B2D-90E3-7771E417FCD3@jerris.com>
 <CAFGRPVpOxHeVtE95CCqb3VmK51_H1N875a9EKRqQF1kgWbVxVA@mail.gmail.com>
 <A7460BF3-DE88-4C9F-B5A8-3033DCB7A3BC@jerris.com>
Message-ID: <CAK6zN=0C8OFK_35fwVjLMcWyqQWP1vJrDW3i_uZesrTDvS1UGA@mail.gmail.com>

A lot of the internal confluence links point back to the wiki.  Here are
some pages I've run across that have a lot of links back to the wiki.

Freeswitch IVR Originate at bottom of page here

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

Almost all the links on the page here:

https://freeswitch.org/confluence/display/FREESWITCH/mod_v8

and here:

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

https://freeswitch.org/confluence/display/FREESWITCH/JavaScript+Event

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

https://freeswitch.org/confluence/display/FREESWITCH/Audio+Codecs

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

https://freeswitch.org/confluence/display/FREESWITCH/Early+Media

On Thu, Jun 7, 2018 at 2:44 PM, Michael Jerris <mike at jerris.com> wrote:

> If there is anything still pointing there it needs to be fixed.
>
> On Jun 7, 2018, at 2:26 PM, David Villasmil <david.villasmil.work at gmail.
> com> wrote:
>
> Yeah, I remember that afterwards... it's just that LOTS of things are
> pointing there...
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
> ᐧ
>
> On Thu, Jun 7, 2018 at 8:25 PM Michael Jerris <mike at jerris.com> wrote:
>
>> wiki.freeswitch.org has been deprecated for many years now.  The correct
>> location for the wiki is https://freeswitch.org/confluence
>>
>> Mike
>>
>> On Jun 6, 2018, at 3:02 PM, David Villasmil <david.villasmil.work at gmail.
>> com> wrote:
>>
>> I can't connect to wiki.freeswitch.org, is the web down? I just get
>> redirected to freeswitch's (new?) commercial web page (freeswitch.com)
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180608/dd3a9c49/attachment-0001.html>

From anthony.minessale at gmail.com  Fri Jun  8 19:58:16 2018
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jun 2018 14:58:16 -0500
Subject: [Freeswitch-users] Hung Sofia Profile
In-Reply-To: <CAKzROs3Um7Vae47zRQLGMYWLKyw7jL9fuHm+TSABf8X4bmc6VQ@mail.gmail.com>
References: <CAKzROs3Um7Vae47zRQLGMYWLKyw7jL9fuHm+TSABf8X4bmc6VQ@mail.gmail.com>
Message-ID: <CAKbxfG93ZihgjXs3k-riO5SQc0Sx+H6WF-On4C47PW516fJ9+A@mail.gmail.com>

You would need to do a thread apply all bt to see what the other threads
are doing.  The thread you posted is stuck on a mutex waiting for something
else, probably your DB in another thread.
If you do get that info file it to jira because its easier to deal with
backtraces there.


On Fri, Jun 8, 2018 at 12:04 PM, Geoff Mina <gmina at connectfirst.com> wrote:

> Greetings,
> Has anyone seen an instance where freeswitch gets hung with the following
> stack trace?  I have about 400 threads in this state and eventually FS
> (1.6.19) just stops responding.
>
> I am running FS against MySQL 5.6.38 on CentOS using ODBC.   Guessing if
> this is known, it probably isn't fixed in 1.6.20... but I can hope.  Any
> input on what a possible solution would be greatly appreciated.  The MySQL
> server and FS are running on the same host connecting via localhost
> interface, so it's not a networking issue.
>
> Thread 445 (Thread 0x7fa2df6db700 (LWP 9395)):
> #0  0x00007fa3712da42d in __lll_lock_wait () from
> /usr/lib64/libpthread.so.0
> #1  0x00007fa3712d5de6 in _L_lock_870 () from /usr/lib64/libpthread.so.0
> #2  0x00007fa3712d5cdf in pthread_mutex_lock () from
> /usr/lib64/libpthread.so.0
> #3  0x00007fa35742468a in sofia_glue_execute_sql_callback () from
> /usr/lib64/freeswitch/mod/mod_sofia.so
> #4  0x00007fa3573df231 in select_from_profile () from
> /usr/lib64/freeswitch/mod/mod_sofia.so
> #5  0x00007fa3573e3b17 in sofia_contact_function () from
> /usr/lib64/freeswitch/mod/mod_sofia.so
> #6  0x00007fa3730594cc in switch_api_execute () from
> /usr/lib64/libfreeswitch.so.1
> #7  0x00007fa372fe69d6 in switch_channel_expand_variables_check () from
> /usr/lib64/libfreeswitch.so.1
> #8  0x00007fa3552d45e4 in user_outgoing_channel () from
> /usr/lib64/freeswitch/mod/mod_dptools.so
> #9  0x00007fa37300b42a in switch_core_session_outgoing_channel () from
> /usr/lib64/libfreeswitch.so.1
> #10 0x00007fa37308dc3b in switch_ivr_originate () from
> /usr/lib64/libfreeswitch.so.1
> #11 0x00007fa3552db0b5 in audio_bridge_function () from
> /usr/lib64/freeswitch/mod/mod_dptools.so
> #12 0x00007fa37300ebcb in switch_core_session_exec () from
> /usr/lib64/libfreeswitch.so.1
> #13 0x00007fa37300f179 in switch_core_session_execute_application_get_flags
> () from /usr/lib64/libfreeswitch.so.1
> #14 0x00007fa373012a34 in switch_core_session_run () from
> /usr/lib64/libfreeswitch.so.1
> #15 0x00007fa37300c33e in switch_core_session_thread () from
> /usr/lib64/libfreeswitch.so.1
> #16 0x00007fa373007d83 in switch_core_session_thread_pool_worker () from
> /usr/lib64/libfreeswitch.so.1
> #17 0x00007fa3732ce7c0 in dummy_worker () from
> /usr/lib64/libfreeswitch.so.1
> #18 0x00007fa3712d3e25 in start_thread () from /usr/lib64/libpthread.so.0
> #19 0x00007fa37092e34d in clone () from /usr/lib64/libc.so.6
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 
Anthony Minessale II
Founder, FreeSWITCH.
http://freeswitch.com


https://youtu.be/l_hOxzCt6X4
https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180608/8fe52cc0/attachment-0001.html>

From anthony.minessale at gmail.com  Fri Jun  8 20:00:14 2018
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jun 2018 15:00:14 -0500
Subject: [Freeswitch-users] Wrong release cause text, bug?
In-Reply-To: <CAFGRPVrJ1tw8V4Q9o7wmLjpnLp+Be01fW-sAue=4mHyHhf_WRg@mail.gmail.com>
References: <CAFGRPVrJ1tw8V4Q9o7wmLjpnLp+Be01fW-sAue=4mHyHhf_WRg@mail.gmail.com>
Message-ID: <CAKbxfG9Bt=8CWf3h18WV1crj9sE=zpm8e8mfa98O=7MFgMuRUw@mail.gmail.com>

If the sip stack hangs up due to its own decision.  it will not use a
custom response.  We can only control the response when its FS who triggers
the bye.


On Thu, Jun 7, 2018 at 4:11 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello guys,
>
> When hanging up a call with a specific release, FS sends the specific
> code, i.e.:
>
> if i release a call like so:
>
> <action inline="true" application="set" data="release_code=483
> EXCHANGE_ROUTING_ERROR"/>
>
> The BYE message is as follows:
>
>    SIP/2.0 483 EXCHANGE_ROUTING_ERROR
>    Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bK0y1H2X8899veB
>    Max-Forwards: 69
>    From: "david.villasmil" <sip:david.villasmil at 1.2.3.4>;tag=t35r9gatZFcjj
>    To: <sip:9999999 at 1.2.3.4:5080>;tag=pB7mFpUrc30gB
>    Call-ID: e2da7a79-e537-1236-5691-02e1e654f872
>    CSeq: 123866386 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>    Content-Length: 0
>    X-FS-Display-Name: 9999999
>    X-FS-Display-Number: sip:9999999 at 1.2.3.4
>    Remote-Party-ID: "9999999" <sip:9999999 at 1.2.3.4>;party=
> calling;privacy=off;screen=no
>
> Another example:
>
> <action inline="true" application="set" data="release_code=410
> NUMBER_CHANGED"/>
>
> The BYE message is SOMETIMES as follows:
>
>    SIP/2.0 410 NUMBER_CHANGED
>    Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKZN8r02Q5c16UF
>    Max-Forwards: 69
>    From: "david.villasmil" <sip:david.villasmil at 1.2.3.4>;tag=Q8Se4ZQF8m9SF
>    To: <sip:9999999 at 1.2.3.4:5080>;tag=mSm3B0SHjHmBm
>    Call-ID: dfbb5497-e537-1236-5691-02e1e654f872
>    CSeq: 123866383 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Reason: Q.850;cause=22;text="NUMBER_CHANGED"
>    Content-Length: 0
>    X-FS-Display-Name: 9999999
>    X-FS-Display-Number: sip:9999999 at 1.2.3.4
>    Remote-Party-ID: "9999999" <sip:9999999 at 54.84.149.30>;
> party=calling;privacy=off;screen=no
>
> And other times as follows:
>
>    SIP/2.0 410 Gone
>    Via: SIP/2.0/UDP 1.2.3.4:32200;branch=z9hG4bK-
> d8754z-86e4295fca2efa0a-1---d8754z-;rport=32200
>    Max-Forwards: 69
>    From: <sip:david.villasmil at 1.2.3.4;transport=UDP>;tag=42169b65
>    To: <sip:9999999 at 1.2.3.4;transport=UDP>;tag=pZ0N246BBcK7K
>    Call-ID: YmU5YjEwODk2ZmZhODJjNDgzNTNlM2FiZWVmMjI2MDg.
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>    Content-Length: 0
>    Remote-Party-ID: "9999999" <sip:9999999 at 1.2.3.4>;party=
> calling;privacy=off;screen=no
>
>
> But I've noticed that when i do:
>
> <action inline="true" application="set" data="release_code=488
> BEARERCAPABILITY_NOTIMPL"/>
>
> The BYE message is as follows:
>
>    SIP/2.0 488 BEARERCAPABILITY_NOTIMPL
>    Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bK3SDv7eUK14Z6D
>    Max-Forwards: 69
>    From: "david.villasmil" <sip:david.villasmil at 1.2.3.4>;tag=F3am0c2594H7a
>    To: <sip:9999999 at 1.2.3.4:5080>;tag=6XDmXaZj5BQ8S
>    Call-ID: aeb6624e-e539-1236-5691-02e1e654f872
>    CSeq: 123866772 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
>    Content-Length: 0
>    X-FS-Display-Name: 9999999
>    X-FS-Display-Number: sip:9999999 at 1.2.3.4
>    Remote-Party-ID: "9999999" <sip:9999999 at 54.84.149.30>;
> party=calling;privacy=off;screen=no
>
>
> So I guess, my question is why sometimes the "text" is set properly, and
> sometimes not??
>
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> ᐧ
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 
Anthony Minessale II
Founder, FreeSWITCH.
http://freeswitch.com


https://youtu.be/l_hOxzCt6X4
https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180608/583f9bad/attachment-0001.html>

From gmina at connectfirst.com  Fri Jun  8 21:12:58 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Fri, 8 Jun 2018 15:12:58 -0600
Subject: [Freeswitch-users] Hung Sofia Profile
In-Reply-To: <CAKbxfG93ZihgjXs3k-riO5SQc0Sx+H6WF-On4C47PW516fJ9+A@mail.gmail.com>
References: <CAKzROs3Um7Vae47zRQLGMYWLKyw7jL9fuHm+TSABf8X4bmc6VQ@mail.gmail.com>
 <CAKbxfG93ZihgjXs3k-riO5SQc0Sx+H6WF-On4C47PW516fJ9+A@mail.gmail.com>
Message-ID: <CAKzROs0qx0Qw-HFo5t+_NhmTkA4egs_+KTuvm1MKOsLa+5WTpQ@mail.gmail.com>

Would you guys suggest switching to Postgres on the back-end?  I am
guessing if there are known MySQL issues, the suggestion is probably native
PGSQL?

On Fri, Jun 8, 2018 at 2:48 PM Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You would need to do a thread apply all bt to see what the other threads
> are doing.  The thread you posted is stuck on a mutex waiting for something
> else, probably your DB in another thread.
> If you do get that info file it to jira because its easier to deal with
> backtraces there.
>
>
> On Fri, Jun 8, 2018 at 12:04 PM, Geoff Mina <gmina at connectfirst.com>
> wrote:
>
>> Greetings,
>> Has anyone seen an instance where freeswitch gets hung with the following
>> stack trace?  I have about 400 threads in this state and eventually FS
>> (1.6.19) just stops responding.
>>
>> I am running FS against MySQL 5.6.38 on CentOS using ODBC.   Guessing if
>> this is known, it probably isn't fixed in 1.6.20... but I can hope.  Any
>> input on what a possible solution would be greatly appreciated.  The MySQL
>> server and FS are running on the same host connecting via localhost
>> interface, so it's not a networking issue.
>>
>> Thread 445 (Thread 0x7fa2df6db700 (LWP 9395)):
>> #0  0x00007fa3712da42d in __lll_lock_wait () from
>> /usr/lib64/libpthread.so.0
>> #1  0x00007fa3712d5de6 in _L_lock_870 () from /usr/lib64/libpthread.so.0
>> #2  0x00007fa3712d5cdf in pthread_mutex_lock () from
>> /usr/lib64/libpthread.so.0
>> #3  0x00007fa35742468a in sofia_glue_execute_sql_callback () from
>> /usr/lib64/freeswitch/mod/mod_sofia.so
>> #4  0x00007fa3573df231 in select_from_profile () from
>> /usr/lib64/freeswitch/mod/mod_sofia.so
>> #5  0x00007fa3573e3b17 in sofia_contact_function () from
>> /usr/lib64/freeswitch/mod/mod_sofia.so
>> #6  0x00007fa3730594cc in switch_api_execute () from
>> /usr/lib64/libfreeswitch.so.1
>> #7  0x00007fa372fe69d6 in switch_channel_expand_variables_check () from
>> /usr/lib64/libfreeswitch.so.1
>> #8  0x00007fa3552d45e4 in user_outgoing_channel () from
>> /usr/lib64/freeswitch/mod/mod_dptools.so
>> #9  0x00007fa37300b42a in switch_core_session_outgoing_channel () from
>> /usr/lib64/libfreeswitch.so.1
>> #10 0x00007fa37308dc3b in switch_ivr_originate () from
>> /usr/lib64/libfreeswitch.so.1
>> #11 0x00007fa3552db0b5 in audio_bridge_function () from
>> /usr/lib64/freeswitch/mod/mod_dptools.so
>> #12 0x00007fa37300ebcb in switch_core_session_exec () from
>> /usr/lib64/libfreeswitch.so.1
>> #13 0x00007fa37300f179 in
>> switch_core_session_execute_application_get_flags () from
>> /usr/lib64/libfreeswitch.so.1
>> #14 0x00007fa373012a34 in switch_core_session_run () from
>> /usr/lib64/libfreeswitch.so.1
>> #15 0x00007fa37300c33e in switch_core_session_thread () from
>> /usr/lib64/libfreeswitch.so.1
>> #16 0x00007fa373007d83 in switch_core_session_thread_pool_worker () from
>> /usr/lib64/libfreeswitch.so.1
>> #17 0x00007fa3732ce7c0 in dummy_worker () from
>> /usr/lib64/libfreeswitch.so.1
>> #18 0x00007fa3712d3e25 in start_thread () from /usr/lib64/libpthread.so.0
>> #19 0x00007fa37092e34d in clone () from /usr/lib64/libc.so.6
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
>
>
> --
> Anthony Minessale II
> Founder, FreeSWITCH.
> http://freeswitch.com
>
>
> https://youtu.be/l_hOxzCt6X4
> https://www.youtube.com/watch?v=oAxXgyx5jUw
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180608/ee1499b4/attachment-0001.html>

From david.villasmil.work at gmail.com  Fri Jun  8 21:43:52 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 8 Jun 2018 23:43:52 +0200
Subject: [Freeswitch-users] Wrong release cause text, bug?
In-Reply-To: <CAKbxfG9Bt=8CWf3h18WV1crj9sE=zpm8e8mfa98O=7MFgMuRUw@mail.gmail.com>
References: <CAFGRPVrJ1tw8V4Q9o7wmLjpnLp+Be01fW-sAue=4mHyHhf_WRg@mail.gmail.com>
 <CAKbxfG9Bt=8CWf3h18WV1crj9sE=zpm8e8mfa98O=7MFgMuRUw@mail.gmail.com>
Message-ID: <CAFGRPVrr0Y2X4pZqVSi=ZR3w-ZzmjLLtCs_sos_qNdu4KPMJeQ@mail.gmail.com>

Yep, I'm hanging up with a custom cause. What surprises me is the
inconsistency of the reason text.

I'm trying to use the causes to act on it, but i.e.
BEARERCAPABILITY_NOTIMPL is not caught by the sending fs, the variable
comes up empty...

On Fri, Jun 8, 2018, 23:18 Anthony Minessale <anthony.minessale at gmail.com>
wrote:

> If the sip stack hangs up due to its own decision.  it will not use a
> custom response.  We can only control the response when its FS who triggers
> the bye.
>
>
> On Thu, Jun 7, 2018 at 4:11 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello guys,
>>
>> When hanging up a call with a specific release, FS sends the specific
>> code, i.e.:
>>
>> if i release a call like so:
>>
>> <action inline="true" application="set" data="release_code=483
>> EXCHANGE_ROUTING_ERROR"/>
>>
>> The BYE message is as follows:
>>
>>    SIP/2.0 483 EXCHANGE_ROUTING_ERROR
>>    Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bK0y1H2X8899veB
>>    Max-Forwards: 69
>>    From: "david.villasmil" <sip:david.villasmil at 1.2.3.4
>> >;tag=t35r9gatZFcjj
>>    To: <sip:9999999 at 1.2.3.4:5080>;tag=pB7mFpUrc30gB
>>    Call-ID: e2da7a79-e537-1236-5691-02e1e654f872
>>    CSeq: 123866386 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>>    Supported: timer, path, replaces
>>    Allow-Events: talk, hold, conference, refer
>>    Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>>    Content-Length: 0
>>    X-FS-Display-Name: 9999999
>>    X-FS-Display-Number: sip:9999999 at 1.2.3.4
>>    Remote-Party-ID: "9999999" <sip:9999999 at 1.2.3.4
>> >;party=calling;privacy=off;screen=no
>>
>> Another example:
>>
>> <action inline="true" application="set" data="release_code=410
>> NUMBER_CHANGED"/>
>>
>> The BYE message is SOMETIMES as follows:
>>
>>    SIP/2.0 410 NUMBER_CHANGED
>>    Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKZN8r02Q5c16UF
>>    Max-Forwards: 69
>>    From: "david.villasmil" <sip:david.villasmil at 1.2.3.4
>> >;tag=Q8Se4ZQF8m9SF
>>    To: <sip:9999999 at 1.2.3.4:5080>;tag=mSm3B0SHjHmBm
>>    Call-ID: dfbb5497-e537-1236-5691-02e1e654f872
>>    CSeq: 123866383 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>>    Supported: timer, path, replaces
>>    Allow-Events: talk, hold, conference, refer
>>    Reason: Q.850;cause=22;text="NUMBER_CHANGED"
>>    Content-Length: 0
>>    X-FS-Display-Name: 9999999
>>    X-FS-Display-Number: sip:9999999 at 1.2.3.4
>>    Remote-Party-ID: "9999999" <sip:9999999 at 54.84.149.30
>> >;party=calling;privacy=off;screen=no
>>
>> And other times as follows:
>>
>>    SIP/2.0 410 Gone
>>    Via: SIP/2.0/UDP 1.2.3.4:32200
>> ;branch=z9hG4bK-d8754z-86e4295fca2efa0a-1---d8754z-;rport=32200
>>    Max-Forwards: 69
>>    From: <sip:david.villasmil at 1.2.3.4;transport=UDP>;tag=42169b65
>>    To: <sip:9999999 at 1.2.3.4;transport=UDP>;tag=pZ0N246BBcK7K
>>    Call-ID: YmU5YjEwODk2ZmZhODJjNDgzNTNlM2FiZWVmMjI2MDg.
>>    CSeq: 1 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>>    Supported: timer, path, replaces
>>    Allow-Events: talk, hold, conference, refer
>>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>>    Content-Length: 0
>>    Remote-Party-ID: "9999999" <sip:9999999 at 1.2.3.4
>> >;party=calling;privacy=off;screen=no
>>
>>
>> But I've noticed that when i do:
>>
>> <action inline="true" application="set" data="release_code=488
>> BEARERCAPABILITY_NOTIMPL"/>
>>
>> The BYE message is as follows:
>>
>>    SIP/2.0 488 BEARERCAPABILITY_NOTIMPL
>>    Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bK3SDv7eUK14Z6D
>>    Max-Forwards: 69
>>    From: "david.villasmil" <sip:david.villasmil at 1.2.3.4
>> >;tag=F3am0c2594H7a
>>    To: <sip:9999999 at 1.2.3.4:5080>;tag=6XDmXaZj5BQ8S
>>    Call-ID: aeb6624e-e539-1236-5691-02e1e654f872
>>    CSeq: 123866772 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY
>>    Supported: timer, path, replaces
>>    Allow-Events: talk, hold, conference, refer
>>    Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
>>    Content-Length: 0
>>    X-FS-Display-Name: 9999999
>>    X-FS-Display-Number: sip:9999999 at 1.2.3.4
>>    Remote-Party-ID: "9999999" <sip:9999999 at 54.84.149.30
>> >;party=calling;privacy=off;screen=no
>>
>>
>> So I guess, my question is why sometimes the "text" is set properly, and
>> sometimes not??
>>
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>> ᐧ
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
>
>
> --
> Anthony Minessale II
> Founder, FreeSWITCH.
> http://freeswitch.com
>
>
> https://youtu.be/l_hOxzCt6X4
> https://www.youtube.com/watch?v=oAxXgyx5jUw
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180608/f457459f/attachment-0001.html>

From gmaruzz at gmail.com  Fri Jun  8 22:00:42 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 9 Jun 2018 00:00:42 +0200
Subject: [Freeswitch-users] wiki and org sites
In-Reply-To: <CAFGRPVr8kVwMdC8wALCEY+KM6PQ6uUGxuguw2HcFCNWu-KUwXA@mail.gmail.com>
References: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>
 <E785B568-3F86-4B2D-90E3-7771E417FCD3@jerris.com>
 <CAFGRPVpOxHeVtE95CCqb3VmK51_H1N875a9EKRqQF1kgWbVxVA@mail.gmail.com>
 <CAFGRPVr8kVwMdC8wALCEY+KM6PQ6uUGxuguw2HcFCNWu-KUwXA@mail.gmail.com>
Message-ID: <CALXCt0qoDe9wjug2qqrxiAD1xJuz5r8gotCUbmVcBTROhpMZJw@mail.gmail.com>

On Fri, Jun 8, 2018, 19:15 David Villasmil <david.villasmil.work at gmail.com>
wrote:

> BTW, I'll be at ClueCon this year, so see you guys in August!
>


David, Cluecon will be in July this year. Be sure to.plan accordingly!

;)

-giovanni







> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
> ᐧ
>
> On Thu, Jun 7, 2018 at 8:26 PM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Yeah, I remember that afterwards... it's just that LOTS of things are
>> pointing there...
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
>> ᐧ
>>
>> On Thu, Jun 7, 2018 at 8:25 PM Michael Jerris <mike at jerris.com> wrote:
>>
>>> wiki.freeswitch.org has been deprecated for many years now.  The
>>> correct location for the wiki is https://freeswitch.org/confluence
>>>
>>> Mike
>>>
>>> On Jun 6, 2018, at 3:02 PM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>> I can't connect to wiki.freeswitch.org, is the web down? I just get
>>> redirected to freeswitch's (new?) commercial web page (freeswitch.com)
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180609/5e2807ff/attachment.html>

From david.villasmil.work at gmail.com  Sat Jun  9 00:48:23 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 9 Jun 2018 02:48:23 +0200
Subject: [Freeswitch-users] Wrong release cause text, bug?
In-Reply-To: <CAFGRPVrr0Y2X4pZqVSi=ZR3w-ZzmjLLtCs_sos_qNdu4KPMJeQ@mail.gmail.com>
References: <CAFGRPVrJ1tw8V4Q9o7wmLjpnLp+Be01fW-sAue=4mHyHhf_WRg@mail.gmail.com>
 <CAKbxfG9Bt=8CWf3h18WV1crj9sE=zpm8e8mfa98O=7MFgMuRUw@mail.gmail.com>
 <CAFGRPVrr0Y2X4pZqVSi=ZR3w-ZzmjLLtCs_sos_qNdu4KPMJeQ@mail.gmail.com>
Message-ID: <CAFGRPVpiedi4+q7ar3Db=bHdgLO1TJB7QvYoQhHY9CCj8EDb-A@mail.gmail.com>

Anthony,

Doing something like:

        case SWITCH_CAUSE_NETWORK_OUT_OF_ORDER:
                return 503;
        case SWITCH_CAUSE_NORMAL_CIRCUIT_CONGESTION:
                return 503;
        case SWITCH_CAUSE_NORMAL_TEMPORARY_FAILURE:
                return 503;
        case SWITCH_CAUSE_REQUESTED_CHAN_UNAVAIL:
                return 503;
        case SWITCH_CAUSE_SWITCH_CONGESTION:
                return 503;
        case SWITCH_CAUSE_GATEWAY_DOWN:
                return 503;

Would go a long way into actually reporting the text properly, ordering
alphabetically and returning the corresponding code.

Would this be wrong?

Regards,

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

ᐧ

On Fri, Jun 8, 2018 at 11:43 PM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Yep, I'm hanging up with a custom cause. What surprises me is the
> inconsistency of the reason text.
>
> I'm trying to use the causes to act on it, but i.e.
> BEARERCAPABILITY_NOTIMPL is not caught by the sending fs, the variable
> comes up empty...
>
> On Fri, Jun 8, 2018, 23:18 Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>
>> If the sip stack hangs up due to its own decision.  it will not use a
>> custom response.  We can only control the response when its FS who triggers
>> the bye.
>>
>>
>> On Thu, Jun 7, 2018 at 4:11 PM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello guys,
>>>
>>> When hanging up a call with a specific release, FS sends the specific
>>> code, i.e.:
>>>
>>> if i release a call like so:
>>>
>>> <action inline="true" application="set" data="release_code=483
>>> EXCHANGE_ROUTING_ERROR"/>
>>>
>>> The BYE message is as follows:
>>>
>>>    SIP/2.0 483 EXCHANGE_ROUTING_ERROR
>>>    Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bK0y1H2X8899veB
>>>    Max-Forwards: 69
>>>    From: "david.villasmil" <sip:david.villasmil at 1.2.3.4
>>> >;tag=t35r9gatZFcjj
>>>    To: <sip:9999999 at 1.2.3.4:5080>;tag=pB7mFpUrc30gB
>>>    Call-ID: e2da7a79-e537-1236-5691-02e1e654f872
>>>    CSeq: 123866386 INVITE
>>>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>>>    Accept: application/sdp
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>>    Supported: timer, path, replaces
>>>    Allow-Events: talk, hold, conference, refer
>>>    Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
>>>    Content-Length: 0
>>>    X-FS-Display-Name: 9999999
>>>    X-FS-Display-Number: sip:9999999 at 1.2.3.4
>>>    Remote-Party-ID: "9999999" <sip:9999999 at 1.2.3.4
>>> >;party=calling;privacy=off;screen=no
>>>
>>> Another example:
>>>
>>> <action inline="true" application="set" data="release_code=410
>>> NUMBER_CHANGED"/>
>>>
>>> The BYE message is SOMETIMES as follows:
>>>
>>>    SIP/2.0 410 NUMBER_CHANGED
>>>    Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bKZN8r02Q5c16UF
>>>    Max-Forwards: 69
>>>    From: "david.villasmil" <sip:david.villasmil at 1.2.3.4
>>> >;tag=Q8Se4ZQF8m9SF
>>>    To: <sip:9999999 at 1.2.3.4:5080>;tag=mSm3B0SHjHmBm
>>>    Call-ID: dfbb5497-e537-1236-5691-02e1e654f872
>>>    CSeq: 123866383 INVITE
>>>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>>>    Accept: application/sdp
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>>    Supported: timer, path, replaces
>>>    Allow-Events: talk, hold, conference, refer
>>>    Reason: Q.850;cause=22;text="NUMBER_CHANGED"
>>>    Content-Length: 0
>>>    X-FS-Display-Name: 9999999
>>>    X-FS-Display-Number: sip:9999999 at 1.2.3.4
>>>    Remote-Party-ID: "9999999" <sip:9999999 at 54.84.149.30
>>> >;party=calling;privacy=off;screen=no
>>>
>>> And other times as follows:
>>>
>>>    SIP/2.0 410 Gone
>>>    Via: SIP/2.0/UDP 1.2.3.4:32200
>>> ;branch=z9hG4bK-d8754z-86e4295fca2efa0a-1---d8754z-;rport=32200
>>>    Max-Forwards: 69
>>>    From: <sip:david.villasmil at 1.2.3.4;transport=UDP>;tag=42169b65
>>>    To: <sip:9999999 at 1.2.3.4;transport=UDP>;tag=pZ0N246BBcK7K
>>>    Call-ID: YmU5YjEwODk2ZmZhODJjNDgzNTNlM2FiZWVmMjI2MDg.
>>>    CSeq: 1 INVITE
>>>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>>>    Accept: application/sdp
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>>    Supported: timer, path, replaces
>>>    Allow-Events: talk, hold, conference, refer
>>>    Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>>>    Content-Length: 0
>>>    Remote-Party-ID: "9999999" <sip:9999999 at 1.2.3.4
>>> >;party=calling;privacy=off;screen=no
>>>
>>>
>>> But I've noticed that when i do:
>>>
>>> <action inline="true" application="set" data="release_code=488
>>> BEARERCAPABILITY_NOTIMPL"/>
>>>
>>> The BYE message is as follows:
>>>
>>>    SIP/2.0 488 BEARERCAPABILITY_NOTIMPL
>>>    Via: SIP/2.0/UDP 1.2.3.4:5080;rport=5080;branch=z9hG4bK3SDv7eUK14Z6D
>>>    Max-Forwards: 69
>>>    From: "david.villasmil" <sip:david.villasmil at 1.2.3.4
>>> >;tag=F3am0c2594H7a
>>>    To: <sip:9999999 at 1.2.3.4:5080>;tag=6XDmXaZj5BQ8S
>>>    Call-ID: aeb6624e-e539-1236-5691-02e1e654f872
>>>    CSeq: 123866772 INVITE
>>>    User-Agent: FreeSWITCH-mod_sofia/1.6.20-37-987c9b9~64bit
>>>    Accept: application/sdp
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY
>>>    Supported: timer, path, replaces
>>>    Allow-Events: talk, hold, conference, refer
>>>    Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
>>>    Content-Length: 0
>>>    X-FS-Display-Name: 9999999
>>>    X-FS-Display-Number: sip:9999999 at 1.2.3.4
>>>    Remote-Party-ID: "9999999" <sip:9999999 at 54.84.149.30
>>> >;party=calling;privacy=off;screen=no
>>>
>>>
>>> So I guess, my question is why sometimes the "text" is set properly, and
>>> sometimes not??
>>>
>>>
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337
>>> ᐧ
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>> Founder, FreeSWITCH.
>> http://freeswitch.com
>>
>>
>> https://youtu.be/l_hOxzCt6X4
>> https://www.youtube.com/watch?v=oAxXgyx5jUw
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180609/4443eb03/attachment-0001.html>

From david.villasmil.work at gmail.com  Sat Jun  9 01:01:39 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 9 Jun 2018 03:01:39 +0200
Subject: [Freeswitch-users] wiki and org sites
In-Reply-To: <CALXCt0qoDe9wjug2qqrxiAD1xJuz5r8gotCUbmVcBTROhpMZJw@mail.gmail.com>
References: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>
 <E785B568-3F86-4B2D-90E3-7771E417FCD3@jerris.com>
 <CAFGRPVpOxHeVtE95CCqb3VmK51_H1N875a9EKRqQF1kgWbVxVA@mail.gmail.com>
 <CAFGRPVr8kVwMdC8wALCEY+KM6PQ6uUGxuguw2HcFCNWu-KUwXA@mail.gmail.com>
 <CALXCt0qoDe9wjug2qqrxiAD1xJuz5r8gotCUbmVcBTROhpMZJw@mail.gmail.com>
Message-ID: <CAFGRPVrz18bbgxRM3WuC9-DG4yOMd4oTqkbVuaU-p-HOqmBdcQ@mail.gmail.com>

Yeah, brainfart! Coming back to Spain on August...
Regards,

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

ᐧ

On Sat, Jun 9, 2018 at 3:00 AM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

>
>
> On Fri, Jun 8, 2018, 19:15 David Villasmil <david.villasmil.work at gmail.com>
> wrote:
>
>> BTW, I'll be at ClueCon this year, so see you guys in August!
>>
>
>
> David, Cluecon will be in July this year. Be sure to.plan accordingly!
>
> ;)
>
> -giovanni
>
>
>
>
>
>
>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
>> ᐧ
>>
>> On Thu, Jun 7, 2018 at 8:26 PM David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Yeah, I remember that afterwards... it's just that LOTS of things are
>>> pointing there...
>>>
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337
>>>
>>> ᐧ
>>>
>>> On Thu, Jun 7, 2018 at 8:25 PM Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> wiki.freeswitch.org has been deprecated for many years now.  The
>>>> correct location for the wiki is https://freeswitch.org/confluence
>>>>
>>>> Mike
>>>>
>>>> On Jun 6, 2018, at 3:02 PM, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>> I can't connect to wiki.freeswitch.org, is the web down? I just get
>>>> redirected to freeswitch's (new?) commercial web page (freeswitch.com)
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>
>>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180609/9497a5d1/attachment-0001.html>

From anthony.minessale at gmail.com  Sat Jun  9 01:42:11 2018
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Jun 2018 20:42:11 -0500
Subject: [Freeswitch-users] Hung Sofia Profile
In-Reply-To: <CAKzROs0qx0Qw-HFo5t+_NhmTkA4egs_+KTuvm1MKOsLa+5WTpQ@mail.gmail.com>
References: <CAKzROs3Um7Vae47zRQLGMYWLKyw7jL9fuHm+TSABf8X4bmc6VQ@mail.gmail.com>
 <CAKbxfG93ZihgjXs3k-riO5SQc0Sx+H6WF-On4C47PW516fJ9+A@mail.gmail.com>
 <CAKzROs0qx0Qw-HFo5t+_NhmTkA4egs_+KTuvm1MKOsLa+5WTpQ@mail.gmail.com>
Message-ID: <CAKbxfG-oe7oxEQsvbHBrck4=fUWQvviKGvARGUZjLodSD_vMNg@mail.gmail.com>

There are some params on confluence that show how to optimize the mysql.
We don't like evangelizing one or the other technology but ya, postgres was
at least better at threading when we first added db support many years ago.


On Fri, Jun 8, 2018 at 4:12 PM, Geoff Mina <gmina at connectfirst.com> wrote:

> Would you guys suggest switching to Postgres on the back-end?  I am
> guessing if there are known MySQL issues, the suggestion is probably native
> PGSQL?
>
> On Fri, Jun 8, 2018 at 2:48 PM Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> You would need to do a thread apply all bt to see what the other threads
>> are doing.  The thread you posted is stuck on a mutex waiting for something
>> else, probably your DB in another thread.
>> If you do get that info file it to jira because its easier to deal with
>> backtraces there.
>>
>>
>> On Fri, Jun 8, 2018 at 12:04 PM, Geoff Mina <gmina at connectfirst.com>
>> wrote:
>>
>>> Greetings,
>>> Has anyone seen an instance where freeswitch gets hung with the
>>> following stack trace?  I have about 400 threads in this state and
>>> eventually FS (1.6.19) just stops responding.
>>>
>>> I am running FS against MySQL 5.6.38 on CentOS using ODBC.   Guessing if
>>> this is known, it probably isn't fixed in 1.6.20... but I can hope.  Any
>>> input on what a possible solution would be greatly appreciated.  The MySQL
>>> server and FS are running on the same host connecting via localhost
>>> interface, so it's not a networking issue.
>>>
>>> Thread 445 (Thread 0x7fa2df6db700 (LWP 9395)):
>>> #0  0x00007fa3712da42d in __lll_lock_wait () from
>>> /usr/lib64/libpthread.so.0
>>> #1  0x00007fa3712d5de6 in _L_lock_870 () from /usr/lib64/libpthread.so.0
>>> #2  0x00007fa3712d5cdf in pthread_mutex_lock () from
>>> /usr/lib64/libpthread.so.0
>>> #3  0x00007fa35742468a in sofia_glue_execute_sql_callback () from
>>> /usr/lib64/freeswitch/mod/mod_sofia.so
>>> #4  0x00007fa3573df231 in select_from_profile () from
>>> /usr/lib64/freeswitch/mod/mod_sofia.so
>>> #5  0x00007fa3573e3b17 in sofia_contact_function () from
>>> /usr/lib64/freeswitch/mod/mod_sofia.so
>>> #6  0x00007fa3730594cc in switch_api_execute () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #7  0x00007fa372fe69d6 in switch_channel_expand_variables_check () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #8  0x00007fa3552d45e4 in user_outgoing_channel () from
>>> /usr/lib64/freeswitch/mod/mod_dptools.so
>>> #9  0x00007fa37300b42a in switch_core_session_outgoing_channel () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #10 0x00007fa37308dc3b in switch_ivr_originate () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #11 0x00007fa3552db0b5 in audio_bridge_function () from
>>> /usr/lib64/freeswitch/mod/mod_dptools.so
>>> #12 0x00007fa37300ebcb in switch_core_session_exec () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #13 0x00007fa37300f179 in switch_core_session_execute_application_get_flags
>>> () from /usr/lib64/libfreeswitch.so.1
>>> #14 0x00007fa373012a34 in switch_core_session_run () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #15 0x00007fa37300c33e in switch_core_session_thread () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #16 0x00007fa373007d83 in switch_core_session_thread_pool_worker ()
>>> from /usr/lib64/libfreeswitch.so.1
>>> #17 0x00007fa3732ce7c0 in dummy_worker () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #18 0x00007fa3712d3e25 in start_thread () from /usr/lib64/libpthread.so.0
>>> #19 0x00007fa37092e34d in clone () from /usr/lib64/libc.so.6
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>> Founder, FreeSWITCH.
>> http://freeswitch.com
>>
>>
>> https://youtu.be/l_hOxzCt6X4
>> https://www.youtube.com/watch?v=oAxXgyx5jUw
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 
Anthony Minessale II
Founder, FreeSWITCH.
http://freeswitch.com


https://youtu.be/l_hOxzCt6X4
https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180608/7f187942/attachment-0001.html>

From gmaruzz at gmail.com  Sat Jun  9 06:11:29 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 9 Jun 2018 08:11:29 +0200
Subject: [Freeswitch-users] Hung Sofia Profile
In-Reply-To: <CAKzROs0qx0Qw-HFo5t+_NhmTkA4egs_+KTuvm1MKOsLa+5WTpQ@mail.gmail.com>
References: <CAKzROs3Um7Vae47zRQLGMYWLKyw7jL9fuHm+TSABf8X4bmc6VQ@mail.gmail.com>
 <CAKbxfG93ZihgjXs3k-riO5SQc0Sx+H6WF-On4C47PW516fJ9+A@mail.gmail.com>
 <CAKzROs0qx0Qw-HFo5t+_NhmTkA4egs_+KTuvm1MKOsLa+5WTpQ@mail.gmail.com>
Message-ID: <CALXCt0qeJE+GK6A8fcv+0mcWqdS6c34mDbzWZHRxtRGbUPJ5JA@mail.gmail.com>

On Sat, Jun 9, 2018, 02:12 Geoff Mina <gmina at connectfirst.com> wrote:

> Would you guys suggest switching to Postgres on the back-end?  I am
> guessing if there are known MySQL issues, the suggestion is probably native
> PGSQL?
>


Problem is with the odbc driver.

 So, short answer is: yes, definitely go with postgresql native support


-giovanni



> On Fri, Jun 8, 2018 at 2:48 PM Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> You would need to do a thread apply all bt to see what the other threads
>> are doing.  The thread you posted is stuck on a mutex waiting for something
>> else, probably your DB in another thread.
>> If you do get that info file it to jira because its easier to deal with
>> backtraces there.
>>
>>
>> On Fri, Jun 8, 2018 at 12:04 PM, Geoff Mina <gmina at connectfirst.com>
>> wrote:
>>
>>> Greetings,
>>> Has anyone seen an instance where freeswitch gets hung with the
>>> following stack trace?  I have about 400 threads in this state and
>>> eventually FS (1.6.19) just stops responding.
>>>
>>> I am running FS against MySQL 5.6.38 on CentOS using ODBC.   Guessing if
>>> this is known, it probably isn't fixed in 1.6.20... but I can hope.  Any
>>> input on what a possible solution would be greatly appreciated.  The MySQL
>>> server and FS are running on the same host connecting via localhost
>>> interface, so it's not a networking issue.
>>>
>>> Thread 445 (Thread 0x7fa2df6db700 (LWP 9395)):
>>> #0  0x00007fa3712da42d in __lll_lock_wait () from
>>> /usr/lib64/libpthread.so.0
>>> #1  0x00007fa3712d5de6 in _L_lock_870 () from /usr/lib64/libpthread.so.0
>>> #2  0x00007fa3712d5cdf in pthread_mutex_lock () from
>>> /usr/lib64/libpthread.so.0
>>> #3  0x00007fa35742468a in sofia_glue_execute_sql_callback () from
>>> /usr/lib64/freeswitch/mod/mod_sofia.so
>>> #4  0x00007fa3573df231 in select_from_profile () from
>>> /usr/lib64/freeswitch/mod/mod_sofia.so
>>> #5  0x00007fa3573e3b17 in sofia_contact_function () from
>>> /usr/lib64/freeswitch/mod/mod_sofia.so
>>> #6  0x00007fa3730594cc in switch_api_execute () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #7  0x00007fa372fe69d6 in switch_channel_expand_variables_check () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #8  0x00007fa3552d45e4 in user_outgoing_channel () from
>>> /usr/lib64/freeswitch/mod/mod_dptools.so
>>> #9  0x00007fa37300b42a in switch_core_session_outgoing_channel () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #10 0x00007fa37308dc3b in switch_ivr_originate () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #11 0x00007fa3552db0b5 in audio_bridge_function () from
>>> /usr/lib64/freeswitch/mod/mod_dptools.so
>>> #12 0x00007fa37300ebcb in switch_core_session_exec () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #13 0x00007fa37300f179 in
>>> switch_core_session_execute_application_get_flags () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #14 0x00007fa373012a34 in switch_core_session_run () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #15 0x00007fa37300c33e in switch_core_session_thread () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #16 0x00007fa373007d83 in switch_core_session_thread_pool_worker () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #17 0x00007fa3732ce7c0 in dummy_worker () from
>>> /usr/lib64/libfreeswitch.so.1
>>> #18 0x00007fa3712d3e25 in start_thread () from /usr/lib64/libpthread.so.0
>>> #19 0x00007fa37092e34d in clone () from /usr/lib64/libc.so.6
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>> Founder, FreeSWITCH.
>> http://freeswitch.com
>>
>>
>> https://youtu.be/l_hOxzCt6X4
>> https://www.youtube.com/watch?v=oAxXgyx5jUw
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180609/d78a0d5c/attachment-0001.html>

From vishalmpai at gmail.com  Sat Jun  9 17:10:31 2018
From: vishalmpai at gmail.com (Vishal Pai)
Date: Sat, 9 Jun 2018 22:40:31 +0530
Subject: [Freeswitch-users] mod_verto
In-Reply-To: <C16D05FD-E04E-4519-A75F-50BADE4FCAE0@jerris.com>
References: <CACOmXtBSxNbtKgH961AP0D=5jEpgtPe3gb+cjR-w4GU3oZKUFQ@mail.gmail.com>
 <C16D05FD-E04E-4519-A75F-50BADE4FCAE0@jerris.com>
Message-ID: <CACOmXtBrb8Z8PMVn1=k0KvZV7qGRQT+dLS5BHRidTnN64KecBg@mail.gmail.com>

Great.....

Thanks for guiding me.

On Thu, 7 Jun 2018 at 9:39 PM, Michael Jerris <mike at jerris.com> wrote:

> Thank you for your question! We answered it live on ClueCon weekly today!
> You can watch it at https://www.youtube.com/watch?v=gIsEiddBbFo
>
> Mike
>
> On Jun 3, 2018, at 4:02 AM, Vishal Pai <vishalmpai at gmail.com> wrote:
>
> Hi All
>
>
> Can we record the video conferencing using mod_verto. If yes how it is
> possible and and what would it’s file format.
>
>
> Thanks
> Vishal Pai
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180609/daf5b564/attachment-0001.html>

From rick at magicmail.mooo.com  Mon Jun 11 21:42:50 2018
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Mon, 11 Jun 2018 22:42:50 +0100
Subject: [Freeswitch-users] Port numbers
Message-ID: <etPan.5b1eecda.17725b58.1dd@magicmail.mooo.com>

Ok kind of FreeSWITCH 101 here I know, but something I’ve never understood completely:

A provider wants to send us calls as number at our-ip-address - fine, but it’s hitting us on port 5060 (whereas external is set up as 5080). So can’t use external, have to use internal profile, right?

So I can create a ACL to allow their IPs onto Internal, ‘inbound’ ACL I think?

But how do I handle call routing for inbound calls from provider’s IPs, vs my registered users dialling out?

If anyone can simplify this for me that would be great :)

Thanks
__
R

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180611/6abec08e/attachment-0001.html>

From s.safarov at gmail.com  Tue Jun 12 07:30:05 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 12 Jun 2018 10:30:05 +0300
Subject: [Freeswitch-users] Wrong IP in ACK
Message-ID: <CAHtxdDcXfiD73Av6wrgybCZk5zDx6k2Tnjif-PWEuDGf3Zy_cQ@mail.gmail.com>

Are anybody experiences issue when for some of calls  ACK send to other IP.
http://prntscr.com/jtxkbz
in this case FS in bypass media with two profiles (1 to internet, 2
intranet).
INVITE is send to DNS name that is resolved to

   1. 10.0.9.32
   2. 10.0.9.33
   3. 10.0.9.34
   4. 10.0.9.35

FS 1.6.19

Contact on 200 message is correct

SIP/2.0 200 OK Via: SIP/2.0/TCP 10.0.9.51:12000;rport=39049;branch=
z9hG4bKBH60NQjSZj36a From: "Anonymous" <sip:+12022553656 at 10.0.9.51>;tag=
rH5cBQarr559D To: <sip:13013688630 at metropolis.ack.one>;tag=UDv7UXv3B53Da
Call-ID: b0e4d7e1-e79e-1236-8cb9-02420a000933 CSeq: 123998414 INVITE
Contact: <sip:13013688630 at 10.0.9.33:11000;transport=tcp> User-Agent:
proxy.ackone.com Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Now for future investigate FS upgrader to 1.6.20 and enabled siptrace and
loglevel

Sergey Safarov
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180612/77f6aaf8/attachment-0001.html>

From s.safarov at gmail.com  Tue Jun 12 08:45:22 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 12 Jun 2018 11:45:22 +0300
Subject: [Freeswitch-users] Wrong IP in ACK
In-Reply-To: <CAHtxdDcXfiD73Av6wrgybCZk5zDx6k2Tnjif-PWEuDGf3Zy_cQ@mail.gmail.com>
References: <CAHtxdDcXfiD73Av6wrgybCZk5zDx6k2Tnjif-PWEuDGf3Zy_cQ@mail.gmail.com>
Message-ID: <CAHtxdDeD1iX0U3-PC2yOyMy6yuFxK=n5PKd1gRnphnvDfkqQ8A@mail.gmail.com>

root of issue
ACK send to IP resolved via DNS for value defined as "fs_path", rather than
defined in contact string.

more details
https://freeswitch.org/jira/browse/FS-11190

вт, 12 июн. 2018 г. в 10:30, Sergey Safarov <s.safarov at gmail.com>:

> Are anybody experiences issue when for some of calls  ACK send to other IP.
> http://prntscr.com/jtxkbz
> in this case FS in bypass media with two profiles (1 to internet, 2
> intranet).
> INVITE is send to DNS name that is resolved to
>
>    1. 10.0.9.32
>    2. 10.0.9.33
>    3. 10.0.9.34
>    4. 10.0.9.35
>
> FS 1.6.19
>
> Contact on 200 message is correct
>
> SIP/2.0 200 OK Via: SIP/2.0/TCP 10.0.9.51:12000;rport=39049;branch=
> z9hG4bKBH60NQjSZj36a From: "Anonymous" <sip:+12022553656 at 10.0.9.51>;tag=
> rH5cBQarr559D To: <sip:13013688630 <(301)%20368-8630>@metropolis.ack.one>;tag=UDv7UXv3B53Da
> Call-ID: b0e4d7e1-e79e-1236-8cb9-02420a000933 CSeq: 123998414 INVITE
> Contact: <sip:13013688630 <(301)%20368-8630>@10.0.9.33:11000;transport=tcp>
> User-Agent: proxy.ackone.com Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
> MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Now for future investigate FS upgrader to 1.6.20 and enabled siptrace and
> loglevel
>
> Sergey Safarov
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180612/debf576b/attachment-0001.html>

From mario_fs at mgtech.com  Tue Jun 12 16:49:27 2018
From: mario_fs at mgtech.com (Mario)
Date: Tue, 12 Jun 2018 09:49:27 -0700
Subject: [Freeswitch-users] When 1.6.21 ?
In-Reply-To: <CFFFCB20-E42D-4525-A0E1-DC11C533BE88@jerris.com>
References: <A34EB9C4-2177-4757-B5A7-8B8C9ECD900E@mgtech.com>
 <CFFFCB20-E42D-4525-A0E1-DC11C533BE88@jerris.com>
Message-ID: <F722295B-0D69-4E0B-B037-DE6FDCBDAB16@mgtech.com>

May be a dumb question… back port usually refers to a previous release but 1.6.20 shows as the current release on the wiki and files.freeswitch.org <http://file.freeswitch.org/>. Does this mean 1.8 is soon? Thanks,
Mario G

> On Jun 8, 2018, at 11:23 AM, Michael Jerris <mike at jerris.com> wrote:
> 
> I plan to put a call for suggestions of what to back port into 1.6.21 in the next few weeks, then it will be just based on my availability to review those and see what is practical to put in.
> 
>> On Jun 8, 2018, at 12:01 PM, Mario <mario_fs at mgtech.com> wrote:
>> 
>> Can anyone answer when 1.6.21 will be available? Just wondering since I haven’t been able to test FS production for macOS wiki since PCRE was updated (FS-11061). Also wanted to start testing on macOS Mojave to see how that goes. Thanks,
>> Mario G
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180612/69619fb4/attachment-0001.html>

From mario_fs at mgtech.com  Tue Jun 12 17:38:30 2018
From: mario_fs at mgtech.com (Mario)
Date: Tue, 12 Jun 2018 10:38:30 -0700
Subject: [Freeswitch-users] wiki and org sites
In-Reply-To: <CAK6zN=0C8OFK_35fwVjLMcWyqQWP1vJrDW3i_uZesrTDvS1UGA@mail.gmail.com>
References: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>
 <E785B568-3F86-4B2D-90E3-7771E417FCD3@jerris.com>
 <CAFGRPVpOxHeVtE95CCqb3VmK51_H1N875a9EKRqQF1kgWbVxVA@mail.gmail.com>
 <A7460BF3-DE88-4C9F-B5A8-3033DCB7A3BC@jerris.com>
 <CAK6zN=0C8OFK_35fwVjLMcWyqQWP1vJrDW3i_uZesrTDvS1UGA@mail.gmail.com>
Message-ID: <23DB1380-EDF9-479E-86C4-8B131D83AD72@mgtech.com>

Way back when the conversion started I offered to scan the data using Rexx and produce a report of links to old pages but no one ever got back to me. I think a report like that would have reduced/eliminated this problem by now.

BTW, I fixed most of the pages you referenced below:
https://freeswitch.org/confluence/display/FREESWITCH/Early+Media <https://freeswitch.org/confluence/display/FREESWITCH/Early+Media>
https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer <https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer>
https://freeswitch.org/confluence/display/FREESWITCH/Audio+Codecs <https://freeswitch.org/confluence/display/FREESWITCH/Audio+Codecs>
https://freeswitch.org/confluence/display/FREESWITCH/IVR <https://freeswitch.org/confluence/display/FREESWITCH/IVR>.   <- except for voicemail.py which I could not find
https://freeswitch.org/confluence/display/FREESWITCH/Dialplan.   <— this was already correct
https://freeswitch.org/confluence/display/FREESWITCH/mod_v8 <https://freeswitch.org/confluence/display/FREESWITCH/mod_v8>.   <- two  links removed since pages combined with others

Mario G


> On Jun 8, 2018, at 11:42 AM, Preston Hagar <prestonh at gmail.com> wrote:
> 
> A lot of the internal confluence links point back to the wiki.  Here are some pages I've run across that have a lot of links back to the wiki.
> 
> Freeswitch IVR Originate at bottom of page here
> 
> https://freeswitch.org/confluence/display/FREESWITCH/Dialplan <https://freeswitch.org/confluence/display/FREESWITCH/Dialplan>.   <— this was already correct
> 
> Almost all the links on the page here:
> 
> https://freeswitch.org/confluence/display/FREESWITCH/mod_v8 <https://freeswitch.org/confluence/display/FREESWITCH/mod_v8>
> 
> and here:
> 
> https://freeswitch.org/confluence/display/FREESWITCH/JavaScript <https://freeswitch.org/confluence/display/FREESWITCH/JavaScript>
> 
> https://freeswitch.org/confluence/display/FREESWITCH/JavaScript+Event <https://freeswitch.org/confluence/display/FREESWITCH/JavaScript+Event>
> 
> https://freeswitch.org/confluence/display/FREESWITCH/IVR <https://freeswitch.org/confluence/display/FREESWITCH/IVR>
> 
> https://freeswitch.org/confluence/display/FREESWITCH/Audio+Codecs <https://freeswitch.org/confluence/display/FREESWITCH/Audio+Codecs>
> 
> https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer <https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer>
> 
> https://freeswitch.org/confluence/display/FREESWITCH/Early+Media <https://freeswitch.org/confluence/display/FREESWITCH/Early+Media>
> 
> On Thu, Jun 7, 2018 at 2:44 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> If there is anything still pointing there it needs to be fixed.  
> 
>> On Jun 7, 2018, at 2:26 PM, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>> 
>> Yeah, I remember that afterwards... it's just that LOTS of things are pointing there...
>> 
>> Regards,
>> 
>> David Villasmil
>> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
>> phone: +34669448337
>> 
>> ᐧ
>> 
>> On Thu, Jun 7, 2018 at 8:25 PM Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> wiki.freeswitch.org <http://wiki.freeswitch.org/> has been deprecated for many years now.  The correct location for the wiki is https://freeswitch.org/confluence <https://freeswitch.org/confluence>
>> 
>> Mike
>> 
>>> On Jun 6, 2018, at 3:02 PM, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> wrote:
>>> 
>>> I can't connect to wiki.freeswitch.org <http://wiki.freeswitch.org/>, is the web down? I just get redirected to freeswitch's (new?) commercial web page (freeswitch.com <http://freeswitch.com/>)
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180612/d6075093/attachment-0001.html>

From gb at cm.nl  Wed Jun 13 09:02:28 2018
From: gb at cm.nl (Grant Bagdasarian)
Date: Wed, 13 Jun 2018 09:02:28 +0000
Subject: [Freeswitch-users] Pyrun limitations
Message-ID: <0f8a299688cb45d5b78a49b98301c047@cm.nl>

Hello,

We're currently using the NEventSocket library to control our freeswitch servers for starting python scripts (using a backgroundjob and pyrun py_module_name) which internally use the freeswitch.py library to originate a call.
However, we're running into some performance limitations where we can't seem to get more than 5 call setups per second.

To quote the documentation (https://freeswitch.org/confluence/display/FREESWITCH/mod_python):
"A single python interpreter is spawned at module startup and used for the lifetime of the freeswitch process."
"Each thread swaps in its "thread state" before executing python code and then swaps it out when finished. Also during blocking calls into freeswitch, a thread will swap out its thread state in order to not block other threads, and then swap it in after the blocking call to freeswitch has finished."

How does this relate to the number of threads freeswitch is allowed to start and run python scripts? If there are hard limitations set, can these be increased? I couldn't find hard numbers in the source files, so this may be controlled somewhere else?

Thanks.

Grant Bagdasarian

Senior Developer



cm.com<https://cm.com>




-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180613/e8341e56/attachment.html>

From ko at sv01.de  Wed Jun 13 09:43:34 2018
From: ko at sv01.de (Kevin Olbrich)
Date: Wed, 13 Jun 2018 11:43:34 +0200
Subject: [Freeswitch-users] Access to release 1.8
Message-ID: <CA+gLzy8c17TaduDHHTasCfPu7g4UYdcWjuLTgBYdjKz=czm=FA@mail.gmail.com>

Hi!

In Jan 2017 release 1.8 was described as "With the release date of 1.8 fast
approaching" or "The result of all of this you ask? FreeSWITCH 1.8, and
after that just think we'll be at FreeSWITCH 2.0!".
As of now, there is only 1.6 available for public access.

Freeswitch 1.6 is working fine but is 1.8 still in the works?

Source:
http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124534.html

Kind regards,
Kevin
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180613/0c4b0728/attachment.html>

From gregor at infomedia.si  Wed Jun 13 21:03:12 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Wed, 13 Jun 2018 23:03:12 +0200
Subject: [Freeswitch-users] Pyrun limitations
In-Reply-To: <0f8a299688cb45d5b78a49b98301c047@cm.nl>
References: <0f8a299688cb45d5b78a49b98301c047@cm.nl>
Message-ID: <CAC7ttFF0CoeK3GVGY7REnQzs4nopx=FmjkRh2U+0k6f99hXPJw@mail.gmail.com>

Grant, how are u satisfied with neventsocket library?

On Wed, Jun 13, 2018, 22:51 Grant Bagdasarian <gb at cm.nl> wrote:

> Hello,
>
>
>
> We’re currently using the NEventSocket library to control our freeswitch
> servers for starting python scripts (using a backgroundjob and pyrun
> py_module_name) which internally use the freeswitch.py library to originate
> a call.
>
> However, we’re running into some performance limitations where we can’t
> seem to get more than 5 call setups per second.
>
>
>
> To quote the documentation (
> https://freeswitch.org/confluence/display/FREESWITCH/mod_python):
>
> “A single python interpreter is spawned at module startup and used for the
> lifetime of the freeswitch process.”
>
> “Each thread swaps in its "thread state" before executing python code and
> then swaps it out when finished. Also during blocking calls into
> freeswitch, a thread will swap out its thread state in order to not block
> other threads, and then swap it in after the blocking call to freeswitch
> has finished.”
>
>
>
> How does this relate to the number of threads freeswitch is allowed to
> start and run python scripts? If there are hard limitations set, can these
> be increased? I couldn’t find hard numbers in the source files, so this may
> be controlled somewhere else?
>
>
>
> Thanks.
>
>
>
> Grant Bagdasarian
>
> Senior Developer
>
>
>
> cm.com
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180613/e5d88a30/attachment-0001.html>

From robcapo at gmail.com  Wed Jun 13 21:04:57 2018
From: robcapo at gmail.com (Rob Capo)
Date: Wed, 13 Jun 2018 17:04:57 -0400
Subject: [Freeswitch-users] Initiate SIP session with MRCPv2 server
Message-ID: <CAJN5KK+Nq2+8ZKMPrYLNCTx-=CPQue_OetoJSfJk1cxRPQ-NxA@mail.gmail.com>

Hello,

I have an MRCPv2 client that I'd like to use in an ASR application which is
connected to FreeSWITCH via eventsocket (application written in Go). I'd
like to use this client for MRCP signaling because it has the fine-grained
control that I need, and the API is easier to use than the unimrcp
eventsocket API.

However, I'd like to have FreeSWITCH take care of answering the incoming
call from the user. I don't care quite as much about RTP and SIP signaling
to/from the ASR, but would like some control over it if possible (e.g.
being able to see the RTP data for recording segments of audio that are
important to me).

There are a few ways I can think of solving this, some hackier than others.

Currently, what I'm doing is probably the hackiest:
1. answer the call with FS
2. Initiate MRCPv2 session with ASR using my client
3. get the call's local media port over ES
4. pcap that port, and proxy the media to the ASR manually

Perhaps a better way would be:
1. answer the call with FS
2. initiate MRCPv2 session with ASR using my client
3. tell FS to proxy audio to the ASR's RTP port from SDP (maybe passing
through a proxy UDP socket that I open)

But I'm not sure if this is possible.

Another way that might be more approachable:
1. answer the call with FS
2. bridge in the ASR using FS
3. get the SDP / MRCPv2 port from FS
4. initiate TCP connection to MRCPv2 port and use my client

The disadvantage would be that I no longer have visibility / control over
the SIP / RTP paths, but I could always use pcap as before if the need
arises (the situations where I need this control are most likely going to
be debugging / non-production purposes).

Does anyone have better recommendations or details on how I could achieve
the second and / or third methods?

Thanks,
Rob
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180613/3038f8e9/attachment-0001.html>

From hawkins at hawkinsegroup.com  Wed Jun 13 22:00:56 2018
From: hawkins at hawkinsegroup.com (Don Hawkins)
Date: Wed, 13 Jun 2018 17:00:56 -0500
Subject: [Freeswitch-users] Port numbers
In-Reply-To: <etPan.5b1eecda.17725b58.1dd@magicmail.mooo.com>
References: <etPan.5b1eecda.17725b58.1dd@magicmail.mooo.com>
Message-ID: <CAO6gFMz_nFe35jfPCbBWzCEpoVc35-3Wo1_LZ+EJJK-08jywkg@mail.gmail.com>

Regardless of how you look at it, all calls are INBOUND calls to
Freeswitch, wether they be from your DID provider or a registered SIP user.

Calls are going into Freeswitch either way, all you have to do is setup
each profile to handle the inbound call (connect to the number dialed,
connect to extension, etc.).

Sent from my NationPCS Galaxy Note 5

On Wed, Jun 13, 2018, 3:50 PM Rick Jarvis <rick at magicmail.mooo.com> wrote:

> Ok kind of FreeSWITCH 101 here I know, but something I’ve never understood
> completely:
>
> A provider wants to send us calls as number at our-ip-address - fine, but
> it’s hitting us on port 5060 (whereas external is set up as 5080). So can’t
> use external, have to use internal profile, right?
>
> So I can create a ACL to allow their IPs onto Internal, ‘inbound’ ACL I
> think?
>
> But how do I handle call routing for inbound calls from provider’s IPs, vs
> my registered users dialling out?
>
> If anyone can simplify this for me that would be great :)
>
> Thanks
> __
> R
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180613/246175d3/attachment-0001.html>

From anthony.minessale at gmail.com  Wed Jun 13 22:17:11 2018
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Jun 2018 17:17:11 -0500
Subject: [Freeswitch-users] When 1.6.21 ?
In-Reply-To: <F722295B-0D69-4E0B-B037-DE6FDCBDAB16@mgtech.com>
References: <A34EB9C4-2177-4757-B5A7-8B8C9ECD900E@mgtech.com>
 <CFFFCB20-E42D-4525-A0E1-DC11C533BE88@jerris.com>
 <F722295B-0D69-4E0B-B037-DE6FDCBDAB16@mgtech.com>
Message-ID: <CAKbxfG_DNxf65KOTZjroeSkcX9kZgRwhg2JRhcuVBStX2ur1HQ@mail.gmail.com>

He means from master branch.


On Tue, Jun 12, 2018 at 11:49 AM, Mario <mario_fs at mgtech.com> wrote:

> May be a dumb question… back port usually refers to a previous release but
> 1.6.20 shows as the current release on the wiki and files.freeswitch.org
> <http://file.freeswitch.org>. Does this mean 1.8 is soon? Thanks,
> Mario G
>
>
> On Jun 8, 2018, at 11:23 AM, Michael Jerris <mike at jerris.com> wrote:
>
> I plan to put a call for suggestions of what to back port into 1.6.21 in
> the next few weeks, then it will be just based on my availability to review
> those and see what is practical to put in.
>
> On Jun 8, 2018, at 12:01 PM, Mario <mario_fs at mgtech.com> wrote:
>
> Can anyone answer when 1.6.21 will be available? Just wondering since I
> haven’t been able to test FS production for macOS wiki since PCRE was
> updated (FS-11061). Also wanted to start testing on macOS Mojave to see how
> that goes. Thanks,
> Mario G
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 
Anthony Minessale II
Founder, FreeSWITCH.
http://freeswitch.com


https://youtu.be/l_hOxzCt6X4
https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180613/c8064455/attachment-0001.html>

From gmina at connectfirst.com  Wed Jun 13 22:42:44 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Wed, 13 Jun 2018 16:42:44 -0600
Subject: [Freeswitch-users] Hung Sofia Profile
In-Reply-To: <CALXCt0qeJE+GK6A8fcv+0mcWqdS6c34mDbzWZHRxtRGbUPJ5JA@mail.gmail.com>
References: <CAKzROs3Um7Vae47zRQLGMYWLKyw7jL9fuHm+TSABf8X4bmc6VQ@mail.gmail.com>
 <CAKbxfG93ZihgjXs3k-riO5SQc0Sx+H6WF-On4C47PW516fJ9+A@mail.gmail.com>
 <CAKzROs0qx0Qw-HFo5t+_NhmTkA4egs_+KTuvm1MKOsLa+5WTpQ@mail.gmail.com>
 <CALXCt0qeJE+GK6A8fcv+0mcWqdS6c34mDbzWZHRxtRGbUPJ5JA@mail.gmail.com>
Message-ID: <CAKzROs3di=WNu7JpS_j3JCr=EeBkU_WA4RSjnAW6DHuN2TaN5g@mail.gmail.com>

Thanks.  I was assuming it was the ODBC driver since MySQL itself isn't
prone to deadlocks.

We have switched all of our backends to PostgreSQL as of last week.  So far
so good.

On Wed, Jun 13, 2018 at 4:31 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> On Sat, Jun 9, 2018, 02:12 Geoff Mina <gmina at connectfirst.com> wrote:
>
>> Would you guys suggest switching to Postgres on the back-end?  I am
>> guessing if there are known MySQL issues, the suggestion is probably native
>> PGSQL?
>>
>
>
> Problem is with the odbc driver.
>
>  So, short answer is: yes, definitely go with postgresql native support
>
>
> -giovanni
>
>
>
>> On Fri, Jun 8, 2018 at 2:48 PM Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> You would need to do a thread apply all bt to see what the other threads
>>> are doing.  The thread you posted is stuck on a mutex waiting for something
>>> else, probably your DB in another thread.
>>> If you do get that info file it to jira because its easier to deal with
>>> backtraces there.
>>>
>>>
>>> On Fri, Jun 8, 2018 at 12:04 PM, Geoff Mina <gmina at connectfirst.com>
>>> wrote:
>>>
>>>> Greetings,
>>>> Has anyone seen an instance where freeswitch gets hung with the
>>>> following stack trace?  I have about 400 threads in this state and
>>>> eventually FS (1.6.19) just stops responding.
>>>>
>>>> I am running FS against MySQL 5.6.38 on CentOS using ODBC.   Guessing
>>>> if this is known, it probably isn't fixed in 1.6.20... but I can hope.  Any
>>>> input on what a possible solution would be greatly appreciated.  The MySQL
>>>> server and FS are running on the same host connecting via localhost
>>>> interface, so it's not a networking issue.
>>>>
>>>> Thread 445 (Thread 0x7fa2df6db700 (LWP 9395)):
>>>> #0  0x00007fa3712da42d in __lll_lock_wait () from
>>>> /usr/lib64/libpthread.so.0
>>>> #1  0x00007fa3712d5de6 in _L_lock_870 () from /usr/lib64/libpthread.so.0
>>>> #2  0x00007fa3712d5cdf in pthread_mutex_lock () from
>>>> /usr/lib64/libpthread.so.0
>>>> #3  0x00007fa35742468a in sofia_glue_execute_sql_callback () from
>>>> /usr/lib64/freeswitch/mod/mod_sofia.so
>>>> #4  0x00007fa3573df231 in select_from_profile () from
>>>> /usr/lib64/freeswitch/mod/mod_sofia.so
>>>> #5  0x00007fa3573e3b17 in sofia_contact_function () from
>>>> /usr/lib64/freeswitch/mod/mod_sofia.so
>>>> #6  0x00007fa3730594cc in switch_api_execute () from
>>>> /usr/lib64/libfreeswitch.so.1
>>>> #7  0x00007fa372fe69d6 in switch_channel_expand_variables_check () from
>>>> /usr/lib64/libfreeswitch.so.1
>>>> #8  0x00007fa3552d45e4 in user_outgoing_channel () from
>>>> /usr/lib64/freeswitch/mod/mod_dptools.so
>>>> #9  0x00007fa37300b42a in switch_core_session_outgoing_channel () from
>>>> /usr/lib64/libfreeswitch.so.1
>>>> #10 0x00007fa37308dc3b in switch_ivr_originate () from
>>>> /usr/lib64/libfreeswitch.so.1
>>>> #11 0x00007fa3552db0b5 in audio_bridge_function () from
>>>> /usr/lib64/freeswitch/mod/mod_dptools.so
>>>> #12 0x00007fa37300ebcb in switch_core_session_exec () from
>>>> /usr/lib64/libfreeswitch.so.1
>>>> #13 0x00007fa37300f179 in
>>>> switch_core_session_execute_application_get_flags () from
>>>> /usr/lib64/libfreeswitch.so.1
>>>> #14 0x00007fa373012a34 in switch_core_session_run () from
>>>> /usr/lib64/libfreeswitch.so.1
>>>> #15 0x00007fa37300c33e in switch_core_session_thread () from
>>>> /usr/lib64/libfreeswitch.so.1
>>>> #16 0x00007fa373007d83 in switch_core_session_thread_pool_worker ()
>>>> from /usr/lib64/libfreeswitch.so.1
>>>> #17 0x00007fa3732ce7c0 in dummy_worker () from
>>>> /usr/lib64/libfreeswitch.so.1
>>>> #18 0x00007fa3712d3e25 in start_thread () from
>>>> /usr/lib64/libpthread.so.0
>>>> #19 0x00007fa37092e34d in clone () from /usr/lib64/libc.so.6
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Services
>>>> sales at freeswitch.com
>>>> https://freeswitch.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> https://freeswitch.com/oss
>>>> https://freeswitch.org/confluence
>>>> https://cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> https://freeswitch.com
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>> Founder, FreeSWITCH.
>>> http://freeswitch.com
>>>
>>>
>>> https://youtu.be/l_hOxzCt6X4
>>> https://www.youtube.com/watch?v=oAxXgyx5jUw
>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Services
>>> sales at freeswitch.com
>>> https://freeswitch.com
>>>
>>> Official FreeSWITCH Sites
>>> https://freeswitch.com/oss
>>> https://freeswitch.org/confluence
>>> https://cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> https://freeswitch.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180613/db9ce906/attachment-0001.html>

From alex at freeswitch.com  Thu Jun 14 00:06:46 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Thu, 14 Jun 2018 09:06:46 +0900
Subject: [Freeswitch-users] Port numbers
In-Reply-To: <etPan.5b1eecda.17725b58.1dd@magicmail.mooo.com>
References: <etPan.5b1eecda.17725b58.1dd@magicmail.mooo.com>
Message-ID: <CAGJLwFg6YgkrOAyWQh-xiHfE3SLaboL0u21dPCD18fC+udKJKg@mail.gmail.com>

Hi,

Most of the time you need to register against your provider. When you
declare your gateway to provider in the external profile FreeSWITCH
will send REGISTER with 5080 port in contact. Provider will send your
calls to 5080 because of it.

If you don't need to register most probably your provider will ask you
for IP to send calls. Don't forget to include 5080 port in that case.

The whole concept is really nice since enterprise can have some phones
in their LAN  and FreeSWITCH will have only private (rfc1918) ip. In
the meantime 5080 can be forwarded from the internet via firewall and
calls will reach the server without possibility of brute force attacks
to 5060.

Also you can use different contexts for profile and for users.

Regards,

Alex

On 6/12/18, Rick Jarvis <rick at magicmail.mooo.com> wrote:
> Ok kind of FreeSWITCH 101 here I know, but something I’ve never understood
> completely:
>
> A provider wants to send us calls as number at our-ip-address - fine, but it’s
> hitting us on port 5060 (whereas external is set up as 5080). So can’t use
> external, have to use internal profile, right?
>
> So I can create a ACL to allow their IPs onto Internal, ‘inbound’ ACL I
> think?
>
> But how do I handle call routing for inbound calls from provider’s IPs, vs
> my registered users dialling out?
>
> If anyone can simplify this for me that would be great :)
>
> Thanks
> __
> R
>
>


-- 
Alex Sibyakin | Support Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.


From mayamatakeshi at gmail.com  Thu Jun 14 00:29:02 2018
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 14 Jun 2018 09:29:02 +0900
Subject: [Freeswitch-users] Channel Hold Info
Message-ID: <CABaNFCZCDuYsj02NcmJf_zBFaZ3UDMEd9v1nYfMLY_iSvrjsqw@mail.gmail.com>

Hello,
I have a requirement to generate reports of for how long calls were left on
hold.
My original plan was to monitor CHANNEL_HOLD/CHANNEL_UNHOLD events and
update a channel variable preserving hold history.
Then, I happened to find:

https://freeswitch.org/confluence/display/FREESWITCH/CDR#CDR-Howlongwasthecallonhold
and realized there are already channel variables for this kind of purpose
like hold_events, hold_accum_seconds and even hold-record.

However, testing we recent commit:
  FreeSWITCH Version 1.9.0+git~20180516T213111Z~cfd99a03d4~64bit (git
cfd99a0 2018-05-16 21:31:11Z 64bit)

I verified all this data from XML CDR is related to the side that put the
call on hold:

Leg1) Side that put the other on hold:
    <hold_stamp>2018-06-14%2008%3A23%3A18</hold_stamp>
    <hold_events>%7B%7B1528932198592029,1528932199869251%7D%7D</hold_events>
    <last_hold_epoch>1528932198</last_hold_epoch>
    <last_hold_uepoch>1528932198592027</last_hold_uepoch>
    <hold_accum_seconds>1</hold_accum_seconds>
    <hold_accum_usec>1277224</hold_accum_usec>
    <hold_accum_ms>1277</hold_accum_ms>
  <hold-record>
    <hold on="1528932198592029" off="1528932199869251"
bridged-to="1adf353f-a4e0-4e16-8953-fbd8beccca99"></hold>
  </hold-record>
      <last_hold_time>1528932198592027</last_hold_time>
      <hold_accum_time>1277224</hold_accum_time>


Leg2) Side put on hold:
    <last_hold_epoch>0</last_hold_epoch>
    <last_hold_uepoch>0</last_hold_uepoch>
    <hold_accum_seconds>0</hold_accum_seconds>
    <hold_accum_usec>0</hold_accum_usec>
    <hold_accum_ms>0</hold_accum_ms>

      <last_hold_time>0</last_hold_time>
      <hold_accum_time>0</hold_accum_time>


I am curious about this.
Wouldn't it be better to have this kind of hold info at the side that was
put on hold?
Was this missed or done on purpose?
I mean, we can with extra processing of hold-record of all Leg1s to resolve
hold history for Leg2 however having this info at the Leg2 itself would be
much simpler (and so I am considering in using the original
CHANNEL_HOLD/CHANNEL_UNHOLD approach).
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180614/856ff3ca/attachment-0001.html>

From ap at gen-ip.fr  Thu Jun 14 09:46:52 2018
From: ap at gen-ip.fr (Alexis)
Date: Thu, 14 Jun 2018 11:46:52 +0200
Subject: [Freeswitch-users] Port numbers
In-Reply-To: <etPan.5b1eecda.17725b58.1dd@magicmail.mooo.com>
References: <etPan.5b1eecda.17725b58.1dd@magicmail.mooo.com>
Message-ID: <79440300-faa2-e057-f840-7273890d42c5@gen-ip.fr>

Hi,

You can create a new sip_profile on a new IP address (on port 5060) and 
allow only your provider's IP to send you packets on this new IP address 
(with IPTables).
That's the easier solution if you have some free ip addresses :)

Alexis Prodhomme

Le 11/06/2018 à 23:42, Rick Jarvis a écrit :
> Ok kind of FreeSWITCH 101 here I know, but something I’ve never 
> understood completely:
>
> A provider wants to send us calls as number at our-ip-address - fine, but 
> it’s hitting us on port 5060 (whereas external is set up as 5080). So 
> can’t use external, have to use internal profile, right?
>
> So I can create a ACL to allow their IPs onto Internal, ‘inbound’ ACL 
> I think?
>
> But how do I handle call routing for inbound calls from provider’s 
> IPs, vs my registered users dialling out?
>
> If anyone can simplify this for me that would be great :)
>
> Thanks
> __
> R
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180614/a3a9b89c/attachment.html>

From francesco at delagarda.com  Thu Jun 14 14:33:34 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Thu, 14 Jun 2018 16:33:34 +0200
Subject: [Freeswitch-users] number occupied
Message-ID: <02a401d403ec$ae040050$0a0c00f0$@delagarda.com>

I'm scripting in javascript

 

How can I tell if the numer I called was occupied / busy?

 

session.answer();

var sessOut = new Session(trunk + dialedNum);

if (session.ready()) {

///  all ok

}

// WHAT IF dialed num was busy or occupied ??

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180614/33f2a9bf/attachment.html>

From david.villasmil.work at gmail.com  Wed Jun 13 21:51:01 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 13 Jun 2018 23:51:01 +0200
Subject: [Freeswitch-users] wiki and org sites
In-Reply-To: <23DB1380-EDF9-479E-86C4-8B131D83AD72@mgtech.com>
References: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>
 <E785B568-3F86-4B2D-90E3-7771E417FCD3@jerris.com>
 <CAFGRPVpOxHeVtE95CCqb3VmK51_H1N875a9EKRqQF1kgWbVxVA@mail.gmail.com>
 <A7460BF3-DE88-4C9F-B5A8-3033DCB7A3BC@jerris.com>
 <CAK6zN=0C8OFK_35fwVjLMcWyqQWP1vJrDW3i_uZesrTDvS1UGA@mail.gmail.com>
 <23DB1380-EDF9-479E-86C4-8B131D83AD72@mgtech.com>
Message-ID: <CAFGRPVr-RZsNFgAu18pJLE2y8__z5Lo1HUB5mCeasj+tefL+bg@mail.gmail.com>

Nice! Great job! Many thanks!

On Wed, Jun 13, 2018, 23:49 Mario <mario_fs at mgtech.com> wrote:

> Way back when the conversion started I offered to scan the data using Rexx
> and produce a report of links to old pages but no one ever got back to me.
> I think a report like that would have reduced/eliminated this problem by
> now.
>
> BTW, I fixed most of the pages you referenced below:
> https://freeswitch.org/confluence/display/FREESWITCH/Early+Media
> https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
> https://freeswitch.org/confluence/display/FREESWITCH/Audio+Codecs
> https://freeswitch.org/confluence/display/FREESWITCH/IVR.   <- except for
> voicemail.py which I could not find
> https://freeswitch.org/confluence/display/FREESWITCH/Dialplan.   <— this
> was already correct
> https://freeswitch.org/confluence/display/FREESWITCH/mod_v8.   <- two
>  links removed since pages combined with others
>
> Mario G
>
>
> On Jun 8, 2018, at 11:42 AM, Preston Hagar <prestonh at gmail.com> wrote:
>
> A lot of the internal confluence links point back to the wiki.  Here are
> some pages I've run across that have a lot of links back to the wiki.
>
> Freeswitch IVR Originate at bottom of page here
>
> https://freeswitch.org/confluence/display/FREESWITCH/Dialplan.   <— this
> was already correct
>
>
> Almost all the links on the page here:
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_v8
>
> and here:
>
> https://freeswitch.org/confluence/display/FREESWITCH/JavaScript
>
> https://freeswitch.org/confluence/display/FREESWITCH/JavaScript+Event
>
> https://freeswitch.org/confluence/display/FREESWITCH/IVR
>
> https://freeswitch.org/confluence/display/FREESWITCH/Audio+Codecs
>
> https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
>
> https://freeswitch.org/confluence/display/FREESWITCH/Early+Media
>
> On Thu, Jun 7, 2018 at 2:44 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> If there is anything still pointing there it needs to be fixed.
>>
>> On Jun 7, 2018, at 2:26 PM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>> Yeah, I remember that afterwards... it's just that LOTS of things are
>> pointing there...
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
>> ᐧ
>>
>> On Thu, Jun 7, 2018 at 8:25 PM Michael Jerris <mike at jerris.com> wrote:
>>
>>> wiki.freeswitch.org has been deprecated for many years now.  The
>>> correct location for the wiki is https://freeswitch.org/confluence
>>>
>>> Mike
>>>
>>> On Jun 6, 2018, at 3:02 PM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>> I can't connect to wiki.freeswitch.org, is the web down? I just get
>>> redirected to freeswitch's (new?) commercial web page (freeswitch.com)
>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180613/1086c622/attachment-0001.html>

From david.villasmil.work at gmail.com  Thu Jun 14 00:16:18 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 14 Jun 2018 02:16:18 +0200
Subject: [Freeswitch-users] Port numbers
In-Reply-To: <etPan.5b1eecda.17725b58.1dd@magicmail.mooo.com>
References: <etPan.5b1eecda.17725b58.1dd@magicmail.mooo.com>
Message-ID: <CAFGRPVq8v==fP2b7mS_h84z20ix8UZzbumhysh+TS8h0qkOTEg@mail.gmail.com>

No, you can just remove the internal profile and change the port of the
external to use port 5080.
Take a look at internal.xml and external.xml you will see the ports in use
and can change them appropiately. Be sure to close up freeswitch properly
with a firewall or iptables or with acl.

Hope that helps.

On Thu, Jun 14, 2018, 00:49 Rick Jarvis <rick at magicmail.mooo.com> wrote:

> Ok kind of FreeSWITCH 101 here I know, but something I’ve never understood
> completely:
>
> A provider wants to send us calls as number at our-ip-address - fine, but
> it’s hitting us on port 5060 (whereas external is set up as 5080). So can’t
> use external, have to use internal profile, right?
>
> So I can create a ACL to allow their IPs onto Internal, ‘inbound’ ACL I
> think?
>
> But how do I handle call routing for inbound calls from provider’s IPs, vs
> my registered users dialling out?
>
> If anyone can simplify this for me that would be great :)
>
> Thanks
> __
> R
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180614/1c376b1b/attachment-0001.html>

From david.villasmil.work at gmail.com  Thu Jun 14 01:50:02 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 14 Jun 2018 03:50:02 +0200
Subject: [Freeswitch-users] python script not running properly
Message-ID: <CAFGRPVq=R1JVpuEM=nHMf8NH91vA3G-nWc=RU6PNu2Ku86Lt=w@mail.gmail.com>

Hello Guys,

I'm writing a script in python which when run via cli works properly. I get
some dta via http (urllib3) and parse it like:

r = http.request('GET', '
https://www.massphoning.net/customer/inMassphoning/getInCampaignInfo?phone='
+ caller)

if r.status == 200:
httpResult = json.loads(r.data.decode('utf-8'))

Then print it like:
print('Response Parsed: %s\n' % CampaignData)
print('CamapignType: %s\n' % CampaignData['campaing_type'])

and via the CLI works nicely.

But when run on freeswitch dialplan
print('Response Parsed: %s\n' % CampaignData)

runs properly and prints on the log the data, but it fails on:
print('CamapignType: %s\n' % CampaignData['campaing_type'])

with:
2018-06-14 03:33:28.971179 [ERR] mod_python.c:164 Python Error by calling
script "foo": <type 'exceptions.TypeError'>
Message: in method 'consoleLog', argument 2 of type 'char *'
Exception: None

Traceback (most recent call last)
File: "/opt/freeswitch/share/freeswitch/scripts/foo.py", line 16, in handler
File: "/opt/freeswitch/share/freeswitch/scripts/foo.py", line 69, in
getCampaignInfo

Any ideas??


Thanks!
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
ᐧ
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180614/7ba19c08/attachment-0001.html>

From prestonh at gmail.com  Thu Jun 14 21:09:27 2018
From: prestonh at gmail.com (Preston Hagar)
Date: Thu, 14 Jun 2018 16:09:27 -0500
Subject: [Freeswitch-users] wiki and org sites
In-Reply-To: <23DB1380-EDF9-479E-86C4-8B131D83AD72@mgtech.com>
References: <CAFGRPVr6NiRr1SPDD+vnZ7rstLQLyfPJcmx9gDCYM_-EO+TsDg@mail.gmail.com>
 <E785B568-3F86-4B2D-90E3-7771E417FCD3@jerris.com>
 <CAFGRPVpOxHeVtE95CCqb3VmK51_H1N875a9EKRqQF1kgWbVxVA@mail.gmail.com>
 <A7460BF3-DE88-4C9F-B5A8-3033DCB7A3BC@jerris.com>
 <CAK6zN=0C8OFK_35fwVjLMcWyqQWP1vJrDW3i_uZesrTDvS1UGA@mail.gmail.com>
 <23DB1380-EDF9-479E-86C4-8B131D83AD72@mgtech.com>
Message-ID: <CAK6zN=0N+6nusKGr2gQ=w_SgW0UQLzXfr8wY-a3kTEcfuvoV+g@mail.gmail.com>

Awesome, thank you!  I agree, some sort of automated scan seem to be the
best solution.  I just know I had been through those pages recently and got
frustrated :)  Thanks again!

On Tue, Jun 12, 2018 at 12:38 PM, Mario <mario_fs at mgtech.com> wrote:

> Way back when the conversion started I offered to scan the data using Rexx
> and produce a report of links to old pages but no one ever got back to me.
> I think a report like that would have reduced/eliminated this problem by
> now.
>
> BTW, I fixed most of the pages you referenced below:
> https://freeswitch.org/confluence/display/FREESWITCH/Early+Media
> https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
> https://freeswitch.org/confluence/display/FREESWITCH/Audio+Codecs
> https://freeswitch.org/confluence/display/FREESWITCH/IVR.   <- except for
> voicemail.py which I could not find
> https://freeswitch.org/confluence/display/FREESWITCH/Dialplan.   <— this
> was already correct
> https://freeswitch.org/confluence/display/FREESWITCH/mod_v8.   <- two
>  links removed since pages combined with others
>
> Mario G
>
>
> On Jun 8, 2018, at 11:42 AM, Preston Hagar <prestonh at gmail.com> wrote:
>
> A lot of the internal confluence links point back to the wiki.  Here are
> some pages I've run across that have a lot of links back to the wiki.
>
> Freeswitch IVR Originate at bottom of page here
>
> https://freeswitch.org/confluence/display/FREESWITCH/Dialplan.   <— this
> was already correct
>
> Almost all the links on the page here:
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_v8
>
> and here:
>
> https://freeswitch.org/confluence/display/FREESWITCH/JavaScript
>
> https://freeswitch.org/confluence/display/FREESWITCH/JavaScript+Event
>
> https://freeswitch.org/confluence/display/FREESWITCH/IVR
>
> https://freeswitch.org/confluence/display/FREESWITCH/Audio+Codecs
>
> https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
>
> https://freeswitch.org/confluence/display/FREESWITCH/Early+Media
>
> On Thu, Jun 7, 2018 at 2:44 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> If there is anything still pointing there it needs to be fixed.
>>
>> On Jun 7, 2018, at 2:26 PM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>> Yeah, I remember that afterwards... it's just that LOTS of things are
>> pointing there...
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
>> ᐧ
>>
>> On Thu, Jun 7, 2018 at 8:25 PM Michael Jerris <mike at jerris.com> wrote:
>>
>>> wiki.freeswitch.org has been deprecated for many years now.  The
>>> correct location for the wiki is https://freeswitch.org/confluence
>>>
>>> Mike
>>>
>>> On Jun 6, 2018, at 3:02 PM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>> I can't connect to wiki.freeswitch.org, is the web down? I just get
>>> redirected to freeswitch's (new?) commercial web page (freeswitch.com)
>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180614/10fdcea0/attachment-0001.html>

From joel at textplus.com  Fri Jun 15 02:25:55 2018
From: joel at textplus.com (Joel Serrano)
Date: Thu, 14 Jun 2018 19:25:55 -0700
Subject: [Freeswitch-users] python script not running properly
In-Reply-To: <CAFGRPVq=R1JVpuEM=nHMf8NH91vA3G-nWc=RU6PNu2Ku86Lt=w@mail.gmail.com>
References: <CAFGRPVq=R1JVpuEM=nHMf8NH91vA3G-nWc=RU6PNu2Ku86Lt=w@mail.gmail.com>
Message-ID: <CAMtXxQnj4r1VX9aNisbXRh=hW3AAEMVLP7G8s3HqmtuST9fhPg@mail.gmail.com>

Hi David,

If that happened to me I would run something like this:

print('Test 1: %s\n' % CampaignData)
print('Test 2: %s\n' % type(CampaignData['campaing_type']))
print('Test 3: %s\n' % str(CampaignData['campaing_type']))


Can you share the output?
Joel.


On Wed, Jun 13, 2018 at 6:50 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello Guys,
>
> I'm writing a script in python which when run via cli works properly. I
> get some dta via http (urllib3) and parse it like:
>
> r = http.request('GET', 'https://www.massphoning.net/
> customer/inMassphoning/getInCampaignInfo?phone=' + caller)
>
> if r.status == 200:
> httpResult = json.loads(r.data.decode('utf-8'))
>
> Then print it like:
> print('Response Parsed: %s\n' % CampaignData)
> print('CamapignType: %s\n' % CampaignData['campaing_type'])
>
> and via the CLI works nicely.
>
> But when run on freeswitch dialplan
> print('Response Parsed: %s\n' % CampaignData)
>
> runs properly and prints on the log the data, but it fails on:
> print('CamapignType: %s\n' % CampaignData['campaing_type'])
>
> with:
> 2018-06-14 03:33:28.971179 [ERR] mod_python.c:164 Python Error by calling
> script "foo": <type 'exceptions.TypeError'>
> Message: in method 'consoleLog', argument 2 of type 'char *'
> Exception: None
>
> Traceback (most recent call last)
> File: "/opt/freeswitch/share/freeswitch/scripts/foo.py", line 16, in
> handler
> File: "/opt/freeswitch/share/freeswitch/scripts/foo.py", line 69, in
> getCampaignInfo
>
> Any ideas??
>
>
> Thanks!
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> ᐧ
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180614/c83e7b28/attachment-0001.html>

From chad at apartmentlines.com  Fri Jun 15 18:50:55 2018
From: chad at apartmentlines.com (Chad Phillips)
Date: Fri, 15 Jun 2018 11:50:55 -0700
Subject: [Freeswitch-users] Using H264 with Verto
Message-ID: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>

I have a slightly older install where I’m using H264 with Verto, and it
works fine. However, I’m not able to get it working on latest master.

I’m guessing there’s some esoteric setting I’m missing, but can’t seem to
find it. I do have the following in verto.conf.xml:

      <param name="outbound-codec-string" value="opus,h264"/>
      <param name=“inbound-codec-string” value="opus,h264"/>

I also have mod_h26x disabled, as I remember that’s passthrough only and
causes issues when loaded in this case.

When I try in Chrome I see it offering H264 in the SDP, but the answer SDP
contains this, and the stream is audio only:

m=video 0 UDP/TLS/RTP/SAVPF 19

What am I missing??
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180615/eff5048e/attachment-0001.html>

From hggh96.hg at gmail.com  Mon Jun 18 05:14:05 2018
From: hggh96.hg at gmail.com (Hossein Gholizadeh)
Date: Mon, 18 Jun 2018 09:44:05 +0430
Subject: [Freeswitch-users] sip.js has no audio
Message-ID: <CAA7fES+fCWP4VWU70FBArs=MNi=s6BoLm_8ErR1bAkBGEY1KYg@mail.gmail.com>

hi all ...  i'm using sip.js for my IVR project ... it could connect to
freeswitch correctly but there is no audio between call legs ... first of
all i have to say i really cannot use Verto . this is my function of when
call button pressed

 * phoneCallButtonPressed : function(sessionid) {*

*            var s      = ctxSip.Sessions[sessionid],*
*                target = $("#numDisplay").val();*

*            if (!s) {*

*                $("#numDisplay").val("");*
*                ctxSip.sipCall(target);*

*            } else if (s.accept && !s.startTime) {*

*                s.accept({*
*                    media : {*
*                        stream      : ctxSip.Stream,*
*                        constraints : { audio : true, video : false },*
*                        render      : {*
*                            remote : { audio: $('#audioRemote').get()[0] }*
*                        },*
*                        RTCConstraints : { "optional": [{
'DtlsSrtpKeyAgreement': 'true'} ]}*
*                    }*
*                });*
*            }*
*        },*

-----------------------------------------------------------------------------------------
also i have to say that i tested it between my phone and x-lite ... X-lite
could receive the voice but my phone could'nt ... i was using Zoiper in my
phone.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/0cfdda44/attachment-0001.html>

From darshanmody at avaya.com  Mon Jun 18 06:17:56 2018
From: darshanmody at avaya.com (Mody, Darshan (Darshan))
Date: Mon, 18 Jun 2018 06:17:56 +0000
Subject: [Freeswitch-users] Memory leak while using BGAPI
Message-ID: <25D2EC755404B4409F263AC6D050FEBB2BCCD8C4@AZ-FFEXMB03.global.avaya.com>

Hi,

We are observing considerable memory leak while using LUA and BGAPI of mod_command.c. Below is the valgrind's output

==21863== 123,170 (45,496 direct, 77,674 indirect) bytes in 517 blocks are definitely lost in loss record 2,370 of 2,398
==21863==    at 0x4C29C23: malloc (vg_replace_malloc.c:299)
==21863==    by 0x548707C: switch_event_create_subclass_detailed (switch_event.c:736)
==21863==    by 0x11631389: bgapi_exec (mod_commands.c:5151)
==21863==    by 0x56EEE8F: dummy_worker (thread.c:151)
==21863==    by 0x7677E24: start_thread (in /usr/lib64/libpthread-2.17.so)
==21863==    by 0x805734C: clone (in /usr/lib64/libc-2.17.so

==21863== 143,478 (18,392 direct, 125,086 indirect) bytes in 209 blocks are definitely lost in loss record 2,374 of 2,398
==21863==    at 0x4C29C23: malloc (vg_replace_malloc.c:299)
==21863==    by 0x548707C: switch_event_create_subclass_detailed (switch_event.c:736)
==21863==    by 0x542AADE: switch_core_session_exec (switch_core_session.c:2874)
==21863==    by 0x542B108: switch_core_session_execute_application_get_flags (switch_core_session.c:2742)
==21863==    by 0x5510C10: CoreSession::execute(char const*, char const*) (switch_cpp.cpp:777)
==21863==    by 0x16FED72C: ??? (mod_lua_wrap.cpp:7289)
==21863==    by 0x17217323: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x17221E56: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x1721774C: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x17216A6D: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x172178D9: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x1721344C: lua_pcall (in /usr/lib64/liblua-5.1.so)

Has some one also observed behavior while using BGAPI?

Thanks
Darshan


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/01d2e53c/attachment.html>

From andrew.keil at visytel.com  Mon Jun 18 06:23:54 2018
From: andrew.keil at visytel.com (Andrew Keil)
Date: Mon, 18 Jun 2018 06:23:54 +0000
Subject: [Freeswitch-users] Recording skipping silence on outbound call
	recording
Message-ID: <MEXPR01MB1223E2FF1CA0FE0D1D6DA4FAF3710@MEXPR01MB1223.ausprd01.prod.outlook.com>

To FreeSWITCH Users,

I am attempting to make a recording of an outbound call using the originate API call which starts a Lua script which then simply records to disk exactly what it hears.

I just noticed an interesting issue (or feature) where the recording skips any silence.  What I am aiming for is for no silence to be skipped inside the recording, if someone can provide some settings for me to try I would be more than happy to experiment further.

The settings for the Lua session:recordFile are length of recording: 60 seconds; silence threshold: 300; silence seconds: 60 seconds  {essentially termination on silence is disabled since the silence seconds = recording seconds}

I also tried with silence threshold = 0 and silence is still skipped inside the recording.

I also tried the setting inside the appropriate profile: <param name="suppress-cng" value="true"/> and this made no difference, silence is still skipped inside the recording.

My setup is one single FreeSWITCH 1.6.20 (production version) box with a simple internal gateway setup to land on the same FreeSWITCH IVR (just for testing).

API command: originate {origination_uuid=121e20a6-1245-439a-8527-53efab009334-out-1,origination_caller_id_number=FLOODOUT,call_timeout=15,ignore_early_media=true,return_ring_ready=true,suppress_cng=true}sofia/gateway/visytel-pc-ivr/01111111111 FLOODOUT

The SDP inside my log is similar to this for the outbound and inbound (Note: The outbound call is performing the recording):

121e20a6-1245-439a-8527-53efab009334-out-1 Local SDP:
121e20a6-1245-439a-8527-53efab009334-out-1 v=0
121e20a6-1245-439a-8527-53efab009334-out-1 o=FreeSWITCH 1529274701 1529274702 IN IP4 192.168.15.15
121e20a6-1245-439a-8527-53efab009334-out-1 s=FreeSWITCH
121e20a6-1245-439a-8527-53efab009334-out-1 c=IN IP4 192.168.15.15
121e20a6-1245-439a-8527-53efab009334-out-1 t=0 0
121e20a6-1245-439a-8527-53efab009334-out-1 m=audio 25994 RTP/AVP 9 8 0 101
121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:9 G722/8000
121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:8 PCMA/8000
121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:0 PCMU/8000
121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:101 telephone-event/8000
121e20a6-1245-439a-8527-53efab009334-out-1 a=fmtp:101 0-16
121e20a6-1245-439a-8527-53efab009334-out-1 a=silenceSupp:off - - - -
121e20a6-1245-439a-8527-53efab009334-out-1 a=ptime:20
121e20a6-1245-439a-8527-53efab009334-out-1 a=sendrecv

6da97c7c-1052-4f78-8dde-83901147a0c1 2018-06-18 15:44:55.596296 [DEBUG] sofia.c:7094 Remote SDP:
6da97c7c-1052-4f78-8dde-83901147a0c1 v=0
6da97c7c-1052-4f78-8dde-83901147a0c1 o=FreeSWITCH 1529274701 1529274702 IN IP4 192.168.15.15
6da97c7c-1052-4f78-8dde-83901147a0c1 s=FreeSWITCH
6da97c7c-1052-4f78-8dde-83901147a0c1 c=IN IP4 192.168.15.15
6da97c7c-1052-4f78-8dde-83901147a0c1 t=0 0
6da97c7c-1052-4f78-8dde-83901147a0c1 m=audio 25994 RTP/AVP 9 8 0 101
6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:9 G722/8000
6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:8 PCMA/8000
6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:0 PCMU/8000
6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:101 telephone-event/8000
6da97c7c-1052-4f78-8dde-83901147a0c1 a=fmtp:101 0-16
6da97c7c-1052-4f78-8dde-83901147a0c1 a=silenceSupp:off - - - -
6da97c7c-1052-4f78-8dde-83901147a0c1 a=ptime:20
6da97c7c-1052-4f78-8dde-83901147a0c1

Kind Regards,

Andrew Keil

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/40ffbaa0/attachment.html>

From francesco at delagarda.com  Sat Jun 16 12:34:27 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Sat, 16 Jun 2018 14:34:27 +0200
Subject: [Freeswitch-users] hangupCause()
Message-ID: <048501d4056e$5f771c50$1e6554f0$@delagarda.com>

I've been trying to the hangupCause() in javascript.

 

I saw that in LUA its

local hcause = session1:hangupCause();

 

is there NO session1.hangupCause()  in javascript??

 

 

Thanks

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180616/1ef8fd41/attachment-0001.html>

From francesco at delagarda.com  Sat Jun 16 12:42:14 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Sat, 16 Jun 2018 14:42:14 +0200
Subject: [Freeswitch-users] causecode - please help
Message-ID: <049201d4056f$750c5c00$5f251400$@delagarda.com>

I am trying the undestand the hangup reason in javascript

I noticed there is no 

session.hangupCause

but there IS a

                session.causecode

 

Docs say it gives the Q.931 code for hangup,

 

But, on occupied calls, I just get "false" when I ask

session.causecode

 

What's happening..

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180616/1782ac8d/attachment.html>

From francesco at delagarda.com  Sun Jun 17 09:12:29 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Sun, 17 Jun 2018 11:12:29 +0200
Subject: [Freeswitch-users] session.cause and session.causecode
Message-ID: <056701d4061b$521ec960$f65c5c20$@delagarda.com>

'Morning all

I can't get session.cause and session.causecode to work.

All I get is false for the code and blank for the cause!

 

Im running FS 1.6.20

 

This is a simplified version of my JS script

 

if (session.ready()) {

 

    session.answer();

 

    var dialedNum = "3332094333"; 

                console_log("notice", "*********************** DIALING");

                var sessOut = new Session("sofia/gateway/realtoneFXO/" +
dialedNum + "@192.168.0.216:5060");

                console_log("notice", "*********************** WAITING FOR
SESSION READY ..");

 

                

                if (sessOut.ready()) {

               console_log("notice", "*********************** SESSION
READY");

               bridge(session, sessOut);

                } else {

                               console_log("notice",
"************************* disconnect cause:" + sessOut.cause + ":" +
sessOut.causeCode);

                }

                sessOut.hangup();

                session.hangup();

    console_log("notice", "*********************** CALL ENDED");

 

}

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180617/04c1a01b/attachment.html>

From francesco at delagarda.com  Sun Jun 17 09:23:51 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Sun, 17 Jun 2018 11:23:51 +0200
Subject: [Freeswitch-users] session.cause and session.causecode
Message-ID: <057401d4061c$e8766070$b9632150$@delagarda.com>

This is what the log says:

2018-06-17 11:18:41.186422 [ALERT] stuff.js:74 ***********************
DIALING

2018-06-17 11:18:41.186422 [NOTICE] switch_channel.c:1104 New Channel
sofia/external/065551234 [411ac79d-e8e0-4f77-b51f-7da24a71ea32]

2018-06-17 11:18:45.606424 [NOTICE] sofia.c:8273 Hangup
sofia/external/065551234 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER]

2018-06-17 11:18:45.626430 [NOTICE] switch_core_session.c:1683 Session 915
(sofia/external/065551234) Ended

2018-06-17 11:18:45.626430 [NOTICE] switch_core_session.c:1687 Close Channel
sofia/external/065551234 [CS_DESTROY]

2018-06-17 11:18:45.626430 [ALERT] stuff.js:74 ***********************
WAITING FOR SESSION READY ..

2018-06-17 11:18:45.626430 [ALERT] stuff.js:74 ***********************
DISCONNECT CAUSE :false:undefined

 

So this is what I want:

2018-06-17 11:18:45.606424 [NOTICE] sofia.c:8273 Hangup
sofia/external/065551234 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER]

 

So why do I get:

False: undefined 

from

console_log("notice", "************************* disconnect cause:" +
sessOut.cause + ":" + sessOut.causeCode);

 

 

From: Francesco Facco de Lagarda <francesco at delagarda.com> 
Sent: domenica 17 giugno 2018 11:12
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: session.cause and session.causecode

 

'Morning all

I can't get session.cause and session.causecode to work.

All I get is false for the code and blank for the cause!

 

Im running FS 1.6.20

 

This is a simplified version of my JS script

 

if (session.ready()) {

 

    session.answer();

 

    var dialedNum = "3332094333"; 

                console_log("notice", "*********************** DIALING");

                var sessOut = new Session("sofia/gateway/realtoneFXO/" +
dialedNum + "@192.168.0.216:5060");

                console_log("notice", "*********************** WAITING FOR
SESSION READY ..");

 

                

                if (sessOut.ready()) {

               console_log("notice", "*********************** SESSION
READY");

               bridge(session, sessOut);

                } else {

                               console_log("notice",
"************************* disconnect cause:" + sessOut.cause + ":" +
sessOut.causeCode);

                }

                sessOut.hangup();

                session.hangup();

    console_log("notice", "*********************** CALL ENDED");

 

}

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180617/ae0bc2db/attachment.html>

From mjlopez at smartic.es  Mon Jun 18 11:51:35 2018
From: mjlopez at smartic.es (=?iso-8859-1?Q?Miguel_Jes=FAs_L=F3pez_Valverde?=)
Date: Mon, 18 Jun 2018 13:51:35 +0200
Subject: [Freeswitch-users] Installing Freeswitch 1.8 on testing mode.
Message-ID: <00e801d406fa$b5f6ad60$21e40820$@smartic.es>

Hello,

 

                Is it possible to complete a Freeswitch 1.8 installation on
Debian 9, as seen in the comments on the link
https://freeswitch.org/confluence/display/FREESWITCH/Debian+9+Stretch ?, Do
you know if there is a recipe for it , even in testing mode ?.

 

On the other hand, does anyone know if there is any date or estimate for the
release of Freeswitch 1.8?

 

Thank you very much.

 

Miguel J. López.

 



---
El software de antivirus Avast ha analizado este correo electrónico en busca de virus.
https://www.avast.com/antivirus
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/57180176/attachment-0001.html>

From matt at supportedbusiness.com  Mon Jun 18 13:32:49 2018
From: matt at supportedbusiness.com (Matt Broad)
Date: Mon, 18 Jun 2018 14:32:49 +0100
Subject: [Freeswitch-users] causecode - please help
In-Reply-To: <049201d4056f$750c5c00$5f251400$@delagarda.com>
References: <049201d4056f$750c5c00$5f251400$@delagarda.com>
Message-ID: <CAEeV61j2ZJp9XeODVNs5KS62r8aF+Z1b1UfagzQw1g8eaEkhtg@mail.gmail.com>

I believe that causecode is false until the call hangsup, you may want to
check originateCause when causecode == false something like below;


if(session.causecode == false)
 {
      this.ogCallCauseCode = session.originateCause;
  }
 else
  {
      this.ogCallCauseCode = session.causecode;
  }

thanks
Matt

Matt Broad
Tel: +44 (0)203 011 1313 <+44%2020%203011%201313>
Web: www.supportedbusiness.com

The content of this email, including any files or documents attached, are
confidential and may also be legally privileged, protected from disclosure
and/or protected by other legal rules. It is written without prejudice and
is intended for the individual specified in the message only. The views and
opinions included in this email belong to their author and do not
necessarily mirror the views and opinions of the company. Full security of
this email cannot be ensured as, despite our best efforts, the data
included in emails could be infected, intercepted, or corrupted. Do not
share any part of this message with any third party, without a written
consent of the sender. If you have received this message in error, please
notify us and remove it from your system.

On 16 June 2018 at 13:42, Francesco Facco de Lagarda <
francesco at delagarda.com> wrote:

> I am trying the undestand the hangup reason in javascript
>
> I noticed there is no
>
> session.hangupCause
>
> but there IS a
>
>                 session.causecode
>
>
>
> Docs say it gives the Q.931 code for hangup,
>
>
>
> But, on occupied calls, I just get “false” when I ask
>
> session.causecode
>
>
>
> What’s happening..
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/d3d61ad5/attachment.html>

From matt at supportedbusiness.com  Mon Jun 18 13:45:19 2018
From: matt at supportedbusiness.com (Matt Broad)
Date: Mon, 18 Jun 2018 14:45:19 +0100
Subject: [Freeswitch-users] session.cause and session.causecode
In-Reply-To: <056701d4061b$521ec960$f65c5c20$@delagarda.com>
References: <056701d4061b$521ec960$f65c5c20$@delagarda.com>
Message-ID: <CAEeV61hkpie1GZWaspccjnCu5F=4KKORYtFCKxHhVyLcjCv4aA@mail.gmail.com>

sessOut.causeCode should be sessOut.causecode

you can get the sessOut object values by adding:

console_log("notice", JSON.stringify(sessOut) );



thanks
Matt

Matt Broad
Tel: +44 (0)203 011 1313 <+44%2020%203011%201313>
Web: www.supportedbusiness.com

The content of this email, including any files or documents attached, are
confidential and may also be legally privileged, protected from disclosure
and/or protected by other legal rules. It is written without prejudice and
is intended for the individual specified in the message only. The views and
opinions included in this email belong to their author and do not
necessarily mirror the views and opinions of the company. Full security of
this email cannot be ensured as, despite our best efforts, the data
included in emails could be infected, intercepted, or corrupted. Do not
share any part of this message with any third party, without a written
consent of the sender. If you have received this message in error, please
notify us and remove it from your system.

On 17 June 2018 at 10:12, Francesco Facco de Lagarda <
francesco at delagarda.com> wrote:

> ‘Morning all
>
> I can’t get session.cause and session.causecode to work.
>
> All I get is false for the code and blank for the cause!
>
>
>
> Im running FS 1.6.20
>
>
>
> This is a simplified version of my JS script
>
>
>
> if (session.ready()) {
>
>
>
>     session.answer();
>
>
>
>     var dialedNum = "3332094333";
>
>                 console_log("notice", "*********************** DIALING");
>
>                 var sessOut = new Session("sofia/gateway/realtoneFXO/" +
> dialedNum + "@192.168.0.216:5060");
>
>                 console_log("notice", "*********************** WAITING FOR
> SESSION READY ..");
>
>
>
>
>
>                 if (sessOut.ready()) {
>
>                console_log("notice", "*********************** SESSION
> READY");
>
>                bridge(session, sessOut);
>
>                 } else {
>
>                                console_log("notice",
> "************************* disconnect cause:" + sessOut.cause + ":" +
> sessOut.causeCode);
>
>                 }
>
>                 sessOut.hangup();
>
>                 session.hangup();
>
>     console_log("notice", "*********************** CALL ENDED");
>
>
>
> }
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/e16d8df0/attachment.html>

From gmaruzz at gmail.com  Mon Jun 18 14:02:38 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 18 Jun 2018 16:02:38 +0200
Subject: [Freeswitch-users] sip.js has no audio
In-Reply-To: <CAA7fES+fCWP4VWU70FBArs=MNi=s6BoLm_8ErR1bAkBGEY1KYg@mail.gmail.com>
References: <CAA7fES+fCWP4VWU70FBArs=MNi=s6BoLm_8ErR1bAkBGEY1KYg@mail.gmail.com>
Message-ID: <CALXCt0rihexsvUB8GLtverDems4JhcEi6FhZk-GpZoVcsbqSBA@mail.gmail.com>

first of all reinstall freeswitch from scratch, usiing the instruction in
https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie ,
without the minimum change

then check if calls are working between two softphones, on two different
machines

after all is working then do the fancy things.

you may want to give attention at external_rtp_ip and external_sip_ip
values in "external" sip profile

anyway, first you must have calls in local LAN working

then calls from other, more complex, scenarios.

you may find the freeswitch books, easily available through the internet,
useful as well as confluence documentation.

-giovanni


On 18 June 2018 at 07:14, Hossein Gholizadeh <hggh96.hg at gmail.com> wrote:

> hi all ...  i'm using sip.js for my IVR project ... it could connect to
> freeswitch correctly but there is no audio between call legs ... first of
> all i have to say i really cannot use Verto . this is my function of when
> call button pressed
>
>  * phoneCallButtonPressed : function(sessionid) {*
>
> *            var s      = ctxSip.Sessions[sessionid],*
> *                target = $("#numDisplay").val();*
>
> *            if (!s) {*
>
> *                $("#numDisplay").val("");*
> *                ctxSip.sipCall(target);*
>
> *            } else if (s.accept && !s.startTime) {*
>
> *                s.accept({*
> *                    media : {*
> *                        stream      : ctxSip.Stream,*
> *                        constraints : { audio : true, video : false },*
> *                        render      : {*
> *                            remote : { audio: $('#audioRemote').get()[0]
> }*
> *                        },*
> *                        RTCConstraints : { "optional": [{
> 'DtlsSrtpKeyAgreement': 'true'} ]}*
> *                    }*
> *                });*
> *            }*
> *        },*
>
> ------------------------------------------------------------
> -----------------------------
> also i have to say that i tested it between my phone and x-lite ... X-lite
> could receive the voice but my phone could'nt ... i was using Zoiper in my
> phone.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/499635c4/attachment-0001.html>

From f.antonini at tiesse.com  Mon Jun 18 15:43:57 2018
From: f.antonini at tiesse.com (fabio)
Date: Mon, 18 Jun 2018 17:43:57 +0200
Subject: [Freeswitch-users] Problem loading mod_lua.so
Message-ID: <3d54a27e-196a-6e61-ca0f-c7a27875f82f@tiesse.com>

Hi everybody

I'm facing a strange problem when I configure mod_lua in my OpenWrt 
framework. Freeswitch version is 1.6.20 and OpenWrt is an 
'attitude_adjustment'.


I added 'mod_lua' to the list of modules to be loaded ( "<load 
module="mod_lua"/>" in the file 
/etc/freeswitch/autoload_configs/modules.conf.xml). The mod_lua is 
successfully compiled, and at the boot FS tries to load it but 
unfortunately it complains with the following message:


*2018-06-18 15:29:17.395100 [CRIT] switch_loadable_module.c:1448 Error 
Loading module /usr/lib/freeswitch/mod_lua.so**
****Unknown error***


The file is placed in that directory with all the other FS modules.


I cannot understand where is problem.


Any help or advice will greatly appreciated.


Thanks in advance


Best regards


fabio

-- 

Fabio Antonini

/Software Engineer (Ph.D)/
f.antonini at tiesse.com <mailto:f.antonini at tiesse.com>
*Tel* +39.0863.455830
*Mob* +39.393.9261941
*Fax* +39.0863.455830
Via Corradini 80
67051 Avezzano (AQ)

	

Logo Tiesse dal 1998 al 2018, vent'anni di Innovazione Made in Italy. 
Clicca per visitare il sito Tiesse <http://www.tiesse.com/>

*Tiesse S.p.A.* - www.tiesse.com <http://www.tiesse.com/>
Via Asti 4, 10015 Ivrea (TO)
Pagina Tiesse su Linkedin, clicca e visitaci 
<https://linkedin.com/company/tiesse-spa>

*Disclaimer:* il contenuto di questa email è riservato e non vincolante 
per Tiesse S.p.A.. Se lo avesse ricevuto per errore, la preghiamo di 
segnalarlo immediatamente al mittente, di non utilizzare e divulgare il 
contenuto e di distruggere ogni copia in suo possesso. Tiesse S.p.A. 
declina ogni responsabilità da qualsiasi conseguenza derivante da 
utilizzi non autorizzati, contraffazioni o manomissioni di email recanti 
riferimenti all'azienda.
*Rispetta l'ambiente. Non stampare questa mail se non è necessario.*

	

	

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/590067e2/attachment.html>

From douglas.davenport at gmail.com  Mon Jun 18 17:12:26 2018
From: douglas.davenport at gmail.com (Douglas Davenport)
Date: Mon, 18 Jun 2018 13:12:26 -0400
Subject: [Freeswitch-users] No T38 SDP in 200 OK after glare
Message-ID: <CAMDsFF3GR02KLStT0JeGkfeXFtoVHOE1XYMSpYBJsjEyC4VeCw@mail.gmail.com>

After T38 reinvite glare on an inbound call, the provider resends their
reinvite after a random delay  (they own the call-ID). Freeswitch correctly
responds with 200 OK because fax_enable_t38=true however it does not
include SDP.  Are there any settings that would affect this behavior or is
this a bug? I'm aware that bug reports go to JIRA but I'm asking first to
know if this is actually a bug or I'm doing something wrong. Here is the
call flow. Note the starred 200 OK with no SDP:


                                Provider
 Freeswitch
      ──────────┬─────────          ──────────┬─────────
  12:07:02.067864   │ ─────── INVITE (SDP) ─────> │
  12:07:02.068654   │ <─────── 100 Trying ─────── │
  12:07:03.304619   │ <────── 200 OK (SDP) ────── │
  12:07:03.305554   │ ──────────── ACK ─────────> │
  12:07:06.506575   │ ────── INVITE (T38 SDP)───> │
  12:07:06.718790   │ <────── INVITE (T38 SDP)─── │
  12:07:06.719063   │ ───── 491 Request Pendi ──> │
  12:07:06.879340   │ <─────── 100 Trying ─────── │
  12:07:06.881990   │ <─────────── ACK ────────── │
  12:07:06.943138   │ <──── 491 Request Pendi ─── │
  12:07:06.943356   │ ──────────── ACK ─────────> │
  12:07:07.133006   │ <────INVITE (T38 SDP) ───── │
  12:07:07.133313   │ ──────── 100 Trying ──────> │
  12:07:07.146828   │ ─*************─ 200 OK ─[*NO SDP*]─> │
*****************
  12:07:07.280018   │ <─────────── ACK ────────── │
  12:07:16.012689   │ <─────────── BYE ────────── │
  12:07:16.026995   │ ────────── 200 OK ────────> │
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/c24c2860/attachment.html>

From david.villasmil.work at gmail.com  Mon Jun 18 12:32:28 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 18 Jun 2018 14:32:28 +0200
Subject: [Freeswitch-users] sip.js has no audio
In-Reply-To: <CAA7fES+fCWP4VWU70FBArs=MNi=s6BoLm_8ErR1bAkBGEY1KYg@mail.gmail.com>
References: <CAA7fES+fCWP4VWU70FBArs=MNi=s6BoLm_8ErR1bAkBGEY1KYg@mail.gmail.com>
Message-ID: <CAFGRPVr6=uzqjW8CgLTr=9fX3ZtTb5egSJwfcoex5XXXXOUtpQ@mail.gmail.com>

I would say this more a question for sipjs mailing list.
Having said that, do you see rtps arriving on freeswitch?

On Mon, Jun 18, 2018, 14:27 Hossein Gholizadeh <hggh96.hg at gmail.com> wrote:

> hi all ...  i'm using sip.js for my IVR project ... it could connect to
> freeswitch correctly but there is no audio between call legs ... first of
> all i have to say i really cannot use Verto . this is my function of when
> call button pressed
>
>  * phoneCallButtonPressed : function(sessionid) {*
>
> *            var s      = ctxSip.Sessions[sessionid],*
> *                target = $("#numDisplay").val();*
>
> *            if (!s) {*
>
> *                $("#numDisplay").val("");*
> *                ctxSip.sipCall(target);*
>
> *            } else if (s.accept && !s.startTime) {*
>
> *                s.accept({*
> *                    media : {*
> *                        stream      : ctxSip.Stream,*
> *                        constraints : { audio : true, video : false },*
> *                        render      : {*
> *                            remote : { audio: $('#audioRemote').get()[0]
> }*
> *                        },*
> *                        RTCConstraints : { "optional": [{
> 'DtlsSrtpKeyAgreement': 'true'} ]}*
> *                    }*
> *                });*
> *            }*
> *        },*
>
>
> -----------------------------------------------------------------------------------------
> also i have to say that i tested it between my phone and x-lite ... X-lite
> could receive the voice but my phone could'nt ... i was using Zoiper in my
> phone.
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/460d4dc2/attachment-0001.html>

From david.villasmil.work at gmail.com  Mon Jun 18 14:14:33 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 18 Jun 2018 16:14:33 +0200
Subject: [Freeswitch-users] python script not running properly
In-Reply-To: <CAMtXxQnj4r1VX9aNisbXRh=hW3AAEMVLP7G8s3HqmtuST9fhPg@mail.gmail.com>
References: <CAFGRPVq=R1JVpuEM=nHMf8NH91vA3G-nWc=RU6PNu2Ku86Lt=w@mail.gmail.com>
 <CAMtXxQnj4r1VX9aNisbXRh=hW3AAEMVLP7G8s3HqmtuST9fhPg@mail.gmail.com>
Message-ID: <CAFGRPVq5JGT7Wr4P1HefJL_GKxrrNa7pMX=DGcPT_gitQNdbzw@mail.gmail.com>

Hey thanks for replying!
Yep, i found that answer, but didn't update the question.

Thanks!

On Mon, Jun 18, 2018, 15:39 Joel Serrano <joel at textplus.com> wrote:

> Hi David,
>
> If that happened to me I would run something like this:
>
> print('Test 1: %s\n' % CampaignData)
> print('Test 2: %s\n' % type(CampaignData['campaing_type']))
> print('Test 3: %s\n' % str(CampaignData['campaing_type']))
>
>
> Can you share the output?
> Joel.
>
>
> On Wed, Jun 13, 2018 at 6:50 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello Guys,
>>
>> I'm writing a script in python which when run via cli works properly. I
>> get some dta via http (urllib3) and parse it like:
>>
>> r = http.request('GET', '
>> https://www.massphoning.net/customer/inMassphoning/getInCampaignInfo?phone=
>> ' + caller)
>>
>> if r.status == 200:
>> httpResult = json.loads(r.data.decode('utf-8'))
>>
>> Then print it like:
>> print('Response Parsed: %s\n' % CampaignData)
>> print('CamapignType: %s\n' % CampaignData['campaing_type'])
>>
>> and via the CLI works nicely.
>>
>> But when run on freeswitch dialplan
>> print('Response Parsed: %s\n' % CampaignData)
>>
>> runs properly and prints on the log the data, but it fails on:
>> print('CamapignType: %s\n' % CampaignData['campaing_type'])
>>
>> with:
>> 2018-06-14 03:33:28.971179 [ERR] mod_python.c:164 Python Error by calling
>> script "foo": <type 'exceptions.TypeError'>
>> Message: in method 'consoleLog', argument 2 of type 'char *'
>> Exception: None
>>
>> Traceback (most recent call last)
>> File: "/opt/freeswitch/share/freeswitch/scripts/foo.py", line 16, in
>> handler
>> File: "/opt/freeswitch/share/freeswitch/scripts/foo.py", line 69, in
>> getCampaignInfo
>>
>> Any ideas??
>>
>>
>> Thanks!
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>> ᐧ
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/5907820e/attachment.html>

From david.villasmil.work at gmail.com  Mon Jun 18 14:15:54 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 18 Jun 2018 16:15:54 +0200
Subject: [Freeswitch-users] hangupCause()
In-Reply-To: <048501d4056e$5f771c50$1e6554f0$@delagarda.com>
References: <048501d4056e$5f771c50$1e6554f0$@delagarda.com>
Message-ID: <CAFGRPVohZS0aEMrP71ebPSb7N3SMoofb9_+JR1qZPjNjAsKLuw@mail.gmail.com>

Have you tried just hangup('NORMAL_CLEARING') ?
Or maybe just execute an api

On Mon, Jun 18, 2018, 15:17 Francesco Facco de Lagarda <
francesco at delagarda.com> wrote:

> I’ve been trying to the hangupCause() in javascript.
>
>
>
> I saw that in LUA its
>
> local hcause = session1:hangupCause();
>
>
>
> is there NO session1.hangupCause()  in javascript??
>
>
>
>
>
> Thanks
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/472cbd8c/attachment.html>

From mike at jerris.com  Mon Jun 18 20:16:16 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Jun 2018 16:16:16 -0400
Subject: [Freeswitch-users] Wrong IP in ACK
In-Reply-To: <CAHtxdDeD1iX0U3-PC2yOyMy6yuFxK=n5PKd1gRnphnvDfkqQ8A@mail.gmail.com>
References: <CAHtxdDcXfiD73Av6wrgybCZk5zDx6k2Tnjif-PWEuDGf3Zy_cQ@mail.gmail.com>
 <CAHtxdDeD1iX0U3-PC2yOyMy6yuFxK=n5PKd1gRnphnvDfkqQ8A@mail.gmail.com>
Message-ID: <D26E5F38-7916-4D66-86E3-D813FC14DDFD@jerris.com>

Setting fs_path like that is used to force a locked in outbound proxy for that dialog.  The behavior you are seeing is intended behavior.

> On Jun 12, 2018, at 4:45 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> root of issue
> ACK send to IP resolved via DNS for value defined as "fs_path", rather than defined in contact string.
> 
> more details
> https://freeswitch.org/jira/browse/FS-11190 <https://freeswitch.org/jira/browse/FS-11190>
> 
> вт, 12 июн. 2018 г. в 10:30, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>>:
> Are anybody experiences issue when for some of calls  ACK send to other IP.
> http://prntscr.com/jtxkbz <http://prntscr.com/jtxkbz>
> in this case FS in bypass media with two profiles (1 to internet, 2 intranet).
> INVITE is send to DNS name that is resolved to
> 10.0.9.32
> 10.0.9.33
> 10.0.9.34
> 10.0.9.35
> FS 1.6.19
> 
> Contact on 200 message is correct
> 
> SIP/2.0 200 OK
> Via: SIP/2.0/TCP 10.0.9.51:1 <http://10.0.9.51:1/>2000;rport=39049;branch=z9hG4bKBH60NQjSZj36a
> From: "Anonymous" <sip:+12022553656 at 10.0.9.51 <mailto:sip%3A%2B12022553656 at 10.0.9.51>>;tag=rH5cBQarr559D
> To: <sip:13013688630 <tel:(301)%20368-8630>@metropolis.ack.one>;tag=UDv7UXv3B53Da
> Call-ID: b0e4d7e1-e79e-1236-8cb9-02420a000933
> CSeq: 123998414 INVITE
> Contact: <sip:13013688630 <tel:(301)%20368-8630>@10.0.9.33:11000;transport=tcp>
> User-Agent: proxy.ackone.com <http://proxy.ackone.com/>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
> 
> Now for future investigate FS upgrader to 1.6.20 and enabled siptrace and loglevel
> 
> Sergey Safarov

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/1aa01f68/attachment-0001.html>

From mike at jerris.com  Mon Jun 18 20:18:58 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Jun 2018 16:18:58 -0400
Subject: [Freeswitch-users] Hung Sofia Profile
In-Reply-To: <CAKzROs3di=WNu7JpS_j3JCr=EeBkU_WA4RSjnAW6DHuN2TaN5g@mail.gmail.com>
References: <CAKzROs3Um7Vae47zRQLGMYWLKyw7jL9fuHm+TSABf8X4bmc6VQ@mail.gmail.com>
 <CAKbxfG93ZihgjXs3k-riO5SQc0Sx+H6WF-On4C47PW516fJ9+A@mail.gmail.com>
 <CAKzROs0qx0Qw-HFo5t+_NhmTkA4egs_+KTuvm1MKOsLa+5WTpQ@mail.gmail.com>
 <CALXCt0qeJE+GK6A8fcv+0mcWqdS6c34mDbzWZHRxtRGbUPJ5JA@mail.gmail.com>
 <CAKzROs3di=WNu7JpS_j3JCr=EeBkU_WA4RSjnAW6DHuN2TaN5g@mail.gmail.com>
Message-ID: <D65AE504-A6CD-4292-875A-0284F44AB0DA@jerris.com>

the history of issues is in the client and odbc drivers, yes.

Mike


> On Jun 13, 2018, at 6:42 PM, Geoff Mina <gmina at connectfirst.com> wrote:
> 
> Thanks.  I was assuming it was the ODBC driver since MySQL itself isn't prone to deadlocks.
> 
> We have switched all of our backends to PostgreSQL as of last week.  So far so good.
> 
> On Wed, Jun 13, 2018 at 4:31 PM Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
> On Sat, Jun 9, 2018, 02:12 Geoff Mina <gmina at connectfirst.com <mailto:gmina at connectfirst.com>> wrote:
> Would you guys suggest switching to Postgres on the back-end?  I am guessing if there are known MySQL issues, the suggestion is probably native PGSQL?
> 
> 
> Problem is with the odbc driver.
> 
>  So, short answer is: yes, definitely go with postgresql native support
> 
> 
> -giovanni
> 
> 
> 
> On Fri, Jun 8, 2018 at 2:48 PM Anthony Minessale <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> You would need to do a thread apply all bt to see what the other threads are doing.  The thread you posted is stuck on a mutex waiting for something else, probably your DB in another thread.
> If you do get that info file it to jira because its easier to deal with backtraces there.
> 
> 
> On Fri, Jun 8, 2018 at 12:04 PM, Geoff Mina <gmina at connectfirst.com <mailto:gmina at connectfirst.com>> wrote:
> Greetings,
> Has anyone seen an instance where freeswitch gets hung with the following stack trace?  I have about 400 threads in this state and eventually FS (1.6.19) just stops responding.  
> 
> I am running FS against MySQL 5.6.38 on CentOS using ODBC.   Guessing if this is known, it probably isn't fixed in 1.6.20... but I can hope.  Any input on what a possible solution would be greatly appreciated.  The MySQL server and FS are running on the same host connecting via localhost interface, so it's not a networking issue.
> 
> Thread 445 (Thread 0x7fa2df6db700 (LWP 9395)):
> #0  0x00007fa3712da42d in __lll_lock_wait () from /usr/lib64/libpthread.so.0
> #1  0x00007fa3712d5de6 in _L_lock_870 () from /usr/lib64/libpthread.so.0
> #2  0x00007fa3712d5cdf in pthread_mutex_lock () from /usr/lib64/libpthread.so.0
> #3  0x00007fa35742468a in sofia_glue_execute_sql_callback () from /usr/lib64/freeswitch/mod/mod_sofia.so
> #4  0x00007fa3573df231 in select_from_profile () from /usr/lib64/freeswitch/mod/mod_sofia.so
> #5  0x00007fa3573e3b17 in sofia_contact_function () from /usr/lib64/freeswitch/mod/mod_sofia.so
> #6  0x00007fa3730594cc in switch_api_execute () from /usr/lib64/libfreeswitch.so.1
> #7  0x00007fa372fe69d6 in switch_channel_expand_variables_check () from /usr/lib64/libfreeswitch.so.1
> #8  0x00007fa3552d45e4 in user_outgoing_channel () from /usr/lib64/freeswitch/mod/mod_dptools.so
> #9  0x00007fa37300b42a in switch_core_session_outgoing_channel () from /usr/lib64/libfreeswitch.so.1
> #10 0x00007fa37308dc3b in switch_ivr_originate () from /usr/lib64/libfreeswitch.so.1
> #11 0x00007fa3552db0b5 in audio_bridge_function () from /usr/lib64/freeswitch/mod/mod_dptools.so
> #12 0x00007fa37300ebcb in switch_core_session_exec () from /usr/lib64/libfreeswitch.so.1
> #13 0x00007fa37300f179 in switch_core_session_execute_application_get_flags () from /usr/lib64/libfreeswitch.so.1
> #14 0x00007fa373012a34 in switch_core_session_run () from /usr/lib64/libfreeswitch.so.1
> #15 0x00007fa37300c33e in switch_core_session_thread () from /usr/lib64/libfreeswitch.so.1
> #16 0x00007fa373007d83 in switch_core_session_thread_pool_worker () from /usr/lib64/libfreeswitch.so.1
> #17 0x00007fa3732ce7c0 in dummy_worker () from /usr/lib64/libfreeswitch.so.1
> #18 0x00007fa3712d3e25 in start_thread () from /usr/lib64/libpthread.so.0
> #19 0x00007fa37092e34d in clone () from /usr/lib64/libc.so.6

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/19497dc6/attachment-0001.html>

From mike at jerris.com  Mon Jun 18 21:01:04 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Jun 2018 17:01:04 -0400
Subject: [Freeswitch-users] Using H264 with Verto
In-Reply-To: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>
References: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>
Message-ID: <3D1DB6A6-D120-4C75-93C3-4D010DE4273A@jerris.com>

load mod_av?  if thats not the issue, I’d need to see a debug log w/ sip trace to tell you more.


> On Jun 15, 2018, at 2:50 PM, Chad Phillips <chad at apartmentlines.com> wrote:
> 
> I have a slightly older install where I’m using H264 with Verto, and it works fine. However, I’m not able to get it working on latest master.
> 
> I’m guessing there’s some esoteric setting I’m missing, but can’t seem to find it. I do have the following in verto.conf.xml:
> 
>       <param name="outbound-codec-string" value="opus,h264"/>
>       <param name=“inbound-codec-string” value="opus,h264"/>
> 
> I also have mod_h26x disabled, as I remember that’s passthrough only and causes issues when loaded in this case.
> 
> When I try in Chrome I see it offering H264 in the SDP, but the answer SDP contains this, and the stream is audio only: 
> 
> m=video 0 UDP/TLS/RTP/SAVPF 19
> 
> What am I missing??



From mike at jerris.com  Mon Jun 18 21:04:24 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Jun 2018 17:04:24 -0400
Subject: [Freeswitch-users] Memory leak while using BGAPI
In-Reply-To: <25D2EC755404B4409F263AC6D050FEBB2BCCD8C4@AZ-FFEXMB03.global.avaya.com>
References: <25D2EC755404B4409F263AC6D050FEBB2BCCD8C4@AZ-FFEXMB03.global.avaya.com>
Message-ID: <108CFF47-7EC4-4562-AB51-1DE1061AFFA8@jerris.com>

If that event is being consumed by something that is blocking maybe?  Is it always that number or if you do many more does it get bigger?  This could just be an event that isnt consumed yet when you shut down and not a big problem.


> On Jun 18, 2018, at 2:17 AM, Mody, Darshan (Darshan) <darshanmody at avaya.com> wrote:
> 
> Hi,
>  
> We are observing considerable memory leak while using LUA and BGAPI of mod_command.c. Below is the valgrind’s output
>  
> ==21863== 123,170 (45,496 direct, 77,674 indirect) bytes in 517 blocks are definitely lost in loss record 2,370 of 2,398
> ==21863==    at 0x4C29C23: malloc (vg_replace_malloc.c:299)
> ==21863==    by 0x548707C: switch_event_create_subclass_detailed (switch_event.c:736)
> ==21863==    by 0x11631389: bgapi_exec (mod_commands.c:5151)
> ==21863==    by 0x56EEE8F: dummy_worker (thread.c:151)
> ==21863==    by 0x7677E24: start_thread (in /usr/lib64/libpthread-2.17.so)
> ==21863==    by 0x805734C: clone (in /usr/lib64/libc-2.17.so
>  
> ==21863== 143,478 (18,392 direct, 125,086 indirect) bytes in 209 blocks are definitely lost in loss record 2,374 of 2,398
> ==21863==    at 0x4C29C23: malloc (vg_replace_malloc.c:299)
> ==21863==    by 0x548707C: switch_event_create_subclass_detailed (switch_event.c:736)
> ==21863==    by 0x542AADE: switch_core_session_exec (switch_core_session.c:2874)
> ==21863==    by 0x542B108: switch_core_session_execute_application_get_flags (switch_core_session.c:2742)
> ==21863==    by 0x5510C10: CoreSession::execute(char const*, char const*) (switch_cpp.cpp:777)
> ==21863==    by 0x16FED72C: ??? (mod_lua_wrap.cpp:7289)
> ==21863==    by 0x17217323: ??? (in /usr/lib64/liblua-5.1.so)
> ==21863==    by 0x17221E56: ??? (in /usr/lib64/liblua-5.1.so)
> ==21863==    by 0x1721774C: ??? (in /usr/lib64/liblua-5.1.so)
> ==21863==    by 0x17216A6D: ??? (in /usr/lib64/liblua-5.1.so)
> ==21863==    by 0x172178D9: ??? (in /usr/lib64/liblua-5.1.so)
> ==21863==    by 0x1721344C: lua_pcall (in /usr/lib64/liblua-5.1.so)
>  
> Has some one also observed behavior while using BGAPI?
>  
> Thanks
> Darshan 
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/6bd07a93/attachment-0001.html>

From mike at jerris.com  Mon Jun 18 21:08:01 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Jun 2018 17:08:01 -0400
Subject: [Freeswitch-users] hangupCause()
In-Reply-To: <048501d4056e$5f771c50$1e6554f0$@delagarda.com>
References: <048501d4056e$5f771c50$1e6554f0$@delagarda.com>
Message-ID: <8DF569C9-4C24-43AC-83A6-2AF3EFD60294@jerris.com>

https://freeswitch.org/confluence/display/FREESWITCH/Session+cause <https://freeswitch.org/confluence/display/FREESWITCH/Session+cause>

> On Jun 16, 2018, at 8:34 AM, Francesco Facco de Lagarda <francesco at delagarda.com> wrote:
> 
> I’ve been trying to the hangupCause() in javascript.
>  
> I saw that in LUA its
> local hcause = session1:hangupCause();
>  
> is there NO session1.hangupCause()  in javascript??
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/5c1c1daf/attachment.html>

From mike at jerris.com  Mon Jun 18 21:08:59 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Jun 2018 17:08:59 -0400
Subject: [Freeswitch-users] causecode - please help
In-Reply-To: <049201d4056f$750c5c00$5f251400$@delagarda.com>
References: <049201d4056f$750c5c00$5f251400$@delagarda.com>
Message-ID: <566D5CC0-F0F1-4688-A978-67F92FF7349A@jerris.com>

https://freeswitch.org/confluence/display/FREESWITCH/Session+cause <https://freeswitch.org/confluence/display/FREESWITCH/Session+cause>


> On Jun 16, 2018, at 8:42 AM, Francesco Facco de Lagarda <francesco at delagarda.com> wrote:
> 
> I am trying the undestand the hangup reason in javascript
> I noticed there is no 
> session.hangupCause
> but there IS a
>                 session.causecode
>  
> Docs say it gives the Q.931 code for hangup,
>  
> But, on occupied calls, I just get “false” when I ask
> 
> session.causecode
>  
> What’s happening..
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/0e20832e/attachment.html>

From mike at jerris.com  Mon Jun 18 21:12:46 2018
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Jun 2018 17:12:46 -0400
Subject: [Freeswitch-users] Installing Freeswitch 1.8 on testing mode.
In-Reply-To: <00e801d406fa$b5f6ad60$21e40820$@smartic.es>
References: <00e801d406fa$b5f6ad60$21e40820$@smartic.es>
Message-ID: <2502420D-C2E9-4262-B3C2-6E5BC2D2EEB2@jerris.com>

its possible to get master to compile with some tricks, but due to some issues in what Debian 9 did with supporting multiple conflicting library versions, the chance of random crashes are high without a bunch of work to override a number of system packages.  I would not use it without those issues being resolved, as it would be unstable.

> On Jun 18, 2018, at 7:51 AM, Miguel Jesús López Valverde <mjlopez at smartic.es> wrote:
> 
> Hello,
>  
>                 Is it possible to complete a Freeswitch 1.8 installation on Debian 9, as seen in the comments on the link https://freeswitch.org/confluence/display/FREESWITCH/Debian+9+Stretch <https://freeswitch.org/confluence/display/FREESWITCH/Debian+9+Stretch> ?, Do you know if there is a recipe for it , even in testing mode ?.
>  

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/e399576e/attachment.html>

From brian at freeswitch.com  Mon Jun 18 21:13:43 2018
From: brian at freeswitch.com (Brian West)
Date: Mon, 18 Jun 2018 16:13:43 -0500
Subject: [Freeswitch-users] No T38 SDP in 200 OK after glare
In-Reply-To: <CAMDsFF3GR02KLStT0JeGkfeXFtoVHOE1XYMSpYBJsjEyC4VeCw@mail.gmail.com>
References: <CAMDsFF3GR02KLStT0JeGkfeXFtoVHOE1XYMSpYBJsjEyC4VeCw@mail.gmail.com>
Message-ID: <CAEJMVkD6hYU58Gg1W_p71WxpPwPQQjXocxsOKU2grvY83MdEzA@mail.gmail.com>

Does this call terminate on FreeSWITCH or are you doing passthru?

/b


On Mon, Jun 18, 2018 at 12:12 PM, Douglas Davenport <
douglas.davenport at gmail.com> wrote:

> After T38 reinvite glare on an inbound call, the provider resends their
> reinvite after a random delay  (they own the call-ID). Freeswitch correctly
> responds with 200 OK because fax_enable_t38=true however it does not
> include SDP.  Are there any settings that would affect this behavior or is
> this a bug? I'm aware that bug reports go to JIRA but I'm asking first to
> know if this is actually a bug or I'm doing something wrong. Here is the
> call flow. Note the starred 200 OK with no SDP:
>
>
>                                 Provider
>    Freeswitch
>       ──────────┬─────────          ──────────┬─────────
>   12:07:02.067864   │ ─────── INVITE (SDP) ─────> │
>   12:07:02.068654   │ <─────── 100 Trying ─────── │
>   12:07:03.304619   │ <────── 200 OK (SDP) ────── │
>   12:07:03.305554   │ ──────────── ACK ─────────> │
>   12:07:06.506575   │ ────── INVITE (T38 SDP)───> │
>   12:07:06.718790   │ <────── INVITE (T38 SDP)─── │
>   12:07:06.719063   │ ───── 491 Request Pendi ──> │
>   12:07:06.879340   │ <─────── 100 Trying ─────── │
>   12:07:06.881990   │ <─────────── ACK ────────── │
>   12:07:06.943138   │ <──── 491 Request Pendi ─── │
>   12:07:06.943356   │ ──────────── ACK ─────────> │
>   12:07:07.133006   │ <────INVITE (T38 SDP) ───── │
>   12:07:07.133313   │ ──────── 100 Trying ──────> │
>   12:07:07.146828   │ ─*************─ 200 OK ─[*NO SDP*]─> │
> *****************
>   12:07:07.280018   │ <─────────── ACK ────────── │
>   12:07:16.012689   │ <─────────── BYE ────────── │
>   12:07:16.026995   │ ────────── 200 OK ────────> │
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 

Brian West | Co-founder and Developer

Need Commercial support? email sales at freeswitch.com

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

Email: brian at freeswitch.com

Mobile: 918-424-9378

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

[image: https://www.facebook.com/signalwireinc?src=email]
<https://www.facebook.com/freeswitch> [image:
https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/fa4289ea/attachment-0001.html>

From alex at freeswitch.com  Tue Jun 19 02:26:16 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Tue, 19 Jun 2018 11:26:16 +0900
Subject: [Freeswitch-users] Problem loading mod_lua.so
In-Reply-To: <3d54a27e-196a-6e61-ca0f-c7a27875f82f@tiesse.com>
References: <3d54a27e-196a-6e61-ca0f-c7a27875f82f@tiesse.com>
Message-ID: <CAGJLwFibpbmanr-gJU+ZCxihCD=W14RgUAkQiMAKjNzXuY7J7Q@mail.gmail.com>

Hi,

Attitude Adjustment released more than 5 years ago, your best shot is
update.

Regards,

Alex

On Tue, Jun 19, 2018 at 12:43 AM, fabio <f.antonini at tiesse.com> wrote:

> Hi everybody
>
> I'm facing a strange problem when I configure mod_lua in my OpenWrt
> framework. Freeswitch version is 1.6.20 and OpenWrt is an
> 'attitude_adjustment'.
>
>
> I added 'mod_lua' to the list of modules to be loaded ( "<load
> module="mod_lua"/>" in the file /etc/freeswitch/autoload_configs/modules.conf.xml).
> The mod_lua is successfully compiled, and at the boot FS tries to load it
> but unfortunately it complains with the following message:
>
>
> *2018-06-18 15:29:17.395100 [CRIT] switch_loadable_module.c:1448 Error
> Loading module /usr/lib/freeswitch/mod_lua.so*
> ***Unknown error***
>
>
> The file is placed in that directory with all the other FS modules.
>
>
> I cannot understand where is problem.
>
>
> Any help or advice will greatly appreciated.
>
>
> Thanks in advance
>
>
> Best regards
>
>
> fabio
> --
>
> Fabio Antonini
>
> *Software Engineer (Ph.D)*
> f.antonini at tiesse.com
> *Tel* +39.0863.455830
> *Mob* +39.393.9261941
> *Fax* +39.0863.455830
> Via Corradini 80
> 67051 Avezzano (AQ)
>
> [image: Logo Tiesse dal 1998 al 2018, vent'anni di Innovazione Made in
> Italy. Clicca per visitare il sito Tiesse] <http://www.tiesse.com/>
>
> *Tiesse S.p.A.* - www.tiesse.com
> Via Asti 4, 10015 Ivrea (TO)
> [image: Pagina Tiesse su Linkedin, clicca e visitaci]
> <https://linkedin.com/company/tiesse-spa>
> *Disclaimer:* il contenuto di questa email è riservato e non vincolante
> per Tiesse S.p.A.. Se lo avesse ricevuto per errore, la preghiamo di
> segnalarlo immediatamente al mittente, di non utilizzare e divulgare il
> contenuto e di distruggere ogni copia in suo possesso. Tiesse S.p.A.
> declina ogni responsabilità da qualsiasi conseguenza derivante da utilizzi
> non autorizzati, contraffazioni o manomissioni di email recanti riferimenti
> all'azienda.
> *Rispetta l'ambiente. Non stampare questa mail se non è necessario.*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 
Alex Sibyakin | Support Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/b551da36/attachment.html>

From andrew.keil at visytel.com  Mon Jun 18 22:07:35 2018
From: andrew.keil at visytel.com (Andrew Keil)
Date: Mon, 18 Jun 2018 22:07:35 +0000
Subject: [Freeswitch-users] Memory leak while using BGAPI
In-Reply-To: <25D2EC755404B4409F263AC6D050FEBB2BCCD8C4@AZ-FFEXMB03.global.avaya.com>
References: <25D2EC755404B4409F263AC6D050FEBB2BCCD8C4@AZ-FFEXMB03.global.avaya.com>
Message-ID: <62B0235C-1568-4746-BBEA-F048437792F5@visytel.com>

Darshan,

Can you provide some Lua sample code showing your BGAPI call and I will try to replicate the issue here.  Also what version of FreeSWITCH are you running?

Andrew

On 18 Jun 2018, at 10:59 pm, Mody, Darshan (Darshan) <darshanmody at avaya.com<mailto:darshanmody at avaya.com>> wrote:

Hi,

We are observing considerable memory leak while using LUA and BGAPI of mod_command.c. Below is the valgrind’s output

==21863== 123,170 (45,496 direct, 77,674 indirect) bytes in 517 blocks are definitely lost in loss record 2,370 of 2,398
==21863==    at 0x4C29C23: malloc (vg_replace_malloc.c:299)
==21863==    by 0x548707C: switch_event_create_subclass_detailed (switch_event.c:736)
==21863==    by 0x11631389: bgapi_exec (mod_commands.c:5151)
==21863==    by 0x56EEE8F: dummy_worker (thread.c:151)
==21863==    by 0x7677E24: start_thread (in /usr/lib64/libpthread-2.17.so)
==21863==    by 0x805734C: clone (in /usr/lib64/libc-2.17.so

==21863== 143,478 (18,392 direct, 125,086 indirect) bytes in 209 blocks are definitely lost in loss record 2,374 of 2,398
==21863==    at 0x4C29C23: malloc (vg_replace_malloc.c:299)
==21863==    by 0x548707C: switch_event_create_subclass_detailed (switch_event.c:736)
==21863==    by 0x542AADE: switch_core_session_exec (switch_core_session.c:2874)
==21863==    by 0x542B108: switch_core_session_execute_application_get_flags (switch_core_session.c:2742)
==21863==    by 0x5510C10: CoreSession::execute(char const*, char const*) (switch_cpp.cpp:777)
==21863==    by 0x16FED72C: ??? (mod_lua_wrap.cpp:7289)
==21863==    by 0x17217323: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x17221E56: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x1721774C: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x17216A6D: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x172178D9: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x1721344C: lua_pcall (in /usr/lib64/liblua-5.1.so)

Has some one also observed behavior while using BGAPI?

Thanks
Darshan


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/0f7b4c88/attachment-0001.html>

From darshanmody at avaya.com  Tue Jun 19 06:26:00 2018
From: darshanmody at avaya.com (Mody, Darshan (Darshan))
Date: Tue, 19 Jun 2018 06:26:00 +0000
Subject: [Freeswitch-users] Memory leak while using BGAPI
In-Reply-To: <108CFF47-7EC4-4562-AB51-1DE1061AFFA8@jerris.com>
References: <25D2EC755404B4409F263AC6D050FEBB2BCCD8C4@AZ-FFEXMB03.global.avaya.com>
 <108CFF47-7EC4-4562-AB51-1DE1061AFFA8@jerris.com>
Message-ID: <25D2EC755404B4409F263AC6D050FEBB2BCCEFCE@AZ-FFEXMB03.global.avaya.com>

Hi Micheal,

We find below code in the switch_event.c. Is there a specific reason as to why are we setting the pointers eventp and event as NULL and not destroying the event?

static switch_status_t switch_event_queue_dispatch_event(switch_event_t **eventp)
{

                switch_event_t *event = *eventp;

                if (!SYSTEM_RUNNING) {
                                return SWITCH_STATUS_FALSE;
                }

                while (event) {
                                int launch = 0;

                                switch_mutex_lock(EVENT_QUEUE_MUTEX);

                                if (!PENDING && switch_queue_size(EVENT_DISPATCH_QUEUE) > (unsigned int)(DISPATCH_QUEUE_LEN * DISPATCH_THREAD_COUNT)) {
                                                if (SOFT_MAX_DISPATCH + 1 < MAX_DISPATCH) {
                                                                launch++;
                                                                PENDING++;
                                                                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "launch = %d & PENDING = %d\n", launch,PENDING);
                                                }
                                }

                                switch_mutex_unlock(EVENT_QUEUE_MUTEX);

                                if (launch) {
                                                if (SOFT_MAX_DISPATCH + 1 < MAX_DISPATCH) {
                                                                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Dispatching %d Thread\n", (SOFT_MAX_DISPATCH + 1));
                                                                switch_event_launch_dispatch_threads(SOFT_MAX_DISPATCH + 1);
                                                }

                                                switch_mutex_lock(EVENT_QUEUE_MUTEX);
                                                PENDING--;
                                                switch_mutex_unlock(EVENT_QUEUE_MUTEX);
                                }

                                *eventp = NULL;
                                switch_queue_push(EVENT_DISPATCH_QUEUE, event);
                                event = NULL;

                }

                return SWITCH_STATUS_SUCCESS;
}

Thanks
Darshan

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, June 19, 2018 2:34 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Memory leak while using BGAPI

If that event is being consumed by something that is blocking maybe?  Is it always that number or if you do many more does it get bigger?  This could just be an event that isnt consumed yet when you shut down and not a big problem.



On Jun 18, 2018, at 2:17 AM, Mody, Darshan (Darshan) <darshanmody at avaya.com<mailto:darshanmody at avaya.com>> wrote:

Hi,

We are observing considerable memory leak while using LUA and BGAPI of mod_command.c. Below is the valgrind’s output

==21863== 123,170 (45,496 direct, 77,674 indirect) bytes in 517 blocks are definitely lost in loss record 2,370 of 2,398
==21863==    at 0x4C29C23: malloc (vg_replace_malloc.c:299)
==21863==    by 0x548707C: switch_event_create_subclass_detailed (switch_event.c:736)
==21863==    by 0x11631389: bgapi_exec (mod_commands.c:5151)
==21863==    by 0x56EEE8F: dummy_worker (thread.c:151)
==21863==    by 0x7677E24: start_thread (in /usr/lib64/libpthread-2.17.so)
==21863==    by 0x805734C: clone (in /usr/lib64/libc-2.17.so

==21863== 143,478 (18,392 direct, 125,086 indirect) bytes in 209 blocks are definitely lost in loss record 2,374 of 2,398
==21863==    at 0x4C29C23: malloc (vg_replace_malloc.c:299)
==21863==    by 0x548707C: switch_event_create_subclass_detailed (switch_event.c:736)
==21863==    by 0x542AADE: switch_core_session_exec (switch_core_session.c:2874)
==21863==    by 0x542B108: switch_core_session_execute_application_get_flags (switch_core_session.c:2742)
==21863==    by 0x5510C10: CoreSession::execute(char const*, char const*) (switch_cpp.cpp:777)
==21863==    by 0x16FED72C: ??? (mod_lua_wrap.cpp:7289)
==21863==    by 0x17217323: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x17221E56: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x1721774C: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x17216A6D: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x172178D9: ??? (in /usr/lib64/liblua-5.1.so)
==21863==    by 0x1721344C: lua_pcall (in /usr/lib64/liblua-5.1.so)

Has some one also observed behavior while using BGAPI?

Thanks
Darshan


_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com<mailto:sales at freeswitch.com>
https://freeswitch.com<https://urldefense.proofpoint.com/v2/url?u=https-3A__freeswitch.com_&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=cf87DDQfp5SEnNdhK6qztdg7bCeSemewRWYu-gCSUuM&e=>

Official FreeSWITCH Sites
https://freeswitch.com/oss<https://urldefense.proofpoint.com/v2/url?u=https-3A__freeswitch.com_oss&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=km3nKWxpOpy0aSeC6s3a2J04SwIbQR8nJF_1h-3w9iE&e=>
https://freeswitch.org/confluence<https://urldefense.proofpoint.com/v2/url?u=https-3A__freeswitch.org_confluence&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=Mj2ZnsEi90La30FIpe-hLKtfSogOiqMoiuljmGH73SU&e=>
https://cluecon.com<https://urldefense.proofpoint.com/v2/url?u=https-3A__cluecon.com_&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=ZFkKDP5CjqxuU0TAIPZlU9zUpys5ch7RgYWUEGaVTsk&e=>

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<https://urldefense.proofpoint.com/v2/url?u=http-3A__lists.freeswitch.org_mailman_listinfo_freeswitch-2Dusers&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=kXdKsgCA_UpVrUS_jBuy4lBpWP5RT9qazUatzWIslsE&e=>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<https://urldefense.proofpoint.com/v2/url?u=http-3A__lists.freeswitch.org_mailman_options_freeswitch-2Dusers&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=J8fwO1L0LvupkX0THYA5RMB66VVagvVYay4ie0-2s68&e=>
https://freeswitch.com<https://urldefense.proofpoint.com/v2/url?u=https-3A__freeswitch.com_&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=cf87DDQfp5SEnNdhK6qztdg7bCeSemewRWYu-gCSUuM&e=>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/22f6c074/attachment-0001.html>

From david.villasmil.work at gmail.com  Mon Jun 18 22:46:17 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 19 Jun 2018 00:46:17 +0200
Subject: [Freeswitch-users] Installing Freeswitch 1.8 on testing mode.
In-Reply-To: <00e801d406fa$b5f6ad60$21e40820$@smartic.es>
References: <00e801d406fa$b5f6ad60$21e40820$@smartic.es>
Message-ID: <CAFGRPVq14NGt8zqTQOxSRkPVpHDOmD19rUs+J97MNJpMFvL4oQ@mail.gmail.com>

There's no recipe yet, install on debian 8, you'll save yourself a lot of
trouble... specially on a test environment.

Regards,

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

ᐧ

On Mon, Jun 18, 2018 at 11:41 PM Miguel Jesús López Valverde <
mjlopez at smartic.es> wrote:

> Hello,
>
>
>
>                 Is it possible to complete a Freeswitch 1.8 installation
> on Debian 9, as seen in the comments on the link
> https://freeswitch.org/confluence/display/FREESWITCH/Debian+9+Stretch ?,
> Do you know if there is a recipe for it , even in testing mode ?.
>
>
>
> On the other hand, does anyone know if there is any date or estimate for
> the release of Freeswitch 1.8?
>
>
>
> Thank you very much.
>
>
>
> Miguel J. López.
>
>
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient> Libre
> de virus. www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=emailclient>
> <#m_-1311653527338938663_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/142cc620/attachment.html>

From s.safarov at gmail.com  Tue Jun 19 07:06:41 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 19 Jun 2018 10:06:41 +0300
Subject: [Freeswitch-users] Wrong IP in ACK
In-Reply-To: <D26E5F38-7916-4D66-86E3-D813FC14DDFD@jerris.com>
References: <CAHtxdDcXfiD73Av6wrgybCZk5zDx6k2Tnjif-PWEuDGf3Zy_cQ@mail.gmail.com>
 <CAHtxdDeD1iX0U3-PC2yOyMy6yuFxK=n5PKd1gRnphnvDfkqQ8A@mail.gmail.com>
 <D26E5F38-7916-4D66-86E3-D813FC14DDFD@jerris.com>
Message-ID: <CAHtxdDd9cJB_JCcmhS_iaCicgHe+AZBpqYMkCi+i9R8w4261Rw@mail.gmail.com>

If FS ignores contact values and use fs_path like "Record-Route" SIP
header, then ACK must be send to same IP used for INVITE.
Without this whole session is broken.
Please look call log. Call is dropped by remote end because ACK not
received.

вт, 19 июн. 2018 г. в 2:18, Michael Jerris <mike at jerris.com>:

> Setting fs_path like that is used to force a locked in outbound proxy for
> that dialog.  The behavior you are seeing is intended behavior.
>
>
> On Jun 12, 2018, at 4:45 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> root of issue
> ACK send to IP resolved via DNS for value defined as "fs_path", rather
> than defined in contact string.
>
> more details
> https://freeswitch.org/jira/browse/FS-11190
>
> вт, 12 июн. 2018 г. в 10:30, Sergey Safarov <s.safarov at gmail.com>:
>
>> Are anybody experiences issue when for some of calls  ACK send to other
>> IP.
>> http://prntscr.com/jtxkbz
>> in this case FS in bypass media with two profiles (1 to internet, 2
>> intranet).
>> INVITE is send to DNS name that is resolved to
>>
>>    1. 10.0.9.32
>>    2. 10.0.9.33
>>    3. 10.0.9.34
>>    4. 10.0.9.35
>>
>> FS 1.6.19
>>
>> Contact on 200 message is correct
>>
>> SIP/2.0 200 OK Via: SIP/2.0/TCP 10.0.9.51:12000;rport=39049;branch=
>> z9hG4bKBH60NQjSZj36a From: "Anonymous" <sip:+12022553656 at 10.0.9.51>;tag=
>> rH5cBQarr559D To: <sip:13013688630 <(301)%20368-8630>@metropolis.ack.one>;tag=UDv7UXv3B53Da
>> Call-ID: b0e4d7e1-e79e-1236-8cb9-02420a000933 CSeq: 123998414 INVITE
>> Contact: <sip:13013688630 <(301)%20368-8630>@10.0.9.33:11000;transport=tcp>
>> User-Agent: proxy.ackone.com Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>> MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Now for future investigate FS upgrader to 1.6.20 and enabled siptrace and
>> loglevel
>>
>> Sergey Safarov
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/621a58ad/attachment.html>

From j4v28bsjtp43hnf865 at gmail.com  Tue Jun 19 04:45:54 2018
From: j4v28bsjtp43hnf865 at gmail.com (Xenia Obolenskaya)
Date: Tue, 19 Jun 2018 04:45:54 +0000
Subject: [Freeswitch-users] Peer Subject Mismatch (incoming connection)
Message-ID: <CAF7mOn5W_XAio5hBdzpzfM6b5cCw_Gt1uxbvGi1SXFjC78KwJg@mail.gmail.com>

Hi, All,

Peer Subject Mismatch (incoming connection) - WHY MISMATCH? WHAT MISMATCH?

Why Peer Certificate Subject 0: "sip.obolenskaya.su" is mismatching realm
(which is correctly resolving with DNS SRV including port) "
sip.obolenskaya.su"?
The same CA certificate was placed in cafile.pem on both peers.

Peer Certificate:
    Data:
        Version: 3 (0x2)
        Serial Number:
            a7:17:ad:32:72:ce:b5:e9
    Signature Algorithm: sha256WithRSAEncryption
        Issuer: O=FS, CN=My_FS
        Validity
            Not Before: Jun 18 07:42:08 2018 GMT
            Not After : Dec  9 07:42:08 2023 GMT
        Subject: O=FS, CN=obolenskaya.su
        Subject Public Key Info:
            Public Key Algorithm: rsaEncryption
                Public-Key: (2048 bit)
                Modulus:
                    00:ef:31:86:e6:dd:13:2d:92:e1:2b:7a:14:7f:de:
                    0f:11:97:2b:d9:de:bf:1e:6b:bc:90:ef:8e:04:2a:
                    99:7b:58:a1:39:94:01:42:00:38:6e:e0:1c:cf:13:
                    7f:25:2d:f2:c0:0f:ed:0d:ad:0e:e9:37:40:32:ae:
                    92:2e:58:b0:79:2b:bf:16:5a:11:86:5b:18:bd:a1:
                    b0:3c:2a:2f:cb:bd:52:c5:dd:a1:94:64:96:18:3a:
                    77:eb:0d:b3:65:24:ff:22:d9:3b:55:a7:13:bc:1c:
                    d2:61:be:40:e1:1c:43:87:c9:78:ab:b4:55:95:fd:
                    52:f6:e7:e3:04:ed:50:86:d3:19:53:67:07:30:d4:
                    08:62:c6:a2:f5:e9:07:71:5e:03:af:96:1a:89:39:
                    db:cf:6f:21:be:46:61:6b:cc:2c:10:99:b4:cf:32:
                    db:c1:a3:0d:03:5d:1f:30:45:4e:ca:ff:f4:ba:ad:
                    d9:4f:6a:5a:f6:42:41:82:d4:3f:39:f9:5a:98:95:
                    1a:c1:e4:4f:5c:a0:e3:a0:dd:1c:a6:65:f9:98:15:
                    f0:8b:18:c5:0d:8f:67:17:4c:5c:ef:ed:fc:b5:42:
                    80:0f:c8:e4:e6:02:73:c6:8d:8f:1c:94:4d:de:90:
                    eb:24:05:94:36:f3:76:8c:62:a2:80:e6:57:24:06:
                    0d:33
                Exponent: 65537 (0x10001)
        X509v3 extensions:
            X509v3 Basic Constraints:
                CA:FALSE
            X509v3 Key Usage:
                Digital Signature, Non Repudiation, Key Encipherment
            X509v3 Subject Alternative Name:
                DNS:sip.obolenskaya.su, DNS:obolenskaya.su
    Signature Algorithm: sha256WithRSAEncryption
         5a:54:62:45:66:71:3c:11:b8:01:21:e2:bb:bb:3c:ca:0d:23:
         c8:d2:3e:5b:9f:93:28:cd:4d:df:b0:82:8f:76:b3:9e:0c:4a:
         91:e1:f5:c7:aa:ae:26:a1:c5:87:a5:16:8f:60:6f:6b:f6:80:
         f8:7f:f9:12:f3:87:bf:63:52:da:1b:35:c7:31:16:d0:4f:7c:
         49:71:f4:77:99:4c:64:97:a0:bb:e3:cb:b5:67:64:64:c4:f4:
         93:7e:55:35:3e:07:ad:9c:b5:a7:01:89:14:a1:e8:2f:44:ea:
         8e:f6:66:79:1b:5d:51:7f:e2:41:b5:cc:97:da:62:db:68:40:
         8f:82:68:8c:5c:da:26:d2:1f:43:0c:ea:3b:14:6d:15:0e:d4:
         12:92:a0:89:8e:42:e5:1e:33:cc:55:64:fc:11:30:5b:f9:15:
         cd:47:61:b5:b3:b4:6d:26:ee:dd:68:6a:6b:b3:15:28:41:d4:
         ee:d5:60:cd:e7:59:3c:91:45:6c:85:79:78:87:a5:24:ca:0f:
         33:ec:b2:03:7f:03:81:a6:b5:8a:22:e2:98:30:32:06:5f:f9:
         7b:ec:c3:d6:a7:aa:e8:db:29:6b:e0:be:97:59:51:be:a0:69:
         00:61:86:e5:ea:93:fb:45:b4:6c:aa:d7:e3:01:54:e7:a6:7d:
         7f:9b:0a:d5

Very simple gateway:
<gateway name="my_freeswitch">
     <param name="username" value="1234"/>
     <param name="realm" value="sip.obolenskaya.su"/>
     <param name="password" value="my_strong_password"/>
     <param name="register-transport" value="tls"/>
</gateway>

Some relevant tls-setings:
<settings>
     <param name="tls" value="true"/>
     <param name="tls-only" value="true"/>
     <param name="tls-bind-params" value="transport=tls"/>
     <param name="tls-sip-port" value="5061"/>
     <param name="tls-cert-dir" value="/usr/local/freeswitch/certs"/>
     <param name="tls-passphrase" value=""/>
     <param name="tls-verify-date" value="true"/>
     <param name="tls-verify-policy" value="subjects_all"/>
     <param name="tls-verify-depth" value="2"/>
     param name="tls-verify-in-subjects" value="CN"/>
     <param name="tls-version" value="tlsv1.2"/>
</settings>

[NOTICE] sofia_reg.c:448 Registering my_freeswitch
tport.c:3257 tport_tsend() tport_tsend(0x7f9c2c094410) tpn =
tls/2a02:b184:3:1a11::2ba1:5061
tport.c:4046 tport_resolve() tport_resolve addrinfo =
[2a02:b184:3:1a11::2ba1]:5061
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7f9c2c094410): not
found by name tls/2a02:b184:3:1a11::2ba1:5061
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7f9c2c094410): not
found by name tls/2a02:b184:3:1a11::2ba1:5061
tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7f9c2c094410):
new secondary tport 0x7f9c2c068f10
tport_type_tcp.c:203 tport_tcp_init_secondary()
tport_tcp_init_secondary(0x7f9c2c068f10): Setting TCP_KEEPIDLE to 30
tport_type_tcp.c:209 tport_tcp_init_secondary()
tport_tcp_init_secondary(0x7f9c2c068f10): Setting TCP_KEEPINTVL to 30
tport_type_tls.c:683 tport_tls_connect() tport_tls_connect(0x7f9c2c068f10):
connecting to tls/[2a02:b184:3:1a11::2ba1]:5061/sips
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c2c068f10): reset timer
tport.c:3782 tport_queue() tport_queue(0x7f9c2c068f10): queueing
0x7f9c2c05d030 for tls/[2a02:b184:3:1a11::2ba1]:5061
tport.c:4160 tport_pend() tport_pend(0x7f9c2c068f10): pending
0x7f9c2c05d030 for tls/[2a02:b184:3:1a11::2ba1]:5061 (already 0)
tport_tls.c:956 tls_connect() tls_connect(0x7f9c2c068f10): events CONNECTING
tport_tls.c:956 tls_connect() tls_connect(0x7f9c2c068f10): events
NEGOTIATING
tport_tls.c:956 tls_connect() tls_connect(0x7f9c2c068f10): events
NEGOTIATING
tport_tls.c:599 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c068f10): TLS cipher chosen (name):
ECDHE-RSA-AES256-GCM-SHA384
tport_tls.c:601 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c068f10): TLS cipher chosen (version):
TLSv1/SSLv3
tport_tls.c:604 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c068f10): TLS cipher chosen
(bits/alg_bits): 256/256
tport_tls.c:607 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c068f10): TLS cipher chosen (description):
ECDHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH     Au=RSA  Enc=AESGCM(256)
Mac=AEAD

tport_tls.c:694 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c068f10): Peer Certificate Subject 0:
sip.obolenskaya.su
tport_tls.c:694 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c068f10): Peer Certificate Subject 1:
obolenskaya.su
tport.c:3923 tport_send_event() tport_send_event(0x7f9c2c068f10) - ready to
send to (tls/[2a02:b184:3:1a11::2ba1]:5061)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
0x7f9c2c0af610 0x7f9c2c09db90 637 (637)
tport.c:3594 tport_vsend() tport_vsend(0x7f9c2c068f10): 637 bytes of 637 to
tls/[2a02:b184:3:1a11::2ba1]:5061
tport.c:3492 tport_send_msg() tport_vsend returned 637
tport_type_tls.c:338 tport_tls_set_events()
tport_tls_set_events(0x7f9c2c068f10): logical events IN real IN
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c2c068f10): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7f9c2c068f10): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7f9c2c068f10)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f9c2c068f10):
tls_read() returned 653
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f9c2c068f10) msg
0x7f9c2c089c70 from (tls/[2a02:b184:3:1a11::2ba1]:5061) has 653 bytes,
veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7f9c2c068f10): msg
0x7f9c2c089c70 (653 bytes) from tls/[2a02:b184:3:1a11::2ba1]:5061/sips
next=(nil)
tport.c:4222 tport_release() tport_release(0x7f9c2c068f10): 0x7f9c2c05d030
by 0x7f9c2c0672c0 with 0x7f9c2c089c70
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c2c068f10): reset timer
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c2c068f10): reset timer
tport.c:3257 tport_tsend() tport_tsend(0x7f9c2c094410) tpn =
tls/2a02:b184:3:1a11::2ba1:5061
tport.c:4046 tport_resolve() tport_resolve addrinfo =
[2a02:b184:3:1a11::2ba1]:5061
tport.c:4677 tport_by_addrinfo() tport_by_addrinfo(0x7f9c2c094410): found
0x7f9c2c068f10 by name tls/2a02:b184:3:1a11::2ba1:5061
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
0x7f9c2c0af610 0x7f9c2c080930 912 (912)
tport.c:3594 tport_vsend() tport_vsend(0x7f9c2c068f10): 912 bytes of 912 to
tls/[2a02:b184:3:1a11::2ba1]:5061
tport.c:3492 tport_send_msg() tport_vsend returned 912
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c2c068f10): reset timer
tport.c:4160 tport_pend() tport_pend(0x7f9c2c068f10): pending
0x7f9c2c05d030 for tls/[2a02:b184:3:1a11::2ba1]:5061 (already 0)
tport.c:2773 tport_wakeup() tport_wakeup(0x7f9c2c068f10): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7f9c2c068f10)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f9c2c068f10):
tls_read() returned 654
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f9c2c068f10) msg
0x7f9c2c089c70 from (tls/[2a02:b184:3:1a11::2ba1]:5061) has 654 bytes,
veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7f9c2c068f10): msg
0x7f9c2c089c70 (654 bytes) from tls/[2a02:b184:3:1a11::2ba1]:5061/sips
next=(nil)
tport.c:4222 tport_release() tport_release(0x7f9c2c068f10): 0x7f9c2c05d030
by 0x7f9c2c067080 with 0x7f9c2c089c70
tport.c:4160 tport_pend() tport_pend(0x7f9c2c068f10): pending (nil) for
tls/[2a02:b184:3:1a11::2ba1]:5061 (already 0)
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c2c068f10): reset timer
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7f9c2c094410): events IN
tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7f9c2c094410):
new secondary tport 0x7f9c2c082960
tport_type_tcp.c:203 tport_tcp_init_secondary()
tport_tcp_init_secondary(0x7f9c2c082960): Setting TCP_KEEPIDLE to 30
tport_type_tcp.c:209 tport_tcp_init_secondary()
tport_tcp_init_secondary(0x7f9c2c082960): Setting TCP_KEEPINTVL to 30
tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x7f9c2c082960):
new connection from tls/[2a02:b184:3:1a11::2ba1]:52886/sips
tport_tls.c:956 tls_connect() tls_connect(0x7f9c2c082960): events
NEGOTIATING
tport_tls.c:956 tls_connect() tls_connect(0x7f9c2c082960): events
NEGOTIATING
tport_tls.c:599 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c082960): TLS cipher chosen (name):
ECDHE-RSA-AES256-GCM-SHA384
tport_tls.c:601 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c082960): TLS cipher chosen (version):
TLSv1/SSLv3
tport_tls.c:604 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c082960): TLS cipher chosen
(bits/alg_bits): 256/256
tport_tls.c:607 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c082960): TLS cipher chosen (description):
ECDHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH     Au=RSA  Enc=AESGCM(256)
Mac=AEAD

tport_tls.c:694 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c082960): Peer Certificate Subject 0:
sip.obolenskaya.su
tport_tls.c:694 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c082960): Peer Certificate Subject 1:
obolenskaya.su
tport_tls.c:721 tls_post_connection_check()
tls_post_connection_check(0x7f9c2c082960): Peer Subject Mismatch (incoming
connection)
tport.c:2090 tport_close() tport_close(0x7f9c2c082960):
tls/[2a02:b184:3:1a11::2ba1]:52886/sips
tport.c:2263 tport_set_secondary_timer() tport(0x7f9c2c082960): set timer
at 0 ms because zap


If this setting changed the mismatch disappear:
<param name="tls-verify-policy" value="subjects_out|in"/>

[NOTICE] sofia_reg.c:448 Registering my_freeswitch
tport.c:3257 tport_tsend() tport_tsend(0x7f9c1c046570) tpn =
tls/2a02:b184:3:1a11::2ba1:5061
tport.c:4046 tport_resolve() tport_resolve addrinfo =
[2a02:b184:3:1a11::2ba1]:5061
tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0x7f9c1c046570): not
found by name tls/2a02:b184:3:1a11::2ba1:5061
tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7f9c1c046570):
new secondary tport 0x7f9c1c07fa60
tport_type_tcp.c:203 tport_tcp_init_secondary()
tport_tcp_init_secondary(0x7f9c1c07fa60): Setting TCP_KEEPIDLE to 30
tport_type_tcp.c:209 tport_tcp_init_secondary()
tport_tcp_init_secondary(0x7f9c1c07fa60): Setting TCP_KEEPINTVL to 30
tport_type_tls.c:683 tport_tls_connect() tport_tls_connect(0x7f9c1c07fa60):
connecting to tls/[2a02:b184:3:1a11::2ba1]:5061/sips
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c1c07fa60): reset timer
tport.c:3782 tport_queue() tport_queue(0x7f9c1c07fa60): queueing
0x7f9c1c03a460 for tls/[2a02:b184:3:1a11::2ba1]:5061
tport.c:4160 tport_pend() tport_pend(0x7f9c1c07fa60): pending
0x7f9c1c03a460 for tls/[2a02:b184:3:1a11::2ba1]:5061 (already 0)
tport_tls.c:956 tls_connect() tls_connect(0x7f9c1c07fa60): events CONNECTING
tport_tls.c:956 tls_connect() tls_connect(0x7f9c1c07fa60): events
NEGOTIATING
tport_tls.c:956 tls_connect() tls_connect(0x7f9c1c07fa60): events
NEGOTIATING
tport_tls.c:599 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c07fa60): TLS cipher chosen (name):
ECDHE-RSA-AES256-GCM-SHA384
tport_tls.c:601 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c07fa60): TLS cipher chosen (version):
TLSv1/SSLv3
tport_tls.c:604 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c07fa60): TLS cipher chosen
(bits/alg_bits): 256/256
tport_tls.c:607 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c07fa60): TLS cipher chosen (description):
ECDHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH     Au=RSA  Enc=AESGCM(256)
Mac=AEAD
tport_tls.c:694 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c07fa60): Peer Certificate Subject 0:
sip.obolenskaya.su
tport_tls.c:694 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c07fa60): Peer Certificate Subject 1:
obolenskaya.su
tport.c:3923 tport_send_event() tport_send_event(0x7f9c1c07fa60) - ready to
send to (tls/[2a02:b184:3:1a11::2ba1]:5061)
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
0x7f9c1c0a4750 0x7f9c1c058cb0 637 (637)
tport.c:3594 tport_vsend() tport_vsend(0x7f9c1c07fa60): 637 bytes of 637 to
tls/[2a02:b184:3:1a11::2ba1]:5061
tport.c:3492 tport_send_msg() tport_vsend returned 637
tport_type_tls.c:338 tport_tls_set_events()
tport_tls_set_events(0x7f9c1c07fa60): logical events IN real IN
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c1c07fa60): reset timer
tport.c:2773 tport_wakeup() tport_wakeup(0x7f9c1c07fa60): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7f9c1c07fa60)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f9c1c07fa60):
tls_read() returned 653
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f9c1c07fa60) msg
0x7f9c1c09c620 from (tls/[2a02:b184:3:1a11::2ba1]:5061) has 653 bytes,
veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7f9c1c07fa60): msg
0x7f9c1c09c620 (653 bytes) from tls/[2a02:b184:3:1a11::2ba1]:5061/sips
next=(nil)
tport.c:4222 tport_release() tport_release(0x7f9c1c07fa60): 0x7f9c1c03a460
by 0x7f9c1c0a4500 with 0x7f9c1c09c620
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c1c07fa60): reset timer
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c1c07fa60): reset timer
tport.c:3257 tport_tsend() tport_tsend(0x7f9c1c046570) tpn =
tls/2a02:b184:3:1a11::2ba1:5061
tport.c:4046 tport_resolve() tport_resolve addrinfo =
[2a02:b184:3:1a11::2ba1]:5061
tport.c:4677 tport_by_addrinfo() tport_by_addrinfo(0x7f9c1c046570): found
0x7f9c1c07fa60 by name tls/2a02:b184:3:1a11::2ba1:5061
tport_type_tls.c:534 tport_tls_send() tport_tls_writevec: vec
0x7f9c1c0a4750 0x7f9c1c09c620 912 (912)
tport.c:3594 tport_vsend() tport_vsend(0x7f9c1c07fa60): 912 bytes of 912 to
tls/[2a02:b184:3:1a11::2ba1]:5061
tport.c:3492 tport_send_msg() tport_vsend returned 912
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c1c07fa60): reset timer
tport.c:4160 tport_pend() tport_pend(0x7f9c1c07fa60): pending
0x7f9c1c058cb0 for tls/[2a02:b184:3:1a11::2ba1]:5061 (already 0)
tport.c:2773 tport_wakeup() tport_wakeup(0x7f9c1c07fa60): events IN
tport.c:2864 tport_recv_event() tport_recv_event(0x7f9c1c07fa60)
tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f9c1c07fa60):
tls_read() returned 654
tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7f9c1c07fa60) msg
0x7f9c1c09ca30 from (tls/[2a02:b184:3:1a11::2ba1]:5061) has 654 bytes,
veclen = 1
tport.c:3023 tport_deliver() tport_deliver(0x7f9c1c07fa60): msg
0x7f9c1c09ca30 (654 bytes) from tls/[2a02:b184:3:1a11::2ba1]:5061/sips
next=(nil)
tport.c:4222 tport_release() tport_release(0x7f9c1c07fa60): 0x7f9c1c058cb0
by 0x7f9c1c09d470 with 0x7f9c1c09ca30
tport.c:4160 tport_pend() tport_pend(0x7f9c1c07fa60): pending (nil) for
tls/[2a02:b184:3:1a11::2ba1]:5061 (already 0)
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c1c07fa60): reset timer
tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7f9c1c046570): events IN
tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7f9c1c046570):
new secondary tport 0x7f9c1c058cb0
tport_type_tcp.c:203 tport_tcp_init_secondary()
tport_tcp_init_secondary(0x7f9c1c058cb0): Setting TCP_KEEPIDLE to 30
tport_type_tcp.c:209 tport_tcp_init_secondary()
tport_tcp_init_secondary(0x7f9c1c058cb0): Setting TCP_KEEPINTVL to 30
tport_type_tls.c:610 tport_tls_accept() tport_tls_accept(0x7f9c1c058cb0):
new connection from tls/[2a02:b184:3:1a11::2ba1]:40928/sips
tport_tls.c:956 tls_connect() tls_connect(0x7f9c1c058cb0): events
NEGOTIATING
tport_tls.c:956 tls_connect() tls_connect(0x7f9c1c058cb0): events
NEGOTIATING
tport_tls.c:599 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c058cb0): TLS cipher chosen (name):
ECDHE-RSA-AES256-GCM-SHA384
tport_tls.c:601 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c058cb0): TLS cipher chosen (version):
TLSv1/SSLv3
tport_tls.c:604 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c058cb0): TLS cipher chosen
(bits/alg_bits): 256/256
tport_tls.c:607 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c058cb0): TLS cipher chosen (description):
ECDHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH     Au=RSA  Enc=AESGCM(256)
Mac=AEAD

tport_tls.c:694 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c058cb0): Peer Certificate Subject 0:
sip.obolenskaya.su
tport_tls.c:694 tls_post_connection_check()
tls_post_connection_check(0x7f9c1c058cb0): Peer Certificate Subject 1:
obolenskaya.su
tport.c:2296 tport_set_secondary_timer() tport(0x7f9c1c058cb0): reset timer

Now the peer can call me.

But what should I do that subjects_in will be checked correctly?

Thank you!

Xenia Obolenskaya
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/86b1edab/attachment-0001.html>

From gb at cm.nl  Tue Jun 19 07:22:48 2018
From: gb at cm.nl (Grant Bagdasarian)
Date: Tue, 19 Jun 2018 07:22:48 +0000
Subject: [Freeswitch-users] Pyrun limitations
In-Reply-To: <CAC7ttFF0CoeK3GVGY7REnQzs4nopx=FmjkRh2U+0k6f99hXPJw@mail.gmail.com>
References: <0f8a299688cb45d5b78a49b98301c047@cm.nl>
 <CAC7ttFF0CoeK3GVGY7REnQzs4nopx=FmjkRh2U+0k6f99hXPJw@mail.gmail.com>
Message-ID: <d80255d2a5d94cbbb5d4f8817d41777c@cm.nl>

Hi Gregor,

We’re very satisfied with the library and have been running it in production for more than 9 months now. I can’t remember having any major issues we couldn’t overcome.
The interface is more programmer friendly compared to the managed_esl library which is packaged with freeswitch. We have mainly used it to control our freeswitch cluster and originate calls, so no real voice(ivr) application programming with complex flows etc.

Definitely recommend it if you’re having a tough time using the managed_esl library.

Regards,

Grant Bagdasarian

Senior Developer

cm.com<https://cm.com>


From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Gregor Nanger
Sent: woensdag 13 juni 2018 23:03
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Pyrun limitations

Grant, how are u satisfied with neventsocket library?
On Wed, Jun 13, 2018, 22:51 Grant Bagdasarian <gb at cm.nl<mailto:gb at cm.nl>> wrote:
Hello,

We’re currently using the NEventSocket library to control our freeswitch servers for starting python scripts (using a backgroundjob and pyrun py_module_name) which internally use the freeswitch.py library to originate a call.
However, we’re running into some performance limitations where we can’t seem to get more than 5 call setups per second.

To quote the documentation (https://freeswitch.org/confluence/display/FREESWITCH/mod_python):
“A single python interpreter is spawned at module startup and used for the lifetime of the freeswitch process.”
“Each thread swaps in its "thread state" before executing python code and then swaps it out when finished. Also during blocking calls into freeswitch, a thread will swap out its thread state in order to not block other threads, and then swap it in after the blocking call to freeswitch has finished.”

How does this relate to the number of threads freeswitch is allowed to start and run python scripts? If there are hard limitations set, can these be increased? I couldn’t find hard numbers in the source files, so this may be controlled somewhere else?

Thanks.

Grant Bagdasarian

Senior Developer



cm.com<https://cm.com>




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

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

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


CTO
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si<http://www.infomedia.si/>


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/1eccd192/attachment.html>

From douglas.davenport at gmail.com  Tue Jun 19 01:34:48 2018
From: douglas.davenport at gmail.com (Douglas Davenport)
Date: Mon, 18 Jun 2018 21:34:48 -0400
Subject: [Freeswitch-users] No T38 SDP in 200 OK after glare
In-Reply-To: <CAEJMVkD6hYU58Gg1W_p71WxpPwPQQjXocxsOKU2grvY83MdEzA@mail.gmail.com>
References: <CAMDsFF3GR02KLStT0JeGkfeXFtoVHOE1XYMSpYBJsjEyC4VeCw@mail.gmail.com>
 <CAEJMVkD6hYU58Gg1W_p71WxpPwPQQjXocxsOKU2grvY83MdEzA@mail.gmail.com>
Message-ID: <CAMDsFF1yy1v-ywR4EokeMxN+=ivx2qsWSnec-oF-CddKgWB1Qg@mail.gmail.com>

Yes, it terminates on freeswitch (rxfax).


On Mon, Jun 18, 2018 at 6:57 PM Brian West <brian at freeswitch.com> wrote:

> Does this call terminate on FreeSWITCH or are you doing passthru?
>
> /b
>
>
> On Mon, Jun 18, 2018 at 12:12 PM, Douglas Davenport <
> douglas.davenport at gmail.com> wrote:
>
>> After T38 reinvite glare on an inbound call, the provider resends their
>> reinvite after a random delay  (they own the call-ID). Freeswitch correctly
>> responds with 200 OK because fax_enable_t38=true however it does not
>> include SDP.  Are there any settings that would affect this behavior or is
>> this a bug? I'm aware that bug reports go to JIRA but I'm asking first to
>> know if this is actually a bug or I'm doing something wrong. Here is the
>> call flow. Note the starred 200 OK with no SDP:
>>
>>
>>                                 Provider
>>    Freeswitch
>>       ──────────┬─────────          ──────────┬─────────
>>   12:07:02.067864   │ ─────── INVITE (SDP) ─────> │
>>   12:07:02.068654   │ <─────── 100 Trying ─────── │
>>   12:07:03.304619   │ <────── 200 OK (SDP) ────── │
>>   12:07:03.305554   │ ──────────── ACK ─────────> │
>>   12:07:06.506575   │ ────── INVITE (T38 SDP)───> │
>>   12:07:06.718790   │ <────── INVITE (T38 SDP)─── │
>>   12:07:06.719063   │ ───── 491 Request Pendi ──> │
>>   12:07:06.879340   │ <─────── 100 Trying ─────── │
>>   12:07:06.881990   │ <─────────── ACK ────────── │
>>   12:07:06.943138   │ <──── 491 Request Pendi ─── │
>>   12:07:06.943356   │ ──────────── ACK ─────────> │
>>   12:07:07.133006   │ <────INVITE (T38 SDP) ───── │
>>   12:07:07.133313   │ ──────── 100 Trying ──────> │
>>   12:07:07.146828   │ ─*************─ 200 OK ─[*NO SDP*]─> │
>> *****************
>>   12:07:07.280018   │ <─────────── ACK ────────── │
>>   12:07:16.012689   │ <─────────── BYE ────────── │
>>   12:07:16.026995   │ ────────── 200 OK ────────> │
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
>
>
> --
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: brian at freeswitch.com
>
> Mobile: 918-424-9378
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: https://www.facebook.com/signalwireinc?src=email]
> <https://www.facebook.com/freeswitch> [image:
> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180618/b83c0315/attachment-0001.html>

From douglas.davenport at gmail.com  Tue Jun 19 14:31:15 2018
From: douglas.davenport at gmail.com (Douglas Davenport)
Date: Tue, 19 Jun 2018 10:31:15 -0400
Subject: [Freeswitch-users] No T38 SDP in 200 OK after glare
In-Reply-To: <CAEJMVkD6hYU58Gg1W_p71WxpPwPQQjXocxsOKU2grvY83MdEzA@mail.gmail.com>
References: <CAMDsFF3GR02KLStT0JeGkfeXFtoVHOE1XYMSpYBJsjEyC4VeCw@mail.gmail.com>
 <CAEJMVkD6hYU58Gg1W_p71WxpPwPQQjXocxsOKU2grvY83MdEzA@mail.gmail.com>
Message-ID: <CAMDsFF3=SZs6jUn5kejKtnS2F_iSfs3xGAMi4_e3qwLWn3bfVQ@mail.gmail.com>

Yes, it terminates on freeswitch (rxfax).

On Mon, Jun 18, 2018 at 6:57 PM Brian West <brian at freeswitch.com> wrote:

> Does this call terminate on FreeSWITCH or are you doing passthru?
>
> /b
>
>
> On Mon, Jun 18, 2018 at 12:12 PM, Douglas Davenport <
> douglas.davenport at gmail.com> wrote:
>
>> After T38 reinvite glare on an inbound call, the provider resends their
>> reinvite after a random delay  (they own the call-ID). Freeswitch correctly
>> responds with 200 OK because fax_enable_t38=true however it does not
>> include SDP.  Are there any settings that would affect this behavior or is
>> this a bug? I'm aware that bug reports go to JIRA but I'm asking first to
>> know if this is actually a bug or I'm doing something wrong. Here is the
>> call flow. Note the starred 200 OK with no SDP:
>>
>>
>>                                 Provider
>>    Freeswitch
>>       ──────────┬─────────          ──────────┬─────────
>>   12:07:02.067864   │ ─────── INVITE (SDP) ─────> │
>>   12:07:02.068654   │ <─────── 100 Trying ─────── │
>>   12:07:03.304619   │ <────── 200 OK (SDP) ────── │
>>   12:07:03.305554   │ ──────────── ACK ─────────> │
>>   12:07:06.506575   │ ────── INVITE (T38 SDP)───> │
>>   12:07:06.718790   │ <────── INVITE (T38 SDP)─── │
>>   12:07:06.719063   │ ───── 491 Request Pendi ──> │
>>   12:07:06.879340   │ <─────── 100 Trying ─────── │
>>   12:07:06.881990   │ <─────────── ACK ────────── │
>>   12:07:06.943138   │ <──── 491 Request Pendi ─── │
>>   12:07:06.943356   │ ──────────── ACK ─────────> │
>>   12:07:07.133006   │ <────INVITE (T38 SDP) ───── │
>>   12:07:07.133313   │ ──────── 100 Trying ──────> │
>>   12:07:07.146828   │ ─*************─ 200 OK ─[*NO SDP*]─> │
>> *****************
>>   12:07:07.280018   │ <─────────── ACK ────────── │
>>   12:07:16.012689   │ <─────────── BYE ────────── │
>>   12:07:16.026995   │ ────────── 200 OK ────────> │
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
>
>
> --
>
> Brian West | Co-founder and Developer
>
> Need Commercial support? email sales at freeswitch.com
>
> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
> <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
>
> Email: brian at freeswitch.com
>
> Mobile: 918-424-9378
>
> Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
>
> [image: https://www.facebook.com/signalwireinc?src=email]
> <https://www.facebook.com/freeswitch> [image:
> https://twitter.com/freeswitch] <https://twitter.com/freeswitch>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/6ab3fd44/attachment.html>

From francesco at delagarda.com  Tue Jun 19 07:46:15 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Tue, 19 Jun 2018 09:46:15 +0200
Subject: [Freeswitch-users] session.cause and session.causecode
In-Reply-To: <CAEeV61hkpie1GZWaspccjnCu5F=4KKORYtFCKxHhVyLcjCv4aA@mail.gmail.com>
References: <056701d4061b$521ec960$f65c5c20$@delagarda.com>
 <CAEeV61hkpie1GZWaspccjnCu5F=4KKORYtFCKxHhVyLcjCv4aA@mail.gmail.com>
Message-ID: <3d9a01d407a1$9b07dd60$d1179820$@delagarda.com>

Thanks Matt .. you saved me… by dumping all the vars I realized that what I really needed to understand why the call had failed cas not the cause or causecode, but the “originateCause”

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Matt Broad
Sent: lunedì 18 giugno 2018 15:45
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] session.cause and session.causecode

 

sessOut.causeCode should be sessOut.causecode

 

you can get the sessOut object values by adding:

 

console_log("notice", JSON.stringify(sessOut) ); 

 

 

 

thanks

Matt




Matt Broad

Tel:  <tel:+44%2020%203011%201313> +44 (0)203 011 1313

Web:  <http://www.supportedbusiness.com/> www.supportedbusiness.com

  <http://www.supportedbusiness.com/coming-soon-logo.png> 

The content of this email, including any files or documents attached, are confidential and may also be legally privileged, protected from disclosure and/or protected by other legal rules. It is written without prejudice and is intended for the individual specified in the message only. The views and opinions included in this email belong to their author and do not necessarily mirror the views and opinions of the company. Full security of this email cannot be ensured as, despite our best efforts, the data included in emails could be infected, intercepted, or corrupted. Do not share any part of this message with any third party, without a written consent of the sender. If you have received this message in error, please notify us and remove it from your system.

 

On 17 June 2018 at 10:12, Francesco Facco de Lagarda <francesco at delagarda.com <mailto:francesco at delagarda.com> > wrote:

‘Morning all

I can’t get session.cause and session.causecode to work.

All I get is false for the code and blank for the cause!

 

Im running FS 1.6.20

 

This is a simplified version of my JS script

 

if (session.ready()) {

 

    session.answer();

 

    var dialedNum = "3332094333"; 

                console_log("notice", "*********************** DIALING");

                var sessOut = new Session("sofia/gateway/realtoneFXO/" +  dialedNum + "@192.168.0.216:5060 <http://192.168.0.216:5060> ");

                console_log("notice", "*********************** WAITING FOR SESSION READY ..");

 

                

                if (sessOut.ready()) {

               console_log("notice", "*********************** SESSION READY");

               bridge(session, sessOut);

                } else {

                               console_log("notice", "************************* disconnect cause:" + sessOut.cause + ":" + sessOut.causeCode);

                }

                sessOut.hangup();

                session.hangup();

    console_log("notice", "*********************** CALL ENDED");

 

}

 

 


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

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

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

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/6125f9f4/attachment-0001.html>

From francesco at delagarda.com  Tue Jun 19 10:28:37 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Tue, 19 Jun 2018 12:28:37 +0200
Subject: [Freeswitch-users] Using H264 with Verto
In-Reply-To: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>
References: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>
Message-ID: <3db101d407b8$499c3d60$dcd4b820$@delagarda.com>

My ten cent’s worth .. this is my verto profiles…

 

 

<profiles>

                <profile name="default-v4">

                               <param name="bind-local" value="$${local_ip_v4}:8081"/>

                               <param name="bind-local" value="$${local_ip_v4}:8082" secure="true"/>

                               <param name="force-register-domain" value="$${domain}"/>

                               <param name="secure-combined" value="$${certs_dir}/wss.pem"/>

                               <param name="secure-chain" value="$${certs_dir}/wss.pem"/>

                               <param name="userauth" value="true"/>

                               <!-- setting this to true will allow anyone to register even with no account so use with care -->

                               <param name="blind-reg" value="false"/>

                               <param name="mcast-ip" value="224.1.1.1"/>

                               <param name="mcast-port" value="1337"/>

                               <param name="rtp-ip" value="$${local_ip_v4}"/>

                               <param name="ext-rtp-ip" value="************MY PUBLIC UP ADDRESS********"/>

                               <param name="local-network" value="localnet.auto"/>

                               <param name="outbound-codec-string" value="opus,vp8"/>

                               <param name="inbound-codec-string" value="opus,vp8"/>

 

                               <param name="apply-candidate-acl" value="localnet.auto"/>

                               <param name="apply-candidate-acl" value="wan_v4.auto"/>

                               <param name="apply-candidate-acl" value="rfc1918.auto"/>

                               <param name="apply-candidate-acl" value="any_v4.auto"/>

                               <param name="timer-name" value="soft"/>

 

                </profile>

 

                <profile name="default-v6">

                               <param name="bind-local" value="[$${local_ip_v6}]:8081"/>

                               <param name="bind-local" value="[$${local_ip_v6}]:8082" secure="true"/>

                               <param name="force-register-domain" value="$${domain}"/>

                               <param name="secure-combined" value="$${certs_dir}/wss.pem"/>

                               <param name="secure-chain" value="$${certs_dir}/wss.pem"/>

                               <param name="userauth" value="true"/>

                               <!-- setting this to true will allow anyone to register even with no account so use with care -->

                               <param name="blind-reg" value="false"/>

                               <param name="rtp-ip" value="$${local_ip_v6}"/>

                               <param name="ext-rtp-ip" value="************MY PUBLIC UP ADDRESS********"/>

                               <param name="outbound-codec-string" value="opus,vp8"/>

                               <param name="inbound-codec-string" value="opus,vp8"/>

 

                               <param name="apply-candidate-acl" value="wan_v6.auto"/>

                               <param name="apply-candidate-acl" value="rfc1918.auto"/>

                               <param name="apply-candidate-acl" value="any_v6.auto"/>

                               <param name="apply-candidate-acl" value="wan_v4.auto"/>

                               <param name="apply-candidate-acl" value="any_v4.auto"/>

                               <param name="timer-name" value="soft"/>

 

                </profile>

</profiles>

 

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Chad Phillips
Sent: venerdì 15 giugno 2018 20:51
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Using H264 with Verto

 

I have a slightly older install where I’m using H264 with Verto, and it works fine. However, I’m not able to get it working on latest master.

 

I’m guessing there’s some esoteric setting I’m missing, but can’t seem to find it. I do have the following in verto.conf.xml:

 

      <param name="outbound-codec-string" value="opus,h264"/>

      <param name=“inbound-codec-string” value="opus,h264"/>

 

I also have mod_h26x disabled, as I remember that’s passthrough only and causes issues when loaded in this case.

 

When I try in Chrome I see it offering H264 in the SDP, but the answer SDP contains this, and the stream is audio only: 

 

m=video 0 UDP/TLS/RTP/SAVPF 19

 

What am I missing??

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/fa53ab67/attachment-0001.html>

From francesco at delagarda.com  Tue Jun 19 11:07:18 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Tue, 19 Jun 2018 13:07:18 +0200
Subject: [Freeswitch-users] gateway selection by prefix
Message-ID: <3de901d407bd$b15be4f0$1413aed0$@delagarda.com>

I have 3 gateways:
Telecom Italia on an analogue adapter

MessageNet (sip provider)
Skype

 

How can I select gateway by a prefix THAT THEN GETS DROPPED before Dialing..

 

Example I would like to have

All calls beginning with 9 Sent to MessageNet , DROPPING THE "9"

All calls beginning with 7 sent to Skype, dropping the "7"

Otherwise use telecom

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/14b3f66b/attachment-0001.html>

From mike at jerris.com  Tue Jun 19 14:56:28 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 Jun 2018 10:56:28 -0400
Subject: [Freeswitch-users] Memory leak while using BGAPI
In-Reply-To: <25D2EC755404B4409F263AC6D050FEBB2BCCEFCE@AZ-FFEXMB03.global.avaya.com>
References: <25D2EC755404B4409F263AC6D050FEBB2BCCD8C4@AZ-FFEXMB03.global.avaya.com>
 <108CFF47-7EC4-4562-AB51-1DE1061AFFA8@jerris.com>
 <25D2EC755404B4409F263AC6D050FEBB2BCCEFCE@AZ-FFEXMB03.global.avaya.com>
Message-ID: <19224735-8F41-410D-941E-8B2A49D7260E@jerris.com>

It is being pushed into the queue to be processed.  It is free’d after it is processed at the other end of that queue.  For an orderly shutdown without the -vg flag you may have a few events that we dont bother flushing out of the queue and freed.  Under normal operation those are processed and free’d.  If that number is not large and growing its fine, I think running with -vg will make that go away, if so its not a problem.  A number of memory free cleanups on shutdown are skipped without the -vg flag, and I think thats all you are seeing.

Mike


> On Jun 19, 2018, at 2:26 AM, Mody, Darshan (Darshan) <darshanmody at avaya.com> wrote:
> 
> Hi Micheal,
>  
> We find below code in the switch_event.c. Is there a specific reason as to why are we setting the pointers eventp and event as NULL and not destroying the event?
>  
> static switch_status_t switch_event_queue_dispatch_event(switch_event_t **eventp)
> {
>  
>                 switch_event_t *event = *eventp;
>  
>                 if (!SYSTEM_RUNNING) {
>                                 return SWITCH_STATUS_FALSE;
>                 }
>  
>                 while (event) {
>                                 int launch = 0;
>  
>                                 switch_mutex_lock(EVENT_QUEUE_MUTEX);
>  
>                                 if (!PENDING && switch_queue_size(EVENT_DISPATCH_QUEUE) > (unsigned int)(DISPATCH_QUEUE_LEN * DISPATCH_THREAD_COUNT)) {
>                                                 if (SOFT_MAX_DISPATCH + 1 < MAX_DISPATCH) {
>                                                                 launch++;
>                                                                 PENDING++;
>                                                                 switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "launch = %d & PENDING = %d\n", launch,PENDING);
>                                                 }
>                                 }
>  
>                                 switch_mutex_unlock(EVENT_QUEUE_MUTEX);
>  
>                                 if (launch) {
>                                                 if (SOFT_MAX_DISPATCH + 1 < MAX_DISPATCH) {
>                                                                 switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Dispatching %d Thread\n", (SOFT_MAX_DISPATCH + 1));
>                                                                 switch_event_launch_dispatch_threads(SOFT_MAX_DISPATCH + 1);
>                                                 }
>  
>                                                 switch_mutex_lock(EVENT_QUEUE_MUTEX);
>                                                 PENDING--;
>                                                 switch_mutex_unlock(EVENT_QUEUE_MUTEX);
>                                 }
>  
>                                 *eventp = NULL;
>                                 switch_queue_push(EVENT_DISPATCH_QUEUE, event);
>                                 event = NULL;
>  
>                 }
>  
>                 return SWITCH_STATUS_SUCCESS;
> }
>  
> Thanks
> Darshan
>   <>
> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Tuesday, June 19, 2018 2:34 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Memory leak while using BGAPI
>  
> If that event is being consumed by something that is blocking maybe?  Is it always that number or if you do many more does it get bigger?  This could just be an event that isnt consumed yet when you shut down and not a big problem.
>  
> 
> 
> On Jun 18, 2018, at 2:17 AM, Mody, Darshan (Darshan) <darshanmody at avaya.com <mailto:darshanmody at avaya.com>> wrote:
>  
> Hi,
>  
> We are observing considerable memory leak while using LUA and BGAPI of mod_command.c. Below is the valgrind’s output
>  
> ==21863== 123,170 (45,496 direct, 77,674 indirect) bytes in 517 blocks are definitely lost in loss record 2,370 of 2,398
> ==21863==    at 0x4C29C23: malloc (vg_replace_malloc.c:299)
> ==21863==    by 0x548707C: switch_event_create_subclass_detailed (switch_event.c:736)
> ==21863==    by 0x11631389: bgapi_exec (mod_commands.c:5151)
> ==21863==    by 0x56EEE8F: dummy_worker (thread.c:151)
> ==21863==    by 0x7677E24: start_thread (in /usr/lib64/libpthread-2.17.so)
> ==21863==    by 0x805734C: clone (in /usr/lib64/libc-2.17.so
>  
> ==21863== 143,478 (18,392 direct, 125,086 indirect) bytes in 209 blocks are definitely lost in loss record 2,374 of 2,398
> ==21863==    at 0x4C29C23: malloc (vg_replace_malloc.c:299)
> ==21863==    by 0x548707C: switch_event_create_subclass_detailed (switch_event.c:736)
> ==21863==    by 0x542AADE: switch_core_session_exec (switch_core_session.c:2874)
> ==21863==    by 0x542B108: switch_core_session_execute_application_get_flags (switch_core_session.c:2742)
> ==21863==    by 0x5510C10: CoreSession::execute(char const*, char const*) (switch_cpp.cpp:777)
> ==21863==    by 0x16FED72C: ??? (mod_lua_wrap.cpp:7289)
> ==21863==    by 0x17217323: ??? (in /usr/lib64/liblua-5.1.so)
> ==21863==    by 0x17221E56: ??? (in /usr/lib64/liblua-5.1.so)
> ==21863==    by 0x1721774C: ??? (in /usr/lib64/liblua-5.1.so)
> ==21863==    by 0x17216A6D: ??? (in /usr/lib64/liblua-5.1.so)
> ==21863==    by 0x172178D9: ??? (in /usr/lib64/liblua-5.1.so)
> ==21863==    by 0x1721344C: lua_pcall (in /usr/lib64/liblua-5.1.so)
>  
> Has some one also observed behavior while using BGAPI?
>  
> Thanks
> Darshan 
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://urldefense.proofpoint.com/v2/url?u=https-3A__freeswitch.com_&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=cf87DDQfp5SEnNdhK6qztdg7bCeSemewRWYu-gCSUuM&e=>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://urldefense.proofpoint.com/v2/url?u=https-3A__freeswitch.com_oss&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=km3nKWxpOpy0aSeC6s3a2J04SwIbQR8nJF_1h-3w9iE&e=>
> https://freeswitch.org/confluence <https://urldefense.proofpoint.com/v2/url?u=https-3A__freeswitch.org_confluence&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=Mj2ZnsEi90La30FIpe-hLKtfSogOiqMoiuljmGH73SU&e=>
> https://cluecon.com <https://urldefense.proofpoint.com/v2/url?u=https-3A__cluecon.com_&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=ZFkKDP5CjqxuU0TAIPZlU9zUpys5ch7RgYWUEGaVTsk&e=>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <https://urldefense.proofpoint.com/v2/url?u=http-3A__lists.freeswitch.org_mailman_listinfo_freeswitch-2Dusers&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=kXdKsgCA_UpVrUS_jBuy4lBpWP5RT9qazUatzWIslsE&e=>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <https://urldefense.proofpoint.com/v2/url?u=http-3A__lists.freeswitch.org_mailman_options_freeswitch-2Dusers&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=J8fwO1L0LvupkX0THYA5RMB66VVagvVYay4ie0-2s68&e=>
> https://freeswitch.com <https://urldefense.proofpoint.com/v2/url?u=https-3A__freeswitch.com_&d=DwMFaQ&c=BFpWQw8bsuKpl1SgiZH64Q&r=bsEULbVnjelD7InzgsegHBEbtXzaIDagy9EuEhJrKfQ&m=doV3DZKRNDthayX1enWG7PgnBOG04sTRZ_pM8b5Xfm0&s=cf87DDQfp5SEnNdhK6qztdg7bCeSemewRWYu-gCSUuM&e=>
>  
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/653a9dc1/attachment-0001.html>

From gregor at infomedia.si  Tue Jun 19 14:55:44 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 19 Jun 2018 16:55:44 +0200
Subject: [Freeswitch-users] Pyrun limitations
In-Reply-To: <d80255d2a5d94cbbb5d4f8817d41777c@cm.nl>
References: <0f8a299688cb45d5b78a49b98301c047@cm.nl>
 <CAC7ttFF0CoeK3GVGY7REnQzs4nopx=FmjkRh2U+0k6f99hXPJw@mail.gmail.com>
 <d80255d2a5d94cbbb5d4f8817d41777c@cm.nl>
Message-ID: <CAC7ttFHO8Lg0zn-joC0aOvjdZ8psfCq55yhuMvS=Ap3VsV0rxg@mail.gmail.com>

Just curios how it works for you, thank you.

We are using it too without any issues. Mainly for event subscribing, not
controling fs and it works good. Not to mention architecture of this lib
compared to managed_esl.

2018-06-19 9:22 GMT+02:00 Grant Bagdasarian <gb at cm.nl>:

> Hi Gregor,
>
>
>
> We’re very satisfied with the library and have been running it in
> production for more than 9 months now. I can’t remember having any major
> issues we couldn’t overcome.
>
> The interface is more programmer friendly compared to the managed_esl
> library which is packaged with freeswitch. We have mainly used it to
> control our freeswitch cluster and originate calls, so no real voice(ivr)
> application programming with complex flows etc.
>
>
>
> Definitely recommend it if you’re having a tough time using the
> managed_esl library.
>
>
>
> Regards,
>
>
>
> Grant Bagdasarian
>
> Senior Developer
>
> cm.com
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Gregor Nanger
> *Sent:* woensdag 13 juni 2018 23:03
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Pyrun limitations
>
>
>
> Grant, how are u satisfied with neventsocket library?
>
> On Wed, Jun 13, 2018, 22:51 Grant Bagdasarian <gb at cm.nl> wrote:
>
> Hello,
>
>
>
> We’re currently using the NEventSocket library to control our freeswitch
> servers for starting python scripts (using a backgroundjob and pyrun
> py_module_name) which internally use the freeswitch.py library to originate
> a call.
>
> However, we’re running into some performance limitations where we can’t
> seem to get more than 5 call setups per second.
>
>
>
> To quote the documentation (https://freeswitch.org/
> confluence/display/FREESWITCH/mod_python):
>
> “A single python interpreter is spawned at module startup and used for the
> lifetime of the freeswitch process.”
>
> “Each thread swaps in its "thread state" before executing python code and
> then swaps it out when finished. Also during blocking calls into
> freeswitch, a thread will swap out its thread state in order to not block
> other threads, and then swap it in after the blocking call to freeswitch
> has finished.”
>
>
>
> How does this relate to the number of threads freeswitch is allowed to
> start and run python scripts? If there are hard limitations set, can these
> be increased? I couldn’t find hard numbers in the source files, so this may
> be controlled somewhere else?
>
>
>
> Thanks.
>
>
>
> Grant Bagdasarian
>
> Senior Developer
>
>
>
> cm.com
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> --
>
> *Gregor Nanger*
>
>
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/f766177a/attachment.html>

From mike at jerris.com  Tue Jun 19 16:32:26 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 Jun 2018 12:32:26 -0400
Subject: [Freeswitch-users] Wrong IP in ACK
In-Reply-To: <CAHtxdDd9cJB_JCcmhS_iaCicgHe+AZBpqYMkCi+i9R8w4261Rw@mail.gmail.com>
References: <CAHtxdDcXfiD73Av6wrgybCZk5zDx6k2Tnjif-PWEuDGf3Zy_cQ@mail.gmail.com>
 <CAHtxdDeD1iX0U3-PC2yOyMy6yuFxK=n5PKd1gRnphnvDfkqQ8A@mail.gmail.com>
 <D26E5F38-7916-4D66-86E3-D813FC14DDFD@jerris.com>
 <CAHtxdDd9cJB_JCcmhS_iaCicgHe+AZBpqYMkCi+i9R8w4261Rw@mail.gmail.com>
Message-ID: <5798A77D-BB91-4E5E-9F41-A5968F08193C@jerris.com>

if you are using fs_path and expecting it to be anything other than a hardcoded outbound proxy then its not going to work like you expect.



> On Jun 19, 2018, at 3:06 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
> 
> If FS ignores contact values and use fs_path like "Record-Route" SIP header, then ACK must be send to same IP used for INVITE.
> Without this whole session is broken.
> Please look call log. Call is dropped by remote end because ACK not received.
> 
> вт, 19 июн. 2018 г. в 2:18, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> Setting fs_path like that is used to force a locked in outbound proxy for that dialog.  The behavior you are seeing is intended behavior.
> 
> 
>> On Jun 12, 2018, at 4:45 AM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
>> 
>> root of issue
>> ACK send to IP resolved via DNS for value defined as "fs_path", rather than defined in contact string.
>> 
>> more details
>> https://freeswitch.org/jira/browse/FS-11190 <https://freeswitch.org/jira/browse/FS-11190>
>> 
>> вт, 12 июн. 2018 г. в 10:30, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>>:
>> Are anybody experiences issue when for some of calls  ACK send to other IP.
>> http://prntscr.com/jtxkbz <http://prntscr.com/jtxkbz>
>> in this case FS in bypass media with two profiles (1 to internet, 2 intranet).
>> INVITE is send to DNS name that is resolved to
>> 10.0.9.32
>> 10.0.9.33
>> 10.0.9.34
>> 10.0.9.35
>> FS 1.6.19
>> 
>> Contact on 200 message is correct
>> 
>> SIP/2.0 200 OK
>> Via: SIP/2.0/TCP 10.0.9.51:1 <http://10.0.9.51:1/>2000;rport=39049;branch=z9hG4bKBH60NQjSZj36a
>> From: "Anonymous" <sip:+12022553656 at 10.0.9.51 <mailto:sip%3A%2B12022553656 at 10.0.9.51>>;tag=rH5cBQarr559D
>> To: <sip:13013688630 <tel:(301)%20368-8630>@metropolis.ack.one>;tag=UDv7UXv3B53Da
>> Call-ID: b0e4d7e1-e79e-1236-8cb9-02420a000933
>> CSeq: 123998414 INVITE
>> Contact: <sip:13013688630 <tel:(301)%20368-8630>@10.0.9.33:11000;transport=tcp>
>> User-Agent: proxy.ackone.com <http://proxy.ackone.com/>
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> 
>> Now for future investigate FS upgrader to 1.6.20 and enabled siptrace and loglevel
>> 
>> Sergey Safarov
> 
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>_________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/a7faaa51/attachment-0001.html>

From chad at apartmentlines.com  Tue Jun 19 15:04:43 2018
From: chad at apartmentlines.com (Chad Phillips)
Date: Tue, 19 Jun 2018 09:04:43 -0600
Subject: [Freeswitch-users] Using H264 with Verto
In-Reply-To: <3D1DB6A6-D120-4C75-93C3-4D010DE4273A@jerris.com>
References: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>
 <3D1DB6A6-D120-4C75-93C3-4D010DE4273A@jerris.com>
Message-ID: <CAJ0sghTvzfrWjihy9hJmX0LWZKCMPNG+oFhOHBsW4PrVdaEVKw@mail.gmail.com>

Yep, that was it, thanks!

I added a section to the Verto docs to clarify all the steps to get H264
working: https://evoluxbr.github.io/verto-docs/tut/adding-h264-support.html

On Mon, Jun 18, 2018 at 4:21 PM Michael Jerris <mike at jerris.com> wrote:

> load mod_av?  if thats not the issue, I’d need to see a debug log w/ sip
> trace to tell you more.
>
>
> > On Jun 15, 2018, at 2:50 PM, Chad Phillips <chad at apartmentlines.com>
> wrote:
> >
> > I have a slightly older install where I’m using H264 with Verto, and it
> works fine. However, I’m not able to get it working on latest master.
> >
> > I’m guessing there’s some esoteric setting I’m missing, but can’t seem
> to find it. I do have the following in verto.conf.xml:
> >
> >       <param name="outbound-codec-string" value="opus,h264"/>
> >       <param name=“inbound-codec-string” value="opus,h264"/>
> >
> > I also have mod_h26x disabled, as I remember that’s passthrough only and
> causes issues when loaded in this case.
> >
> > When I try in Chrome I see it offering H264 in the SDP, but the answer
> SDP contains this, and the stream is audio only:
> >
> > m=video 0 UDP/TLS/RTP/SAVPF 19
> >
> > What am I missing??
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/ebc916d7/attachment.html>

From mike at jerris.com  Tue Jun 19 16:34:26 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 Jun 2018 12:34:26 -0400
Subject: [Freeswitch-users] Using H264 with Verto
In-Reply-To: <CAJ0sghTvzfrWjihy9hJmX0LWZKCMPNG+oFhOHBsW4PrVdaEVKw@mail.gmail.com>
References: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>
 <3D1DB6A6-D120-4C75-93C3-4D010DE4273A@jerris.com>
 <CAJ0sghTvzfrWjihy9hJmX0LWZKCMPNG+oFhOHBsW4PrVdaEVKw@mail.gmail.com>
Message-ID: <00FF96CE-F893-4B56-8FC9-BE667D1F217F@jerris.com>

Awesome.  No problem.

> On Jun 19, 2018, at 11:04 AM, Chad Phillips <chad at apartmentlines.com> wrote:
> 
> Yep, that was it, thanks!
> 
> I added a section to the Verto docs to clarify all the steps to get H264 working: https://evoluxbr.github.io/verto-docs/tut/adding-h264-support.html <https://evoluxbr.github.io/verto-docs/tut/adding-h264-support.html>
> On Mon, Jun 18, 2018 at 4:21 PM Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> load mod_av?  if thats not the issue, I’d need to see a debug log w/ sip trace to tell you more.
> 
> 
> > On Jun 15, 2018, at 2:50 PM, Chad Phillips <chad at apartmentlines.com <mailto:chad at apartmentlines.com>> wrote:
> > 
> > I have a slightly older install where I’m using H264 with Verto, and it works fine. However, I’m not able to get it working on latest master.
> > 
> > I’m guessing there’s some esoteric setting I’m missing, but can’t seem to find it. I do have the following in verto.conf.xml:
> > 
> >       <param name="outbound-codec-string" value="opus,h264"/>
> >       <param name=“inbound-codec-string” value="opus,h264"/>
> > 
> > I also have mod_h26x disabled, as I remember that’s passthrough only and causes issues when loaded in this case.
> > 
> > When I try in Chrome I see it offering H264 in the SDP, but the answer SDP contains this, and the stream is audio only: 
> > 
> > m=video 0 UDP/TLS/RTP/SAVPF 19
> > 
> > What am I missing??
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/5acdba15/attachment.html>

From asilva at wirelessmundi.com  Tue Jun 19 16:49:02 2018
From: asilva at wirelessmundi.com (antonio)
Date: Tue, 19 Jun 2018 18:49:02 +0200
Subject: [Freeswitch-users] Sending SIP Messages from LUA
In-Reply-To: <3bb9f14310dc3c83d826f0dfbbba332f@themenz.biz>
References: <4757e45e0bfeb5156b2a0d149ccaf6b5@themenz.biz>
 <a8f3c338805581ccd97384465fcfba8d@themenz.biz>
 <3bb9f14310dc3c83d826f0dfbbba332f@themenz.biz>
Message-ID: <7087fe9d-d4bd-ea4e-355c-70462b056b7f@wirelessmundi.com>

i had the same problem.. i solve it setting a sip uri in "to"  where the 
user is registered... i'm doing it on fs master, not sure if it will 
work on 1.6


example:
event:addHeader("to", "sip:200 at 192.168.10.2:38592");


To get the uri in lua:

contact = api:execute("sofia_contact", "user/200 at lab.local");
profile = contact:match("sofia/(.*)/") or "";
to = contact:match("sofia/.*/(.*)") or "";


Probably there is a easier way...


On 03/13/2018 04:03 PM, Ronnie Beck wrote:
>
>
> Hi All,
>
> I am trying to write a LUA script which will slot into a chatplan 
> whose goal is to simple take a simple text message (SIMPLE MESSAGE) 
> and then distribute it to multiple recipients.  I wrote a script which 
> takes some arguments (sender, recipieant list and the body) then would 
> generate new SIP Messages to be sent to each of the recipient.  The 
> code I use to generate the message (which is more or less the example 
> given in mod_sms for sending an SMS, not exactly what I want to do):
>
>
> local event = freeswitch.Event("CUSTOM", "SMS::SEND_MESSAGE");
> event:addHeader("proto", "global");
> event:addHeader("dest_proto", "sip");
> event:addHeader("from", source);
> event:addHeader("to", domain.."/"..recipient);
> event:addHeader("type", "text/plain");
> event:addHeader("skip_global_process", "true");
> event:addBody( body );
> event:fire();
>
>
> This gives the error:
>
> [WARNING] sofia_presence.c:221 Not sending to local box for 
> XXXX at 10.11.12.13 <#NOP>
> [ERR] sofia_presence.c:272 Chat proto [sip]
> from [<sip:YYYY at 10.11.12.13>;tag=uUQudJKEl1SKjq8wdjqwi68768]
> to [XXXX at 10.11.12.13]
> Some message text
> Nobody to send to: Profile internal
>
>
> But the message actually arrives successfully.  This is fine but I 
> would prefer not to have lots of error messages.  So I tried to find a 
> more correct way to do this.  I tried the following (based on what I 
> could find in freeswitch 1.6.20 source code):
>
> local event = freeswitch.Event("SWITCH_EVENT_SEND_MESSAGE");
> event:addHeader("user", recipient );
> event:addHeader("host", domain );
> event:addHeader("profile", domain );
> event:addHeader("subject", "SIMPLE MESSAGE");
> event:addHeader("content-type", "text/plain");
> event:addHeader("from", source);
> event:addBody( body );
> event:fire();
>
> But this gives the same error.
>
> How can I send a simple SIP Message with LUA to a locally registered user?
>
> Many thanks,
>
> Aaron
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Saludos / Regards / Cumprimentos
Anónio Silva

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/82838990/attachment-0001.html>

From kowalma at gmail.com  Tue Jun 19 16:50:03 2018
From: kowalma at gmail.com (Marcin Kowalczyk)
Date: Tue, 19 Jun 2018 18:50:03 +0200
Subject: [Freeswitch-users] DNS cache on invite/register
Message-ID: <CAFg=aF5pMuE6-dDPV3M+4Ur0Ui+JVBnL9GCX+yS51QNbCv0Xeg@mail.gmail.com>

Hi,

I'm having problem setting-up trunk with provider who does have muliple
sip-proxies with not shared nonce db.

sip.freeconet.pl resolves to 3 diffrent IP and FS does not cache/remeber IP
that send back request.

So we have a case
FS->P1: Invite
P1 -> FS -> 407 + nonce
FS does again DNS lookup and
FS->P2 Invite with nonce from P1
P2->FS 407
FS fails call.

Is there a way to force FS to send all messages within dialog to one IP
only ?

Regards
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/50138e73/attachment-0001.html>

From ap at gen-ip.fr  Tue Jun 19 17:09:54 2018
From: ap at gen-ip.fr (Alexis)
Date: Tue, 19 Jun 2018 19:09:54 +0200
Subject: [Freeswitch-users] gateway selection by prefix
In-Reply-To: <3de901d407bd$b15be4f0$1413aed0$@delagarda.com>
References: <3de901d407bd$b15be4f0$1413aed0$@delagarda.com>
Message-ID: <0774e250-ad23-d4b9-02b7-ff4ad25c5852@gen-ip.fr>

Hi,

You can do that with this dialplan :

<extension name='MessageNet'>
     <condition field='${destination_number}' expression='^9(\d+)'>
         <action application='bridge' data='sofia/gateway/your MessageNet gateway name/$1'/>
     </condition>
</extension>

<extension name='Skype'>
     <condition field='${destination_number}' expression='^7(\d+)'>
         <action application='bridge' data='sofia/gateway/your Skype gateway name/$1'/>
     </condition>
</extension>

<extension name='TelecomItalia'>
     <condition>
         <action application='bridge' data='sofia/gateway/your TelecomItalia gateway name/${destination_number}'/>
     </condition>
</extension>

You can find more informations here : 
https://freeswitch.org/confluence/display/FREESWITCH/Regular+Expression#RegularExpression-ClusteringvsCapturing

And this page give you exactly what you want : 
https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-Conditions 
("Example 1: Capturing Digits")

Alexis

Le 19/06/2018 à 13:07, Francesco Facco de Lagarda a écrit :
>
> I have 3 gateways:
> Telecom Italia on an analogue adapter
>
> MessageNet (sip provider)
> Skype
>
> How can I select gateway by a prefix THAT THEN GETS DROPPED before 
> Dialing..
>
> Example I would like to have
>
> All calls beginning with 9 Sent to MessageNet , DROPPING THE “9”
>
> All calls beginning with 7 sent to Skype, dropping the “7”
>
> Otherwise use telecom
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/6fc98bdd/attachment.html>

From mike at jerris.com  Tue Jun 19 18:26:53 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 Jun 2018 14:26:53 -0400
Subject: [Freeswitch-users] DNS cache on invite/register
In-Reply-To: <CAFg=aF5pMuE6-dDPV3M+4Ur0Ui+JVBnL9GCX+yS51QNbCv0Xeg@mail.gmail.com>
References: <CAFg=aF5pMuE6-dDPV3M+4Ur0Ui+JVBnL9GCX+yS51QNbCv0Xeg@mail.gmail.com>
Message-ID: <1B9C6462-565F-433A-A878-5B4DF35173BB@jerris.com>

if they are doing nonces on a shared realm, they should be sharing those nonces across their cluster.  If they are not, every time it hits a diff member of the cluster you will get a stale nonce and have to go through that juggle again.  This is just a side effect of how sip auth works.


> On Jun 19, 2018, at 12:50 PM, Marcin Kowalczyk <kowalma at gmail.com> wrote:
> 
> Hi,
> 
> I'm having problem setting-up trunk with provider who does have muliple sip-proxies with not shared nonce db. 
> 
> sip.freeconet.pl <http://sip.freeconet.pl/> resolves to 3 diffrent IP and FS does not cache/remeber IP that send back request. 
> 
> So we have a case 
> FS->P1: Invite
> P1 -> FS -> 407 + nonce
> FS does again DNS lookup and 
> FS->P2 Invite with nonce from P1
> P2->FS 407
> FS fails call.
> 
> Is there a way to force FS to send all messages within dialog to one IP only ?
> 
> Regards
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/606a47d8/attachment.html>

From magnus.kelly at gmail.com  Tue Jun 19 18:40:57 2018
From: magnus.kelly at gmail.com (Magnus)
Date: Tue, 19 Jun 2018 19:40:57 +0100
Subject: [Freeswitch-users] Using H264 with Verto
In-Reply-To: <00FF96CE-F893-4B56-8FC9-BE667D1F217F@jerris.com>
References: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>
 <3D1DB6A6-D120-4C75-93C3-4D010DE4273A@jerris.com>
 <CAJ0sghTvzfrWjihy9hJmX0LWZKCMPNG+oFhOHBsW4PrVdaEVKw@mail.gmail.com>
 <00FF96CE-F893-4B56-8FC9-BE667D1F217F@jerris.com>
Message-ID: <36D9E69E-90EF-4845-87D2-3D0141A21ABC@gmail.com>

 Chad, good addition to documents, by chance have you also discovered how to enable mod_av, but not use H264? I ask as I observe that enabling mod_av adds the h264 codec even if it’s not listed in vars.xml. In my particular case I would like to use mod_av but select when to use h.264 by destination as opposed to it being included in every sip sdp, but I have not yet worked out how to do so - by chance have you any tips from your testing ?

Regards
Magnus


> On 19 Jun 2018, at 17:34, Michael Jerris <mike at jerris.com> wrote:
> 
> Awesome.  No problem.
> 
>> On Jun 19, 2018, at 11:04 AM, Chad Phillips <chad at apartmentlines.com> wrote:
>> 
>> Yep, that was it, thanks!
>> 
>> I added a section to the Verto docs to clarify all the steps to get H264 working: https://evoluxbr.github.io/verto-docs/tut/adding-h264-support.html
>> 
>>> On Mon, Jun 18, 2018 at 4:21 PM Michael Jerris <mike at jerris.com> wrote:
>>> load mod_av?  if thats not the issue, I’d need to see a debug log w/ sip trace to tell you more.
>>> 
>>> 
>>> > On Jun 15, 2018, at 2:50 PM, Chad Phillips <chad at apartmentlines.com> wrote:
>>> > 
>>> > I have a slightly older install where I’m using H264 with Verto, and it works fine. However, I’m not able to get it working on latest master.
>>> > 
>>> > I’m guessing there’s some esoteric setting I’m missing, but can’t seem to find it. I do have the following in verto.conf.xml:
>>> > 
>>> >       <param name="outbound-codec-string" value="opus,h264"/>
>>> >       <param name=“inbound-codec-string” value="opus,h264"/>
>>> > 
>>> > I also have mod_h26x disabled, as I remember that’s passthrough only and causes issues when loaded in this case.
>>> > 
>>> > When I try in Chrome I see it offering H264 in the SDP, but the answer SDP contains this, and the stream is audio only: 
>>> > 
>>> > m=video 0 UDP/TLS/RTP/SAVPF 19
>>> > 
>>> > What am I missing??
>>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/92d2acde/attachment.html>

From chad at apartmentlines.com  Tue Jun 19 20:16:53 2018
From: chad at apartmentlines.com (Chad Phillips)
Date: Tue, 19 Jun 2018 14:16:53 -0600
Subject: [Freeswitch-users] Using H264 with Verto
In-Reply-To: <36D9E69E-90EF-4845-87D2-3D0141A21ABC@gmail.com>
References: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>
 <3D1DB6A6-D120-4C75-93C3-4D010DE4273A@jerris.com>
 <CAJ0sghTvzfrWjihy9hJmX0LWZKCMPNG+oFhOHBsW4PrVdaEVKw@mail.gmail.com>
 <00FF96CE-F893-4B56-8FC9-BE667D1F217F@jerris.com>
 <36D9E69E-90EF-4845-87D2-3D0141A21ABC@gmail.com>
Message-ID: <CAJ0sghQVB9Wjzqt-JbczvZ3yGO9=9g1ad56p-cD7QvWWTkAwHA@mail.gmail.com>

No idea how H264 plays with SIP. In Verto, codecs are tried in the order
they are listed in verto.conf.xml, pretty sure it uses the first codec that
the client supports.

On Tue, Jun 19, 2018 at 2:06 PM Magnus <magnus.kelly at gmail.com> wrote:

>  Chad, good addition to documents, by chance have you also discovered how
> to enable mod_av, but not use H264? I ask as I observe that enabling mod_av
> adds the h264 codec even if it’s not listed in vars.xml. In my particular
> case I would like to use mod_av but select when to use h.264 by destination
> as opposed to it being included in every sip sdp, but I have not yet worked
> out how to do so - by chance have you any tips from your testing ?
>
> Regards
> Magnus
>
>
> On 19 Jun 2018, at 17:34, Michael Jerris <mike at jerris.com> wrote:
>
> Awesome.  No problem.
>
> On Jun 19, 2018, at 11:04 AM, Chad Phillips <chad at apartmentlines.com>
> wrote:
>
> Yep, that was it, thanks!
>
> I added a section to the Verto docs to clarify all the steps to get H264
> working:
> https://evoluxbr.github.io/verto-docs/tut/adding-h264-support.html
>
> On Mon, Jun 18, 2018 at 4:21 PM Michael Jerris <mike at jerris.com> wrote:
>
>> load mod_av?  if thats not the issue, I’d need to see a debug log w/ sip
>> trace to tell you more.
>>
>>
>> > On Jun 15, 2018, at 2:50 PM, Chad Phillips <chad at apartmentlines.com>
>> wrote:
>> >
>> > I have a slightly older install where I’m using H264 with Verto, and it
>> works fine. However, I’m not able to get it working on latest master.
>> >
>> > I’m guessing there’s some esoteric setting I’m missing, but can’t seem
>> to find it. I do have the following in verto.conf.xml:
>> >
>> >       <param name="outbound-codec-string" value="opus,h264"/>
>> >       <param name=“inbound-codec-string” value="opus,h264"/>
>> >
>> > I also have mod_h26x disabled, as I remember that’s passthrough only
>> and causes issues when loaded in this case.
>> >
>> > When I try in Chrome I see it offering H264 in the SDP, but the answer
>> SDP contains this, and the stream is audio only:
>> >
>> > m=video 0 UDP/TLS/RTP/SAVPF 19
>> >
>> > What am I missing??
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/57f78c3a/attachment-0001.html>

From andrew.keil at visytel.com  Wed Jun 20 03:20:39 2018
From: andrew.keil at visytel.com (Andrew Keil)
Date: Wed, 20 Jun 2018 03:20:39 +0000
Subject: [Freeswitch-users] Recording skipping silence on outbound call
	recording
Message-ID: <MEXPR01MB122366DCA0BD9B96BED24509F3770@MEXPR01MB1223.ausprd01.prod.outlook.com>

To FreeSWITCH Users,

I managed to find and fix the issue and thought I would provide my solution here since it may help someone else.

Prior to the recording I set the following inside my outbound Lua script:

session:setVariable("record_fill_cng","1400")

Note: 1400 matches the waste_resources default setting inside the switch_ivr_record_file (switch_ivr_play_say.c)

This fixed the issue and now my recording contains the silence periods correctly (which is what I was after).  I also noticed that since the Lua destination script that my outbound was calling contains periods of session:sleep(...), when waiting for DTMF, and it was this period that was not being recorded.  Now with this fix everything is fine and those periods of session:sleep(...) are matched inside my outbound recorded audio file.

Andrew

--- My original e-mail to FreeSWITCH Users ---

To FreeSWITCH Users,

I am attempting to make a recording of an outbound call using the originate API call which starts a Lua script which then simply records to disk exactly what it hears.

I just noticed an interesting issue (or feature) where the recording skips any silence.  What I am aiming for is for no silence to be skipped inside the recording, if someone can provide some settings for me to try I would be more than happy to experiment further.

The settings for the Lua session:recordFile are length of recording: 60 seconds; silence threshold: 300; silence seconds: 60 seconds  {essentially termination on silence is disabled since the silence seconds = recording seconds}

I also tried with silence threshold = 0 and silence is still skipped inside the recording.

I also tried the setting inside the appropriate profile: <param name="suppress-cng" value="true"/> and this made no difference, silence is still skipped inside the recording.

My setup is one single FreeSWITCH 1.6.20 (production version) box with a simple internal gateway setup to land on the same FreeSWITCH IVR (just for testing).

API command: originate {origination_uuid=121e20a6-1245-439a-8527-53efab009334-out-1,origination_caller_id_number=FLOODOUT,call_timeout=15,ignore_early_media=true,return_ring_ready=true,suppress_cng=true}sofia/gateway/visytel-pc-ivr/01111111111 FLOODOUT

The SDP inside my log is similar to this for the outbound and inbound (Note: The outbound call is performing the recording):

121e20a6-1245-439a-8527-53efab009334-out-1 Local SDP:
121e20a6-1245-439a-8527-53efab009334-out-1 v=0
121e20a6-1245-439a-8527-53efab009334-out-1 o=FreeSWITCH 1529274701 1529274702 IN IP4 192.168.15.15
121e20a6-1245-439a-8527-53efab009334-out-1 s=FreeSWITCH
121e20a6-1245-439a-8527-53efab009334-out-1 c=IN IP4 192.168.15.15
121e20a6-1245-439a-8527-53efab009334-out-1 t=0 0
121e20a6-1245-439a-8527-53efab009334-out-1 m=audio 25994 RTP/AVP 9 8 0 101
121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:9 G722/8000
121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:8 PCMA/8000
121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:0 PCMU/8000
121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:101 telephone-event/8000
121e20a6-1245-439a-8527-53efab009334-out-1 a=fmtp:101 0-16
121e20a6-1245-439a-8527-53efab009334-out-1 a=silenceSupp:off - - - -
121e20a6-1245-439a-8527-53efab009334-out-1 a=ptime:20
121e20a6-1245-439a-8527-53efab009334-out-1 a=sendrecv

6da97c7c-1052-4f78-8dde-83901147a0c1 2018-06-18 15:44:55.596296 [DEBUG] sofia.c:7094 Remote SDP:
6da97c7c-1052-4f78-8dde-83901147a0c1 v=0
6da97c7c-1052-4f78-8dde-83901147a0c1 o=FreeSWITCH 1529274701 1529274702 IN IP4 192.168.15.15
6da97c7c-1052-4f78-8dde-83901147a0c1 s=FreeSWITCH
6da97c7c-1052-4f78-8dde-83901147a0c1 c=IN IP4 192.168.15.15
6da97c7c-1052-4f78-8dde-83901147a0c1 t=0 0
6da97c7c-1052-4f78-8dde-83901147a0c1 m=audio 25994 RTP/AVP 9 8 0 101
6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:9 G722/8000
6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:8 PCMA/8000
6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:0 PCMU/8000
6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:101 telephone-event/8000
6da97c7c-1052-4f78-8dde-83901147a0c1 a=fmtp:101 0-16
6da97c7c-1052-4f78-8dde-83901147a0c1 a=silenceSupp:off - - - -
6da97c7c-1052-4f78-8dde-83901147a0c1 a=ptime:20
6da97c7c-1052-4f78-8dde-83901147a0c1

Kind Regards,

Andrew Keil

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180620/487d44d7/attachment.html>

From chernetsov.artyom at gmail.com  Wed Jun 20 07:21:59 2018
From: chernetsov.artyom at gmail.com (Artyom Chernetzov)
Date: Wed, 20 Jun 2018 10:21:59 +0300
Subject: [Freeswitch-users] NORMAL_CIRCUIT_CONGESTION error limiting
	simultaneous calls
Message-ID: <CAEduXd+n+JpU42E8SeB-A_JNodPTNNBHx6o44mQLesGMN1FjmA@mail.gmail.com>

We got NORMAL_CIRCUIT_CONGESTION error trying to perform 100 simultaneous
calls, and only 40 simultaneous calls actually happen (all other fail with
NORMAL_CIRCUIT_CONGESTION). We have 1gbit/s network, all default rtp ports
are open, and switch.conf.xml sessions-per-second=1000 (most freeswitch and
gateway configs are default). We run dedicated server. Provider tell that
500 lines are available. Is it possible that I missed some config that
could limit gateway channels, or this is definitely provider issue?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180620/7863f85c/attachment.html>

From kowalma at gmail.com  Wed Jun 20 04:37:42 2018
From: kowalma at gmail.com (Marcin Kowalczyk)
Date: Wed, 20 Jun 2018 06:37:42 +0200
Subject: [Freeswitch-users] DNS cache on invite/register
In-Reply-To: <mailman.1553.1529445091.1211.freeswitch-users@lists.freeswitch.org>
References: <mailman.1553.1529445091.1211.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAFg=aF5TCEfT-55nwP0C9Ayc1BBpF3qTZBXFezpsCxf65u1+oA@mail.gmail.com>

Hi,

 I agree they should have shared nonce, but still I think FS should use
same IP for initial and challenged invite unless TTL for domain is not
expired.

Is there a way to force it to such behaviour? I've tried with fs_path but
seems this does not work for challenged invite neither

Somebody posted bug FS-8820 on same case but with different carrier.

Regards

Date: Tue, 19 Jun 2018 14:26:53 -0400
> if they are doing nonces on a shared realm, they should be sharing those
> nonces across their cluster.  If they are not, every time it hits a diff
> member of the cluster you will get a stale nonce and have to go through
> that juggle again.  This is just a side effect of how sip auth works.
>
>
> On Jun 19, 2018, at 12:50 PM, Marcin Kowalczyk <kowalma at gmail.com> wrote:
>
> Hi,
>
> I'm having problem setting-up trunk with provider who does have muliple
> sip-proxies with not shared nonce db.
>
> sip.freeconet.pl resolves to 3 diffrent IP and FS does not cache/remeber
> IP that send back request.
>
> So we have a case
> FS->P1: Invite
> P1 -> FS -> 407 + nonce
> FS does again DNS lookup and
> FS->P2 Invite with nonce from P1
> P2->FS 407
> FS fails call.
>
> Is there a way to force FS to send all messages within dialog to one IP
> only ?
>
> Regards
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180620/119f5e20/attachment.html>

From leandro.campos at nvoip.com.br  Tue Jun 19 22:26:15 2018
From: leandro.campos at nvoip.com.br (Leandro Campos)
Date: Tue, 19 Jun 2018 19:26:15 -0300
Subject: [Freeswitch-users] Problems with NAT
Message-ID: <3DBAB7D6-5210-415A-B5FD-28114D7A93D9@nvoip.com.br>

Recently we are testing the Freeswitch with the Astpp for change us servers (we use asterisk with a2billing).
In the migration process, we check some customers don’t register or don’t receive inbound calls. In a2billing works fine.
I think is something about NAT. Someone help me?


Atenciosamente,

Leandro Campos
CEO
(11) 4118-6267
 
https://www.nvoip.com.br

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/d55f2ec0/attachment-0001.html>

From mike at jerris.com  Tue Jun 19 22:12:35 2018
From: mike at jerris.com (Michael Jerris)
Date: Tue, 19 Jun 2018 18:12:35 -0400
Subject: [Freeswitch-users] Using H264 with Verto
In-Reply-To: <36D9E69E-90EF-4845-87D2-3D0141A21ABC@gmail.com>
References: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>
 <3D1DB6A6-D120-4C75-93C3-4D010DE4273A@jerris.com>
 <CAJ0sghTvzfrWjihy9hJmX0LWZKCMPNG+oFhOHBsW4PrVdaEVKw@mail.gmail.com>
 <00FF96CE-F893-4B56-8FC9-BE667D1F217F@jerris.com>
 <36D9E69E-90EF-4845-87D2-3D0141A21ABC@gmail.com>
Message-ID: <898C30C2-3EB8-4163-BBC2-E1234352EB88@jerris.com>

Just configure codec preferences appropriately.  We filter prefs based on whats loaded so that would be why you wouldnt see it until module is loaded.

> On Jun 19, 2018, at 2:40 PM, Magnus <magnus.kelly at gmail.com> wrote:
> 
>  Chad, good addition to documents, by chance have you also discovered how to enable mod_av, but not use H264? I ask as I observe that enabling mod_av adds the h264 codec even if it’s not listed in vars.xml. In my particular case I would like to use mod_av but select when to use h.264 by destination as opposed to it being included in every sip sdp, but I have not yet worked out how to do so - by chance have you any tips from your testing ?
> 
> Regards
> Magnus
> 
> 
> On 19 Jun 2018, at 17:34, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> 
>> Awesome.  No problem.
>> 
>>> On Jun 19, 2018, at 11:04 AM, Chad Phillips <chad at apartmentlines.com <mailto:chad at apartmentlines.com>> wrote:
>>> 
>>> Yep, that was it, thanks!
>>> 
>>> I added a section to the Verto docs to clarify all the steps to get H264 working: https://evoluxbr.github.io/verto-docs/tut/adding-h264-support.html <https://evoluxbr.github.io/verto-docs/tut/adding-h264-support.html>
>>> On Mon, Jun 18, 2018 at 4:21 PM Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>> load mod_av?  if thats not the issue, I’d need to see a debug log w/ sip trace to tell you more.
>>> 
>>> 
>>> > On Jun 15, 2018, at 2:50 PM, Chad Phillips <chad at apartmentlines.com <mailto:chad at apartmentlines.com>> wrote:
>>> > 
>>> > I have a slightly older install where I’m using H264 with Verto, and it works fine. However, I’m not able to get it working on latest master.
>>> > 
>>> > I’m guessing there’s some esoteric setting I’m missing, but can’t seem to find it. I do have the following in verto.conf.xml:
>>> > 
>>> >       <param name="outbound-codec-string" value="opus,h264"/>
>>> >       <param name=“inbound-codec-string” value="opus,h264"/>
>>> > 
>>> > I also have mod_h26x disabled, as I remember that’s passthrough only and causes issues when loaded in this case.
>>> > 
>>> > When I try in Chrome I see it offering H264 in the SDP, but the answer SDP contains this, and the stream is audio only: 
>>> > 
>>> > m=video 0 UDP/TLS/RTP/SAVPF 19
>>> > 
>>> > What am I missing??
>>> 
>> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180619/b55663b8/attachment.html>

From andrew.keil at visytel.com  Wed Jun 20 22:52:42 2018
From: andrew.keil at visytel.com (Andrew Keil)
Date: Wed, 20 Jun 2018 22:52:42 +0000
Subject: [Freeswitch-users] NORMAL_CIRCUIT_CONGESTION error
	limiting	simultaneous calls
In-Reply-To: <CAEduXd+n+JpU42E8SeB-A_JNodPTNNBHx6o44mQLesGMN1FjmA@mail.gmail.com>
References: <CAEduXd+n+JpU42E8SeB-A_JNodPTNNBHx6o44mQLesGMN1FjmA@mail.gmail.com>
Message-ID: <MEXPR01MB1223B52CA2CFA0AAD649EC98F3770@MEXPR01MB1223.ausprd01.prod.outlook.com>

Artyom.

This could be also related to the speed (Call Attempts Per Second - CAPS) that you are making the outbound calls.  If you slow down the outbound calls being made and increase their duration does this increase your simultaneous calls?

My guess is similar to your one that this is a provider issue.  I know BT in the UK restrict the CAPS on their SIP Trunks to avoid network errors (or basically their SBCs cannot handle more CAPS).

Obviously if the provider is generating the NORMAL_CIRCUIT_CONGESTION error (check the SIP Trace) then it is a provider issue and they should be involved.

Andrew

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of Artyom Chernetzov
Sent: Wednesday, 20 June 2018 5:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] NORMAL_CIRCUIT_CONGESTION error limiting simultaneous calls

We got NORMAL_CIRCUIT_CONGESTION error trying to perform 100 simultaneous calls, and only 40 simultaneous calls actually happen (all other fail with NORMAL_CIRCUIT_CONGESTION). We have 1gbit/s network, all default rtp ports are open, and switch.conf<http://switch.conf>.xml sessions-per-second=1000 (most freeswitch and gateway configs are default). We run dedicated server. Provider tell that 500 lines are available. Is it possible that I missed some config that could limit gateway channels, or this is definitely provider issue?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180620/6d407bee/attachment.html>

From magnus.kelly at gmail.com  Wed Jun 20 15:38:51 2018
From: magnus.kelly at gmail.com (Magnus Kelly)
Date: Wed, 20 Jun 2018 16:38:51 +0100
Subject: [Freeswitch-users] Using H264 with Verto
In-Reply-To: <898C30C2-3EB8-4163-BBC2-E1234352EB88@jerris.com>
References: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>
 <3D1DB6A6-D120-4C75-93C3-4D010DE4273A@jerris.com>
 <CAJ0sghTvzfrWjihy9hJmX0LWZKCMPNG+oFhOHBsW4PrVdaEVKw@mail.gmail.com>
 <00FF96CE-F893-4B56-8FC9-BE667D1F217F@jerris.com>
 <36D9E69E-90EF-4845-87D2-3D0141A21ABC@gmail.com>
 <898C30C2-3EB8-4163-BBC2-E1234352EB88@jerris.com>
Message-ID: <CAAPhwbdFkZ1_NGutcv4sKxpNSAEF6z3fZfxp7Nas2vVAdTQ34w@mail.gmail.com>

Well the issue I have is I trying to remove H.264 from the codecs listed in
the SDP,  as although its not listed in the codec section of vars.xml at
all, enabling mod_av appears to mean that by default H.264 is added in the
SDP for each outbound SIP invite, resulting in calls to PSTN SIP G/W
failing with unacceptable codec. Advice on how to prevent or make selective
per SIP profile most welcome.
Thanks
Magnus

On Wed, 20 Jun 2018 at 14:49, Michael Jerris <mike at jerris.com> wrote:

> Just configure codec preferences appropriately.  We filter prefs based on
> whats loaded so that would be why you wouldnt see it until module is loaded.
>
> On Jun 19, 2018, at 2:40 PM, Magnus <magnus.kelly at gmail.com> wrote:
>
>  Chad, good addition to documents, by chance have you also discovered how
> to enable mod_av, but not use H264? I ask as I observe that enabling mod_av
> adds the h264 codec even if it’s not listed in vars.xml. In my particular
> case I would like to use mod_av but select when to use h.264 by destination
> as opposed to it being included in every sip sdp, but I have not yet worked
> out how to do so - by chance have you any tips from your testing ?
>
> Regards
> Magnus
>
>
> On 19 Jun 2018, at 17:34, Michael Jerris <mike at jerris.com> wrote:
>
> Awesome.  No problem.
>
> On Jun 19, 2018, at 11:04 AM, Chad Phillips <chad at apartmentlines.com>
> wrote:
>
> Yep, that was it, thanks!
>
> I added a section to the Verto docs to clarify all the steps to get H264
> working:
> https://evoluxbr.github.io/verto-docs/tut/adding-h264-support.html
>
> On Mon, Jun 18, 2018 at 4:21 PM Michael Jerris <mike at jerris.com> wrote:
>
>> load mod_av?  if thats not the issue, I’d need to see a debug log w/ sip
>> trace to tell you more.
>>
>>
>> > On Jun 15, 2018, at 2:50 PM, Chad Phillips <chad at apartmentlines.com>
>> wrote:
>> >
>> > I have a slightly older install where I’m using H264 with Verto, and it
>> works fine. However, I’m not able to get it working on latest master.
>> >
>> > I’m guessing there’s some esoteric setting I’m missing, but can’t seem
>> to find it. I do have the following in verto.conf.xml:
>> >
>> >       <param name="outbound-codec-string" value="opus,h264"/>
>> >       <param name=“inbound-codec-string” value="opus,h264"/>
>> >
>> > I also have mod_h26x disabled, as I remember that’s passthrough only
>> and causes issues when loaded in this case.
>> >
>> > When I try in Chrome I see it offering H264 in the SDP, but the answer
>> SDP contains this, and the stream is audio only:
>> >
>> > m=video 0 UDP/TLS/RTP/SAVPF 19
>> >
>> > What am I missing??
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180620/7a858442/attachment-0001.html>

From s.safarov at gmail.com  Wed Jun 20 15:30:27 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 20 Jun 2018 18:30:27 +0300
Subject: [Freeswitch-users] Wrong IP in ACK
In-Reply-To: <5798A77D-BB91-4E5E-9F41-A5968F08193C@jerris.com>
References: <CAHtxdDcXfiD73Av6wrgybCZk5zDx6k2Tnjif-PWEuDGf3Zy_cQ@mail.gmail.com>
 <CAHtxdDeD1iX0U3-PC2yOyMy6yuFxK=n5PKd1gRnphnvDfkqQ8A@mail.gmail.com>
 <D26E5F38-7916-4D66-86E3-D813FC14DDFD@jerris.com>
 <CAHtxdDd9cJB_JCcmhS_iaCicgHe+AZBpqYMkCi+i9R8w4261Rw@mail.gmail.com>
 <5798A77D-BB91-4E5E-9F41-A5968F08193C@jerris.com>
Message-ID: <CAHtxdDfN5rPT2wXMH0TkohRJuJPDts45KRxK49bYYOFJp3MJ1A@mail.gmail.com>

May be i misunderstanding something with call routing.
I want route calls from FS inside private network to "google.com" (as
example) via two FS (load balancing and redundancy) at edge of private
network.

I cannot use fs_path to route call via "edge FS", because session is
maintained.
May i can use other way to reach my goal?

Sergey


вт, 19 июн. 2018 г. в 20:43, Michael Jerris <mike at jerris.com>:

> if you are using fs_path and expecting it to be anything other than a
> hardcoded outbound proxy then its not going to work like you expect.
>
>
>
> On Jun 19, 2018, at 3:06 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> If FS ignores contact values and use fs_path like "Record-Route" SIP
> header, then ACK must be send to same IP used for INVITE.
> Without this whole session is broken.
> Please look call log. Call is dropped by remote end because ACK not
> received.
>
> вт, 19 июн. 2018 г. в 2:18, Michael Jerris <mike at jerris.com>:
>
>> Setting fs_path like that is used to force a locked in outbound proxy for
>> that dialog.  The behavior you are seeing is intended behavior.
>>
>>
>> On Jun 12, 2018, at 4:45 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
>>
>> root of issue
>> ACK send to IP resolved via DNS for value defined as "fs_path", rather
>> than defined in contact string.
>>
>> more details
>> https://freeswitch.org/jira/browse/FS-11190
>>
>> вт, 12 июн. 2018 г. в 10:30, Sergey Safarov <s.safarov at gmail.com>:
>>
>>> Are anybody experiences issue when for some of calls  ACK send to other
>>> IP.
>>> http://prntscr.com/jtxkbz
>>> in this case FS in bypass media with two profiles (1 to internet, 2
>>> intranet).
>>> INVITE is send to DNS name that is resolved to
>>>
>>>    1. 10.0.9.32
>>>    2. 10.0.9.33
>>>    3. 10.0.9.34
>>>    4. 10.0.9.35
>>>
>>> FS 1.6.19
>>>
>>> Contact on 200 message is correct
>>>
>>> SIP/2.0 200 OK Via: SIP/2.0/TCP 10.0.9.51:12000;rport=39049;branch=
>>> z9hG4bKBH60NQjSZj36a From: "Anonymous" <sip:+12022553656 at 10.0.9.51>;tag=
>>> rH5cBQarr559D To: <sip:13013688630 <(301)%20368-8630>@metropolis.ack.one>;tag=UDv7UXv3B53Da
>>> Call-ID: b0e4d7e1-e79e-1236-8cb9-02420a000933 CSeq: 123998414 INVITE
>>> Contact: <sip:13013688630 <(301)%20368-8630>@10.0.9.33:11000;transport=tcp>
>>> User-Agent: proxy.ackone.com Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>>> MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> Now for future investigate FS upgrader to 1.6.20 and enabled siptrace
>>> and loglevel
>>>
>>> Sergey Safarov
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180620/bebf0c29/attachment.html>

From s.safarov at gmail.com  Thu Jun 21 05:18:41 2018
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 21 Jun 2018 08:18:41 +0300
Subject: [Freeswitch-users] Wrong IP in ACK
In-Reply-To: <5798A77D-BB91-4E5E-9F41-A5968F08193C@jerris.com>
References: <CAHtxdDcXfiD73Av6wrgybCZk5zDx6k2Tnjif-PWEuDGf3Zy_cQ@mail.gmail.com>
 <CAHtxdDeD1iX0U3-PC2yOyMy6yuFxK=n5PKd1gRnphnvDfkqQ8A@mail.gmail.com>
 <D26E5F38-7916-4D66-86E3-D813FC14DDFD@jerris.com>
 <CAHtxdDd9cJB_JCcmhS_iaCicgHe+AZBpqYMkCi+i9R8w4261Rw@mail.gmail.com>
 <5798A77D-BB91-4E5E-9F41-A5968F08193C@jerris.com>
Message-ID: <CAHtxdDdeL_CRFUm4eLfofRS6hNnD+H8Xg25r-RuVmzHw-HRLkg@mail.gmail.com>

Michael, you right FS send ACK to defined by "fs_path"
But i use "domain" as "fs_path" value, and and "domain" is resolved to
several IP via "A" records.
In this case "INVITE" send to one IP of "hardcoded outbound proxy" and ACK
send to other IP of "hardcoded outbound proxy".
But this this IPs is not same host. This is two FS used for load balancing
and HA.
Because FS is B2BUA and no one of "hardcoded outbound proxy" have
established properly call, then call is dropped.

As workaround I can manually create two legs/calls. One via first IP
"hardcoded outbound proxy" directly and second call via second IP.
But this will be "hardcoded IP" in dialplan. That also not fine.

Sergey

вт, 19 июн. 2018 г. в 20:43, Michael Jerris <mike at jerris.com>:

> if you are using fs_path and expecting it to be anything other than a
> hardcoded outbound proxy then its not going to work like you expect.
>
>
>
> On Jun 19, 2018, at 3:06 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
>
> If FS ignores contact values and use fs_path like "Record-Route" SIP
> header, then ACK must be send to same IP used for INVITE.
> Without this whole session is broken.
> Please look call log. Call is dropped by remote end because ACK not
> received.
>
> вт, 19 июн. 2018 г. в 2:18, Michael Jerris <mike at jerris.com>:
>
>> Setting fs_path like that is used to force a locked in outbound proxy for
>> that dialog.  The behavior you are seeing is intended behavior.
>>
>>
>> On Jun 12, 2018, at 4:45 AM, Sergey Safarov <s.safarov at gmail.com> wrote:
>>
>> root of issue
>> ACK send to IP resolved via DNS for value defined as "fs_path", rather
>> than defined in contact string.
>>
>> more details
>> https://freeswitch.org/jira/browse/FS-11190
>>
>> вт, 12 июн. 2018 г. в 10:30, Sergey Safarov <s.safarov at gmail.com>:
>>
>>> Are anybody experiences issue when for some of calls  ACK send to other
>>> IP.
>>> http://prntscr.com/jtxkbz
>>> in this case FS in bypass media with two profiles (1 to internet, 2
>>> intranet).
>>> INVITE is send to DNS name that is resolved to
>>>
>>>    1. 10.0.9.32
>>>    2. 10.0.9.33
>>>    3. 10.0.9.34
>>>    4. 10.0.9.35
>>>
>>> FS 1.6.19
>>>
>>> Contact on 200 message is correct
>>>
>>> SIP/2.0 200 OK Via: SIP/2.0/TCP 10.0.9.51:12000;rport=39049;branch=
>>> z9hG4bKBH60NQjSZj36a From: "Anonymous" <sip:+12022553656 at 10.0.9.51>;tag=
>>> rH5cBQarr559D To: <sip:13013688630 <(301)%20368-8630>@metropolis.ack.one>;tag=UDv7UXv3B53Da
>>> Call-ID: b0e4d7e1-e79e-1236-8cb9-02420a000933 CSeq: 123998414 INVITE
>>> Contact: <sip:13013688630 <(301)%20368-8630>@10.0.9.33:11000;transport=tcp>
>>> User-Agent: proxy.ackone.com Allow: INVITE, ACK, BYE, CANCEL, OPTIONS,
>>> MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>> Now for future investigate FS upgrader to 1.6.20 and enabled siptrace
>>> and loglevel
>>>
>>> Sergey Safarov
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180621/74cdfad5/attachment-0001.html>

From gmaruzz at gmail.com  Thu Jun 21 06:33:55 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 21 Jun 2018 08:33:55 +0200
Subject: [Freeswitch-users] Problems with NAT
In-Reply-To: <3DBAB7D6-5210-415A-B5FD-28114D7A93D9@nvoip.com.br>
References: <3DBAB7D6-5210-415A-B5FD-28114D7A93D9@nvoip.com.br>
Message-ID: <CALXCt0o0D=j-W_P1LFTxSxAp8qfKJ_OBBnMpxU32qise4+VjYw@mail.gmail.com>

Leandro,

you obviously need to give much more details.

-giovanni


On 20 June 2018 at 00:26, Leandro Campos <leandro.campos at nvoip.com.br>
wrote:

> Recently we are testing the Freeswitch with the Astpp for change us
> servers (we use asterisk with a2billing).
> In the migration process, we check some customers don’t register or don’t
> receive inbound calls. In a2billing works fine.
> I think is something about NAT. Someone help me?
>
>
> Atenciosamente,
>
> [image: Image]
>
> *Leandro Campos*
>
> *CEO*
>
> (11) 4118-6267
> [image: Nvoip] <https://www.nvoip.com.br/> [image: LinkedIn]
> <https://www.linkedin.com/in/leandro-campos/>
> https://www.nvoip.com.br
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180621/a2a96c6f/attachment.html>

From mike at jerris.com  Thu Jun 21 06:35:00 2018
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 Jun 2018 02:35:00 -0400
Subject: [Freeswitch-users] Using H264 with Verto
In-Reply-To: <CAAPhwbdFkZ1_NGutcv4sKxpNSAEF6z3fZfxp7Nas2vVAdTQ34w@mail.gmail.com>
References: <CAJ0sghRNE5xbQU=h6Hux67xRcJGfZsbkrZm63oBG5owCV-iyUw@mail.gmail.com>
 <3D1DB6A6-D120-4C75-93C3-4D010DE4273A@jerris.com>
 <CAJ0sghTvzfrWjihy9hJmX0LWZKCMPNG+oFhOHBsW4PrVdaEVKw@mail.gmail.com>
 <00FF96CE-F893-4B56-8FC9-BE667D1F217F@jerris.com>
 <36D9E69E-90EF-4845-87D2-3D0141A21ABC@gmail.com>
 <898C30C2-3EB8-4163-BBC2-E1234352EB88@jerris.com>
 <CAAPhwbdFkZ1_NGutcv4sKxpNSAEF6z3fZfxp7Nas2vVAdTQ34w@mail.gmail.com>
Message-ID: <7494283A-ADBD-471D-8882-E2E050CEC042@jerris.com>

It would not be added just because the module is loaded (it would be removed potentially if its not loaded), its either coming from the other leg, or in codec preferences.

> On Jun 20, 2018, at 11:38 AM, Magnus Kelly <magnus.kelly at gmail.com> wrote:
> 
> Well the issue I have is I trying to remove H.264 from the codecs listed in the SDP,  as although its not listed in the codec section of vars.xml at all, enabling mod_av appears to mean that by default H.264 is added in the SDP for each outbound SIP invite, resulting in calls to PSTN SIP G/W failing with unacceptable codec. Advice on how to prevent or make selective per SIP profile most welcome.
> Thanks
> Magnus
> 
> On Wed, 20 Jun 2018 at 14:49, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Just configure codec preferences appropriately.  We filter prefs based on whats loaded so that would be why you wouldnt see it until module is loaded.
> 
>> On Jun 19, 2018, at 2:40 PM, Magnus <magnus.kelly at gmail.com <mailto:magnus.kelly at gmail.com>> wrote:
>> 
>>  Chad, good addition to documents, by chance have you also discovered how to enable mod_av, but not use H264? I ask as I observe that enabling mod_av adds the h264 codec even if it’s not listed in vars.xml. In my particular case I would like to use mod_av but select when to use h.264 by destination as opposed to it being included in every sip sdp, but I have not yet worked out how to do so - by chance have you any tips from your testing ?
>> 
>> Regards
>> Magnus
>> 
>> 
>> On 19 Jun 2018, at 17:34, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> 
>>> Awesome.  No problem.
>>> 
>>>> On Jun 19, 2018, at 11:04 AM, Chad Phillips <chad at apartmentlines.com <mailto:chad at apartmentlines.com>> wrote:
>>>> 
>>>> Yep, that was it, thanks!
>>>> 
>>>> I added a section to the Verto docs to clarify all the steps to get H264 working: https://evoluxbr.github.io/verto-docs/tut/adding-h264-support.html <https://evoluxbr.github.io/verto-docs/tut/adding-h264-support.html>
>>>> On Mon, Jun 18, 2018 at 4:21 PM Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>>> load mod_av?  if thats not the issue, I’d need to see a debug log w/ sip trace to tell you more.
>>>> 
>>>> 
>>>> > On Jun 15, 2018, at 2:50 PM, Chad Phillips <chad at apartmentlines.com <mailto:chad at apartmentlines.com>> wrote:
>>>> > 
>>>> > I have a slightly older install where I’m using H264 with Verto, and it works fine. However, I’m not able to get it working on latest master.
>>>> > 
>>>> > I’m guessing there’s some esoteric setting I’m missing, but can’t seem to find it. I do have the following in verto.conf.xml:
>>>> > 
>>>> >       <param name="outbound-codec-string" value="opus,h264"/>
>>>> >       <param name=“inbound-codec-string” value="opus,h264"/>
>>>> > 
>>>> > I also have mod_h26x disabled, as I remember that’s passthrough only and causes issues when loaded in this case.
>>>> > 
>>>> > When I try in Chrome I see it offering H264 in the SDP, but the answer SDP contains this, and the stream is audio only: 
>>>> > 
>>>> > m=video 0 UDP/TLS/RTP/SAVPF 19
>>>> > 
>>>> > What am I missing??
>>>> 
>>> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180621/1b734ee3/attachment.html>

From mbrancaleoni at voismart.it  Wed Jun 20 09:50:26 2018
From: mbrancaleoni at voismart.it (Matteo)
Date: Wed, 20 Jun 2018 11:50:26 +0200 (CEST)
Subject: [Freeswitch-users] Enterprise/Production Quality?
In-Reply-To: <1527706840.542281.1390853752.2050A1B1@webmail.messagingengine.com>
References: <1A170859-F608-498C-AB99-C41E17E5B706@connectfirst.com>
 <AF3066FB-3622-4462-A16D-D35E69B288EE@jerris.com>
 <1527697018.3678744.1390660224.09199347@webmail.messagingengine.com>
 <6EF6FCCE-4ED1-49B2-BC48-2B87099FDC7D@jerris.com>
 <DAAD6B7C-2ED7-4BFE-9E89-E1FEC4F29CD1@stratusvideo.com>
 <1527706840.542281.1390853752.2050A1B1@webmail.messagingengine.com>
Message-ID: <964413179.25277.1529488226644.JavaMail.zimbra@voismart.it>

Well I agree too.

Since the introduction of the FSA (term not cited anywhere on website), we started evaluating a subscription.

But a lot of things are really, really vague and keeping us away:

* What is in the FSA? What we get for the money?
* What about prices?
* What is the release cycle? When things gets pushed into the OSS version?
* What is the FSA? please add a blog post to clarify this shift in policy and how things "works" from now on.

Basically there's no communication about that and everything appears really vague.

Other OSS projects with similar policy have a very clear release path, what is in, what is not,
when new things will appear in OSS and so on.

And then a suggestion:
Why not offer a "basic" FSA with just access to latest git?
And not provide dedicated support, but use traditional channels?

This should be cheaper and interesting for companies that are autonomous into investigating and fixing issues by themselves.

And about bugfixes... well I agree with others... they should be released because affects something that's already out.

Not doing that may backfire because the OSS product may be perceive buggy (especially if using latest features which may not be debugged properly yet)

Keep in mind that from a company perspective, if the things are free you can be loose on how the project is handled, because is free after all.
But if you want to get payed (which is completely natural and needed to keep things go on), you must be very clear and communicative on what you give for the price.

Just my 2 cents.

Matteo

----- Il 30-mag-18, alle 21:00, Michael Avers michael at mailworks.org ha scritto:

> I agree. I think they are doing it wrong. I obviously believe they need to make
> money and get paid for their work, I have no problem with that at all, and my
> company has bought quite a few licenses of their commercial modules and we
> attend Cluecon every other year. So we have no problem paying but we need to
> know what we are paying FOR.
> 
> This kind of vague state of things where you just don't know if bugs are now
> going to be fixed for everyone or just for the privileged ones is not a good
> path to go down.
> 
> Bug fixes really should be made available to everyone (assuming they are in
> previously public modules, of course).
> 
> The Freeswitch team should focus their efforts on creating commercial modules,
> ready-made apps and setups, pro versions of older modules, say enhanced
> mod_callcenter or whatever. Things like that. But to tell someone to get a
> premium subscription just to get a bug fixed... that's simply wrong.
> 
> Just my 2 cents
> 
> Mike
> 
> 
> 
> 
> On Wed, May 30, 2018, at 11:37 AM, William Simon wrote:
> 
> 
> 
> I thought stability fixes were going to be included in the open source release
> whereas new features are covered under FSA release cycle. What I am seeing here
> is unfortunate. Open source users should just accept instability for the
> 18-month release cycle?
> 
> 
> 
> 
> 
> On May 30, 2018, at 12:35 PM, Michael Jerris < mike at jerris.com > wrote:
> 
> Michael-
> 
> This is a specific bug that I know we have fixed. We spent months of work
> tracking it down, I am very familiar with the issue. This issue is not at all
> with verto, and is specifically with the sip secure web socket support. We have
> never recommended the use of sip web socket support for webrtc, we think that
> verto is typically a better solution, and is more stable.
> 
> Mike
> 
> 
> 
> 
> 
> On May 30, 2018, at 12:16 PM, Michael Avers < michael at mailworks.org > wrote:
> 
> LOL... The issue he describes can be due to several different reasons. I don't
> even think we have enough information at this point to determine it let alone
> narrow it down to one specific bug fix.
> 
> Is saying "Oh this is fixed in our paid product" going to be the standard moving
> forward?
> 
> In any case, a response such as this basically tells potential Verto users that
> they shouldn't bother because it's going to break at 150 concurrent users
> anyway.
> 
> Mike
> 
> 
> On Wed, May 30, 2018, at 8:58 AM, Michael Jerris wrote:
> 
> 
> 
> Geoff-
> 
> I believe this issue is fixed in https://freeswitch.org/jira/browse/FS-10762
> which is available to FSA customers.
> 
> Mike
> 
> 
> 
> 
> On May 30, 2018, at 10:45 AM, Geoff Mina < gmina at connectfirst.com > wrote:
> 
> Is anyone out there actually using FS successfully in an enterprise environment?
> 
> We have deployed a handful of servers in an extremely simple configuration to
> allow standard SIP infrastructure to communicate with WebRTC clients.
> 
> We run ~150 concurrent users per host and we can’t go a week without something
> in the core of Sofia failing. We have seen hung profiles that simply don’t
> respond to REGISTER requests (tried every suggested tweak to no avail) as well
> as seeing FS hang every call in a RINGING state without actually ringing the
> end client.
> 
> These seem like pretty fundamental components we are struggling with. Our use
> case seems quite simple - yet the software (1.6.19) seems like it has never
> even seen a production deployment.
> 
> Anyone out there have a drastically different experience?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> “The information transmitted is intended only for the person or entity to which
> it is addressed and may contain proprietary, business-confidential and/or
> privileged material. If you are not the intended recipient of this message you
> are hereby notified that any use, review, retransmission, dissemination,
> distribution, reproduction or any action taken in reliance upon this message is
> prohibited. If you received this in error, please contact the sender and delete
> the material from any computer.”
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com


From david.villasmil.work at gmail.com  Thu Jun 21 13:03:57 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 21 Jun 2018 15:03:57 +0200
Subject: [Freeswitch-users] Problems with NAT
In-Reply-To: <CALXCt0o0D=j-W_P1LFTxSxAp8qfKJ_OBBnMpxU32qise4+VjYw@mail.gmail.com>
References: <3DBAB7D6-5210-415A-B5FD-28114D7A93D9@nvoip.com.br>
 <CALXCt0o0D=j-W_P1LFTxSxAp8qfKJ_OBBnMpxU32qise4+VjYw@mail.gmail.com>
Message-ID: <CAFGRPVruvKQV5F-O15TKKsrNpr+Q7nf4LhYHcBv90ZinZVWKRA@mail.gmail.com>

Yeah, what's the network look like? Is fs behind nat or the clients or
both? Fs is usually VERY good at overcoming NAT issues. You might want to
set the ext-sip-ip and rtp-ip to the public ip if fs is behind nat.

On Thu, Jun 21, 2018, 10:00 Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> Leandro,
>
> you obviously need to give much more details.
>
> -giovanni
>
>
> On 20 June 2018 at 00:26, Leandro Campos <leandro.campos at nvoip.com.br>
> wrote:
>
>> Recently we are testing the Freeswitch with the Astpp for change us
>> servers (we use asterisk with a2billing).
>> In the migration process, we check some customers don’t register or don’t
>> receive inbound calls. In a2billing works fine.
>> I think is something about NAT. Someone help me?
>>
>>
>> Atenciosamente,
>>
>> [image: Image]
>>
>> *Leandro Campos*
>>
>> *CEO*
>>
>> (11) 4118-6267
>> [image: Nvoip] <https://www.nvoip.com.br/> [image: LinkedIn]
>> <https://www.linkedin.com/in/leandro-campos/>
>> https://www.nvoip.com.br
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180621/05b21972/attachment-0001.html>

From david.villasmil.work at gmail.com  Thu Jun 21 13:07:08 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 21 Jun 2018 15:07:08 +0200
Subject: [Freeswitch-users] NORMAL_CIRCUIT_CONGESTION error limiting
 simultaneous calls
In-Reply-To: <MEXPR01MB1223B52CA2CFA0AAD649EC98F3770@MEXPR01MB1223.ausprd01.prod.outlook.com>
References: <CAEduXd+n+JpU42E8SeB-A_JNodPTNNBHx6o44mQLesGMN1FjmA@mail.gmail.com>
 <MEXPR01MB1223B52CA2CFA0AAD649EC98F3770@MEXPR01MB1223.ausprd01.prod.outlook.com>
Message-ID: <CAFGRPVozdnGBNAr9NiYVT-dARMB1i6ZG3nik7GADTFpbGdVZpQ@mail.gmail.com>

You need traces to see who is rejecting the calls, is it freeswitch or the
termination provider?

On Thu, Jun 21, 2018, 09:37 Andrew Keil <andrew.keil at visytel.com> wrote:

> Artyom.
>
>
>
> This could be also related to the speed (Call Attempts Per Second - CAPS)
> that you are making the outbound calls.  If you slow down the outbound
> calls being made and increase their duration does this increase your
> simultaneous calls?
>
>
>
> My guess is similar to your one that this is a provider issue.  I know BT
> in the UK restrict the CAPS on their SIP Trunks to avoid network errors (or
> basically their SBCs cannot handle more CAPS).
>
>
>
> Obviously if the provider is generating the NORMAL_CIRCUIT_CONGESTION
> error (check the SIP Trace) then it is a provider issue and they should be
> involved.
>
>
>
> Andrew
>
>
>
> *From:* FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> *On
> Behalf Of *Artyom Chernetzov
> *Sent:* Wednesday, 20 June 2018 5:22 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] NORMAL_CIRCUIT_CONGESTION error limiting
> simultaneous calls
>
>
>
> We got NORMAL_CIRCUIT_CONGESTION error trying to perform 100 simultaneous
> calls, and only 40 simultaneous calls actually happen (all other fail with
> NORMAL_CIRCUIT_CONGESTION). We have 1gbit/s network, all default rtp ports
> are open, and switch.conf.xml sessions-per-second=1000 (most freeswitch
> and gateway configs are default). We run dedicated server. Provider tell
> that 500 lines are available. Is it possible that I missed some config that
> could limit gateway channels, or this is definitely provider issue?
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180621/9ac8b4d0/attachment-0001.html>

From vishalmpai at gmail.com  Thu Jun 21 17:29:29 2018
From: vishalmpai at gmail.com (Vishal Pai)
Date: Thu, 21 Jun 2018 22:59:29 +0530
Subject: [Freeswitch-users] mod_verto
In-Reply-To: <88C5D191-E7F2-4ACF-AE27-61816DF12384@jerris.com>
References: <CACOmXtBSxNbtKgH961AP0D=5jEpgtPe3gb+cjR-w4GU3oZKUFQ@mail.gmail.com>
 <88C5D191-E7F2-4ACF-AE27-61816DF12384@jerris.com>
Message-ID: <CACOmXtB8RkB9cAfT_snKsq0WywHn0MEgjPzgNfMq8q78QjRbXQ@mail.gmail.com>

I am able to record the conference. Now I need to know what will be server
requirements if I want to have 5 conference with 10 participants each
simultaneously.



On Mon, Jun 4, 2018 at 11:16 PM, Michael Jerris <mike at jerris.com> wrote:

> mod_av supports mp4 and others.
>
>
> > On Jun 3, 2018, at 4:02 AM, Vishal Pai <vishalmpai at gmail.com> wrote:
> >
> > Can we record the video conferencing using mod_verto. If yes how it is
> possible and and what would it’s file format.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180621/9f5b62ee/attachment-0001.html>

From davidswalkabout at gmail.com  Fri Jun 22 05:09:24 2018
From: davidswalkabout at gmail.com (David P)
Date: Thu, 21 Jun 2018 22:09:24 -0700
Subject: [Freeswitch-users] RECORD_STEREO should not make both channels the
	same
Message-ID: <CAE23trsV4Qwcc+GJcS-hUB86fioOptDyfjmnT2FkNzB2ka9qgg@mail.gmail.com>

We'd like to record the two legs of our calls separately (both 16kHz). I
followed
https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO
and added
<action application="set" data="RECORD_STEREO=true"/>
to our dialplan. I now get two channels instead of one in our mp4's but the
data in the two channels is identical, containing speech from both legs of
the call.

This is contrary to what the confluence page says to expect.

Btw, I wanted to include our FS version here, and I tried to get it by
running the cli and using /help, but it's not shown there.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180621/33350982/attachment-0001.html>

From shakumarsoftware at gmail.com  Fri Jun 22 21:45:35 2018
From: shakumarsoftware at gmail.com (Sharath Kumar)
Date: Fri, 22 Jun 2018 15:45:35 -0600
Subject: [Freeswitch-users] Incoming RTP with different sequence number and
	host
Message-ID: <CA+wj8uaAq0yhCwzy5=63nW4e3yZog1_-W4d-6S-4uftSf7WvYg@mail.gmail.com>

Hello,
I am running FS 1.6.18 and have a provider that is sending me a stream of
incoming RTP that changes it's host IP, and RTP sequence number and
Timestamps after 10 mins in the call. The SSRC does not change. As a result
I am getting either silence or garbled audio. The Sample rate is 20ms as in
the original stream. Is there any setting in the FS that can make this work
? Have any of seen this work with inbound stream ?
Thank you,
Shaks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180622/35b402b7/attachment-0001.html>

From j4v28bsjtp43hnf865 at gmail.com  Sat Jun 23 08:17:23 2018
From: j4v28bsjtp43hnf865 at gmail.com (Xenia Obolenskaya)
Date: Sat, 23 Jun 2018 08:17:23 +0000
Subject: [Freeswitch-users] Peer Subject Mismatch (incoming connection)
Message-ID: <CAF7mOn4vxkNq_dx0=2dg3e0HURtnYpGng86tF8iZRnJ827crtA@mail.gmail.com>

Hi All,

It is strange, very strange history.

When I made the blank settings of value, type of
<param name="tls-verify-in-subjects" value=""/>
the check of peer name in the manner
<param name="tls-verify-policy" value="subjects_all"/>
works good.

The same favorable outcome I can observe if value settins are of kind
<param name="tls-verify-in-subjects" value="sip.obolenskaya.su"/>
that is to say pattern-matching filtering.

Now Peer Subject in incoming connection matches!

But I nowhere could find the similar clarifications.

Thank All!

Regards,
Xenia Obolenskaya
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180623/50626246/attachment-0001.html>

From gregor at infomedia.si  Sat Jun 23 11:07:42 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Sat, 23 Jun 2018 13:07:42 +0200
Subject: [Freeswitch-users] Freeswitch windows build - libpng
Message-ID: <CAC7ttFFMUGXbs9Qj3Doi3MRuqFRohh5i-9MERm1E8sWQwF=okQ@mail.gmail.com>

Hi!

After a year I wanted to build FS on windows and make video tutorial or for
FS wiki.

I am using latest VS2017 and everything builds except libpng project. Below
is error if someone can help me, because I don't have c++ experience:

1>------ Build started: Project: Download zlib, Configuration: Release
Win32 ------
1>Downloading zlib.
2>------ Build started: Project: libpng, Configuration: Release x64 ------
2>Generating pnglibconf.h
2>        1 file(s) copied.
2>png.c
2>pngerror.c
2>pngget.c
2>pngmem.c
2>pngpread.c
2>pngread.c
2>pngrio.c
2>pngrtran.c
2>pngrutil.c
2>pngset.c
2>pngtrans.c
2>pngwio.c
2>pngwrite.c
2>pngwtran.c
2>pngwutil.c
2>Generating Code...
2>zlib.lib(inflate.obj) : MSIL .netmodule or module compiled with /GL
found; restarting link with /LTCG; add /LTCG to the link command line to
improve linker performance
2>   Creating library D:\Git\freeswitch\x64\Release\libpng16.lib and object
D:\Git\freeswitch\x64\Release\libpng16.exp
2>Generating code
2>d:\git\freeswitch\libs\zlib\deflate.c(2097): error C2220: warning treated
as error - no 'executable' file generated
2>d:\git\freeswitch\libs\zlib\deflate.c(2097): warning C5045: Compiler will
insert Spectre mitigation for memory load if /Qspectre switch specified
2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: index 'dist' range
checked by comparison on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: feeds memory load on
this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1987): warning C5045: Compiler will
insert Spectre mitigation for memory load if /Qspectre switch specified
2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: index 'dist' range
checked by comparison on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: feeds memory load on
this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1862): warning C5045: Compiler will
insert Spectre mitigation for memory load if /Qspectre switch specified
2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: index 'dist' range
checked by comparison on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: feeds memory load on
this line
2>LINK : fatal error LNK1257: code generation failed
2>Done building project "libpng.vcxproj" -- FAILED.

Best regards, Gregor
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180623/3a77c7f8/attachment-0001.html>

From davidswalkabout at gmail.com  Sat Jun 23 16:37:38 2018
From: davidswalkabout at gmail.com (David P)
Date: Sat, 23 Jun 2018 09:37:38 -0700
Subject: [Freeswitch-users] RECORD_STEREO should not make both channels
	the same
In-Reply-To: <CAE23trsV4Qwcc+GJcS-hUB86fioOptDyfjmnT2FkNzB2ka9qgg@mail.gmail.com>
References: <CAE23trsV4Qwcc+GJcS-hUB86fioOptDyfjmnT2FkNzB2ka9qgg@mail.gmail.com>
Message-ID: <CAE23truerUone9KzMmAj2dNsz52gLHvsew8Rw0QirM7VjU6_YQ@mail.gmail.com>

Is recording aleg and bleg audio to separate channels possible while using
a conference that records aleg video?

On Thu, 21 Jun 2018, 10:09 pm David P, <davidswalkabout at gmail.com> wrote:

> We'd like to record the two legs of our calls separately (both 16kHz). I
> followed
> https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO
> and added
> <action application="set" data="RECORD_STEREO=true"/>
> to our dialplan. I now get two channels instead of one in our mp4's but
> the data in the two channels is identical, containing speech from both legs
> of the call.
>
> This is contrary to what the confluence page says to expect.
>
> Btw, I wanted to include our FS version here, and I tried to get it by
> running the cli and using /help, but it's not shown there.
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180623/f758ac9c/attachment-0001.html>

From chad at apartmentlines.com  Sun Jun 24 00:49:51 2018
From: chad at apartmentlines.com (Chad Phillips)
Date: Sat, 23 Jun 2018 17:49:51 -0700
Subject: [Freeswitch-users] Conference audio issues when Verto user sends
	video to audio-only conference
Message-ID: <CAJ0sghSkGrQikoddXhYAHNiHWs_h2P=bDm+eMVDi8G62ASZCcw@mail.gmail.com>

I’ve got an audio-only conference configured, and some Verto clients (iOS
to be specific) are also sending video to that conference.

I had hoped that the video would simply be discarded, but instead I get
this error, and audio doesn’t seem to be sent/received properly:

[WARNING] switch_core_media.c:13944 verto.rtc/3521 has no video codec

The only solution I’ve found is to configure the conference in question for
video as well, but that’s not ideal.

Wondering if this is a bug to report to Jira, or possibly there’s some
config thing I can do to fix it? Because of iOS limitations, I’m not seeing
how to have that client send audio only in my case.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180623/1566f286/attachment-0001.html>

From sukithaj at gmail.com  Sun Jun 24 05:02:56 2018
From: sukithaj at gmail.com (sukitha jayasinghe)
Date: Sun, 24 Jun 2018 10:32:56 +0530
Subject: [Freeswitch-users] freeswitch internal profile hang freequently
Message-ID: <CAAwzf1mL2GMgmZTh6G_gUwX3=w5dWEwdjn-S5WO+5=VGb=FdRw@mail.gmail.com>

Dear All,

I have a freeswitch (1.6.20) server in production with db configured to
postgresql through pgpool2. This system hangs time to time and has become a
major issue. It happen only to the internal profile where our UACs connects
through webrtc. After the incident profile stop responding to cli commands,
only solution is restarting the entire application.  I managed to get the
back trace which indicate system waiting forever for profile database
mutex. Further, It seems thread #19 is waiting for result from pgsql server
cause this issue. Please find the back trace link below,

https://pastebin.com/Uqe8HjZn

Has anyone experienced this issue before, what is the solution for this.

Best Regards,
Sukitha.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180624/1138390f/attachment.html>

From atulthosar at gmail.com  Sun Jun 24 11:08:43 2018
From: atulthosar at gmail.com (Atul Thosar)
Date: Sun, 24 Jun 2018 16:38:43 +0530
Subject: [Freeswitch-users] [FreeSWITCH v1.6.20] Memory increasing
	continuously
Message-ID: <CAKT7MpqD-KSfGZO7MQeyXcNmEVhr43MSQOitJTPFgHmvy+Wk+A@mail.gmail.com>

Hi All,
I am verifying if we can use FreeSWITCH in our Production environment for
one of our IVR use case. Our use case involves INVITE containing MIME as a
payload, which contains XML and SDP parts.
When I ran 10 CPS load with MIME as a payload, I see Memory is continuously
growing. Where as same load without MIME (only SDP) payload shows no
increase in memory.

Is this a known issue? If yes, is this already fixed or patch is available
for the same?

I am using FreeSWITCH Version 1.6.20+git~20180123T214909Z~987c9b9a2a~64bit
(git 987c9b9 2018-01-23 21:49:09Z 64bit)
Attached output of top command for both the runs.

Btw Following are the changes made in FS configuration -

conf/vars.xml
+  <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.56.101"/>
+  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/>

conf/dialplan/public/00_inbound_did.xml changed as following

<include>
  <extension name="public_did">
    <condition field="destination_number" expression="^500$">
        <action application="answer"/>
        <action application="sched_hangup" data="+5 alloted_timeout"/>
        <action application="park"/>
    </condition>
  </extension>
</include>


conf/autoload_configs/acl.conf.xml

+ <node type="allow" cidr="192.168.56.102/24"/>

Thanks in advance.

--
Atul
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180624/0f3d28a5/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: top_w_mime.7z
Type: application/octet-stream
Size: 479707 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180624/0f3d28a5/attachment-0002.obj>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: top_wo_mime.7z
Type: application/octet-stream
Size: 529349 bytes
Desc: not available
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180624/0f3d28a5/attachment-0003.obj>

From davidswalkabout at gmail.com  Sun Jun 24 17:43:52 2018
From: davidswalkabout at gmail.com (David P)
Date: Sun, 24 Jun 2018 10:43:52 -0700
Subject: [Freeswitch-users] RECORD_STEREO should not make both channels
	the same
In-Reply-To: <CAE23truerUone9KzMmAj2dNsz52gLHvsew8Rw0QirM7VjU6_YQ@mail.gmail.com>
References: <CAE23trsV4Qwcc+GJcS-hUB86fioOptDyfjmnT2FkNzB2ka9qgg@mail.gmail.com>
 <CAE23truerUone9KzMmAj2dNsz52gLHvsew8Rw0QirM7VjU6_YQ@mail.gmail.com>
Message-ID: <CAE23trtbB8sEYfxprYNvr+wY4Q-z-fsNNRYc3wvPwm9MBOCDoQ@mail.gmail.com>

I've looked in the source, and found where flags are set for stereo,
write_only, etc but didn't find where the contents of the leg streams are
populated or mixed.

On Sat, 23 Jun 2018, 9:37 am David P, <davidswalkabout at gmail.com> wrote:

> Is recording aleg and bleg audio to separate channels possible while using
> a conference that records aleg video?
>
> On Thu, 21 Jun 2018, 10:09 pm David P, <davidswalkabout at gmail.com> wrote:
>
>> We'd like to record the two legs of our calls separately (both 16kHz). I
>> followed
>> https://freeswitch.org/confluence/display/FREESWITCH/RECORD_STEREO
>> and added
>> <action application="set" data="RECORD_STEREO=true"/>
>> to our dialplan. I now get two channels instead of one in our mp4's but
>> the data in the two channels is identical, containing speech from both legs
>> of the call.
>>
>> This is contrary to what the confluence page says to expect.
>>
>> Btw, I wanted to include our FS version here, and I tried to get it by
>> running the cli and using /help, but it's not shown there.
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180624/4ef8d1de/attachment.html>

From atulthosar at gmail.com  Mon Jun 25 12:49:00 2018
From: atulthosar at gmail.com (Atul Thosar)
Date: Mon, 25 Jun 2018 18:19:00 +0530
Subject: [Freeswitch-users] [FreeSWITCH v1.6.20] Memory increasing
	continuously
In-Reply-To: <CAKT7MpqD-KSfGZO7MQeyXcNmEVhr43MSQOitJTPFgHmvy+Wk+A@mail.gmail.com>
References: <CAKT7MpqD-KSfGZO7MQeyXcNmEVhr43MSQOitJTPFgHmvy+Wk+A@mail.gmail.com>
Message-ID: <CAKT7Mppe2HD9_knJ6Gz6WhgyxWG2wnQhmA7mTAj1jweL+_0+KA@mail.gmail.com>

I have ran the same scenario (10 CPS, With Mime as payload) on master
branch (FreeSWITCH Version 1.9.0+git~20180619T173242Z~25e9376b29~64bit (git
25e9376 2018-06-19 17:32:42Z 64bit)) and observed same issue of memory
increasing continuously.


--
Atul


On 24 June 2018 at 16:38, Atul Thosar <atulthosar at gmail.com> wrote:

> Hi All,
> I am verifying if we can use FreeSWITCH in our Production environment for
> one of our IVR use case. Our use case involves INVITE containing MIME as a
> payload, which contains XML and SDP parts.
> When I ran 10 CPS load with MIME as a payload, I see Memory is
> continuously growing. Where as same load without MIME (only SDP) payload
> shows no increase in memory.
>
> Is this a known issue? If yes, is this already fixed or patch is available
> for the same?
>
> I am using FreeSWITCH Version 1.6.20+git~20180123T214909Z~987c9b9a2a~64bit
> (git 987c9b9 2018-01-23 21:49:09Z 64bit)
> Attached output of top command for both the runs.
>
> Btw Following are the changes made in FS configuration -
>
> conf/vars.xml
> +  <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.56.101"/>
> +  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/>
>
> conf/dialplan/public/00_inbound_did.xml changed as following
>
> <include>
>   <extension name="public_did">
>     <condition field="destination_number" expression="^500$">
>         <action application="answer"/>
>         <action application="sched_hangup" data="+5 alloted_timeout"/>
>         <action application="park"/>
>     </condition>
>   </extension>
> </include>
>
>
> conf/autoload_configs/acl.conf.xml
>
> + <node type="allow" cidr="192.168.56.102/24"/>
>
> Thanks in advance.
>
> --
> Atul
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180625/b1091747/attachment-0001.html>

From kkothari157 at gmail.com  Mon Jun 25 12:59:35 2018
From: kkothari157 at gmail.com (Ketan Kothari)
Date: Mon, 25 Jun 2018 18:29:35 +0530
Subject: [Freeswitch-users] Any feature access code of disconnect B-leg
Message-ID: <CAGqqW9-SWNX5bnv1mgyL_EQX-4_gM+4YRTmhkckH7VJQ6m5=0g@mail.gmail.com>

Hello All,

I have configured one IVR using that IVR customer can i dial destnation
number.

The flow is like that.

Customer Dial DID number  ---> Freeswitch --> Its will ask to dial
destination number using  DTMF --> [USA]Route to Destination

A-leg  =  Customer ----> Freeswitch
B leg  = Freeswitch ---> Destination[USA]

Again generate B- leg
B leg  = Freeswitch ---> Destination[INDIA]

In this case customer are talking with  [USA]destination number and now
urgently he want to dial one more destination number[INDIA] and want
disconnect[USA]

So is there any access code or suggestion using that i can achieve above
call flow ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180625/b61107f8/attachment.html>

From asonesh at gmail.com  Fri Jun 22 14:31:27 2018
From: asonesh at gmail.com (Ari Sonesh)
Date: Fri, 22 Jun 2018 10:31:27 -0400
Subject: [Freeswitch-users] No video when calling into conference with Cisco
	Jabber
Message-ID: <CAEoNJ_zQYqyioyqhPYYMFecEeQnMTaw8+A1oZiqofbZMuysjNA@mail.gmail.com>

We are getting audio only but no video when calling into conference with
Cisco Jabber.

Other SIP video clients (e.g Jabber) calling into the conference are OK.

Sip negotiations are fine, SDP is fine and all parties agreeing on H.264
codec.

The log (at debug level) doesn't show any relevant errors or warnings.

Any ideas what can be wrong and/or how to troubleshoot the problem?

Thank you

*Ari *
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180622/f924b880/attachment.html>

From darshanmody at avaya.com  Thu Jun 21 23:13:17 2018
From: darshanmody at avaya.com (Mody, Darshan (Darshan))
Date: Thu, 21 Jun 2018 23:13:17 +0000
Subject: [Freeswitch-users] Memory leak with MIME
Message-ID: <25D2EC755404B4409F263AC6D050FEBB2BCD1F27@AZ-FFEXMB03.global.avaya.com>

Hi

We are observing memory leak when we have MIME data in the incoming invite. The resident memory usage increases constantly. When we tried the traffic without MIME and only SDP the usage stays constant.

Attaching the freeswitch logs for the same.

Thanks
Darshan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180621/1ba4423a/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: freeswitch.log
Type: application/octet-stream
Size: 1146619 bytes
Desc: freeswitch.log
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180621/1ba4423a/attachment-0001.obj>

From gmina at connectfirst.com  Tue Jun 26 01:50:40 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Mon, 25 Jun 2018 19:50:40 -0600
Subject: [Freeswitch-users] freeswitch internal profile hang freequently
In-Reply-To: <CAAwzf1mL2GMgmZTh6G_gUwX3=w5dWEwdjn-S5WO+5=VGb=FdRw@mail.gmail.com>
References: <CAAwzf1mL2GMgmZTh6G_gUwX3=w5dWEwdjn-S5WO+5=VGb=FdRw@mail.gmail.com>
Message-ID: <CAKzROs0H2mo77pSwew3NA-3GQogY+yJN9K9acysiAQhTUUQMBA@mail.gmail.com>

We had the exact same issue with MySQL and were assured it was a fault with
the ODBC driver on CentOS.

I’m concerned to hear this is actually a FS issue and not a DB driver
issue. Concerned. Not surprised.

Are you using xml_curl registrations by chance?

On Mon, Jun 25, 2018 at 7:43 PM sukitha jayasinghe <sukithaj at gmail.com>
wrote:

> Dear All,
>
> I have a freeswitch (1.6.20) server in production with db configured to
> postgresql through pgpool2. This system hangs time to time and has become a
> major issue. It happen only to the internal profile where our UACs connects
> through webrtc. After the incident profile stop responding to cli commands,
> only solution is restarting the entire application.  I managed to get the
> back trace which indicate system waiting forever for profile database
> mutex. Further, It seems thread #19 is waiting for result from pgsql server
> cause this issue. Please find the back trace link below,
>
> https://pastebin.com/Uqe8HjZn
>
> Has anyone experienced this issue before, what is the solution for this.
>
> Best Regards,
> Sukitha.
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
GEOFF MINA
Chief Executive Officer
Connect First / Contact Center Solutions, Built Better.

2545 Central Ave #200, Boulder, CO 80301
720.335.5924
Connect First / Contact Center Solutions, Built Better
www.connectfirst.com

This email and any files transmitted with it are confidential and are
intended solely for the use of the individual or entity to whom they are
addressed. If you have received this email in error, please notify the
system manager.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180625/d4691c6d/attachment-0001.html>

From thetsinling at outlook.com  Tue Jun 26 03:22:31 2018
From: thetsinling at outlook.com (bob. chen)
Date: Tue, 26 Jun 2018 03:22:31 +0000
Subject: [Freeswitch-users] =?utf-8?b?562U5aSNOiAgRnJlZXN3aXRjaCB3aW5kb3dz?=
 =?utf-8?q?_build_-_libpng?=
In-Reply-To: <CAC7ttFFMUGXbs9Qj3Doi3MRuqFRohh5i-9MERm1E8sWQwF=okQ@mail.gmail.com>
References: <CAC7ttFFMUGXbs9Qj3Doi3MRuqFRohh5i-9MERm1E8sWQwF=okQ@mail.gmail.com>
Message-ID: <PS1PR03MB15165E0C66223480F25A31F2A9490@PS1PR03MB1516.apcprd03.prod.outlook.com>

try vs2015 ;)


发件人: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] 代表 Gregor Nanger
发送时间: 2018年6月23日 19:08
收件人: FreeSWITCH Users Help
主题: [Freeswitch-users] Freeswitch windows build - libpng

Hi!

After a year I wanted to build FS on windows and make video tutorial or for FS wiki.

I am using latest VS2017 and everything builds except libpng project. Below is error if someone can help me, because I don't have c++ experience:

1>------ Build started: Project: Download zlib, Configuration: Release Win32 ------
1>Downloading zlib.
2>------ Build started: Project: libpng, Configuration: Release x64 ------
2>Generating pnglibconf.h
2>        1 file(s) copied.
2>png.c
2>pngerror.c
2>pngget.c
2>pngmem.c
2>pngpread.c
2>pngread.c
2>pngrio.c
2>pngrtran.c
2>pngrutil.c
2>pngset.c
2>pngtrans.c
2>pngwio.c
2>pngwrite.c
2>pngwtran.c
2>pngwutil.c
2>Generating Code...
2>zlib.lib(inflate.obj) : MSIL .netmodule or module compiled with /GL found; restarting link with /LTCG; add /LTCG to the link command line to improve linker performance
2>   Creating library D:\Git\freeswitch\x64\Release\libpng16.lib and object D:\Git\freeswitch\x64\Release\libpng16.exp
2>Generating code
2>d:\git\freeswitch\libs\zlib\deflate.c(2097): error C2220: warning treated as error - no 'executable' file generated
2>d:\git\freeswitch\libs\zlib\deflate.c(2097): warning C5045: Compiler will insert Spectre mitigation for memory load if /Qspectre switch specified
2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: index 'dist' range checked by comparison on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: feeds memory load on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1987): warning C5045: Compiler will insert Spectre mitigation for memory load if /Qspectre switch specified
2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: index 'dist' range checked by comparison on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: feeds memory load on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1862): warning C5045: Compiler will insert Spectre mitigation for memory load if /Qspectre switch specified
2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: index 'dist' range checked by comparison on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: feeds memory load on this line
2>LINK : fatal error LNK1257: code generation failed
2>Done building project "libpng.vcxproj" -- FAILED.

Best regards, Gregor

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180626/9505a023/attachment-0001.html>

From vishalmpai at gmail.com  Tue Jun 26 03:42:40 2018
From: vishalmpai at gmail.com (Vishal Pai)
Date: Tue, 26 Jun 2018 09:12:40 +0530
Subject: [Freeswitch-users] mod_verto
In-Reply-To: <88C5D191-E7F2-4ACF-AE27-61816DF12384@jerris.com>
References: <CACOmXtBSxNbtKgH961AP0D=5jEpgtPe3gb+cjR-w4GU3oZKUFQ@mail.gmail.com>
 <88C5D191-E7F2-4ACF-AE27-61816DF12384@jerris.com>
Message-ID: <CACOmXtAtoD4MpZ=u=XFPUzNG9FJMwDb_m8R-3F3ishPoEhF-Ng@mail.gmail.com>

I am able to record the conference. Now I need to know what will be server
requirements if I want to have 5 conference with 10 participants each
simultaneously.

On Mon, 4 Jun 2018 at 11:48 PM, Michael Jerris <mike at jerris.com> wrote:

> mod_av supports mp4 and others.
>
>
> > On Jun 3, 2018, at 4:02 AM, Vishal Pai <vishalmpai at gmail.com> wrote:
> >
> > Can we record the video conferencing using mod_verto. If yes how it is
> possible and and what would it’s file format.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180626/66c367d9/attachment-0001.html>

From albert.hsueh at cloudigit.com  Tue Jun 26 03:47:36 2018
From: albert.hsueh at cloudigit.com (=?UTF-8?B?6Jab5YWJ5a6P?=)
Date: Tue, 26 Jun 2018 11:47:36 +0800
Subject: [Freeswitch-users] video on conference with PC and MAC browser
 works but fail with iPhone safari via webrtc
Message-ID: <CABHWxvw0YvoWqmvx8_jBYQTU=E+Q7d-cy3K3tsR-Lc2pMr+DWA@mail.gmail.com>

Hi All,

here is our simple architecture
<client A>---<FreeSWITCH>
<client B>----/

- client A and B are both registered on FreeSWITCH via websocket (using
sipjs 0.9.2)
- we enable video-mode= transcode and load mod_av on FreeSWITCH.
- if client A uses chrome on PC and client B use safari on *MAC*, they can
see each other, talk to each other in the same conference.
- if client A uses chrome on PC and client B use safari on *iPhone*(8plus,
iOS 11.4), they can NOT see each other, talk to each other in the same
conference.

I though safari on MAC works, then safari on iPhone should work, but not.
Is it possible to fix that with changing configuration of FreeSWITCH? or
any suggestion?

Thanks in advance.

FreeSWITCH Version 1.6.20
Ubuntu Server Ubuntu 14.04.1 LTS 64bit
libav-11.4 or libav-12

See below configurations:
1. enable video-mode= transcode
<param name="conference-flags"
value="transcode-video|minimize-video-encoding"/>
<param name="video-mode" value="transcode"/>

2.enable VP8/H264 codec support
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,VP8,H264"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,VP8,H264"/>

3.enable mod_av and disable mod_h26x
<load module="mod_av"/>
<!--<load module="mod_h26x"/>-->
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180626/c9ecd5ec/attachment-0001.html>

From kkothari157 at gmail.com  Tue Jun 26 04:37:04 2018
From: kkothari157 at gmail.com (Ketan Kothari)
Date: Tue, 26 Jun 2018 10:07:04 +0530
Subject: [Freeswitch-users] Problems with NAT
In-Reply-To: <CAFGRPVruvKQV5F-O15TKKsrNpr+Q7nf4LhYHcBv90ZinZVWKRA@mail.gmail.com>
References: <3DBAB7D6-5210-415A-B5FD-28114D7A93D9@nvoip.com.br>
 <CALXCt0o0D=j-W_P1LFTxSxAp8qfKJ_OBBnMpxU32qise4+VjYw@mail.gmail.com>
 <CAFGRPVruvKQV5F-O15TKKsrNpr+Q7nf4LhYHcBv90ZinZVWKRA@mail.gmail.com>
Message-ID: <CAGqqW9-8BFyBXQLPu6movPdThqR1H0gbKadMZ1LevJFmDS02fA@mail.gmail.com>

Registration issue : Please capture sip trace of registration and pass it
here

Inbound call issue : If you are using public DID then you  must need to add
DID-Provider ip in ipsetting(ACL) for authenitcation also you can check
freeswitch.logs which error you are getting while call.
ASTPP web-portal ----> Switch --> ip settings
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180626/cbb87500/attachment-0001.html>

From naveen.khanna.bm at gmail.com  Tue Jun 26 05:22:46 2018
From: naveen.khanna.bm at gmail.com (Naveen Khanna)
Date: Tue, 26 Jun 2018 01:22:46 -0400
Subject: [Freeswitch-users] Incoming RTP with different sequence number
 and host
In-Reply-To: <CA+wj8uaAq0yhCwzy5=63nW4e3yZog1_-W4d-6S-4uftSf7WvYg@mail.gmail.com>
References: <CA+wj8uaAq0yhCwzy5=63nW4e3yZog1_-W4d-6S-4uftSf7WvYg@mail.gmail.com>
Message-ID: <CAE8fD+3zQrQDczyHXqJhdmMvTkjwc_w0McqP=FnTCSteAJrDkQ@mail.gmail.com>

Try  <param name="rtp-rewrite-timestamps" value="true”/> in the sip profile.

Regards,

Naveen Khanna

On 26 June 2018 at 6:37:06 AM, Sharath Kumar (shakumarsoftware at gmail.com)
wrote:

Hello,
I am running FS 1.6.18 and have a provider that is sending me a stream of
incoming RTP that changes it's host IP, and RTP sequence number and
Timestamps after 10 mins in the call. The SSRC does not change. As a result
I am getting either silence or garbled audio. The Sample rate is 20ms as in
the original stream. Is there any setting in the FS that can make this work
? Have any of seen this work with inbound stream ?
Thank you,
Shaks
_________________________________________________________________________
Professional FreeSWITCH Services
sales at freeswitch.com
https://freeswitch.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180626/9143c47d/attachment-0001.html>

From david.villasmil.work at gmail.com  Tue Jun 26 09:19:16 2018
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 26 Jun 2018 11:19:16 +0200
Subject: [Freeswitch-users] Enterprise/Production Quality?
In-Reply-To: <mailman.1631.1529968857.1211.freeswitch-users@lists.freeswitch.org>
References: <1A170859-F608-498C-AB99-C41E17E5B706@connectfirst.com>
 <AF3066FB-3622-4462-A16D-D35E69B288EE@jerris.com>
 <1527697018.3678744.1390660224.09199347@webmail.messagingengine.com>
 <6EF6FCCE-4ED1-49B2-BC48-2B87099FDC7D@jerris.com>
 <DAAD6B7C-2ED7-4BFE-9E89-E1FEC4F29CD1@stratusvideo.com>
 <1527706840.542281.1390853752.2050A1B1@webmail.messagingengine.com>
 <mailman.1631.1529968857.1211.freeswitch-users@lists.freeswitch.org>
Message-ID: <CAFGRPVqcDfnmGDLk=+CeKwSJTdwUpA9yGfEN_R9D7qJBod-TTQ@mail.gmail.com>

Hello all,

Is this true??? Bug fixes aren't available for the general open source
community for EIGHTEEN MONTHS???
This seems REALLY wrong to me... I'm pretty sure there are better ways of
making money from freeSWITCH...

Now I'm very concerned, and probably a lot of people will, if/when they
find out about this...

David

On Tue, Jun 26, 2018, 02:57 Matteo via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:

>
>
>
> ---------- Forwarded message ----------
> From: Matteo <mbrancaleoni at voismart.it>
> To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Wed, 20 Jun 2018 11:50:26 +0200 (CEST)
> Subject: Re: [Freeswitch-users] Enterprise/Production Quality?
> Well I agree too.
>
> Since the introduction of the FSA (term not cited anywhere on website), we
> started evaluating a subscription.
>
> But a lot of things are really, really vague and keeping us away:
>
> * What is in the FSA? What we get for the money?
> * What about prices?
> * What is the release cycle? When things gets pushed into the OSS version?
> * What is the FSA? please add a blog post to clarify this shift in policy
> and how things "works" from now on.
>
> Basically there's no communication about that and everything appears
> really vague.
>
> Other OSS projects with similar policy have a very clear release path,
> what is in, what is not,
> when new things will appear in OSS and so on.
>
> And then a suggestion:
> Why not offer a "basic" FSA with just access to latest git?
> And not provide dedicated support, but use traditional channels?
>
> This should be cheaper and interesting for companies that are autonomous
> into investigating and fixing issues by themselves.
>
> And about bugfixes... well I agree with others... they should be released
> because affects something that's already out.
>
> Not doing that may backfire because the OSS product may be perceive buggy
> (especially if using latest features which may not be debugged properly yet)
>
> Keep in mind that from a company perspective, if the things are free you
> can be loose on how the project is handled, because is free after all.
> But if you want to get payed (which is completely natural and needed to
> keep things go on), you must be very clear and communicative on what you
> give for the price.
>
> Just my 2 cents.
>
> Matteo
>
> ----- Il 30-mag-18, alle 21:00, Michael Avers michael at mailworks.org ha
> scritto:
>
> > I agree. I think they are doing it wrong. I obviously believe they need
> to make
> > money and get paid for their work, I have no problem with that at all,
> and my
> > company has bought quite a few licenses of their commercial modules and
> we
> > attend Cluecon every other year. So we have no problem paying but we
> need to
> > know what we are paying FOR.
> >
> > This kind of vague state of things where you just don't know if bugs are
> now
> > going to be fixed for everyone or just for the privileged ones is not a
> good
> > path to go down.
> >
> > Bug fixes really should be made available to everyone (assuming they are
> in
> > previously public modules, of course).
> >
> > The Freeswitch team should focus their efforts on creating commercial
> modules,
> > ready-made apps and setups, pro versions of older modules, say enhanced
> > mod_callcenter or whatever. Things like that. But to tell someone to get
> a
> > premium subscription just to get a bug fixed... that's simply wrong.
> >
> > Just my 2 cents
> >
> > Mike
> >
> >
> >
> >
> > On Wed, May 30, 2018, at 11:37 AM, William Simon wrote:
> >
> >
> >
> > I thought stability fixes were going to be included in the open source
> release
> > whereas new features are covered under FSA release cycle. What I am
> seeing here
> > is unfortunate. Open source users should just accept instability for the
> > 18-month release cycle?
> >
> >
> >
> >
> >
> > On May 30, 2018, at 12:35 PM, Michael Jerris < mike at jerris.com > wrote:
> >
> > Michael-
> >
> > This is a specific bug that I know we have fixed. We spent months of work
> > tracking it down, I am very familiar with the issue. This issue is not
> at all
> > with verto, and is specifically with the sip secure web socket support.
> We have
> > never recommended the use of sip web socket support for webrtc, we think
> that
> > verto is typically a better solution, and is more stable.
> >
> > Mike
> >
> >
> >
> >
> >
> > On May 30, 2018, at 12:16 PM, Michael Avers < michael at mailworks.org >
> wrote:
> >
> > LOL... The issue he describes can be due to several different reasons. I
> don't
> > even think we have enough information at this point to determine it let
> alone
> > narrow it down to one specific bug fix.
> >
> > Is saying "Oh this is fixed in our paid product" going to be the
> standard moving
> > forward?
> >
> > In any case, a response such as this basically tells potential Verto
> users that
> > they shouldn't bother because it's going to break at 150 concurrent users
> > anyway.
> >
> > Mike
> >
> >
> > On Wed, May 30, 2018, at 8:58 AM, Michael Jerris wrote:
> >
> >
> >
> > Geoff-
> >
> > I believe this issue is fixed in
> https://freeswitch.org/jira/browse/FS-10762
> > which is available to FSA customers.
> >
> > Mike
> >
> >
> >
> >
> > On May 30, 2018, at 10:45 AM, Geoff Mina < gmina at connectfirst.com >
> wrote:
> >
> > Is anyone out there actually using FS successfully in an enterprise
> environment?
> >
> > We have deployed a handful of servers in an extremely simple
> configuration to
> > allow standard SIP infrastructure to communicate with WebRTC clients.
> >
> > We run ~150 concurrent users per host and we can’t go a week without
> something
> > in the core of Sofia failing. We have seen hung profiles that simply
> don’t
> > respond to REGISTER requests (tried every suggested tweak to no avail)
> as well
> > as seeing FS hang every call in a RINGING state without actually ringing
> the
> > end client.
> >
> > These seem like pretty fundamental components we are struggling with.
> Our use
> > case seems quite simple - yet the software (1.6.19) seems like it has
> never
> > even seen a production deployment.
> >
> > Anyone out there have a drastically different experience?
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> >
> > “The information transmitted is intended only for the person or entity
> to which
> > it is addressed and may contain proprietary, business-confidential and/or
> > privileged material. If you are not the intended recipient of this
> message you
> > are hereby notified that any use, review, retransmission, dissemination,
> > distribution, reproduction or any action taken in reliance upon this
> message is
> > prohibited. If you received this in error, please contact the sender and
> delete
> > the material from any computer.”
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
>
>
>
>
> ---------- Forwarded message ----------
> From: Matteo via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org>
> To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
> Cc:
> Bcc:
> Date: Mon, 25 Jun 2018 17:57:02 -0700 (PDT)
> Subject: Re: [Freeswitch-users] Enterprise/Production Quality?
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180626/ebaff898/attachment-0001.html>

From asilva at wirelessmundi.com  Tue Jun 26 10:05:20 2018
From: asilva at wirelessmundi.com (antonio)
Date: Tue, 26 Jun 2018 12:05:20 +0200
Subject: h264 video call between endpoints with one end at bad resolution of
 352x288
Message-ID: <b40b3c4a-1dfa-d427-c59d-e8a98532e12e@wirelessmundi.com>

Hi,

Calling between two Grandstream GXV3275 i expected to have a video call 
at 1280x720, but the video that is present at destination as the 
resolution of 352x288 and the image as poor quality, originator is 
sending video at 1280×720.  If i swap the devices the result is the 
same, the destination always get a bad resolution of 352x288 but the 
originator receives the video at expected resolution of 1280x720.

If i set rtp_direct between the two endpoints it work as expected.

I also tested a conference room and force the it resolution to 1280*x720 
it work well,  if i don't force the canvas-size i see:

2018-06-26 13:55:02.077824 [WARNING] mod_conference.c:3157 Unspecified 
video-canvas-size, falling back to 1280x720

And the conference is done at 1280x720.


I've open a jira where you can find a full log with this behavior:

https://freeswitch.org/jira/browse/FS-11140


I don't understand why there is transcoding and mod_av is used, the 
codec negotiated is the same.. i understand that the resolution is at 
352x288 for the first frames sent by the phone but then it send them at 
1280x720,  mod_av should detected and adjust the new resolution, no?


Anyone experience this behavior?

-- 
Saludos / Regards / Cumprimentos
Anónio Silva



From italo at freeswitch.org  Tue Jun 26 12:18:42 2018
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 26 Jun 2018 09:18:42 -0300
Subject: [Freeswitch-users] [FreeSWITCH v1.6.20] Memory increasing
	continuously
In-Reply-To: <CAKT7Mppe2HD9_knJ6Gz6WhgyxWG2wnQhmA7mTAj1jweL+_0+KA@mail.gmail.com>
References: <CAKT7MpqD-KSfGZO7MQeyXcNmEVhr43MSQOitJTPFgHmvy+Wk+A@mail.gmail.com>
 <CAKT7Mppe2HD9_knJ6Gz6WhgyxWG2wnQhmA7mTAj1jweL+_0+KA@mail.gmail.com>
Message-ID: <CABMX_w-NPwmW3DAQP4e25nKxzVFXUJKg9yj3ufZ=71v298fBfg@mail.gmail.com>

Please report bugs to JIRA. Make sure to attach the scenario and all the
info you have.

On Mon, Jun 25, 2018 at 9:49 AM, Atul Thosar <atulthosar at gmail.com> wrote:

> I have ran the same scenario (10 CPS, With Mime as payload) on master
> branch (FreeSWITCH Version 1.9.0+git~20180619T173242Z~25e9376b29~64bit
> (git 25e9376 2018-06-19 17:32:42Z 64bit)) and observed same issue of memory
> increasing continuously.
>
>
> --
> Atul
>
>
> On 24 June 2018 at 16:38, Atul Thosar <atulthosar at gmail.com> wrote:
>
>> Hi All,
>> I am verifying if we can use FreeSWITCH in our Production environment for
>> one of our IVR use case. Our use case involves INVITE containing MIME as a
>> payload, which contains XML and SDP parts.
>> When I ran 10 CPS load with MIME as a payload, I see Memory is
>> continuously growing. Where as same load without MIME (only SDP) payload
>> shows no increase in memory.
>>
>> Is this a known issue? If yes, is this already fixed or patch is
>> available for the same?
>>
>> I am using FreeSWITCH Version 1.6.20+git~20180123T214909Z~987c9b9a2a~64bit
>> (git 987c9b9 2018-01-23 21:49:09Z 64bit)
>> Attached output of top command for both the runs.
>>
>> Btw Following are the changes made in FS configuration -
>>
>> conf/vars.xml
>> +  <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.56.101"/>
>> +  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/>
>>
>> conf/dialplan/public/00_inbound_did.xml changed as following
>>
>> <include>
>>   <extension name="public_did">
>>     <condition field="destination_number" expression="^500$">
>>         <action application="answer"/>
>>         <action application="sched_hangup" data="+5 alloted_timeout"/>
>>         <action application="park"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>>
>> conf/autoload_configs/acl.conf.xml
>>
>> + <node type="allow" cidr="192.168.56.102/24"/>
>>
>> Thanks in advance.
>>
>> --
>> Atul
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 
Ítalo Rossi
italo at freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180626/9332356b/attachment-0001.html>

From victor.bogatyryev at gmail.com  Wed Jun 27 07:55:32 2018
From: victor.bogatyryev at gmail.com (Victor Bogatyryev)
Date: Wed, 27 Jun 2018 10:55:32 +0300
Subject: [Freeswitch-users] m=audio 0 RTP/AVP 19
Message-ID: <c5aee132-257a-f270-ee78-46a2fd72f21a@gmail.com>

Hello,

I apologize for my bad English.

Three years ago there was a message 
http://lists.freeswitch.org/pipermail/freeswitch-users/2015-June/113829.html 
and here I am faced with the same problem.
A stupid provider sends crypto in RTP/AVP and to ensure processing this 
request, I inserted in the profile the parameter 
NDLB-allow-crypto-in-avp=true
But Freeswitch answers to INVITE inserting in OK this line: m=audio 0 
RTP/AVP 19
and the provider understands this as "no audio" and does not send ACK 
and does not send rtp.

How to deal with this I do not know.

Regards.

Victor Bogatyryev

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180627/a706ecff/attachment-0001.html>

From mjlopez at smartic.es  Wed Jun 27 12:11:53 2018
From: mjlopez at smartic.es (=?UTF-8?Q?Miguel_Jes=C3=BAs_L=C3=B3pez_Valverde?=)
Date: Wed, 27 Jun 2018 14:11:53 +0200
Subject: [Freeswitch-users] freeswitch internal profile hang freequently
In-Reply-To: <CAAwzf1mL2GMgmZTh6G_gUwX3=w5dWEwdjn-S5WO+5=VGb=FdRw@mail.gmail.com>
References: <CAAwzf1mL2GMgmZTh6G_gUwX3=w5dWEwdjn-S5WO+5=VGb=FdRw@mail.gmail.com>
Message-ID: <05e001d40e10$099da920$1cd8fb60$@smartic.es>

Hello Sukitha:

 

There is a bug recognized about the version 1.6 that produces this situation and, at present, the correction to this bug on this version has not been applied. 

 

The patch is applied on versions 1.8 and 1.9 but this versions are not yet available to the community.

 

 

 

De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de sukitha jayasinghe
Enviado el: domingo, 24 de junio de 2018 7:03
Para: freeswitch-users at lists.freeswitch.org
Asunto: [Freeswitch-users] freeswitch internal profile hang freequently

 

Dear All,

 

I have a freeswitch (1.6.20) server in production with db configured to postgresql through pgpool2. This system hangs time to time and has become a major issue. It happen only to the internal profile where our UACs connects through webrtc. After the incident profile stop responding to cli commands, only solution is restarting the entire application.  I managed to get the back trace which indicate system waiting forever for profile database mutex. Further, It seems thread #19 is waiting for result from pgsql server cause this issue. Please find the back trace link below,

 

https://pastebin.com/Uqe8HjZn

 

Has anyone experienced this issue before, what is the solution for this. 

 

Best Regards,

Sukitha. 



---
El software de antivirus Avast ha analizado este correo electrónico en busca de virus.
https://www.avast.com/antivirus
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180627/044d8754/attachment-0001.html>

From joelists at tm.net.uk  Wed Jun 27 12:33:37 2018
From: joelists at tm.net.uk (Joseph Waite)
Date: Wed, 27 Jun 2018 13:33:37 +0100
Subject: [Freeswitch-users] Manipulate the sip: string in the INVITE header
	on an outgoing Bridge
Message-ID: <A6BC6EF1-C8A3-4110-BBAF-1796DD09CFB4@tm.net.uk>

Hi Guys

Were having issues with Asterisk boxes registering to a trunk on our FreeSwitch box.

We are sending the DID number called down the trunk in the to: field, however Asterisk doesn’t seem to like this.

Is there anyway to manipulate the INVITE to be INVITE sip:DID@ <sip:DID@> instead of what is currently INVITE sip:user@ <sip:user@>

Im hoping there is something like sip_invite_to_uri, however google is failing me.

Regards
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180627/c3c7ae3a/attachment-0001.html>

From melekoktay at gmail.com  Thu Jun 28 07:37:21 2018
From: melekoktay at gmail.com (Melek Oktay)
Date: Thu, 28 Jun 2018 09:37:21 +0200
Subject: [Freeswitch-users] Send display update after RTP session is
	establihed ?
Message-ID: <CAOopgPXP3bTmT77P2LMOsYF+wgpXkQYAvxa-t+LiYhxEaK2d-g@mail.gmail.com>

Hi,

I realize very interesting problem of my software stack (FreeSwitch +
Kamailio)  with  Yealink 41P Voip phone..

After 3-4 days research, I realize Yealink phone does not try to start RTP
session if it gets UPDATE packet before ACK packet !!

In Freeswitch side, do we have a option that send UPDATE packet after RTP
session is established? (we have a option in FreeSwitch side: do not send
UPDATE packet  however this is not good case for me, I update CallerID
with this UPDATE packet)
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180628/e7570132/attachment-0001.html>

From shakumarsoftware at gmail.com  Tue Jun 26 17:07:32 2018
From: shakumarsoftware at gmail.com (Sharath Kumar)
Date: Tue, 26 Jun 2018 11:07:32 -0600
Subject: [Freeswitch-users] Incoming RTP with different sequence number
 and host
In-Reply-To: <CAE8fD+3zQrQDczyHXqJhdmMvTkjwc_w0McqP=FnTCSteAJrDkQ@mail.gmail.com>
References: <CA+wj8uaAq0yhCwzy5=63nW4e3yZog1_-W4d-6S-4uftSf7WvYg@mail.gmail.com>
 <CAE8fD+3zQrQDczyHXqJhdmMvTkjwc_w0McqP=FnTCSteAJrDkQ@mail.gmail.com>
Message-ID: <CA+wj8uajv5+Gj8acx2sFzZLKG9XBqDLOMB1sbMaVD7tOZ4isbg@mail.gmail.com>

Thanks Naveen! I will try this but I thought that was only for packets that
FS generates.

On Mon, Jun 25, 2018 at 11:22 PM, Naveen Khanna <naveen.khanna.bm at gmail.com>
wrote:

> Try  <param name="rtp-rewrite-timestamps" value="true”/> in the sip
> profile.
>
> Regards,
>
> Naveen Khanna
>
> On 26 June 2018 at 6:37:06 AM, Sharath Kumar (shakumarsoftware at gmail.com)
> wrote:
>
> Hello,
> I am running FS 1.6.18 and have a provider that is sending me a stream of
> incoming RTP that changes it's host IP, and RTP sequence number and
> Timestamps after 10 mins in the call. The SSRC does not change. As a result
> I am getting either silence or garbled audio. The Sample rate is 20ms as in
> the original stream. Is there any setting in the FS that can make this work
> ? Have any of seen this work with inbound stream ?
> Thank you,
> Shaks
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180626/c6b221d7/attachment-0001.html>

From mircea.bh at outlook.com  Tue Jun 26 18:57:32 2018
From: mircea.bh at outlook.com (Mircea BH)
Date: Tue, 26 Jun 2018 18:57:32 +0000
Subject: [Freeswitch-users] Voicemail mwi multiple mailboxes notify one user
Message-ID: <VI1PR0301MB2365B31786AB1B2D6ADB0483E9490@VI1PR0301MB2365.eurprd03.prod.outlook.com>

Hello,

I would like to know if there is any possibility to notify one user when messages are left in multiple mailboxes?

I want to have just one registered user which should monitor multiple voicemail mailboxes.

Thank you for your time.

Best regards,
Mircea
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180626/c6f053ad/attachment-0001.html>

From davidswalkabout at gmail.com  Wed Jun 27 05:46:28 2018
From: davidswalkabout at gmail.com (David P)
Date: Tue, 26 Jun 2018 22:46:28 -0700
Subject: [Freeswitch-users] Silent calls, non-silent recordings
Message-ID: <CAE23trvPwRqdOQNoTn+NerJw2cL01ZT9u2-fe1mSEusRRBRB8Q@mail.gmail.com>

We've started seeing calls that aren't audible but in which both legs of
the video conference *are* audible in recordings made at the same time.
Here are some of the variables:

   1. We have two clouds with one FS in each
   2. Both clouds have been configured for TLSv1.2 + SRTP
   3. The network public/private IPs and port openings are the same
   4. Bria for Windows works fine with "verify TLS certificate" on
   5. Bria for Mac gets no audio regardless of security setting
   6. A verto page pointed at one of the clouds works for Chrome/Windows
   and Safari/MacOS but doesn't work for either when pointed at the other cloud
   7. I have looked at the FS log at debug level for one of these silent
   calls, and all of the codec negotiations go as expected and there are no
   errors.

The problem I'd most like help with is finding possible reasons why webRTC
calls are silent but have non-silent recordings.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180626/933b7af9/attachment-0001.html>

From gregor at infomedia.si  Wed Jun 27 14:38:17 2018
From: gregor at infomedia.si (Gregor Nanger)
Date: Wed, 27 Jun 2018 16:38:17 +0200
Subject: [Freeswitch-users] Recording calls
Message-ID: <CAC7ttFGbChZTgDWu5Zb-2KOefMrt5Qe8OdciPNfzYWYE++svww@mail.gmail.com>

Hi!

One question.

Can I record calls with http_api or http_cache modules and post it to
webserver?

If yes, does anyone one have any example?

Best regards, Gregor
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180627/b2d20ddd/attachment-0001.html>

From davidswalkabout at gmail.com  Wed Jun 27 15:41:41 2018
From: davidswalkabout at gmail.com (David P)
Date: Wed, 27 Jun 2018 08:41:41 -0700
Subject: [Freeswitch-users] Silent calls, non-silent recordings
In-Reply-To: <CAE23trvPwRqdOQNoTn+NerJw2cL01ZT9u2-fe1mSEusRRBRB8Q@mail.gmail.com>
References: <CAE23trvPwRqdOQNoTn+NerJw2cL01ZT9u2-fe1mSEusRRBRB8Q@mail.gmail.com>
Message-ID: <CAE23trs4HQSyH3=RQg1bz0CjXVyHh3K+zvQDYZP++mSkLSHcXg@mail.gmail.com>

Someone speculated that this might be due to the STUN server providing an
IPv6 address and that this is being cached.

On Tue, 26 Jun 2018, 10:46 pm David P, <davidswalkabout at gmail.com> wrote:

> We've started seeing calls that aren't audible but in which both legs of
> the video conference *are* audible in recordings made at the same time.
> Here are some of the variables:
>
>    1. We have two clouds with one FS in each
>    2. Both clouds have been configured for TLSv1.2 + SRTP
>    3. The network public/private IPs and port openings are the same
>    4. Bria for Windows works fine with "verify TLS certificate" on
>    5. Bria for Mac gets no audio regardless of security setting
>    6. A verto page pointed at one of the clouds works for Chrome/Windows
>    and Safari/MacOS but doesn't work for either when pointed at the other cloud
>    7. I have looked at the FS log at debug level for one of these silent
>    calls, and all of the codec negotiations go as expected and there are no
>    errors.
>
> The problem I'd most like help with is finding possible reasons why webRTC
> calls are silent but have non-silent recordings.
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180627/d1a6721d/attachment-0001.html>

From davidswalkabout at gmail.com  Wed Jun 27 23:47:31 2018
From: davidswalkabout at gmail.com (David P)
Date: Wed, 27 Jun 2018 16:47:31 -0700
Subject: [Freeswitch-users] Silent calls, non-silent recordings
In-Reply-To: <CAE23trs4HQSyH3=RQg1bz0CjXVyHh3K+zvQDYZP++mSkLSHcXg@mail.gmail.com>
References: <CAE23trvPwRqdOQNoTn+NerJw2cL01ZT9u2-fe1mSEusRRBRB8Q@mail.gmail.com>
 <CAE23trs4HQSyH3=RQg1bz0CjXVyHh3K+zvQDYZP++mSkLSHcXg@mail.gmail.com>
Message-ID: <CAE23trtO=wvQobnT3StKuJt0uFo7yGPuo7fGT2n1Xcw3H-GzZg@mail.gmail.com>

It seems that sip_profiles/internal.xml might be relevant here. But in
https://freeswitch.org/confluence/display/FREESWITCH/Configuring+FreeSWITCH#ConfiguringFreeSWITCH-Internal
this link is broken:
https://wiki.freeswitch.org/wiki/Getting_Started_Guide#directory

and so is: https://wiki.freeswitch.org/wiki/Getting_Started_Guide#contexts

and https://wiki.freeswitch.org/wiki/Dialplan_Recipes

On Wed, Jun 27, 2018 at 8:41 AM David P <davidswalkabout at gmail.com> wrote:

> Someone speculated that this might be due to the STUN server providing an
> IPv6 address and that this is being cached.
>
> On Tue, 26 Jun 2018, 10:46 pm David P, <davidswalkabout at gmail.com> wrote:
>
>> We've started seeing calls that aren't audible but in which both legs of
>> the video conference *are* audible in recordings made at the same time.
>> Here are some of the variables:
>>
>>    1. We have two clouds with one FS in each
>>    2. Both clouds have been configured for TLSv1.2 + SRTP
>>    3. The network public/private IPs and port openings are the same
>>    4. Bria for Windows works fine with "verify TLS certificate" on
>>    5. Bria for Mac gets no audio regardless of security setting
>>    6. A verto page pointed at one of the clouds works for Chrome/Windows
>>    and Safari/MacOS but doesn't work for either when pointed at the other cloud
>>    7. I have looked at the FS log at debug level for one of these silent
>>    calls, and all of the codec negotiations go as expected and there are no
>>    errors.
>>
>> The problem I'd most like help with is finding possible reasons why
>> webRTC calls are silent but have non-silent recordings.
>>
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180627/c8e21c0e/attachment-0001.html>

From cong.wang.itsherpa at gmail.com  Thu Jun 28 07:30:00 2018
From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=)
Date: Thu, 28 Jun 2018 16:30:00 +0900
Subject: [Freeswitch-users] About Verto
Message-ID: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com>

Hey all,

I had tried to build a WebRTC video call service via mod_verto and Verto Comunicator. 
Now I could make calls from (to) an Android device with Chrome, but it didn’t work on iOS with Safari.
Is there any point to be modified to run on Safari?

Regards.

From mouli123 at gmail.com  Thu Jun 28 14:15:46 2018
From: mouli123 at gmail.com (Chandramouli P)
Date: Thu, 28 Jun 2018 19:45:46 +0530
Subject: [Freeswitch-users] Unable to find modules.conf file after
	installation through RPM file
Message-ID: <CALR0W6_eyNJkn4U+0AKR7idLo1gSPmE1SUwodn=dGLcYHOjS4w@mail.gmail.com>

Hello,

I installed FreeSwitch through RPM file as below:

sudo rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-6.noarch.rpm

sudo yum install freeswitch-config-vanilla sox freeswitch-sounds*
freeswitch-lang* freeswitch-lua freeswitch-xml-cdr

Everything is working fine. But, when I am trying to enable or disable the
modules, I could not able to find the "modules.conf" files anywhere. Can
anybody tell me how to enable/disable the Freeswitch modules, if we install
the FreeSwitch through RPM?

Thanks in advance.

Regards,
CM.

<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
Virus-free.
www.avg.com
<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
<#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180628/fb23528e/attachment.html>

From francesco at delagarda.com  Thu Jun 28 14:52:34 2018
From: francesco at delagarda.com (Francesco Facco de Lagarda)
Date: Thu, 28 Jun 2018 16:52:34 +0200
Subject: [Freeswitch-users] About Verto
In-Reply-To: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com>
References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com>
Message-ID: <F7B9EAD5-39A7-4508-89D6-50666174460B@delagarda.com>

iOS safari dies not support webrtc yet....
Apple keeps promising but....

Francesco Facco de Lagarda


> On 28 Jun 2018, at 09:30, 王聡 <cong.wang.itsherpa at gmail.com> wrote:
> 
> Hey all,
> 
> I had tried to build a WebRTC video call service via mod_verto and Verto Comunicator. 
> Now I could make calls from (to) an Android device with Chrome, but it didn’t work on iOS with Safari.
> Is there any point to be modified to run on Safari?
> 
> Regards.
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com



From gmaruzz at gmail.com  Thu Jun 28 15:00:23 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 Jun 2018 17:00:23 +0200
Subject: [Freeswitch-users] About Verto
In-Reply-To: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com>
References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com>
Message-ID: <CALXCt0oJBX3PRRfKigU0w9w=Gm3jHEEU8f8BLJ99eVqecYrt1A@mail.gmail.com>

On 28 June 2018 at 09:30, 王聡 <cong.wang.itsherpa at gmail.com> wrote:

>
> I had tried to build a WebRTC video call service via mod_verto and Verto
> Comunicator.
> Now I could make calls from (to) an Android device with Chrome, but it
> didn’t work on iOS with Safari.
> Is there any point to be modified to run on Safari?
>

be sure to enable H264 codec, both in vars.conf.xml and in verto.conf.xml





>
> Regards.
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com




-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180628/c510000c/attachment.html>

From gmina at connectfirst.com  Thu Jun 28 21:02:15 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Thu, 28 Jun 2018 15:02:15 -0600
Subject: [Freeswitch-users] FireFox 61 + Freeswitch 1.6.20
Message-ID: <CAKzROs2e6RoMeHRn6nOqmJmX8Qs9Tr5jTUyRBvn27HwRV=H7VQ@mail.gmail.com>

Is anyone else having issues with WebRTC and the latest release of
FireFox?  We are showing one way media  - with media flowing properly TO
the browser, but no media coming back from the browser.

Everything in the logs looks OK - no errors.  We are using SIP.js as the
library of choice for signaling.

We are showing SRTP packets flowing bi-directionally (using tcpdump), but
since they are DTLS encrypted, I am unable to see exactly what is in each
packet to determine if there is media being sent from the browser or they
are just empty packets.

Any ideas and/or other folks seeing the same issue - would love to hear
your thoughts.

Thanks,
Geoff
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180628/e8ba0240/attachment.html>

From nathandownes at hotmail.com  Thu Jun 28 21:50:55 2018
From: nathandownes at hotmail.com (Nathan Downes)
Date: Thu, 28 Jun 2018 21:50:55 +0000
Subject: [Freeswitch-users] Manipulate the sip: string in the INVITE
 header	on an outgoing Bridge
In-Reply-To: <A6BC6EF1-C8A3-4110-BBAF-1796DD09CFB4@tm.net.uk>
References: <A6BC6EF1-C8A3-4110-BBAF-1796DD09CFB4@tm.net.uk>
Message-ID: <FR1P152MB1287CDFA77DDC459C5993C47D74F0@FR1P152MB1287.LAMP152.PROD.OUTLOOK.COM>

Hi Joseph,

I use the below for similar use case, on an inbound route, call comes in, gets bridged direct to pbx.

{sip_invite_to_uri=<sip:${destination_number}@${domain}>}user/12341234@${domain}

12341234 being the ext you register pbx as on FS

I have used another that sends call direct to IP, but found if the pbx was behind NAT this way didn’t use the NDLB type NAT fixes, this one still takes them into consideration.


From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joseph Waite
Sent: Wednesday, 27 June 2018 10:34 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Manipulate the sip: string in the INVITE header on an outgoing Bridge

Hi Guys

Were having issues with Asterisk boxes registering to a trunk on our FreeSwitch box.

We are sending the DID number called down the trunk in the to: field, however Asterisk doesn’t seem to like this.

Is there anyway to manipulate the INVITE to be INVITE sip:DID@ instead of what is currently INVITE sip:user@

Im hoping there is something like sip_invite_to_uri, however google is failing me.

Regards
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180628/584a224a/attachment-0001.html>

From andrew.keil at visytel.com  Thu Jun 28 22:47:45 2018
From: andrew.keil at visytel.com (Andrew Keil)
Date: Thu, 28 Jun 2018 22:47:45 +0000
Subject: [Freeswitch-users]
 =?utf-8?b?562U5aSNOiAgRnJlZXN3aXRjaCB3aW5kb3dz?=
 =?utf-8?q?_build_-_libpng?=
In-Reply-To: <PS1PR03MB15165E0C66223480F25A31F2A9490@PS1PR03MB1516.apcprd03.prod.outlook.com>
References: <CAC7ttFFMUGXbs9Qj3Doi3MRuqFRohh5i-9MERm1E8sWQwF=okQ@mail.gmail.com>
 <PS1PR03MB15165E0C66223480F25A31F2A9490@PS1PR03MB1516.apcprd03.prod.outlook.com>
Message-ID: <MEXPR01MB1223A719E07EB9E2C7FFDDA5F34F0@MEXPR01MB1223.ausprd01.prod.outlook.com>

Gregor,

Or you can make the following change to the zlib project for All Configurations and All Releases - C/C++ - All Options:
- Disable Specific Warnings (add to end): ;5045
- Whole Program Optimization: No

Hope that helps.

Andrew

From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> On Behalf Of bob. chen
Sent: Tuesday, 26 June 2018 1:23 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] 答复: Freeswitch windows build - libpng

try vs2015 ;)


发件人: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] 代表 Gregor Nanger
发送时间: 2018年6月23日 19:08
收件人: FreeSWITCH Users Help
主题: [Freeswitch-users] Freeswitch windows build - libpng

Hi!

After a year I wanted to build FS on windows and make video tutorial or for FS wiki.

I am using latest VS2017 and everything builds except libpng project. Below is error if someone can help me, because I don't have c++ experience:

1>------ Build started: Project: Download zlib, Configuration: Release Win32 ------
1>Downloading zlib.
2>------ Build started: Project: libpng, Configuration: Release x64 ------
2>Generating pnglibconf.h
2>        1 file(s) copied.
2>png.c
2>pngerror.c
2>pngget.c
2>pngmem.c
2>pngpread.c
2>pngread.c
2>pngrio.c
2>pngrtran.c
2>pngrutil.c
2>pngset.c
2>pngtrans.c
2>pngwio.c
2>pngwrite.c
2>pngwtran.c
2>pngwutil.c
2>Generating Code...
2>zlib.lib(inflate.obj) : MSIL .netmodule or module compiled with /GL found; restarting link with /LTCG; add /LTCG to the link command line to improve linker performance
2>   Creating library D:\Git\freeswitch\x64\Release\libpng16.lib and object D:\Git\freeswitch\x64\Release\libpng16.exp
2>Generating code
2>d:\git\freeswitch\libs\zlib\deflate.c(2097): error C2220: warning treated as error - no 'executable' file generated
2>d:\git\freeswitch\libs\zlib\deflate.c(2097): warning C5045: Compiler will insert Spectre mitigation for memory load if /Qspectre switch specified
2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: index 'dist' range checked by comparison on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(2097) : note: feeds memory load on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1987): warning C5045: Compiler will insert Spectre mitigation for memory load if /Qspectre switch specified
2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: index 'dist' range checked by comparison on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1987) : note: feeds memory load on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1862): warning C5045: Compiler will insert Spectre mitigation for memory load if /Qspectre switch specified
2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: index 'dist' range checked by comparison on this line
2>d:\git\freeswitch\libs\zlib\deflate.c(1862) : note: feeds memory load on this line
2>LINK : fatal error LNK1257: code generation failed
2>Done building project "libpng.vcxproj" -- FAILED.

Best regards, Gregor

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180628/2992b518/attachment-0001.html>

From gmaruzz at gmail.com  Thu Jun 28 23:38:42 2018
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 29 Jun 2018 01:38:42 +0200
Subject: [Freeswitch-users] About Verto
In-Reply-To: <F7B9EAD5-39A7-4508-89D6-50666174460B@delagarda.com>
References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com>
 <F7B9EAD5-39A7-4508-89D6-50666174460B@delagarda.com>
Message-ID: <CALXCt0qXNpGU=JhEkaH-03nib-dLdZ6Gyhg5YX7_UA5uPkMTHg@mail.gmail.com>

I believe Safari do support webrtc in current ios, it just do not support
vp8, h264 only.


On Thu, Jun 28, 2018, 23:38 Francesco Facco de Lagarda <
francesco at delagarda.com> wrote:

> iOS safari dies not support webrtc yet....
> Apple keeps promising but....
>
> Francesco Facco de Lagarda
>
>
> > On 28 Jun 2018, at 09:30, 王聡 <cong.wang.itsherpa at gmail.com> wrote:
> >
> > Hey all,
> >
> > I had tried to build a WebRTC video call service via mod_verto and Verto
> Comunicator.
> > Now I could make calls from (to) an Android device with Chrome, but it
> didn’t work on iOS with Safari.
> > Is there any point to be modified to run on Safari?
> >
> > Regards.
> > _________________________________________________________________________
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> >
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180629/cbd08e0e/attachment-0001.html>

From alex at freeswitch.com  Fri Jun 29 00:44:22 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Fri, 29 Jun 2018 09:44:22 +0900
Subject: [Freeswitch-users] Unable to find modules.conf file after
 installation through RPM file
In-Reply-To: <CALR0W6_eyNJkn4U+0AKR7idLo1gSPmE1SUwodn=dGLcYHOjS4w@mail.gmail.com>
References: <CALR0W6_eyNJkn4U+0AKR7idLo1gSPmE1SUwodn=dGLcYHOjS4w@mail.gmail.com>
Message-ID: <CAGJLwFjnsPHthkE8Yn013YY9hN5CgiGX6SovnKUp_UUHe=VxVg@mail.gmail.com>

Hello,

"modules.conf" is used on compile time only, you need modules.conf.xml
somewhere in /etc/freeswitch/autoload_configs/

Regards,

Alex

On Thu, Jun 28, 2018 at 11:15 PM, Chandramouli P <mouli123 at gmail.com> wrote:

> Hello,
>
> I installed FreeSwitch through RPM file as below:
>
> sudo rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-6.noarch.
> rpm
>
> sudo yum install freeswitch-config-vanilla sox freeswitch-sounds*
> freeswitch-lang* freeswitch-lua freeswitch-xml-cdr
>
> Everything is working fine. But, when I am trying to enable or disable the
> modules, I could not able to find the "modules.conf" files anywhere. Can
> anybody tell me how to enable/disable the Freeswitch modules, if we install
> the FreeSwitch through RPM?
>
> Thanks in advance.
>
> Regards,
> CM.
>
>
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free.
> www.avg.com
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
> <#m_-9201595567498981960_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 
Alex Sibyakin | Support Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180629/1f8ae3ae/attachment-0001.html>

From alex at freeswitch.com  Fri Jun 29 01:10:02 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Fri, 29 Jun 2018 10:10:02 +0900
Subject: [Freeswitch-users] Manipulate the sip: string in the INVITE
 header on an outgoing Bridge
In-Reply-To: <A6BC6EF1-C8A3-4110-BBAF-1796DD09CFB4@tm.net.uk>
References: <A6BC6EF1-C8A3-4110-BBAF-1796DD09CFB4@tm.net.uk>
Message-ID: <CAGJLwFh5HgN=xR2q=jRTzdtTLDednznqG2d4c_s__m3hLdQL2Q@mail.gmail.com>

Hi,

I'm not sure that I got your scenario right but you can try to do something
like this:

<action application="bridge" data="{sip_req_uri=$1}user/foobar"/> (you may
need another var, full list is in Confluence)

Regards,

Alex



On Wed, Jun 27, 2018 at 9:33 PM, Joseph Waite <joelists at tm.net.uk> wrote:

> Hi Guys
>
> Were having issues with Asterisk boxes registering to a trunk on our
> FreeSwitch box.
>
> We are sending the DID number called down the trunk in the to: field,
> however Asterisk doesn’t seem to like this.
>
> Is there anyway to manipulate the INVITE to be INVITE sip:DID@ instead of
> what is currently INVITE sip:user@
>
> Im hoping there is something like sip_invite_to_uri, however google is
> failing me.
>
> Regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 
Alex Sibyakin | Support Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180629/c6e52db0/attachment-0001.html>

From alex at freeswitch.com  Fri Jun 29 02:19:47 2018
From: alex at freeswitch.com (Alexey Sibyakin)
Date: Fri, 29 Jun 2018 11:19:47 +0900
Subject: [Freeswitch-users] video on conference with PC and MAC browser
 works but fail with iPhone safari via webrtc
In-Reply-To: <CABHWxvw0YvoWqmvx8_jBYQTU=E+Q7d-cy3K3tsR-Lc2pMr+DWA@mail.gmail.com>
References: <CABHWxvw0YvoWqmvx8_jBYQTU=E+Q7d-cy3K3tsR-Lc2pMr+DWA@mail.gmail.com>
Message-ID: <CAGJLwFjtRY8b=Q=40LXh-QCxPUh_5x+vf=Rppqdehc-cWirmmA@mail.gmail.com>

Hi,

try mux instead of transcode

Regards,

Alex

On Tue, Jun 26, 2018 at 12:47 PM, 薛光宏 <albert.hsueh at cloudigit.com> wrote:

> Hi All,
>
> here is our simple architecture
> <client A>---<FreeSWITCH>
> <client B>----/
>
> - client A and B are both registered on FreeSWITCH via websocket (using
> sipjs 0.9.2)
> - we enable video-mode= transcode and load mod_av on FreeSWITCH.
> - if client A uses chrome on PC and client B use safari on *MAC*, they
> can see each other, talk to each other in the same conference.
> - if client A uses chrome on PC and client B use safari on *iPhone*(8plus,
> iOS 11.4), they can NOT see each other, talk to each other in the same
> conference.
>
> I though safari on MAC works, then safari on iPhone should work, but not.
> Is it possible to fix that with changing configuration of FreeSWITCH? or
> any suggestion?
>
> Thanks in advance.
>
> FreeSWITCH Version 1.6.20
> Ubuntu Server Ubuntu 14.04.1 LTS 64bit
> libav-11.4 or libav-12
>
> See below configurations:
> 1. enable video-mode= transcode
> <param name="conference-flags" value="transcode-video|
> minimize-video-encoding"/>
> <param name="video-mode" value="transcode"/>
>
> 2.enable VP8/H264 codec support
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,VP8,H264"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,VP8,H264"/>
>
> 3.enable mod_av and disable mod_h26x
> <load module="mod_av"/>
> <!--<load module="mod_h26x"/>-->
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>



-- 
Alex Sibyakin | Support Engineer
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
<https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: alex at freeswitch.com
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
Need commercial support? Contact sales at freeswitch.com for details.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180629/5bc6a3d2/attachment-0001.html>

From atulthosar at gmail.com  Fri Jun 29 04:59:39 2018
From: atulthosar at gmail.com (Atul Thosar)
Date: Fri, 29 Jun 2018 10:29:39 +0530
Subject: [Freeswitch-users] [FreeSWITCH v1.6.20] Memory increasing
	continuously
In-Reply-To: <CABMX_w-NPwmW3DAQP4e25nKxzVFXUJKg9yj3ufZ=71v298fBfg@mail.gmail.com>
References: <CAKT7MpqD-KSfGZO7MQeyXcNmEVhr43MSQOitJTPFgHmvy+Wk+A@mail.gmail.com>
 <CAKT7Mppe2HD9_knJ6Gz6WhgyxWG2wnQhmA7mTAj1jweL+_0+KA@mail.gmail.com>
 <CABMX_w-NPwmW3DAQP4e25nKxzVFXUJKg9yj3ufZ=71v298fBfg@mail.gmail.com>
Message-ID: <CAKT7MprXW7WgiwqktoNhJqdWn2TQataq9VK0i66bma_7RpA7Gw@mail.gmail.com>

Hi
​
Ítalo Rossi,
My colleague has reported this - https://freeswitch.org/jira/browse/FS-11205


--
BR,
Atul


On 26 June 2018 at 17:48,
​​
Ítalo Rossi <italo at freeswitch.org> wrote:

> Please report bugs to JIRA. Make sure to attach the scenario and all the
> info you have.
>
> On Mon, Jun 25, 2018 at 9:49 AM, Atul Thosar <atulthosar at gmail.com> wrote:
>
>> I have ran the same scenario (10 CPS, With Mime as payload) on master
>> branch (FreeSWITCH Version 1.9.0+git~20180619T173242Z~25e9376b29~64bit
>> (git 25e9376 2018-06-19 17:32:42Z 64bit)) and observed same issue of memory
>> increasing continuously.
>>
>>
>> --
>> Atul
>>
>>
>> On 24 June 2018 at 16:38, Atul Thosar <atulthosar at gmail.com> wrote:
>>
>>> Hi All,
>>> I am verifying if we can use FreeSWITCH in our Production environment
>>> for one of our IVR use case. Our use case involves INVITE containing MIME
>>> as a payload, which contains XML and SDP parts.
>>> When I ran 10 CPS load with MIME as a payload, I see Memory is
>>> continuously growing. Where as same load without MIME (only SDP) payload
>>> shows no increase in memory.
>>>
>>> Is this a known issue? If yes, is this already fixed or patch is
>>> available for the same?
>>>
>>> I am using FreeSWITCH Version 1.6.20+git~20180123T214909Z~987c9b9a2a~64bit
>>> (git 987c9b9 2018-01-23 21:49:09Z 64bit)
>>> Attached output of top command for both the runs.
>>>
>>> Btw Following are the changes made in FS configuration -
>>>
>>> conf/vars.xml
>>> +  <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.56.101"/>
>>> +  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/>
>>>
>>> conf/dialplan/public/00_inbound_did.xml changed as following
>>>
>>> <include>
>>>   <extension name="public_did">
>>>     <condition field="destination_number" expression="^500$">
>>>         <action application="answer"/>
>>>         <action application="sched_hangup" data="+5 alloted_timeout"/>
>>>         <action application="park"/>
>>>     </condition>
>>>   </extension>
>>> </include>
>>>
>>>
>>> conf/autoload_configs/acl.conf.xml
>>>
>>> + <node type="allow" cidr="192.168.56.102/24"/>
>>>
>>> Thanks in advance.
>>>
>>> --
>>> Atul
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Services
>> sales at freeswitch.com
>> https://freeswitch.com
>>
>> Official FreeSWITCH Sites
>> https://freeswitch.com/oss
>> https://freeswitch.org/confluence
>> https://cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> https://freeswitch.com
>>
>
>
>
> --
> Ítalo Rossi
> italo at freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
>
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180629/a6308398/attachment-0001.html>

From domrumsey at hotmail.com  Fri Jun 29 08:53:46 2018
From: domrumsey at hotmail.com (Dom Rumsey)
Date: Fri, 29 Jun 2018 08:53:46 +0000
Subject: [Freeswitch-users] About Verto
In-Reply-To: <CALXCt0oJBX3PRRfKigU0w9w=Gm3jHEEU8f8BLJ99eVqecYrt1A@mail.gmail.com>
References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com>,
 <CALXCt0oJBX3PRRfKigU0w9w=Gm3jHEEU8f8BLJ99eVqecYrt1A@mail.gmail.com>
Message-ID: <CY1PR16MB031695774F2C7269A991D28CAC4E0@CY1PR16MB0316.namprd16.prod.outlook.com>

Yeah, iOS won't work if you're using VP8 codec, you to connect with h264.

Also, does anyone know how to change Verto to 16:9 aspect ratio?


________________________________
From: FreeSWITCH-users <freeswitch-users-bounces at lists.freeswitch.org> on behalf of Giovanni Maruzzelli <gmaruzz at gmail.com>
Sent: Thursday, June 28, 2018 3:00 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] About Verto

On 28 June 2018 at 09:30, 王聡 <cong.wang.itsherpa at gmail.com<mailto:cong.wang.itsherpa at gmail.com>> wrote:

I had tried to build a WebRTC video call service via mod_verto and Verto Comunicator.
Now I could make calls from (to) an Android device with Chrome, but it didn’t work on iOS with Safari.
Is there any point to be modified to run on Safari?

be sure to enable H264 codec, both in vars.conf.xml and in verto.conf.xml





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

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

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



--

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180629/de1d0a64/attachment-0001.html>

From paul.muaddib83 at gmail.com  Fri Jun 29 17:38:54 2018
From: paul.muaddib83 at gmail.com (Paul Muaddib)
Date: Fri, 29 Jun 2018 19:38:54 +0200
Subject: [Freeswitch-users] Failed Registration with status DNS Error [503]
Message-ID: <CAEjNe_HTrNCWz4dEtXdYoTatYk4mneOnH=kVD-EHyshzQAhBsA@mail.gmail.com>

Hi,

how do I fix this problem:
Failed Registration with status DNS Error [503]?

I already search the net but couldn't find a proper answer. The only thing
that I found was something about DNS SRV Resource Record but this was fixed
in freeswitch a long time ago.

*My Setup:*
FreeSWITCH Version 1.6.19
Debian 7 wheezy
Freeswitch(Debian) -> Router(OpenBSD) -> Internet

*Failed Registration with status DNS Error [503]*
2018-06-29 19:27:21.732868 [NOTICE] sofia_reg.c:448 Registering telekom_voip
nua.c:622 nua_register() nua: nua_register: entering
nua_stack.c:529 nua_signal() nua(0x8c8c0c8): sent signal r_register
nua_stack.c:569 nua_stack_signal() nua(0x8c8c0c8): recv signal r_register
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:403 soa_set_params() soa_set_params(static::0xb5c1a298, ...) called
nua_dialog.c:338 nua_dialog_usage_add() nua(0x8c8c0c8): adding register
usage
nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0xb6279478)
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, NAPTR, "
sip-trunk.telekom.de.") called
sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, NAPTR, "
sip-trunk.telekom.de.") returned 1 entries
nta.c:10398 outgoing_query_naptr() nta: for "sip-trunk.telekom.de" query "
sip-trunk.telekom.de" NAPTR (cached)
sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, SRV, "_sip._
udp.sip-trunk.telekom.de.") called
sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, SRV, "_sip._
udp.sip-trunk.telekom.de.") returned 1 entries
nta.c:10598 outgoing_query_srv() nta: for "sip-trunk.telekom.de" query
"_sip._udp.sip-trunk.telekom.de" SRV (cached)
sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, A, "
sip-trunk.telekom.de.") called
sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, A, "
sip-trunk.telekom.de.") returned 1 entries
nta.c:10803 outgoing_query_a() nta: for "sip-trunk.telekom.de" query "
sip-trunk.telekom.de" A (cached)
nta.c:1350 set_timeout() nta: timer set to 32000 ms
nta.c:1348 set_timeout() nta: timer shortened to 5000 ms
nua_stack.c:271 nua_stack_event() nua(0x8c8c0c8): event r_register 503 DNS
Error
nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x8c8c0c8): removing
register usage
nta.c:4470 nta_leg_destroy() nta_leg_destroy(0xb6279478)
nua_stack.c:359 nua_application_event() nua: nua_application_event: entering
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2018-06-29 19:27:21.732868 [ERR] sofia_reg.c:2447 telekom_voip Failed
Registration with status DNS Error [503]. failure #34
nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
2018-06-29 19:27:22.772957 [WARNING] sofia_reg.c:505 telekom_voip Failed
Registration [503], setting retry to 30 seconds.
nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate REGISTER
(124810556)
nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil),
0xb60911e0)
nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout,
1/1 term, 1/1 free
nta.c:1289 agent_timer() nta: timer not set

*sofia_dig --tcp reg.sip-trunk.telekom.de <http://reg.sip-trunk.telekom.de>*
Preference        Weight     Transport          Port       Address
================================================================================
         1         1.000           tcp          5060    217.0.26.165
         2         1.000           tcp          5060    217.0.26.163
         3         1.000           tcp          5060    217.0.26.133

*Gateway settings*
<include>
 <gateway name="telekom_voip">
  <param name="username" value=„XXXXXXXX“/>
  <param name="password" value=„XXXXXXXX“/>

  <param name="proxy" value="reg.sip-trunk.telekom.de"/>
  <param name="register-proxy" value="sip-trunk.telekom.de"/>
  <param name="register" value="true"/>
  <param name="register-transport" value="tcp"/>
  <param name="expire-seconds" value="3600"/>

 </gateway>
</include>

Regards,
Paul
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180629/a026ce24/attachment-0001.html>

From paul.muaddib83 at gmail.com  Fri Jun 29 21:06:01 2018
From: paul.muaddib83 at gmail.com (Paul Muaddib)
Date: Fri, 29 Jun 2018 23:06:01 +0200
Subject: [Freeswitch-users] Failed Registration with status DNS Error
	[503]
In-Reply-To: <CAEjNe_HTrNCWz4dEtXdYoTatYk4mneOnH=kVD-EHyshzQAhBsA@mail.gmail.com>
References: <CAEjNe_HTrNCWz4dEtXdYoTatYk4mneOnH=kVD-EHyshzQAhBsA@mail.gmail.com>
Message-ID: <CAEjNe_FUB1xdyOnfjY5qUyjfOAMXHWhWZk4TK-v9uGc4XGXPHw@mail.gmail.com>

I am very sorry. It turned out to be a firewall setting :P

2018-06-29 19:38 GMT+02:00 Paul Muaddib <paul.muaddib83 at gmail.com>:

> Hi,
>
> how do I fix this problem:
> Failed Registration with status DNS Error [503]?
>
> I already search the net but couldn't find a proper answer. The only thing
> that I found was something about DNS SRV Resource Record but this was fixed
> in freeswitch a long time ago.
>
> *My Setup:*
> FreeSWITCH Version 1.6.19
> Debian 7 wheezy
> Freeswitch(Debian) -> Router(OpenBSD) -> Internet
>
> *Failed Registration with status DNS Error [503]*
> 2018-06-29 19:27:21.732868 [NOTICE] sofia_reg.c:448 Registering
> telekom_voip
> nua.c:622 nua_register() nua: nua_register: entering
> nua_stack.c:529 nua_signal() nua(0x8c8c0c8): sent signal r_register
> nua_stack.c:569 nua_stack_signal() nua(0x8c8c0c8): recv signal r_register
> nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
> soa.c:403 soa_set_params() soa_set_params(static::0xb5c1a298, ...) called
> nua_dialog.c:338 nua_dialog_usage_add() nua(0x8c8c0c8): adding register
> usage
> nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0xb6279478)
> nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
> sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, NAPTR, "
> sip-trunk.telekom.de.") called
> sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, NAPTR, "
> sip-trunk.telekom.de.") returned 1 entries
> nta.c:10398 outgoing_query_naptr() nta: for "sip-trunk.telekom.de" query "
> sip-trunk.telekom.de" NAPTR (cached)
> sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, SRV, "_sip._
> udp.sip-trunk.telekom.de.") called
> sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, SRV, "_sip._
> udp.sip-trunk.telekom.de.") returned 1 entries
> nta.c:10598 outgoing_query_srv() nta: for "sip-trunk.telekom.de" query
> "_sip._udp.sip-trunk.telekom.de" SRV (cached)
> sres_cache.c:272 sres_cache_get() sres_cache_get(0x8ccd7b0, A, "
> sip-trunk.telekom.de.") called
> sres_cache.c:318 sres_cache_get() sres_cache_get(0x8ccd7b0, A, "
> sip-trunk.telekom.de.") returned 1 entries
> nta.c:10803 outgoing_query_a() nta: for "sip-trunk.telekom.de" query "
> sip-trunk.telekom.de" A (cached)
> nta.c:1350 set_timeout() nta: timer set to 32000 ms
> nta.c:1348 set_timeout() nta: timer shortened to 5000 ms
> nua_stack.c:271 nua_stack_event() nua(0x8c8c0c8): event r_register 503 DNS
> Error
> nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x8c8c0c8): removing
> register usage
> nta.c:4470 nta_leg_destroy() nta_leg_destroy(0xb6279478)
> nua_stack.c:359 nua_application_event() nua: nua_application_event:
> entering
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2018-06-29 19:27:21.732868 [ERR] sofia_reg.c:2447 telekom_voip Failed
> Registration with status DNS Error [503]. failure #34
> nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering
> 2018-06-29 19:27:22.772957 [WARNING] sofia_reg.c:505 telekom_voip Failed
> Registration [503], setting retry to 30 seconds.
> nta.c:9101 outgoing_timer_dk() nta: timer K fired, terminate REGISTER
> (124810556)
> nta.c:8799 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil),
> 0xb60911e0)
> nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/0 tout,
> 1/1 term, 1/1 free
> nta.c:1289 agent_timer() nta: timer not set
>
> *sofia_dig --tcp reg.sip-trunk.telekom.de
> <http://reg.sip-trunk.telekom.de>*
> Preference        Weight     Transport          Port       Address
> ============================================================
> ====================
>          1         1.000           tcp          5060    217.0.26.165
>          2         1.000           tcp          5060    217.0.26.163
>          3         1.000           tcp          5060    217.0.26.133
>
> *Gateway settings*
> <include>
>  <gateway name="telekom_voip">
>   <param name="username" value=„XXXXXXXX“/>
>   <param name="password" value=„XXXXXXXX“/>
>
>   <param name="proxy" value="reg.sip-trunk.telekom.de"/>
>   <param name="register-proxy" value="sip-trunk.telekom.de"/>
>   <param name="register" value="true"/>
>   <param name="register-transport" value="tcp"/>
>   <param name="expire-seconds" value="3600"/>
>
>  </gateway>
> </include>
>
> Regards,
> Paul
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180629/57468c16/attachment-0001.html>

From paul.muaddib83 at gmail.com  Sat Jun 30 12:18:09 2018
From: paul.muaddib83 at gmail.com (Paul Muaddib)
Date: Sat, 30 Jun 2018 14:18:09 +0200
Subject: [Freeswitch-users] Failed Registration [908]
Message-ID: <CAEjNe_FKDDQpCO6QZpgD6rKZawddiLxaBWUMmZVPDRAw48ZwHQ@mail.gmail.com>

Hi,

can someone please look at it and tell my what I am doing wrong?

NOTICE] sofia_reg.c:448 Registering sip-trunk.telekom.de
[WARNING] sofia_reg.c:484 Timeout Registering sip-trunk.telekom.de
[WARNING] sofia_reg.c:505 sip-trunk.telekom.de Failed Registration [908],
setting retry to 30 seconds.

My setup:
########

I have a static IP address for my router. My router does not have UPnP or
NAT-PMP.
Phones and Freeswitch server are on the same local network. The local
network is behind NAT

PHONE -> FS -> NAT -> Public Internet
I only want to connect to the PSTN via my sip gateway provider.

According to this manual I did the following setting:
https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal

vars.xml
######

<X-PRE-PROCESS cmd="set" data="external_rtp_ip=X.X.X.X“/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=X.X.X.X"/>

internal.xml
#########

<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

external.xml
##########

<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>

( What is a little bit strange though is that nat_map status is not working
[ERR] mod_commands.c:751 nat_map API called while NAT not initialized )

Is the reason why it does not work the port specification in the Contact
field (sngrep output)?
Contact: <sip:gw+sip-trunk.telekom.de at X.X.X.X:5080;transport=tcp;gw=
sip-trunk.telekom.de>

I do not want to setup port forwarding in the firewall. The connection
should remain open via a keep alive signal.

sngrep output
###########

2018/06/30 13:52:37.211946 10.0.200.2:51813 -> 217.0.26.165:5060
REGISTER sip:reg.sip-trunk.telekom.de;transport=tcp SIP/2.0
Via: SIP/2.0/TCP X.X.X.X:5080;rport;branch=z2hG4bK588cZXSvrN1De
Max-Forwards: 70
From: <sip:username at reg.sip-trunk.telekom.de>;tag=9aldsj8n67a
To: <sip:username at reg.sip-trunk.telekom.de>
Call-ID:
CSeq: 111844514 REGISTER
Contact: <sip:gw+sip-trunk.telekom.de at X.X.X.X:5080;transport=tcp;gw=
sip-trunk.telekom.de>
Expires: 3600
User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~38f568d343~32bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, path, replaces
Content-Length: 0

Provider settings:
#############

output proxy: reg.sip-trunk.telekom.de
registrar: sip-trunk.telekom.de

sip_profiles/external/telekom_voip.xml
##############################

<include>

 <gateway name="sip-trunk.telekom.de">
  <param name="username" value=„XXXXXX“/>
  <param name="password" value=„XXXXXX“/>

  <param name="realm" value="sip-trunk.telekom.de"/>
  <param name="proxy" value="reg.sip-trunk.telekom.de"/>
  <param name="register" value="true"/>
  <param name="register-transport" value="tcp"/>
  <param name="expire-seconds" value="3600"/>

 </gateway>
</include>

Firewall settings
#############

# block all traffic in all directions
block log

# loopback interface is not filtered
set skip on lo0

voip_tcp_client_out = "{ssh, ftp, domain, http, https, sip}"
voip_udp_client_out = "{domain, ntp, sip, 3478}"

match out on $wan_if from any nat-to ($wan_if)

pass on $voip_if inet proto tcp from any to port $voip_tcp_client_out
pass on $voip_if inet proto udp from any to port $voip_udp_client_out

pass out on $wan_if inet proto tcp from any to port $tcp_client_out
pass out on $wan_if inet proto udp from any to port $udp_client_out


Thank you for helping :)

Regards,
Paul
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180630/c5989d29/attachment-0001.html>

From paul.muaddib83 at gmail.com  Sat Jun 30 20:57:37 2018
From: paul.muaddib83 at gmail.com (Paul Muaddib)
Date: Sat, 30 Jun 2018 22:57:37 +0200
Subject: [Freeswitch-users] Failed Registration [908]
In-Reply-To: <CAEjNe_FKDDQpCO6QZpgD6rKZawddiLxaBWUMmZVPDRAw48ZwHQ@mail.gmail.com>
References: <CAEjNe_FKDDQpCO6QZpgD6rKZawddiLxaBWUMmZVPDRAw48ZwHQ@mail.gmail.com>
Message-ID: <CAEjNe_HQ=m+3fpz12fR2dw35ZLFBr4F9pG2Xwz1SnKCLtyaMyA@mail.gmail.com>

sofia loglevel all 9

#####################################################################

2018-06-30 22:31:43.746032 [NOTICE] sofia_reg.c:448 Registering
sip-trunk.telekom.de
nua.c:622 nua_register() nua: nua_register: entering
nua_stack.c:529 nua_signal() nua(0xb4e2ec50): sent signal r_register
nua_stack.c:569 nua_stack_signal() nua(0xb4e2ec50): recv signal r_register
nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering
soa.c:280 soa_clone() soa_clone(static::0xb6255520, 0x88b30b0, 0xb4e2ec50)
called
soa.c:403 soa_set_params() soa_set_params(static::0xb5923410, ...) called
soa.c:403 soa_set_params() soa_set_params(static::0xb5923410, ...) called
nua_dialog.c:338 nua_dialog_usage_add() nua(0xb4e2ec50): adding register
usage
nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0xb5906508)
nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip
sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, SRV, "_sip._
tcp.reg.sip-trunk.telekom.de.") called
sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, SRV, "_sip._
tcp.reg.sip-trunk.telekom.de.") returned 3 entries
nta.c:10598 outgoing_query_srv() nta: for "reg.sip-trunk.telekom.de" query
"_sip._tcp.reg.sip-trunk.telekom.de" SRV (cached)
nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de
IN SRV 10 5  5060 d-ipr-a02.sip-trunk.telekom.de. (tcp)
nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de
IN SRV 11 5  5060 b-ipr-a01.sip-trunk.telekom.de. (tcp)
nta.c:10694 outgoing_answer_srv() nta: _sip._tcp.reg.sip-trunk.telekom.de
IN SRV 11 5  5060 b-ipr-a02.sip-trunk.telekom.de. (tcp)
sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, "
d-ipr-a02.sip-trunk.telekom.de.") called
sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, A, "
d-ipr-a02.sip-trunk.telekom.de.") returned 1 entries
nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query "
d-ipr-a02.sip-trunk.telekom.de." A (cached)
nta.c:10856 outgoing_answer_a() nta: d-ipr-a02.sip-trunk.telekom.de. IN A
217.0.26.133
tport.c:4588 tport_by_name() tport(0xb625dc58): found 0x880d978 by name tcp/
217.0.26.133:5060
tport.c:3257 tport_tsend() tport_tsend(0x880d978) tpn = tcp/
217.0.26.133:5060
tport.c:3594 tport_vsend() tport_vsend(0x880d978): 677 bytes of 677 to tcp/
217.0.26.133:5060
tport.c:3492 tport_send_msg() tport_vsend returned 677
tport.c:2296 tport_set_secondary_timer() tport(0x880d978): reset timer
nta.c:8304 outgoing_send() nta: sent REGISTER (124859287) to tcp/
217.0.26.133:5060
tport.c:4160 tport_pend() tport_pend(0x880d978): pending 0xb6263a08 for tcp/
217.0.26.133:5060 (already 1)
nta.c:8982 outgoing_timer_bf() nta: timer F fired, timeout REGISTER
(124859241)
nta.c:9035 outgoing_timeout() nta(0xb6264318): try next after timeout
nta.c:10200 outgoing_graylist() nta: graylisting
d-ipr-a02.sip-trunk.telekom.de.:5060;transport=tcp
nta.c:10227 outgoing_graylist() nta: reduced priority of 1 _sip._
tcp.reg.sip-trunk.telekom.de SRV records (increase value to 11)
sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, "
reg.sip-trunk.telekom.de.") called
nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query "
reg.sip-trunk.telekom.de" A
sres.c:968 sres_query() sres_query(0xb624a8b8, 0xb6264318, A, "
reg.sip-trunk.telekom.de") called
sres.c:2730 sres_send_dns_query() sres_send_dns_query(0xb624a8b8,
0x88653d0) called
sres.c:2819 sres_send_dns_query() sres_send_dns_query(0xb624a8b8,
0x88653d0) id=12982 A reg.sip-trunk.telekom.de (to [217.237.148.70]:53)
nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/2 tout,
0/0 term, 0/2 free
nta.c:1296 agent_timer() nta: timer set next to 29536 ms
sres.c:3467 sres_resolver_receive() sres_resolver_receive(0xb624a8b8, 31)
called
sres.c:3781 sres_create_record() AUTHORITY RR received sip-trunk.telekom.de.
SOA IN 151 rdlen=64
sres.c:3572 sres_log_response() sres_resolver_receive(0xb624a8b8,
0x88653d0) id=12982 (from [217.237.148.70]:53)
sres.c:2987 sres_query_report_error() sres(q=0x88653d0): reporting error
RECORD_ERR for A reg.sip-trunk.telekom.de
tport.c:4222 tport_release() tport_release(0x880d978): 0xb623b2e8 by
0xb6264318 with (nil)
nta.c:8722 outgoing_free() nta: outgoing_free(0xb6264318)
nta.c:8982 outgoing_timer_bf() nta: timer F fired, timeout REGISTER
(124859287)
nta.c:9035 outgoing_timeout() nta(0xb628fb30): try next after timeout
nta.c:10200 outgoing_graylist() nta: graylisting
d-ipr-a02.sip-trunk.telekom.de.:5060;transport=tcp
nta.c:10227 outgoing_graylist() nta: reduced priority of 1 _sip._
tcp.reg.sip-trunk.telekom.de SRV records (increase value to 12)
sres_cache.c:272 sres_cache_get() sres_cache_get(0x88aa8e0, A, "
b-ipr-a01.sip-trunk.telekom.de.") called
sres_cache.c:318 sres_cache_get() sres_cache_get(0x88aa8e0, A, "
b-ipr-a01.sip-trunk.telekom.de.") returned 1 entries
nta.c:10803 outgoing_query_a() nta: for "reg.sip-trunk.telekom.de" query "
b-ipr-a01.sip-trunk.telekom.de." A (cached)
nta.c:10856 outgoing_answer_a() nta: b-ipr-a01.sip-trunk.telekom.de. IN A
217.0.26.163
tport.c:4588 tport_by_name() tport(0xb625dc58): found 0x87fa8a8 by name tcp/
217.0.26.163:5060
tport.c:4222 tport_release() tport_release(0x880d978): 0xb6263a08 by
0xb628fb30 with (nil)
tport.c:2296 tport_set_secondary_timer() tport(0x880d978): reset timer
tport.c:3257 tport_tsend() tport_tsend(0x87fa8a8) tpn = tcp/
217.0.26.163:5060
tport.c:3594 tport_vsend() tport_vsend(0x87fa8a8): 677 bytes of 677 to tcp/
217.0.26.163:5060
tport.c:3492 tport_send_msg() tport_vsend returned 677
tport.c:2296 tport_set_secondary_timer() tport(0x87fa8a8): reset timer
nta.c:8304 outgoing_send() nta: sent REGISTER (124859287) to tcp/
217.0.26.163:5060
tport.c:4160 tport_pend() tport_pend(0x87fa8a8): pending 0xb6263a08 for tcp/
217.0.26.163:5060 (already 0)
nta.c:8929 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout,
0/0 term, 0/1 free
nta.c:1296 agent_timer() nta: timer set next to 32000 ms
2018-06-30 22:32:44.083197 [WARNING] sofia_reg.c:484 Timeout Registering
sip-trunk.telekom.de
nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering
nua_stack.c:569 nua_stack_signal() nua(0xb4e2ec50): recv signal r_destroy
nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0xb4e2ec50): removing
register usage
nta.c:4470 nta_leg_destroy() nta_leg_destroy(0xb5906508)
soa.c:356 soa_destroy() soa_destroy(static::0xb5923410) called
nua_stack.c:529 nua_signal() nua(0xb4e2ec50): sent signal r_destroy
2018-06-30 22:32:45.123287 [WARNING] sofia_reg.c:505 sip-trunk.telekom.de
Failed Registration [908], setting retry to 30 seconds.

#############################################################################

2018-06-30 14:18 GMT+02:00 Paul Muaddib <paul.muaddib83 at gmail.com>:

> Hi,
>
> can someone please look at it and tell my what I am doing wrong?
>
> NOTICE] sofia_reg.c:448 Registering sip-trunk.telekom.de
> [WARNING] sofia_reg.c:484 Timeout Registering sip-trunk.telekom.de
> [WARNING] sofia_reg.c:505 sip-trunk.telekom.de Failed Registration [908],
> setting retry to 30 seconds.
>
> My setup:
> ########
>
> I have a static IP address for my router. My router does not have UPnP or
> NAT-PMP.
> Phones and Freeswitch server are on the same local network. The local
> network is behind NAT
>
> PHONE -> FS -> NAT -> Public Internet
> I only want to connect to the PSTN via my sip gateway provider.
>
> According to this manual I did the following setting:
> https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal
>
> vars.xml
> ######
>
> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=X.X.X.X“/>
> <X-PRE-PROCESS cmd="set" data="external_sip_ip=X.X.X.X"/>
>
> internal.xml
> #########
>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>
> external.xml
> ##########
>
> <param name="ext-sip-ip" value="$${external_sip_ip}"/>
> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>
> ( What is a little bit strange though is that nat_map status is not working
> [ERR] mod_commands.c:751 nat_map API called while NAT not initialized )
>
> Is the reason why it does not work the port specification in the Contact
> field (sngrep output)?
> Contact: <sip:gw+sip-trunk.telekom.de at X.X.X.X:5080;transport=tcp;gw=
> sip-trunk.telekom.de>
>
> I do not want to setup port forwarding in the firewall. The connection
> should remain open via a keep alive signal.
>
> sngrep output
> ###########
>
> 2018/06/30 13:52:37.211946 10.0.200.2:51813 -> 217.0.26.165:5060
> REGISTER sip:reg.sip-trunk.telekom.de;transport=tcp SIP/2.0
> Via: SIP/2.0/TCP X.X.X.X:5080;rport;branch=z2hG4bK588cZXSvrN1De
> Max-Forwards: 70
> From: <sip:username at reg.sip-trunk.telekom.de>;tag=9aldsj8n67a
> To: <sip:username at reg.sip-trunk.telekom.de>
> Call-ID:
> CSeq: 111844514 REGISTER
> Contact: <sip:gw+sip-trunk.telekom.de at X.X.X.X:5080;transport=tcp;gw=
> sip-trunk.telekom.de>
> Expires: 3600
> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~
> 38f568d343~32bit
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY
> Supported: timer, path, replaces
> Content-Length: 0
>
> Provider settings:
> #############
>
> output proxy: reg.sip-trunk.telekom.de
> registrar: sip-trunk.telekom.de
>
> sip_profiles/external/telekom_voip.xml
> ##############################
>
> <include>
>
>  <gateway name="sip-trunk.telekom.de">
>   <param name="username" value=„XXXXXX“/>
>   <param name="password" value=„XXXXXX“/>
>
>   <param name="realm" value="sip-trunk.telekom.de"/>
>   <param name="proxy" value="reg.sip-trunk.telekom.de"/>
>   <param name="register" value="true"/>
>   <param name="register-transport" value="tcp"/>
>   <param name="expire-seconds" value="3600"/>
>
>  </gateway>
> </include>
>
> Firewall settings
> #############
>
> # block all traffic in all directions
> block log
>
> # loopback interface is not filtered
> set skip on lo0
>
> voip_tcp_client_out = "{ssh, ftp, domain, http, https, sip}"
> voip_udp_client_out = "{domain, ntp, sip, 3478}"
>
> match out on $wan_if from any nat-to ($wan_if)
>
> pass on $voip_if inet proto tcp from any to port $voip_tcp_client_out
> pass on $voip_if inet proto udp from any to port $voip_udp_client_out
>
> pass out on $wan_if inet proto tcp from any to port $tcp_client_out
> pass out on $wan_if inet proto udp from any to port $udp_client_out
>
>
> Thank you for helping :)
>
> Regards,
> Paul
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180630/87c56e7b/attachment-0001.html>

From cong.wang.itsherpa at gmail.com  Fri Jun 29 08:18:40 2018
From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=)
Date: Fri, 29 Jun 2018 17:18:40 +0900
Subject: [Freeswitch-users] About Verto
In-Reply-To: <CALXCt0oJBX3PRRfKigU0w9w=Gm3jHEEU8f8BLJ99eVqecYrt1A@mail.gmail.com>
References: <424FFF6C-73B4-4293-A4FE-8F45A24815CD@gmail.com>
 <CALXCt0oJBX3PRRfKigU0w9w=Gm3jHEEU8f8BLJ99eVqecYrt1A@mail.gmail.com>
Message-ID: <695562B1-39BB-4816-B7C3-2EA9D1663389@gmail.com>

Thanks for your reply.

I had added H264 codec in verto.conf.xml, and now it’s able to make call between iOS users.
But it still didn’t work between Android and iOS users, is there any suggestions?

Regards.

> 在 2018年6月29日，00:00，Giovanni Maruzzelli <gmaruzz at gmail.com> 写道：
> 
> On 28 June 2018 at 09:30, 王聡 <cong.wang.itsherpa at gmail.com <mailto:cong.wang.itsherpa at gmail.com>> wrote:
> 
> I had tried to build a WebRTC video call service via mod_verto and Verto Comunicator. 
> Now I could make calls from (to) an Android device with Chrome, but it didn’t work on iOS with Safari.
> Is there any point to be modified to run on Safari?
> 
> be sure to enable H264 codec, both in vars.conf.xml and in verto.conf.xml 
> 
> 
> 
>  
> 
> Regards.
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com <mailto:sales at freeswitch.com>
> https://freeswitch.com <https://freeswitch.com/>
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss <https://freeswitch.com/oss>
> https://freeswitch.org/confluence <https://freeswitch.org/confluence>
> https://cluecon.com <https://cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> https://freeswitch.com <https://freeswitch.com/>
> 
> 
> -- 
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180629/00173b8a/attachment-0001.html>

From mouli123 at gmail.com  Fri Jun 29 11:45:01 2018
From: mouli123 at gmail.com (Chandramouli P)
Date: Fri, 29 Jun 2018 17:15:01 +0530
Subject: [Freeswitch-users] Unable to find modules.conf file after
	installation through RPM file
In-Reply-To: <CALR0W6_eyNJkn4U+0AKR7idLo1gSPmE1SUwodn=dGLcYHOjS4w@mail.gmail.com>
References: <CALR0W6_eyNJkn4U+0AKR7idLo1gSPmE1SUwodn=dGLcYHOjS4w@mail.gmail.com>
Message-ID: <CALR0W69gf92YaWHEyE9nfbm2_-L7mn1+pNQZ-4xni2yrdgxLSg@mail.gmail.com>

Hello Moderator,

I am not receiving any emails from users. Please look in to the issue.

Thank you.

Regards,
CM.

On Thu, Jun 28, 2018 at 7:45 PM, Chandramouli P <mouli123 at gmail.com> wrote:

> Hello,
>
> I installed FreeSwitch through RPM file as below:
>
> sudo rpm -Uvh http://files.freeswitch.org/freeswitch-release-1-6.noarch.
> rpm
>
> sudo yum install freeswitch-config-vanilla sox freeswitch-sounds*
> freeswitch-lang* freeswitch-lua freeswitch-xml-cdr
>
> Everything is working fine. But, when I am trying to enable or disable the
> modules, I could not able to find the "modules.conf" files anywhere. Can
> anybody tell me how to enable/disable the Freeswitch modules, if we install
> the FreeSwitch through RPM?
>
> Thanks in advance.
>
> Regards,
> CM.
>
>
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail> Virus-free.
> www.avg.com
> <http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail>
> <#m_6479024870473712214_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180629/eca6b3fa/attachment-0001.html>

From gmina at connectfirst.com  Fri Jun 29 14:45:04 2018
From: gmina at connectfirst.com (Geoff Mina)
Date: Fri, 29 Jun 2018 08:45:04 -0600
Subject: [Freeswitch-users] FireFox 61 + Freeswitch 1.6.20
In-Reply-To: <CAKzROs2e6RoMeHRn6nOqmJmX8Qs9Tr5jTUyRBvn27HwRV=H7VQ@mail.gmail.com>
References: <CAKzROs2e6RoMeHRn6nOqmJmX8Qs9Tr5jTUyRBvn27HwRV=H7VQ@mail.gmail.com>
Message-ID: <CAKzROs2OMSWK35wpwd6pAZxDA2TuAGXoVcDMqJkr1BQ2uyPqCg@mail.gmail.com>

Not sure if there are any folks on the list that can help, but here are
some logs.

I have poured over them and can't find anything different in either the
console logs or freeswitch logs across the two versions of FireFox.

[One Way Audio - Failure]
https://nofile.io/f/VnF5cFGFseg/freeswitch-log-failure-FF61.log
https://nofile.io/f/yjT6OCfAP7w/console-log-failure-FF61.txt

[Two Way Audio - Success]
https://nofile.io/f/HVXzCmE1VVx/freeswitch-log-success-FF60.log
https://nofile.io/f/GzyJrSjs7aS/console-log-success-FF60.txt

The call flow we have in this example is that a normal SIP over UDP call
comes into FreeSwitch using G.711 codec.  Freeswitch invites to the WSS
client (using Opus codec) and bridges the media.

In the Freeswitch logs, you will see the ingress INVITE as well as the
egress INVITE to the browser.  On the "failure" calls - there is no audible
media being delivered, but we are seeing RTP packets using tcpdump, so we
don't have a NAT/Firewall issue or anything.



On Thu, Jun 28, 2018 at 3:02 PM Geoff Mina <gmina at connectfirst.com> wrote:

> Is anyone else having issues with WebRTC and the latest release of
> FireFox?  We are showing one way media  - with media flowing properly TO
> the browser, but no media coming back from the browser.
>
> Everything in the logs looks OK - no errors.  We are using SIP.js as the
> library of choice for signaling.
>
> We are showing SRTP packets flowing bi-directionally (using tcpdump), but
> since they are DTLS encrypted, I am unable to see exactly what is in each
> packet to determine if there is media being sent from the browser or they
> are just empty packets.
>
> Any ideas and/or other folks seeing the same issue - would love to hear
> your thoughts.
>
> Thanks,
> Geoff
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20180629/a4c966f1/attachment-0001.html>

From zen at freedbms.net  Sat Jun 30 11:59:28 2018
From: zen at freedbms.net (Zenaan Harkness)
Date: Sat, 30 Jun 2018 21:59:28 +1000
Subject: [Freeswitch-users] Debian 9 and Freeswitch 1.8
In-Reply-To: <4B263F26-BC14-43AD-8869-DA0F8EE6E022@jerris.com>
References: <5ed9762c-0735-0135-682b-d1458daf891e@williamcollsassoc.ca>
 <4B263F26-BC14-43AD-8869-DA0F8EE6E022@jerris.com>
Message-ID: <20180630115928.r5blmw4t7kdetmwt@eye.freedbms.net>

I too am interested in compiling fs for debian 9;

>From here:

 https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie

running this command (one line):

 wget -O - https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add -

ends with this error:

 Cannot write to ‘-’ (Broken pipe).


but running this command (one line):

 wget -O - https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | sudo apt-key add -

seems ok,

BUT, then running this command:

 git clone https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch.gits

produces this error:

 Cloning into 'freeswitch.gits'...
 fatal: unable to access
 'https://freeswitch.org/stash/scm/fs/freeswitch.git/':
 gnutls_handshake() failed: Public key signature verification has
 failed.


So there is some tls problem which "should" be simple to solve... for
someone who knows how :)
So this would be my first question - how to fix this error.

My second question is what environment do folks normally use for
development? E.g. debian sid/unstable?

(my apologies if this should go to the dev list, I have applied to
join, a few days ago, just waiting to be approved...)



On Mon, Feb 05, 2018 at 11:13:00AM -0500, Michael Jerris wrote:
> There are some known issues with Debian 9 and how its handled the
> openssl upgrade that we are still working through to confirm what
> exactly needs to get done to fix it.  The problem is, a bunch of
> system libs were not ready for the new openssl when deb9 was locked
> down, so they took a half way there approach and its caused issues
> where multiple versions of openssl end up in the same process, and
> we expect that this may cause some issues.  I expect this to take a
> little while to verify completely.
> 
> 
> > On Feb 3, 2018, at 5:40 PM, William Colls <william at williamcollsassoc.ca> wrote:
> > 
> > 
> > Is there an expected date when Freeswitch 1.8/Debian 9 will become the preferred production configuration?
> 
> 


From zen at freedbms.net  Sat Jun 30 12:03:56 2018
From: zen at freedbms.net (Zenaan Harkness)
Date: Sat, 30 Jun 2018 22:03:56 +1000
Subject: [Freeswitch-users] Debian repos
In-Reply-To: <68A1DF24-5BBE-4D6C-95D6-24E99A631B82@gmail.com>
References: <feef8955-4106-41a6-9a68-1939190f7c63@Spark>
 <68A1DF24-5BBE-4D6C-95D6-24E99A631B82@gmail.com>
Message-ID: <20180630120356.5rotziic4c3j2rlh@eye.freedbms.net>

Is there a suggested or (semi?) official "anonymous git" protocol git
repo - e.g. where the following command would work:

 git clone https://freeswitch.org/stash/scm/fs/freeswitch.git

(like how github works...)?



On Wed, Nov 29, 2017 at 10:13:36AM +0900, 王聡 wrote:
> You can compile FS by youself if you want.
> 
> For FS 1.6.8:
> 
> git clone https://freeswitch.org/stash/scm/fs/freeswitch.git
> cd /usr/src/freeswitch/
> 
> git checkout -b v1.6.8 refs/tags/v1.6.8
> ./bootstrap.sh -j
> Also you can compile any version as you wish.
> 
> > 2017/11/28 22:52、Igor Olhovskiy <igorolhovskiy at gmail.com>のメール:
> > 
> > Hi!
> > 
> > Is there a way to set up different version of FreeSWITCH from repos? 
> > Means for ex I want to use not 1.6.19, but 1.6.8?
> > Or compile is the only way?
> > 
> > Regards, Igor
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20171129/b121cd3d/attachment-0001.html>


From zen at freedbms.net  Sat Jun 30 12:09:52 2018
From: zen at freedbms.net (Zenaan Harkness)
Date: Sat, 30 Jun 2018 22:09:52 +1000
Subject: [Freeswitch-users] Debian repos
In-Reply-To: <20180630120356.5rotziic4c3j2rlh@eye.freedbms.net>
References: <feef8955-4106-41a6-9a68-1939190f7c63@Spark>
 <68A1DF24-5BBE-4D6C-95D6-24E99A631B82@gmail.com>
 <20180630120356.5rotziic4c3j2rlh@eye.freedbms.net>
Message-ID: <20180630120952.re4t4ujro7vi45yq@eye.freedbms.net>

Found this:
https://freeswitch.org/confluence/display/FREESWITCH/Tips+For+Using+Git


On Sat, Jun 30, 2018 at 10:03:56PM +1000, Zenaan Harkness wrote:
> Is there a suggested or (semi?) official "anonymous git" protocol git
> repo - e.g. where the following command would work:
> 
>  git clone https://freeswitch.org/stash/scm/fs/freeswitch.git
> 
> (like how github works...)?
> 
> 
> 
> On Wed, Nov 29, 2017 at 10:13:36AM +0900, 王聡 wrote:
> > You can compile FS by youself if you want.
> > 
> > For FS 1.6.8:
> > 
> > git clone https://freeswitch.org/stash/scm/fs/freeswitch.git
> > cd /usr/src/freeswitch/
> > 
> > git checkout -b v1.6.8 refs/tags/v1.6.8
> > ./bootstrap.sh -j
> > Also you can compile any version as you wish.
> > 
> > > 2017/11/28 22:52、Igor Olhovskiy <igorolhovskiy at gmail.com>のメール:
> > > 
> > > Hi!
> > > 
> > > Is there a way to set up different version of FreeSWITCH from repos? 
> > > Means for ex I want to use not 1.6.19, but 1.6.8?
> > > Or compile is the only way?
> > > 
> > > Regards, Igor
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > > 
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://confluence.freeswitch.org
> > > http://www.cluecon.com
> > > 
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > 
> > -------------- next part --------------
> > An HTML attachment was scrubbed...
> > URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20171129/b121cd3d/attachment-0001.html>


From vma at vallimamod.org  Thu Jun 28 14:16:33 2018
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Thu, 28 Jun 2018 16:16:33 +0200
Subject: [Freeswitch-users] Enterprise/Production Quality?
In-Reply-To: <CAFGRPVqcDfnmGDLk=+CeKwSJTdwUpA9yGfEN_R9D7qJBod-TTQ@mail.gmail.com>
References: <1A170859-F608-498C-AB99-C41E17E5B706@connectfirst.com>
 <AF3066FB-3622-4462-A16D-D35E69B288EE@jerris.com>
 <1527697018.3678744.1390660224.09199347@webmail.messagingengine.com>
 <6EF6FCCE-4ED1-49B2-BC48-2B87099FDC7D@jerris.com>
 <DAAD6B7C-2ED7-4BFE-9E89-E1FEC4F29CD1@stratusvideo.com>
 <1527706840.542281.1390853752.2050A1B1@webmail.messagingengine.com>
 <mailman.1631.1529968857.1211.freeswitch-users@lists.freeswitch.org>
 <CAFGRPVqcDfnmGDLk=+CeKwSJTdwUpA9yGfEN_R9D7qJBod-TTQ@mail.gmail.com>
Message-ID: <8CB94817-A9A8-408E-97D5-2A64EA9A6DBD@vallimamod.org>

Hi,

Not sure if it is related, but when I tried to look at the commits correcting a bug I was interested on (https://freeswitch.org/jira/browse/FS-10904), I got the message "You don't have access to view all related commits. Please contact your administrator." Looks like the reporter got the same message too.

screenshot: https://imgur.com/y29ysyT


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

> On 26 Jun 2018, at 11:19, David Villasmil <david.villasmil.work at gmail.com> wrote:
> 
> Hello all,
> Is this true??? Bug fixes aren't available for the general open source community for EIGHTEEN MONTHS???
> This seems REALLY wrong to me... I'm pretty sure there are better ways of making money from freeSWITCH...
> 
> Now I'm very concerned, and probably a lot of people will, if/when they find out about this...
> 
> David
> 
> 
> On Tue, Jun 26, 2018, 02:57 Matteo via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org> wrote:
> 
> 
> 
> ---------- Forwarded message ----------
> From: Matteo <mbrancaleoni at voismart.it>
> To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Bcc: 
> Date: Wed, 20 Jun 2018 11:50:26 +0200 (CEST)
> Subject: Re: [Freeswitch-users] Enterprise/Production Quality?
> Well I agree too.
> 
> Since the introduction of the FSA (term not cited anywhere on website), we started evaluating a subscription.
> 
> But a lot of things are really, really vague and keeping us away:
> 
> * What is in the FSA? What we get for the money?
> * What about prices?
> * What is the release cycle? When things gets pushed into the OSS version?
> * What is the FSA? please add a blog post to clarify this shift in policy and how things "works" from now on.
> 
> Basically there's no communication about that and everything appears really vague.
> 
> Other OSS projects with similar policy have a very clear release path, what is in, what is not,
> when new things will appear in OSS and so on.
> 
> And then a suggestion:
> Why not offer a "basic" FSA with just access to latest git?
> And not provide dedicated support, but use traditional channels?
> 
> This should be cheaper and interesting for companies that are autonomous into investigating and fixing issues by themselves.
> 
> And about bugfixes... well I agree with others... they should be released because affects something that's already out.
> 
> Not doing that may backfire because the OSS product may be perceive buggy (especially if using latest features which may not be debugged properly yet)
> 
> Keep in mind that from a company perspective, if the things are free you can be loose on how the project is handled, because is free after all.
> But if you want to get payed (which is completely natural and needed to keep things go on), you must be very clear and communicative on what you give for the price.
> 
> Just my 2 cents.
> 
> Matteo
> 
> ----- Il 30-mag-18, alle 21:00, Michael Avers michael at mailworks.org ha scritto:
> 
> > I agree. I think they are doing it wrong. I obviously believe they need to make
> > money and get paid for their work, I have no problem with that at all, and my
> > company has bought quite a few licenses of their commercial modules and we
> > attend Cluecon every other year. So we have no problem paying but we need to
> > know what we are paying FOR.
> > 
> > This kind of vague state of things where you just don't know if bugs are now
> > going to be fixed for everyone or just for the privileged ones is not a good
> > path to go down.
> > 
> > Bug fixes really should be made available to everyone (assuming they are in
> > previously public modules, of course).
> > 
> > The Freeswitch team should focus their efforts on creating commercial modules,
> > ready-made apps and setups, pro versions of older modules, say enhanced
> > mod_callcenter or whatever. Things like that. But to tell someone to get a
> > premium subscription just to get a bug fixed... that's simply wrong.
> > 
> > Just my 2 cents
> > 
> > Mike
> > 
> > 
> > 
> > 
> > On Wed, May 30, 2018, at 11:37 AM, William Simon wrote:
> > 
> > 
> > 
> > I thought stability fixes were going to be included in the open source release
> > whereas new features are covered under FSA release cycle. What I am seeing here
> > is unfortunate. Open source users should just accept instability for the
> > 18-month release cycle?
> > 
> > 
> > 
> > 
> > 
> > On May 30, 2018, at 12:35 PM, Michael Jerris < mike at jerris.com > wrote:
> > 
> > Michael-
> > 
> > This is a specific bug that I know we have fixed. We spent months of work
> > tracking it down, I am very familiar with the issue. This issue is not at all
> > with verto, and is specifically with the sip secure web socket support. We have
> > never recommended the use of sip web socket support for webrtc, we think that
> > verto is typically a better solution, and is more stable.
> > 
> > Mike
> > 
> > 
> > 
> > 
> > 
> > On May 30, 2018, at 12:16 PM, Michael Avers < michael at mailworks.org > wrote:
> > 
> > LOL... The issue he describes can be due to several different reasons. I don't
> > even think we have enough information at this point to determine it let alone
> > narrow it down to one specific bug fix.
> > 
> > Is saying "Oh this is fixed in our paid product" going to be the standard moving
> > forward?
> > 
> > In any case, a response such as this basically tells potential Verto users that
> > they shouldn't bother because it's going to break at 150 concurrent users
> > anyway.
> > 
> > Mike
> > 
> > 
> > On Wed, May 30, 2018, at 8:58 AM, Michael Jerris wrote:
> > 
> > 
> > 
> > Geoff-
> > 
> > I believe this issue is fixed in https://freeswitch.org/jira/browse/FS-10762
> > which is available to FSA customers.
> > 
> > Mike
> > 
> > 
> > 
> > 
> > On May 30, 2018, at 10:45 AM, Geoff Mina < gmina at connectfirst.com > wrote:
> > 
> > Is anyone out there actually using FS successfully in an enterprise environment?
> > 
> > We have deployed a handful of servers in an extremely simple configuration to
> > allow standard SIP infrastructure to communicate with WebRTC clients.
> > 
> > We run ~150 concurrent users per host and we can’t go a week without something
> > in the core of Sofia failing. We have seen hung profiles that simply don’t
> > respond to REGISTER requests (tried every suggested tweak to no avail) as well
> > as seeing FS hang every call in a RINGING state without actually ringing the
> > end client.
> > 
> > These seem like pretty fundamental components we are struggling with. Our use
> > case seems quite simple - yet the software (1.6.19) seems like it has never
> > even seen a production deployment.
> > 
> > Anyone out there have a drastically different experience?
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > 
> > 
> > “The information transmitted is intended only for the person or entity to which
> > it is addressed and may contain proprietary, business-confidential and/or
> > privileged material. If you are not the intended recipient of this message you
> > are hereby notified that any use, review, retransmission, dissemination,
> > distribution, reproduction or any action taken in reliance upon this message is
> > prohibited. If you received this in error, please contact the sender and delete
> > the material from any computer.”
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Services
> > sales at freeswitch.com
> > https://freeswitch.com
> > 
> > Official FreeSWITCH Sites
> > https://freeswitch.com/oss
> > https://freeswitch.org/confluence
> > https://cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > https://freeswitch.com
> 
> 
> 
> 
> ---------- Forwarded message ----------
> From: Matteo via FreeSWITCH-users <freeswitch-users at lists.freeswitch.org>
> To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
> Cc: 
> Bcc: 
> Date: Mon, 25 Jun 2018 17:57:02 -0700 (PDT)
> Subject: Re: [Freeswitch-users] Enterprise/Production Quality?
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
> _________________________________________________________________________
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com