From gmaruzz at gmail.com  Sat Jul  1 00:42:18 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 1 Jul 2017 02:42:18 +0200
Subject: [Freeswitch-users] Audio issue in Amazon EC2 server
In-Reply-To: <CAGqqW99U6MCCJdZAQD1hRMJSLyn=B83-TUYwmpZ9atr7PKy81w@mail.gmail.com>
References: <CAGqqW980iZWcwrFLv4KcRTZkGscoRHop_rv2+rmJSik_NWHXqg@mail.gmail.com>
 <CALXCt0qJNCZb3ygVcAkyoVFwnurzoZ8PBriuO6gxK+EbsZOZZA@mail.gmail.com>
 <CAGqqW99jm4PEiCMBORUXraNkUF_38Hq9hq+qg_oZPq9x=i=muQ@mail.gmail.com>
 <CALXCt0qDGrO6uhNReQG_6vTCJuUbC235La-bomF2VjuuJCLg_g@mail.gmail.com>
 <CAGqqW9-WsJOoqP_obYX5BGPzgXBL-EYA7cgozVE0iVvAs=07aw@mail.gmail.com>
 <CALXCt0q9YFJXr=aH7DMuNP27j6PNFQW9mArUd+bPcCVUUF0f5g@mail.gmail.com>
 <CAGqqW99U6MCCJdZAQD1hRMJSLyn=B83-TUYwmpZ9atr7PKy81w@mail.gmail.com>
Message-ID: <CALXCt0qrF09s-U026FX4tfp35_z0w_W9Os1iPERF5_C3E2eSEQ@mail.gmail.com>

On 1 July 2017 at 01:33, Ketan Kothari <kkothari157 at gmail.com> wrote:

> Hello Giovanni,
>
> Here is my "sofia status profile external"
>
> freeswitch@> sofia status profile external
> ============================================================
> =====================================
> Name                 external
> Domain Name          N/A
> Auto-NAT             false
> DBName               sofia_reg_external
> Pres Hosts           192.168.1.101
> Dialplan             XML
> Context              default
> Challenge Realm      auto_from
> RTP-IP               192.168.1.101
> Ext-RTP-IP          128.22.54.32
> SIP-IP               192.168.1.101
> Ext-SIP-IP           128.22.54.32
> URL                  sip:mod_sofia at 128.22.54.32:5060
> BIND-URL             sip:mod_sofia at 128.22.54.32:5060;maddr=192.168.1.101;
> transport=udp,tcp
> HOLD-MUSIC           local_stream://moh
> OUTBOUND-PROXY       N/A
> CODECS IN            PCMA,PCMU
> CODECS OUT           PCMA,PCMU
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             false
> PROXY-MEDIA          false
> ZRTP-PASSTHRU        false
> AGGRESSIVENAT        true
> CALLS-IN             2
> FAILED-CALLS-IN      0
> CALLS-OUT            2
> FAILED-CALLS-OUT     0
> REGISTRATIONS        0
>
> ===============
>
>
>
seems you have altered the default XML configuration. Can you reinstall the
default configuration, without making any change, then only edit the two
sip profiles for the external addresses?

Also, how is your gateway definition?

What is the result of "sofia status"?






>
> On Sat, Jul 1, 2017 at 4:23 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>>
>>
>> On 1 July 2017 at 00:28, Ketan Kothari <kkothari157 at gmail.com> wrote:
>>
>>> Hello Giovanni,
>>>
>>> Yeah i'm sure. I have checked in sip-profile ext-rtp-ip and ext-sip-ip
>>> both are set to public address but still while calling freeswitch sending
>>> local ip to gateway side.
>>>
>>
>> you edited "external.xml" as well? And restarted?
>>
>> can you post here the result of:
>>
>> "sofia status"
>>
>> and
>>
>> "sofia status profile external"
>>
>> from fs_cli?
>>
>>  -giovanni
>>
>>
>>>
>>> On Sat, Jul 1, 2017 at 3:50 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>>
>>>>
>>>> On 30 June 2017 at 23:53, Ketan Kothari <kkothari157 at gmail.com> wrote:
>>>>
>>>>> Hello Giovanni,
>>>>>
>>>>> I have already set same in sip-profile but still getting same issue
>>>>> which mention.
>>>>>
>>>>
>>>> not possible.
>>>> check again, you must edit "/usr/local/freeswitch/conf/sip_profiles/internal.xml"
>>>> and "external.xml" files, then restart freeswitch
>>>>
>>>>
>>>> Also, is a very strange private address 192.168.1.101 in AWS... Are you
>>>> sure?
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Jun 29, 2017 at 6:46 PM, Giovanni Maruzzelli <
>>>>> gmaruzz at gmail.com> wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On 29 June 2017 at 15:10, Ketan Kothari <kkothari157 at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Hi All,
>>>>>>>
>>>>>>> I have installed freeswitch on Amazon EC2 server and i followed *https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2
>>>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2>*
>>>>>>>
>>>>>>> Once we tried to call outbound call getting *no audio in both side*
>>>>>>> also FreeSWITCH passing my server local IP as RTP-IP  to gateway side,
>>>>>>>
>>>>>>
>>>>>>
>>>>>> set your ext-rtp-ip and ext-sip-ip in all your sip_profiles to the
>>>>>> EXTERNAL (public) ip address of AWS instance
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From joelists at tm.net.uk  Sat Jul  1 14:09:26 2017
From: joelists at tm.net.uk (Jospeh Waite)
Date: Sat, 1 Jul 2017 15:09:26 +0100
Subject: [Freeswitch-users] mod_rad_auth config file not found
Message-ID: <F427185C-3060-4169-8F7E-4FF7A3036136@tm.net.uk>

Hi Guys

Following some pointers from the mailing list a couple of weeks back I am attempting to configure and use mod_rad_auth to do radius authentication from the dial plan.

I have compiled and installed the module and created the config file as per the confluence page, however the confluence page does not tell me where the config file should be located.

I have tried in /usr/local/freeswitch/conf  and also /usr/local/freeswitch/conf/autoload_configs 

However I get the following error

2017-07-01 15:08:27.024695 [ERR] mod_rad_auth.c:762 open of rad_auth.conf failed

I have even tried looking at the source of the module but can’t seem to get an idea of where it should be

Any pointers would be greatly appreciated.

Regards
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From s.safarov at gmail.com  Sat Jul  1 17:48:08 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 01 Jul 2017 17:48:08 +0000
Subject: [Freeswitch-users] mod_rad_auth config file not found
In-Reply-To: <F427185C-3060-4169-8F7E-4FF7A3036136@tm.net.uk>
References: <F427185C-3060-4169-8F7E-4FF7A3036136@tm.net.uk>
Message-ID: <CAHtxdDdgNEDoT1LWZZFhaFAGcvkWXvUcH7Y8gfgBuFWLXHkciQ@mail.gmail.com>

really looked /etc/freeswitch/autoload_configs/rad_auth.conf.xml

[root at node1.sbc ~]# head /etc/freeswitch/autoload_configs/rad_auth.conf.xml

<configuration name="rad_auth.conf" description="radius authentification
module">

 <settings>

    <!-- backward compatibility to allow radiusclient config file instead
of an embedded config -->

    <!--  <param name="radius_config"
value="/usr/local/etc/radiusclient/radiusclient.conf"/>   -->

 </settings>


 <client>

   <param name="authserver" value="bill.rcsnet.ru:1814:x7h6t4g"/>

   <param name="dictionary" value="/etc/radiusclient/dictionary.all"/>

   <param name="seqfile" value="/var/run/radius.seq"/>



сб, 1 июл. 2017 г. в 17:14, Jospeh Waite <joelists at tm.net.uk>:

> Hi Guys
>
> Following some pointers from the mailing list a couple of weeks back I am
> attempting to configure and use mod_rad_auth to do radius authentication
> from the dial plan.
>
> I have compiled and installed the module and created the config file as
> per the confluence page, however the confluence page does not tell me where
> the config file should be located.
>
> I have tried in /usr/local/freeswitch/conf  and also
> /usr/local/freeswitch/conf/autoload_configs
>
> However I get the following error
>
> 2017-07-01 15:08:27.024695 [ERR] mod_rad_auth.c:762 open of rad_auth.conf
> failed
>
> I have even tried looking at the source of the module but can’t seem to
> get an idea of where it should be
>
> Any pointers would be greatly appreciated.
>
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From michael at mailworks.org  Sat Jul  1 22:51:46 2017
From: michael at mailworks.org (Michael Avers)
Date: Sat, 01 Jul 2017 15:51:46 -0700
Subject: [Freeswitch-users] Originate and then transfer on answer but not on
	VM
In-Reply-To: <1494961415.1484944.978717552.3EE447BC@webmail.messagingengine.com>
References: <1494961415.1484944.978717552.3EE447BC@webmail.messagingengine.com>
Message-ID: <1498949506.4142771.1027676440.6E4144D3@webmail.messagingengine.com>

Hello,

What would be a good approach if I want to originate a call to a local user, then once they pick up - bridge to an external number (or anywhere else). I tried several ways, but they all end up bridging even if the user's VM picked up the call.

For example I tried:

originate with execute_on_answer='transfer NEXT_STEP XML context'
originate statement followed by &bridge or even just a the next step extension as the destination

These work, but I want to avoid proceeding to the next step if the call was picked up by the user's VM.

I also tried parking it but then how would I know if it ended up in VM?

Thanks
-Mike


From michael at mailworks.org  Sat Jul  1 23:00:41 2017
From: michael at mailworks.org (Michael Avers)
Date: Sat, 01 Jul 2017 16:00:41 -0700
Subject: [Freeswitch-users] Originate and then transfer on answer but
 not on VM
In-Reply-To: <1498949506.4142771.1027676440.6E4144D3@webmail.messagingengine.com>
References: <1494961415.1484944.978717552.3EE447BC@webmail.messagingengine.com>
 <1498949506.4142771.1027676440.6E4144D3@webmail.messagingengine.com>
Message-ID: <1498950041.641472.1027681328.387BC186@webmail.messagingengine.com>

My bad, at some point in my tests today I switched to using sofia/external/extension at domain vs. user/extension at domain. Using the latter, a proper USER_BUSY response will not proceed to the next step, as desired.

Enjoy your weekend everyone!
Mike


On Sat, Jul 1, 2017, at 03:51 PM, Michael Avers wrote:
> Hello,
> 
> What would be a good approach if I want to originate a call to a local user, then once they pick up - bridge to an external number (or anywhere else). I tried several ways, but they all end up bridging even if the user's VM picked up the call.
> 
> For example I tried:
> 
> originate with execute_on_answer='transfer NEXT_STEP XML context'
> originate statement followed by &bridge or even just a the next step extension as the destination
> 
> These work, but I want to avoid proceeding to the next step if the call was picked up by the user's VM.
> 
> I also tried parking it but then how would I know if it ended up in VM?
> 
> Thanks
> -Mike
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gregor at infomedia.si  Sun Jul  2 00:44:45 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 2 Jul 2017 02:44:45 +0200
Subject: [Freeswitch-users] Music on hold
Message-ID: <CAC7ttFEA92XB4A35ETxndS6xswKCdOLcwKLi9yLZruZC1576mQ@mail.gmail.com>

Need some advice.

I have multi tenant setup, driven with xml_curl. What is best approach to
make music on hold configurable for every customer? I mean from resource
point of view. Music files should be read from http, not file system.

Thank you, Gregor
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From siju.irs at gmail.com  Sun Jul  2 05:25:16 2017
From: siju.irs at gmail.com (Siju Nair)
Date: Sun, 2 Jul 2017 10:55:16 +0530
Subject: [Freeswitch-users] Load testing
Message-ID: <4B572C48-A705-4583-90BB-7F02A10731DD@gmail.com>

Hi team,

Any approximate figure regarding FS Server load balance? 

How much concurrent calls server can handle also minimum hardware configuration that required to handle 500 concurrent calls at a time ? 

Sent from my iPhone

> On 05-Apr-2017, at 9:18 AM, Siju Nair <siju.irs at gmail.com> wrote:
> 
> Hi Micheal 
> 
> Thank you very much for ur reply. 
> Can I know what all info u are looking for I am using  FS version 1.2.23 
> 
> Am running call center module in FS with 4 agents taking calls .. for 3 days it worked fine but then it get hanged up and I was unable to log in to server .. 
> 
> I could see above screenshot logs . 
> 
> Sent from my iPhone
> 
>> On 05-Apr-2017, at 1:29 AM, Michael Jerris <mike at jerris.com> wrote:
>> 
>> We’ve looked at the provided information.  This information is not enough to provide any feedback on this issue.
>> 
>>> On Apr 3, 2017, at 11:34 PM, Siju Nair <siju.irs at gmail.com> wrote:
>>> 
>>> Hi team 
>>> Any advice on below issue, please!!
>>> 
>>> Sent from my iPhone
>>> 
>>>> On 03-Apr-2017, at 1:00 PM, Siju Nair <siju.irs at gmail.com> wrote:
>>>> 
>>>> Hi team,
>>>> Please help me on below issue !!
>>>> 
>>>> Sent from my iPhone
>>>> 
>>>>> On 09-Mar-2017, at 4:44 PM, Siju Nair <siju.irs at gmail.com> wrote:
>>>>> 
>>>>> Hi team,
>>>>> 
>>>>> Please help on below issue ! Also my CPU avg is 42 on calls ! 
>>>>> 
>>>>> Sent from my iPhone
>>>>> 
>>>>>> On 08-Mar-2017, at 7:53 PM, Siju Nair <siju.irs at gmail.com> wrote:
>>>>>> 
>>>>>> Please check the attached screen shot for more clarification
>>>>>> 
>>>>>>> On Wed, Mar 8, 2017 at 7:48 PM, Siju Nair <siju.irs at gmail.com> wrote:
>>>>>>> Hi team,
>>>>>>> 
>>>>>>> My FS server got hung, under server logs due freeswitch server got
>>>>>>> hung up. Below is the log i found..
>>>>>>> 
>>>>>>> echo 0 > /proc/sys/kernal/hung_task_timeout_secs"disables this message
>>>>>>> info: task freeswitch: 3034 blocked for more than 120 secs.
>>>>>>> 
>>>>>>> and then
>>>>>>> freeswitch > waiting for background process to be ready.
>>>>>>> 
>>>>>>> Please help me on the same.
>>>>>>> 
>>>>>>> Sent from my iPhone
>>>>>> 
>>>>>> <ccmmydqld_hang.PNG>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
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From s.safarov at gmail.com  Sun Jul  2 14:12:14 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 02 Jul 2017 14:12:14 +0000
Subject: [Freeswitch-users] FreeSwitch base docker container
Message-ID: <CAHtxdDdEixpP7uV=0mRYbuQ-XSw6-Ba5E_7JrZRyJgmkXQ1siQ@mail.gmail.com>

Hello
I created minimized version of could you look pull request 1322
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1322>
or
test docker container safarov/freeswitch/
<https://hub.docker.com/r/safarov/freeswitch/>
I ask FreeSwitch core team to consider the possibility of making it an
official container on the Docker Hub

Sergey afarov
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From joelists at tm.net.uk  Sun Jul  2 16:30:24 2017
From: joelists at tm.net.uk (Jospeh Waite)
Date: Sun, 2 Jul 2017 17:30:24 +0100
Subject: [Freeswitch-users] mod_rad_auth dial plan condition to match radius
	result
Message-ID: <7CAC6D12-1BD1-42B4-9D5A-A5263BE2127B@tm.net.uk>

Hi Guys

Nearly there with mod_rad_auth, I have it being called in my dial plan and its sending the Radisu auth and working as expected.

I now want to put a condition so that in the event of Radius Auth failing the dial plan will end and not allow the call to be made.

I have put the following in the dial plan

        <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>

  <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
  <action application="log" data="INFO  credit_time=${credit_time}"/>
  <action application="log" data="INFO  return_code=${return_code}"/>
   <condition field="${return_code}" expression="^16$">

            <action application="set" data="execute_on_answer=sched_hangup +${credit_time} alloted_timeout" />
            <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060" />
            <action application="hangup" data="${bridge_hangup_cause}"/>
        </condition>


And I check the freeswitch log and see

EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO  AUTH_RESULT=OK)
2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  AUTH_RESULT=OK
EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO  credit_time=7199)
2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  credit_time=7199
EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO  return_code=16)
2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  return_code=16


But it fails to execute the bit within the condition.


Any Ideas of how to make this work?

Regards
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From vma at vallimamod.org  Sun Jul  2 17:28:36 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Sun, 2 Jul 2017 19:28:36 +0200
Subject: [Freeswitch-users] mod_rad_auth dial plan condition to match
 radius result
In-Reply-To: <7CAC6D12-1BD1-42B4-9D5A-A5263BE2127B@tm.net.uk>
References: <7CAC6D12-1BD1-42B4-9D5A-A5263BE2127B@tm.net.uk>
Message-ID: <A465D9D4-196B-42E2-AE99-550A48B8A44D@vallimamod.org>

Hi,

Try adding inline="true" to the action returning the result you want to check.
There is a nice explanation on the wiki on how the dialplan works and why this is necessary if you want to use the result of an action in a subsequent condition.

Best regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.

> On 2 Jul 2017, at 18:30, Jospeh Waite <joelists at tm.net.uk> wrote:
> 
> Hi Guys
> 
> Nearly there with mod_rad_auth, I have it being called in my dial plan and its sending the Radisu auth and working as expected.
> 
> I now want to put a condition so that in the event of Radius Auth failing the dial plan will end and not allow the call to be made.
> 
> I have put the following in the dial plan
> 
>         <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
> 
>   <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>   <action application="log" data="INFO  credit_time=${credit_time}"/>
>   <action application="log" data="INFO  return_code=${return_code}"/>
>    <condition field="${return_code}" expression="^16$">
> 
>             <action application="set" data="execute_on_answer=sched_hangup +${credit_time} alloted_timeout" />
>             <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060" />
>             <action application="hangup" data="${bridge_hangup_cause}"/>
>         </condition>
> 
> 
> And I check the freeswitch log and see
> 
> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  AUTH_RESULT=OK)
> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  AUTH_RESULT=OK
> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  credit_time=7199)
> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  credit_time=7199
> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  return_code=16)
> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  return_code=16
> 
> 
> But it fails to execute the bit within the condition.
> 
> 
> Any Ideas of how to make this work?
> 
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From findmeinwland at gmail.com  Sun Jul  2 17:59:35 2017
From: findmeinwland at gmail.com (Artur Mega)
Date: Sun, 2 Jul 2017 22:59:35 +0500
Subject: [Freeswitch-users] 000298C appends to the destination number
Message-ID: <CAOLEQ7fgtpu6jHjXcc0jZ1nCWYFH+heRXsOdoPAzscLrHzM-9g@mail.gmail.com>

Hello, anybody knows why destination number starts with 000298C (in several
cases)? I think it is encoded plus sign.

-- 
​Regards, Arthur​
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From joelists at tm.net.uk  Sun Jul  2 21:45:20 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Sun, 2 Jul 2017 22:45:20 +0100
Subject: [Freeswitch-users] mod_rad_auth dial plan condition to match
	radius result
In-Reply-To: <A465D9D4-196B-42E2-AE99-550A48B8A44D@vallimamod.org>
References: <7CAC6D12-1BD1-42B4-9D5A-A5263BE2127B@tm.net.uk>
 <A465D9D4-196B-42E2-AE99-550A48B8A44D@vallimamod.org>
Message-ID: <B19941B2-6461-4E80-8A57-E00260877102@tm.net.uk>

If I put the inline=“true” in the action that calls auth_function then it fails to return any results in the log output.


> On 2 Jul 2017, at 18:28, Vallimamod Abdullah <vma at vallimamod.org> wrote:
> 
> Hi,
> 
> Try adding inline="true" to the action returning the result you want to check.
> There is a nice explanation on the wiki on how the dialplan works and why this is necessary if you want to use the result of an action in a subsequent condition.
> 
> Best regards,
> -- 
> Vallimamod Abdullah
> SIP Solutions
> vma at sipsolutions.fr <mailto:vma at sipsolutions.fr>.
> 
>> On 2 Jul 2017, at 18:30, Jospeh Waite <joelists at tm.net.uk <mailto:joelists at tm.net.uk>> wrote:
>> 
>> Hi Guys
>> 
>> Nearly there with mod_rad_auth, I have it being called in my dial plan and its sending the Radisu auth and working as expected.
>> 
>> I now want to put a condition so that in the event of Radius Auth failing the dial plan will end and not allow the call to be made.
>> 
>> I have put the following in the dial plan
>> 
>>         <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>> 
>>   <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>>   <action application="log" data="INFO  credit_time=${credit_time}"/>
>>   <action application="log" data="INFO  return_code=${return_code}"/>
>>    <condition field="${return_code}" expression="^16$">
>> 
>>             <action application="set" data="execute_on_answer=sched_hangup +${credit_time} alloted_timeout" />
>>             <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060 <sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060>" />
>>             <action application="hangup" data="${bridge_hangup_cause}"/>
>>         </condition>
>> 
>> 
>> And I check the freeswitch log and see
>> 
>> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  AUTH_RESULT=OK)
>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  AUTH_RESULT=OK
>> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  credit_time=7199)
>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  credit_time=7199
>> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  return_code=16)
>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  return_code=16
>> 
>> 
>> But it fails to execute the bit within the condition.
>> 
>> 
>> Any Ideas of how to make this work?
>> 
>> Regards
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nathan.port at gmail.com  Mon Jul  3 03:24:35 2017
From: nathan.port at gmail.com (Nate)
Date: Mon, 3 Jul 2017 15:24:35 +1200
Subject: [Freeswitch-users] Dialplan: Outbound route:
 destination_number: regex question
Message-ID: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>

Hi Roy, many thanks for responding.

I am simply trying to finishing configuring the server to make/receive
calls from the PSTN.

My reference to the regex is from trying to configure an outbound route.

Can anyone offer any advice on a regex that will work for me to reach any
of the following phone numbers:

International: +64 22 333 4444 <+64%2022%20333%204444>
non-local: 022 333 4444
local: 333 4444

>From the number assigned to my account? +64 9 xxx yyyy

Many many thanks!

Nate.

On Sat, Jul 1, 2017 at 9:17 AM, <royj at yandex.ru> wrote:

> Not quite clear. What are you trying to achieve?
>
> 30.06.2017, 11:11, "Nate" <nathan.port at gmail.com>:
>
> Good day/evening everyone,
>
> Apologies for not being able to figure this out on my own. I've been
> searching and trying for several days to get inbound/outbound working but
> have yet to see success.
>
> At this stage I need help determining a proper regex expression for
> handling New Zealand phone numbers.
>
> For instance, there are three different ways of expressing numbers here in
> NZ:
>
> International: +64 22 333 4444 <+64%2022%20333%204444>
> non-local: 022 333 4444
> local: 333 4444
>
> A couple questions related to the SIP proxy:
>
> If the SIP proxy is located in Hong Kong, but the phone number is a New
> Zealand number, does the location of the proxy have any impact on the
> number of characters in the string for inbound/outbound calls?
>
> Again, apologies for the rudimentary nature of these questions, but having
> nearly exhausted all other options (docs, searches, IRC), I am now spending
> a large amount of time guessing and trial and error without any progress.
>
> Many thanks for any feedback at all.
>
> Nate
> ,
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ljjimenez at gmail.com  Mon Jul  3 11:19:11 2017
From: ljjimenez at gmail.com (Luis Jimenez)
Date: Mon, 3 Jul 2017 07:19:11 -0400
Subject: [Freeswitch-users] Dialplan: Outbound route:
 destination_number: regex question
In-Reply-To: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>
References: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>
Message-ID: <CAKUOwoFi55pCh=2_8WxUOSg4-OTJbMTBOYT_JrmGSi4tqvFyAA@mail.gmail.com>

^(\+)?((64|0)22)?(3334444)$



On Sun, Jul 2, 2017 at 11:24 PM, Nate <nathan.port at gmail.com> wrote:

> Hi Roy, many thanks for responding.
>
> I am simply trying to finishing configuring the server to make/receive
> calls from the PSTN.
>
> My reference to the regex is from trying to configure an outbound route.
>
> Can anyone offer any advice on a regex that will work for me to reach any
> of the following phone numbers:
>
> International: +64 22 333 4444 <+64%2022%20333%204444>
> non-local: 022 333 4444
> local: 333 4444
>
> From the number assigned to my account? +64 9 xxx yyyy
>
> Many many thanks!
>
> Nate.
>
> On Sat, Jul 1, 2017 at 9:17 AM, <royj at yandex.ru> wrote:
>
>> Not quite clear. What are you trying to achieve?
>>
>> 30.06.2017, 11:11, "Nate" <nathan.port at gmail.com>:
>>
>> Good day/evening everyone,
>>
>> Apologies for not being able to figure this out on my own. I've been
>> searching and trying for several days to get inbound/outbound working but
>> have yet to see success.
>>
>> At this stage I need help determining a proper regex expression for
>> handling New Zealand phone numbers.
>>
>> For instance, there are three different ways of expressing numbers here
>> in NZ:
>>
>> International: +64 22 333 4444 <+64%2022%20333%204444>
>> non-local: 022 333 4444
>> local: 333 4444
>>
>> A couple questions related to the SIP proxy:
>>
>> If the SIP proxy is located in Hong Kong, but the phone number is a New
>> Zealand number, does the location of the proxy have any impact on the
>> number of characters in the string for inbound/outbound calls?
>>
>> Again, apologies for the rudimentary nature of these questions, but
>> having nearly exhausted all other options (docs, searches, IRC), I am now
>> spending a large amount of time guessing and trial and error without any
>> progress.
>>
>> Many thanks for any feedback at all.
>>
>> Nate
>> ,
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vma at vallimamod.org  Mon Jul  3 11:44:22 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Mon, 3 Jul 2017 13:44:22 +0200
Subject: [Freeswitch-users] Dialplan: Outbound route:
 destination_number: regex question
In-Reply-To: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>
References: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>
Message-ID: <478D2D4F-2967-4A18-BFAC-5AD1C35D0E47@vallimamod.org>

Hi,

IMHO, instead of trying to find a single complicated regex working in all cases, a more readable way is to use condition regex="any" to stack all your possible matching regexes and let freeswitch try them one by one:

<condition regex="any">
  <regex field="destination_number" expression="^(\+6422\d{7})$"/>
  <regex field="destination_number" expression="^(022\d{7})$"/>
  <regex field="destination_number" expression="^(\d{7})$"/>
  <action ...>
</condition>

Hope this helps.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.


> On 3 Jul 2017, at 05:24, Nate <nathan.port at gmail.com> wrote:
> 
> Hi Roy, many thanks for responding.
> 
> I am simply trying to finishing configuring the server to make/receive calls from the PSTN.
> 
> My reference to the regex is from trying to configure an outbound route.
> 
> Can anyone offer any advice on a regex that will work for me to reach any of the following phone numbers:
> 
> International: +64 22 333 4444
> non-local: 022 333 4444
> local: 333 4444
> 
> From the number assigned to my account? +64 9 xxx yyyy
> 
> Many many thanks!
> 
> Nate.
> 
> On Sat, Jul 1, 2017 at 9:17 AM, <royj at yandex.ru> wrote:
> Not quite clear. What are you trying to achieve?
>  
> 30.06.2017, 11:11, "Nate" <nathan.port at gmail.com>:
>> Good day/evening everyone,
>>  
>> Apologies for not being able to figure this out on my own. I've been searching and trying for several days to get inbound/outbound working but have yet to see success.
>>  
>> At this stage I need help determining a proper regex expression for handling New Zealand phone numbers.
>>  
>> For instance, there are three different ways of expressing numbers here in NZ:
>>  
>> International: +64 22 333 4444
>> non-local: 022 333 4444
>> local: 333 4444
>>  
>> A couple questions related to the SIP proxy:
>>  
>> If the SIP proxy is located in Hong Kong, but the phone number is a New Zealand number, does the location of the proxy have any impact on the number of characters in the string for inbound/outbound calls?
>>  
>> Again, apologies for the rudimentary nature of these questions, but having nearly exhausted all other options (docs, searches, IRC), I am now spending a large amount of time guessing and trial and error without any progress.
>>  
>> Many thanks for any feedback at all.
>>  
>> Nate
>> ,
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From findmeinwland at gmail.com  Mon Jul  3 13:42:52 2017
From: findmeinwland at gmail.com (Artur Mega)
Date: Mon, 3 Jul 2017 18:42:52 +0500
Subject: [Freeswitch-users] Dialplan: Outbound route:
 destination_number: regex question
In-Reply-To: <478D2D4F-2967-4A18-BFAC-5AD1C35D0E47@vallimamod.org>
References: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>
 <478D2D4F-2967-4A18-BFAC-5AD1C35D0E47@vallimamod.org>
Message-ID: <CAOLEQ7fkZ0ytU7PVsQubbShZrm5hcqwQk_oKFVm1gbqtQgCz5Q@mail.gmail.com>

Sorry, I asked a question to freeswitch users list, but why I don't see
this question here? It is my second question, and second time question not
goes here.

2017-07-03 16:44 GMT+05:00 Vallimamod Abdullah <vma at vallimamod.org>:

> Hi,
>
> IMHO, instead of trying to find a single complicated regex working in all
> cases, a more readable way is to use condition regex="any" to stack all
> your possible matching regexes and let freeswitch try them one by one:
>
> <condition regex="any">
>   <regex field="destination_number" expression="^(\+6422\d{7})$"/>
>   <regex field="destination_number" expression="^(022\d{7})$"/>
>   <regex field="destination_number" expression="^(\d{7})$"/>
>   <action ...>
> </condition>
>
> Hope this helps.
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sipsolutions.fr
> .
>
>
> > On 3 Jul 2017, at 05:24, Nate <nathan.port at gmail.com> wrote:
> >
> > Hi Roy, many thanks for responding.
> >
> > I am simply trying to finishing configuring the server to make/receive
> calls from the PSTN.
> >
> > My reference to the regex is from trying to configure an outbound route.
> >
> > Can anyone offer any advice on a regex that will work for me to reach
> any of the following phone numbers:
> >
> > International: +64 22 333 4444
> > non-local: 022 333 4444
> > local: 333 4444
> >
> > From the number assigned to my account? +64 9 xxx yyyy
> >
> > Many many thanks!
> >
> > Nate.
> >
> > On Sat, Jul 1, 2017 at 9:17 AM, <royj at yandex.ru> wrote:
> > Not quite clear. What are you trying to achieve?
> >
> > 30.06.2017, 11:11, "Nate" <nathan.port at gmail.com>:
> >> Good day/evening everyone,
> >>
> >> Apologies for not being able to figure this out on my own. I've been
> searching and trying for several days to get inbound/outbound working but
> have yet to see success.
> >>
> >> At this stage I need help determining a proper regex expression for
> handling New Zealand phone numbers.
> >>
> >> For instance, there are three different ways of expressing numbers here
> in NZ:
> >>
> >> International: +64 22 333 4444
> >> non-local: 022 333 4444
> >> local: 333 4444
> >>
> >> A couple questions related to the SIP proxy:
> >>
> >> If the SIP proxy is located in Hong Kong, but the phone number is a New
> Zealand number, does the location of the proxy have any impact on the
> number of characters in the string for inbound/outbound calls?
> >>
> >> Again, apologies for the rudimentary nature of these questions, but
> having nearly exhausted all other options (docs, searches, IRC), I am now
> spending a large amount of time guessing and trial and error without any
> progress.
> >>
> >> Many thanks for any feedback at all.
> >>
> >> Nate
> >> ,
> >> ____________________________________________________________
> _____________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
> options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
​С уважением, ​
Артур
​Regards, Arthur​
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From rick at magicmail.mooo.com  Mon Jul  3 15:54:08 2017
From: rick at magicmail.mooo.com (Rick Jarvis)
Date: Mon, 3 Jul 2017 16:54:08 +0100
Subject: [Freeswitch-users] Max transmissions
Message-ID: <EB0BC787-0AA9-43D0-93A3-C55DE13904AD@magicmail.mooo.com>

Where should I start looking to fault find this? I’m getting choppy audio (FS->External) at the start of a call, which I’m guessing might be connected?

2017-07-03 16:51:00.631873 [DEBUG] switch_rtp.c:1464  [  zrtp utils]: 	Send <HELLO> ssrc=2371749770 seq=53538 size=148. Stream 2:CLEAR:START
2017-07-03 16:51:00.831873 [DEBUG] switch_rtp.c:1464  [  zrtp utils]: 	Send <HELLO> ssrc=2371749770 seq=53539 size=148. Stream 2:CLEAR:START
2017-07-03 16:51:01.031875 [DEBUG] switch_rtp.c:1464  [  zrtp utils]: 	Send <HELLO> ssrc=2371749770 seq=53540 size=148. Stream 2:CLEAR:START
2017-07-03 16:51:01.231873 [DEBUG] switch_rtp.c:1464  [  zrtp utils]: 	Send <HELLO> ssrc=2371749770 seq=53541 size=148. Stream 2:CLEAR:START
2017-07-03 16:51:01.431886 [DEBUG] switch_rtp.c:1464  [  zrtp utils]: 	Send <HELLO> ssrc=2371749770 seq=53542 size=148. Stream 2:CLEAR:START
2017-07-03 16:51:01.631874 [DEBUG] switch_rtp.c:1464  [  zrtp utils]: 	Send <HELLO> ssrc=2371749770 seq=53543 size=148. Stream 2:CLEAR:START
2017-07-03 16:51:01.831873 [DEBUG] switch_rtp.c:1464  [ zrtp engine]: WARNING! HELLO Max retransmissions count reached (20 retries). ID=2
2017-07-03 16:51:01.831873 [DEBUG] switch_rtp.c:1464  [        zrtp]: 	Stream ID=2 CLEAR switching <START> ---> <NOZRTP>.
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From sagar.malam at ecosmob.com  Mon Jul  3 13:54:10 2017
From: sagar.malam at ecosmob.com (Sagar Malam)
Date: Mon, 03 Jul 2017 13:54:10 +0000
Subject: [Freeswitch-users] Need channel variables stored in DB in realtime
Message-ID: <CAL9P3ciJUmjy45bHhcufQZ5pyLQve9rXXWvxXSCZJVHpgGpLEQ@mail.gmail.com>

Hello,

I want to have all the channel variables stored in DB in realtime.I tried
making  ODBC configuration for MySQL for FS core data but i did not find
any table with channel variable information.

Is there any inbuilt way to do so ? Please help.

Thanks in advance
-- 
Thanks & Regards,
Sagar Malam
Team Lead | Ecosmob Technologies Pvt. Ltd.
(+91)9601533171 | www.ecosmob.com
<http://www.google.com/url?q=http%3A%2F%2Fwww.hodusoft.com&sa=D&sntz=1&usg=AFQjCNHXhIaelhkmhqcPU8D1lt3QoYpm2w>
Skype: sagar.ecosmob
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From mark.melling at savageminds.com  Mon Jul  3 15:43:28 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Mon, 03 Jul 2017 15:43:28 +0000
Subject: [Freeswitch-users] Making DTMF tone 'silent' to caller
Message-ID: <CAJp-f1NK7rV+_QLqU7X=f3LspnxRLsuex-VNO+JuA1ap5iSRkg@mail.gmail.com>

Hi,

I have a scenario where a mobile phone makes a 'normal' call (not SIP/VoIP)
into a Freeswitch conference (via a SIP trunk). The dialled number includes
DTMF tones to be transmitted after the call is dialled.

When sent these DTMF tones can be heard by the mobile phone user. What I
want to achieve is that these tones are not audible to the mobile caller.

In a previous incarnation of this system the mobile phone dialled into a
Twilio hosted conference. In the set-up when the DTMF tones were
transmitted the mobile phone user could not hear them.

Is it the case that Freeswitch is generating the DTMF tones and if so how
can I stop this?

If not, what are the possible explanations - e.g. perhaps DTMF tones are
played in-band in one scenario versus out-of-band in the other?

Any suggestions and thoughts most appreciated.

Thanks

Mark
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From steveayre at gmail.com  Mon Jul  3 20:30:06 2017
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 3 Jul 2017 21:30:06 +0100
Subject: [Freeswitch-users] Dialplan: Outbound route:
 destination_number: regex question
In-Reply-To: <478D2D4F-2967-4A18-BFAC-5AD1C35D0E47@vallimamod.org>
References: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>
 <478D2D4F-2967-4A18-BFAC-5AD1C35D0E47@vallimamod.org>
Message-ID: <CAFiqYukbBGtKOVGtM+RwDnGmtZ+D37Dqqy_XL5-Xtd52NxEkRA@mail.gmail.com>

"local: 333 4444" is the one that's going to cause you problems really.
It's easy enough to spot a prefix, but hard to spot no prefix (eg 3334444 a
local number or an international call to France?)

You could spot the international and national dialing prefixes, and treat
anything else as local perhaps?


On 3 July 2017 at 12:44, Vallimamod Abdullah <vma at vallimamod.org> wrote:

> Hi,
>
> IMHO, instead of trying to find a single complicated regex working in all
> cases, a more readable way is to use condition regex="any" to stack all
> your possible matching regexes and let freeswitch try them one by one:
>
> <condition regex="any">
>   <regex field="destination_number" expression="^(\+6422\d{7})$"/>
>   <regex field="destination_number" expression="^(022\d{7})$"/>
>   <regex field="destination_number" expression="^(\d{7})$"/>
>   <action ...>
> </condition>
>
> Hope this helps.
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sipsolutions.fr
> .
>
>
> > On 3 Jul 2017, at 05:24, Nate <nathan.port at gmail.com> wrote:
> >
> > Hi Roy, many thanks for responding.
> >
> > I am simply trying to finishing configuring the server to make/receive
> calls from the PSTN.
> >
> > My reference to the regex is from trying to configure an outbound route.
> >
> > Can anyone offer any advice on a regex that will work for me to reach
> any of the following phone numbers:
> >
> > International: +64 22 333 4444
> > non-local: 022 333 4444
> > local: 333 4444
> >
> > From the number assigned to my account? +64 9 xxx yyyy
> >
> > Many many thanks!
> >
> > Nate.
> >
> > On Sat, Jul 1, 2017 at 9:17 AM, <royj at yandex.ru> wrote:
> > Not quite clear. What are you trying to achieve?
> >
> > 30.06.2017, 11:11, "Nate" <nathan.port at gmail.com>:
> >> Good day/evening everyone,
> >>
> >> Apologies for not being able to figure this out on my own. I've been
> searching and trying for several days to get inbound/outbound working but
> have yet to see success.
> >>
> >> At this stage I need help determining a proper regex expression for
> handling New Zealand phone numbers.
> >>
> >> For instance, there are three different ways of expressing numbers here
> in NZ:
> >>
> >> International: +64 22 333 4444
> >> non-local: 022 333 4444
> >> local: 333 4444
> >>
> >> A couple questions related to the SIP proxy:
> >>
> >> If the SIP proxy is located in Hong Kong, but the phone number is a New
> Zealand number, does the location of the proxy have any impact on the
> number of characters in the string for inbound/outbound calls?
> >>
> >> Again, apologies for the rudimentary nature of these questions, but
> having nearly exhausted all other options (docs, searches, IRC), I am now
> spending a large amount of time guessing and trial and error without any
> progress.
> >>
> >> Many thanks for any feedback at all.
> >>
> >> Nate
> >> ,
> >> ____________________________________________________________
> _____________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
> options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joelists at tm.net.uk  Mon Jul  3 20:48:22 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Mon, 3 Jul 2017 21:48:22 +0100
Subject: [Freeswitch-users] mod_rad_auth dial plan condition to match
	radius result
In-Reply-To: <B19941B2-6461-4E80-8A57-E00260877102@tm.net.uk>
References: <7CAC6D12-1BD1-42B4-9D5A-A5263BE2127B@tm.net.uk>
 <A465D9D4-196B-42E2-AE99-550A48B8A44D@vallimamod.org>
 <B19941B2-6461-4E80-8A57-E00260877102@tm.net.uk>
Message-ID: <93977B89-A45F-4CC8-B30E-B6029E07BCB5@tm.net.uk>

Ok, So I have done a lot of playing and it seems that I cannot run the Auth_function inline as it won’t allow me to. I get an error in the log.

It therefore means that the result of it is not available for when the condition is parsed.

So my question is, How do I make Freeswitch hang up the call if the Auth fails?

Any help would be much appreciated as I am banging my head against this!

Regards
> On 2 Jul 2017, at 22:45, Joseph Waite <joelists at tm.net.uk> wrote:
> 
> If I put the inline=“true” in the action that calls auth_function then it fails to return any results in the log output.
> 
> 
>> On 2 Jul 2017, at 18:28, Vallimamod Abdullah <vma at vallimamod.org <mailto:vma at vallimamod.org>> wrote:
>> 
>> Hi,
>> 
>> Try adding inline="true" to the action returning the result you want to check.
>> There is a nice explanation on the wiki on how the dialplan works and why this is necessary if you want to use the result of an action in a subsequent condition.
>> 
>> Best regards,
>> -- 
>> Vallimamod Abdullah
>> SIP Solutions
>> vma at sipsolutions.fr <mailto:vma at sipsolutions.fr>.
>> 
>>> On 2 Jul 2017, at 18:30, Jospeh Waite <joelists at tm.net.uk <mailto:joelists at tm.net.uk>> wrote:
>>> 
>>> Hi Guys
>>> 
>>> Nearly there with mod_rad_auth, I have it being called in my dial plan and its sending the Radisu auth and working as expected.
>>> 
>>> I now want to put a condition so that in the event of Radius Auth failing the dial plan will end and not allow the call to be made.
>>> 
>>> I have put the following in the dial plan
>>> 
>>>         <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>> 
>>>   <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>>>   <action application="log" data="INFO  credit_time=${credit_time}"/>
>>>   <action application="log" data="INFO  return_code=${return_code}"/>
>>>    <condition field="${return_code}" expression="^16$">
>>> 
>>>             <action application="set" data="execute_on_answer=sched_hangup +${credit_time} alloted_timeout" />
>>>             <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060 <sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060>" />
>>>             <action application="hangup" data="${bridge_hangup_cause}"/>
>>>         </condition>
>>> 
>>> 
>>> And I check the freeswitch log and see
>>> 
>>> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  AUTH_RESULT=OK)
>>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  AUTH_RESULT=OK
>>> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  credit_time=7199)
>>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  credit_time=7199
>>> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  return_code=16)
>>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  return_code=16
>>> 
>>> 
>>> But it fails to execute the bit within the condition.
>>> 
>>> 
>>> Any Ideas of how to make this work?
>>> 
>>> Regards
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From covici at ccs.covici.com  Mon Jul  3 20:58:14 2017
From: covici at ccs.covici.com (John Covici)
Date: Mon, 03 Jul 2017 16:58:14 -0400
Subject: [Freeswitch-users] mod_rad_auth dial plan condition to
	match	radius result
In-Reply-To: <93977B89-A45F-4CC8-B30E-B6029E07BCB5@tm.net.uk>
References: <7CAC6D12-1BD1-42B4-9D5A-A5263BE2127B@tm.net.uk>
 <A465D9D4-196B-42E2-AE99-550A48B8A44D@vallimamod.org>
 <B19941B2-6461-4E80-8A57-E00260877102@tm.net.uk>
 <93977B89-A45F-4CC8-B30E-B6029E07BCB5@tm.net.uk>
Message-ID: <m3lgo5nspl.wl-covici@ccs.covici.com>

I would say you have to do that logic using some programming language,
lua, c# or whatever, this is too much for the xml dialplan.

On Mon, 03 Jul 2017 16:48:22 -0400,
Joseph Waite wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; utf-8 (quoted-printable)>]
> [1.2  <text/html; utf-8 (quoted-printable)>]
> Ok, So I have done a lot of playing and it seems that I cannot run the Auth_function inline as it won’t allow me to. I get an error in the log.
> 
> It therefore means that the result of it is not available for when the condition is parsed.
> 
> So my question is, How do I make Freeswitch hang up the call if the Auth fails?
> 
> Any help would be much appreciated as I am banging my head against this!
> 
> Regards
> 
>  On 2 Jul 2017, at 22:45, Joseph Waite <joelists at tm.net.uk> wrote:
> 
>  If I put the inline=“true” in the action that calls auth_function then it fails to return any results in the log output.
> 
>  On 2 Jul 2017, at 18:28, Vallimamod Abdullah <vma at vallimamod.org> wrote:
> 
>  Hi,
> 
>  Try adding inline="true" to the action returning the result you want to check.
>  There is a nice explanation on the wiki on how the dialplan works and why this is necessary if you want to use the result of an action in a subsequent condition.
> 
>  Best regards,
>  -- 
>  Vallimamod Abdullah
>  SIP Solutions
>  vma at sipsolutions.fr
>  .
> 
>  On 2 Jul 2017, at 18:30, Jospeh Waite <joelists at tm.net.uk> wrote:
> 
>  Hi Guys
> 
>  Nearly there with mod_rad_auth, I have it being called in my dial plan and its sending the Radisu auth and working as expected.
> 
>  I now want to put a condition so that in the event of Radius Auth failing the dial plan will end and not allow the call to be made.
> 
>  I have put the following in the dial plan
> 
>  <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
> 
>  <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/>
>  <action application="log" data="INFO credit_time=${credit_time}"/>
>  <action application="log" data="INFO return_code=${return_code}"/>
>  <condition field="${return_code}" expression="^16$">
> 
>  <action application="set" data="execute_on_answer=sched_hangup +${credit_time} alloted_timeout" />
>  <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060" />
>  <action application="hangup" data="${bridge_hangup_cause}"/>
>  </condition>
> 
>  And I check the freeswitch log and see
> 
>  EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO AUTH_RESULT=OK)
>  2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 AUTH_RESULT=OK
>  EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO credit_time=7199)
>  2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 credit_time=7199
>  EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO return_code=16)
>  2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 return_code=16
> 
>  But it fails to execute the bit within the condition.
> 
>  Any Ideas of how to make this work?
> 
>  Regards
>  _________________________________________________________________________
>  Professional FreeSWITCH Consulting Services:
>  consulting at freeswitch.org
>  http://www.freeswitchsolutions.com
> 
>  Official FreeSWITCH Sites
>  http://www.freeswitch.org
>  http://confluence.freeswitch.org
>  http://www.cluecon.com
> 
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
> 
>  _________________________________________________________________________
>  Professional FreeSWITCH Consulting Services:
>  consulting at freeswitch.org
>  http://www.freeswitchsolutions.com
> 
>  Official FreeSWITCH Sites
>  http://www.freeswitch.org
>  http://confluence.freeswitch.org
>  http://www.cluecon.com
> 
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
> 
>  _________________________________________________________________________
>  Professional FreeSWITCH Consulting Services:
>  consulting at freeswitch.org
>  http://www.freeswitchsolutions.com
> 
>  Official FreeSWITCH Sites
>  http://www.freeswitch.org
>  http://confluence.freeswitch.org
>  http://www.cluecon.com
> 
>  FreeSWITCH-users mailing list
>  FreeSWITCH-users at lists.freeswitch.org
>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
> 
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From joelists at tm.net.uk  Mon Jul  3 21:14:02 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Mon, 3 Jul 2017 22:14:02 +0100
Subject: [Freeswitch-users] mod_rad_auth dial plan condition to
	match	radius result
In-Reply-To: <m3lgo5nspl.wl-covici@ccs.covici.com>
References: <7CAC6D12-1BD1-42B4-9D5A-A5263BE2127B@tm.net.uk>
 <A465D9D4-196B-42E2-AE99-550A48B8A44D@vallimamod.org>
 <B19941B2-6461-4E80-8A57-E00260877102@tm.net.uk>
 <93977B89-A45F-4CC8-B30E-B6029E07BCB5@tm.net.uk>
 <m3lgo5nspl.wl-covici@ccs.covici.com>
Message-ID: <4B08B696-FF16-4ABD-9A43-EF3319DFF7FC@tm.net.uk>

Really? for a simple If auth fails, hangup?

> On 3 Jul 2017, at 21:58, John Covici <covici at ccs.covici.com> wrote:
> 
> I would say you have to do that logic using some programming language,
> lua, c# or whatever, this is too much for the xml dialplan.
> 
> On Mon, 03 Jul 2017 16:48:22 -0400,
> Joseph Waite wrote:
>> 
>> [1  <multipart/alternative (7bit)>]
>> [1.1  <text/plain; utf-8 (quoted-printable)>]
>> [1.2  <text/html; utf-8 (quoted-printable)>]
>> Ok, So I have done a lot of playing and it seems that I cannot run the Auth_function inline as it won’t allow me to. I get an error in the log.
>> 
>> It therefore means that the result of it is not available for when the condition is parsed.
>> 
>> So my question is, How do I make Freeswitch hang up the call if the Auth fails?
>> 
>> Any help would be much appreciated as I am banging my head against this!
>> 
>> Regards
>> 
>> On 2 Jul 2017, at 22:45, Joseph Waite <joelists at tm.net.uk> wrote:
>> 
>> If I put the inline=“true” in the action that calls auth_function then it fails to return any results in the log output.
>> 
>> On 2 Jul 2017, at 18:28, Vallimamod Abdullah <vma at vallimamod.org> wrote:
>> 
>> Hi,
>> 
>> Try adding inline="true" to the action returning the result you want to check.
>> There is a nice explanation on the wiki on how the dialplan works and why this is necessary if you want to use the result of an action in a subsequent condition.
>> 
>> Best regards,
>> -- 
>> Vallimamod Abdullah
>> SIP Solutions
>> vma at sipsolutions.fr
>> .
>> 
>> On 2 Jul 2017, at 18:30, Jospeh Waite <joelists at tm.net.uk> wrote:
>> 
>> Hi Guys
>> 
>> Nearly there with mod_rad_auth, I have it being called in my dial plan and its sending the Radisu auth and working as expected.
>> 
>> I now want to put a condition so that in the event of Radius Auth failing the dial plan will end and not allow the call to be made.
>> 
>> I have put the following in the dial plan
>> 
>> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>> 
>> <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/>
>> <action application="log" data="INFO credit_time=${credit_time}"/>
>> <action application="log" data="INFO return_code=${return_code}"/>
>> <condition field="${return_code}" expression="^16$">
>> 
>> <action application="set" data="execute_on_answer=sched_hangup +${credit_time} alloted_timeout" />
>> <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060" />
>> <action application="hangup" data="${bridge_hangup_cause}"/>
>> </condition>
>> 
>> And I check the freeswitch log and see
>> 
>> EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO AUTH_RESULT=OK)
>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 AUTH_RESULT=OK
>> EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO credit_time=7199)
>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 credit_time=7199
>> EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO return_code=16)
>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 return_code=16
>> 
>> But it fails to execute the bit within the condition.
>> 
>> Any Ideas of how to make this work?
>> 
>> Regards
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> [2  <text/plain; utf-8 (base64)>]
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>         John Covici
>         covici at ccs.covici.com
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From joel at gogii.net  Mon Jul  3 21:43:46 2017
From: joel at gogii.net (Joel Serrano)
Date: Mon, 03 Jul 2017 21:43:46 +0000
Subject: [Freeswitch-users] Need channel variables stored in DB in
	realtime
In-Reply-To: <CAL9P3ciJUmjy45bHhcufQZ5pyLQve9rXXWvxXSCZJVHpgGpLEQ@mail.gmail.com>
References: <CAL9P3ciJUmjy45bHhcufQZ5pyLQve9rXXWvxXSCZJVHpgGpLEQ@mail.gmail.com>
Message-ID: <CAMtXxQkoZ4axn4M453z_1PJHp=V-11jwCFqBXrL5Y7gJ=cqrkw@mail.gmail.com>

Have you tried listening for events with a lua script or something to
insert/update the db?

On Mon, Jul 3, 2017 at 10:37 Sagar Malam <sagar.malam at ecosmob.com> wrote:

> Hello,
>
> I want to have all the channel variables stored in DB in realtime.I tried
> making  ODBC configuration for MySQL for FS core data but i did not find
> any table with channel variable information.
>
> Is there any inbuilt way to do so ? Please help.
>
> Thanks in advance
> --
> Thanks & Regards,
> Sagar Malam
> Team Lead | Ecosmob Technologies Pvt. Ltd.
> (+91)9601533171 | www.ecosmob.com
> <http://www.google.com/url?q=http%3A%2F%2Fwww.hodusoft.com&sa=D&sntz=1&usg=AFQjCNHXhIaelhkmhqcPU8D1lt3QoYpm2w>
> Skype: sagar.ecosmob
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Mon Jul  3 21:51:52 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Jul 2017 17:51:52 -0400
Subject: [Freeswitch-users] Need channel variables stored in DB in
 realtime
In-Reply-To: <CAMtXxQkoZ4axn4M453z_1PJHp=V-11jwCFqBXrL5Y7gJ=cqrkw@mail.gmail.com>
References: <CAL9P3ciJUmjy45bHhcufQZ5pyLQve9rXXWvxXSCZJVHpgGpLEQ@mail.gmail.com>
 <CAMtXxQkoZ4axn4M453z_1PJHp=V-11jwCFqBXrL5Y7gJ=cqrkw@mail.gmail.com>
Message-ID: <D8804279-9029-4092-99D4-D8E97EB5F1F1@jerris.com>

No need to help these guys, they sell FreeSWITCH support and do nothing at all to support the project.


> On Jul 3, 2017, at 5:43 PM, Joel Serrano <joel at gogii.net> wrote:
> 
> Have you tried listening for events with a lua script or something to insert/update the db?
> 
> On Mon, Jul 3, 2017 at 10:37 Sagar Malam <sagar.malam at ecosmob.com <mailto:sagar.malam at ecosmob.com>> wrote:
> Hello,
> 
> I want to have all the channel variables stored in DB in realtime.I tried making  ODBC configuration for MySQL for FS core data but i did not find any table with channel variable information. 
> 
> Is there any inbuilt way to do so ? Please help.
> 
> Thanks in advance
> -- 
> Thanks & Regards,
> Sagar Malam
> Team Lead | Ecosmob Technologies Pvt. Ltd.
> (+91)9601533171 | www.ecosmob.com <http://www.google.com/url?q=http%3A%2F%2Fwww.hodusoft.com&sa=D&sntz=1&usg=AFQjCNHXhIaelhkmhqcPU8D1lt3QoYpm2w>
> Skype: sagar.ecosmob

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From covici at ccs.covici.com  Mon Jul  3 22:00:09 2017
From: covici at ccs.covici.com (John Covici)
Date: Mon, 03 Jul 2017 18:00:09 -0400
Subject: [Freeswitch-users] mod_rad_auth dial plan condition
	to	match	radius result
In-Reply-To: <4B08B696-FF16-4ABD-9A43-EF3319DFF7FC@tm.net.uk>
References: <7CAC6D12-1BD1-42B4-9D5A-A5263BE2127B@tm.net.uk>
 <A465D9D4-196B-42E2-AE99-550A48B8A44D@vallimamod.org>
 <B19941B2-6461-4E80-8A57-E00260877102@tm.net.uk>
 <93977B89-A45F-4CC8-B30E-B6029E07BCB5@tm.net.uk>
 <m3lgo5nspl.wl-covici@ccs.covici.com>
 <4B08B696-FF16-4ABD-9A43-EF3319DFF7FC@tm.net.uk>
Message-ID: <m3inj9npue.wl-covici@ccs.covici.com>

Yep, the dialplan is for very simple things, what you want to do is if
auth fails, don't do the bridge, I am not a Lua expert, but maybe that
would work.

On Mon, 03 Jul 2017 17:14:02 -0400,
Joseph Waite wrote:
> 
> Really? for a simple If auth fails, hangup?
> 
> > On 3 Jul 2017, at 21:58, John Covici <covici at ccs.covici.com> wrote:
> > 
> > I would say you have to do that logic using some programming language,
> > lua, c# or whatever, this is too much for the xml dialplan.
> > 
> > On Mon, 03 Jul 2017 16:48:22 -0400,
> > Joseph Waite wrote:
> >> 
> >> [1  <multipart/alternative (7bit)>]
> >> [1.1  <text/plain; utf-8 (quoted-printable)>]
> >> [1.2  <text/html; utf-8 (quoted-printable)>]
> >> Ok, So I have done a lot of playing and it seems that I cannot run the Auth_function inline as it won’t allow me to. I get an error in the log.
> >> 
> >> It therefore means that the result of it is not available for when the condition is parsed.
> >> 
> >> So my question is, How do I make Freeswitch hang up the call if the Auth fails?
> >> 
> >> Any help would be much appreciated as I am banging my head against this!
> >> 
> >> Regards
> >> 
> >> On 2 Jul 2017, at 22:45, Joseph Waite <joelists at tm.net.uk> wrote:
> >> 
> >> If I put the inline=“true” in the action that calls auth_function then it fails to return any results in the log output.
> >> 
> >> On 2 Jul 2017, at 18:28, Vallimamod Abdullah <vma at vallimamod.org> wrote:
> >> 
> >> Hi,
> >> 
> >> Try adding inline="true" to the action returning the result you want to check.
> >> There is a nice explanation on the wiki on how the dialplan works and why this is necessary if you want to use the result of an action in a subsequent condition.
> >> 
> >> Best regards,
> >> -- 
> >> Vallimamod Abdullah
> >> SIP Solutions
> >> vma at sipsolutions.fr
> >> .
> >> 
> >> On 2 Jul 2017, at 18:30, Jospeh Waite <joelists at tm.net.uk> wrote:
> >> 
> >> Hi Guys
> >> 
> >> Nearly there with mod_rad_auth, I have it being called in my dial plan and its sending the Radisu auth and working as expected.
> >> 
> >> I now want to put a condition so that in the event of Radius Auth failing the dial plan will end and not allow the call to be made.
> >> 
> >> I have put the following in the dial plan
> >> 
> >> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
> >> 
> >> <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/>
> >> <action application="log" data="INFO credit_time=${credit_time}"/>
> >> <action application="log" data="INFO return_code=${return_code}"/>
> >> <condition field="${return_code}" expression="^16$">
> >> 
> >> <action application="set" data="execute_on_answer=sched_hangup +${credit_time} alloted_timeout" />
> >> <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060" />
> >> <action application="hangup" data="${bridge_hangup_cause}"/>
> >> </condition>
> >> 
> >> And I check the freeswitch log and see
> >> 
> >> EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO AUTH_RESULT=OK)
> >> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 AUTH_RESULT=OK
> >> EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO credit_time=7199)
> >> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 credit_time=7199
> >> EXECUTE sofia/external/07966677711 at 185.8.92.3 log(INFO return_code=16)
> >> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724 return_code=16
> >> 
> >> But it fails to execute the bit within the condition.
> >> 
> >> Any Ideas of how to make this work?
> >> 
> >> Regards
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> >> [2  <text/plain; utf-8 (base64)>]
> >> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >> 
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >> 
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > 
> > -- 
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> > 
> >         John Covici
> >         covici at ccs.covici.com
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From alexandr.popov at iqoption.com  Tue Jul  4 08:58:05 2017
From: alexandr.popov at iqoption.com (Alexandr Popov)
Date: Tue, 4 Jul 2017 11:58:05 +0300
Subject: [Freeswitch-users] Error sending queries to pgsql
In-Reply-To: <f76beadd-9014-d279-1fb8-ea9d0048b240@wirelessmundi.com>
References: <d2f4526f-a783-a216-405d-126c253e34fd@wirelessmundi.com>
 <31066AE1-BC31-4C3F-8809-6C5B93C3B0E8@jerris.com>
 <f76beadd-9014-d279-1fb8-ea9d0048b240@wirelessmundi.com>
Message-ID: <CAKeZRhvyLu+uT4_c_ZSEQVxW5zYZgrgeuAf10Qxpqn_5B-XRTA@mail.gmail.com>

Seems its trouble with sockets. I have the same problem appeared about a
month ago.

2017-06-29 19:31 GMT+03:00 Antonio Silva <asilva at wirelessmundi.com>:

> Hi Michael,
>
> Yes, i'm trying to figure it out if is an issue in FS or external.. but
> the message from PG i can't translate it.. i just enable more logs to see
> if i got extra hints...
>
>
> Thanks.
>
> Saludos / Regards / Cumprimentos,
> António silva
>
> On 06/29/2017 06:22 PM, Michael Jerris wrote:
>
>> If you can figure out a reliable way to reproduce this issue, please file
>> a jira with details on what causes it.
>>
>> On Jun 29, 2017, at 7:22 AM, Antonio Silva <asilva at wirelessmundi.com>
>>> wrote:
>>>
>>> Hi all,
>>>
>>> i use pgsql in core and from time to time i see critical messages like
>>> fail to send query, example:
>>>
>>> [CRIT] switch_pgsql.c:255 Failed to send query (update
>>> sip_authentication set expires='1498726568',last_nc=364 where
>>> nonce='3c20da2e-b663-46fe-8edc-c756f9166d07') to database: server
>>> closed the connection unexpectedly
>>>
>>> This was recently, i did an update to current master the previous
>>> version was from April, not sure if it could be an error on FS o some other
>>> issue on my box..
>>>
>>>
>>> PG is installed on the same server and the only thing i see from pg is
>>> "postgres[2236]: FATAL:  invalid frontend message type 21", PG is installed
>>> on the same server, running on /dev/shm with the same prio as FS and  the
>>> process never stopped.
>>>
>>>
>>> anyone has experience this error before? any idea what it could be the
>>> cause?
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From asilva at wirelessmundi.com  Tue Jul  4 09:34:38 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Tue, 4 Jul 2017 11:34:38 +0200
Subject: [Freeswitch-users] Error sending queries to pgsql
In-Reply-To: <CAKeZRhvyLu+uT4_c_ZSEQVxW5zYZgrgeuAf10Qxpqn_5B-XRTA@mail.gmail.com>
References: <d2f4526f-a783-a216-405d-126c253e34fd@wirelessmundi.com>
 <31066AE1-BC31-4C3F-8809-6C5B93C3B0E8@jerris.com>
 <f76beadd-9014-d279-1fb8-ea9d0048b240@wirelessmundi.com>
 <CAKeZRhvyLu+uT4_c_ZSEQVxW5zYZgrgeuAf10Qxpqn_5B-XRTA@mail.gmail.com>
Message-ID: <39b55634-01fd-afea-a08f-27094f2e237c@wirelessmundi.com>

In this same box i also have data corruption with sqlite databases, 
special the ones that i use with mod_lua.. but I think that it could be 
something related with kernel.. i recently update to 4.9.x, and there 
was some changes in ext4.

I don't really know how to debug this, so today i'm reverting to the old 
kernel, 4.4.x and check it happens again..

If it happens again, my next move will try to reproduce this in FS..

Saludos / Regards / Cumprimentos,
António silva

On 07/04/2017 10:58 AM, Alexandr Popov wrote:
> Seems its trouble with sockets. I have the same problem appeared about 
> a month ago.
>
> 2017-06-29 19:31 GMT+03:00 Antonio Silva <asilva at wirelessmundi.com 
> <mailto:asilva at wirelessmundi.com>>:
>
>     Hi Michael,
>
>     Yes, i'm trying to figure it out if is an issue in FS or
>     external.. but the message from PG i can't translate it.. i just
>     enable more logs to see if i got extra hints...
>
>
>     Thanks.
>
>     Saludos / Regards / Cumprimentos,
>     António silva
>
>     On 06/29/2017 06:22 PM, Michael Jerris wrote:
>
>         If you can figure out a reliable way to reproduce this issue,
>         please file a jira with details on what causes it.
>
>             On Jun 29, 2017, at 7:22 AM, Antonio Silva
>             <asilva at wirelessmundi.com
>             <mailto:asilva at wirelessmundi.com>> wrote:
>
>             Hi all,
>
>             i use pgsql in core and from time to time i see critical
>             messages like fail to send query, example:
>
>             [CRIT] switch_pgsql.c:255 Failed to send query (update
>             sip_authentication set expires='1498726568',last_nc=364
>             where nonce='3c20da2e-b663-46fe-8edc-c756f9166d07') to
>             database: server closed the connection unexpectedly
>
>             This was recently, i did an update to current master the
>             previous version was from April, not sure if it could be
>             an error on FS o some other issue on my box..
>
>
>             PG is installed on the same server and the only thing i
>             see from pg is "postgres[2236]: FATAL: invalid frontend
>             message type 21", PG is installed on the same server,
>             running on /dev/shm with the same prio as FS and  the
>             process never stopped.
>
>
>             anyone has experience this error before? any idea what it
>             could be the cause?
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>         <http://www.freeswitchsolutions.com>
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://confluence.freeswitch.org
>         <http://confluence.freeswitch.org>
>         http://www.cluecon.com
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>         http://www.freeswitch.org
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.villasmil.work at gmail.com  Tue Jul  4 10:23:43 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 04 Jul 2017 10:23:43 +0000
Subject: [Freeswitch-users] Need channel variables stored in DB in
	realtime
In-Reply-To: <D8804279-9029-4092-99D4-D8E97EB5F1F1@jerris.com>
References: <CAL9P3ciJUmjy45bHhcufQZ5pyLQve9rXXWvxXSCZJVHpgGpLEQ@mail.gmail.com>
 <CAMtXxQkoZ4axn4M453z_1PJHp=V-11jwCFqBXrL5Y7gJ=cqrkw@mail.gmail.com>
 <D8804279-9029-4092-99D4-D8E97EB5F1F1@jerris.com>
Message-ID: <CAFGRPVrfKa+zoC9Hb58JMOsrsaNe=GLbKHM0tE21WwzRHK_EDw@mail.gmail.com>

Shame on them!
On Mon, Jul 3, 2017 at 11:52 PM Michael Jerris <mike at jerris.com> wrote:

> No need to help these guys, they sell FreeSWITCH support and do nothing at
> all to support the project.
>
>
> On Jul 3, 2017, at 5:43 PM, Joel Serrano <joel at gogii.net> wrote:
>
> Have you tried listening for events with a lua script or something to
> insert/update the db?
>
> On Mon, Jul 3, 2017 at 10:37 Sagar Malam <sagar.malam at ecosmob.com> wrote:
>
>> Hello,
>>
>> I want to have all the channel variables stored in DB in realtime.I tried
>> making  ODBC configuration for MySQL for FS core data but i did not find
>> any table with channel variable information.
>>
>> Is there any inbuilt way to do so ? Please help.
>>
>> Thanks in advance
>> --
>> Thanks & Regards,
>> Sagar Malam
>> Team Lead | Ecosmob Technologies Pvt. Ltd.
>> (+91)9601533171 | www.ecosmob.com
>> <http://www.google.com/url?q=http%3A%2F%2Fwww.hodusoft.com&sa=D&sntz=1&usg=AFQjCNHXhIaelhkmhqcPU8D1lt3QoYpm2w>
>> Skype: sagar.ecosmob
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From vma at vallimamod.org  Tue Jul  4 16:09:39 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Tue, 4 Jul 2017 18:09:39 +0200
Subject: [Freeswitch-users] mod_rad_auth dial plan condition to match
 radius result
In-Reply-To: <93977B89-A45F-4CC8-B30E-B6029E07BCB5@tm.net.uk>
References: <7CAC6D12-1BD1-42B4-9D5A-A5263BE2127B@tm.net.uk>
 <A465D9D4-196B-42E2-AE99-550A48B8A44D@vallimamod.org>
 <B19941B2-6461-4E80-8A57-E00260877102@tm.net.uk>
 <93977B89-A45F-4CC8-B30E-B6029E07BCB5@tm.net.uk>
Message-ID: <B9081754-73EB-470C-9628-F2562DF1BCAE@vallimamod.org>

Hi,

In this case, use two extensions: in the first one, do the authentication and set the return code as a variable then transfer the call (with the transfer app) to a second extension where you test this variable to bridge or hangup the call accordingly.

Hope this helps.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
.

> On 3 Jul 2017, at 22:48, Joseph Waite <joelists at tm.net.uk> wrote:
> 
> Ok, So I have done a lot of playing and it seems that I cannot run the Auth_function inline as it won’t allow me to. I get an error in the log.
> 
> It therefore means that the result of it is not available for when the condition is parsed.
> 
> So my question is, How do I make Freeswitch hang up the call if the Auth fails?
> 
> Any help would be much appreciated as I am banging my head against this!
> 
> Regards
>> On 2 Jul 2017, at 22:45, Joseph Waite <joelists at tm.net.uk <mailto:joelists at tm.net.uk>> wrote:
>> 
>> If I put the inline=“true” in the action that calls auth_function then it fails to return any results in the log output.
>> 
>> 
>>> On 2 Jul 2017, at 18:28, Vallimamod Abdullah <vma at vallimamod.org <mailto:vma at vallimamod.org>> wrote:
>>> 
>>> Hi,
>>> 
>>> Try adding inline="true" to the action returning the result you want to check.
>>> There is a nice explanation on the wiki on how the dialplan works and why this is necessary if you want to use the result of an action in a subsequent condition.
>>> 
>>> Best regards,
>>> -- 
>>> Vallimamod Abdullah
>>> SIP Solutions
>>> vma at sipsolutions.fr <mailto:vma at sipsolutions.fr>.
>>> 
>>>> On 2 Jul 2017, at 18:30, Jospeh Waite <joelists at tm.net.uk <mailto:joelists at tm.net.uk>> wrote:
>>>> 
>>>> Hi Guys
>>>> 
>>>> Nearly there with mod_rad_auth, I have it being called in my dial plan and its sending the Radisu auth and working as expected.
>>>> 
>>>> I now want to put a condition so that in the event of Radius Auth failing the dial plan will end and not allow the call to be made.
>>>> 
>>>> I have put the following in the dial plan
>>>> 
>>>>         <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>> 
>>>>   <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>   <action application="log" data="INFO  credit_time=${credit_time}"/>
>>>>   <action application="log" data="INFO  return_code=${return_code}"/>
>>>>    <condition field="${return_code}" expression="^16$">
>>>> 
>>>>             <action application="set" data="execute_on_answer=sched_hangup +${credit_time} alloted_timeout" />
>>>>             <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060 <sip:${sip_from_user}@${sip_network_ip}}sofia/internal/${destination_number}@185.35.229.30:5060>" />
>>>>             <action application="hangup" data="${bridge_hangup_cause}"/>
>>>>         </condition>
>>>> 
>>>> 
>>>> And I check the freeswitch log and see
>>>> 
>>>> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  AUTH_RESULT=OK)
>>>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  AUTH_RESULT=OK
>>>> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  credit_time=7199)
>>>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  credit_time=7199
>>>> EXECUTE sofia/external/07966677711 at 185.8.92.3 <mailto:sofia/external/07966677711 at 185.8.92.3> log(INFO  return_code=16)
>>>> 2017-07-02 17:28:12.885037 [INFO] mod_dptools.c:1724  return_code=16
>>>> 
>>>> 
>>>> But it fails to execute the bit within the condition.
>>>> 
>>>> 
>>>> Any Ideas of how to make this work?
>>>> 
>>>> Regards
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From deepikay at iiitd.ac.in  Tue Jul  4 16:31:55 2017
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Tue, 4 Jul 2017 22:01:55 +0530
Subject: [Freeswitch-users] Freeswitch registration error with GSM gateway
Message-ID: <CAM99_YkwCiD2htv8tfwzR6_3kWuf9xir9zB9ML=vJ1CMDp80Hg@mail.gmail.com>

Hi,

I have a VOIP-GSM 16 port Dinstar gateway installed. It has its local IP,
SIM inserted and receiving signals but somehow it is not getting registered
with Freeswitch as external profile.

I have tried all the things found at different sources such as :

https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo

<https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
http://www.voipon.co.uk/documents/DWG_with_FreeSwitch.pdf
<https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>

<https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
but no luck :(

*Error at console* - Failed Registration with status Request Timeout [408]


*SIP trace is as follows*:

Freeswitch Server IP - *192.168.2.71*
Gateway IP - *192.1.68.2.69*


SUBSCRIBE sip:dwg2008 at 192.168.2.71 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.69;branch=z9hG4bKf22aec329cf8392a03d1fb7ac5907373;rport
From: <sip:dwg2008 at 192.168.2.71>;tag=d2e191310d53ca480a99c32d072b6907
To: <sip:dwg2008 at 192.168.2.71>
Call-ID: 7901887bd57699805a51f44e8e162cd5 at 192.168.2.69
CSeq: 479122789 SUBSCRIBE
Contact: <sip:dwg2008 at 192.168.2.69>
Event: message-summary
Supported: eventlist
Expires: 1
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Max-Forwards: 70
Accept: application/simple-message-summary
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKXcrNrjD10j6BN
From: <sip:dwg2008 at 192.168.2.71>;tag=5umvaME1a8N6
To: <sip:dwg2008 at 192.168.2.71>;tag=8b79a39dab9333d980ae172b0ddf0b10
Call-ID: c2ceb8fb9e2c2c1bfdba005ce2f3d223 at 192.168.2.69
CSeq: 797846101 NOTIFY
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKyNHetDy4XUvyg
From: <sip:dwg2008 at 192.168.2.71>;tag=DaCYUMXLeOft
To: <sip:dwg2008 at 192.168.2.71>;tag=d2e191310d53ca480a99c32d072b6907
Call-ID: 7901887bd57699805a51f44e8e162cd5 at 192.168.2.69
CSeq: 797846103 NOTIFY
Content-Length: 0

REGISTER sip:192.168.2.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.69:5060
;branch=z9hG4bKac7ee3cb94dbc6021702a40d789e140c;rport
From: <sip:dwg2008 at 192.168.2.71>;tag=8c1a438aeba7d28784d1958a90940ec1
To: <sip:dwg2008 at 192.168.2.71>
Call-ID: f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69
CSeq: 479121941 REGISTER
Contact: <sip:dwg2008 at 192.168.2.69:5060>
Expires: 1800
User-Agent: DWG2000F Business  20231220
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Max-Forwards: 70
Content-Length: 0

REGISTER sip:192.168.2.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.69:5060
;branch=z9hG4bKc2d768329e8115ca788edb78700b7db5;rport
From: <sip:dwg2008 at 192.168.2.71>;tag=8c1a438aeba7d28784d1958a90940ec1
To: <sip:dwg2008 at 192.168.2.71>
Call-ID: f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69
CSeq: 479121942 REGISTER
Contact: <sip:dwg2008 at 192.168.2.69:5060>
Authorization: Digest username="dwg2008", realm="192.168.2.71",
nonce="c4248cb2-60a7-11e7-b9ec-236b0b6c342f", uri="sip:192.168.2.71",
response="c0b6a86fcb1553a891d79eaafc2951e5", algorithm=MD5,
cnonce="6ea20a9604ad6faa602fa2e5802c9d72", qop=auth, nc=00000001
Expires: 1800
User-Agent: DWG2000F Business  20231220
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
Max-Forwards: 70
Content-Length: 0



-- 
Regards
Deepika
https://www.iiitd.edu.in/~deepikay/
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From shaun.stokes at itec-support.co.uk  Tue Jul  4 19:08:32 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Tue, 4 Jul 2017 19:08:32 +0000
Subject: [Freeswitch-users] Limit
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86791EC@mbx-01.sysconfig.co.uk>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678F9A@mbx-01.sysconfig.co.uk>
 <6FD2F8B5BB72834E9939AEDF9FB802A901E8678FDB@mbx-01.sysconfig.co.uk>
 <CAFLo2QyxyBwcVDxkPyMgPE7KnDAV_ZnOU1KC-iU0ODLD2ryK1g@mail.gmail.com>
 <6FD2F8B5BB72834E9939AEDF9FB802A901E86791EC@mbx-01.sysconfig.co.uk>
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E867AF65@mbx-01.sysconfig.co.uk>

During testing I’ve come across another way for the limit to apply on B leg with-out transferring the call out of the LUA script.

We simply execute this line before the bridge:
session:execute("export","nolocal:execute_on_pre_answer=limit hash "..domain_name.."_bleg "..extension.." "..limit_max.." "..limit_destination);

This results in the limit being applied on B leg before the call is answered.

We continue to apply a separate limit on A leg as before, the above is be used as an additional measure to limit B leg in the event that extensions use features such as attended transfer to exceed their available channels.


From: Shaun Stokes
Sent: 30 June 2017 16:23
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: RE: [Freeswitch-users] Limit

I should also point out, the extensions are transferring to external PSTN numbers not local to FreeSWITCH.

I’ll give this a try and will provide feedback, we’ll need a new dialplan to bridge the gateway.

Thanks,
Shaun

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis Daniel Lucio Quiroz
Sent: 30 June 2017 12:45
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Limit

In your script, do a transfer instead a bridge. Let the dialplan deal with the bridge

Le 30 juin 2017 7:05 AM, "Shaun Stokes" <shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>> a écrit :
Hi All,

In our environment we restrict the number of current channels (using limit) per extension, however using attended transfer allows extensions to exceed the limit since leg a ends once the transfer completes which resets the limit to 0.

I believe the solution is to apply the limit on leg b, however leg b is initiated via a bridge (with-in our outbound LUA script).

How can we apply the limit to leg b, or is there a better solution?

Thanks,
Shaun
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]

Shaun Stokes - Infrastructure Analyst

T :

01453 700713

E :

shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>

W :

www.itec-support.co.uk<http://www.itec-support.co.uk>


Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
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FreeSWITCH-users mailing list
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______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
Shaun Stokes - Infrastructure Analyst

T :     01453 700713
E :     shaun.stokes at itec-support.co.uk
W :     www.itec-support.co.uk

Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.

______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
______________________________________________________________________
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From bipin at xbipin.com  Tue Jul  4 19:42:20 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Tue, 04 Jul 2017 23:42:20 +0400
Subject: [Freeswitch-users] Freeswitch registration error with GSM
	gateway
In-Reply-To: <CAM99_YkwCiD2htv8tfwzR6_3kWuf9xir9zB9ML=vJ1CMDp80Hg@mail.gmail.com>
References: <CAM99_YkwCiD2htv8tfwzR6_3kWuf9xir9zB9ML=vJ1CMDp80Hg@mail.gmail.com>
Message-ID: <15d0f1ff960.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Hi,

We use a matrix GSM gateway but there is no need for it to register to FS 
in any way as long as both are on same LAN. From FS we simply route the 
calls to matrix on it's IP for sim calls and any calls on the sim come to 
FS on internal profile, u could route it to external also, we have added 
the matrix IP in FS with IP authentication.




On July 4, 2017 8:35:22 PM Deepika Yadav <deepikay at iiitd.ac.in> wrote:

> Hi,
>
> I have a VOIP-GSM 16 port Dinstar gateway installed. It has its local IP,
> SIM inserted and receiving signals but somehow it is not getting registered
> with Freeswitch as external profile.
>
> I have tried all the things found at different sources such as :
>
> https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo
>
> <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
> http://www.voipon.co.uk/documents/DWG_with_FreeSwitch.pdf
> <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
>
> <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
> but no luck :(
>
> *Error at console* - Failed Registration with status Request Timeout [408]
>
>
> *SIP trace is as follows*:
>
> Freeswitch Server IP - *192.168.2.71*
> Gateway IP - *192.1.68.2.69*
>
>
> SUBSCRIBE sip:dwg2008 at 192.168.2.71 SIP/2.0
> Via: SIP/2.0/UDP
> 192.168.2.69;branch=z9hG4bKf22aec329cf8392a03d1fb7ac5907373;rport
> From: <sip:dwg2008 at 192.168.2.71>;tag=d2e191310d53ca480a99c32d072b6907
> To: <sip:dwg2008 at 192.168.2.71>
> Call-ID: 7901887bd57699805a51f44e8e162cd5 at 192.168.2.69
> CSeq: 479122789 SUBSCRIBE
> Contact: <sip:dwg2008 at 192.168.2.69>
> Event: message-summary
> Supported: eventlist
> Expires: 1
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
> Max-Forwards: 70
> Accept: application/simple-message-summary
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKXcrNrjD10j6BN
> From: <sip:dwg2008 at 192.168.2.71>;tag=5umvaME1a8N6
> To: <sip:dwg2008 at 192.168.2.71>;tag=8b79a39dab9333d980ae172b0ddf0b10
> Call-ID: c2ceb8fb9e2c2c1bfdba005ce2f3d223 at 192.168.2.69
> CSeq: 797846101 NOTIFY
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKyNHetDy4XUvyg
> From: <sip:dwg2008 at 192.168.2.71>;tag=DaCYUMXLeOft
> To: <sip:dwg2008 at 192.168.2.71>;tag=d2e191310d53ca480a99c32d072b6907
> Call-ID: 7901887bd57699805a51f44e8e162cd5 at 192.168.2.69
> CSeq: 797846103 NOTIFY
> Content-Length: 0
>
> REGISTER sip:192.168.2.71 SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.69:5060
> ;branch=z9hG4bKac7ee3cb94dbc6021702a40d789e140c;rport
> From: <sip:dwg2008 at 192.168.2.71>;tag=8c1a438aeba7d28784d1958a90940ec1
> To: <sip:dwg2008 at 192.168.2.71>
> Call-ID: f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69
> CSeq: 479121941 REGISTER
> Contact: <sip:dwg2008 at 192.168.2.69:5060>
> Expires: 1800
> User-Agent: DWG2000F Business  20231220
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
> Max-Forwards: 70
> Content-Length: 0
>
> REGISTER sip:192.168.2.71 SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.69:5060
> ;branch=z9hG4bKc2d768329e8115ca788edb78700b7db5;rport
> From: <sip:dwg2008 at 192.168.2.71>;tag=8c1a438aeba7d28784d1958a90940ec1
> To: <sip:dwg2008 at 192.168.2.71>
> Call-ID: f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69
> CSeq: 479121942 REGISTER
> Contact: <sip:dwg2008 at 192.168.2.69:5060>
> Authorization: Digest username="dwg2008", realm="192.168.2.71",
> nonce="c4248cb2-60a7-11e7-b9ec-236b0b6c342f", uri="sip:192.168.2.71",
> response="c0b6a86fcb1553a891d79eaafc2951e5", algorithm=MD5,
> cnonce="6ea20a9604ad6faa602fa2e5802c9d72", qop=auth, nc=00000001
> Expires: 1800
> User-Agent: DWG2000F Business  20231220
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
> Max-Forwards: 70
> Content-Length: 0
>
>
>
> --
> Regards
> Deepika
> https://www.iiitd.edu.in/~deepikay/
>
>
>
> ----------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mario_fs at mgtech.com  Tue Jul  4 19:42:57 2017
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 4 Jul 2017 12:42:57 -0700
Subject: [Freeswitch-users] Music on hold
In-Reply-To: <CAC7ttFEA92XB4A35ETxndS6xswKCdOLcwKLi9yLZruZC1576mQ@mail.gmail.com>
References: <CAC7ttFEA92XB4A35ETxndS6xswKCdOLcwKLi9yLZruZC1576mQ@mail.gmail.com>
Message-ID: <8B1CBEA1-0062-4151-96E0-3EC266176B1C@mgtech.com>

I use LUA to customize the ringback which includes hello messages and MOH files. You could do that with regular MOH as well. When the call comes in set a variable for the MOH filename.

> On Jul 1, 2017, at 5:44 PM, Gregor Nanger <gregor at infomedia.si> wrote:
> 
> Need some advice. 
> 
> I have multi tenant setup, driven with xml_curl. What is best approach to make music on hold configurable for every customer? I mean from resource point of view. Music files should be read from http, not file system.
> 
> Thank you, Gregor
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bipin at xbipin.com  Tue Jul  4 19:51:06 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Tue, 04 Jul 2017 23:51:06 +0400
Subject: [Freeswitch-users] Limit
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E867AF65@mbx-01.sysconfig.co.uk>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678F9A@mbx-01.sysconfig.co.uk>
 <6FD2F8B5BB72834E9939AEDF9FB802A901E8678FDB@mbx-01.sysconfig.co.uk>
 <CAFLo2QyxyBwcVDxkPyMgPE7KnDAV_ZnOU1KC-iU0ODLD2ryK1g@mail.gmail.com>
 <6FD2F8B5BB72834E9939AEDF9FB802A901E86791EC@mbx-01.sysconfig.co.uk>
 <6FD2F8B5BB72834E9939AEDF9FB802A901E867AF65@mbx-01.sysconfig.co.uk>
Message-ID: <15d0f280010.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Hi,

Recently we had a similar issue with limit and as you know it doesn't work 
across dialplan so we started using hash where we increment a counter for 
the caller and callee and once they hang up we decrement it and in transfer 
situation we call the attended transfer extension in features dialplan 
where we take the input of the extension to be transferred to then we call 
execute extension to parse the features dialplan again and then we 
increment the counter for the callee where call is to be transferred.

This way the counter for all parties is incremented properly and we use api 
hangup hook to decrement the hash value in the bridge statement so when 
that party hangs up the counter is automatically decremented.




On July 4, 2017 11:11:43 PM Shaun Stokes <shaun.stokes at itec-support.co.uk> 
wrote:

> During testing I’ve come across another way for the limit to apply on B leg 
> with-out transferring the call out of the LUA script.
>
> We simply execute this line before the bridge:
> session:execute("export","nolocal:execute_on_pre_answer=limit hash 
> "..domain_name.."_bleg "..extension.." "..limit_max.." "..limit_destination);
>
> This results in the limit being applied on B leg before the call is answered.
>
> We continue to apply a separate limit on A leg as before, the above is be 
> used as an additional measure to limit B leg in the event that extensions 
> use features such as attended transfer to exceed their available channels.
>
>
> From: Shaun Stokes
> Sent: 30 June 2017 16:23
> To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> Subject: RE: [Freeswitch-users] Limit
>
> I should also point out, the extensions are transferring to external PSTN 
> numbers not local to FreeSWITCH.
>
> I’ll give this a try and will provide feedback, we’ll need a new dialplan 
> to bridge the gateway.
>
> Thanks,
> Shaun
>
> From: FreeSWITCH-users 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis 
> Daniel Lucio Quiroz
> Sent: 30 June 2017 12:45
> To: FreeSWITCH Users Help 
> <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] Limit
>
> In your script, do a transfer instead a bridge. Let the dialplan deal with 
> the bridge
>
> Le 30 juin 2017 7:05 AM, "Shaun Stokes" 
> <shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>> a 
> écrit :
> Hi All,
>
> In our environment we restrict the number of current channels (using limit) 
> per extension, however using attended transfer allows extensions to exceed 
> the limit since leg a ends once the transfer completes which resets the 
> limit to 0.
>
> I believe the solution is to apply the limit on leg b, however leg b is 
> initiated via a bridge (with-in our outbound LUA script).
>
> How can we apply the limit to leg b, or is there a better solution?
>
> Thanks,
> Shaun
> [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
>
> Shaun Stokes - Infrastructure Analyst
>
> T :
>
> 01453 700713
>
> E :
>
> shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>
>
> W :
>
> www.itec-support.co.uk<http://www.itec-support.co.uk>
>
>
> Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, 
> Stroud, Gloucestershire GL5 3QF
> Company No. 06908001
>
> CONFIDENTIALITY NOTICE
> This communication and the information it contains are intended for the 
> person or organisation to which it is addressed. Its contents are 
> confidential and may be protected in law. Unauthorised use, copying or 
> disclosure of any of it may be unlawful. If you are not the intended 
> recipient, please contact us immediately.
> The contents of any attachments in this e-mail may contain software 
> viruses, which could damage your own computer system. While ITEC Support 
> has taken every reasonable precaution to minimise this risk, we cannot 
> accept liability for any damage which you sustain as a result of software 
> viruses. You should carry out your own virus checking procedure before 
> opening any attachment.
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus 
> Scanning Service.
> ______________________________________________________________________
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus 
> Scanning Service.
> ______________________________________________________________________
> [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
> Shaun Stokes - Infrastructure Analyst
>
> T :     01453 700713
> E :     shaun.stokes at itec-support.co.uk
> W :     www.itec-support.co.uk
>
> Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, 
> Stroud, Gloucestershire GL5 3QF
> Company No. 06908001
>
> CONFIDENTIALITY NOTICE
> This communication and the information it contains are intended for the 
> person or organisation to which it is addressed. Its contents are 
> confidential and may be protected in law. Unauthorised use, copying or 
> disclosure of any of it may be unlawful. If you are not the intended 
> recipient, please contact us immediately.
> The contents of any attachments in this e-mail may contain software 
> viruses, which could damage your own computer system. While ITEC Support 
> has taken every reasonable precaution to minimise this risk, we cannot 
> accept liability for any damage which you sustain as a result of software 
> viruses. You should carry out your own virus checking procedure before 
> opening any attachment.
>
> ______________________________________________________________________
> This message has been checked for all known viruses by  MessageLabs Virus 
> Scanning Service.
> ______________________________________________________________________
>
>
> ----------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gregor at infomedia.si  Tue Jul  4 20:01:05 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 04 Jul 2017 20:01:05 +0000
Subject: [Freeswitch-users] Music on hold
In-Reply-To: <8B1CBEA1-0062-4151-96E0-3EC266176B1C@mgtech.com>
References: <CAC7ttFEA92XB4A35ETxndS6xswKCdOLcwKLi9yLZruZC1576mQ@mail.gmail.com>
 <8B1CBEA1-0062-4151-96E0-3EC266176B1C@mgtech.com>
Message-ID: <CAC7ttFF=FAye6RsrB-pnq=-MFsnbw7r3P2BobFCoT-PBG+UFjA@mail.gmail.com>

Can you please give me example or link? Can I point to external mp3 file?
Does this mean that for each incoming call there would be own stream?

Thank you, Gregor

On Tue, Jul 4, 2017, 21:44 Mario G <mario_fs at mgtech.com> wrote:

> I use LUA to customize the ringback which includes hello messages and MOH
> files. You could do that with regular MOH as well. When the call comes in
> set a variable for the MOH filename.
>
> On Jul 1, 2017, at 5:44 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>
> Need some advice.
>
> I have multi tenant setup, driven with xml_curl. What is best approach to
> make music on hold configurable for every customer? I mean from resource
> point of view. Music files should be read from http, not file system.
>
> Thank you, Gregor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From brandon at cryy.com  Tue Jul  4 20:09:55 2017
From: brandon at cryy.com (Brandon Armstead)
Date: Tue, 04 Jul 2017 20:09:55 +0000
Subject: [Freeswitch-users] Music on hold
In-Reply-To: <CAC7ttFF=FAye6RsrB-pnq=-MFsnbw7r3P2BobFCoT-PBG+UFjA@mail.gmail.com>
References: <CAC7ttFEA92XB4A35ETxndS6xswKCdOLcwKLi9yLZruZC1576mQ@mail.gmail.com>
 <8B1CBEA1-0062-4151-96E0-3EC266176B1C@mgtech.com>
 <CAC7ttFF=FAye6RsrB-pnq=-MFsnbw7r3P2BobFCoT-PBG+UFjA@mail.gmail.com>
Message-ID: <CACr1DbD0ZN07KFJJ6PYxEF_L9wkYcS6VU4OAGJWJSZoA9ssXpw@mail.gmail.com>

Look at mod_http_cache I believe is the model name.

On Tue, Jul 4, 2017 at 1:02 PM Gregor Nanger <gregor at infomedia.si> wrote:

> Can you please give me example or link? Can I point to external mp3 file?
> Does this mean that for each incoming call there would be own stream?
>
> Thank you, Gregor
>
> On Tue, Jul 4, 2017, 21:44 Mario G <mario_fs at mgtech.com> wrote:
>
>> I use LUA to customize the ringback which includes hello messages and MOH
>> files. You could do that with regular MOH as well. When the call comes in
>> set a variable for the MOH filename.
>>
>> On Jul 1, 2017, at 5:44 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>>
>> Need some advice.
>>
>> I have multi tenant setup, driven with xml_curl. What is best approach to
>> make music on hold configurable for every customer? I mean from resource
>> point of view. Music files should be read from http, not file system.
>>
>> Thank you, Gregor
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
> • www.infomedia.si
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Sent from Gmail Mobile
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From gregor at infomedia.si  Tue Jul  4 20:52:13 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 4 Jul 2017 22:52:13 +0200
Subject: [Freeswitch-users] Music on hold
In-Reply-To: <CACr1DbD0ZN07KFJJ6PYxEF_L9wkYcS6VU4OAGJWJSZoA9ssXpw@mail.gmail.com>
References: <CAC7ttFEA92XB4A35ETxndS6xswKCdOLcwKLi9yLZruZC1576mQ@mail.gmail.com>
 <8B1CBEA1-0062-4151-96E0-3EC266176B1C@mgtech.com>
 <CAC7ttFF=FAye6RsrB-pnq=-MFsnbw7r3P2BobFCoT-PBG+UFjA@mail.gmail.com>
 <CACr1DbD0ZN07KFJJ6PYxEF_L9wkYcS6VU4OAGJWJSZoA9ssXpw@mail.gmail.com>
Message-ID: <CAC7ttFGwc_73zYpMQrNz2uYEcXuLKFRTvurFSQ0-j3Up7X6BVw@mail.gmail.com>

I see, Brandon. I can use http_cache, to copy files localy and on web
server where file is stored I can use max-age to limit how long will stay
in cache.

2017-07-04 22:09 GMT+02:00 Brandon Armstead <brandon at cryy.com>:

> Look at mod_http_cache I believe is the model name.
>
> On Tue, Jul 4, 2017 at 1:02 PM Gregor Nanger <gregor at infomedia.si> wrote:
>
>> Can you please give me example or link? Can I point to external mp3 file?
>> Does this mean that for each incoming call there would be own stream?
>>
>> Thank you, Gregor
>>
>> On Tue, Jul 4, 2017, 21:44 Mario G <mario_fs at mgtech.com> wrote:
>>
>>> I use LUA to customize the ringback which includes hello messages and
>>> MOH files. You could do that with regular MOH as well. When the call comes
>>> in set a variable for the MOH filename.
>>>
>>> On Jul 1, 2017, at 5:44 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>>>
>>> Need some advice.
>>>
>>> I have multi tenant setup, driven with xml_curl. What is best approach
>>> to make music on hold configurable for every customer? I mean from resource
>>> point of view. Music files should be read from http, not file system.
>>>
>>> Thank you, Gregor
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> --
>> Gregor Nanger
>>
>> *CTO*
>> t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
>> • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
>> • www.infomedia.si
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Sent from Gmail Mobile
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From deepikay at iiitd.ac.in  Wed Jul  5 05:49:03 2017
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Wed, 5 Jul 2017 11:19:03 +0530
Subject: [Freeswitch-users] Freeswitch registration error with GSM
	gateway
In-Reply-To: <15d0f1ff960.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
References: <CAM99_YkwCiD2htv8tfwzR6_3kWuf9xir9zB9ML=vJ1CMDp80Hg@mail.gmail.com>
 <15d0f1ff960.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
Message-ID: <CAM99_YkN6MYjr0xQRAm=zXgwS9=gmRBiJV4NoRWAg9_US+yz5w@mail.gmail.com>

Could please help me with the command:

currently I use the external SIP profile namely MySIP as follows:

*originate
sofia/gateway/MySIP/919716517818 &conference("xyz"+flags("unmute"))*


How should modify it to direct it to the gateway'IP which is having SIP
user ID and password?


On Jul 5, 2017 1:15 AM, "Bipin Patel" <bipin at xbipin.com> wrote:

Hi,

We use a matrix GSM gateway but there is no need for it to register to FS
in any way as long as both are on same LAN. From FS we simply route the
calls to matrix on it's IP for sim calls and any calls on the sim come to
FS on internal profile, u could route it to external also, we have added
the matrix IP in FS with IP authentication.

On July 4, 2017 8:35:22 PM Deepika Yadav <deepikay at iiitd.ac.in> wrote:

> Hi,
>
> I have a VOIP-GSM 16 port Dinstar gateway installed. It has its local IP,
> SIM inserted and receiving signals but somehow it is not getting registered
> with Freeswitch as external profile.
>
> I have tried all the things found at different sources such as :
>
> https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo
>
> <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
> http://www.voipon.co.uk/documents/DWG_with_FreeSwitch.pdf
> <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
>
> <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
> but no luck :(
>
> *Error at console* - Failed Registration with status Request Timeout [408]
>
>
> *SIP trace is as follows*:
>
> Freeswitch Server IP - *192.168.2.71*
> Gateway IP - *192.1.68.2.69*
>
>
> SUBSCRIBE sip:dwg2008 at 192.168.2.71 SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.69;branch=z9hG4bKf22
> aec329cf8392a03d1fb7ac5907373;rport
> From: <sip:dwg2008 at 192.168.2.71>;tag=d2e191310d53ca480a99c32d072b6907
> To: <sip:dwg2008 at 192.168.2.71>
> Call-ID: 7901887bd57699805a51f44e8e162cd5 at 192.168.2.69
> CSeq: 479122789 SUBSCRIBE
> Contact: <sip:dwg2008 at 192.168.2.69>
> Event: message-summary
> Supported: eventlist
> Expires: 1
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
> Max-Forwards: 70
> Accept: application/simple-message-summary
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKXcrNrjD10j6BN
> From: <sip:dwg2008 at 192.168.2.71>;tag=5umvaME1a8N6
> To: <sip:dwg2008 at 192.168.2.71>;tag=8b79a39dab9333d980ae172b0ddf0b10
> Call-ID: c2ceb8fb9e2c2c1bfdba005ce2f3d223 at 192.168.2.69
> CSeq: 797846101 NOTIFY
> Content-Length: 0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKyNHetDy4XUvyg
> From: <sip:dwg2008 at 192.168.2.71>;tag=DaCYUMXLeOft
> To: <sip:dwg2008 at 192.168.2.71>;tag=d2e191310d53ca480a99c32d072b6907
> Call-ID: 7901887bd57699805a51f44e8e162cd5 at 192.168.2.69
> CSeq: 797846103 NOTIFY
> Content-Length: 0
>
> REGISTER sip:192.168.2.71 SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.69:5060;branch=z9hG4
> bKac7ee3cb94dbc6021702a40d789e140c;rport
> From: <sip:dwg2008 at 192.168.2.71>;tag=8c1a438aeba7d28784d1958a90940ec1
> To: <sip:dwg2008 at 192.168.2.71>
> Call-ID: f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69
> CSeq: 479121941 REGISTER
> Contact: <sip:dwg2008 at 192.168.2.69:5060>
> Expires: 1800
> User-Agent: DWG2000F Business  20231220
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
> Max-Forwards: 70
> Content-Length: 0
>
> REGISTER sip:192.168.2.71 SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.69:5060;branch=z9hG4
> bKc2d768329e8115ca788edb78700b7db5;rport
> From: <sip:dwg2008 at 192.168.2.71>;tag=8c1a438aeba7d28784d1958a90940ec1
> To: <sip:dwg2008 at 192.168.2.71>
> Call-ID: f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69
> CSeq: 479121942 REGISTER
> Contact: <sip:dwg2008 at 192.168.2.69:5060>
> Authorization: Digest username="dwg2008", realm="192.168.2.71",
> nonce="c4248cb2-60a7-11e7-b9ec-236b0b6c342f", uri="sip:192.168.2.71",
> response="c0b6a86fcb1553a891d79eaafc2951e5", algorithm=MD5,
> cnonce="6ea20a9604ad6faa602fa2e5802c9d72", qop=auth, nc=00000001
> Expires: 1800
> User-Agent: DWG2000F Business  20231220
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
> Max-Forwards: 70
> Content-Length: 0
>
>
>
> --
> Regards
> Deepika
> https://www.iiitd.edu.in/~deepikay/
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From bipin at xbipin.com  Wed Jul  5 06:24:36 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Wed, 5 Jul 2017 10:24:36 +0400
Subject: [Freeswitch-users] Freeswitch registration error with GSM
	gateway
In-Reply-To: <CAM99_YkN6MYjr0xQRAm=zXgwS9=gmRBiJV4NoRWAg9_US+yz5w@mail.gmail.com>
References: <CAM99_YkwCiD2htv8tfwzR6_3kWuf9xir9zB9ML=vJ1CMDp80Hg@mail.gmail.com>
 <15d0f1ff960.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <CAM99_YkN6MYjr0xQRAm=zXgwS9=gmRBiJV4NoRWAg9_US+yz5w@mail.gmail.com>
Message-ID: <1d6c7722-aa7f-f271-925b-6a6cea001847@xbipin.com>

hi,

im not too good with originate commands but to send calls to a gateway 
IP without registering to it u can use the below

sofia/external/919716517818 at 192.168.2.69

and regarding adding id/pass for authentication u can add the below to 
the parameters

sip_auth_username=username,sip_auth_password=password


Regards,
Bipin


------------------------------------------------------------------------
-------- Original Message --------
Subject: Re: [Freeswitch-users] Freeswitch registration error with GSM 
gateway
From: Deepika Yadav <deepikay at iiitd.ac.in>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 7/5/2017, 9:49:03 AM
> Could please help me with the command:
>
> currently I use the external SIP profile namely MySIP as follows:
>
>     *originate
>     sofia/gateway/MySIP/919716517818 &conference("xyz"+flags("unmute"))*
>     *
>     *
>
>
> How should modify it to direct it to the gateway'IP which is having 
> SIP user ID and password?
>
>
> On Jul 5, 2017 1:15 AM, "Bipin Patel" <bipin at xbipin.com 
> <mailto:bipin at xbipin.com>> wrote:
>
>     Hi,
>
>     We use a matrix GSM gateway but there is no need for it to
>     register to FS in any way as long as both are on same LAN. From FS
>     we simply route the calls to matrix on it's IP for sim calls and
>     any calls on the sim come to FS on internal profile, u could route
>     it to external also, we have added the matrix IP in FS with IP
>     authentication.
>
>     On July 4, 2017 8:35:22 PM Deepika Yadav <deepikay at iiitd.ac.in
>     <mailto:deepikay at iiitd.ac.in>> wrote:
>
>>     Hi,
>>
>>     I have a VOIP-GSM 16 port Dinstar gateway installed. It has its
>>     local IP, SIM inserted and receiving signals but somehow it is
>>     not getting registered with Freeswitch as external profile.
>>
>>     I have tried all the things found at different sources such as :
>>
>>     https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo
>>     <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
>>
>>     <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
>>     http://www.voipon.co.uk/documents/DWG_with_FreeSwitch.pdf
>>     <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
>>
>>     <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
>>     but no luck :(
>>
>>     *Error at console* - Failed Registration with status Request
>>     Timeout [408]
>>
>>
>>     *SIP trace is as follows*:
>>
>>     Freeswitch Server IP - *192.168.2.71*
>>     Gateway IP - *192.1.68.2.69*
>>
>>
>>     SUBSCRIBE sip:dwg2008 at 192.168.2.71
>>     <mailto:sip%3Adwg2008 at 192.168.2.71> SIP/2.0
>>     Via: SIP/2.0/UDP
>>     192.168.2.69;branch=z9hG4bKf22aec329cf8392a03d1fb7ac5907373;rport
>>     From: <sip:dwg2008 at 192.168.2.71
>>     <mailto:sip%3Adwg2008 at 192.168.2.71>>;tag=d2e191310d53ca480a99c32d072b6907
>>     To: <sip:dwg2008 at 192.168.2.71 <mailto:sip%3Adwg2008 at 192.168.2.71>>
>>     Call-ID: 7901887bd57699805a51f44e8e162cd5 at 192.168.2.69
>>     <mailto:7901887bd57699805a51f44e8e162cd5 at 192.168.2.69>
>>     CSeq: 479122789 SUBSCRIBE
>>     Contact: <sip:dwg2008 at 192.168.2.69
>>     <mailto:sip%3Adwg2008 at 192.168.2.69>>
>>     Event: message-summary
>>     Supported: eventlist
>>     Expires: 1
>>     Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
>>     Max-Forwards: 70
>>     Accept: application/simple-message-summary
>>     Content-Length: 0
>>
>>     SIP/2.0 200 OK
>>     Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKXcrNrjD10j6BN
>>     From: <sip:dwg2008 at 192.168.2.71
>>     <mailto:sip%3Adwg2008 at 192.168.2.71>>;tag=5umvaME1a8N6
>>     To: <sip:dwg2008 at 192.168.2.71
>>     <mailto:sip%3Adwg2008 at 192.168.2.71>>;tag=8b79a39dab9333d980ae172b0ddf0b10
>>     Call-ID: c2ceb8fb9e2c2c1bfdba005ce2f3d223 at 192.168.2.69
>>     <mailto:c2ceb8fb9e2c2c1bfdba005ce2f3d223 at 192.168.2.69>
>>     CSeq: 797846101 NOTIFY
>>     Content-Length: 0
>>
>>     SIP/2.0 200 OK
>>     Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKyNHetDy4XUvyg
>>     From: <sip:dwg2008 at 192.168.2.71
>>     <mailto:sip%3Adwg2008 at 192.168.2.71>>;tag=DaCYUMXLeOft
>>     To: <sip:dwg2008 at 192.168.2.71
>>     <mailto:sip%3Adwg2008 at 192.168.2.71>>;tag=d2e191310d53ca480a99c32d072b6907
>>     Call-ID: 7901887bd57699805a51f44e8e162cd5 at 192.168.2.69
>>     <mailto:7901887bd57699805a51f44e8e162cd5 at 192.168.2.69>
>>     CSeq: 797846103 NOTIFY
>>     Content-Length: 0
>>
>>     REGISTER sip:192.168.2.71 SIP/2.0
>>     Via: SIP/2.0/UDP
>>     192.168.2.69:5060;branch=z9hG4bKac7ee3cb94dbc6021702a40d789e140c;rport
>>     From: <sip:dwg2008 at 192.168.2.71
>>     <mailto:sip%3Adwg2008 at 192.168.2.71>>;tag=8c1a438aeba7d28784d1958a90940ec1
>>     To: <sip:dwg2008 at 192.168.2.71 <mailto:sip%3Adwg2008 at 192.168.2.71>>
>>     Call-ID: f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69
>>     <mailto:f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69>
>>     CSeq: 479121941 REGISTER
>>     Contact: <sip:dwg2008 at 192.168.2.69:5060
>>     <http://sip:dwg2008 at 192.168.2.69:5060>>
>>     Expires: 1800
>>     User-Agent: DWG2000F Business  20231220
>>     Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
>>     Max-Forwards: 70
>>     Content-Length: 0
>>
>>     REGISTER sip:192.168.2.71 SIP/2.0
>>     Via: SIP/2.0/UDP
>>     192.168.2.69:5060;branch=z9hG4bKc2d768329e8115ca788edb78700b7db5;rport
>>     From: <sip:dwg2008 at 192.168.2.71
>>     <mailto:sip%3Adwg2008 at 192.168.2.71>>;tag=8c1a438aeba7d28784d1958a90940ec1
>>     To: <sip:dwg2008 at 192.168.2.71 <mailto:sip%3Adwg2008 at 192.168.2.71>>
>>     Call-ID: f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69
>>     <mailto:f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69>
>>     CSeq: 479121942 REGISTER
>>     Contact: <sip:dwg2008 at 192.168.2.69:5060
>>     <http://sip:dwg2008 at 192.168.2.69:5060>>
>>     Authorization: Digest username="dwg2008", realm="192.168.2.71",
>>     nonce="c4248cb2-60a7-11e7-b9ec-236b0b6c342f",
>>     uri="sip:192.168.2.71",
>>     response="c0b6a86fcb1553a891d79eaafc2951e5", algorithm=MD5,
>>     cnonce="6ea20a9604ad6faa602fa2e5802c9d72", qop=auth, nc=00000001
>>     Expires: 1800
>>     User-Agent: DWG2000F Business  20231220
>>     Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
>>     Max-Forwards: 70
>>     Content-Length: 0
>>
>>
>>
>>     -- 
>>     Regards
>>     Deepika
>>     https://www.iiitd.edu.in/~deepikay/
>>     <https://www.iiitd.edu.in/%7Edeepikay/>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting%40freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>     <http://www.freeswitchsolutions.com>
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users%40lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>     http://www.freeswitch.org
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From deepikay at iiitd.ac.in  Wed Jul  5 06:41:02 2017
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Wed, 5 Jul 2017 12:11:02 +0530
Subject: [Freeswitch-users] Freeswitch registration error with GSM
	gateway
In-Reply-To: <1d6c7722-aa7f-f271-925b-6a6cea001847@xbipin.com>
References: <CAM99_YkwCiD2htv8tfwzR6_3kWuf9xir9zB9ML=vJ1CMDp80Hg@mail.gmail.com>
 <15d0f1ff960.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <CAM99_YkN6MYjr0xQRAm=zXgwS9=gmRBiJV4NoRWAg9_US+yz5w@mail.gmail.com>
 <1d6c7722-aa7f-f271-925b-6a6cea001847@xbipin.com>
Message-ID: <CAM99_YkR3TPS+EoAgvdx4ARw4eeG7QEenTLEprnRCWEgFJNVpQ@mail.gmail.com>

Thanks, yes this way it worked :)

On Wed, Jul 5, 2017 at 11:54 AM, Bipin Patel <bipin at xbipin.com> wrote:

> hi,
>
> im not too good with originate commands but to send calls to a gateway IP
> without registering to it u can use the below
>
> sofia/external/919716517818 at 192.168.2.69
>
> and regarding adding id/pass for authentication u can add the below to the
> parameters
>
> sip_auth_username=username,sip_auth_password=password
>
>
> Regards,
> Bipin
>
>
> ------------------------------
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] Freeswitch registration error with GSM
> gateway
> From: Deepika Yadav <deepikay at iiitd.ac.in> <deepikay at iiitd.ac.in>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Date: 7/5/2017, 9:49:03 AM
>
> Could please help me with the command:
>
> currently I use the external SIP profile namely MySIP as follows:
>
> *originate
> sofia/gateway/MySIP/919716517818 &conference("xyz"+flags("unmute"))*
>
>
> How should modify it to direct it to the gateway'IP which is having SIP
> user ID and password?
>
>
> On Jul 5, 2017 1:15 AM, "Bipin Patel" <bipin at xbipin.com> wrote:
>
> Hi,
>
> We use a matrix GSM gateway but there is no need for it to register to FS
> in any way as long as both are on same LAN. From FS we simply route the
> calls to matrix on it's IP for sim calls and any calls on the sim come to
> FS on internal profile, u could route it to external also, we have added
> the matrix IP in FS with IP authentication.
>
> On July 4, 2017 8:35:22 PM Deepika Yadav <deepikay at iiitd.ac.in> wrote:
>
>> Hi,
>>
>> I have a VOIP-GSM 16 port Dinstar gateway installed. It has its local IP,
>> SIM inserted and receiving signals but somehow it is not getting registered
>> with Freeswitch as external profile.
>>
>> I have tried all the things found at different sources such as :
>>
>> https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo
>>
>> <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
>> http://www.voipon.co.uk/documents/DWG_with_FreeSwitch.pdf
>> <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
>>
>> <https://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo>
>> but no luck :(
>>
>> *Error at console* - Failed Registration with status Request Timeout
>> [408]
>>
>>
>> *SIP trace is as follows*:
>>
>> Freeswitch Server IP - *192.168.2.71*
>> Gateway IP - *192.1.68.2.69*
>>
>>
>> SUBSCRIBE sip:dwg2008 at 192.168.2.71 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.2.69;branch=z9hG4bKf22
>> aec329cf8392a03d1fb7ac5907373;rport
>> From: <sip:dwg2008 at 192.168.2.71>;tag=d2e191310d53ca480a99c32d072b6907
>> To: <sip:dwg2008 at 192.168.2.71>
>> Call-ID: 7901887bd57699805a51f44e8e162cd5 at 192.168.2.69
>> CSeq: 479122789 SUBSCRIBE
>> Contact: <sip:dwg2008 at 192.168.2.69>
>> Event: message-summary
>> Supported: eventlist
>> Expires: 1
>> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
>> Max-Forwards: 70
>> Accept: application/simple-message-summary
>> Content-Length: 0
>>
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKXcrNrjD10j6BN
>> From: <sip:dwg2008 at 192.168.2.71>;tag=5umvaME1a8N6
>> To: <sip:dwg2008 at 192.168.2.71>;tag=8b79a39dab9333d980ae172b0ddf0b10
>> Call-ID: c2ceb8fb9e2c2c1bfdba005ce2f3d223 at 192.168.2.69
>> CSeq: 797846101 NOTIFY
>> Content-Length: 0
>>
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 192.168.2.71;rport=5060;branch=z9hG4bKyNHetDy4XUvyg
>> From: <sip:dwg2008 at 192.168.2.71>;tag=DaCYUMXLeOft
>> To: <sip:dwg2008 at 192.168.2.71>;tag=d2e191310d53ca480a99c32d072b6907
>> Call-ID: 7901887bd57699805a51f44e8e162cd5 at 192.168.2.69
>> CSeq: 797846103 NOTIFY
>> Content-Length: 0
>>
>> REGISTER sip:192.168.2.71 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.2.69:5060;branch=z9hG4
>> bKac7ee3cb94dbc6021702a40d789e140c;rport
>> From: <sip:dwg2008 at 192.168.2.71>;tag=8c1a438aeba7d28784d1958a90940ec1
>> To: <sip:dwg2008 at 192.168.2.71>
>> Call-ID: f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69
>> CSeq: 479121941 REGISTER
>> Contact: <sip:dwg2008 at 192.168.2.69:5060>
>> Expires: 1800
>> User-Agent: DWG2000F Business  20231220
>> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
>> Max-Forwards: 70
>> Content-Length: 0
>>
>> REGISTER sip:192.168.2.71 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.2.69:5060;branch=z9hG4
>> bKc2d768329e8115ca788edb78700b7db5;rport
>> From: <sip:dwg2008 at 192.168.2.71>;tag=8c1a438aeba7d28784d1958a90940ec1
>> To: <sip:dwg2008 at 192.168.2.71>
>> Call-ID: f58080d7fd4106bc3673e1295108eb09 at 192.168.2.69
>> CSeq: 479121942 REGISTER
>> Contact: <sip:dwg2008 at 192.168.2.69:5060>
>> Authorization: Digest username="dwg2008", realm="192.168.2.71",
>> nonce="c4248cb2-60a7-11e7-b9ec-236b0b6c342f", uri="sip:192.168.2.71",
>> response="c0b6a86fcb1553a891d79eaafc2951e5", algorithm=MD5,
>> cnonce="6ea20a9604ad6faa602fa2e5802c9d72", qop=auth, nc=00000001
>> Expires: 1800
>> User-Agent: DWG2000F Business  20231220
>> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, NOTIFY, REFER
>> Max-Forwards: 70
>> Content-Length: 0
>>
>>
>>
>> --
>> Regards
>> Deepika
>> https://www.iiitd.edu.in/~deepikay/
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>


-- 
Regards
Deepika
https://www.iiitd.edu.in/~deepikay/
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From mark.melling at savageminds.com  Tue Jul  4 21:55:15 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Tue, 04 Jul 2017 21:55:15 +0000
Subject: [Freeswitch-users] Verto: no inbound audio when verto client called
	from conference
Message-ID: <CAJp-f1ONFgNwhHs7MLuC+xfy8gEwKvOwJ8+CYq2DmcNB+xkp_g@mail.gmail.com>

Hi,

I have a dialplan where a user calls a number which results in them being
placed in a conference room, and an outbound called made to a verto client
when the conference is created, by defining in the dialplan:

    <action application="conference_set_auto_outcall" data="user/1000@
$${domain}"/>

The verto client (which is basically taken from the verto demo) rings and
is answered.

The original inbound caller can hear the verto client speaking, but the
verto client can't hear the the inbound caller (although the audio from the
inbound caller is getting to the conference room).

If I change the dialling order - so the verto client makes the inbound call
and the SIP client is dialled from the conference using the same technique
as before (conference_set_auto_outcall). Then everything appears to work
fine, i.e. each caller can here the other.

Any suggestions as to what might be the issue or steps I could take to help
identify the problem.

Thanks

Mark
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From nathan.port at gmail.com  Tue Jul  4 22:29:04 2017
From: nathan.port at gmail.com (Nate)
Date: Wed, 5 Jul 2017 10:29:04 +1200
Subject: [Freeswitch-users] Dialplan: Outbound route:
 destination_number: regex question
In-Reply-To: <478D2D4F-2967-4A18-BFAC-5AD1C35D0E47@vallimamod.org>
References: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>
 <478D2D4F-2967-4A18-BFAC-5AD1C35D0E47@vallimamod.org>
Message-ID: <CAB+hG+iEqsnnCHcAdrGm2uXSiubJcJ56CwsVot2FEuOK5MSq4g@mail.gmail.com>

Ah many thanks for that.
Right now I am just trying to isolate the issue that is preventing
inbound/outbound calls.
I have another closed voip phone system here in the office that can reach
my mobile phone with the string 022 333 4444.

But when I tried using the regex "^(022\d{7})$" without quotes for the
outbound destination_number and now get a new error stating:
#487 Missed Call.

Now checking the logs to see if there is anything else I can find in there.

On Mon, Jul 3, 2017 at 11:44 PM, Vallimamod Abdullah <vma at vallimamod.org>
wrote:

> Hi,
>
> IMHO, instead of trying to find a single complicated regex working in all
> cases, a more readable way is to use condition regex="any" to stack all
> your possible matching regexes and let freeswitch try them one by one:
>
> <condition regex="any">
>   <regex field="destination_number" expression="^(\+6422\d{7})$"/>
>   <regex field="destination_number" expression="^(022\d{7})$"/>
>   <regex field="destination_number" expression="^(\d{7})$"/>
>   <action ...>
> </condition>
>
> Hope this helps.
>
> Best Regards,
> --
> Vallimamod Abdullah
> SIP Solutions
> vma at sipsolutions.fr
> .
>
>
> > On 3 Jul 2017, at 05:24, Nate <nathan.port at gmail.com> wrote:
> >
> > Hi Roy, many thanks for responding.
> >
> > I am simply trying to finishing configuring the server to make/receive
> calls from the PSTN.
> >
> > My reference to the regex is from trying to configure an outbound route.
> >
> > Can anyone offer any advice on a regex that will work for me to reach
> any of the following phone numbers:
> >
> > International: +64 22 333 4444
> > non-local: 022 333 4444
> > local: 333 4444
> >
> > From the number assigned to my account? +64 9 xxx yyyy
> >
> > Many many thanks!
> >
> > Nate.
> >
> > On Sat, Jul 1, 2017 at 9:17 AM, <royj at yandex.ru> wrote:
> > Not quite clear. What are you trying to achieve?
> >
> > 30.06.2017, 11:11, "Nate" <nathan.port at gmail.com>:
> >> Good day/evening everyone,
> >>
> >> Apologies for not being able to figure this out on my own. I've been
> searching and trying for several days to get inbound/outbound working but
> have yet to see success.
> >>
> >> At this stage I need help determining a proper regex expression for
> handling New Zealand phone numbers.
> >>
> >> For instance, there are three different ways of expressing numbers here
> in NZ:
> >>
> >> International: +64 22 333 4444
> >> non-local: 022 333 4444
> >> local: 333 4444
> >>
> >> A couple questions related to the SIP proxy:
> >>
> >> If the SIP proxy is located in Hong Kong, but the phone number is a New
> Zealand number, does the location of the proxy have any impact on the
> number of characters in the string for inbound/outbound calls?
> >>
> >> Again, apologies for the rudimentary nature of these questions, but
> having nearly exhausted all other options (docs, searches, IRC), I am now
> spending a large amount of time guessing and trial and error without any
> progress.
> >>
> >> Many thanks for any feedback at all.
> >>
> >> Nate
> >> ,
> >> ____________________________________________________________
> _____________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
> options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nathan.port at gmail.com  Tue Jul  4 23:25:01 2017
From: nathan.port at gmail.com (Nate)
Date: Wed, 5 Jul 2017 11:25:01 +1200
Subject: [Freeswitch-users] Dialplan: Outbound route:
 destination_number: regex question
In-Reply-To: <CAB+hG+iEqsnnCHcAdrGm2uXSiubJcJ56CwsVot2FEuOK5MSq4g@mail.gmail.com>
References: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>
 <478D2D4F-2967-4A18-BFAC-5AD1C35D0E47@vallimamod.org>
 <CAB+hG+iEqsnnCHcAdrGm2uXSiubJcJ56CwsVot2FEuOK5MSq4g@mail.gmail.com>
Message-ID: <CAB+hG+j0h_UABGKa08jwpzSfOS5b5nw+rLH86vuYPtM4N5evQQ@mail.gmail.com>

So I just created a blank log and performed a test call to my mobile and
came up with the following snippet from the log. What stands out to me is
the declaration that "IP 192.168.1.92 Rejected by acl "domains"." But I
wonder if that is just normal as the system is configured to auth by
credentials rather than IP address.

Can anyone please comment on whether there is anything apparently wrong
here?

 [DEBUG] sofia.c:9837 sofia/internal/200 at 192.168.1.13 receiving invite from
192.168.1.92:6000 version: 1.6.18 -35-6e79667 64bit
 [DEBUG] sofia.c:10008 IP 192.168.1.92 Rejected by acl "domains". Falling
back to Digest auth.
 [DEBUG] sofia.c:2334 detaching session 1ea84eeb-e5ba-49c8-83dc-c479eedd6f1c
 [WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile
'internal' for [0223334444 at 192.168.1.13] from ip 192.168.1.92
 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/200 at 192.168.1.13)
State NEW
 [DEBUG] sofia.c:2442 Re-attaching to session
1ea84eeb-e5ba-49c8-83dc-c479eedd6f1c
 [DEBUG] sofia.c:9837 sofia/internal/200 at 192.168.1.13 receiving invite from
192.168.1.92:6000 version: 1.6.18 -35-6e79667 64bit
 [DEBUG] sofia.c:10008 IP 192.168.1.92 Rejected by acl "domains". Falling
back to Digest auth.
 [DEBUG] sofia.c:7048 Channel sofia/internal/200 at 192.168.1.13 entering
state [received][100]


I also noted the following snippet from the log where the call finally
fails which I hope someone might also be able to identify where I may have
something configured incorrectly:

 [DEBUG] mod_sofia.c:143 sofia/external/0223334444 SOFIA ROUTING
 [DEBUG] switch_ivr_originate.c:67 (sofia/external/0223334444) State Change
CS_ROUTING -> CS_CONSUME_MEDIA
 [DEBUG] switch_core_state_machine.c:643 (sofia/external/0223334444) State
ROUTING going to sleep
 [DEBUG] switch_core_state_machine.c:584 (sofia/external/0223334444)
Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 10)
 [DEBUG] switch_core_state_machine.c:662 (sofia/external/0223334444) State
CONSUME_MEDIA
 [DEBUG] switch_core_state_machine.c:662 (sofia/external/0223334444) State
CONSUME_MEDIA going to sleep
 [DEBUG] sofia.c:7048 Channel sofia/external/0223334444 entering state
[calling][0]
 [DEBUG] sofia.c:7048 Channel sofia/external/0223334444 entering state
[terminated][487]
 [NOTICE] sofia.c:8237 Hangup sofia/external/0223334444 [CS_CONSUME_MEDIA]
[ORIGINATOR_CANCEL]
 [DEBUG] switch_core_state_machine.c:584 (sofia/external/0223334444)
Running State Change CS_HANGUP (Cur 2 Tot 10)
 [DEBUG] switch_core_state_machine.c:850 (sofia/external/0223334444)
Callstate Change DOWN -> HANGUP
 [DEBUG] switch_core_state_machine.c:852 (sofia/external/0223334444) State
HANGUP
 [DEBUG] mod_sofia.c:438 Channel sofia/external/0223334444 hanging up,
cause: ORIGINATOR_CANCEL
 [DEBUG] switch_core_state_machine.c:60 sofia/external/0223334444 Standard
HANGUP, cause: ORIGINATOR_CANCEL
 [DEBUG] switch_core_state_machine.c:852 (sofia/external/0223334444) State
HANGUP going to sleep
 [DEBUG] switch_core_state_machine.c:619 (sofia/external/0223334444) State
Change CS_HANGUP -> CS_REPORTING
 [DEBUG] switch_core_state_machine.c:584 (sofia/external/0223334444)
Running State Change CS_REPORTING (Cur 2 Tot 10)
 [DEBUG] switch_core_state_machine.c:938 (sofia/external/0223334444) State
REPORTING
 [DEBUG] switch_core_state_machine.c:174 sofia/external/0223334444 Standard
REPORTING, cause: ORIGINATOR_CANCEL
 [DEBUG] switch_core_state_machine.c:938 (sofia/external/0223334444) State
REPORTING going to sleep
 [DEBUG] switch_core_state_machine.c:610 (sofia/external/0223334444) State
Change CS_REPORTING -> CS_DESTROY
 [DEBUG] switch_core_session.c:1664 Session 10 (sofia/external/0223334444)
Locked, Waiting on external entities
 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 487
[ORIGINATOR_CANCEL]
 [INFO] mod_dptools.c:3418 Originate Failed.  Cause: ORIGINATOR_CANCEL


On Wed, Jul 5, 2017 at 10:29 AM, Nate <nathan.port at gmail.com> wrote:

> Ah many thanks for that.
> Right now I am just trying to isolate the issue that is preventing
> inbound/outbound calls.
> I have another closed voip phone system here in the office that can reach
> my mobile phone with the string 022 333 4444.
>
> But when I tried using the regex "^(022\d{7})$" without quotes for the
> outbound destination_number and now get a new error stating:
> #487 Missed Call.
>
> Now checking the logs to see if there is anything else I can find in there.
>
> On Mon, Jul 3, 2017 at 11:44 PM, Vallimamod Abdullah <vma at vallimamod.org>
> wrote:
>
>> Hi,
>>
>> IMHO, instead of trying to find a single complicated regex working in all
>> cases, a more readable way is to use condition regex="any" to stack all
>> your possible matching regexes and let freeswitch try them one by one:
>>
>> <condition regex="any">
>>   <regex field="destination_number" expression="^(\+6422\d{7})$"/>
>>   <regex field="destination_number" expression="^(022\d{7})$"/>
>>   <regex field="destination_number" expression="^(\d{7})$"/>
>>   <action ...>
>> </condition>
>>
>> Hope this helps.
>>
>> Best Regards,
>> --
>> Vallimamod Abdullah
>> SIP Solutions
>> vma at sipsolutions.fr
>> .
>>
>>
>> > On 3 Jul 2017, at 05:24, Nate <nathan.port at gmail.com> wrote:
>> >
>> > Hi Roy, many thanks for responding.
>> >
>> > I am simply trying to finishing configuring the server to make/receive
>> calls from the PSTN.
>> >
>> > My reference to the regex is from trying to configure an outbound route.
>> >
>> > Can anyone offer any advice on a regex that will work for me to reach
>> any of the following phone numbers:
>> >
>> > International: +64 22 333 4444
>> > non-local: 022 333 4444
>> > local: 333 4444
>> >
>> > From the number assigned to my account? +64 9 xxx yyyy
>> >
>> > Many many thanks!
>> >
>> > Nate.
>> >
>> > On Sat, Jul 1, 2017 at 9:17 AM, <royj at yandex.ru> wrote:
>> > Not quite clear. What are you trying to achieve?
>> >
>> > 30.06.2017, 11:11, "Nate" <nathan.port at gmail.com>:
>> >> Good day/evening everyone,
>> >>
>> >> Apologies for not being able to figure this out on my own. I've been
>> searching and trying for several days to get inbound/outbound working but
>> have yet to see success.
>> >>
>> >> At this stage I need help determining a proper regex expression for
>> handling New Zealand phone numbers.
>> >>
>> >> For instance, there are three different ways of expressing numbers
>> here in NZ:
>> >>
>> >> International: +64 22 333 4444
>> >> non-local: 022 333 4444
>> >> local: 333 4444
>> >>
>> >> A couple questions related to the SIP proxy:
>> >>
>> >> If the SIP proxy is located in Hong Kong, but the phone number is a
>> New Zealand number, does the location of the proxy have any impact on the
>> number of characters in the string for inbound/outbound calls?
>> >>
>> >> Again, apologies for the rudimentary nature of these questions, but
>> having nearly exhausted all other options (docs, searches, IRC), I am now
>> spending a large amount of time guessing and trial and error without any
>> progress.
>> >>
>> >> Many thanks for any feedback at all.
>> >>
>> >> Nate
>> >> ,
>> >> ____________________________________________________________
>> _____________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>> freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> > ____________________________________________________________
>> _____________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>> freeswitch-users
>> > http://www.freeswitch.org
>> >
>> > ____________________________________________________________
>> _____________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>> freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From nathan.port at gmail.com  Tue Jul  4 23:35:04 2017
From: nathan.port at gmail.com (Nate)
Date: Wed, 5 Jul 2017 11:35:04 +1200
Subject: [Freeswitch-users] Dialplan: Outbound route:
 destination_number: regex question
In-Reply-To: <CAB+hG+j0h_UABGKa08jwpzSfOS5b5nw+rLH86vuYPtM4N5evQQ@mail.gmail.com>
References: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>
 <478D2D4F-2967-4A18-BFAC-5AD1C35D0E47@vallimamod.org>
 <CAB+hG+iEqsnnCHcAdrGm2uXSiubJcJ56CwsVot2FEuOK5MSq4g@mail.gmail.com>
 <CAB+hG+j0h_UABGKa08jwpzSfOS5b5nw+rLH86vuYPtM4N5evQQ@mail.gmail.com>
Message-ID: <CAB+hG+iCPEdxGX0UJJFD-+mjVWo8YCO7o03LL8-Vt6XxODh3HQ@mail.gmail.com>

Ah, I also just setup another gateway and watched the logs as it tried to
register and found the following entries that appear to indicate I don't
have the proper external IP addresses assigned somewhere.

This system is NAT'd with SIP forwarded:

2017-07-05 11:29:22.655729 [DEBUG] sofia.c:4450 rtp-ip [192.168.1.13]
2017-07-05 11:29:22.655729 [DEBUG] sofia.c:4450 sip-ip [192.168.1.13]
2017-07-05 11:29:22.655729 [DEBUG] sofia.c:4450 ext-rtp-ip [192.168.1.13]
2017-07-05 11:29:22.655729 [DEBUG] sofia.c:4450 ext-sip-ip [192.168.1.13]


On Wed, Jul 5, 2017 at 11:25 AM, Nate <nathan.port at gmail.com> wrote:

> So I just created a blank log and performed a test call to my mobile and
> came up with the following snippet from the log. What stands out to me is
> the declaration that "IP 192.168.1.92 Rejected by acl "domains"." But I
> wonder if that is just normal as the system is configured to auth by
> credentials rather than IP address.
>
> Can anyone please comment on whether there is anything apparently wrong
> here?
>
>  [DEBUG] sofia.c:9837 sofia/internal/200 at 192.168.1.13 receiving invite
> from 192.168.1.92:6000 version: 1.6.18 -35-6e79667 64bit
>  [DEBUG] sofia.c:10008 IP 192.168.1.92 Rejected by acl "domains". Falling
> back to Digest auth.
>  [DEBUG] sofia.c:2334 detaching session 1ea84eeb-e5ba-49c8-83dc-
> c479eedd6f1c
>  [WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile
> 'internal' for [0223334444 at 192.168.1.13] from ip 192.168.1.92
>  [DEBUG] switch_core_state_machine.c:603 (sofia/internal/200 at 192.168.1.13)
> State NEW
>  [DEBUG] sofia.c:2442 Re-attaching to session 1ea84eeb-e5ba-49c8-83dc-
> c479eedd6f1c
>  [DEBUG] sofia.c:9837 sofia/internal/200 at 192.168.1.13 receiving invite
> from 192.168.1.92:6000 version: 1.6.18 -35-6e79667 64bit
>  [DEBUG] sofia.c:10008 IP 192.168.1.92 Rejected by acl "domains". Falling
> back to Digest auth.
>  [DEBUG] sofia.c:7048 Channel sofia/internal/200 at 192.168.1.13 entering
> state [received][100]
>
>
> I also noted the following snippet from the log where the call finally
> fails which I hope someone might also be able to identify where I may have
> something configured incorrectly:
>
>  [DEBUG] mod_sofia.c:143 sofia/external/0223334444 SOFIA ROUTING
>  [DEBUG] switch_ivr_originate.c:67 (sofia/external/0223334444) State
> Change CS_ROUTING -> CS_CONSUME_MEDIA
>  [DEBUG] switch_core_state_machine.c:643 (sofia/external/0223334444)
> State ROUTING going to sleep
>  [DEBUG] switch_core_state_machine.c:584 (sofia/external/0223334444)
> Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 10)
>  [DEBUG] switch_core_state_machine.c:662 (sofia/external/0223334444)
> State CONSUME_MEDIA
>  [DEBUG] switch_core_state_machine.c:662 (sofia/external/0223334444)
> State CONSUME_MEDIA going to sleep
>  [DEBUG] sofia.c:7048 Channel sofia/external/0223334444 entering state
> [calling][0]
>  [DEBUG] sofia.c:7048 Channel sofia/external/0223334444 entering state
> [terminated][487]
>  [NOTICE] sofia.c:8237 Hangup sofia/external/0223334444 [CS_CONSUME_MEDIA]
> [ORIGINATOR_CANCEL]
>  [DEBUG] switch_core_state_machine.c:584 (sofia/external/0223334444)
> Running State Change CS_HANGUP (Cur 2 Tot 10)
>  [DEBUG] switch_core_state_machine.c:850 (sofia/external/0223334444)
> Callstate Change DOWN -> HANGUP
>  [DEBUG] switch_core_state_machine.c:852 (sofia/external/0223334444)
> State HANGUP
>  [DEBUG] mod_sofia.c:438 Channel sofia/external/0223334444 hanging up,
> cause: ORIGINATOR_CANCEL
>  [DEBUG] switch_core_state_machine.c:60 sofia/external/0223334444 Standard
> HANGUP, cause: ORIGINATOR_CANCEL
>  [DEBUG] switch_core_state_machine.c:852 (sofia/external/0223334444)
> State HANGUP going to sleep
>  [DEBUG] switch_core_state_machine.c:619 (sofia/external/0223334444)
> State Change CS_HANGUP -> CS_REPORTING
>  [DEBUG] switch_core_state_machine.c:584 (sofia/external/0223334444)
> Running State Change CS_REPORTING (Cur 2 Tot 10)
>  [DEBUG] switch_core_state_machine.c:938 (sofia/external/0223334444)
> State REPORTING
>  [DEBUG] switch_core_state_machine.c:174 sofia/external/0223334444
> Standard REPORTING, cause: ORIGINATOR_CANCEL
>  [DEBUG] switch_core_state_machine.c:938 (sofia/external/0223334444)
> State REPORTING going to sleep
>  [DEBUG] switch_core_state_machine.c:610 (sofia/external/0223334444)
> State Change CS_REPORTING -> CS_DESTROY
>  [DEBUG] switch_core_session.c:1664 Session 10 (sofia/external/0223334444)
> Locked, Waiting on external entities
>  [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause:
> 487 [ORIGINATOR_CANCEL]
>  [INFO] mod_dptools.c:3418 Originate Failed.  Cause: ORIGINATOR_CANCEL
>
>
> On Wed, Jul 5, 2017 at 10:29 AM, Nate <nathan.port at gmail.com> wrote:
>
>> Ah many thanks for that.
>> Right now I am just trying to isolate the issue that is preventing
>> inbound/outbound calls.
>> I have another closed voip phone system here in the office that can reach
>> my mobile phone with the string 022 333 4444.
>>
>> But when I tried using the regex "^(022\d{7})$" without quotes for the
>> outbound destination_number and now get a new error stating:
>> #487 Missed Call.
>>
>> Now checking the logs to see if there is anything else I can find in
>> there.
>>
>> On Mon, Jul 3, 2017 at 11:44 PM, Vallimamod Abdullah <vma at vallimamod.org>
>> wrote:
>>
>>> Hi,
>>>
>>> IMHO, instead of trying to find a single complicated regex working in
>>> all cases, a more readable way is to use condition regex="any" to stack all
>>> your possible matching regexes and let freeswitch try them one by one:
>>>
>>> <condition regex="any">
>>>   <regex field="destination_number" expression="^(\+6422\d{7})$"/>
>>>   <regex field="destination_number" expression="^(022\d{7})$"/>
>>>   <regex field="destination_number" expression="^(\d{7})$"/>
>>>   <action ...>
>>> </condition>
>>>
>>> Hope this helps.
>>>
>>> Best Regards,
>>> --
>>> Vallimamod Abdullah
>>> SIP Solutions
>>> vma at sipsolutions.fr
>>> .
>>>
>>>
>>> > On 3 Jul 2017, at 05:24, Nate <nathan.port at gmail.com> wrote:
>>> >
>>> > Hi Roy, many thanks for responding.
>>> >
>>> > I am simply trying to finishing configuring the server to make/receive
>>> calls from the PSTN.
>>> >
>>> > My reference to the regex is from trying to configure an outbound
>>> route.
>>> >
>>> > Can anyone offer any advice on a regex that will work for me to reach
>>> any of the following phone numbers:
>>> >
>>> > International: +64 22 333 4444
>>> > non-local: 022 333 4444
>>> > local: 333 4444
>>> >
>>> > From the number assigned to my account? +64 9 xxx yyyy
>>> >
>>> > Many many thanks!
>>> >
>>> > Nate.
>>> >
>>> > On Sat, Jul 1, 2017 at 9:17 AM, <royj at yandex.ru> wrote:
>>> > Not quite clear. What are you trying to achieve?
>>> >
>>> > 30.06.2017, 11:11, "Nate" <nathan.port at gmail.com>:
>>> >> Good day/evening everyone,
>>> >>
>>> >> Apologies for not being able to figure this out on my own. I've been
>>> searching and trying for several days to get inbound/outbound working but
>>> have yet to see success.
>>> >>
>>> >> At this stage I need help determining a proper regex expression for
>>> handling New Zealand phone numbers.
>>> >>
>>> >> For instance, there are three different ways of expressing numbers
>>> here in NZ:
>>> >>
>>> >> International: +64 22 333 4444
>>> >> non-local: 022 333 4444
>>> >> local: 333 4444
>>> >>
>>> >> A couple questions related to the SIP proxy:
>>> >>
>>> >> If the SIP proxy is located in Hong Kong, but the phone number is a
>>> New Zealand number, does the location of the proxy have any impact on the
>>> number of characters in the string for inbound/outbound calls?
>>> >>
>>> >> Again, apologies for the rudimentary nature of these questions, but
>>> having nearly exhausted all other options (docs, searches, IRC), I am now
>>> spending a large amount of time guessing and trial and error without any
>>> progress.
>>> >>
>>> >> Many thanks for any feedback at all.
>>> >>
>>> >> Nate
>>> >> ,
>>> >> ____________________________________________________________
>>> _____________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://confluence.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>> switch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> > ____________________________________________________________
>>> _____________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>> switch-users
>>> > http://www.freeswitch.org
>>> >
>>> > ____________________________________________________________
>>> _____________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>> switch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From gmaruzz at gmail.com  Wed Jul  5 07:55:39 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 5 Jul 2017 09:55:39 +0200
Subject: [Freeswitch-users] Verto: no inbound audio when verto client
 called from conference
In-Reply-To: <CAJp-f1ONFgNwhHs7MLuC+xfy8gEwKvOwJ8+CYq2DmcNB+xkp_g@mail.gmail.com>
References: <CAJp-f1ONFgNwhHs7MLuC+xfy8gEwKvOwJ8+CYq2DmcNB+xkp_g@mail.gmail.com>
Message-ID: <CALXCt0qojw_FEP=ZveB8_=DGUSEA_BCi2gYwUxu1yYWos1nASA@mail.gmail.com>

On 4 July 2017 at 23:55, Mark Melling <mark.melling at savageminds.com> wrote:

Hi,
>
> I have a dialplan where a user calls a number which results in them being
> placed in a conference room, and an outbound called made to a verto client
> when the conference is created, by defining in the dialplan:
>
>     <action application="conference_set_auto_outcall" data="user/1000@
> $${domain}"/>
>
> The verto client (which is basically taken from the verto demo) rings and
> is answered.
>
> The original inbound caller can hear the verto client speaking, but the
> verto client can't hear the the inbound caller (although the audio from the
> inbound caller is getting to the conference room).
>
> If I change the dialling order - so the verto client makes the inbound
> call and the SIP client is dialled from the conference using the same
> technique as before (conference_set_auto_outcall). Then everything appears
> to work fine, i.e. each caller can here the other.
>
> Any suggestions as to what might be the issue or steps I could take to
> help identify the problem.
>


Maybe this is because verto (webrtc) takes time to establish audio because
of stun, etc etc

Try this: instead of generating autocall from inside conference (eg instead
of using autocall),originate call to user, wait for her to answer, then
(after she answer) sleep for 2 seconds, then transfer her to the conf

-giovanni



>
> Thanks
>
> Mark
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From david.villasmil.work at gmail.com  Wed Jul  5 12:07:12 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 05 Jul 2017 12:07:12 +0000
Subject: [Freeswitch-users] Dialplan: Outbound route:
 destination_number: regex question
In-Reply-To: <CAB+hG+iCPEdxGX0UJJFD-+mjVWo8YCO7o03LL8-Vt6XxODh3HQ@mail.gmail.com>
References: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>
 <478D2D4F-2967-4A18-BFAC-5AD1C35D0E47@vallimamod.org>
 <CAB+hG+iEqsnnCHcAdrGm2uXSiubJcJ56CwsVot2FEuOK5MSq4g@mail.gmail.com>
 <CAB+hG+j0h_UABGKa08jwpzSfOS5b5nw+rLH86vuYPtM4N5evQQ@mail.gmail.com>
 <CAB+hG+iCPEdxGX0UJJFD-+mjVWo8YCO7o03LL8-Vt6XxODh3HQ@mail.gmail.com>
Message-ID: <CAFGRPVqPCiSCevF=Rkr+S_XHPz_07yOOrwMgAVJf211JjSDEOw@mail.gmail.com>

Fs will fallback to Digest from ACL when the source ip is not allowed, this
is normal.

Looking at your log, it doesn't seem like the client is actually responding
to the challenge.

Can you take a trace and look at it? You should see the challenge and then
an INVITE with credentials.
On Wed, Jul 5, 2017 at 9:48 AM Nate <nathan.port at gmail.com> wrote:

> Ah, I also just setup another gateway and watched the logs as it tried to
> register and found the following entries that appear to indicate I don't
> have the proper external IP addresses assigned somewhere.
>
> This system is NAT'd with SIP forwarded:
>
> 2017-07-05 11:29:22.655729 [DEBUG] sofia.c:4450 rtp-ip [192.168.1.13]
> 2017-07-05 11:29:22.655729 [DEBUG] sofia.c:4450 sip-ip [192.168.1.13]
> 2017-07-05 11:29:22.655729 [DEBUG] sofia.c:4450 ext-rtp-ip [192.168.1.13]
> 2017-07-05 11:29:22.655729 [DEBUG] sofia.c:4450 ext-sip-ip [192.168.1.13]
>
>
> On Wed, Jul 5, 2017 at 11:25 AM, Nate <nathan.port at gmail.com> wrote:
>
>> So I just created a blank log and performed a test call to my mobile and
>> came up with the following snippet from the log. What stands out to me is
>> the declaration that "IP 192.168.1.92 Rejected by acl "domains"." But I
>> wonder if that is just normal as the system is configured to auth by
>> credentials rather than IP address.
>>
>> Can anyone please comment on whether there is anything apparently wrong
>> here?
>>
>>  [DEBUG] sofia.c:9837 sofia/internal/200 at 192.168.1.13 receiving invite
>> from 192.168.1.92:6000 version: 1.6.18 -35-6e79667 64bit
>>  [DEBUG] sofia.c:10008 IP 192.168.1.92 Rejected by acl "domains". Falling
>> back to Digest auth.
>>  [DEBUG] sofia.c:2334 detaching session
>> 1ea84eeb-e5ba-49c8-83dc-c479eedd6f1c
>>  [WARNING] sofia_reg.c:1792 SIP auth challenge (INVITE) on sofia profile
>> 'internal' for [0223334444 at 192.168.1.13] from ip 192.168.1.92
>>  [DEBUG] switch_core_state_machine.c:603 (sofia/internal/200 at 192.168.1.13)
>> State NEW
>>  [DEBUG] sofia.c:2442 Re-attaching to session
>> 1ea84eeb-e5ba-49c8-83dc-c479eedd6f1c
>>  [DEBUG] sofia.c:9837 sofia/internal/200 at 192.168.1.13 receiving invite
>> from 192.168.1.92:6000 version: 1.6.18 -35-6e79667 64bit
>>  [DEBUG] sofia.c:10008 IP 192.168.1.92 Rejected by acl "domains". Falling
>> back to Digest auth.
>>  [DEBUG] sofia.c:7048 Channel sofia/internal/200 at 192.168.1.13 entering
>> state [received][100]
>>
>>
>> I also noted the following snippet from the log where the call finally
>> fails which I hope someone might also be able to identify where I may have
>> something configured incorrectly:
>>
>>  [DEBUG] mod_sofia.c:143 sofia/external/0223334444 SOFIA ROUTING
>>  [DEBUG] switch_ivr_originate.c:67 (sofia/external/0223334444) State
>> Change CS_ROUTING -> CS_CONSUME_MEDIA
>>  [DEBUG] switch_core_state_machine.c:643 (sofia/external/0223334444)
>> State ROUTING going to sleep
>>  [DEBUG] switch_core_state_machine.c:584 (sofia/external/0223334444)
>> Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 10)
>>  [DEBUG] switch_core_state_machine.c:662 (sofia/external/0223334444)
>> State CONSUME_MEDIA
>>  [DEBUG] switch_core_state_machine.c:662 (sofia/external/0223334444)
>> State CONSUME_MEDIA going to sleep
>>  [DEBUG] sofia.c:7048 Channel sofia/external/0223334444 entering state
>> [calling][0]
>>  [DEBUG] sofia.c:7048 Channel sofia/external/0223334444 entering state
>> [terminated][487]
>>  [NOTICE] sofia.c:8237 Hangup sofia/external/0223334444
>> [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>>  [DEBUG] switch_core_state_machine.c:584 (sofia/external/0223334444)
>> Running State Change CS_HANGUP (Cur 2 Tot 10)
>>  [DEBUG] switch_core_state_machine.c:850 (sofia/external/0223334444)
>> Callstate Change DOWN -> HANGUP
>>  [DEBUG] switch_core_state_machine.c:852 (sofia/external/0223334444)
>> State HANGUP
>>  [DEBUG] mod_sofia.c:438 Channel sofia/external/0223334444 hanging up,
>> cause: ORIGINATOR_CANCEL
>>  [DEBUG] switch_core_state_machine.c:60 sofia/external/0223334444
>> Standard HANGUP, cause: ORIGINATOR_CANCEL
>>  [DEBUG] switch_core_state_machine.c:852 (sofia/external/0223334444)
>> State HANGUP going to sleep
>>  [DEBUG] switch_core_state_machine.c:619 (sofia/external/0223334444)
>> State Change CS_HANGUP -> CS_REPORTING
>>  [DEBUG] switch_core_state_machine.c:584 (sofia/external/0223334444)
>> Running State Change CS_REPORTING (Cur 2 Tot 10)
>>  [DEBUG] switch_core_state_machine.c:938 (sofia/external/0223334444)
>> State REPORTING
>>  [DEBUG] switch_core_state_machine.c:174 sofia/external/0223334444
>> Standard REPORTING, cause: ORIGINATOR_CANCEL
>>  [DEBUG] switch_core_state_machine.c:938 (sofia/external/0223334444)
>> State REPORTING going to sleep
>>  [DEBUG] switch_core_state_machine.c:610 (sofia/external/0223334444)
>> State Change CS_REPORTING -> CS_DESTROY
>>  [DEBUG] switch_core_session.c:1664 Session 10
>> (sofia/external/0223334444) Locked, Waiting on external entities
>>  [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause:
>> 487 [ORIGINATOR_CANCEL]
>>  [INFO] mod_dptools.c:3418 Originate Failed.  Cause: ORIGINATOR_CANCEL
>>
>>
>> On Wed, Jul 5, 2017 at 10:29 AM, Nate <nathan.port at gmail.com> wrote:
>>
>>> Ah many thanks for that.
>>> Right now I am just trying to isolate the issue that is preventing
>>> inbound/outbound calls.
>>> I have another closed voip phone system here in the office that can
>>> reach my mobile phone with the string 022 333 4444.
>>>
>>> But when I tried using the regex "^(022\d{7})$" without quotes for the
>>> outbound destination_number and now get a new error stating:
>>> #487 Missed Call.
>>>
>>> Now checking the logs to see if there is anything else I can find in
>>> there.
>>>
>>> On Mon, Jul 3, 2017 at 11:44 PM, Vallimamod Abdullah <vma at vallimamod.org
>>> > wrote:
>>>
>>>> Hi,
>>>>
>>>> IMHO, instead of trying to find a single complicated regex working in
>>>> all cases, a more readable way is to use condition regex="any" to stack all
>>>> your possible matching regexes and let freeswitch try them one by one:
>>>>
>>>> <condition regex="any">
>>>>   <regex field="destination_number" expression="^(\+6422\d{7})$"/>
>>>>   <regex field="destination_number" expression="^(022\d{7})$"/>
>>>>   <regex field="destination_number" expression="^(\d{7})$"/>
>>>>   <action ...>
>>>> </condition>
>>>>
>>>> Hope this helps.
>>>>
>>>> Best Regards,
>>>> --
>>>> Vallimamod Abdullah
>>>> SIP Solutions
>>>> vma at sipsolutions.fr
>>>> .
>>>>
>>>>
>>>> > On 3 Jul 2017, at 05:24, Nate <nathan.port at gmail.com> wrote:
>>>> >
>>>> > Hi Roy, many thanks for responding.
>>>> >
>>>> > I am simply trying to finishing configuring the server to
>>>> make/receive calls from the PSTN.
>>>> >
>>>> > My reference to the regex is from trying to configure an outbound
>>>> route.
>>>> >
>>>> > Can anyone offer any advice on a regex that will work for me to reach
>>>> any of the following phone numbers:
>>>> >
>>>> > International: +64 22 333 4444
>>>> > non-local: 022 333 4444
>>>> > local: 333 4444
>>>> >
>>>> > From the number assigned to my account? +64 9 xxx yyyy
>>>> >
>>>> > Many many thanks!
>>>> >
>>>> > Nate.
>>>> >
>>>> > On Sat, Jul 1, 2017 at 9:17 AM, <royj at yandex.ru> wrote:
>>>> > Not quite clear. What are you trying to achieve?
>>>> >
>>>> > 30.06.2017, 11:11, "Nate" <nathan.port at gmail.com>:
>>>> >> Good day/evening everyone,
>>>> >>
>>>> >> Apologies for not being able to figure this out on my own. I've been
>>>> searching and trying for several days to get inbound/outbound working but
>>>> have yet to see success.
>>>> >>
>>>> >> At this stage I need help determining a proper regex expression for
>>>> handling New Zealand phone numbers.
>>>> >>
>>>> >> For instance, there are three different ways of expressing numbers
>>>> here in NZ:
>>>> >>
>>>> >> International: +64 22 333 4444
>>>> >> non-local: 022 333 4444
>>>> >> local: 333 4444
>>>> >>
>>>> >> A couple questions related to the SIP proxy:
>>>> >>
>>>> >> If the SIP proxy is located in Hong Kong, but the phone number is a
>>>> New Zealand number, does the location of the proxy have any impact on the
>>>> number of characters in the string for inbound/outbound calls?
>>>> >>
>>>> >> Again, apologies for the rudimentary nature of these questions, but
>>>> having nearly exhausted all other options (docs, searches, IRC), I am now
>>>> spending a large amount of time guessing and trial and error without any
>>>> progress.
>>>> >>
>>>> >> Many thanks for any feedback at all.
>>>> >>
>>>> >> Nate
>>>> >> ,
>>>> >>
>>>> _________________________________________________________________________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> consulting at freeswitch.org
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://confluence.freeswitch.org
>>>> >> http://www.cluecon.com
>>>> >>
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://confluence.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>> _________________________________________________________________________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://confluence.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From alex at digitalmail.com  Wed Jul  5 15:42:02 2017
From: alex at digitalmail.com (Alex Lake)
Date: Wed, 05 Jul 2017 15:42:02 +0000
Subject: [Freeswitch-users] Simple script for ringtone
Message-ID: <6bbe39c2-2af7-1111-7ec5-d5c3f6ddbba8@digitalmail.com>

I want to set up a number that just rings and never answers. 
Unfortunately my freeswitch is very rusty!

Can someone help me out with what must be a 2 or 3 line script? ;-)

Alex


From david.ponzone at gmail.com  Wed Jul  5 15:03:13 2017
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 5 Jul 2017 17:03:13 +0200
Subject: [Freeswitch-users] G729 annexb=no not forwarded to leg B
Message-ID: <F43A397B-BE5D-4B39-AFF7-B18B5032E3AB@gmail.com>

All,

If I receive an INVITE from Leg A (Patton GW) with G729 and annexb=no in the SDP, I would expect annexb=no to be automatically inserted in the SDP to Leg B.
It seems either with export absolute_codec_string=G729, and without it (so I send G729 and G711 to Leg B), annexb=no is not sent to Leg B.

Is there any parameters or any configuration issue which may lead to annex=no being suppressed (which is an issue, as some carriers default to annexb=yes), or is it the normal behavior, meaning I have to manually export it to leg B if it was requested by A ?

Thank you




From mark.melling at savageminds.com  Wed Jul  5 16:08:11 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Wed, 05 Jul 2017 16:08:11 +0000
Subject: [Freeswitch-users] Verto: no inbound audio when verto client
 called from conference
In-Reply-To: <CALXCt0qojw_FEP=ZveB8_=DGUSEA_BCi2gYwUxu1yYWos1nASA@mail.gmail.com>
References: <CAJp-f1ONFgNwhHs7MLuC+xfy8gEwKvOwJ8+CYq2DmcNB+xkp_g@mail.gmail.com>
 <CALXCt0qojw_FEP=ZveB8_=DGUSEA_BCi2gYwUxu1yYWos1nASA@mail.gmail.com>
Message-ID: <CAJp-f1NJv3_Etvx2ThvGW3pR9i9jx-s+YUZOBY2JkA3OAwpJLg@mail.gmail.com>

Thanks Giovanni for the suggestion.

I tried some more experiments and basically if I call a verto client and
add them to a conference then they don't hear the audio (although the
conference is detecting audio when they speak).

But if they dial into the conference then everything appears fine and they
do hear audio.

Specifically from fs_cli I entered:

originate <call-url> &conference(<conf-name>@default)

If call-url is a sip client then you hear the conference music, but if
call-url is a verto client you don't hear any conference music. But the
conference does detect when the verto client is speaking (at least the
status in the verto web page indicates the user is talking).

Whereas if you dialled into a conference from a verto client then you would
hear the conference music.

So I'm not sure how I can work around this.







On Wed, 5 Jul 2017 at 08:57 Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> On 4 July 2017 at 23:55, Mark Melling <mark.melling at savageminds.com>
> wrote:
>
> Hi,
>>
>> I have a dialplan where a user calls a number which results in them being
>> placed in a conference room, and an outbound called made to a verto client
>> when the conference is created, by defining in the dialplan:
>>
>>     <action application="conference_set_auto_outcall" data="user/1000@
>> $${domain}"/>
>>
>> The verto client (which is basically taken from the verto demo) rings and
>> is answered.
>>
>> The original inbound caller can hear the verto client speaking, but the
>> verto client can't hear the the inbound caller (although the audio from the
>> inbound caller is getting to the conference room).
>>
>> If I change the dialling order - so the verto client makes the inbound
>> call and the SIP client is dialled from the conference using the same
>> technique as before (conference_set_auto_outcall). Then everything appears
>> to work fine, i.e. each caller can here the other.
>>
>> Any suggestions as to what might be the issue or steps I could take to
>> help identify the problem.
>>
>
>
> Maybe this is because verto (webrtc) takes time to establish audio because
> of stun, etc etc
>
> Try this: instead of generating autocall from inside conference (eg
> instead of using autocall),originate call to user, wait for her to answer,
> then (after she answer) sleep for 2 seconds, then transfer her to the conf
>
> -giovanni
>
>
>
>>
>> Thanks
>>
>> Mark
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From rahul.ultimate at gmail.com  Wed Jul  5 18:25:57 2017
From: rahul.ultimate at gmail.com (Rahul MathuR)
Date: Wed, 5 Jul 2017 23:55:57 +0530
Subject: [Freeswitch-users] Switching profiles upon 3xx Redirection
In-Reply-To: <CAM8wsqKJM2PwQuokGR7VQn6Wd3tDBV0at68nLaqEhEEfV4sg-w@mail.gmail.com>
References: <CAM8wsqKnH6r=tvW+wrJqBZb3gqiNM9P1znTLPtyx=2YUGh59rQ@mail.gmail.com>
 <CAM8wsqLtcKO5VQGRkQZhbzSGfzWGwYmk-ZggkPvr=f-D8wdCnQ@mail.gmail.com>
 <CAM8wsqJU82f4y+kEAi9Jbz8x+-XKzL=UZQqtAsWTHHLe7bc_ig@mail.gmail.com>
 <CAM8wsqKWqVbxRgAAeEMwugUQrVBncjBappGNO2UqMFnBCFsefg@mail.gmail.com>
 <CAM8wsqLRBeBPeNzF7CqrJtSr3yJQz216sB-etdvJn=QzPFBJpw@mail.gmail.com>
 <CAM8wsq+XrtZrEEmMWEP0TX0EQfNjngW9wuAkC-Jdsk=vQHyQxw@mail.gmail.com>
 <CAM8wsqKOphmWpOdfevp57m6ub27U0YVwK+tHLp=6=yxnOhbXdg@mail.gmail.com>
 <CAM8wsqLsHMNsteYStWqLyLjfrQ2P4zxBmJD8M1eowW1Dw7MoPg@mail.gmail.com>
 <CAM8wsq+4auoyC5xM4A9OM2GU2_Dg4HTOBwq3fzKSdfi-7jP5yw@mail.gmail.com>
 <CAM8wsqJ5sJF9zZDtckm5FehSHw+y6p=7w4K9-wFw7WcwmVp5UQ@mail.gmail.com>
 <CAM8wsqK80ywS6UtDOaXFShRWqt9UtTiat7Np3PB56cY6yk2k_w@mail.gmail.com>
 <CAM8wsq+nd-D=4LRz2J=Bfzw_8Rd6wWsRKvrdO0E6-RBMubsu8Q@mail.gmail.com>
 <CAM8wsq+tu=5iT5MB=gWuHqg51H-bQTbVe1cBhZGmDgq7qi7fOQ@mail.gmail.com>
 <CAM8wsq+Pxw3HP6iYj3U86WsoN4Xjtr2pLLcmnfV3J1769XWFhA@mail.gmail.com>
 <CAM8wsqL0AEP_bFRpdYL5465vnmFxH0uSbtvqjBOs2y4nQHn7Bw@mail.gmail.com>
 <CAM8wsqKyw8oCZQzQYn5VbyD43ER2nJ9Cw8Cd0wnKVA8Lcqm=dQ@mail.gmail.com>
 <CAM8wsqKJM2PwQuokGR7VQn6Wd3tDBV0at68nLaqEhEEfV4sg-w@mail.gmail.com>
Message-ID: <CAM8wsq+Qn5D7oEjdHfUVDggRGy4KgmsB723u5mdDXDqWZo6D+w@mail.gmail.com>

Any comments guys ??

On Jul 1, 2017 12:11 AM, "Rahul MathuR" <rahul.ultimate at gmail.com> wrote:

> Hello guys,
>
> Thanks for a wonderful product !
> I'm loving it.
>
> I am working on a situation where FS is installed on a centos7 having 2 ip
> address internal and public. On internal, I receive  INVITE and create a
> b-leg and send to a Redirect server. This server sends 3xx and populates
> Contact header. When I receive it, I can see that external profile has
> already been set. And I can't change it to send it to another profile.
>
> Is there a way to do that ?
> My external profile listens on internal ip. I tried different options
> mentioned in confluence page but to no avail.
>
> Am I missing anything or trying to accomplish something impossible ?
>
> Thanks in anticipation.
>
> Rahul
>
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From gmaruzz at gmail.com  Wed Jul  5 19:10:51 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 5 Jul 2017 21:10:51 +0200
Subject: [Freeswitch-users] Verto: no inbound audio when verto client
 called from conference
In-Reply-To: <CAJp-f1NJv3_Etvx2ThvGW3pR9i9jx-s+YUZOBY2JkA3OAwpJLg@mail.gmail.com>
References: <CAJp-f1ONFgNwhHs7MLuC+xfy8gEwKvOwJ8+CYq2DmcNB+xkp_g@mail.gmail.com>
 <CALXCt0qojw_FEP=ZveB8_=DGUSEA_BCi2gYwUxu1yYWos1nASA@mail.gmail.com>
 <CAJp-f1NJv3_Etvx2ThvGW3pR9i9jx-s+YUZOBY2JkA3OAwpJLg@mail.gmail.com>
Message-ID: <CALXCt0oykcTR0qaQHrLAAJmacTXsCoHkkUBP+ErY6VF27V_Hwg@mail.gmail.com>

On 5 July 2017 at 18:08, Mark Melling <mark.melling at savageminds.com> wrote:

> Thanks Giovanni for the suggestion.
>
> I tried some more experiments and basically if I call a verto client and
> add them to a conference then they don't hear the audio (although the
> conference is detecting audio when they speak).
>
> But if they dial into the conference then everything appears fine and they
> do hear audio.
>
> Specifically from fs_cli I entered:
>
> originate <call-url> &conference(<conf-name>@default)
>
> If call-url is a sip client then you hear the conference music, but if
> call-url is a verto client you don't hear any conference music. But the
> conference does detect when the verto client is speaking (at least the
> status in the verto web page indicates the user is talking).
>
> Whereas if you dialled into a conference from a verto client then you
> would hear the conference music.
>
> So I'm not sure how I can work around this.
>
>
>
Have you tried what I suggested?



>>
>> Maybe this is because verto (webrtc) takes time to establish audio
>> because of stun, etc etc
>>
>> Try this: instead of generating autocall from inside conference (eg
>> instead of using autocall),originate call to user, wait for her to answer,
>> then (after she answer) sleep for 2 seconds, then transfer her to the conf
>>
>> -giovanni
>>
>
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From joelists at tm.net.uk  Wed Jul  5 19:20:58 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Wed, 5 Jul 2017 20:20:58 +0100
Subject: [Freeswitch-users] Switching profiles upon 3xx Redirection
In-Reply-To: <CAM8wsqKJM2PwQuokGR7VQn6Wd3tDBV0at68nLaqEhEEfV4sg-w@mail.gmail.com>
References: <CAM8wsqKnH6r=tvW+wrJqBZb3gqiNM9P1znTLPtyx=2YUGh59rQ@mail.gmail.com>
 <CAM8wsqLtcKO5VQGRkQZhbzSGfzWGwYmk-ZggkPvr=f-D8wdCnQ@mail.gmail.com>
 <CAM8wsqJU82f4y+kEAi9Jbz8x+-XKzL=UZQqtAsWTHHLe7bc_ig@mail.gmail.com>
 <CAM8wsqKWqVbxRgAAeEMwugUQrVBncjBappGNO2UqMFnBCFsefg@mail.gmail.com>
 <CAM8wsqLRBeBPeNzF7CqrJtSr3yJQz216sB-etdvJn=QzPFBJpw@mail.gmail.com>
 <CAM8wsq+XrtZrEEmMWEP0TX0EQfNjngW9wuAkC-Jdsk=vQHyQxw@mail.gmail.com>
 <CAM8wsqKOphmWpOdfevp57m6ub27U0YVwK+tHLp=6=yxnOhbXdg@mail.gmail.com>
 <CAM8wsqLsHMNsteYStWqLyLjfrQ2P4zxBmJD8M1eowW1Dw7MoPg@mail.gmail.com>
 <CAM8wsq+4auoyC5xM4A9OM2GU2_Dg4HTOBwq3fzKSdfi-7jP5yw@mail.gmail.com>
 <CAM8wsqJ5sJF9zZDtckm5FehSHw+y6p=7w4K9-wFw7WcwmVp5UQ@mail.gmail.com>
 <CAM8wsqK80ywS6UtDOaXFShRWqt9UtTiat7Np3PB56cY6yk2k_w@mail.gmail.com>
 <CAM8wsq+nd-D=4LRz2J=Bfzw_8Rd6wWsRKvrdO0E6-RBMubsu8Q@mail.gmail.com>
 <CAM8wsq+tu=5iT5MB=gWuHqg51H-bQTbVe1cBhZGmDgq7qi7fOQ@mail.gmail.com>
 <CAM8wsq+Pxw3HP6iYj3U86WsoN4Xjtr2pLLcmnfV3J1769XWFhA@mail.gmail.com>
 <CAM8wsqL0AEP_bFRpdYL5465vnmFxH0uSbtvqjBOs2y4nQHn7Bw@mail.gmail.com>
 <CAM8wsqKyw8oCZQzQYn5VbyD43ER2nJ9Cw8Cd0wnKVA8Lcqm=dQ@mail.gmail.com>
 <CAM8wsqKJM2PwQuokGR7VQn6Wd3tDBV0at68nLaqEhEEfV4sg-w@mail.gmail.com>
Message-ID: <6BC2657E-D247-4FD5-BE90-8BC411966C5A@tm.net.uk>

Can you send the b-leg to the redirect server on the internal profile and then the re-directed call will go out on the internal profile.

The other option is to catch the re-direct into another context and handle it from there.

> On 30 Jun 2017, at 19:41, Rahul MathuR <rahul.ultimate at gmail.com> wrote:
> 
> Hello guys,
> 
> Thanks for a wonderful product !
> I'm loving it.
> 
> I am working on a situation where FS is installed on a centos7 having 2 ip address internal and public. On internal, I receive  INVITE and create a b-leg and send to a Redirect server. This server sends 3xx and populates Contact header. When I receive it, I can see that external profile has already been set. And I can't change it to send it to another profile.
> 
> Is there a way to do that ?
> My external profile listens on internal ip. I tried different options mentioned in confluence page but to no avail.
> 
> Am I missing anything or trying to accomplish something impossible ?
> 
> Thanks in anticipation.
> 
> Rahul
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From italo at freeswitch.org  Wed Jul  5 19:25:32 2017
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 5 Jul 2017 16:25:32 -0300
Subject: [Freeswitch-users] G729 annexb=no not forwarded to leg B
In-Reply-To: <F43A397B-BE5D-4B39-AFF7-B18B5032E3AB@gmail.com>
References: <F43A397B-BE5D-4B39-AFF7-B18B5032E3AB@gmail.com>
Message-ID: <CABMX_w9XubhPMSDVyG5_ksP4mnBfxEjLaubAiP_jSm=zG8RDrQ@mail.gmail.com>

Try:

<action application="export" data="rtp_append_audio_sdp=a=fmtp:18
annexb=no"/>

On Wed, Jul 5, 2017 at 12:03 PM, David Ponzone <david.ponzone at gmail.com>
wrote:

> All,
>
> If I receive an INVITE from Leg A (Patton GW) with G729 and annexb=no in
> the SDP, I would expect annexb=no to be automatically inserted in the SDP
> to Leg B.
> It seems either with export absolute_codec_string=G729, and without it (so
> I send G729 and G711 to Leg B), annexb=no is not sent to Leg B.
>
> Is there any parameters or any configuration issue which may lead to
> annex=no being suppressed (which is an issue, as some carriers default to
> annexb=yes), or is it the normal behavior, meaning I have to manually
> export it to leg B if it was requested by A ?
>
> Thank you
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 
Ítalo Rossi
italo at freeswitch.org
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From ljjimenez at gmail.com  Wed Jul  5 19:59:17 2017
From: ljjimenez at gmail.com (Luis Jimenez)
Date: Wed, 5 Jul 2017 15:59:17 -0400
Subject: [Freeswitch-users] Simple script for ringtone
In-Reply-To: <6bbe39c2-2af7-1111-7ec5-d5c3f6ddbba8@digitalmail.com>
References: <6bbe39c2-2af7-1111-7ec5-d5c3f6ddbba8@digitalmail.com>
Message-ID: <CAKUOwoHYYwGrrr6dOQhNxbY2KiGw48RT04ax1_N724UT-d9WvA@mail.gmail.com>

<condition field="destination_number" expression="^(.+)$">
 <action application="ring_ready"/>
 <action application="sleep" data="86400000"/>
 <action application="respond" data="408 Try again later"/>
 <action application="hangup"/>
</condition>





On Wed, Jul 5, 2017 at 11:42 AM, Alex Lake <alex at digitalmail.com> wrote:

> I want to set up a number that just rings and never answers. Unfortunately
> my freeswitch is very rusty!
>
> Can someone help me out with what must be a 2 or 3 line script? ;-)
>
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mark.melling at savageminds.com  Wed Jul  5 21:21:50 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Wed, 05 Jul 2017 21:21:50 +0000
Subject: [Freeswitch-users] Verto: no inbound audio when verto client
 called from conference
In-Reply-To: <CALXCt0oykcTR0qaQHrLAAJmacTXsCoHkkUBP+ErY6VF27V_Hwg@mail.gmail.com>
References: <CAJp-f1ONFgNwhHs7MLuC+xfy8gEwKvOwJ8+CYq2DmcNB+xkp_g@mail.gmail.com>
 <CALXCt0qojw_FEP=ZveB8_=DGUSEA_BCi2gYwUxu1yYWos1nASA@mail.gmail.com>
 <CAJp-f1NJv3_Etvx2ThvGW3pR9i9jx-s+YUZOBY2JkA3OAwpJLg@mail.gmail.com>
 <CALXCt0oykcTR0qaQHrLAAJmacTXsCoHkkUBP+ErY6VF27V_Hwg@mail.gmail.com>
Message-ID: <CAJp-f1MvetuC4vWL5UM2uDda99FPPYZWTahbn6W9miPVBF-LQg@mail.gmail.com>

Your suggestion does work, I did the following manually from fs_cli.

So with a user dialled into a conference room I was able to do:

> originate  <call-url> &park

Where the call-url referring to a verto client, then

> uuid_transfer <verto-call-uuid> conference:<conf-name> inline

And that worked, I could hear audio from both clients connected to the
conference.

Thanks for your help.



On Wed, 5 Jul 2017 at 20:12 Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> On 5 July 2017 at 18:08, Mark Melling <mark.melling at savageminds.com>
> wrote:
>
>> Thanks Giovanni for the suggestion.
>>
>> I tried some more experiments and basically if I call a verto client and
>> add them to a conference then they don't hear the audio (although the
>> conference is detecting audio when they speak).
>>
>> But if they dial into the conference then everything appears fine and
>> they do hear audio.
>>
>> Specifically from fs_cli I entered:
>>
>> originate <call-url> &conference(<conf-name>@default)
>>
>> If call-url is a sip client then you hear the conference music, but if
>> call-url is a verto client you don't hear any conference music. But the
>> conference does detect when the verto client is speaking (at least the
>> status in the verto web page indicates the user is talking).
>>
>> Whereas if you dialled into a conference from a verto client then you
>> would hear the conference music.
>>
>> So I'm not sure how I can work around this.
>>
>>
>>
> Have you tried what I suggested?
>
>
>
>>>
>>> Maybe this is because verto (webrtc) takes time to establish audio
>>> because of stun, etc etc
>>>
>>> Try this: instead of generating autocall from inside conference (eg
>>> instead of using autocall),originate call to user, wait for her to answer,
>>> then (after she answer) sleep for 2 seconds, then transfer her to the conf
>>>
>>> -giovanni
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From asilva at wirelessmundi.com  Wed Jul  5 21:27:47 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Wed, 5 Jul 2017 23:27:47 +0200
Subject: [Freeswitch-users] Simple script for ringtone
In-Reply-To: <6bbe39c2-2af7-1111-7ec5-d5c3f6ddbba8@digitalmail.com>
References: <6bbe39c2-2af7-1111-7ec5-d5c3f6ddbba8@digitalmail.com>
Message-ID: <e9cf6fc0-78f1-671d-98db-867ba57ad481@wirelessmundi.com>

You can try, this will ring forever but the line/remote must support 
early media in order to work


<extension name="ringing on early media" continue="false">
<condition field="destination_number" expression=".*">
<action application="pre_answer"/>
<action application="endless_playback" 
data="tone_stream://%(1500,3000,425.0)"/>
</condition>
</extension>


Regards,
António

On 07/05/2017 05:42 PM, Alex Lake wrote:
> I want to set up a number that just rings and never answers. 
> Unfortunately my freeswitch is very rusty!
>
> Can someone help me out with what must be a 2 or 3 line script? ;-)
>
> Alex
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From nandy1925 at gmail.com  Thu Jul  6 04:01:54 2017
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 6 Jul 2017 04:01:54 +0000
Subject: [Freeswitch-users] Dialplan: destination_number: regex question
In-Reply-To: <3821491498857447@web5m.yandex.ru>
References: <CAB+hG+gCkMaAEH9g02hjLTJyuLndN-7DFvWdWT6m+i=EXZn3-A@mail.gmail.com>
 <3821491498857447@web5m.yandex.ru>
Message-ID: <CAHKqMb5gb4=fwb8=TK=Wd1Jxy4tRuuLvE+NFLnsdDoRXym2Ecw@mail.gmail.com>

I can cite the recommendation of mod_lcr, convert all numbers to the
International format starting with +64 or 64.  IMO, the location of proxy
will determine what numbers to include.

In your example, a proxy in HK, CC (64) and AC (22) are needed to call NZ
phone.

So, you're asking for the regex .... How about this?

<condition field="destination_number" expression="(\+?64\d{9})"/>

Nandy Dagondon <https://camiguintelekom.wordpress.com/about-the-author/>


On Fri, Jun 30, 2017 at 9:17 PM, <royj at yandex.ru> wrote:

> Not quite clear. What are you trying to achieve?
>
> 30.06.2017, 11:11, "Nate" <nathan.port at gmail.com>:
>
> Good day/evening everyone,
>
> Apologies for not being able to figure this out on my own. I've been
> searching and trying for several days to get inbound/outbound working but
> have yet to see success.
>
> At this stage I need help determining a proper regex expression for
> handling New Zealand phone numbers.
>
> For instance, there are three different ways of expressing numbers here in
> NZ:
>
> International: +64 22 333 4444 <+64%2022%20333%204444>
> non-local: 022 333 4444
> local: 333 4444
>
> A couple questions related to the SIP proxy:
>
> If the SIP proxy is located in Hong Kong, but the phone number is a New
> Zealand number, does the location of the proxy have any impact on the
> number of characters in the string for inbound/outbound calls?
>
> Again, apologies for the rudimentary nature of these questions, but having
> nearly exhausted all other options (docs, searches, IRC), I am now spending
> a large amount of time guessing and trial and error without any progress.
>
> Many thanks for any feedback at all.
>
> Nate
> ,
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nandy1925 at gmail.com  Thu Jul  6 04:13:52 2017
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 6 Jul 2017 04:13:52 +0000
Subject: [Freeswitch-users] VOIP-GSM Gateway configuration Error
In-Reply-To: <PS1PR03MB1516E745CC0AB3DEB01C63A3A9DD0@PS1PR03MB1516.apcprd03.prod.outlook.com>
References: <CAM99_YmYaGbdz51NT2Fm3pahXEy8DjgMxRdir3xqa6_mSX5=mw@mail.gmail.com>
 <PS1PR03MB1516E745CC0AB3DEB01C63A3A9DD0@PS1PR03MB1516.apcprd03.prod.outlook.com>
Message-ID: <CAHKqMb7=sRHK783YPVPH1kxQSwaC5LoSdskQrsd+pcdOtBMcWg@mail.gmail.com>

It looks the GSM gateway is registering to the Freeswitch. Try to change
all directory values from 192.168.2.69 to 192.168.2.71.

Nandy Dagondon <https://camiguintelekom.wordpress.com/about-the-author/>


On Wed, Jun 28, 2017 at 4:25 AM, bob. chen <thetsinling at outlook.com> wrote:

> https://freeswitch.org/confluence/display/FREESWITCH/Dinstar+GSM+gateway+
> FreeSwitch+HowTo
>
> try  peer to peer ;)
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Deepika Yadav
> *Sent:* Tuesday, June 27, 2017 7:06 PM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] VOIP-GSM Gateway configuration Error
>
>
>
> Hi,
>
>
>
> I have been trying for a while the configuration of a VOIP-GSM gateway
> that I have bought but the registration fails every time.
>
>
>
> Error at Freeswitch console
>
>
>
> 2017-06-27 16:27:10.191913 [WARNING] sofia_reg.c:2906 Can't find user [*testadmin at 192.168.2.71
> <testadmin at 192.168.2.71>*] from *192.168.2.69*
>
> You must define a domain called '192.168.2.71' in your directory and add a
> user with the id="*testadmin*" attribute
>
> and you must configure your device to use the proper domain in it's
> authentication credentials.
>
> 2017-06-27 16:27:14.071916 [ERR] sofia_reg.c:2447 MySIP Failed
> Registration with status Request Timeout [408]. failure #1
>
> 2017-06-27 16:27:14.111913 [WARNING] sofia_reg.c:505 MySIP Failed
> Registration [408], setting retry to 30 seconds.
>
>
>
> Here *'192.168.2.71*' is the local IP of the Freeswitch server and
> *192.168.2.69* that of the gateway attached to the LAN port of the
> Freeswitch server.
>
>
>
> I have put SIP UserID in the gateway admin panel as "*testadmin*"
> alongwith its passoword.
>
>
>
> my external SIP profile is as follows:
>
>
>
> <include>
>
>
>
> <gateway name="MySIP">
>
>
>
>   <param name="username" value="testadmin"/>
>
>
>
>   <param name="realm" value="192.168.2.69"/>
>
>
>
>   <param name="from-user" value="testadmin"/>
>
>
>
>   <param name="from-domain" value="192.168.2.69"/>
>
>
>
>   <param name="password" value="****"/>
>
>
>
>   <param name="extension" value="5888"/>
>
>
>
>   <param name="proxy" value="192.168.2.69"/>
>
>
>
>   <param name="expire-seconds" value="120"/>
>
>
>
>   <param name="register" value="true"/>
>
>
>
>   <param name="caller-id-in-from" value="true"/>
>
> </gateway>
>
>
>
> I have tried number of permutations and combinations but the gateway is
> not getting registered.
>
>
>
>
>
> Regards
> Deepika
> https://www.iiitd.edu.in/~deepikay/
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nandy1925 at gmail.com  Thu Jul  6 04:47:19 2017
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 6 Jul 2017 04:47:19 +0000
Subject: [Freeswitch-users] Conditional call forward
In-Reply-To: <CAA_Nwu67OMwjoD+6ZN496U9Qzino_y8FK3k3=rPff41CNDpsXw@mail.gmail.com>
References: <CAA_Nwu4U1n6cj+Js3Mb0E0c-eSUpade+3C_VmEsvQXyc0Yp8ig@mail.gmail.com>
 <F83CBE07-C94D-4964-8FE4-DFC84A194997@gmail.com>
 <CAA_Nwu67OMwjoD+6ZN496U9Qzino_y8FK3k3=rPff41CNDpsXw@mail.gmail.com>
Message-ID: <CAHKqMb47XuXr12v_o2gVVmYTOTRvRiUidFvQu7TgZG-CVKqNXA@mail.gmail.com>

The XML dialplan logic would be like this:

<extension name="group-allowed-cf"/>
<condition field="group_field" expression="group_name"/>
<condition field="destination_number" expression="extn_number">
   <actions where call is forwarded when bridge fail>
</condition>

<extension name="all-other-callers"/>
<condition field="destination_number" expression="extn_number">
   <actions where call is hangup when bridge fail>
</condition>

Hope this is what you want. :-)

Nandy Dagondon <https://camiguintelekom.wordpress.com/about-the-author/>



On Thu, May 18, 2017 at 8:46 AM, Ashwin Rath <ashwinrath at gmail.com> wrote:

> Actually this related to fusion PBX. CF is configured on FS as a custom
> dial-string in the directory . The idea is to not forward if a call comes
> to a ring group but forward if it comes directly to the number.
>
> is there some way the dialplan can be modified to change the dial-string ?
>
> On 16 May 2017 at 22:20, Srigo Kana <ksrigo at gmail.com> wrote:
>
>> Hi,
>>
>> Is the callforward configured on the phone?
>> If you get 302 redirect from a phone, you can jst catch it in a dialplan
>> and do whatever you want.
>>
>> Srigo
>>
>> Sent from my iPhone
>>
>> > On 13 May 2017, at 19:14, Ashwin Rath <ashwinrath at gmail.com> wrote:
>> >
>> > Hi
>> >
>> > I have an extension which has call forward setup BUT i would like the
>> call forward to work only when dialed from a certain number and not from
>> another numbers. Can this be achieved ?
>> >
>> > --
>> > Ashwin Kumar Rath
>> > ____________________________________________________________
>> _____________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>> freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Ashwin Kumar Rath
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nandy1925 at gmail.com  Thu Jul  6 04:59:58 2017
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 6 Jul 2017 04:59:58 +0000
Subject: [Freeswitch-users] Verto calls
In-Reply-To: <CAKThY04NEa0oYTk7p=bDS6agcQvP1FuUqOm+uQ-cpLdBsevQmg@mail.gmail.com>
References: <CAKThY04NEa0oYTk7p=bDS6agcQvP1FuUqOm+uQ-cpLdBsevQmg@mail.gmail.com>
Message-ID: <CAHKqMb6LEYG+RpbWa0mjq1NbBNVaxPFb0ckpKmuaGdxcrj7VTA@mail.gmail.com>

Show your directory entries for Verto clients and the verto.js.  I notice
there's no data in the bridge() application.

Nandy Dagondon <https://camiguintelekom.wordpress.com/about-the-author/>


On Tue, May 16, 2017 at 3:19 PM, Agustí Ubalde <aubalde at presenceco.com>
wrote:

> Hi all,
>
>
>
> I am trying to call from Verto extension to another Verto extension. *Both
> are successfully registered* (Verto status show the successfully
> register) but the call between is not established.
>
> The call remains in ring state.
>
>
>
> This is the last dialplan function executed (calling 1000 to 1001):
>
> *EXECUTE verto.rtc/1001 bridge()*
>
>
>
> Regards,
>
> PRESENCE TECHNOLOGY
>
> An ENGHOUSE INTERACTIVE Company
>
> Agustí Ubalde Bellot
>
> Chief Developer
>
> C/ Comte Urgell 240 3º-A
>
> Barcelona 08036
>
> aubalde at presenceco.com
>
> Ph: +34 93 10 10 322
>
> Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH
>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
>
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at gmail.com  Thu Jul  6 05:24:48 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 6 Jul 2017 07:24:48 +0200
Subject: [Freeswitch-users] Verto calls
In-Reply-To: <CAKThY04NEa0oYTk7p=bDS6agcQvP1FuUqOm+uQ-cpLdBsevQmg@mail.gmail.com>
References: <CAKThY04NEa0oYTk7p=bDS6agcQvP1FuUqOm+uQ-cpLdBsevQmg@mail.gmail.com>
Message-ID: <CALXCt0q01BjikpS-hzbe13ZQP3xig0QcsLc57UQkHYAjJqNgoQ@mail.gmail.com>

On 16 May 2017 at 17:19, Agustí Ubalde <aubalde at presenceco.com> wrote:

> Hi all,
>
>
>
> I am trying to call from Verto extension to another Verto extension. *Both
> are successfully registered* (Verto status show the successfully
> register) but the call between is not established.
>
> The call remains in ring state.
>
>
>
> This is the last dialplan function executed (calling 1000 to 1001):
>
> *EXECUTE verto.rtc/1001 bridge()*
>
>
>
You are calling the bridge application in a wrong way.

Look at the demo dialplan, Local_Extensions: it uses the contruct:

user/1001 bridge()

this because the correct dialstring is taken from user directory

If you want to do it yourself (eg, without use of the "user" construct, and
directory defined dialstring), then you must use the

verto_contact() API

Again, look at its use in the demo configuration.

Concept is: you have a verto registered user. You call from fs_cli
verto_contact("username"). This gives you the dialstring you must use.

>From dialplan, you would use: ${verto_contact(${dialed_user}@${dialed_domain})}
as dialstring






>
> Regards,
>
> PRESENCE TECHNOLOGY
>
> An ENGHOUSE INTERACTIVE Company
>
> Agustí Ubalde Bellot
>
> Chief Developer
>
> C/ Comte Urgell 240 3º-A
>
> Barcelona 08036
>
> aubalde at presenceco.com
>
> Ph: +34 93 10 10 322
>
> Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH
>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
>
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From nandy1925 at gmail.com  Thu Jul  6 05:31:12 2017
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 6 Jul 2017 05:31:12 +0000
Subject: [Freeswitch-users] from Asterisk sip.conf to FS
In-Reply-To: <CAGu7vdaK7-iktLDw+v_ASY5XWR49jE3sWnxBaTX0_pow1Eckfg@mail.gmail.com>
References: <CAGu7vdaK7-iktLDw+v_ASY5XWR49jE3sWnxBaTX0_pow1Eckfg@mail.gmail.com>
Message-ID: <CAHKqMb4oCvUDm1sBXF9G8yECOnt2tBWc8nFCN551k7jvMDS-iA@mail.gmail.com>

Pls see my comments below. Hope it helps.

Nandy Dagondon <https://camiguintelekom.wordpress.com/about-the-author/>


On Tue, Apr 25, 2017 at 9:23 PM, Valter Nogueira <valter at fastway.com.br>
wrote:

> I have an app that runs in Asterisk and I am trying port it to FS.
>
> Below, I pasted a small sip.conf that sums up all my needs.
>
> 1. We register out in a VOIP_PROVIDER
>
> 2. An external asterisk register in our asterisk as CLIENT01
>
> 3. Calls from CLIENT01 route to VOIP_PROVIDER
>
> 4. Yealink's, with ACL restrictions, also uses VOIP_PROVIDER to external
> calls
>
> 5. VOIP PROVIDER and Yealinks use one NIC (eth0)
>
> 6. CLIENT01 uses another NIC (eth1)
>
> In Asterisk all configs are keep together in sip.conf.
>
> However, in FS it seems I should spread out things in different configs
> parts.
>
> So:
>
> Should Yealink's accounts go into /usr/local/freeswitch/
> conf/directory/default?
>
No need since Yealink is registered to VOIP_PROVIDER, right?  Your public
dialplan should take care of the incoming calls - e.g. destination_number

>
> Should I put VOIP_PROVIDER in sofia's external profile?
>
Yes.

>
> Should I create an additional external profile to CLIENT01? Where should I
> define CLIENT01 account to allow it register in? In
> /usr/local/freeswitch/conf/directory/default?
>
CLIENT01 in directory/default but ... Yes, create a new profile so that it
will listen with the WAN IP address and new port (e.g. WAN_IP:5070 since
5060 listens to LAN clients registering to Freeswitch, port 5080 for
FreeSwitch to register to VOIP_PROVIDER. Search for "doublenat" profile
example in the Wiki.

>
> Thanks for any help.
>
> Valter
>
> === SIP.CONF ===
>
> [general]
> disallow=all
> context=default                 ; Default context for incoming calls
> allowoverlap=no                 ; Disable overlap dialing support.
> (Default is yes)
> bindport=5060                   ; UDP Port to bind to (SIP standard port
> is 5060)
> bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to
> all)
> srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
> alwaysauthreject=yes
> allowguest=no
>
> ;voip provider
> register => 185999:1010 at xxx.xxx.xxx.xxx:5060
>
> [VOIP_PROVIDER]
> type=peer
> context=FDTRONCO_94
> host=xxx.xxx.xxx.xxx
> port=5060
> username=185999
> authuser=185999
> authname=185999
> secret=XXXXXXXXXXXXXX
> fromdomain=xxx.xxx.xxx.xxx
> fromuser=185999
> insecure=port,invite
> canreinvite=no
> nat=no
> disallow=all
> allow=ulaw:30
> dtmfmode=rfc2833
> ignoreregexpire=yes
> language=pt_BR
> call-limit=9999
>
> ;EXTERNAL ASTERISK ACCOUNT - CALLS GOES THRU VOIP_PROVIDER
> [CLIENT01]
> type=peer
> context=FDTRONCO_60
> host=dynamic
> port=5060
> username=CLIENT01
> authuser=CLIENT01
> authname=CLIENT01
> secret=XXXXXXX
> insecure=port,invite
> canreinvite=no
> nat=yes
> disallow=all
> allow=g729
> dtmfmode=rfc2833
> ignoreregexpire=yes
> language=pt_BR
> call-limit=440
>
> ;yealinks ip phones
> [1000]
> type=friend
> secret=XXXX
> host=dynamic
> deny=0.0.0.0/0.0.0.0
> permit=192.168.0.0/16
> username=1000
> context=FASTDIALER_RAMAIS
> callerid=testefone <1000>
> requirecalltoken=no
> nat=no
> canreinvite=no
> qualify=yes
> disallow=all
> allow=alaw
> call-limit=2
> dtmfmode=RFC2833
> language=pt_BR
>
> Callgroup=1
> pickupgroup=1
>
> [2012]
> type=friend
> secret=XXXX
> host=dynamic
> deny=0.0.0.0/0.0.0.0
> permit=192.168.0.0/255.255.0.0
> username=2012
> context=FASTDIALER_RAMAIS
> callerid=Telefone sem fio <2012>
> requirecalltoken=no
> nat=no
> canreinvite=no
> qualify=no
> disallow=all
> allow=alaw
> call-limit=2
> dtmfmode=RFC2833
> language=pt_BR
>
> Callgroup=1
> pickupgroup=1
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From varghesepaul87 at gmail.com  Thu Jul  6 05:50:56 2017
From: varghesepaul87 at gmail.com (Varghese Paul)
Date: Thu, 6 Jul 2017 11:20:56 +0530
Subject: [Freeswitch-users] Any plan to use DTLSv1.2
Message-ID: <CAM33Ch4WkijvcitCqZM-RccjDc_O93nerxJOznL4Xbif9G-gGA@mail.gmail.com>

Hi all,

Right now we are hard coded for using DTLSv1.0 which use TLS1.1. In the
latest master I can see we are using DTLS1.0 and DTLS1.2 method if the SSL
version is 1.1.0.

Do we have any plan to add support for DTLSv1.2 ( TLS1.2) ?.

Regards

Varghese Paul
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From stefano.favaro at edistar.com  Thu Jul  6 10:11:12 2017
From: stefano.favaro at edistar.com (Stefano Favaro)
Date: Thu, 6 Jul 2017 12:11:12 +0200 (CEST)
Subject: [Freeswitch-users] freeswitch.Session with LUA
In-Reply-To: <764431494.171.1499335160683.JavaMail.sfa@EDISTAR-SFA>
Message-ID: <249445539.185.1499335849571.JavaMail.sfa@EDISTAR-SFA>




Hello, 


I'm writing a simple script with LUA to connect an inbound call to e new outbound call. 
Everything work fine but I can't set the ringback tone nor a waiting music on the leg A and also the call_timeout variable. 
The legA is in silence while the legB is ringing and the call_timeout is always the default 60 seconds. 
What is wrong with this code? 


Thanks 


This is the code: 
... 

legA:streamFile("/usr/share/freeswitch/sounds/wait.wav") 


legA:setVariable("ringback", "$${uk-ring}") 
legA:setVariable("call_timeout", 30) 


legB = freeswitch.Session("{ignore_early_media=true, hangup_after_bridge=true, origination_caller_id_number=1234}sofia/internal/0123456789 at myserver) 


if legB:ready() then 
freeswitch.bridge(legA, legB) 
end 





Stefano Favaro 
Sviluppo Servizi ed Applicazioni 
_____________________ 



Edistar Srl 

a socio unico soggetta a direzione e coordinamento di YourVoice SpA 

Via Artigianato 1 – I – 31050 Vedelago (TV) Italy 
Phone +39 0423 7331 – Fax +39 0423 733133 

skype: stefanofavaro 

www.edistar.com 
	

	

		

															


Le informazioni trasmesse attraverso la presente e-mail ed i suoi allegati sono dirette esclusivamente al destinatario e devono ritenersi riservate con divieto di diffusione e di uso nei giudizi salva espressa autorizzazione; nel caso di utilizzo senza espressa autorizzazione, potrà essere effettuata denuncia alla competente Autorità. La diffusione e la comunicazione da parte di soggetto diverso dal destinatario è vietata dall’art. 616 e ss. c.p. e dal d. l.vo n. 196/03. Se la presente e-mail e i suoi allegati fossero stati ricevuti per errore da persona diversa dal destinatario preghiamo di distruggere quanto ricevuto e di rinviare al mittente con lo stesso mezzo. Grazie per la collaborazione. 

This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this for the addressee, you must not use, copy, disclose or take any action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply e-mail and delete this message. Thank you for your cooperation. 
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From jurijs.ivolga at gmail.com  Thu Jul  6 11:01:37 2017
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Thu, 6 Jul 2017 14:01:37 +0300
Subject: [Freeswitch-users] freeswitch.Session with LUA
In-Reply-To: <249445539.185.1499335849571.JavaMail.sfa@EDISTAR-SFA>
References: <764431494.171.1499335160683.JavaMail.sfa@EDISTAR-SFA>
 <249445539.185.1499335849571.JavaMail.sfa@EDISTAR-SFA>
Message-ID: <CAOrA2UbEMQLk5qXJg6dRnqsT_gQKaJiY5_eK1Faa8kV467Av_g@mail.gmail.com>

Hi,

Not sure if it helps, but I got similar problem and I made workaround when
I bridge to loopback inside Lua something like this:

session:execute("bridge",
"{ignore_early_media=true,timeout="..timeout.."}loopback/wait")

And in dialplan:

<extension name="wait">      <condition field="destination_number"
expression="^wait$">
        <action application="set" data="ringback=$${uk-ring}"/>
 	<action application="pre_answer"/>        <!-- dial command to B leg
sofia/internal/0123456789 at myserver -->
       </condition></extension>

In this case you should be able to hear UK ring-back tones.

P.S. I never tested this code, but you should get an overall idea.

With kind regards,


Jurijs

On Thu, Jul 6, 2017 at 1:11 PM, Stefano Favaro <stefano.favaro at edistar.com>
wrote:

>
> Hello,
>
> I'm writing a simple script with LUA to connect an inbound call to e new
> outbound call.
> Everything work fine but I can't set the *ringback* tone nor a waiting
> music on the leg A and also the *call_timeout* variable.
> The legA is in silence while the legB is ringing and the call_timeout is
> always the default 60 seconds.
> What is wrong with this code?
>
> Thanks
>
> This is the code:
> ...
> legA:streamFile("/usr/share/freeswitch/sounds/wait.wav")
>
> legA:setVariable("ringback", "$${uk-ring}")
> legA:setVariable("call_timeout", 30)
>
> legB = freeswitch.Session("{ignore_early_media=true,
> hangup_after_bridge=true, origination_caller_id_number=
> 1234}sofia/internal/0123456789 at myserver)
>
> if legB:ready() then
> freeswitch.bridge(legA, legB)
> end
>
>
> Stefano Favaro
> Sviluppo Servizi ed Applicazioni
> _____________________
>
>
>
> Edistar Srl
>
> a socio unico soggetta a direzione e coordinamento di YourVoice SpA
>
> Via Artigianato 1 – I – 31050 Vedelago (TV) Italy
>
> Phone +39 0423 7331 <+39%200423%207331> – Fax +39 0423 733133
> <+39%200423%20733133>
>
>
> skype: stefanofavaro
>
> www.edistar.com
>
>
> <http://twitter.com/#%21/edistar>
>
>
> <http://www.linkedin.com/company/239920>
>
>
>
> Le informazioni trasmesse attraverso la presente e-mail ed i suoi allegati
> sono dirette esclusivamente al destinatario e devono ritenersi riservate
> con divieto di diffusione e di uso nei giudizi salva espressa
> autorizzazione; nel caso di utilizzo senza espressa autorizzazione, potrà
> essere effettuata denuncia alla competente Autorità. La diffusione e la
> comunicazione da parte di soggetto diverso dal destinatario è vietata
> dall’art. 616 e ss. c.p. e dal d. l.vo n. 196/03. Se la presente e-mail e i
> suoi allegati fossero stati ricevuti per errore da persona diversa dal
> destinatario preghiamo di distruggere quanto ricevuto e di rinviare al
> mittente con lo stesso mezzo. Grazie per la collaborazione.
>
> This message may contain confidential and/or privileged information. If
> you are not the addressee or authorized to receive this for the addressee,
> you must not use, copy, disclose or take any action based on this message
> or any information herein. If you have received this message in error,
> please advise the sender immediately by reply e-mail and delete this
> message. Thank you for your cooperation.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From shaun.stokes at itec-support.co.uk  Thu Jul  6 12:08:50 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Thu, 6 Jul 2017 12:08:50 +0000
Subject: [Freeswitch-users] mod_callcenter - Unloading queue with abandoned
	members
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E867BE92@mbx-01.sysconfig.co.uk>

When using the command 'callcenter_config queue unload queuename at context' this works great except when there are abandoned members in the queue, in this situation we get the following error repeatedly:
[WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this member
[WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this member
[WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this member
[WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this member
[WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this member

The only solution that seems to work is re-add the queue and wait for the abandoned members to disappear as defined by 'discard-abandoned-after'.

Is this normal behaviour in which case what's the command to clear abandoned members from the queue before it's unloaded, or is this a bug?

Thanks,
Shaun
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
Shaun Stokes - Infrastructure Analyst

T :     01453 700713
E :     shaun.stokes at itec-support.co.uk
W :     www.itec-support.co.uk

Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.

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From asilva at wirelessmundi.com  Thu Jul  6 12:26:19 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Thu, 6 Jul 2017 14:26:19 +0200
Subject: [Freeswitch-users] Error sending queries to pgsql
In-Reply-To: <39b55634-01fd-afea-a08f-27094f2e237c@wirelessmundi.com>
References: <d2f4526f-a783-a216-405d-126c253e34fd@wirelessmundi.com>
 <31066AE1-BC31-4C3F-8809-6C5B93C3B0E8@jerris.com>
 <f76beadd-9014-d279-1fb8-ea9d0048b240@wirelessmundi.com>
 <CAKeZRhvyLu+uT4_c_ZSEQVxW5zYZgrgeuAf10Qxpqn_5B-XRTA@mail.gmail.com>
 <39b55634-01fd-afea-a08f-27094f2e237c@wirelessmundi.com>
Message-ID: <1d4ae6f9-d857-4cba-c780-dc5ceaee004f@wirelessmundi.com>

Hi,

same results on old kernel..
i'm trying to reproduce it on isolate machine.. for now i put my 
findings in jira https://freeswitch.org/jira/browse/FS-10474.



Alexandr:

What SO are you running? did you get some errors from postgres before fs 
critical message?


Saludos / Regards / Cumprimentos,
António silva

On 07/04/2017 11:34 AM, Antonio Silva wrote:
> In this same box i also have data corruption with sqlite databases, 
> special the ones that i use with mod_lua.. but I think that it could 
> be something related with kernel.. i recently update to 4.9.x, and 
> there was some changes in ext4.
>
> I don't really know how to debug this, so today i'm reverting to the 
> old kernel, 4.4.x and check it happens again..
>
> If it happens again, my next move will try to reproduce this in FS..
>
> Saludos / Regards / Cumprimentos,
> António silva
> On 07/04/2017 10:58 AM, Alexandr Popov wrote:
>> Seems its trouble with sockets. I have the same problem appeared 
>> about a month ago.
>>
>> 2017-06-29 19:31 GMT+03:00 Antonio Silva <asilva at wirelessmundi.com 
>> <mailto:asilva at wirelessmundi.com>>:
>>
>>     Hi Michael,
>>
>>     Yes, i'm trying to figure it out if is an issue in FS or
>>     external.. but the message from PG i can't translate it.. i just
>>     enable more logs to see if i got extra hints...
>>
>>
>>     Thanks.
>>
>>     Saludos / Regards / Cumprimentos,
>>     António silva
>>
>>     On 06/29/2017 06:22 PM, Michael Jerris wrote:
>>
>>         If you can figure out a reliable way to reproduce this issue,
>>         please file a jira with details on what causes it.
>>
>>             On Jun 29, 2017, at 7:22 AM, Antonio Silva
>>             <asilva at wirelessmundi.com
>>             <mailto:asilva at wirelessmundi.com>> wrote:
>>
>>             Hi all,
>>
>>             i use pgsql in core and from time to time i see critical
>>             messages like fail to send query, example:
>>
>>             [CRIT] switch_pgsql.c:255 Failed to send query (update
>>             sip_authentication set expires='1498726568',last_nc=364
>>             where nonce='3c20da2e-b663-46fe-8edc-c756f9166d07') to
>>             database: server closed the connection unexpectedly
>>
>>             This was recently, i did an update to current master the
>>             previous version was from April, not sure if it could be
>>             an error on FS o some other issue on my box..
>>
>>
>>             PG is installed on the same server and the only thing i
>>             see from pg is "postgres[2236]: FATAL: invalid frontend
>>             message type 21", PG is installed on the same server,
>>             running on /dev/shm with the same prio as FS and  the
>>             process never stopped.
>>
>>
>>             anyone has experience this error before? any idea what it
>>             could be the cause?
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>         <http://www.freeswitchsolutions.com>
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://confluence.freeswitch.org
>>         <http://confluence.freeswitch.org>
>>         http://www.cluecon.com
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>         http://www.freeswitch.org
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>     <http://www.freeswitchsolutions.com>
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>     http://www.freeswitch.org
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From alexandr.popov at iqoption.com  Thu Jul  6 14:13:43 2017
From: alexandr.popov at iqoption.com (Alexandr Popov)
Date: Thu, 6 Jul 2017 17:13:43 +0300
Subject: [Freeswitch-users] Error sending queries to pgsql
In-Reply-To: <1d4ae6f9-d857-4cba-c780-dc5ceaee004f@wirelessmundi.com>
References: <d2f4526f-a783-a216-405d-126c253e34fd@wirelessmundi.com>
 <31066AE1-BC31-4C3F-8809-6C5B93C3B0E8@jerris.com>
 <f76beadd-9014-d279-1fb8-ea9d0048b240@wirelessmundi.com>
 <CAKeZRhvyLu+uT4_c_ZSEQVxW5zYZgrgeuAf10Qxpqn_5B-XRTA@mail.gmail.com>
 <39b55634-01fd-afea-a08f-27094f2e237c@wirelessmundi.com>
 <1d4ae6f9-d857-4cba-c780-dc5ceaee004f@wirelessmundi.com>
Message-ID: <CAKeZRhvm=NgcOxXQSBJkQQbOGvJEdwyhRunXEW8f6cskEa6d7g@mail.gmail.com>

i'm receiving two type of of error at FS are [FATAL:  invalid frontend
message type 21
]
and
  This probably means the server terminated abnormally
before or while processing the request.
[Error sending query!]


at postgers log i getting only -- invalid frontend message type 21


2017-07-06 15:26 GMT+03:00 Antonio Silva <asilva at wirelessmundi.com>:

> Hi,
>
> same results on old kernel..
> i'm trying to reproduce it on isolate machine.. for now i put my findings
> in jira https://freeswitch.org/jira/browse/FS-10474.
>
>
>
> Alexandr:
>
> What SO are you running? did you get some errors from postgres before fs
> critical message?
>
>
> Saludos / Regards / Cumprimentos,
> António silva
>
> On 07/04/2017 11:34 AM, Antonio Silva wrote:
>
> In this same box i also have data corruption with sqlite databases,
> special the ones that i use with mod_lua.. but I think that it could be
> something related with kernel.. i recently update to 4.9.x, and there was
> some changes in ext4.
>
> I don't really know how to debug this, so today i'm reverting to the old
> kernel, 4.4.x and check it happens again..
>
> If it happens again, my next move will try to reproduce this in FS..
>
> Saludos / Regards / Cumprimentos,
> António silva
>
> On 07/04/2017 10:58 AM, Alexandr Popov wrote:
>
> Seems its trouble with sockets. I have the same problem appeared about a
> month ago.
>
> 2017-06-29 19:31 GMT+03:00 Antonio Silva <asilva at wirelessmundi.com>:
>
>> Hi Michael,
>>
>> Yes, i'm trying to figure it out if is an issue in FS or external.. but
>> the message from PG i can't translate it.. i just enable more logs to see
>> if i got extra hints...
>>
>>
>> Thanks.
>>
>> Saludos / Regards / Cumprimentos,
>> António silva
>>
>> On 06/29/2017 06:22 PM, Michael Jerris wrote:
>>
>>> If you can figure out a reliable way to reproduce this issue, please
>>> file a jira with details on what causes it.
>>>
>>> On Jun 29, 2017, at 7:22 AM, Antonio Silva <asilva at wirelessmundi.com>
>>>> wrote:
>>>>
>>>> Hi all,
>>>>
>>>> i use pgsql in core and from time to time i see critical messages like
>>>> fail to send query, example:
>>>>
>>>> [CRIT] switch_pgsql.c:255 Failed to send query (update
>>>> sip_authentication set expires='1498726568',last_nc=364 where
>>>> nonce='3c20da2e-b663-46fe-8edc-c756f9166d07') to database: server
>>>> closed the connection unexpectedly
>>>>
>>>> This was recently, i did an update to current master the previous
>>>> version was from April, not sure if it could be an error on FS o some other
>>>> issue on my box..
>>>>
>>>>
>>>> PG is installed on the same server and the only thing i see from pg is
>>>> "postgres[2236]: FATAL:  invalid frontend message type 21", PG is installed
>>>> on the same server, running on /dev/shm with the same prio as FS and  the
>>>> process never stopped.
>>>>
>>>>
>>>> anyone has experience this error before? any idea what it could be the
>>>> cause?
>>>>
>>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bipin at xbipin.com  Thu Jul  6 15:39:52 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Thu, 6 Jul 2017 19:39:52 +0400
Subject: [Freeswitch-users] running FS issue without make install
Message-ID: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>

hi,

this might sound a basic question but would appreciate if any1 can guide 
me. I have 2 systems which are exactly same in software and hardware, 
both running debian with exact same packages and libraries installed. I 
git pull on one of them the latest FS master and compile it and then 
make install and everything works good, now to avoid compiling the same 
on the second machine, i just copy over the complete 
/usr/local/freeswitch to the second machine and just modify the profile 
to reflect its IP but when i run freeswitch i see it loading modules etc 
but it comes to the part loading mod_dialplan_xml and then its just 
stuck there forever and starts consuming 100% cpu.

firstly is it necessary to compile and make install on all systems and 
does make install do anything additional other than create the folders 
and copy the binaries?
secondly how do i debug whats causing the issue when running FS coz 
there is no error shown, it just comes to that line and its stuck there 
for ever


-- 
Regards,
Bipin


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From nathan.port at gmail.com  Thu Jul  6 04:44:44 2017
From: nathan.port at gmail.com (Nate)
Date: Thu, 6 Jul 2017 16:44:44 +1200
Subject: [Freeswitch-users] Dialplan: Outbound route:
 destination_number: regex question
In-Reply-To: <CAFGRPVqPCiSCevF=Rkr+S_XHPz_07yOOrwMgAVJf211JjSDEOw@mail.gmail.com>
References: <CAB+hG+gxLfN0ARXysM4HrUgaLZY0jauON01wt=D7jjNU8G8O0g@mail.gmail.com>
 <478D2D4F-2967-4A18-BFAC-5AD1C35D0E47@vallimamod.org>
 <CAB+hG+iEqsnnCHcAdrGm2uXSiubJcJ56CwsVot2FEuOK5MSq4g@mail.gmail.com>
 <CAB+hG+j0h_UABGKa08jwpzSfOS5b5nw+rLH86vuYPtM4N5evQQ@mail.gmail.com>
 <CAB+hG+iCPEdxGX0UJJFD-+mjVWo8YCO7o03LL8-Vt6XxODh3HQ@mail.gmail.com>
 <CAFGRPVqPCiSCevF=Rkr+S_XHPz_07yOOrwMgAVJf211JjSDEOw@mail.gmail.com>
Message-ID: <CAB+hG+i2vjYiJUq0kNP56KXD8xeQEUrh73OTc-OendXKvZioAw@mail.gmail.com>

Heaps of thanks David, and apologies for my lack of certainty regarding
what you mean by "trace".

I have increased debugging on all aspects and seem have turned up the
following entries from the log and thought I would ask for a sanity check
here.
Does the context look okay to you? 0223334444 is an alias for my mobile
number.

I can see where certain decisions are being made such as responding to my
INVITE with 480 at the end, but why? Is there another log I can set up or
something?

...
2017-07-06 16:10:40.694638 [DEBUG] mod_sofia.c:143 sofia/internal/
200 at 192.168.1.13 SOFIA ROUTING
2017-07-06 16:10:40.694638 [DEBUG] switch_core_state_machine.c:236
sofia/internal/200 at 192.168.1.13 Standard ROUTING
2017-07-06 16:10:40.694638 [INFO] mod_dialplan_xml.c:637 Processing 200
<200>->0223334444 in context 192.168.1.13
Dialplan: sofia/internal/200 at 192.168.1.13 parsing
[192.168.1.13->user_exists] continue=true
Dialplan: sofia/internal/200 at 192.168.1.13 Regex (PASS) [user_exists] () =~
// break=on-false
Dialplan: sofia/internal/200 at 192.168.1.13 Action
set(user_exists=${user_exists id ${destination_number} ${domain_name}})
INLINE
2017-07-06 16:10:40.733859 [DEBUG] freeswitch_lua.cpp:365 DBH handle
0x7f2ba40aa750 Connected.
2017-07-06 16:10:40.733859 [DEBUG] freeswitch_lua.cpp:382 DBH handle
0x7f2ba40aa750 released.
EXECUTE sofia/internal/200 at 192.168.1.13 set(user_exists=false)
2017-07-06 16:10:40.733859 [DEBUG] mod_dptools.c:1530 SET sofia/internal/
200 at 192.168.1.13 [user_exists]=[false]
...
///Note: after a whole lot of conditionals and such, things appear to get
interesting again:
...
2017-07-06 16:10:40.753859 [DEBUG] mod_sofia.c:198 sofia/internal/
200 at 192.168.1.13 SOFIA EXECUTE
2017-07-06 16:10:40.753859 [DEBUG] switch_core_state_machine.c:328
sofia/internal/200 at 192.168.1.13 Standard EXECUTE
ed34877a-2bed-4478-947a-13bc01ef90a7 EXECUTE sofia/internal/200 at 192.168.1.13
set(call_direction=local)
 2017-07-06 16:10:40.753859 [DEBUG] mod_dptools.c:1530 SET sofia/internal/
200 at 192.168.1.13 [call_direction]=[local]
EXECUTE sofia/internal/200 at 192.168.1.13 export(origination_callee_id_name=
0223334444)
  2017-07-06 16:10:40.753859 [DEBUG] switch_channel.c:1296 EXPORT
(export_vars) [origination_callee_id_name]=[0223334444]
EXECUTE sofia/internal/200 at 192.168.1.13 set(RFC2822_DATE=Thu, 06 Jul 2017
16:10:40 +1200)
  2017-07-06 16:10:40.753859 [DEBUG] mod_dptools.c:1530 SET sofia/internal/
200 at 192.168.1.13 [RFC2822_DATE]=[Thu, 06 Jul 2017 16:10:40 +1200]
34877a-2bed-4478-947a-13bc01ef90a7
EXECUTE sofia/internal/200 at 192.168.1.13
hash(insert/192.168.1.13-last_dial/200/0226391300)
ed34877a-2bed-4478-947a-13bc01ef90a7 EXECUTE sofia/internal/200 at 192.168.1.13
eval(not_secure)
  2017-07-06 16:10:40.753859 [NOTICE] switch_core_state_machine.c:385
sofia/internal/200 at 192.168.1.13 has executed the last dialplan instruction,
hanging up.
  2017-07-06 16:10:40.753859 [NOTICE] switch_core_state_machine.c:387
Hangup sofia/internal/200 at 192.168.1.13 [CS_EXECUTE] [NORMAL_CLEARING]
  2017-07-06 16:10:40.753859 [DEBUG] switch_core_state_machine.c:650
(sofia/internal/200 at 192.168.1.13) State EXECUTE going to sleep
  2017-07-06 16:10:40.753859 [DEBUG] switch_core_state_machine.c:584
(sofia/internal/200 at 192.168.1.13) Running State Change CS_HANGUP (Cur 1 Tot
15)
  2017-07-06 16:10:40.753859 [DEBUG] switch_core_state_machine.c:850
(sofia/internal/200 at 192.168.1.13) Callstate Change RINGING -> HANGUP
  2017-07-06 16:10:40.753859 [DEBUG] switch_core_state_machine.c:852
(sofia/internal/200 at 192.168.1.13) State HANGUP
  2017-07-06 16:10:40.753859 [DEBUG] mod_sofia.c:438 Channel sofia/internal/
200 at 192.168.1.13 hanging up, cause: NORMAL_CLEARING
2017-07-06 16:10:40.753859 [DEBUG] mod_sofia.c:577 Responding to INVITE
with: 480

Many many thanks for looking this over!

Nate.

On Thu, Jul 6, 2017 at 12:07 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Fs will fallback to Digest from ACL when the source ip is not allowed,
> this is normal.
>
> Looking at your log, it doesn't seem like the client is actually
> responding to the challenge.
>
> Can you take a trace and look at it? You should see the challenge and then
> an INVITE with credentials.
>
>
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From david.ponzone at gmail.com  Thu Jul  6 07:12:06 2017
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 6 Jul 2017 09:12:06 +0200
Subject: [Freeswitch-users] G729 annexb=no not forwarded to leg B
In-Reply-To: <CABMX_w9XubhPMSDVyG5_ksP4mnBfxEjLaubAiP_jSm=zG8RDrQ@mail.gmail.com>
References: <F43A397B-BE5D-4B39-AFF7-B18B5032E3AB@gmail.com>
 <CABMX_w9XubhPMSDVyG5_ksP4mnBfxEjLaubAiP_jSm=zG8RDrQ@mail.gmail.com>
Message-ID: <2B1D31C5-AC67-45B7-AB02-63C65AB8094A@gmail.com>

Italo,

That’s already what I use in a regexp condition on ${switch_r_sdp}.

My point was more to understand if it was the expected behavior. One would expect this parameter if requested by A to be forwarded to B by default.


> Le 5 juil. 2017 à 21:25, Ítalo Rossi <italo at freeswitch.org> a écrit :
> 
> Try:
> 
> <action application="export" data="rtp_append_audio_sdp=a=fmtp:18 annexb=no"/>
> 
> On Wed, Jul 5, 2017 at 12:03 PM, David Ponzone <david.ponzone at gmail.com <mailto:david.ponzone at gmail.com>> wrote:
> All,
> 
> If I receive an INVITE from Leg A (Patton GW) with G729 and annexb=no in the SDP, I would expect annexb=no to be automatically inserted in the SDP to Leg B.
> It seems either with export absolute_codec_string=G729, and without it (so I send G729 and G711 to Leg B), annexb=no is not sent to Leg B.
> 
> Is there any parameters or any configuration issue which may lead to annex=no being suppressed (which is an issue, as some carriers default to annexb=yes), or is it the normal behavior, meaning I have to manually export it to leg B if it was requested by A ?
> 
> Thank you
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> -- 
> Ítalo Rossi
> italo at freeswitch.org <mailto:italo at freeswitch.org>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From aviv at ironsip.com  Thu Jul  6 13:48:19 2017
From: aviv at ironsip.com (Aviv Shaham)
Date: Thu, 06 Jul 2017 06:48:19 -0700
Subject: [Freeswitch-users] freeswitch.Session with LUA
In-Reply-To: <249445539.185.1499335849571.JavaMail.sfa@EDISTAR-SFA>
References: <249445539.185.1499335849571.JavaMail.sfa@EDISTAR-SFA>
Message-ID: <1499348899.737544.1032327552.4D26C137@webmail.messagingengine.com>

Have you tried transfer_ringback?

https://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones#Transfer_Ringback

On Thu, Jul 6, 2017, at 03:11 AM, Stefano Favaro wrote:
> 
> Hello,
> 
> I'm writing a simple script with LUA to connect an inbound call to e new outbound call.> Everything work fine but I can't set the *ringback* tone nor a waiting music on the leg A and also the *call_timeout* variable.> The legA is in silence while the legB is ringing and the call_timeout is always the default 60 seconds.> What is wrong with this code?
> 
> Thanks 
> 
> This is the code:
> ...
> legA:streamFile("/usr/share/freeswitch/sounds/wait.wav")
> 
> legA:setVariable("ringback", "$${uk-ring}")
> legA:setVariable("call_timeout", 30)
> 
> legB = freeswitch.Session("{ignore_early_media=true, hangup_after_bridge=true, origination_caller_id_number=1234}sofia/internal/0123456789 at myserver)> 
> if legB:ready() then
> freeswitch.bridge(legA, legB)
> end
> 
> 
> 
> Stefano Favaro 
> Sviluppo Servizi ed Applicazioni
> _____________________
> 
> 
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From italo at freeswitch.org  Thu Jul  6 16:24:32 2017
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 06 Jul 2017 16:24:32 +0000
Subject: [Freeswitch-users] G729 annexb=no not forwarded to leg B
In-Reply-To: <2B1D31C5-AC67-45B7-AB02-63C65AB8094A@gmail.com>
References: <F43A397B-BE5D-4B39-AFF7-B18B5032E3AB@gmail.com>
 <CABMX_w9XubhPMSDVyG5_ksP4mnBfxEjLaubAiP_jSm=zG8RDrQ@mail.gmail.com>
 <2B1D31C5-AC67-45B7-AB02-63C65AB8094A@gmail.com>
Message-ID: <CABMX_w8BdxV7QtNEP_0UwmB_ZfvYkYbVd3vjuGYh7CKOfOd0jw@mail.gmail.com>

Got it. I think there's different behavior when using late negotiation and
early negotiation. Mike can explain a little more. We *may* have a bug.
Em qui, 6 de jul de 2017 às 12:59, David Ponzone <david.ponzone at gmail.com>
escreveu:

> Italo,
>
> That’s already what I use in a regexp condition on ${switch_r_sdp}.
>
> My point was more to understand if it was the expected behavior. One would
> expect this parameter if requested by A to be forwarded to B by default.
>
>
>
> Le 5 juil. 2017 à 21:25, Ítalo Rossi <italo at freeswitch.org> a écrit :
>
> Try:
>
> <action application="export" data="rtp_append_audio_sdp=a=fmtp:18
> annexb=no"/>
>
> On Wed, Jul 5, 2017 at 12:03 PM, David Ponzone <david.ponzone at gmail.com>
> wrote:
>
>> All,
>>
>> If I receive an INVITE from Leg A (Patton GW) with G729 and annexb=no in
>> the SDP, I would expect annexb=no to be automatically inserted in the SDP
>> to Leg B.
>> It seems either with export absolute_codec_string=G729, and without it
>> (so I send G729 and G711 to Leg B), annexb=no is not sent to Leg B.
>>
>> Is there any parameters or any configuration issue which may lead to
>> annex=no being suppressed (which is an issue, as some carriers default to
>> annexb=yes), or is it the normal behavior, meaning I have to manually
>> export it to leg B if it was requested by A ?
>>
>> Thank you
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> --
> Ítalo Rossi
> italo at freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From italo at freeswitch.org  Thu Jul  6 16:25:54 2017
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 06 Jul 2017 16:25:54 +0000
Subject: [Freeswitch-users] mod_callcenter - Unloading queue with
 abandoned members
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E867BE92@mbx-01.sysconfig.co.uk>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E867BE92@mbx-01.sysconfig.co.uk>
Message-ID: <CABMX_w_i36hBQBrsv4-RPev1gQp_619=DN1j7UzwsXDqoQzt8w@mail.gmail.com>

Fixed yesterday.
Em qui, 6 de jul de 2017 às 09:10, Shaun Stokes <
shaun.stokes at itec-support.co.uk> escreveu:

> When using the command ‘callcenter_config queue unload queuename at context’
> this works great except when there are abandoned members in the queue, in
> this situation we get the following error repeatedly:
>
> [WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this
> member
>
> [WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this
> member
>
> [WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this
> member
>
> [WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this
> member
>
> [WARNING] mod_callcenter.c:2031 Queue queue.2 not found locally, skip this
> member
>
>
>
> The only solution that seems to work is re-add the queue and wait for the
> abandoned members to disappear as defined by ‘discard-abandoned-after’.
>
>
>
> Is this normal behaviour in which case what’s the command to clear
> abandoned members from the queue before it’s unloaded, or is this a bug?
>
>
>
> Thanks,
>
> Shaun
> Shaun Stokes - Infrastructure Analyst
> T : 01453 700713
> E : shaun.stokes at itec-support.co.uk
> W : www.itec-support.co.uk
> Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road,
> Stroud, Gloucestershire GL5 3QF
> Company No. 06908001
>
> CONFIDENTIALITY NOTICE
> This communication and the information it contains are intended for the
> person or organisation to which it is addressed. Its contents are
> confidential and may be protected in law. Unauthorised use, copying or
> disclosure of any of it may be unlawful. If you are not the intended
> recipient, please contact us immediately.
> The contents of any attachments in this e-mail may contain software
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> has taken every reasonable precaution to minimise this risk, we cannot
> accept liability for any damage which you sustain as a result of software
> viruses. You should carry out your own virus checking procedure before
> opening any attachment.
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From covici at ccs.covici.com  Thu Jul  6 16:59:50 2017
From: covici at ccs.covici.com (John Covici)
Date: Thu, 06 Jul 2017 12:59:50 -0400
Subject: [Freeswitch-users] running FS issue without make install
In-Reply-To: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
References: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
Message-ID: <m34lupo60p.wl-covici@ccs.covici.com>

I think make install does some things, so its bets to compile on all
systems.  Just in case you have some slight differences or something.

On Thu, 06 Jul 2017 11:39:52 -0400,
Bipin Patel wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; windows-1252 (7bit)>]
> [1.2  <text/html; windows-1252 (7bit)>]
> hi,
> 
> this might sound a basic question but would appreciate if any1 can guide me. I have 2 systems which are exactly same in software and hardware, both running debian with exact same packages and libraries installed. I git pull on one of
> them the latest FS master and compile it and then make install and everything works good, now to avoid compiling the same on the second machine, i just copy over the complete /usr/local/freeswitch to the second machine and just modify
> the profile to reflect its IP but when i run freeswitch i see it loading modules etc but it comes to the part loading mod_dialplan_xml and then its just stuck there forever and starts consuming 100% cpu.
> 
> firstly is it necessary to compile and make install on all systems and does make install do anything additional other than create the folders and copy the binaries?
> secondly how do i debug whats causing the issue when running FS coz there is no error shown, it just comes to that line and its stuck there for ever
> 
> -- 
> Regards,
> Bipin
> 
> ---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From rmundkowsky at ets.org  Thu Jul  6 16:55:27 2017
From: rmundkowsky at ets.org (Mundkowsky, Robert)
Date: Thu, 6 Jul 2017 16:55:27 +0000
Subject: [Freeswitch-users] running FS issue without make install
In-Reply-To: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
References: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
Message-ID: <BN6PR07MB35562707A6721BBD8A3999C3BED50@BN6PR07MB3556.namprd07.prod.outlook.com>

This might be easier:

./configure --prefix=/foobar/freeswitch
make
make install

Then everything except for boot scripts in under /foobar/freeswitch

And hopefully that will pick-up anything you missed.

Robert Mundkowsky

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bipin Patel
Sent: Thursday, July 6, 2017 11:40 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] running FS issue without make install

hi,

this might sound a basic question but would appreciate if any1 can guide me. I have 2 systems which are exactly same in software and hardware, both running debian with exact same packages and libraries installed. I git pull on one of them the latest FS master and compile it and then make install and everything works good, now to avoid compiling the same on the second machine, i just copy over the complete /usr/local/freeswitch to the second machine and just modify the profile to reflect its IP but when i run freeswitch i see it loading modules etc but it comes to the part loading mod_dialplan_xml and then its just stuck there forever and starts consuming 100% cpu.

firstly is it necessary to compile and make install on all systems and does make install do anything additional other than create the folders and copy the binaries?
secondly how do i debug whats causing the issue when running FS coz there is no error shown, it just comes to that line and its stuck there for ever

--
Regards,
Bipin

________________________________

________________________________

This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited.


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From bipin at xbipin.com  Thu Jul  6 20:41:44 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 07 Jul 2017 00:41:44 +0400
Subject: [Freeswitch-users] running FS issue without make install
In-Reply-To: <m34lupo60p.wl-covici@ccs.covici.com>
References: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
 <m34lupo60p.wl-covici@ccs.covici.com>
Message-ID: <15d19a31340.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Hi,

Well there is no variation in software or libraries between the two, we 
always run identical commands on both. It would be great if anyone can shed 
some more light on what make install does extra, like add any symbolic 
links etc or something different with gets missed out on manually copying 
over the files as I plan to span out multiple installs as quickly as I can 
rather than have to compile in each machine. Might as well document what I 
learn in the process as I have already noted down things I did so far and 
more over FS seems to run fine until it's trying to load mod dialplan xml, 
it gets the IP and also the wan IP and sets up the profile and codecs etc 
just fine so if it was some dependency stuff some error would have popped up.




On July 6, 2017 9:02:28 PM John Covici <covici at ccs.covici.com> wrote:

> I think make install does some things, so its bets to compile on all
> systems.  Just in case you have some slight differences or something.
>
> On Thu, 06 Jul 2017 11:39:52 -0400,
> Bipin Patel wrote:
>>
>> [1  <multipart/alternative (7bit)>]
>> [1.1  <text/plain; windows-1252 (7bit)>]
>> [1.2  <text/html; windows-1252 (7bit)>]
>> hi,
>>
>> this might sound a basic question but would appreciate if any1 can guide 
>> me. I have 2 systems which are exactly same in software and hardware, both 
>> running debian with exact same packages and libraries installed. I git pull 
>> on one of
>> them the latest FS master and compile it and then make install and 
>> everything works good, now to avoid compiling the same on the second 
>> machine, i just copy over the complete /usr/local/freeswitch to the second 
>> machine and just modify
>> the profile to reflect its IP but when i run freeswitch i see it loading 
>> modules etc but it comes to the part loading mod_dialplan_xml and then its 
>> just stuck there forever and starts consuming 100% cpu.
>>
>> firstly is it necessary to compile and make install on all systems and does 
>> make install do anything additional other than create the folders and copy 
>> the binaries?
>> secondly how do i debug whats causing the issue when running FS coz there 
>> is no error shown, it just comes to that line and its stuck there for ever
>>
>> --
>> Regards,
>> Bipin
>>
>> ---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
>> [2  <text/plain; utf-8 (base64)>]
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici
>          covici at ccs.covici.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From bipin at xbipin.com  Thu Jul  6 20:44:19 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 07 Jul 2017 00:44:19 +0400
Subject: [Freeswitch-users] running FS issue without make install
In-Reply-To: <BN6PR07MB35562707A6721BBD8A3999C3BED50@BN6PR07MB3556.namprd07.prod.outlook.com>
References: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
 <BN6PR07MB35562707A6721BBD8A3999C3BED50@BN6PR07MB3556.namprd07.prod.outlook.com>
Message-ID: <15d19a570b8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Well I'm trying to avoid making and not so great in Linux I don't know what 
that configure command would do


On July 6, 2017 11:51:26 PM "Mundkowsky, Robert" <rmundkowsky at ets.org> wrote:

> This might be easier:
>
> ./configure --prefix=/foobar/freeswitch
> make
> make install
>
> Then everything except for boot scripts in under /foobar/freeswitch
>
> And hopefully that will pick-up anything you missed.
>
> Robert Mundkowsky
>
> From: FreeSWITCH-users 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bipin Patel
> Sent: Thursday, July 6, 2017 11:40 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] running FS issue without make install
>
> hi,
>
> this might sound a basic question but would appreciate if any1 can guide 
> me. I have 2 systems which are exactly same in software and hardware, both 
> running debian with exact same packages and libraries installed. I git pull 
> on one of them the latest FS master and compile it and then make install 
> and everything works good, now to avoid compiling the same on the second 
> machine, i just copy over the complete /usr/local/freeswitch to the second 
> machine and just modify the profile to reflect its IP but when i run 
> freeswitch i see it loading modules etc but it comes to the part loading 
> mod_dialplan_xml and then its just stuck there forever and starts consuming 
> 100% cpu.
>
> firstly is it necessary to compile and make install on all systems and does 
> make install do anything additional other than create the folders and copy 
> the binaries?
> secondly how do i debug whats causing the issue when running FS coz there 
> is no error shown, it just comes to that line and its stuck there for ever
>
> --
> Regards,
> Bipin
>
> ________________________________
>
> ________________________________
>
> This e-mail and any files transmitted with it may contain privileged or 
> confidential information. It is solely for use by the individual for whom 
> it is intended, even if addressed incorrectly. If you received this e-mail 
> in error, please notify the sender; do not disclose, copy, distribute, or 
> take any action in reliance on the contents of this information; and delete 
> it from your system. Any other use of this e-mail is prohibited.
>
>
> Thank you for your compliance.
>
> ________________________________
>
>
>
> ----------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Thu Jul  6 20:51:12 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 6 Jul 2017 16:51:12 -0400
Subject: [Freeswitch-users] running FS issue without make install
In-Reply-To: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
References: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
Message-ID: <B8687012-F3A0-4863-AA83-C8658C6A648D@jerris.com>

This is what packages are for.

> On Jul 6, 2017, at 11:39 AM, Bipin Patel <bipin at xbipin.com> wrote:
> 
> hi,
> 
> this might sound a basic question but would appreciate if any1 can guide me. I have 2 systems which are exactly same in software and hardware, both running debian with exact same packages and libraries installed. I git pull on one of them the latest FS master and compile it and then make install and everything works good, now to avoid compiling the same on the second machine, i just copy over the complete /usr/local/freeswitch to the second machine and just modify the profile to reflect its IP but when i run freeswitch i see it loading modules etc but it comes to the part loading mod_dialplan_xml and then its just stuck there forever and starts consuming 100% cpu.
> 
> firstly is it necessary to compile and make install on all systems and does make install do anything additional other than create the folders and copy the binaries?
> secondly how do i debug whats causing the issue when running FS coz there is no error shown, it just comes to that line and its stuck there for ever
> 

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From luca.pradovera at gmail.com  Thu Jul  6 20:57:16 2017
From: luca.pradovera at gmail.com (Luca Pradovera)
Date: Thu, 6 Jul 2017 22:57:16 +0200
Subject: [Freeswitch-users] Mobile WebRTC session
Message-ID: <CAKBQ8W01VVm8FJ8fByTLx1aMsJKncV=+6bUXG5uPLBNMhw0Y8g@mail.gmail.com>

Hello,
I am trying to diagnose the reason why we only have one-way audio (browser
to device, not the reverse) in a WebRTC call using Verto, which does not
really matter in this particular case as signaling works great.

This is where the issue starts appearing:
https://gist.github.com/lpradovera/607de959022d06adf4ff6f8fcfc97510#file-gistfile1-txt-L153

Based on the fact that payload 96 might be SRTP, I am postulating this is
is a TLS problem and not a codec issue, but I am really grasping at straws.

Does anyone have a pointer, please?

Thanks!

Luca
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From bipin at xbipin.com  Thu Jul  6 21:05:49 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 07 Jul 2017 01:05:49 +0400
Subject: [Freeswitch-users] running FS issue without make install
In-Reply-To: <B8687012-F3A0-4863-AA83-C8658C6A648D@jerris.com>
References: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
 <B8687012-F3A0-4863-AA83-C8658C6A648D@jerris.com>
Message-ID: <15d19b91fc8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Yes I understand the importance of packages but that's fine for Debian 
installs and I have many small clients who would have less than 10 exts and 
in our testing giving them fs on raspbian is just fine and hardware like 
the raspberry pi works brilliant for calls, fax and most other basic things 
used in a PBX environment so if copying binary works fine on Debian my next 
task is to do the same on raspbian and I love to be on the bleeding edge of 
the latest code rather than use old packages which I guess for raspbian is 
a very old package of fs




On July 7, 2017 12:53:25 AM Michael Jerris <mike at jerris.com> wrote:

> This is what packages are for.
>
>> On Jul 6, 2017, at 11:39 AM, Bipin Patel <bipin at xbipin.com> wrote:
>>
>> hi,
>>
>> this might sound a basic question but would appreciate if any1 can guide 
>> me. I have 2 systems which are exactly same in software and hardware, both 
>> running debian with exact same packages and libraries installed. I git pull 
>> on one of them the latest FS master and compile it and then make install 
>> and everything works good, now to avoid compiling the same on the second 
>> machine, i just copy over the complete /usr/local/freeswitch to the second 
>> machine and just modify the profile to reflect its IP but when i run 
>> freeswitch i see it loading modules etc but it comes to the part loading 
>> mod_dialplan_xml and then its just stuck there forever and starts consuming 
>> 100% cpu.
>>
>> firstly is it necessary to compile and make install on all systems and does 
>> make install do anything additional other than create the folders and copy 
>> the binaries?
>> secondly how do i debug whats causing the issue when running FS coz there 
>> is no error shown, it just comes to that line and its stuck there for ever
>>
>
>
>
>
> ----------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Thu Jul  6 21:08:41 2017
From: mike at jerris.com (Michael Jerris)
Date: Thu, 6 Jul 2017 17:08:41 -0400
Subject: [Freeswitch-users] running FS issue without make install
In-Reply-To: <15d19b91fc8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
References: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
 <B8687012-F3A0-4863-AA83-C8658C6A648D@jerris.com>
 <15d19b91fc8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
Message-ID: <08FC3381-8334-46AB-8F3E-819AAFED45AD@jerris.com>

so build packages… 

> On Jul 6, 2017, at 5:05 PM, Bipin Patel <bipin at xbipin.com> wrote:
> 
> Yes I understand the importance of packages but that's fine for Debian installs and I have many small clients who would have less than 10 exts and in our testing giving them fs on raspbian is just fine and hardware like the raspberry pi works brilliant for calls, fax and most other basic things used in a PBX environment so if copying binary works fine on Debian my next task is to do the same on raspbian and I love to be on the bleeding edge of the latest code rather than use old packages which I guess for raspbian is a very old package of fs
> 
> On July 7, 2017 12:53:25 AM Michael Jerris <mike at jerris.com> wrote:
> 
>> This is what packages are for.
>> 
>>> On Jul 6, 2017, at 11:39 AM, Bipin Patel <bipin at xbipin.com <mailto:bipin at xbipin.com>> wrote:
>>> 
>>> hi,
>>> 
>>> this might sound a basic question but would appreciate if any1 can guide me. I have 2 systems which are exactly same in software and hardware, both running debian with exact same packages and libraries installed. I git pull on one of them the latest FS master and compile it and then make install and everything works good, now to avoid compiling the same on the second machine, i just copy over the complete /usr/local/freeswitch to the second machine and just modify the profile to reflect its IP but when i run freeswitch i see it loading modules etc but it comes to the part loading mod_dialplan_xml and then its just stuck there forever and starts consuming 100% cpu.
>>> 
>>> firstly is it necessary to compile and make install on all systems and does make install do anything additional other than create the folders and copy the binaries?
>>> secondly how do i debug whats causing the issue when running FS coz there is no error shown, it just comes to that line and its stuck there for ever
>>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting%40freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users%40lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.villasmil.work at gmail.com  Thu Jul  6 21:11:40 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 6 Jul 2017 23:11:40 +0200
Subject: [Freeswitch-users] sip to webrtc - sdp invalid description
Message-ID: <CAFGRPVoQB8mH3a8JYbY3wP1L=L3x-zdZF-kK1o+ocW5GzSYt-A@mail.gmail.com>

Hello guys,

I have this setup:

Zoiper-->Kamailio->fs->kamailio->webrtc client(s)

Whenever webrtc clients call each other, calls are ok.
But when the zoiper (regular sip/tcp) calls, the browsers complaint about:

"no ice-ufrag" (firefox) or
"Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
(Chrome).

I am setting the sdp in freeswitch as:

      <action application="set" data="proxy_media=false"/>
      <action application="set" data="bypass_media=false"/>
      <action application="export" data="zrtp_secure_media=false"/>
      <action application="export" data="nolocal:rtp_secure_media=true"/>

and the actual sdp is:

v=0
o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4
s=FreeSWITCH
c=IN IP4 1.2.3.4
t=0 0
m=audio 40954 RTP/SAVP 8 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVrVfjA0Im
Wgu5WyLK0vtg60RTk6Koe8c0sRkjU
a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+D5NVoKFFw06I0Xok/9KbRystQ
a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+
ETPuCOiOIxjFLRjbFbDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg
a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:evU8MzAtiSHwKb95s4V9IAMpmok06k
W9ZGDgH3/Lc3ZytVn2SR4
a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:KKDkT0DssohSeKFsX6tbixRhwYdiIh
E6r3u5CCVA
a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:D3JgGuOlIxXfHGdqf7lKWqNDAIiJrb
OqOKb+erlhPQtBKF4wzomjbN0sBIiE4w
a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:c12ebWWzZ1cqZN0v5C5uYzdvtfnw6A
ARU3+jGA0WzTSDlDd20vI
a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:nPZku7S6hlz2OPAff8T9I8sNFzuZzi
Ng64KuvfNS
a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3dhTqD0X
W8FAOLNtmP
a=ptime:20
a=nortpproxy:yes


I'm pretty new to the webrtc stuff, so any help is greatly appreciated!

Thanks!


David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337 <+34%20669%2044%2083%2037>
ᐧ
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From joelists at tm.net.uk  Thu Jul  6 22:35:55 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Thu, 6 Jul 2017 23:35:55 +0100
Subject: [Freeswitch-users] redirect context not working
Message-ID: <D7E55E6C-D1C7-41EA-AABC-BDE84C0D34E8@tm.net.uk>

Hi Guys

I have setup a context called redirect to catch calls sent back from a redirect server with a 300 multiple choices message.

However I get the following error when a call goes through.

2017-07-06 23:14:27.403150 [DEBUG] switch_core_state_machine.c:236 sofia/external/sipp at 185.35.228.51:5060 Standard ROUTING
2017-07-06 23:14:27.403150 [INFO] mod_dialplan_xml.c:637 Processing sipp <sipp>->441554555666 in context redirect
2017-07-06 23:14:27.403150 [INFO] switch_core_state_machine.c:311 No Route, Aborting
2017-07-06 23:14:27.403150 [NOTICE] switch_core_state_machine.c:312 Hangup sofia/external/sipp at 185.35.228.51:5060 [CS_ROUTING] [NO_ROUTE_DESTINATION]


I have a redirect.xml file as follows in my conf/dialplan

<include>
  <context name="redirect">

   <X-PRE-PROCESS cmd="include" data="redirect/*.xml"/>
  </context>
</include>


and then a 00_redirect.xml in conf/dialplan/redirect

<include>
  <context name="redirect">
    <extension name="Redirect Calls">
        <condition>
  <action application="log" data="INFO  Were in the Redirect Context!!"/>

            <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${sip_redirect_dialstring}" />
            <action application="hangup" data="${bridge_hangup_cause}"/>


        </condition>
    </extension>
  </context>
</include>

Can anyone give me an idea why the call is not being picked up in the Redirect Calls extension?

Regards
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From asilva at wirelessmundi.com  Thu Jul  6 22:50:04 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Fri, 7 Jul 2017 00:50:04 +0200
Subject: [Freeswitch-users] redirect context not working
In-Reply-To: <D7E55E6C-D1C7-41EA-AABC-BDE84C0D34E8@tm.net.uk>
References: <D7E55E6C-D1C7-41EA-AABC-BDE84C0D34E8@tm.net.uk>
Message-ID: <d9f253a2-8b57-21f3-200f-bcb979febbfe@wirelessmundi.com>

it looks you are including a context inside a context..


<include>
<context name="redirect">
<X-PRE-PROCESS cmd="include" data="redirect/*.xml"/>
</context>
</include>


the correct one:

<include>
   <extension name="Redirect Calls">
       <condition>
          <action application="log" data="INFO  Were in the Redirect 
Context!!"/>
           <action application="bridge" 
data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${sip_redirect_dialstring}" 
/>
           <action application="hangup" data="${bridge_hangup_cause}"/>
       </condition>
   </extension>
</include>



On 07/07/2017 12:35 AM, Joseph Waite wrote:
> Hi Guys
>
> I have setup a context called redirect to catch calls sent back from a 
> redirect server with a 300 multiple choices message.
>
> However I get the following error when a call goes through.
>
> 2017-07-06 23:14:27.403150 [DEBUG] switch_core_state_machine.c:236 
> sofia/external/sipp at 185.35.228.51 
> <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard ROUTING
> 2017-07-06 23:14:27.403150 [INFO] mod_dialplan_xml.c:637 Processing 
> sipp <sipp>->441554555666 in context redirect
> 2017-07-06 23:14:27.403150 [INFO] switch_core_state_machine.c:311 No 
> Route, Aborting
> 2017-07-06 23:14:27.403150 [NOTICE] switch_core_state_machine.c:312 
> Hangup sofia/external/sipp at 185.35.228.51 
> <mailto:sofia/external/sipp at 185.35.228.51>:5060 [CS_ROUTING] 
> [NO_ROUTE_DESTINATION]
>
>
> I have a redirect.xml file as follows in my conf/dialplan
>
> <include>
> <context name="redirect">
>
> <X-PRE-PROCESS cmd="include" data="redirect/*.xml"/>
> </context>
> </include>
>
>
> and then a 00_redirect.xml in conf/dialplan/redirect
>
> <include>
>   <context name="redirect">
>     <extension name="Redirect Calls">
>         <condition>
>   <action application="log" data="INFO  Were in the Redirect Context!!"/>
>
>             <action application="bridge" 
> data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${sip_redirect_dialstring}" 
> />
>             <action application="hangup" data="${bridge_hangup_cause}"/>
>
>
>         </condition>
>     </extension>
>   </context>
> </include>
>
> Can anyone give me an idea why the call is not being picked up in the 
> Redirect Calls extension?
>
> Regards
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From smrdoshi at gmail.com  Fri Jul  7 03:00:09 2017
From: smrdoshi at gmail.com (Samir Doshi)
Date: Fri, 7 Jul 2017 08:30:09 +0530
Subject: [Freeswitch-users] G729 annexb=no not forwarded to leg B
In-Reply-To: <CABMX_w9XubhPMSDVyG5_ksP4mnBfxEjLaubAiP_jSm=zG8RDrQ@mail.gmail.com>
References: <F43A397B-BE5D-4B39-AFF7-B18B5032E3AB@gmail.com>
 <CABMX_w9XubhPMSDVyG5_ksP4mnBfxEjLaubAiP_jSm=zG8RDrQ@mail.gmail.com>
Message-ID: <CAKnxTonoBh_FYn1pegrVS92_EUhu2nK-uN_pX7a0G-YQO=kKpA@mail.gmail.com>

Along with that try,

https://wiki.freeswitch.org/wiki/Variable_inherit_codec


On Jul 6, 2017 12:57 AM, "Ítalo Rossi" <italo at freeswitch.org> wrote:

> Try:
>
> <action application="export" data="rtp_append_audio_sdp=a=fmtp:18
> annexb=no"/>
>
> On Wed, Jul 5, 2017 at 12:03 PM, David Ponzone <david.ponzone at gmail.com>
> wrote:
>
>> All,
>>
>> If I receive an INVITE from Leg A (Patton GW) with G729 and annexb=no in
>> the SDP, I would expect annexb=no to be automatically inserted in the SDP
>> to Leg B.
>> It seems either with export absolute_codec_string=G729, and without it
>> (so I send G729 and G711 to Leg B), annexb=no is not sent to Leg B.
>>
>> Is there any parameters or any configuration issue which may lead to
>> annex=no being suppressed (which is an issue, as some carriers default to
>> annexb=yes), or is it the normal behavior, meaning I have to manually
>> export it to leg B if it was requested by A ?
>>
>> Thank you
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> --
> Ítalo Rossi
> italo at freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bipin at xbipin.com  Fri Jul  7 04:47:29 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 07 Jul 2017 08:47:29 +0400
Subject: [Freeswitch-users] running FS issue without make install
In-Reply-To: <08FC3381-8334-46AB-8F3E-819AAFED45AD@jerris.com>
References: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
 <B8687012-F3A0-4863-AA83-C8658C6A648D@jerris.com>
 <15d19b91fc8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <08FC3381-8334-46AB-8F3E-819AAFED45AD@jerris.com>
Message-ID: <15d1b5fcae8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>

Only if I knew how to I would but I don't.

So I guess I won't be getting a answer to the main question or probably 
other users also don't know what make install does additional than copy 
binaries




On July 7, 2017 1:10:34 AM Michael Jerris <mike at jerris.com> wrote:

> so build packages…
>
>> On Jul 6, 2017, at 5:05 PM, Bipin Patel <bipin at xbipin.com> wrote:
>>
>> Yes I understand the importance of packages but that's fine for Debian 
>> installs and I have many small clients who would have less than 10 exts and 
>> in our testing giving them fs on raspbian is just fine and hardware like 
>> the raspberry pi works brilliant for calls, fax and most other basic things 
>> used in a PBX environment so if copying binary works fine on Debian my next 
>> task is to do the same on raspbian and I love to be on the bleeding edge of 
>> the latest code rather than use old packages which I guess for raspbian is 
>> a very old package of fs
>>
>> On July 7, 2017 12:53:25 AM Michael Jerris <mike at jerris.com> wrote:
>>
>>> This is what packages are for.
>>>
>>>> On Jul 6, 2017, at 11:39 AM, Bipin Patel <bipin at xbipin.com 
>>>> <mailto:bipin at xbipin.com>> wrote:
>>>>
>>>> hi,
>>>>
>>>> this might sound a basic question but would appreciate if any1 can guide 
>>>> me. I have 2 systems which are exactly same in software and hardware, both 
>>>> running debian with exact same packages and libraries installed. I git pull 
>>>> on one of them the latest FS master and compile it and then make install 
>>>> and everything works good, now to avoid compiling the same on the second 
>>>> machine, i just copy over the complete /usr/local/freeswitch to the second 
>>>> machine and just modify the profile to reflect its IP but when i run 
>>>> freeswitch i see it loading modules etc but it comes to the part loading 
>>>> mod_dialplan_xml and then its just stuck there forever and starts consuming 
>>>> 100% cpu.
>>>>
>>>> firstly is it necessary to compile and make install on all systems and does 
>>>> make install do anything additional other than create the folders and copy 
>>>> the binaries?
>>>> secondly how do i debug whats causing the issue when running FS coz there 
>>>> is no error shown, it just comes to that line and its stuck there for ever
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting%40freeswitch.org>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org 
>>> <mailto:FreeSWITCH-users%40lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
>>> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org 
>>> <http://www.freeswitch.org/>_________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> ----------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Fri Jul  7 05:22:13 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 7 Jul 2017 00:22:13 -0500
Subject: [Freeswitch-users] ClueCon Hotel is filling up! ACT NOW and save
	$300
Message-ID: <CAKbxfG8DqKRDYxqpWbneViPyJEOQdOAr6YAr98ZZjpZwwsyGfA@mail.gmail.com>

Use the code CCJulySM2017 and save $300
Help support the FS Community and have fun in the process!


​

-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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

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

https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
​
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From joelists at tm.net.uk  Fri Jul  7 05:43:42 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Fri, 7 Jul 2017 06:43:42 +0100
Subject: [Freeswitch-users] redirect context not working
In-Reply-To: <d9f253a2-8b57-21f3-200f-bcb979febbfe@wirelessmundi.com>
References: <D7E55E6C-D1C7-41EA-AABC-BDE84C0D34E8@tm.net.uk>
 <d9f253a2-8b57-21f3-200f-bcb979febbfe@wirelessmundi.com>
Message-ID: <7F0D46A2-B06E-4765-83C6-D4B1D66FDE1B@tm.net.uk>

Thank you, hd been looking at that for over an hour and couldn’t work it out. Knew it would be something stupid I had done like that!

Regards
> On 6 Jul 2017, at 23:50, Antonio Silva <asilva at wirelessmundi.com> wrote:
> 
> it looks you are including a context inside a context..
> 
> 
> <include>
>   <context name="redirect">
>    <X-PRE-PROCESS cmd="include" data="redirect/*.xml"/>
>   </context>
> </include>
> 
> 
> the correct one:
> 
> <include>
>     <extension name="Redirect Calls">
>         <condition>
>              <action application="log" data="INFO  Were in the Redirect Context!!"/>
>             <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${sip_redirect_dialstring}" />
>             <action application="hangup" data="${bridge_hangup_cause}"/>
>         </condition>
>     </extension>
> </include>
> 
> 
> 
> On 07/07/2017 12:35 AM, Joseph Waite wrote:
>> Hi Guys
>> 
>> I have setup a context called redirect to catch calls sent back from a redirect server with a 300 multiple choices message.
>> 
>> However I get the following error when a call goes through.
>> 
>> 2017-07-06 23:14:27.403150 [DEBUG] switch_core_state_machine.c:236 sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 Standard ROUTING
>> 2017-07-06 23:14:27.403150 [INFO] mod_dialplan_xml.c:637 Processing sipp <sipp>->441554555666 in context redirect
>> 2017-07-06 23:14:27.403150 [INFO] switch_core_state_machine.c:311 No Route, Aborting
>> 2017-07-06 23:14:27.403150 [NOTICE] switch_core_state_machine.c:312 Hangup sofia/external/sipp at 185.35.228.51 <mailto:sofia/external/sipp at 185.35.228.51>:5060 [CS_ROUTING] [NO_ROUTE_DESTINATION]
>> 
>> 
>> I have a redirect.xml file as follows in my conf/dialplan
>> 
>> <include>
>>   <context name="redirect">
>> 
>>    <X-PRE-PROCESS cmd="include" data="redirect/*.xml"/>
>>   </context>
>> </include>
>> 
>> 
>> and then a 00_redirect.xml in conf/dialplan/redirect
>> 
>> <include>
>>   <context name="redirect">
>>     <extension name="Redirect Calls">
>>         <condition>
>>   <action application="log" data="INFO  Were in the Redirect Context!!"/>
>> 
>>             <action application="bridge" data="{sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${sip_redirect_dialstring}" />
>>             <action application="hangup" data="${bridge_hangup_cause}"/>
>> 
>> 
>>         </condition>
>>     </extension>
>>   </context>
>> </include>
>> 
>> Can anyone give me an idea why the call is not being picked up in the Redirect Calls extension?
>> 
>> Regards
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sebastian_ml at gmx.net  Fri Jul  7 06:17:39 2017
From: sebastian_ml at gmx.net (Sebastian Kemper)
Date: Fri, 7 Jul 2017 08:17:39 +0200
Subject: [Freeswitch-users] running FS issue without make install
In-Reply-To: <15d1b5fcae8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
References: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
 <B8687012-F3A0-4863-AA83-C8658C6A648D@jerris.com>
 <15d19b91fc8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <08FC3381-8334-46AB-8F3E-819AAFED45AD@jerris.com>
 <15d1b5fcae8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
Message-ID: <20170707061739.GA2631@wolfgang.lan>

On Fri, Jul 07, 2017 at 08:47:29AM +0400, Bipin Patel wrote:
> Only if I knew how to I would but I don't.
> 
> So I guess I won't be getting a answer to the main question or
> probably other users also don't know what make install does additional
> than copy binaries
> 
> 

Hi Bipin,

You could try out OpenWrt/LEDE. They have many target devices. There are
also images for Raspberry Pi.

Right now FS is only built for the trunk snapshots. But the next major
release should include FS as well.

For model 3 the target snapshot is here:
https://downloads.lede-project.org/snapshots/targets/brcm2708/bcm2710/

The telephony packages for model 3 are here if you want to check whether
all modules you need are included:
https://downloads.lede-project.org/snapshots/packages/aarch64_cortex-a53_neon-vfpv4/telephony/

Kind regards,
Sebastian

> 
> 
> On July 7, 2017 1:10:34 AM Michael Jerris <mike at jerris.com> wrote:
> 
> > so build packages…
> >
> >> On Jul 6, 2017, at 5:05 PM, Bipin Patel <bipin at xbipin.com> wrote:
> >>
> >> Yes I understand the importance of packages but that's fine for
> >> Debian installs and I have many small clients who would have less
> >> than 10 exts and in our testing giving them fs on raspbian is just
> >> fine and hardware like the raspberry pi works brilliant for calls,
> >> fax and most other basic things used in a PBX environment so if
> >> copying binary works fine on Debian my next task is to do the same
> >> on raspbian and I love to be on the bleeding edge of the latest
> >> code rather than use old packages which I guess for raspbian is a
> >> very old package of fs
> >>
> >> On July 7, 2017 12:53:25 AM Michael Jerris <mike at jerris.com> wrote:
> >>
> >>> This is what packages are for.
> >>>
> >>>> On Jul 6, 2017, at 11:39 AM, Bipin Patel <bipin at xbipin.com
> >>>> <mailto:bipin at xbipin.com>> wrote:
> >>>>
> >>>> hi,
> >>>>
> >>>> this might sound a basic question but would appreciate if any1
> >>>> can guide me. I have 2 systems which are exactly same in software
> >>>> and hardware, both running debian with exact same packages and
> >>>> libraries installed. I git pull on one of them the latest FS
> >>>> master and compile it and then make install and everything works
> >>>> good, now to avoid compiling the same on the second machine, i
> >>>> just copy over the complete /usr/local/freeswitch to the second
> >>>> machine and just modify the profile to reflect its IP but when i
> >>>> run freeswitch i see it loading modules etc but it comes to the
> >>>> part loading mod_dialplan_xml and then its just stuck there
> >>>> forever and starts consuming 100% cpu.
> >>>>
> >>>> firstly is it necessary to compile and make install on all
> >>>> systems and does make install do anything additional other than
> >>>> create the folders and copy the binaries?  secondly how do i
> >>>> debug whats causing the issue when running FS coz there is no
> >>>> error shown, it just comes to that line and its stuck there for
> >>>> ever


From bipin at xbipin.com  Fri Jul  7 07:21:06 2017
From: bipin at xbipin.com (Bipin Patel)
Date: Fri, 7 Jul 2017 11:21:06 +0400
Subject: [Freeswitch-users] running FS issue without make install
In-Reply-To: <20170707061739.GA2631@wolfgang.lan>
References: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
 <B8687012-F3A0-4863-AA83-C8658C6A648D@jerris.com>
 <15d19b91fc8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <08FC3381-8334-46AB-8F3E-819AAFED45AD@jerris.com>
 <15d1b5fcae8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
 <20170707061739.GA2631@wolfgang.lan>
Message-ID: <205bc55c-6440-7a52-e0fb-8718bb139c6c@xbipin.com>

hi,

thanks for that info, ill look into it, for now i managed to get it to 
work, the reason it was getting stuck was due to spandsp module, 
disabling that makes it run just fine without issue, next step is to 
figure out why spandsp wont load

Regards,
Bipin



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

-------- Original Message  --------
Subject: Re: [Freeswitch-users] running FS issue without make install
From: Sebastian Kemper <sebastian_ml at gmx.net>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Date: 7/7/2017, 10:17:39 AM

> On Fri, Jul 07, 2017 at 08:47:29AM +0400, Bipin Patel wrote:
>> Only if I knew how to I would but I don't.
>>
>> So I guess I won't be getting a answer to the main question or
>> probably other users also don't know what make install does additional
>> than copy binaries
>>
>>
> 
> Hi Bipin,
> 
> You could try out OpenWrt/LEDE. They have many target devices. There are
> also images for Raspberry Pi.
> 
> Right now FS is only built for the trunk snapshots. But the next major
> release should include FS as well.
> 
> For model 3 the target snapshot is here:
> https://downloads.lede-project.org/snapshots/targets/brcm2708/bcm2710/
> 
> The telephony packages for model 3 are here if you want to check whether
> all modules you need are included:
> https://downloads.lede-project.org/snapshots/packages/aarch64_cortex-a53_neon-vfpv4/telephony/
> 
> Kind regards,
> Sebastian
> 
>>
>>
>> On July 7, 2017 1:10:34 AM Michael Jerris <mike at jerris.com> wrote:
>>
>>> so build packages…
>>>
>>>> On Jul 6, 2017, at 5:05 PM, Bipin Patel <bipin at xbipin.com> wrote:
>>>>
>>>> Yes I understand the importance of packages but that's fine for
>>>> Debian installs and I have many small clients who would have less
>>>> than 10 exts and in our testing giving them fs on raspbian is just
>>>> fine and hardware like the raspberry pi works brilliant for calls,
>>>> fax and most other basic things used in a PBX environment so if
>>>> copying binary works fine on Debian my next task is to do the same
>>>> on raspbian and I love to be on the bleeding edge of the latest
>>>> code rather than use old packages which I guess for raspbian is a
>>>> very old package of fs
>>>>
>>>> On July 7, 2017 12:53:25 AM Michael Jerris <mike at jerris.com> wrote:
>>>>
>>>>> This is what packages are for.
>>>>>
>>>>>> On Jul 6, 2017, at 11:39 AM, Bipin Patel <bipin at xbipin.com
>>>>>> <mailto:bipin at xbipin.com>> wrote:
>>>>>>
>>>>>> hi,
>>>>>>
>>>>>> this might sound a basic question but would appreciate if any1
>>>>>> can guide me. I have 2 systems which are exactly same in software
>>>>>> and hardware, both running debian with exact same packages and
>>>>>> libraries installed. I git pull on one of them the latest FS
>>>>>> master and compile it and then make install and everything works
>>>>>> good, now to avoid compiling the same on the second machine, i
>>>>>> just copy over the complete /usr/local/freeswitch to the second
>>>>>> machine and just modify the profile to reflect its IP but when i
>>>>>> run freeswitch i see it loading modules etc but it comes to the
>>>>>> part loading mod_dialplan_xml and then its just stuck there
>>>>>> forever and starts consuming 100% cpu.
>>>>>>
>>>>>> firstly is it necessary to compile and make install on all
>>>>>> systems and does make install do anything additional other than
>>>>>> create the folders and copy the binaries?  secondly how do i
>>>>>> debug whats causing the issue when running FS coz there is no
>>>>>> error shown, it just comes to that line and its stuck there for
>>>>>> ever
> 



From mark.melling at savageminds.com  Fri Jul  7 09:44:11 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Fri, 07 Jul 2017 09:44:11 +0000
Subject: [Freeswitch-users] How do you determine if DTMFs received are
	in-band or out-of-band?
Message-ID: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>

Hi,

How can you determine if Freeswitch is receiving DTMFs in-band or
out-of-band?

I'm not entirely clear on terminology here, but is out-of-band the same as
rfc 2833?

Thanks

Mark
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From canadamcpherson at gmail.com  Fri Jul  7 02:18:33 2017
From: canadamcpherson at gmail.com (John McPherson)
Date: Thu, 6 Jul 2017 23:18:33 -0300
Subject: [Freeswitch-users] SRTP Issue
Message-ID: <CAEDP5rtJrFBpamRe+CYBdZoUNJ9AE9LDMC4PAnNu11zVviCXtA@mail.gmail.com>

Hello All,

*I had the following issue:*

https://pastebin.freeswitch.org/view/8722d895#L30
issue is here :[WARNING] switch_core_media.c:4451 Crypto not negotiated but
required.
This is the log for my issue. The external forwarding does not work when I
use rtp_secure_media=true. Everything else works. The only workaround is to
put rtp_secure_media=false in the outbound dialplan and make the
bridge=loopback instead of outbound. I don't like this workaround. Is there
a more elegant solution?
I want the bridge=outbound and not loopback

*Then I was told that I can use rtp_secure_media=optional*

It made me happy as I thought the problem is solved.


*But now the problem is rtp_secure_media=optional does not work when the
devices only have SRTP_Encryption =ALWAYS or NEVER where the "Optional" is
not an option and they cannot receive calls. I know that we can separate
inbound and outbound. However, forwarding is both inbound and outbound.*

Do you have a solution for me?


With my thanks,

John McPherson
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From rmundkowsky at ets.org  Thu Jul  6 22:07:56 2017
From: rmundkowsky at ets.org (Mundkowsky, Robert)
Date: Thu, 6 Jul 2017 22:07:56 +0000
Subject: [Freeswitch-users] running FS issue without make install
In-Reply-To: <15d19a570b8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
References: <375ef10a-4530-0910-61e4-9f32bf4cc075@xbipin.com>
 <BN6PR07MB35562707A6721BBD8A3999C3BED50@BN6PR07MB3556.namprd07.prod.outlook.com>
 <15d19a570b8.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com>
Message-ID: <BN6PR07MB35561B4705DB7AB4F81E4234BED50@BN6PR07MB3556.namprd07.prod.outlook.com>

If you have no reason to make (build from source) then do what Michael suggested and use apt-get (or yum) to install FreeSWITCH RPM (package).

See https://freeswitch.org/confluence/display/FREESWITCH/Linux




Robert Mundkowsky

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bipin Patel
Sent: Thursday, July 6, 2017 4:44 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] running FS issue without make install


Well I'm trying to avoid making and not so great in Linux I don't know what that configure command would do

On July 6, 2017 11:51:26 PM "Mundkowsky, Robert" <rmundkowsky at ets.org<mailto:rmundkowsky at ets.org>> wrote:
This might be easier:

./configure --prefix=/foobar/freeswitch
make
make install

Then everything except for boot scripts in under /foobar/freeswitch

And hopefully that will pick-up anything you missed.

Robert Mundkowsky

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bipin Patel
Sent: Thursday, July 6, 2017 11:40 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] running FS issue without make install

hi,

this might sound a basic question but would appreciate if any1 can guide me. I have 2 systems which are exactly same in software and hardware, both running debian with exact same packages and libraries installed. I git pull on one of them the latest FS master and compile it and then make install and everything works good, now to avoid compiling the same on the second machine, i just copy over the complete /usr/local/freeswitch to the second machine and just modify the profile to reflect its IP but when i run freeswitch i see it loading modules etc but it comes to the part loading mod_dialplan_xml and then its just stuck there forever and starts consuming 100% cpu.

firstly is it necessary to compile and make install on all systems and does make install do anything additional other than create the folders and copy the binaries?
secondly how do i debug whats causing the issue when running FS coz there is no error shown, it just comes to that line and its stuck there for ever


--
Regards,
Bipin


________________________________

________________________________

This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited.


Thank you for your compliance.

________________________________
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting%40freeswitch.org>
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________________________________

This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited.


Thank you for your compliance.

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From gmaruzz at gmail.com  Fri Jul  7 13:42:15 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 7 Jul 2017 15:42:15 +0200
Subject: [Freeswitch-users] SRTP Issue
In-Reply-To: <CAEDP5rtJrFBpamRe+CYBdZoUNJ9AE9LDMC4PAnNu11zVviCXtA@mail.gmail.com>
References: <CAEDP5rtJrFBpamRe+CYBdZoUNJ9AE9LDMC4PAnNu11zVviCXtA@mail.gmail.com>
Message-ID: <CALXCt0r_hQYrvt+ngKXTN8=3-fCTuN_Vq3JMq_4Ggt+x4veY8Q@mail.gmail.com>

On 7 July 2017 at 04:18, John McPherson <canadamcpherson at gmail.com> wrote:

> Hello All,
>
> *I had the following issue:*
>
> https://pastebin.freeswitch.org/view/8722d895#L30
> issue is here :[WARNING] switch_core_media.c:4451 Crypto not negotiated
> but required.
> This is the log for my issue. The external forwarding does not work when I
> use rtp_secure_media=true. Everything else works. The only workaround is to
> put rtp_secure_media=false in the outbound dialplan and make the
> bridge=loopback instead of outbound. I don't like this workaround. Is there
> a more elegant solution?
> I want the bridge=outbound and not loopback
>
> *Then I was told that I can use rtp_secure_media=optional*
>
> It made me happy as I thought the problem is solved.
>
>
> *But now the problem is rtp_secure_media=optional does not work when the
> devices only have SRTP_Encryption =ALWAYS or NEVER where the "Optional" is
> not an option and they cannot receive calls. I know that we can separate
> inbound and outbound. However, forwarding is both inbound and outbound.*
>

John,

what exactly is your problem?

What is that you want to achieve?

What dialplan, what endpoints?

What you exactly do, how we can reproduce the problem?

Please state all the step we need to replicate your problem.

-giovanni






>
> Do you have a solution for me?
>
>
> With my thanks,
>
> John McPherson
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From rbetancor at gmail.com  Fri Jul  7 13:50:32 2017
From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=)
Date: Fri, 7 Jul 2017 14:50:32 +0100
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
Message-ID: <CAKsfQBgSBv8ukqFAiQuBhvAR79kM36oDuosPb8v=WcMHus0=AA@mail.gmail.com>

Out of band allways means 'out of the audio RTP stream' ... could be
RFC2833 or could be SIP-INFO

2017-07-07 10:44 GMT+01:00 Mark Melling <mark.melling at savageminds.com>:

> Hi,
>
> How can you determine if Freeswitch is receiving DTMFs in-band or
> out-of-band?
>
> I'm not entirely clear on terminology here, but is out-of-band the same as
> rfc 2833?
>
> Thanks
>
> Mark
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mark.melling at savageminds.com  Fri Jul  7 13:59:55 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Fri, 07 Jul 2017 13:59:55 +0000
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAKsfQBgSBv8ukqFAiQuBhvAR79kM36oDuosPb8v=WcMHus0=AA@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAKsfQBgSBv8ukqFAiQuBhvAR79kM36oDuosPb8v=WcMHus0=AA@mail.gmail.com>
Message-ID: <CAJp-f1McibBfgDh4gxtsfxBv5OXsp5=y9iMLyvmuHacuKvoUWA@mail.gmail.com>

OK, Thanks

So if the DTMF was received as SIP-INFO I assume I could switch on sip
tracing (sofia global siptrace on) and I would presumably see the relevant
SIP packets.

But what can I do, check for in the logs, or check say in terms of a
channel variable, that would enable me verify that a DTMF was being
received as RFC2833 and not received in-band?



On Fri, 7 Jul 2017 at 14:51 Raúl Alexis Betancor Santana <
rbetancor at gmail.com> wrote:

> Out of band allways means 'out of the audio RTP stream' ... could be
> RFC2833 or could be SIP-INFO
>
> 2017-07-07 10:44 GMT+01:00 Mark Melling <mark.melling at savageminds.com>:
>
>> Hi,
>>
>> How can you determine if Freeswitch is receiving DTMFs in-band or
>> out-of-band?
>>
>> I'm not entirely clear on terminology here, but is out-of-band the same
>> as rfc 2833?
>>
>> Thanks
>>
>> Mark
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From stefano.favaro at edistar.com  Fri Jul  7 14:19:35 2017
From: stefano.favaro at edistar.com (Stefano Favaro)
Date: Fri, 7 Jul 2017 16:19:35 +0200 (CEST)
Subject: [Freeswitch-users] freeswitch.Session with LUA
In-Reply-To: <1499348899.737544.1032327552.4D26C137@webmail.messagingengine.com>
Message-ID: <1787994869.228.1499437154383.JavaMail.sfa@EDISTAR-SFA>



Thanks, 


actually there is an error in the freeswitch.Session method 


The documentation is not complete 


the correct syntax is 


legB = freeswitch.Session("{ignore_early_media=true, hangup_after_bridge=true, origination_caller_id_number=1234}sofia/internal/0123456789 at myserver, legA ) 


The last parameter is the legA session. 


Using this syntax everything works fine. 







Stefano Favaro 
Sviluppo Servizi ed Applicazioni 
_____________________ 



Edistar Srl 

a socio unico soggetta a direzione e coordinamento di YourVoice SpA 

Via Artigianato 1 – I – 31050 Vedelago (TV) Italy 
Phone +39 0423 7331 – Fax +39 0423 733133 

skype: stefanofavaro 

www.edistar.com 
	

	

		

															


Le informazioni trasmesse attraverso la presente e-mail ed i suoi allegati sono dirette esclusivamente al destinatario e devono ritenersi riservate con divieto di diffusione e di uso nei giudizi salva espressa autorizzazione; nel caso di utilizzo senza espressa autorizzazione, potrà essere effettuata denuncia alla competente Autorità. La diffusione e la comunicazione da parte di soggetto diverso dal destinatario è vietata dall’art. 616 e ss. c.p. e dal d. l.vo n. 196/03. Se la presente e-mail e i suoi allegati fossero stati ricevuti per errore da persona diversa dal destinatario preghiamo di distruggere quanto ricevuto e di rinviare al mittente con lo stesso mezzo. Grazie per la collaborazione. 

This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this for the addressee, you must not use, copy, disclose or take any action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply e-mail and delete this message. Thank you for your cooperation. 
----- Messaggio originale -----

Da: "Aviv Shaham" <aviv at ironsip.com> 
A: freeswitch-users at lists.freeswitch.org 
Inviato: Giovedì, 6 luglio 2017 15:48:19 
Oggetto: Re: [Freeswitch-users] freeswitch.Session with LUA 


Have you tried transfer_ringback? 



https://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones#Transfer_Ringback 





On Thu, Jul 6, 2017, at 03:11 AM, Stefano Favaro wrote: 






Hello, 



I'm writing a simple script with LUA to connect an inbound call to e new outbound call. 

Everything work fine but I can't set the ringback tone nor a waiting music on the leg A and also the call_timeout variable. 

The legA is in silence while the legB is ringing and the call_timeout is always the default 60 seconds. 

What is wrong with this code? 



Thanks 



This is the code: 

... 


legA:streamFile("/usr/share/freeswitch/sounds/wait.wav") 



legA:setVariable("ringback", "$${uk-ring}") 

legA:setVariable("call_timeout", 30) 



legB = freeswitch.Session("{ignore_early_media=true, hangup_after_bridge=true, origination_caller_id_number=1234}sofia/internal/0123456789 at myserver) 



if legB:ready() then 

freeswitch.bridge(legA, legB) 

end 









Stefano Favaro 
Sviluppo Servizi ed Applicazioni 
_____________________ 







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

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

FreeSWITCH-users mailing list 
FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
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From aqsyounas at gmail.com  Fri Jul  7 14:41:16 2017
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 7 Jul 2017 19:41:16 +0500
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAJp-f1McibBfgDh4gxtsfxBv5OXsp5=y9iMLyvmuHacuKvoUWA@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAKsfQBgSBv8ukqFAiQuBhvAR79kM36oDuosPb8v=WcMHus0=AA@mail.gmail.com>
 <CAJp-f1McibBfgDh4gxtsfxBv5OXsp5=y9iMLyvmuHacuKvoUWA@mail.gmail.com>
Message-ID: <CAGVOjVWa=stBVCOiWqu5MZ6xUY8BboSM-1+W23s263VOfyyL9g@mail.gmail.com>

if you see "101 telephone-event" in SDP, you are getting DTMF as rfc2833.

On 7 July 2017 at 18:59, Mark Melling <mark.melling at savageminds.com> wrote:

> OK, Thanks
>
> So if the DTMF was received as SIP-INFO I assume I could switch on sip
> tracing (sofia global siptrace on) and I would presumably see the relevant
> SIP packets.
>
> But what can I do, check for in the logs, or check say in terms of a
> channel variable, that would enable me verify that a DTMF was being
> received as RFC2833 and not received in-band?
>
>
>
> On Fri, 7 Jul 2017 at 14:51 Raúl Alexis Betancor Santana <
> rbetancor at gmail.com> wrote:
>
>> Out of band allways means 'out of the audio RTP stream' ... could be
>> RFC2833 or could be SIP-INFO
>>
>> 2017-07-07 10:44 GMT+01:00 Mark Melling <mark.melling at savageminds.com>:
>>
>>> Hi,
>>>
>>> How can you determine if Freeswitch is receiving DTMFs in-band or
>>> out-of-band?
>>>
>>> I'm not entirely clear on terminology here, but is out-of-band the same
>>> as rfc 2833?
>>>
>>> Thanks
>>>
>>> Mark
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rbetancor at gmail.com  Fri Jul  7 15:01:38 2017
From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=)
Date: Fri, 7 Jul 2017 16:01:38 +0100
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAGVOjVWa=stBVCOiWqu5MZ6xUY8BboSM-1+W23s263VOfyyL9g@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAKsfQBgSBv8ukqFAiQuBhvAR79kM36oDuosPb8v=WcMHus0=AA@mail.gmail.com>
 <CAJp-f1McibBfgDh4gxtsfxBv5OXsp5=y9iMLyvmuHacuKvoUWA@mail.gmail.com>
 <CAGVOjVWa=stBVCOiWqu5MZ6xUY8BboSM-1+W23s263VOfyyL9g@mail.gmail.com>
Message-ID: <CAKsfQBigKjw0=RTOFuw4JE53GGDWsJNs2Amo0kb7XLVJv=X8+Q@mail.gmail.com>

DTMF could be 'negotiated', so seen 101 is not always true ... ;)

2017-07-07 15:41 GMT+01:00 Aqs Younas <aqsyounas at gmail.com>:

> if you see "101 telephone-event" in SDP, you are getting DTMF as rfc2833.
>
> On 7 July 2017 at 18:59, Mark Melling <mark.melling at savageminds.com>
> wrote:
>
>> OK, Thanks
>>
>> So if the DTMF was received as SIP-INFO I assume I could switch on sip
>> tracing (sofia global siptrace on) and I would presumably see the relevant
>> SIP packets.
>>
>> But what can I do, check for in the logs, or check say in terms of a
>> channel variable, that would enable me verify that a DTMF was being
>> received as RFC2833 and not received in-band?
>>
>>
>>
>> On Fri, 7 Jul 2017 at 14:51 Raúl Alexis Betancor Santana <
>> rbetancor at gmail.com> wrote:
>>
>>> Out of band allways means 'out of the audio RTP stream' ... could be
>>> RFC2833 or could be SIP-INFO
>>>
>>> 2017-07-07 10:44 GMT+01:00 Mark Melling <mark.melling at savageminds.com>:
>>>
>>>> Hi,
>>>>
>>>> How can you determine if Freeswitch is receiving DTMFs in-band or
>>>> out-of-band?
>>>>
>>>> I'm not entirely clear on terminology here, but is out-of-band the same
>>>> as rfc 2833?
>>>>
>>>> Thanks
>>>>
>>>> Mark
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brians at iptel.co  Fri Jul  7 15:17:55 2017
From: brians at iptel.co (Brian :)
Date: Fri, 7 Jul 2017 16:17:55 +0100
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
Message-ID: <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>

Hi Mark

>From fs_cli

/event plain dtmf

This will give you console logging of every DTMF event that FS handles
and I'm pretty sure will give you the DTMF source.



On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
<mark.melling at savageminds.com> wrote:
> Hi,
>
> How can you determine if Freeswitch is receiving DTMFs in-band or
> out-of-band?
>
> I'm not entirely clear on terminology here, but is out-of-band the same as
> rfc 2833?
>
> Thanks
>
> Mark
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mark.melling at savageminds.com  Fri Jul  7 15:28:48 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Fri, 07 Jul 2017 15:28:48 +0000
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
Message-ID: <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>

Thanks Brian,

Having done that it says

DTMF-Source: RTP

So I assume that means that the DTMF is in-band, is that right?



On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:

> Hi Mark
>
> From fs_cli
>
> /event plain dtmf
>
> This will give you console logging of every DTMF event that FS handles
> and I'm pretty sure will give you the DTMF source.
>
>
>
> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
> <mark.melling at savageminds.com> wrote:
> > Hi,
> >
> > How can you determine if Freeswitch is receiving DTMFs in-band or
> > out-of-band?
> >
> > I'm not entirely clear on terminology here, but is out-of-band the same
> as
> > rfc 2833?
> >
> > Thanks
> >
> > Mark
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From brians at iptel.co  Fri Jul  7 15:50:29 2017
From: brians at iptel.co (Brian :)
Date: Fri, 7 Jul 2017 16:50:29 +0100
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
 <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
Message-ID: <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>

I believe thats 2833

>From the source:

switch(dtmf->source) {

                        case SWITCH_DTMF_INBAND_AUDIO:  /* From audio */

                                dtmf_source_str = "INBAND_AUDIO";

                                break;

                        case SWITCH_DTMF_RTP:                   /*
>From RTP as a telephone event */

                                dtmf_source_str = "RTP";

                                break;

                        case SWITCH_DTMF_ENDPOINT:              /*
>From endpoint signaling */

                                dtmf_source_str = "ENDPOINT";

                                break;

                        case SWITCH_DTMF_APP:                   /*
Injected by application */

                                dtmf_source_str = "APP";

                                break;

                        case SWITCH_DTMF_UNKNOWN:               /*
Unknown source */

                        default:

                                dtmf_source_str = "UNKNOWN";

                                break;

On Fri, Jul 7, 2017 at 4:28 PM, Mark Melling
<mark.melling at savageminds.com> wrote:
> Thanks Brian,
>
> Having done that it says
>
> DTMF-Source: RTP
>
> So I assume that means that the DTMF is in-band, is that right?
>
>
>
> On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:
>>
>> Hi Mark
>>
>> From fs_cli
>>
>> /event plain dtmf
>>
>> This will give you console logging of every DTMF event that FS handles
>> and I'm pretty sure will give you the DTMF source.
>>
>>
>>
>> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
>> <mark.melling at savageminds.com> wrote:
>> > Hi,
>> >
>> > How can you determine if Freeswitch is receiving DTMFs in-band or
>> > out-of-band?
>> >
>> > I'm not entirely clear on terminology here, but is out-of-band the same
>> > as
>> > rfc 2833?
>> >
>> > Thanks
>> >
>> > Mark
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mark.melling at savageminds.com  Fri Jul  7 16:14:44 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Fri, 07 Jul 2017 16:14:44 +0000
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
 <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
 <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>
Message-ID: <CAJp-f1Ne+31RsY0hmWXdZwCXF5Q3hJugshgEa8npN5iJ+EPvXw@mail.gmail.com>

Thanks Brian.


On Fri, 7 Jul 2017 at 16:51 Brian : <brians at iptel.co> wrote:

> I believe thats 2833
>
> From the source:
>
> switch(dtmf->source) {
>
>                         case SWITCH_DTMF_INBAND_AUDIO:  /* From audio */
>
>                                 dtmf_source_str = "INBAND_AUDIO";
>
>                                 break;
>
>                         case SWITCH_DTMF_RTP:                   /*
> From RTP as a telephone event */
>
>                                 dtmf_source_str = "RTP";
>
>                                 break;
>
>                         case SWITCH_DTMF_ENDPOINT:              /*
> From endpoint signaling */
>
>                                 dtmf_source_str = "ENDPOINT";
>
>                                 break;
>
>                         case SWITCH_DTMF_APP:                   /*
> Injected by application */
>
>                                 dtmf_source_str = "APP";
>
>                                 break;
>
>                         case SWITCH_DTMF_UNKNOWN:               /*
> Unknown source */
>
>                         default:
>
>                                 dtmf_source_str = "UNKNOWN";
>
>                                 break;
>
> On Fri, Jul 7, 2017 at 4:28 PM, Mark Melling
> <mark.melling at savageminds.com> wrote:
> > Thanks Brian,
> >
> > Having done that it says
> >
> > DTMF-Source: RTP
> >
> > So I assume that means that the DTMF is in-band, is that right?
> >
> >
> >
> > On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:
> >>
> >> Hi Mark
> >>
> >> From fs_cli
> >>
> >> /event plain dtmf
> >>
> >> This will give you console logging of every DTMF event that FS handles
> >> and I'm pretty sure will give you the DTMF source.
> >>
> >>
> >>
> >> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
> >> <mark.melling at savageminds.com> wrote:
> >> > Hi,
> >> >
> >> > How can you determine if Freeswitch is receiving DTMFs in-band or
> >> > out-of-band?
> >> >
> >> > I'm not entirely clear on terminology here, but is out-of-band the
> same
> >> > as
> >> > rfc 2833?
> >> >
> >> > Thanks
> >> >
> >> > Mark
> >> >
> >> >
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://confluence.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mark.melling at savageminds.com  Fri Jul  7 16:22:57 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Fri, 07 Jul 2017 16:22:57 +0000
Subject: [Freeswitch-users] Controlling tones for RFC2833 DTMFs
Message-ID: <CAJp-f1MUoL-xdHfdmokUrOBx_abBCQvfyLZKBFtNhHzsTHsnnA@mail.gmail.com>

Hi,

If Freeswitch receives rfc2833 DTMFs, given that these DTMFs are
out-of-band, does Freeswitch generate a tone to indicate to the user that a
key has been pressed?

Assuming it does, is there a way to prevent the tone from being generated,
i.e. so the user doesn't hear anything when a key is pressed?

Thanks

Mark
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From mark.melling at savageminds.com  Fri Jul  7 16:34:24 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Fri, 07 Jul 2017 16:34:24 +0000
Subject: [Freeswitch-users] Verto: no inbound audio when verto client
 called from conference
In-Reply-To: <CAJp-f1MvetuC4vWL5UM2uDda99FPPYZWTahbn6W9miPVBF-LQg@mail.gmail.com>
References: <CAJp-f1ONFgNwhHs7MLuC+xfy8gEwKvOwJ8+CYq2DmcNB+xkp_g@mail.gmail.com>
 <CALXCt0qojw_FEP=ZveB8_=DGUSEA_BCi2gYwUxu1yYWos1nASA@mail.gmail.com>
 <CAJp-f1NJv3_Etvx2ThvGW3pR9i9jx-s+YUZOBY2JkA3OAwpJLg@mail.gmail.com>
 <CALXCt0oykcTR0qaQHrLAAJmacTXsCoHkkUBP+ErY6VF27V_Hwg@mail.gmail.com>
 <CAJp-f1MvetuC4vWL5UM2uDda99FPPYZWTahbn6W9miPVBF-LQg@mail.gmail.com>
Message-ID: <CAJp-f1OD1VN=rbZnNTmUxG6qWJXAr4mkTMzVpoi=Ysvg08qbAQ@mail.gmail.com>

Based on this I implemented a solution using lua:

The key bits being:

    session = freeswitch.Session(<call url>)
    ...
    session:execute("sleep", 1000)
    session:execute("conference", <conference-name>)

The only problem is that when the verto user is added to the conference
they appear (in the conference) with the originating caller id and name (as
set in the freeswitch.Session call), rather than their own.

Is there anyway around this?

Thanks

Mark


On Wed, 5 Jul 2017 at 22:21 Mark Melling <mark.melling at savageminds.com>
wrote:

>
> Your suggestion does work, I did the following manually from fs_cli.
>
> So with a user dialled into a conference room I was able to do:
>
> > originate  <call-url> &park
>
> Where the call-url referring to a verto client, then
>
> > uuid_transfer <verto-call-uuid> conference:<conf-name> inline
>
> And that worked, I could hear audio from both clients connected to the
> conference.
>
> Thanks for your help.
>
>
>
> On Wed, 5 Jul 2017 at 20:12 Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>
>> On 5 July 2017 at 18:08, Mark Melling <mark.melling at savageminds.com>
>> wrote:
>>
>>> Thanks Giovanni for the suggestion.
>>>
>>> I tried some more experiments and basically if I call a verto client and
>>> add them to a conference then they don't hear the audio (although the
>>> conference is detecting audio when they speak).
>>>
>>> But if they dial into the conference then everything appears fine and
>>> they do hear audio.
>>>
>>> Specifically from fs_cli I entered:
>>>
>>> originate <call-url> &conference(<conf-name>@default)
>>>
>>> If call-url is a sip client then you hear the conference music, but if
>>> call-url is a verto client you don't hear any conference music. But the
>>> conference does detect when the verto client is speaking (at least the
>>> status in the verto web page indicates the user is talking).
>>>
>>> Whereas if you dialled into a conference from a verto client then you
>>> would hear the conference music.
>>>
>>> So I'm not sure how I can work around this.
>>>
>>>
>>>
>> Have you tried what I suggested?
>>
>>
>>
>>>>
>>>> Maybe this is because verto (webrtc) takes time to establish audio
>>>> because of stun, etc etc
>>>>
>>>> Try this: instead of generating autocall from inside conference (eg
>>>> instead of using autocall),originate call to user, wait for her to answer,
>>>> then (after she answer) sleep for 2 seconds, then transfer her to the conf
>>>>
>>>> -giovanni
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Fri Jul  7 16:39:24 2017
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 Jul 2017 11:39:24 -0500
Subject: [Freeswitch-users] SRTP Issue
In-Reply-To: <CALXCt0r_hQYrvt+ngKXTN8=3-fCTuN_Vq3JMq_4Ggt+x4veY8Q@mail.gmail.com>
References: <CAEDP5rtJrFBpamRe+CYBdZoUNJ9AE9LDMC4PAnNu11zVviCXtA@mail.gmail.com>
 <CALXCt0r_hQYrvt+ngKXTN8=3-fCTuN_Vq3JMq_4Ggt+x4veY8Q@mail.gmail.com>
Message-ID: <CAEJMVkB0X8Sj2eah=F1KTMxPCUC+WBuMhpqGe0gYbRrfNLJbWw@mail.gmail.com>

Open up vars.xml in 1.6.18 or master repo, and read the outlined SRTP
documentation.

/b


On Fri, Jul 7, 2017 at 8:42 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

>
>
> On 7 July 2017 at 04:18, John McPherson <canadamcpherson at gmail.com> wrote:
>
>> Hello All,
>>
>> *I had the following issue:*
>>
>> https://pastebin.freeswitch.org/view/8722d895#L30
>> issue is here :[WARNING] switch_core_media.c:4451 Crypto not negotiated
>> but required.
>> This is the log for my issue. The external forwarding does not work when
>> I use rtp_secure_media=true. Everything else works. The only workaround is
>> to put rtp_secure_media=false in the outbound dialplan and make the
>> bridge=loopback instead of outbound. I don't like this workaround. Is there
>> a more elegant solution?
>> I want the bridge=outbound and not loopback
>>
>> *Then I was told that I can use rtp_secure_media=optional*
>>
>> It made me happy as I thought the problem is solved.
>>
>>
>> *But now the problem is rtp_secure_media=optional does not work when the
>> devices only have SRTP_Encryption =ALWAYS or NEVER where the "Optional" is
>> not an option and they cannot receive calls. I know that we can separate
>> inbound and outbound. However, forwarding is both inbound and outbound.*
>>
>
> John,
>
> what exactly is your problem?
>
> What is that you want to achieve?
>
> What dialplan, what endpoints?
>
> What you exactly do, how we can reproduce the problem?
>
> Please state all the step we need to replicate your problem.
>
> -giovanni
>
>
>
>
>
>
>>
>> Do you have a solution for me?
>>
>>
>> With my thanks,
>>
>> John McPherson
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From david.villasmil.work at gmail.com  Fri Jul  7 16:46:00 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 7 Jul 2017 18:46:00 +0200
Subject: [Freeswitch-users] sip to webrtc - sdp invalid description
In-Reply-To: <CAFGRPVoQB8mH3a8JYbY3wP1L=L3x-zdZF-kK1o+ocW5GzSYt-A@mail.gmail.com>
References: <CAFGRPVoQB8mH3a8JYbY3wP1L=L3x-zdZF-kK1o+ocW5GzSYt-A@mail.gmail.com>
Message-ID: <CAFGRPVrNFQ6rKYVehMX1bopaUqKqT0NQeDQkGigCGC5e5h7FUg@mail.gmail.com>

Hello guys,

Any help on this?

Thanks

David
ᐧ

Regards,

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

On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello guys,
>
> I have this setup:
>
> Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
>
> Whenever webrtc clients call each other, calls are ok.
> But when the zoiper (regular sip/tcp) calls, the browsers complaint about:
>
> "no ice-ufrag" (firefox) or
> "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."
> (Chrome).
>
> I am setting the sdp in freeswitch as:
>
>       <action application="set" data="proxy_media=false"/>
>       <action application="set" data="bypass_media=false"/>
>       <action application="export" data="zrtp_secure_media=false"/>
>       <action application="export" data="nolocal:rtp_secure_media=true"/>
>
> and the actual sdp is:
>
> v=0
> o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4
> s=FreeSWITCH
> c=IN IP4 1.2.3.4
> t=0 0
> m=audio 40954 RTP/SAVP 8 101 13
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVr
> VfjA0ImWgu5WyLK0vtg60RTk6Koe8c0sRkjU
> a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+D5NVoKFFw0
> 6I0Xok/9KbRystQ
> a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+ETPuCOiOIxjFLRjbF
> bDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg
> a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:evU8MzAtiSHwKb95s4V9IAM
> pmok06kW9ZGDgH3/Lc3ZytVn2SR4
> a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:KKDkT0DssohSeKFsX6tbixR
> hwYdiIhE6r3u5CCVA
> a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:D3JgGuOlIxXfHGdqf7lKWqN
> DAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w
> a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:c12ebWWzZ1cqZN0v5C5uYzd
> vtfnw6AARU3+jGA0WzTSDlDd20vI
> a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:nPZku7S6hlz2OPAff8T9I8s
> NFzuZziNg64KuvfNS
> a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3
> dhTqD0XW8FAOLNtmP
> a=ptime:20
> a=nortpproxy:yes
>
>
> I'm pretty new to the webrtc stuff, so any help is greatly appreciated!
>
> Thanks!
>
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337 <+34%20669%2044%2083%2037>
> ᐧ
>
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From mirkobrankovic at gmail.com  Fri Jul  7 17:18:11 2017
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Fri, 7 Jul 2017 19:18:11 +0200
Subject: [Freeswitch-users] Controlling tones for RFC2833 DTMFs
In-Reply-To: <CAJp-f1MUoL-xdHfdmokUrOBx_abBCQvfyLZKBFtNhHzsTHsnnA@mail.gmail.com>
References: <CAJp-f1MUoL-xdHfdmokUrOBx_abBCQvfyLZKBFtNhHzsTHsnnA@mail.gmail.com>
Message-ID: <CAND18T36TwTz=G7kbwp_uVwnSEw05YoSUUL00Fn_w8d-s-U2EA@mail.gmail.com>

Well every dtmf will have end rtp event with final duration and then tone
will be played to endpoint, and in between you will receive more rtp event
with increasing duration.
I'm not sure if you can ignore it of you negotiated rfc2833 type. Maybe
there is some channel var that you can set on the fly

On Jul 7, 2017 18:23, "Mark Melling" <mark.melling at savageminds.com> wrote:

> Hi,
>
> If Freeswitch receives rfc2833 DTMFs, given that these DTMFs are
> out-of-band, does Freeswitch generate a tone to indicate to the user that a
> key has been pressed?
>
> Assuming it does, is there a way to prevent the tone from being generated,
> i.e. so the user doesn't hear anything when a key is pressed?
>
> Thanks
>
> Mark
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at gmail.com  Fri Jul  7 17:51:46 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 7 Jul 2017 19:51:46 +0200
Subject: [Freeswitch-users] Verto: no inbound audio when verto client
 called from conference
In-Reply-To: <CAJp-f1OD1VN=rbZnNTmUxG6qWJXAr4mkTMzVpoi=Ysvg08qbAQ@mail.gmail.com>
References: <CAJp-f1ONFgNwhHs7MLuC+xfy8gEwKvOwJ8+CYq2DmcNB+xkp_g@mail.gmail.com>
 <CALXCt0qojw_FEP=ZveB8_=DGUSEA_BCi2gYwUxu1yYWos1nASA@mail.gmail.com>
 <CAJp-f1NJv3_Etvx2ThvGW3pR9i9jx-s+YUZOBY2JkA3OAwpJLg@mail.gmail.com>
 <CALXCt0oykcTR0qaQHrLAAJmacTXsCoHkkUBP+ErY6VF27V_Hwg@mail.gmail.com>
 <CAJp-f1MvetuC4vWL5UM2uDda99FPPYZWTahbn6W9miPVBF-LQg@mail.gmail.com>
 <CAJp-f1OD1VN=rbZnNTmUxG6qWJXAr4mkTMzVpoi=Ysvg08qbAQ@mail.gmail.com>
Message-ID: <CALXCt0o+_grFm7Uwd_+UNQJ8M8Dih6XJgGAdOB32JJuJ-PcaUQ@mail.gmail.com>

Set the channel caller id variables before joining the conference

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT

On Jul 7, 2017 18:35, "Mark Melling" <mark.melling at savageminds.com> wrote:

> Based on this I implemented a solution using lua:
>
> The key bits being:
>
>     session = freeswitch.Session(<call url>)
>     ...
>     session:execute("sleep", 1000)
>     session:execute("conference", <conference-name>)
>
> The only problem is that when the verto user is added to the conference
> they appear (in the conference) with the originating caller id and name (as
> set in the freeswitch.Session call), rather than their own.
>
> Is there anyway around this?
>
> Thanks
>
> Mark
>
>
> On Wed, 5 Jul 2017 at 22:21 Mark Melling <mark.melling at savageminds.com>
> wrote:
>
>>
>> Your suggestion does work, I did the following manually from fs_cli.
>>
>> So with a user dialled into a conference room I was able to do:
>>
>> > originate  <call-url> &park
>>
>> Where the call-url referring to a verto client, then
>>
>> > uuid_transfer <verto-call-uuid> conference:<conf-name> inline
>>
>> And that worked, I could hear audio from both clients connected to the
>> conference.
>>
>> Thanks for your help.
>>
>>
>>
>> On Wed, 5 Jul 2017 at 20:12 Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>
>>> On 5 July 2017 at 18:08, Mark Melling <mark.melling at savageminds.com>
>>> wrote:
>>>
>>>> Thanks Giovanni for the suggestion.
>>>>
>>>> I tried some more experiments and basically if I call a verto client
>>>> and add them to a conference then they don't hear the audio (although the
>>>> conference is detecting audio when they speak).
>>>>
>>>> But if they dial into the conference then everything appears fine and
>>>> they do hear audio.
>>>>
>>>> Specifically from fs_cli I entered:
>>>>
>>>> originate <call-url> &conference(<conf-name>@default)
>>>>
>>>> If call-url is a sip client then you hear the conference music, but if
>>>> call-url is a verto client you don't hear any conference music. But the
>>>> conference does detect when the verto client is speaking (at least the
>>>> status in the verto web page indicates the user is talking).
>>>>
>>>> Whereas if you dialled into a conference from a verto client then you
>>>> would hear the conference music.
>>>>
>>>> So I'm not sure how I can work around this.
>>>>
>>>>
>>>>
>>> Have you tried what I suggested?
>>>
>>>
>>>
>>>>>
>>>>> Maybe this is because verto (webrtc) takes time to establish audio
>>>>> because of stun, etc etc
>>>>>
>>>>> Try this: instead of generating autocall from inside conference (eg
>>>>> instead of using autocall),originate call to user, wait for her to answer,
>>>>> then (after she answer) sleep for 2 seconds, then transfer her to the conf
>>>>>
>>>>> -giovanni
>>>>>
>>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From grcamauer at gmail.com  Fri Jul  7 18:19:59 2017
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Fri, 7 Jul 2017 15:19:59 -0300
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAJp-f1Ne+31RsY0hmWXdZwCXF5Q3hJugshgEa8npN5iJ+EPvXw@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
 <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
 <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>
 <CAJp-f1Ne+31RsY0hmWXdZwCXF5Q3hJugshgEa8npN5iJ+EPvXw@mail.gmail.com>
Message-ID: <CAPNTnnqMg_-w8q3nH_CfhRtoz0H_mFo7PZO5Yp6Ynb0ZRnHeOg@mail.gmail.com>

RFC-2833 sends the DTMF as a special packet within the RTP stream.
SIP-INFO sends it in the SIP dialog.
INBAND DTMF come as a series of packets within the RTP stream that must be
analyzed to find the frequencies corresponding to the different DTMF.  I
believe spandsp takes cares of this.

Guillermo

On Fri, Jul 7, 2017 at 1:14 PM, Mark Melling <mark.melling at savageminds.com>
wrote:

> Thanks Brian.
>
>
> On Fri, 7 Jul 2017 at 16:51 Brian : <brians at iptel.co> wrote:
>
>> I believe thats 2833
>>
>> From the source:
>>
>> switch(dtmf->source) {
>>
>>                         case SWITCH_DTMF_INBAND_AUDIO:  /* From audio */
>>
>>                                 dtmf_source_str = "INBAND_AUDIO";
>>
>>                                 break;
>>
>>                         case SWITCH_DTMF_RTP:                   /*
>> From RTP as a telephone event */
>>
>>                                 dtmf_source_str = "RTP";
>>
>>                                 break;
>>
>>                         case SWITCH_DTMF_ENDPOINT:              /*
>> From endpoint signaling */
>>
>>                                 dtmf_source_str = "ENDPOINT";
>>
>>                                 break;
>>
>>                         case SWITCH_DTMF_APP:                   /*
>> Injected by application */
>>
>>                                 dtmf_source_str = "APP";
>>
>>                                 break;
>>
>>                         case SWITCH_DTMF_UNKNOWN:               /*
>> Unknown source */
>>
>>                         default:
>>
>>                                 dtmf_source_str = "UNKNOWN";
>>
>>                                 break;
>>
>> On Fri, Jul 7, 2017 at 4:28 PM, Mark Melling
>> <mark.melling at savageminds.com> wrote:
>> > Thanks Brian,
>> >
>> > Having done that it says
>> >
>> > DTMF-Source: RTP
>> >
>> > So I assume that means that the DTMF is in-band, is that right?
>> >
>> >
>> >
>> > On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:
>> >>
>> >> Hi Mark
>> >>
>> >> From fs_cli
>> >>
>> >> /event plain dtmf
>> >>
>> >> This will give you console logging of every DTMF event that FS handles
>> >> and I'm pretty sure will give you the DTMF source.
>> >>
>> >>
>> >>
>> >> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
>> >> <mark.melling at savageminds.com> wrote:
>> >> > Hi,
>> >> >
>> >> > How can you determine if Freeswitch is receiving DTMFs in-band or
>> >> > out-of-band?
>> >> >
>> >> > I'm not entirely clear on terminology here, but is out-of-band the
>> same
>> >> > as
>> >> > rfc 2833?
>> >> >
>> >> > Thanks
>> >> >
>> >> > Mark
>> >> >
>> >> >
>> >> >
>> >> > ____________________________________________________________
>> _____________
>> >> > Professional FreeSWITCH Consulting Services:
>> >> > consulting at freeswitch.org
>> >> > http://www.freeswitchsolutions.com
>> >> >
>> >> > Official FreeSWITCH Sites
>> >> > http://www.freeswitch.org
>> >> > http://confluence.freeswitch.org
>> >> > http://www.cluecon.com
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>> options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >>
>> >> ____________________________________________________________
>> _____________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>> options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > ____________________________________________________________
>> _____________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>> options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Guillermo Ruiz Camauer
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From matt.nichols at westtel.com  Fri Jul  7 18:03:09 2017
From: matt.nichols at westtel.com (Matthew Nichols)
Date: Fri, 7 Jul 2017 18:03:09 +0000
Subject: [Freeswitch-users] PostgreSQL BDR Schema for HA
Message-ID: <DM5PR1001MB2155921B0DBBD076B9DDAAF583AA0@DM5PR1001MB2155.namprd10.prod.outlook.com>

Is there documentation for a PostgreSQL BDR Schema that works with FreeSWITCH HA? The closest I have found is https://gist.github.com/DigiDaz/1cfe3d5d32080a8e3d75a20bb5bc4fb5 and while the RTP streams transfer over the call/channel state and what not don't, and I get errors like this:

2017-07-07 11:42:04.296420 [DEBUG] switch_pgsql.c:415 Query (insert into channels (uuid,direction,created,created_epoch, name,state,callstate,dialplan,context,hostname,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context) values('eac21399-ecee-40c4-a4e3-9a8f9e4aae7a','outbound','2017-07-07 11:42:04','1499449324','sofia/internal/1002 at 10.1.1.71','CS_INIT','DOWN','XML','default','VFS1','1001','1001','10.1.1.71','1002','XML','default')) returned PGRES_FATAL_ERROR
2017-07-07 11:42:04.296420 [DEBUG] switch_pgsql.c:415 Query (insert into channels (uuid,direction,created,created_epoch, name,state,callstate,dialplan,context,hostname,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context) values('09112fab-0c0f-4ee1-9254-424af3948658','inbound','2017-07-07 11:42:04','1499449324','sofia/internal/1001 at 10.1.1.46','CS_INIT','ACTIVE','XML','default','VFS1','1001','1001','10.1.1.70','1002','XML','default')) returned PGRES_FATAL_ERROR
2017-07-07 11:42:04.296420 [ERR] switch_pgsql.c:656 Error executing query:
ERROR:  current transaction is aborted, commands ignored until end of transaction block

2017-07-07 11:42:04.496420 [DEBUG] switch_pgsql.c:415 Query (insert into calls (call_uuid,call_created,call_created_epoch,caller_uuid,callee_uuid,hostname) values ('09112fab-0c0f-4ee1-9254-424af3948658','2017-07-07 11:42:04','1499449324','09112fab-0c0f-4ee1-9254-424af3948658','eac21399-ecee-40c4-a4e3-9a8f9e4aae7a','VFS1')) returned PGRES_FATAL_ERROR
2017-07-07 11:42:04.496420 [DEBUG] switch_pgsql.c:415 Query (update channels set read_codec='G722',read_rate='16000',read_bit_rate='64000',write_codec='G722',write_rate='16000',write_bit_rate='64000' where uuid='eac21399-ecee-40c4-a4e3-9a8f9e4aae7a') returned PGRES_FATAL_ERROR
2017-07-07 11:42:04.496420 [ERR] switch_pgsql.c:656 Error executing query:
ERROR:  current transaction is aborted, commands ignored until end of transaction block


It looks like it's trying to insert into the channels/calls tables with the same UUID as before which of course doesn't work since it is a primary key. Does the primary key need to include the hostname as well (and is that only for the calls/channels tables or other tables as well?). Since BDR requires a primary key I'd like to know what FreeSWITCH expects, since it doesn't autogenerate one.


From mark.melling at savageminds.com  Fri Jul  7 18:42:44 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Fri, 07 Jul 2017 18:42:44 +0000
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAPNTnnqMg_-w8q3nH_CfhRtoz0H_mFo7PZO5Yp6Ynb0ZRnHeOg@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
 <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
 <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>
 <CAJp-f1Ne+31RsY0hmWXdZwCXF5Q3hJugshgEa8npN5iJ+EPvXw@mail.gmail.com>
 <CAPNTnnqMg_-w8q3nH_CfhRtoz0H_mFo7PZO5Yp6Ynb0ZRnHeOg@mail.gmail.com>
Message-ID: <CAJp-f1PsJhMzAJVaFbWFQWvOpo=qCNLAwaZGZ8uoikG-3V0=QQ@mail.gmail.com>

Thanks for that, that is useful info.

I posted a separate, but related question that you might have some
suggestions about.

Namely I'm assuming that for RFC-2833 DTMFs Freeswitch is generating a
corresponding tone for the received DTMF (I don't know if this is actually
true or not).  If this is the case whether it is possible to disable this
tone so that basically a user pressing a key (that generates a DTMF)
doesn't hear anything.

Thanks


On Fri, 7 Jul 2017 at 19:22 Guillermo Ruiz Camauer <grcamauer at gmail.com>
wrote:

> RFC-2833 sends the DTMF as a special packet within the RTP stream.
> SIP-INFO sends it in the SIP dialog.
> INBAND DTMF come as a series of packets within the RTP stream that must be
> analyzed to find the frequencies corresponding to the different DTMF.  I
> believe spandsp takes cares of this.
>
> Guillermo
>
> On Fri, Jul 7, 2017 at 1:14 PM, Mark Melling <mark.melling at savageminds.com
> > wrote:
>
>> Thanks Brian.
>>
>>
>> On Fri, 7 Jul 2017 at 16:51 Brian : <brians at iptel.co> wrote:
>>
>>> I believe thats 2833
>>>
>>> From the source:
>>>
>>> switch(dtmf->source) {
>>>
>>>                         case SWITCH_DTMF_INBAND_AUDIO:  /* From audio */
>>>
>>>                                 dtmf_source_str = "INBAND_AUDIO";
>>>
>>>                                 break;
>>>
>>>                         case SWITCH_DTMF_RTP:                   /*
>>> From RTP as a telephone event */
>>>
>>>                                 dtmf_source_str = "RTP";
>>>
>>>                                 break;
>>>
>>>                         case SWITCH_DTMF_ENDPOINT:              /*
>>> From endpoint signaling */
>>>
>>>                                 dtmf_source_str = "ENDPOINT";
>>>
>>>                                 break;
>>>
>>>                         case SWITCH_DTMF_APP:                   /*
>>> Injected by application */
>>>
>>>                                 dtmf_source_str = "APP";
>>>
>>>                                 break;
>>>
>>>                         case SWITCH_DTMF_UNKNOWN:               /*
>>> Unknown source */
>>>
>>>                         default:
>>>
>>>                                 dtmf_source_str = "UNKNOWN";
>>>
>>>                                 break;
>>>
>>> On Fri, Jul 7, 2017 at 4:28 PM, Mark Melling
>>> <mark.melling at savageminds.com> wrote:
>>> > Thanks Brian,
>>> >
>>> > Having done that it says
>>> >
>>> > DTMF-Source: RTP
>>> >
>>> > So I assume that means that the DTMF is in-band, is that right?
>>> >
>>> >
>>> >
>>> > On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:
>>> >>
>>> >> Hi Mark
>>> >>
>>> >> From fs_cli
>>> >>
>>> >> /event plain dtmf
>>> >>
>>> >> This will give you console logging of every DTMF event that FS handles
>>> >> and I'm pretty sure will give you the DTMF source.
>>> >>
>>> >>
>>> >>
>>> >> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
>>> >> <mark.melling at savageminds.com> wrote:
>>> >> > Hi,
>>> >> >
>>> >> > How can you determine if Freeswitch is receiving DTMFs in-band or
>>> >> > out-of-band?
>>> >> >
>>> >> > I'm not entirely clear on terminology here, but is out-of-band the
>>> same
>>> >> > as
>>> >> > rfc 2833?
>>> >> >
>>> >> > Thanks
>>> >> >
>>> >> > Mark
>>> >> >
>>> >> >
>>> >> >
>>> >> >
>>> _________________________________________________________________________
>>> >> > Professional FreeSWITCH Consulting Services:
>>> >> > consulting at freeswitch.org
>>> >> > http://www.freeswitchsolutions.com
>>> >> >
>>> >> > Official FreeSWITCH Sites
>>> >> > http://www.freeswitch.org
>>> >> > http://confluence.freeswitch.org
>>> >> > http://www.cluecon.com
>>> >> >
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://confluence.freeswitch.org
>>> >> http://www.cluecon.com
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://confluence.freeswitch.org
>>> > http://www.cluecon.com
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Guillermo Ruiz Camauer
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From rbetancor at gmail.com  Fri Jul  7 18:57:29 2017
From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=)
Date: Fri, 7 Jul 2017 19:57:29 +0100
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAJp-f1PsJhMzAJVaFbWFQWvOpo=qCNLAwaZGZ8uoikG-3V0=QQ@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
 <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
 <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>
 <CAJp-f1Ne+31RsY0hmWXdZwCXF5Q3hJugshgEa8npN5iJ+EPvXw@mail.gmail.com>
 <CAPNTnnqMg_-w8q3nH_CfhRtoz0H_mFo7PZO5Yp6Ynb0ZRnHeOg@mail.gmail.com>
 <CAJp-f1PsJhMzAJVaFbWFQWvOpo=qCNLAwaZGZ8uoikG-3V0=QQ@mail.gmail.com>
Message-ID: <CAKsfQBgMDEJ4qUheB9JsQp4J913DfvxpMvkmj8uC9=h3ygn8ew@mail.gmail.com>

I don't know if FS allow you tu 'filter' the DTMFs and don't send them to
the B leg ... but it doesn't matter either ... as if the B leg is hearing
the DTMF it's because it's forwarded to it and it's the B leg SIP endpoint
the responsible of 'let them hear it'.

2017-07-07 19:42 GMT+01:00 Mark Melling <mark.melling at savageminds.com>:

> Thanks for that, that is useful info.
>
> I posted a separate, but related question that you might have some
> suggestions about.
>
> Namely I'm assuming that for RFC-2833 DTMFs Freeswitch is generating a
> corresponding tone for the received DTMF (I don't know if this is actually
> true or not).  If this is the case whether it is possible to disable this
> tone so that basically a user pressing a key (that generates a DTMF)
> doesn't hear anything.
>
> Thanks
>
>
> On Fri, 7 Jul 2017 at 19:22 Guillermo Ruiz Camauer <grcamauer at gmail.com>
> wrote:
>
>> RFC-2833 sends the DTMF as a special packet within the RTP stream.
>> SIP-INFO sends it in the SIP dialog.
>> INBAND DTMF come as a series of packets within the RTP stream that must
>> be analyzed to find the frequencies corresponding to the different DTMF.  I
>> believe spandsp takes cares of this.
>>
>> Guillermo
>>
>> On Fri, Jul 7, 2017 at 1:14 PM, Mark Melling <
>> mark.melling at savageminds.com> wrote:
>>
>>> Thanks Brian.
>>>
>>>
>>> On Fri, 7 Jul 2017 at 16:51 Brian : <brians at iptel.co> wrote:
>>>
>>>> I believe thats 2833
>>>>
>>>> From the source:
>>>>
>>>> switch(dtmf->source) {
>>>>
>>>>                         case SWITCH_DTMF_INBAND_AUDIO:  /* From audio */
>>>>
>>>>                                 dtmf_source_str = "INBAND_AUDIO";
>>>>
>>>>                                 break;
>>>>
>>>>                         case SWITCH_DTMF_RTP:                   /*
>>>> From RTP as a telephone event */
>>>>
>>>>                                 dtmf_source_str = "RTP";
>>>>
>>>>                                 break;
>>>>
>>>>                         case SWITCH_DTMF_ENDPOINT:              /*
>>>> From endpoint signaling */
>>>>
>>>>                                 dtmf_source_str = "ENDPOINT";
>>>>
>>>>                                 break;
>>>>
>>>>                         case SWITCH_DTMF_APP:                   /*
>>>> Injected by application */
>>>>
>>>>                                 dtmf_source_str = "APP";
>>>>
>>>>                                 break;
>>>>
>>>>                         case SWITCH_DTMF_UNKNOWN:               /*
>>>> Unknown source */
>>>>
>>>>                         default:
>>>>
>>>>                                 dtmf_source_str = "UNKNOWN";
>>>>
>>>>                                 break;
>>>>
>>>> On Fri, Jul 7, 2017 at 4:28 PM, Mark Melling
>>>> <mark.melling at savageminds.com> wrote:
>>>> > Thanks Brian,
>>>> >
>>>> > Having done that it says
>>>> >
>>>> > DTMF-Source: RTP
>>>> >
>>>> > So I assume that means that the DTMF is in-band, is that right?
>>>> >
>>>> >
>>>> >
>>>> > On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:
>>>> >>
>>>> >> Hi Mark
>>>> >>
>>>> >> From fs_cli
>>>> >>
>>>> >> /event plain dtmf
>>>> >>
>>>> >> This will give you console logging of every DTMF event that FS
>>>> handles
>>>> >> and I'm pretty sure will give you the DTMF source.
>>>> >>
>>>> >>
>>>> >>
>>>> >> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
>>>> >> <mark.melling at savageminds.com> wrote:
>>>> >> > Hi,
>>>> >> >
>>>> >> > How can you determine if Freeswitch is receiving DTMFs in-band or
>>>> >> > out-of-band?
>>>> >> >
>>>> >> > I'm not entirely clear on terminology here, but is out-of-band the
>>>> same
>>>> >> > as
>>>> >> > rfc 2833?
>>>> >> >
>>>> >> > Thanks
>>>> >> >
>>>> >> > Mark
>>>> >> >
>>>> >> >
>>>> >> >
>>>> >> > ____________________________________________________________
>>>> _____________
>>>> >> > Professional FreeSWITCH Consulting Services:
>>>> >> > consulting at freeswitch.org
>>>> >> > http://www.freeswitchsolutions.com
>>>> >> >
>>>> >> > Official FreeSWITCH Sites
>>>> >> > http://www.freeswitch.org
>>>> >> > http://confluence.freeswitch.org
>>>> >> > http://www.cluecon.com
>>>> >> >
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >>
>>>> >> ____________________________________________________________
>>>> _____________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> consulting at freeswitch.org
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://confluence.freeswitch.org
>>>> >> http://www.cluecon.com
>>>> >>
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > ____________________________________________________________
>>>> _____________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://confluence.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Guillermo Ruiz Camauer
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From grcamauer at gmail.com  Fri Jul  7 19:04:53 2017
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Fri, 7 Jul 2017 16:04:53 -0300
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAJp-f1PsJhMzAJVaFbWFQWvOpo=qCNLAwaZGZ8uoikG-3V0=QQ@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
 <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
 <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>
 <CAJp-f1Ne+31RsY0hmWXdZwCXF5Q3hJugshgEa8npN5iJ+EPvXw@mail.gmail.com>
 <CAPNTnnqMg_-w8q3nH_CfhRtoz0H_mFo7PZO5Yp6Ynb0ZRnHeOg@mail.gmail.com>
 <CAJp-f1PsJhMzAJVaFbWFQWvOpo=qCNLAwaZGZ8uoikG-3V0=QQ@mail.gmail.com>
Message-ID: <CAPNTnnobB1HOqoWSMdD+__iCw895dktyRFmOeCXA1dthYhf7GA@mail.gmail.com>

Do you mead that the user ON THE OTHER END doesn't hear anything?  Because
I believe most devices will generate their own audible tone on the
generating end.

On Fri, Jul 7, 2017 at 3:42 PM, Mark Melling <mark.melling at savageminds.com>
wrote:

> Thanks for that, that is useful info.
>
> I posted a separate, but related question that you might have some
> suggestions about.
>
> Namely I'm assuming that for RFC-2833 DTMFs Freeswitch is generating a
> corresponding tone for the received DTMF (I don't know if this is actually
> true or not).  If this is the case whether it is possible to disable this
> tone so that basically a user pressing a key (that generates a DTMF)
> doesn't hear anything.
>
> Thanks
>
>
> On Fri, 7 Jul 2017 at 19:22 Guillermo Ruiz Camauer <grcamauer at gmail.com>
> wrote:
>
>> RFC-2833 sends the DTMF as a special packet within the RTP stream.
>> SIP-INFO sends it in the SIP dialog.
>> INBAND DTMF come as a series of packets within the RTP stream that must
>> be analyzed to find the frequencies corresponding to the different DTMF.  I
>> believe spandsp takes cares of this.
>>
>> Guillermo
>>
>> On Fri, Jul 7, 2017 at 1:14 PM, Mark Melling <
>> mark.melling at savageminds.com> wrote:
>>
>>> Thanks Brian.
>>>
>>>
>>> On Fri, 7 Jul 2017 at 16:51 Brian : <brians at iptel.co> wrote:
>>>
>>>> I believe thats 2833
>>>>
>>>> From the source:
>>>>
>>>> switch(dtmf->source) {
>>>>
>>>>                         case SWITCH_DTMF_INBAND_AUDIO:  /* From audio */
>>>>
>>>>                                 dtmf_source_str = "INBAND_AUDIO";
>>>>
>>>>                                 break;
>>>>
>>>>                         case SWITCH_DTMF_RTP:                   /*
>>>> From RTP as a telephone event */
>>>>
>>>>                                 dtmf_source_str = "RTP";
>>>>
>>>>                                 break;
>>>>
>>>>                         case SWITCH_DTMF_ENDPOINT:              /*
>>>> From endpoint signaling */
>>>>
>>>>                                 dtmf_source_str = "ENDPOINT";
>>>>
>>>>                                 break;
>>>>
>>>>                         case SWITCH_DTMF_APP:                   /*
>>>> Injected by application */
>>>>
>>>>                                 dtmf_source_str = "APP";
>>>>
>>>>                                 break;
>>>>
>>>>                         case SWITCH_DTMF_UNKNOWN:               /*
>>>> Unknown source */
>>>>
>>>>                         default:
>>>>
>>>>                                 dtmf_source_str = "UNKNOWN";
>>>>
>>>>                                 break;
>>>>
>>>> On Fri, Jul 7, 2017 at 4:28 PM, Mark Melling
>>>> <mark.melling at savageminds.com> wrote:
>>>> > Thanks Brian,
>>>> >
>>>> > Having done that it says
>>>> >
>>>> > DTMF-Source: RTP
>>>> >
>>>> > So I assume that means that the DTMF is in-band, is that right?
>>>> >
>>>> >
>>>> >
>>>> > On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:
>>>> >>
>>>> >> Hi Mark
>>>> >>
>>>> >> From fs_cli
>>>> >>
>>>> >> /event plain dtmf
>>>> >>
>>>> >> This will give you console logging of every DTMF event that FS
>>>> handles
>>>> >> and I'm pretty sure will give you the DTMF source.
>>>> >>
>>>> >>
>>>> >>
>>>> >> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
>>>> >> <mark.melling at savageminds.com> wrote:
>>>> >> > Hi,
>>>> >> >
>>>> >> > How can you determine if Freeswitch is receiving DTMFs in-band or
>>>> >> > out-of-band?
>>>> >> >
>>>> >> > I'm not entirely clear on terminology here, but is out-of-band the
>>>> same
>>>> >> > as
>>>> >> > rfc 2833?
>>>> >> >
>>>> >> > Thanks
>>>> >> >
>>>> >> > Mark
>>>> >> >
>>>> >> >
>>>> >> >
>>>> >> > ____________________________________________________________
>>>> _____________
>>>> >> > Professional FreeSWITCH Consulting Services:
>>>> >> > consulting at freeswitch.org
>>>> >> > http://www.freeswitchsolutions.com
>>>> >> >
>>>> >> > Official FreeSWITCH Sites
>>>> >> > http://www.freeswitch.org
>>>> >> > http://confluence.freeswitch.org
>>>> >> > http://www.cluecon.com
>>>> >> >
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >>
>>>> >> ____________________________________________________________
>>>> _____________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> consulting at freeswitch.org
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://confluence.freeswitch.org
>>>> >> http://www.cluecon.com
>>>> >>
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > ____________________________________________________________
>>>> _____________
>>>> > Professional FreeSWITCH Consulting Services:
>>>> > consulting at freeswitch.org
>>>> > http://www.freeswitchsolutions.com
>>>> >
>>>> > Official FreeSWITCH Sites
>>>> > http://www.freeswitch.org
>>>> > http://confluence.freeswitch.org
>>>> > http://www.cluecon.com
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Guillermo Ruiz Camauer
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Guillermo Ruiz Camauer
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From brian at freeswitch.org  Fri Jul  7 19:12:54 2017
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 Jul 2017 14:12:54 -0500
Subject: [Freeswitch-users] Software Dev Job
Message-ID: <CAEJMVkA9hfFRE61wA00dX8sy10wWjbiMHuHyFcS5ujQbE5W_Tw@mail.gmail.com>

FreeSWITCHers,

https://freeswitch-solutions-llc.workable.com/jobs/523308

Thanks,

-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From mark.melling at savageminds.com  Fri Jul  7 19:17:19 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Fri, 07 Jul 2017 19:17:19 +0000
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAPNTnnobB1HOqoWSMdD+__iCw895dktyRFmOeCXA1dthYhf7GA@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
 <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
 <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>
 <CAJp-f1Ne+31RsY0hmWXdZwCXF5Q3hJugshgEa8npN5iJ+EPvXw@mail.gmail.com>
 <CAPNTnnqMg_-w8q3nH_CfhRtoz0H_mFo7PZO5Yp6Ynb0ZRnHeOg@mail.gmail.com>
 <CAJp-f1PsJhMzAJVaFbWFQWvOpo=qCNLAwaZGZ8uoikG-3V0=QQ@mail.gmail.com>
 <CAPNTnnobB1HOqoWSMdD+__iCw895dktyRFmOeCXA1dthYhf7GA@mail.gmail.com>
Message-ID: <CAJp-f1N1h19-_qa=WM=FOdoeziarTY3HJ4XAmFjqhwNB1CnMsQ@mail.gmail.com>

Yes, I mean that the caller who is pressing the buttons (although in this
case the button presses are automated as they are included at the end of
the original dial string).

The reason why I believe this is possible is that in a different set-up,
not using Freeswitch (rather using Twilio) we were successfully able to
transmit DTMFs (as part of dial string) and they were not audible to the
caller.

I'm just wondering as I type this, if the fact that they were part of the
dial string is significant?



On Fri, 7 Jul 2017 at 20:05 Guillermo Ruiz Camauer <grcamauer at gmail.com>
wrote:

> Do you mead that the user ON THE OTHER END doesn't hear anything?  Because
> I believe most devices will generate their own audible tone on the
> generating end.
>
> On Fri, Jul 7, 2017 at 3:42 PM, Mark Melling <mark.melling at savageminds.com
> > wrote:
>
>> Thanks for that, that is useful info.
>>
>> I posted a separate, but related question that you might have some
>> suggestions about.
>>
>> Namely I'm assuming that for RFC-2833 DTMFs Freeswitch is generating a
>> corresponding tone for the received DTMF (I don't know if this is actually
>> true or not).  If this is the case whether it is possible to disable this
>> tone so that basically a user pressing a key (that generates a DTMF)
>> doesn't hear anything.
>>
>> Thanks
>>
>>
>> On Fri, 7 Jul 2017 at 19:22 Guillermo Ruiz Camauer <grcamauer at gmail.com>
>> wrote:
>>
>>> RFC-2833 sends the DTMF as a special packet within the RTP stream.
>>> SIP-INFO sends it in the SIP dialog.
>>> INBAND DTMF come as a series of packets within the RTP stream that must
>>> be analyzed to find the frequencies corresponding to the different DTMF.  I
>>> believe spandsp takes cares of this.
>>>
>>> Guillermo
>>>
>>> On Fri, Jul 7, 2017 at 1:14 PM, Mark Melling <
>>> mark.melling at savageminds.com> wrote:
>>>
>>>> Thanks Brian.
>>>>
>>>>
>>>> On Fri, 7 Jul 2017 at 16:51 Brian : <brians at iptel.co> wrote:
>>>>
>>>>> I believe thats 2833
>>>>>
>>>>> From the source:
>>>>>
>>>>> switch(dtmf->source) {
>>>>>
>>>>>                         case SWITCH_DTMF_INBAND_AUDIO:  /* From audio
>>>>> */
>>>>>
>>>>>                                 dtmf_source_str = "INBAND_AUDIO";
>>>>>
>>>>>                                 break;
>>>>>
>>>>>                         case SWITCH_DTMF_RTP:                   /*
>>>>> From RTP as a telephone event */
>>>>>
>>>>>                                 dtmf_source_str = "RTP";
>>>>>
>>>>>                                 break;
>>>>>
>>>>>                         case SWITCH_DTMF_ENDPOINT:              /*
>>>>> From endpoint signaling */
>>>>>
>>>>>                                 dtmf_source_str = "ENDPOINT";
>>>>>
>>>>>                                 break;
>>>>>
>>>>>                         case SWITCH_DTMF_APP:                   /*
>>>>> Injected by application */
>>>>>
>>>>>                                 dtmf_source_str = "APP";
>>>>>
>>>>>                                 break;
>>>>>
>>>>>                         case SWITCH_DTMF_UNKNOWN:               /*
>>>>> Unknown source */
>>>>>
>>>>>                         default:
>>>>>
>>>>>                                 dtmf_source_str = "UNKNOWN";
>>>>>
>>>>>                                 break;
>>>>>
>>>>> On Fri, Jul 7, 2017 at 4:28 PM, Mark Melling
>>>>> <mark.melling at savageminds.com> wrote:
>>>>> > Thanks Brian,
>>>>> >
>>>>> > Having done that it says
>>>>> >
>>>>> > DTMF-Source: RTP
>>>>> >
>>>>> > So I assume that means that the DTMF is in-band, is that right?
>>>>> >
>>>>> >
>>>>> >
>>>>> > On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:
>>>>> >>
>>>>> >> Hi Mark
>>>>> >>
>>>>> >> From fs_cli
>>>>> >>
>>>>> >> /event plain dtmf
>>>>> >>
>>>>> >> This will give you console logging of every DTMF event that FS
>>>>> handles
>>>>> >> and I'm pretty sure will give you the DTMF source.
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
>>>>> >> <mark.melling at savageminds.com> wrote:
>>>>> >> > Hi,
>>>>> >> >
>>>>> >> > How can you determine if Freeswitch is receiving DTMFs in-band or
>>>>> >> > out-of-band?
>>>>> >> >
>>>>> >> > I'm not entirely clear on terminology here, but is out-of-band
>>>>> the same
>>>>> >> > as
>>>>> >> > rfc 2833?
>>>>> >> >
>>>>> >> > Thanks
>>>>> >> >
>>>>> >> > Mark
>>>>> >> >
>>>>> >> >
>>>>> >> >
>>>>> >> >
>>>>> _________________________________________________________________________
>>>>> >> > Professional FreeSWITCH Consulting Services:
>>>>> >> > consulting at freeswitch.org
>>>>> >> > http://www.freeswitchsolutions.com
>>>>> >> >
>>>>> >> > Official FreeSWITCH Sites
>>>>> >> > http://www.freeswitch.org
>>>>> >> > http://confluence.freeswitch.org
>>>>> >> > http://www.cluecon.com
>>>>> >> >
>>>>> >> > FreeSWITCH-users mailing list
>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> > http://www.freeswitch.org
>>>>> >>
>>>>> >>
>>>>> _________________________________________________________________________
>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>> >> consulting at freeswitch.org
>>>>> >> http://www.freeswitchsolutions.com
>>>>> >>
>>>>> >> Official FreeSWITCH Sites
>>>>> >> http://www.freeswitch.org
>>>>> >> http://confluence.freeswitch.org
>>>>> >> http://www.cluecon.com
>>>>> >>
>>>>> >> FreeSWITCH-users mailing list
>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> http://www.freeswitch.org
>>>>> >
>>>>> >
>>>>> >
>>>>> _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at freeswitch.org
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
>>>>> > http://confluence.freeswitch.org
>>>>> > http://www.cluecon.com
>>>>> >
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Guillermo Ruiz Camauer
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Guillermo Ruiz Camauer
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From rbetancor at gmail.com  Fri Jul  7 19:42:43 2017
From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=)
Date: Fri, 7 Jul 2017 20:42:43 +0100
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAJp-f1N1h19-_qa=WM=FOdoeziarTY3HJ4XAmFjqhwNB1CnMsQ@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
 <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
 <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>
 <CAJp-f1Ne+31RsY0hmWXdZwCXF5Q3hJugshgEa8npN5iJ+EPvXw@mail.gmail.com>
 <CAPNTnnqMg_-w8q3nH_CfhRtoz0H_mFo7PZO5Yp6Ynb0ZRnHeOg@mail.gmail.com>
 <CAJp-f1PsJhMzAJVaFbWFQWvOpo=qCNLAwaZGZ8uoikG-3V0=QQ@mail.gmail.com>
 <CAPNTnnobB1HOqoWSMdD+__iCw895dktyRFmOeCXA1dthYhf7GA@mail.gmail.com>
 <CAJp-f1N1h19-_qa=WM=FOdoeziarTY3HJ4XAmFjqhwNB1CnMsQ@mail.gmail.com>
Message-ID: <CAKsfQBiy+2HsQPEHaUq4RMFLZFM3hV3n5PCLfvCU4x3DXfHcBw@mail.gmail.com>

So what your are looking it's to do some kind of callthrought, but don't
whant the caller to hear the DTMF ... regardless of if you could, on FS,
mute the DTMF or not ... if the caller phone plays them, and most of
smartphones DOES, they will be able to hear the tones.

2017-07-07 20:17 GMT+01:00 Mark Melling <mark.melling at savageminds.com>:

> Yes, I mean that the caller who is pressing the buttons (although in this
> case the button presses are automated as they are included at the end of
> the original dial string).
>
> The reason why I believe this is possible is that in a different set-up,
> not using Freeswitch (rather using Twilio) we were successfully able to
> transmit DTMFs (as part of dial string) and they were not audible to the
> caller.
>
> I'm just wondering as I type this, if the fact that they were part of the
> dial string is significant?
>
>
>
> On Fri, 7 Jul 2017 at 20:05 Guillermo Ruiz Camauer <grcamauer at gmail.com>
> wrote:
>
>> Do you mead that the user ON THE OTHER END doesn't hear anything?
>> Because I believe most devices will generate their own audible tone on the
>> generating end.
>>
>> On Fri, Jul 7, 2017 at 3:42 PM, Mark Melling <
>> mark.melling at savageminds.com> wrote:
>>
>>> Thanks for that, that is useful info.
>>>
>>> I posted a separate, but related question that you might have some
>>> suggestions about.
>>>
>>> Namely I'm assuming that for RFC-2833 DTMFs Freeswitch is generating a
>>> corresponding tone for the received DTMF (I don't know if this is actually
>>> true or not).  If this is the case whether it is possible to disable this
>>> tone so that basically a user pressing a key (that generates a DTMF)
>>> doesn't hear anything.
>>>
>>> Thanks
>>>
>>>
>>> On Fri, 7 Jul 2017 at 19:22 Guillermo Ruiz Camauer <grcamauer at gmail.com>
>>> wrote:
>>>
>>>> RFC-2833 sends the DTMF as a special packet within the RTP stream.
>>>> SIP-INFO sends it in the SIP dialog.
>>>> INBAND DTMF come as a series of packets within the RTP stream that must
>>>> be analyzed to find the frequencies corresponding to the different DTMF.  I
>>>> believe spandsp takes cares of this.
>>>>
>>>> Guillermo
>>>>
>>>> On Fri, Jul 7, 2017 at 1:14 PM, Mark Melling <
>>>> mark.melling at savageminds.com> wrote:
>>>>
>>>>> Thanks Brian.
>>>>>
>>>>>
>>>>> On Fri, 7 Jul 2017 at 16:51 Brian : <brians at iptel.co> wrote:
>>>>>
>>>>>> I believe thats 2833
>>>>>>
>>>>>> From the source:
>>>>>>
>>>>>> switch(dtmf->source) {
>>>>>>
>>>>>>                         case SWITCH_DTMF_INBAND_AUDIO:  /* From audio
>>>>>> */
>>>>>>
>>>>>>                                 dtmf_source_str = "INBAND_AUDIO";
>>>>>>
>>>>>>                                 break;
>>>>>>
>>>>>>                         case SWITCH_DTMF_RTP:                   /*
>>>>>> From RTP as a telephone event */
>>>>>>
>>>>>>                                 dtmf_source_str = "RTP";
>>>>>>
>>>>>>                                 break;
>>>>>>
>>>>>>                         case SWITCH_DTMF_ENDPOINT:              /*
>>>>>> From endpoint signaling */
>>>>>>
>>>>>>                                 dtmf_source_str = "ENDPOINT";
>>>>>>
>>>>>>                                 break;
>>>>>>
>>>>>>                         case SWITCH_DTMF_APP:                   /*
>>>>>> Injected by application */
>>>>>>
>>>>>>                                 dtmf_source_str = "APP";
>>>>>>
>>>>>>                                 break;
>>>>>>
>>>>>>                         case SWITCH_DTMF_UNKNOWN:               /*
>>>>>> Unknown source */
>>>>>>
>>>>>>                         default:
>>>>>>
>>>>>>                                 dtmf_source_str = "UNKNOWN";
>>>>>>
>>>>>>                                 break;
>>>>>>
>>>>>> On Fri, Jul 7, 2017 at 4:28 PM, Mark Melling
>>>>>> <mark.melling at savageminds.com> wrote:
>>>>>> > Thanks Brian,
>>>>>> >
>>>>>> > Having done that it says
>>>>>> >
>>>>>> > DTMF-Source: RTP
>>>>>> >
>>>>>> > So I assume that means that the DTMF is in-band, is that right?
>>>>>> >
>>>>>> >
>>>>>> >
>>>>>> > On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:
>>>>>> >>
>>>>>> >> Hi Mark
>>>>>> >>
>>>>>> >> From fs_cli
>>>>>> >>
>>>>>> >> /event plain dtmf
>>>>>> >>
>>>>>> >> This will give you console logging of every DTMF event that FS
>>>>>> handles
>>>>>> >> and I'm pretty sure will give you the DTMF source.
>>>>>> >>
>>>>>> >>
>>>>>> >>
>>>>>> >> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
>>>>>> >> <mark.melling at savageminds.com> wrote:
>>>>>> >> > Hi,
>>>>>> >> >
>>>>>> >> > How can you determine if Freeswitch is receiving DTMFs in-band or
>>>>>> >> > out-of-band?
>>>>>> >> >
>>>>>> >> > I'm not entirely clear on terminology here, but is out-of-band
>>>>>> the same
>>>>>> >> > as
>>>>>> >> > rfc 2833?
>>>>>> >> >
>>>>>> >> > Thanks
>>>>>> >> >
>>>>>> >> > Mark
>>>>>> >> >
>>>>>> >> >
>>>>>> >> >
>>>>>> >> > ____________________________________________________________
>>>>>> _____________
>>>>>> >> > Professional FreeSWITCH Consulting Services:
>>>>>> >> > consulting at freeswitch.org
>>>>>> >> > http://www.freeswitchsolutions.com
>>>>>> >> >
>>>>>> >> > Official FreeSWITCH Sites
>>>>>> >> > http://www.freeswitch.org
>>>>>> >> > http://confluence.freeswitch.org
>>>>>> >> > http://www.cluecon.com
>>>>>> >> >
>>>>>> >> > FreeSWITCH-users mailing list
>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>>>> options/freeswitch-users
>>>>>> >> > http://www.freeswitch.org
>>>>>> >>
>>>>>> >> ____________________________________________________________
>>>>>> _____________
>>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>>> >> consulting at freeswitch.org
>>>>>> >> http://www.freeswitchsolutions.com
>>>>>> >>
>>>>>> >> Official FreeSWITCH Sites
>>>>>> >> http://www.freeswitch.org
>>>>>> >> http://confluence.freeswitch.org
>>>>>> >> http://www.cluecon.com
>>>>>> >>
>>>>>> >> FreeSWITCH-users mailing list
>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>>>> options/freeswitch-users
>>>>>> >> http://www.freeswitch.org
>>>>>> >
>>>>>> >
>>>>>> > ____________________________________________________________
>>>>>> _____________
>>>>>> > Professional FreeSWITCH Consulting Services:
>>>>>> > consulting at freeswitch.org
>>>>>> > http://www.freeswitchsolutions.com
>>>>>> >
>>>>>> > Official FreeSWITCH Sites
>>>>>> > http://www.freeswitch.org
>>>>>> > http://confluence.freeswitch.org
>>>>>> > http://www.cluecon.com
>>>>>> >
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>>>> options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>>>> options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>>> options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Guillermo Ruiz Camauer
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Guillermo Ruiz Camauer
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mark.melling at savageminds.com  Fri Jul  7 20:07:11 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Fri, 07 Jul 2017 20:07:11 +0000
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAKsfQBiy+2HsQPEHaUq4RMFLZFM3hV3n5PCLfvCU4x3DXfHcBw@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
 <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
 <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>
 <CAJp-f1Ne+31RsY0hmWXdZwCXF5Q3hJugshgEa8npN5iJ+EPvXw@mail.gmail.com>
 <CAPNTnnqMg_-w8q3nH_CfhRtoz0H_mFo7PZO5Yp6Ynb0ZRnHeOg@mail.gmail.com>
 <CAJp-f1PsJhMzAJVaFbWFQWvOpo=qCNLAwaZGZ8uoikG-3V0=QQ@mail.gmail.com>
 <CAPNTnnobB1HOqoWSMdD+__iCw895dktyRFmOeCXA1dthYhf7GA@mail.gmail.com>
 <CAJp-f1N1h19-_qa=WM=FOdoeziarTY3HJ4XAmFjqhwNB1CnMsQ@mail.gmail.com>
 <CAKsfQBiy+2HsQPEHaUq4RMFLZFM3hV3n5PCLfvCU4x3DXfHcBw@mail.gmail.com>
Message-ID: <CAJp-f1MoOj1E3Cv_JkpKpUPVvu2ut-2HXF-_74HR9yxAa4Xmsg@mail.gmail.com>

> So what your are looking it's to do some kind of callthrought, but don't
whant the caller to hear the DTMF

This is what I'm trying to achieve. And to be clear the call is initiated
from an Android phone. The DTMF digits are part of the dial string of the
form

<phone-number>,<some-digits>#

I'm not disagreeing with your comment that DTMF tones are audible at the
smartphone end. I may have been approaching this problem in the wrong way,
but what I do know is that with the previous system - which wasn't using
Freeswitch, but Twilio we could successfully make calls from an android
device and the DTMF tones were not audible at the android end.

So what is going on that makes that possible?



On Fri, 7 Jul 2017 at 20:43 Raúl Alexis Betancor Santana <
rbetancor at gmail.com> wrote:

> So what your are looking it's to do some kind of callthrought, but don't
> whant the caller to hear the DTMF ... regardless of if you could, on FS,
> mute the DTMF or not ... if the caller phone plays them, and most of
> smartphones DOES, they will be able to hear the tones.
>
> 2017-07-07 20:17 GMT+01:00 Mark Melling <mark.melling at savageminds.com>:
>
>> Yes, I mean that the caller who is pressing the buttons (although in this
>> case the button presses are automated as they are included at the end of
>> the original dial string).
>>
>> The reason why I believe this is possible is that in a different set-up,
>> not using Freeswitch (rather using Twilio) we were successfully able to
>> transmit DTMFs (as part of dial string) and they were not audible to the
>> caller.
>>
>> I'm just wondering as I type this, if the fact that they were part of the
>> dial string is significant?
>>
>>
>>
>> On Fri, 7 Jul 2017 at 20:05 Guillermo Ruiz Camauer <grcamauer at gmail.com>
>> wrote:
>>
>>> Do you mead that the user ON THE OTHER END doesn't hear anything?
>>> Because I believe most devices will generate their own audible tone on the
>>> generating end.
>>>
>>> On Fri, Jul 7, 2017 at 3:42 PM, Mark Melling <
>>> mark.melling at savageminds.com> wrote:
>>>
>>>> Thanks for that, that is useful info.
>>>>
>>>> I posted a separate, but related question that you might have some
>>>> suggestions about.
>>>>
>>>> Namely I'm assuming that for RFC-2833 DTMFs Freeswitch is generating a
>>>> corresponding tone for the received DTMF (I don't know if this is actually
>>>> true or not).  If this is the case whether it is possible to disable this
>>>> tone so that basically a user pressing a key (that generates a DTMF)
>>>> doesn't hear anything.
>>>>
>>>> Thanks
>>>>
>>>>
>>>> On Fri, 7 Jul 2017 at 19:22 Guillermo Ruiz Camauer <grcamauer at gmail.com>
>>>> wrote:
>>>>
>>>>> RFC-2833 sends the DTMF as a special packet within the RTP stream.
>>>>> SIP-INFO sends it in the SIP dialog.
>>>>> INBAND DTMF come as a series of packets within the RTP stream that
>>>>> must be analyzed to find the frequencies corresponding to the different
>>>>> DTMF.  I believe spandsp takes cares of this.
>>>>>
>>>>> Guillermo
>>>>>
>>>>> On Fri, Jul 7, 2017 at 1:14 PM, Mark Melling <
>>>>> mark.melling at savageminds.com> wrote:
>>>>>
>>>>>> Thanks Brian.
>>>>>>
>>>>>>
>>>>>> On Fri, 7 Jul 2017 at 16:51 Brian : <brians at iptel.co> wrote:
>>>>>>
>>>>>>> I believe thats 2833
>>>>>>>
>>>>>>> From the source:
>>>>>>>
>>>>>>> switch(dtmf->source) {
>>>>>>>
>>>>>>>                         case SWITCH_DTMF_INBAND_AUDIO:  /* From
>>>>>>> audio */
>>>>>>>
>>>>>>>                                 dtmf_source_str = "INBAND_AUDIO";
>>>>>>>
>>>>>>>                                 break;
>>>>>>>
>>>>>>>                         case SWITCH_DTMF_RTP:                   /*
>>>>>>> From RTP as a telephone event */
>>>>>>>
>>>>>>>                                 dtmf_source_str = "RTP";
>>>>>>>
>>>>>>>                                 break;
>>>>>>>
>>>>>>>                         case SWITCH_DTMF_ENDPOINT:              /*
>>>>>>> From endpoint signaling */
>>>>>>>
>>>>>>>                                 dtmf_source_str = "ENDPOINT";
>>>>>>>
>>>>>>>                                 break;
>>>>>>>
>>>>>>>                         case SWITCH_DTMF_APP:                   /*
>>>>>>> Injected by application */
>>>>>>>
>>>>>>>                                 dtmf_source_str = "APP";
>>>>>>>
>>>>>>>                                 break;
>>>>>>>
>>>>>>>                         case SWITCH_DTMF_UNKNOWN:               /*
>>>>>>> Unknown source */
>>>>>>>
>>>>>>>                         default:
>>>>>>>
>>>>>>>                                 dtmf_source_str = "UNKNOWN";
>>>>>>>
>>>>>>>                                 break;
>>>>>>>
>>>>>>> On Fri, Jul 7, 2017 at 4:28 PM, Mark Melling
>>>>>>> <mark.melling at savageminds.com> wrote:
>>>>>>> > Thanks Brian,
>>>>>>> >
>>>>>>> > Having done that it says
>>>>>>> >
>>>>>>> > DTMF-Source: RTP
>>>>>>> >
>>>>>>> > So I assume that means that the DTMF is in-band, is that right?
>>>>>>> >
>>>>>>> >
>>>>>>> >
>>>>>>> > On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:
>>>>>>> >>
>>>>>>> >> Hi Mark
>>>>>>> >>
>>>>>>> >> From fs_cli
>>>>>>> >>
>>>>>>> >> /event plain dtmf
>>>>>>> >>
>>>>>>> >> This will give you console logging of every DTMF event that FS
>>>>>>> handles
>>>>>>> >> and I'm pretty sure will give you the DTMF source.
>>>>>>> >>
>>>>>>> >>
>>>>>>> >>
>>>>>>> >> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
>>>>>>> >> <mark.melling at savageminds.com> wrote:
>>>>>>> >> > Hi,
>>>>>>> >> >
>>>>>>> >> > How can you determine if Freeswitch is receiving DTMFs in-band
>>>>>>> or
>>>>>>> >> > out-of-band?
>>>>>>> >> >
>>>>>>> >> > I'm not entirely clear on terminology here, but is out-of-band
>>>>>>> the same
>>>>>>> >> > as
>>>>>>> >> > rfc 2833?
>>>>>>> >> >
>>>>>>> >> > Thanks
>>>>>>> >> >
>>>>>>> >> > Mark
>>>>>>> >> >
>>>>>>> >> >
>>>>>>> >> >
>>>>>>> >> >
>>>>>>> _________________________________________________________________________
>>>>>>> >> > Professional FreeSWITCH Consulting Services:
>>>>>>> >> > consulting at freeswitch.org
>>>>>>> >> > http://www.freeswitchsolutions.com
>>>>>>> >> >
>>>>>>> >> > Official FreeSWITCH Sites
>>>>>>> >> > http://www.freeswitch.org
>>>>>>> >> > http://confluence.freeswitch.org
>>>>>>> >> > http://www.cluecon.com
>>>>>>> >> >
>>>>>>> >> > FreeSWITCH-users mailing list
>>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> > UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >> > http://www.freeswitch.org
>>>>>>> >>
>>>>>>> >>
>>>>>>> _________________________________________________________________________
>>>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>>>> >> consulting at freeswitch.org
>>>>>>> >> http://www.freeswitchsolutions.com
>>>>>>> >>
>>>>>>> >> Official FreeSWITCH Sites
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >> http://confluence.freeswitch.org
>>>>>>> >> http://www.cluecon.com
>>>>>>> >>
>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> >> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> >> http://www.freeswitch.org
>>>>>>> >
>>>>>>> >
>>>>>>> >
>>>>>>> _________________________________________________________________________
>>>>>>> > Professional FreeSWITCH Consulting Services:
>>>>>>> > consulting at freeswitch.org
>>>>>>> > http://www.freeswitchsolutions.com
>>>>>>> >
>>>>>>> > Official FreeSWITCH Sites
>>>>>>> > http://www.freeswitch.org
>>>>>>> > http://confluence.freeswitch.org
>>>>>>> > http://www.cluecon.com
>>>>>>> >
>>>>>>> > FreeSWITCH-users mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> > UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> > http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Guillermo Ruiz Camauer
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Guillermo Ruiz Camauer
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mark.melling at savageminds.com  Fri Jul  7 20:18:49 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Fri, 07 Jul 2017 20:18:49 +0000
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAJp-f1MoOj1E3Cv_JkpKpUPVvu2ut-2HXF-_74HR9yxAa4Xmsg@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
 <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
 <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>
 <CAJp-f1Ne+31RsY0hmWXdZwCXF5Q3hJugshgEa8npN5iJ+EPvXw@mail.gmail.com>
 <CAPNTnnqMg_-w8q3nH_CfhRtoz0H_mFo7PZO5Yp6Ynb0ZRnHeOg@mail.gmail.com>
 <CAJp-f1PsJhMzAJVaFbWFQWvOpo=qCNLAwaZGZ8uoikG-3V0=QQ@mail.gmail.com>
 <CAPNTnnobB1HOqoWSMdD+__iCw895dktyRFmOeCXA1dthYhf7GA@mail.gmail.com>
 <CAJp-f1N1h19-_qa=WM=FOdoeziarTY3HJ4XAmFjqhwNB1CnMsQ@mail.gmail.com>
 <CAKsfQBiy+2HsQPEHaUq4RMFLZFM3hV3n5PCLfvCU4x3DXfHcBw@mail.gmail.com>
 <CAJp-f1MoOj1E3Cv_JkpKpUPVvu2ut-2HXF-_74HR9yxAa4Xmsg@mail.gmail.com>
Message-ID: <CAJp-f1MDkF_MdUQx+X=-d0sTUMKTdGndm_whN+rE-e4m6Jg_2A@mail.gmail.com>

Just to add to my previous comment - it was because with our previous
set-up where DTMF digits at the end of the dial string were not audible
that I assumed that it must be Freeswitch that was generating a tone for
each received DTMF - may be this assumption is wrong.



On Fri, 7 Jul 2017 at 21:07 Mark Melling <mark.melling at savageminds.com>
wrote:

> > So what your are looking it's to do some kind of callthrought, but
> don't whant the caller to hear the DTMF
>
> This is what I'm trying to achieve. And to be clear the call is initiated
> from an Android phone. The DTMF digits are part of the dial string of the
> form
>
> <phone-number>,<some-digits>#
>
> I'm not disagreeing with your comment that DTMF tones are audible at the
> smartphone end. I may have been approaching this problem in the wrong way,
> but what I do know is that with the previous system - which wasn't using
> Freeswitch, but Twilio we could successfully make calls from an android
> device and the DTMF tones were not audible at the android end.
>
> So what is going on that makes that possible?
>
>
>
> On Fri, 7 Jul 2017 at 20:43 Raúl Alexis Betancor Santana <
> rbetancor at gmail.com> wrote:
>
>> So what your are looking it's to do some kind of callthrought, but don't
>> whant the caller to hear the DTMF ... regardless of if you could, on FS,
>> mute the DTMF or not ... if the caller phone plays them, and most of
>> smartphones DOES, they will be able to hear the tones.
>>
>> 2017-07-07 20:17 GMT+01:00 Mark Melling <mark.melling at savageminds.com>:
>>
>>> Yes, I mean that the caller who is pressing the buttons (although in
>>> this case the button presses are automated as they are included at the end
>>> of the original dial string).
>>>
>>> The reason why I believe this is possible is that in a different set-up,
>>> not using Freeswitch (rather using Twilio) we were successfully able to
>>> transmit DTMFs (as part of dial string) and they were not audible to the
>>> caller.
>>>
>>> I'm just wondering as I type this, if the fact that they were part of
>>> the dial string is significant?
>>>
>>>
>>>
>>> On Fri, 7 Jul 2017 at 20:05 Guillermo Ruiz Camauer <grcamauer at gmail.com>
>>> wrote:
>>>
>>>> Do you mead that the user ON THE OTHER END doesn't hear anything?
>>>> Because I believe most devices will generate their own audible tone on the
>>>> generating end.
>>>>
>>>> On Fri, Jul 7, 2017 at 3:42 PM, Mark Melling <
>>>> mark.melling at savageminds.com> wrote:
>>>>
>>>>> Thanks for that, that is useful info.
>>>>>
>>>>> I posted a separate, but related question that you might have some
>>>>> suggestions about.
>>>>>
>>>>> Namely I'm assuming that for RFC-2833 DTMFs Freeswitch is generating a
>>>>> corresponding tone for the received DTMF (I don't know if this is actually
>>>>> true or not).  If this is the case whether it is possible to disable this
>>>>> tone so that basically a user pressing a key (that generates a DTMF)
>>>>> doesn't hear anything.
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> On Fri, 7 Jul 2017 at 19:22 Guillermo Ruiz Camauer <
>>>>> grcamauer at gmail.com> wrote:
>>>>>
>>>>>> RFC-2833 sends the DTMF as a special packet within the RTP stream.
>>>>>> SIP-INFO sends it in the SIP dialog.
>>>>>> INBAND DTMF come as a series of packets within the RTP stream that
>>>>>> must be analyzed to find the frequencies corresponding to the different
>>>>>> DTMF.  I believe spandsp takes cares of this.
>>>>>>
>>>>>> Guillermo
>>>>>>
>>>>>> On Fri, Jul 7, 2017 at 1:14 PM, Mark Melling <
>>>>>> mark.melling at savageminds.com> wrote:
>>>>>>
>>>>>>> Thanks Brian.
>>>>>>>
>>>>>>>
>>>>>>> On Fri, 7 Jul 2017 at 16:51 Brian : <brians at iptel.co> wrote:
>>>>>>>
>>>>>>>> I believe thats 2833
>>>>>>>>
>>>>>>>> From the source:
>>>>>>>>
>>>>>>>> switch(dtmf->source) {
>>>>>>>>
>>>>>>>>                         case SWITCH_DTMF_INBAND_AUDIO:  /* From
>>>>>>>> audio */
>>>>>>>>
>>>>>>>>                                 dtmf_source_str = "INBAND_AUDIO";
>>>>>>>>
>>>>>>>>                                 break;
>>>>>>>>
>>>>>>>>                         case SWITCH_DTMF_RTP:                   /*
>>>>>>>> From RTP as a telephone event */
>>>>>>>>
>>>>>>>>                                 dtmf_source_str = "RTP";
>>>>>>>>
>>>>>>>>                                 break;
>>>>>>>>
>>>>>>>>                         case SWITCH_DTMF_ENDPOINT:              /*
>>>>>>>> From endpoint signaling */
>>>>>>>>
>>>>>>>>                                 dtmf_source_str = "ENDPOINT";
>>>>>>>>
>>>>>>>>                                 break;
>>>>>>>>
>>>>>>>>                         case SWITCH_DTMF_APP:                   /*
>>>>>>>> Injected by application */
>>>>>>>>
>>>>>>>>                                 dtmf_source_str = "APP";
>>>>>>>>
>>>>>>>>                                 break;
>>>>>>>>
>>>>>>>>                         case SWITCH_DTMF_UNKNOWN:               /*
>>>>>>>> Unknown source */
>>>>>>>>
>>>>>>>>                         default:
>>>>>>>>
>>>>>>>>                                 dtmf_source_str = "UNKNOWN";
>>>>>>>>
>>>>>>>>                                 break;
>>>>>>>>
>>>>>>>> On Fri, Jul 7, 2017 at 4:28 PM, Mark Melling
>>>>>>>> <mark.melling at savageminds.com> wrote:
>>>>>>>> > Thanks Brian,
>>>>>>>> >
>>>>>>>> > Having done that it says
>>>>>>>> >
>>>>>>>> > DTMF-Source: RTP
>>>>>>>> >
>>>>>>>> > So I assume that means that the DTMF is in-band, is that right?
>>>>>>>> >
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:
>>>>>>>> >>
>>>>>>>> >> Hi Mark
>>>>>>>> >>
>>>>>>>> >> From fs_cli
>>>>>>>> >>
>>>>>>>> >> /event plain dtmf
>>>>>>>> >>
>>>>>>>> >> This will give you console logging of every DTMF event that FS
>>>>>>>> handles
>>>>>>>> >> and I'm pretty sure will give you the DTMF source.
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
>>>>>>>> >> <mark.melling at savageminds.com> wrote:
>>>>>>>> >> > Hi,
>>>>>>>> >> >
>>>>>>>> >> > How can you determine if Freeswitch is receiving DTMFs in-band
>>>>>>>> or
>>>>>>>> >> > out-of-band?
>>>>>>>> >> >
>>>>>>>> >> > I'm not entirely clear on terminology here, but is out-of-band
>>>>>>>> the same
>>>>>>>> >> > as
>>>>>>>> >> > rfc 2833?
>>>>>>>> >> >
>>>>>>>> >> > Thanks
>>>>>>>> >> >
>>>>>>>> >> > Mark
>>>>>>>> >> >
>>>>>>>> >> >
>>>>>>>> >> >
>>>>>>>> >> >
>>>>>>>> _________________________________________________________________________
>>>>>>>> >> > Professional FreeSWITCH Consulting Services:
>>>>>>>> >> > consulting at freeswitch.org
>>>>>>>> >> > http://www.freeswitchsolutions.com
>>>>>>>> >> >
>>>>>>>> >> > Official FreeSWITCH Sites
>>>>>>>> >> > http://www.freeswitch.org
>>>>>>>> >> > http://confluence.freeswitch.org
>>>>>>>> >> > http://www.cluecon.com
>>>>>>>> >> >
>>>>>>>> >> > FreeSWITCH-users mailing list
>>>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> > UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >> > http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> _________________________________________________________________________
>>>>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>>>>> >> consulting at freeswitch.org
>>>>>>>> >> http://www.freeswitchsolutions.com
>>>>>>>> >>
>>>>>>>> >> Official FreeSWITCH Sites
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >> http://confluence.freeswitch.org
>>>>>>>> >> http://www.cluecon.com
>>>>>>>> >>
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>> >
>>>>>>>> _________________________________________________________________________
>>>>>>>> > Professional FreeSWITCH Consulting Services:
>>>>>>>> > consulting at freeswitch.org
>>>>>>>> > http://www.freeswitchsolutions.com
>>>>>>>> >
>>>>>>>> > Official FreeSWITCH Sites
>>>>>>>> > http://www.freeswitch.org
>>>>>>>> > http://confluence.freeswitch.org
>>>>>>>> > http://www.cluecon.com
>>>>>>>> >
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> > http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Guillermo Ruiz Camauer
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Guillermo Ruiz Camauer
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From nbhatti at gmail.com  Fri Jul  7 20:32:37 2017
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Fri, 7 Jul 2017 13:32:37 -0700
Subject: [Freeswitch-users] How do you determine if DTMFs received are
 in-band or out-of-band?
In-Reply-To: <CAJp-f1MDkF_MdUQx+X=-d0sTUMKTdGndm_whN+rE-e4m6Jg_2A@mail.gmail.com>
References: <CAJp-f1O4HuMizTGU+s6A-mFKNkTiJ2iSHd+n88Ns4oUVMdnyyA@mail.gmail.com>
 <CAGPQfi8HkBmZUbLLyKwDjHTKaU5gYDEmZazEqEwHkqTXUtYv=w@mail.gmail.com>
 <CAJp-f1Nk23icqubPdDxzWciDBZDf3nQb48LM9e4iEEb2k2=1JQ@mail.gmail.com>
 <CAGPQfi9CDRzu5gV-DYTCNS-gGGFeLTTPJmAExpmtQuhHyECzxw@mail.gmail.com>
 <CAJp-f1Ne+31RsY0hmWXdZwCXF5Q3hJugshgEa8npN5iJ+EPvXw@mail.gmail.com>
 <CAPNTnnqMg_-w8q3nH_CfhRtoz0H_mFo7PZO5Yp6Ynb0ZRnHeOg@mail.gmail.com>
 <CAJp-f1PsJhMzAJVaFbWFQWvOpo=qCNLAwaZGZ8uoikG-3V0=QQ@mail.gmail.com>
 <CAPNTnnobB1HOqoWSMdD+__iCw895dktyRFmOeCXA1dthYhf7GA@mail.gmail.com>
 <CAJp-f1N1h19-_qa=WM=FOdoeziarTY3HJ4XAmFjqhwNB1CnMsQ@mail.gmail.com>
 <CAKsfQBiy+2HsQPEHaUq4RMFLZFM3hV3n5PCLfvCU4x3DXfHcBw@mail.gmail.com>
 <CAJp-f1MoOj1E3Cv_JkpKpUPVvu2ut-2HXF-_74HR9yxAa4Xmsg@mail.gmail.com>
 <CAJp-f1MDkF_MdUQx+X=-d0sTUMKTdGndm_whN+rE-e4m6Jg_2A@mail.gmail.com>
Message-ID: <CACak_yxKMmFs2BrZXgmkuoZJc+s1Ck2iQsQ2POQe1ftjVsxZLw@mail.gmail.com>

Maybe you are looking for drop_dtmf()
https://wiki.freeswitch.org/wiki/Variable_drop_dtmf


-- 

Sent with Airmail

From: Mark Melling <mark.melling at savageminds.com>
<mark.melling at savageminds.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
<freeswitch-users at lists.freeswitch.org>
Date: July 8, 2017 at 1:23:37 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
<freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] How do you determine if DTMFs received are
in-band or out-of-band?

Just to add to my previous comment - it was because with our previous
set-up where DTMF digits at the end of the dial string were not audible
that I assumed that it must be Freeswitch that was generating a tone for
each received DTMF - may be this assumption is wrong.



On Fri, 7 Jul 2017 at 21:07 Mark Melling <mark.melling at savageminds.com>
wrote:

> > So what your are looking it's to do some kind of callthrought, but
> don't whant the caller to hear the DTMF
>
> This is what I'm trying to achieve. And to be clear the call is initiated
> from an Android phone. The DTMF digits are part of the dial string of the
> form
>
> <phone-number>,<some-digits>#
>
> I'm not disagreeing with your comment that DTMF tones are audible at the
> smartphone end. I may have been approaching this problem in the wrong way,
> but what I do know is that with the previous system - which wasn't using
> Freeswitch, but Twilio we could successfully make calls from an android
> device and the DTMF tones were not audible at the android end.
>
> So what is going on that makes that possible?
>
>
>
> On Fri, 7 Jul 2017 at 20:43 Raúl Alexis Betancor Santana <
> rbetancor at gmail.com> wrote:
>
>> So what your are looking it's to do some kind of callthrought, but don't
>> whant the caller to hear the DTMF ... regardless of if you could, on FS,
>> mute the DTMF or not ... if the caller phone plays them, and most of
>> smartphones DOES, they will be able to hear the tones.
>>
>> 2017-07-07 20:17 GMT+01:00 Mark Melling <mark.melling at savageminds.com>:
>>
>>> Yes, I mean that the caller who is pressing the buttons (although in
>>> this case the button presses are automated as they are included at the end
>>> of the original dial string).
>>>
>>> The reason why I believe this is possible is that in a different set-up,
>>> not using Freeswitch (rather using Twilio) we were successfully able to
>>> transmit DTMFs (as part of dial string) and they were not audible to the
>>> caller.
>>>
>>> I'm just wondering as I type this, if the fact that they were part of
>>> the dial string is significant?
>>>
>>>
>>>
>>> On Fri, 7 Jul 2017 at 20:05 Guillermo Ruiz Camauer <grcamauer at gmail.com>
>>> wrote:
>>>
>>>> Do you mead that the user ON THE OTHER END doesn't hear anything?
>>>> Because I believe most devices will generate their own audible tone on the
>>>> generating end.
>>>>
>>>> On Fri, Jul 7, 2017 at 3:42 PM, Mark Melling <
>>>> mark.melling at savageminds.com> wrote:
>>>>
>>>>> Thanks for that, that is useful info.
>>>>>
>>>>> I posted a separate, but related question that you might have some
>>>>> suggestions about.
>>>>>
>>>>> Namely I'm assuming that for RFC-2833 DTMFs Freeswitch is generating a
>>>>> corresponding tone for the received DTMF (I don't know if this is actually
>>>>> true or not).  If this is the case whether it is possible to disable this
>>>>> tone so that basically a user pressing a key (that generates a DTMF)
>>>>> doesn't hear anything.
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> On Fri, 7 Jul 2017 at 19:22 Guillermo Ruiz Camauer <
>>>>> grcamauer at gmail.com> wrote:
>>>>>
>>>>>> RFC-2833 sends the DTMF as a special packet within the RTP stream.
>>>>>> SIP-INFO sends it in the SIP dialog.
>>>>>> INBAND DTMF come as a series of packets within the RTP stream that
>>>>>> must be analyzed to find the frequencies corresponding to the different
>>>>>> DTMF.  I believe spandsp takes cares of this.
>>>>>>
>>>>>> Guillermo
>>>>>>
>>>>>> On Fri, Jul 7, 2017 at 1:14 PM, Mark Melling <
>>>>>> mark.melling at savageminds.com> wrote:
>>>>>>
>>>>>>> Thanks Brian.
>>>>>>>
>>>>>>>
>>>>>>> On Fri, 7 Jul 2017 at 16:51 Brian : <brians at iptel.co> wrote:
>>>>>>>
>>>>>>>> I believe thats 2833
>>>>>>>>
>>>>>>>> From the source:
>>>>>>>>
>>>>>>>> switch(dtmf->source) {
>>>>>>>>
>>>>>>>>                         case SWITCH_DTMF_INBAND_AUDIO:  /* From
>>>>>>>> audio */
>>>>>>>>
>>>>>>>>                                 dtmf_source_str = "INBAND_AUDIO";
>>>>>>>>
>>>>>>>>                                 break;
>>>>>>>>
>>>>>>>>                         case SWITCH_DTMF_RTP:                   /*
>>>>>>>> From RTP as a telephone event */
>>>>>>>>
>>>>>>>>                                 dtmf_source_str = "RTP";
>>>>>>>>
>>>>>>>>                                 break;
>>>>>>>>
>>>>>>>>                         case SWITCH_DTMF_ENDPOINT:              /*
>>>>>>>> From endpoint signaling */
>>>>>>>>
>>>>>>>>                                 dtmf_source_str = "ENDPOINT";
>>>>>>>>
>>>>>>>>                                 break;
>>>>>>>>
>>>>>>>>                         case SWITCH_DTMF_APP:                   /*
>>>>>>>> Injected by application */
>>>>>>>>
>>>>>>>>                                 dtmf_source_str = "APP";
>>>>>>>>
>>>>>>>>                                 break;
>>>>>>>>
>>>>>>>>                         case SWITCH_DTMF_UNKNOWN:               /*
>>>>>>>> Unknown source */
>>>>>>>>
>>>>>>>>                         default:
>>>>>>>>
>>>>>>>>                                 dtmf_source_str = "UNKNOWN";
>>>>>>>>
>>>>>>>>                                 break;
>>>>>>>>
>>>>>>>> On Fri, Jul 7, 2017 at 4:28 PM, Mark Melling
>>>>>>>> <mark.melling at savageminds.com> wrote:
>>>>>>>> > Thanks Brian,
>>>>>>>> >
>>>>>>>> > Having done that it says
>>>>>>>> >
>>>>>>>> > DTMF-Source: RTP
>>>>>>>> >
>>>>>>>> > So I assume that means that the DTMF is in-band, is that right?
>>>>>>>> >
>>>>>>>> >
>>>>>>>> >
>>>>>>>> > On Fri, 7 Jul 2017 at 16:19 Brian : <brians at iptel.co> wrote:
>>>>>>>> >>
>>>>>>>> >> Hi Mark
>>>>>>>> >>
>>>>>>>> >> From fs_cli
>>>>>>>> >>
>>>>>>>> >> /event plain dtmf
>>>>>>>> >>
>>>>>>>> >> This will give you console logging of every DTMF event that FS
>>>>>>>> handles
>>>>>>>> >> and I'm pretty sure will give you the DTMF source.
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> >> On Fri, Jul 7, 2017 at 10:44 AM, Mark Melling
>>>>>>>> >> <mark.melling at savageminds.com> wrote:
>>>>>>>> >> > Hi,
>>>>>>>> >> >
>>>>>>>> >> > How can you determine if Freeswitch is receiving DTMFs in-band
>>>>>>>> or
>>>>>>>> >> > out-of-band?
>>>>>>>> >> >
>>>>>>>> >> > I'm not entirely clear on terminology here, but is out-of-band
>>>>>>>> the same
>>>>>>>> >> > as
>>>>>>>> >> > rfc 2833?
>>>>>>>> >> >
>>>>>>>> >> > Thanks
>>>>>>>> >> >
>>>>>>>> >> > Mark
>>>>>>>> >> >
>>>>>>>> >> >
>>>>>>>> >> >
>>>>>>>> >> >
>>>>>>>> _________________________________________________________________________
>>>>>>>> >> > Professional FreeSWITCH Consulting Services:
>>>>>>>> >> > consulting at freeswitch.org
>>>>>>>> >> > http://www.freeswitchsolutions.com
>>>>>>>> >> >
>>>>>>>> >> > Official FreeSWITCH Sites
>>>>>>>> >> > http://www.freeswitch.org
>>>>>>>> >> > http://confluence.freeswitch.org
>>>>>>>> >> > http://www.cluecon.com
>>>>>>>> >> >
>>>>>>>> >> > FreeSWITCH-users mailing list
>>>>>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> > UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >> > http://www.freeswitch.org
>>>>>>>> >>
>>>>>>>> >>
>>>>>>>> _________________________________________________________________________
>>>>>>>> >> Professional FreeSWITCH Consulting Services:
>>>>>>>> >> consulting at freeswitch.org
>>>>>>>> >> http://www.freeswitchsolutions.com
>>>>>>>> >>
>>>>>>>> >> Official FreeSWITCH Sites
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >> http://confluence.freeswitch.org
>>>>>>>> >> http://www.cluecon.com
>>>>>>>> >>
>>>>>>>> >> FreeSWITCH-users mailing list
>>>>>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> >> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> >> http://www.freeswitch.org
>>>>>>>> >
>>>>>>>> >
>>>>>>>> >
>>>>>>>> _________________________________________________________________________
>>>>>>>> > Professional FreeSWITCH Consulting Services:
>>>>>>>> > consulting at freeswitch.org
>>>>>>>> > http://www.freeswitchsolutions.com
>>>>>>>> >
>>>>>>>> > Official FreeSWITCH Sites
>>>>>>>> > http://www.freeswitch.org
>>>>>>>> > http://confluence.freeswitch.org
>>>>>>>> > http://www.cluecon.com
>>>>>>>> >
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> > http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Guillermo Ruiz Camauer
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Guillermo Ruiz Camauer
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From joelists at tm.net.uk  Sat Jul  8 18:00:21 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Sat, 8 Jul 2017 19:00:21 +0100
Subject: [Freeswitch-users] Generate a custom log alert on failed SIP
	Registration
Message-ID: <85073F2B-E3FB-4B1C-B951-0DAF475D00D6@tm.net.uk>

Hi Guys

Wondering if it is possible, and if so how, to create a custom log ALERT entry on a failed SIP Registration?

Basically I want to output an ALERT to the log file with “Auth Failed ${ip of register}

I want to drop logging level down to Alert as this is going to be a high volume SBC, but I want to be able to use fail2ban to block on failed registers.

Any help would be much appreciated.

Regards

From ssinyagin at gmail.com  Sat Jul  8 19:28:55 2017
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sat, 8 Jul 2017 21:28:55 +0200
Subject: [Freeswitch-users] SIM7100 LTE modem with USB audio
Message-ID: <CACOYK=jtpcNSyzveoWKVRssHad-_6CwJtZ8H72d=qnHNS5niVA@mail.gmail.com>

Simcom has recently released a new 4G/LTE modem, and it has USB audio support.
You can find sim7100_usb_audio_application_note_v0.01.pdf   with
details at the vendor site, or at techship.com  after registration.

It transmits 8kHZ, 16-bit PCM audio via a USB UART simulation.

So, in theory, gsmopen module may be adapted to it (or maybe a new
module is worth starting).

I ordered a sample, will check it out soon.

cheers,
stanislav


From rich.freeswitch at branham.us  Sat Jul  8 20:56:41 2017
From: rich.freeswitch at branham.us (Richard-Freeswitch)
Date: Sat, 08 Jul 2017 16:56:41 -0400
Subject: [Freeswitch-users] One-way audio on first leg of attended transfer
Message-ID: <16945347435b095d106c63fab950fce2@branham.us>

I'm experiencing one-way audio on the first leg of an attended transfer. 
  The originator of the transfer hears audio but the destination user 
does not.

The one clue I have so far:  the FS log shows a PGRES_FATAL_ERROR and 
the Postgresql log says 'ERROR:  duplicate key value violates unique 
constraint "calls_pkey"', and the channels record update also fails as a 
result of the calls update failure.

Any ideas on the cause of this issue?  Log lines from fs_cli are pasted 
below.

Thanks!





2017-07-07 16:15:28.819969 [NOTICE] switch_rtp.c:1275 Auto Changing 
audio stun/rtp/dtls port from 111.111.111.111:54590 to 10.10.10.10:54590

2017-07-07 16:15:28.829991 [DEBUG] switch_pgsql.c:415 Query (insert into 
calls 
(call_uuid,call_created,call_created_epoch,caller_uuid,callee_uuid,hostname) 
values ('f292dc4c-e1af-47b9-914d-cf8e7953e033','2017-07-07 
16:15:28','1499544928','14fbd555-a49d-45c2-81c1-6a149446d6db','659c9953-685e-4e74-bd89-2da2f0973d79','freeswitch')) 
returned PGRES_FATAL_ERROR

2017-07-07 16:15:28.829991 [DEBUG] switch_pgsql.c:415 Query (update 
channels set 
read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000' 
where uuid='659c9953-685e-4e74-bd89-2da2f0973d79') returned 
PGRES_FATAL_ERROR

2017-07-07 16:15:28.829991 [ERR] switch_pgsql.c:656 Error executing 
query:  ERROR:  current transaction is aborted, commands ignored until 
end of transaction block


From joel at gogii.net  Sat Jul  8 23:14:34 2017
From: joel at gogii.net (Joel Serrano)
Date: Sat, 8 Jul 2017 16:14:34 -0700
Subject: [Freeswitch-users] Generate a custom log alert on failed SIP
	Registration
In-Reply-To: <85073F2B-E3FB-4B1C-B951-0DAF475D00D6@tm.net.uk>
References: <85073F2B-E3FB-4B1C-B951-0DAF475D00D6@tm.net.uk>
Message-ID: <CAMtXxQ=P5xh4kiRPadzFMd0J5oXAtYH1-mEPfFO=iLeFPmA9fg@mail.gmail.com>

mod_fail2ban?

https://freeswitch.org/confluence/display/FREESWITCH/mod_fail2ban



On Sat, Jul 8, 2017 at 11:00 AM, Joseph Waite <joelists at tm.net.uk> wrote:

> Hi Guys
>
> Wondering if it is possible, and if so how, to create a custom log ALERT
> entry on a failed SIP Registration?
>
> Basically I want to output an ALERT to the log file with “Auth Failed ${ip
> of register}
>
> I want to drop logging level down to Alert as this is going to be a high
> volume SBC, but I want to be able to use fail2ban to block on failed
> registers.
>
> Any help would be much appreciated.
>
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From joel at gogii.net  Sun Jul  9 02:56:59 2017
From: joel at gogii.net (Joel Serrano)
Date: Sat, 8 Jul 2017 19:56:59 -0700
Subject: [Freeswitch-users] sip to webrtc - sdp invalid description
In-Reply-To: <CAFGRPVrNFQ6rKYVehMX1bopaUqKqT0NQeDQkGigCGC5e5h7FUg@mail.gmail.com>
References: <CAFGRPVoQB8mH3a8JYbY3wP1L=L3x-zdZF-kK1o+ocW5GzSYt-A@mail.gmail.com>
 <CAFGRPVrNFQ6rKYVehMX1bopaUqKqT0NQeDQkGigCGC5e5h7FUg@mail.gmail.com>
Message-ID: <CAMtXxQm643PE7EJr11Tf7rkvoXuURicDfkR3GeuROQUSPkiECQ@mail.gmail.com>

Can it be that Zoiper is using SRTP-3DES instead of SRTP-DTLS?

WebRTC requires SRTP-DTLS.



On Fri, Jul 7, 2017 at 9:46 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello guys,
>
> Any help on this?
>
> Thanks
>
> David
> ᐧ
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337 <+34%20669%2044%2083%2037>
>
> On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello guys,
>>
>> I have this setup:
>>
>> Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
>>
>> Whenever webrtc clients call each other, calls are ok.
>> But when the zoiper (regular sip/tcp) calls, the browsers complaint about:
>>
>> "no ice-ufrag" (firefox) or
>> "Failed to set remote offer sdp: Called with SDP without DTLS
>> fingerprint." (Chrome).
>>
>> I am setting the sdp in freeswitch as:
>>
>>       <action application="set" data="proxy_media=false"/>
>>       <action application="set" data="bypass_media=false"/>
>>       <action application="export" data="zrtp_secure_media=false"/>
>>       <action application="export" data="nolocal:rtp_secure_media=true"/>
>>
>> and the actual sdp is:
>>
>> v=0
>> o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4
>> s=FreeSWITCH
>> c=IN IP4 1.2.3.4
>> t=0 0
>> m=audio 40954 RTP/SAVP 8 101 13
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVr
>> VfjA0ImWgu5WyLK0vtg60RTk6Koe8c0sRkjU
>> a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+D5NVoKFFw0
>> 6I0Xok/9KbRystQ
>> a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+ETPuCOiOIxjFLRjbF
>> bDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg
>> a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:evU8MzAtiSHwKb95s4V9IAM
>> pmok06kW9ZGDgH3/Lc3ZytVn2SR4
>> a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:KKDkT0DssohSeKFsX6tbixR
>> hwYdiIhE6r3u5CCVA
>> a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:D3JgGuOlIxXfHGdqf7lKWqN
>> DAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w
>> a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:c12ebWWzZ1cqZN0v5C5uYzd
>> vtfnw6AARU3+jGA0WzTSDlDd20vI
>> a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:nPZku7S6hlz2OPAff8T9I8s
>> NFzuZziNg64KuvfNS
>> a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3
>> dhTqD0XW8FAOLNtmP
>> a=ptime:20
>> a=nortpproxy:yes
>>
>>
>> I'm pretty new to the webrtc stuff, so any help is greatly appreciated!
>>
>> Thanks!
>>
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337 <+34%20669%2044%2083%2037>
>> ᐧ
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joel at gogii.net  Sun Jul  9 05:35:51 2017
From: joel at gogii.net (Joel Serrano)
Date: Sun, 09 Jul 2017 05:35:51 +0000
Subject: [Freeswitch-users] sip to webrtc - sdp invalid description
In-Reply-To: <CAMtXxQm643PE7EJr11Tf7rkvoXuURicDfkR3GeuROQUSPkiECQ@mail.gmail.com>
References: <CAFGRPVoQB8mH3a8JYbY3wP1L=L3x-zdZF-kK1o+ocW5GzSYt-A@mail.gmail.com>
 <CAFGRPVrNFQ6rKYVehMX1bopaUqKqT0NQeDQkGigCGC5e5h7FUg@mail.gmail.com>
 <CAMtXxQm643PE7EJr11Tf7rkvoXuURicDfkR3GeuROQUSPkiECQ@mail.gmail.com>
Message-ID: <CAMtXxQkB+cvj_HUF2p_ud8orbvTfPB-nL_=4se86WRoENkbc9A@mail.gmail.com>

Nevermind, I missed the part that FS is in the middle. I'm going to test
that and see what I find.

On Sat, Jul 8, 2017 at 19:56 Joel Serrano <joel at gogii.net> wrote:

> Can it be that Zoiper is using SRTP-3DES instead of SRTP-DTLS?
>
> WebRTC requires SRTP-DTLS.
>
>
>
> On Fri, Jul 7, 2017 at 9:46 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Hello guys,
>>
>> Any help on this?
>>
>> Thanks
>>
>> David
>> ᐧ
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>
>> On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello guys,
>>>
>>> I have this setup:
>>>
>>> Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
>>>
>>> Whenever webrtc clients call each other, calls are ok.
>>> But when the zoiper (regular sip/tcp) calls, the browsers complaint
>>> about:
>>>
>>> "no ice-ufrag" (firefox) or
>>> "Failed to set remote offer sdp: Called with SDP without DTLS
>>> fingerprint." (Chrome).
>>>
>>> I am setting the sdp in freeswitch as:
>>>
>>>       <action application="set" data="proxy_media=false"/>
>>>       <action application="set" data="bypass_media=false"/>
>>>       <action application="export" data="zrtp_secure_media=false"/>
>>>       <action application="export" data="nolocal:rtp_secure_media=true"/>
>>>
>>> and the actual sdp is:
>>>
>>> v=0
>>> o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4
>>> s=FreeSWITCH
>>> c=IN IP4 1.2.3.4
>>> t=0 0
>>> m=audio 40954 RTP/SAVP 8 101 13
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=rtpmap:13 CN/8000
>>> a=crypto:1 AEAD_AES_256_GCM_8
>>> inline:9XidL0Z5VYb0L5CegRZaYVrVfjA0ImWgu5WyLK0vtg60RTk6Koe8c0sRkjU
>>> a=crypto:2 AEAD_AES_128_GCM_8
>>> inline:fLKE1lxhRoVw+D5NVoKFFw06I0Xok/9KbRystQ
>>> a=crypto:3 AES_CM_256_HMAC_SHA1_80
>>> inline:w/yDN+ETPuCOiOIxjFLRjbFbDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg
>>> a=crypto:4 AES_CM_192_HMAC_SHA1_80
>>> inline:evU8MzAtiSHwKb95s4V9IAMpmok06kW9ZGDgH3/Lc3ZytVn2SR4
>>> a=crypto:5 AES_CM_128_HMAC_SHA1_80
>>> inline:KKDkT0DssohSeKFsX6tbixRhwYdiIhE6r3u5CCVA
>>> a=crypto:6 AES_CM_256_HMAC_SHA1_32
>>> inline:D3JgGuOlIxXfHGdqf7lKWqNDAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w
>>> a=crypto:7 AES_CM_192_HMAC_SHA1_32
>>> inline:c12ebWWzZ1cqZN0v5C5uYzdvtfnw6AARU3+jGA0WzTSDlDd20vI
>>> a=crypto:8 AES_CM_128_HMAC_SHA1_32
>>> inline:nPZku7S6hlz2OPAff8T9I8sNFzuZziNg64KuvfNS
>>> a=crypto:9 AES_CM_128_NULL_AUTH
>>> inline:9MNdj7xaingoGY14NUg8iS3dhTqD0XW8FAOLNtmP
>>> a=ptime:20
>>> a=nortpproxy:yes
>>>
>>>
>>> I'm pretty new to the webrtc stuff, so any help is greatly appreciated!
>>>
>>> Thanks!
>>>
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>> ᐧ
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From infos at madovsky.org  Sun Jul  9 20:21:19 2017
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 Jul 2017 13:21:19 -0700
Subject: [Freeswitch-users] mod_odbc_query
Message-ID: <0eb25e20-4155-4e2d-1b59-6b2e55a18eaa@madovsky.org>

Hi Folks,

is it a normal behavior that mod_odbc_query

accept only odbc-dsn like

|<||param| |name||=||"odbc-dsn"| 
|value||=||"freeswitch:freeswitch:secret"||/>|

|and not|


|<||param| |name||=||"odbc-dsn"| |value||=||"pgsql://xxxxxx"||/>|

|
|

|?|

|
|

|thanks|

|
|

|Franck
|

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From joel at gogii.net  Sun Jul  9 23:44:30 2017
From: joel at gogii.net (Joel Serrano)
Date: Sun, 9 Jul 2017 16:44:30 -0700
Subject: [Freeswitch-users] mod_odbc_query
In-Reply-To: <0eb25e20-4155-4e2d-1b59-6b2e55a18eaa@madovsky.org>
References: <0eb25e20-4155-4e2d-1b59-6b2e55a18eaa@madovsky.org>
Message-ID: <CAMtXxQkVeawNx1t25x=kfCvKLQVH3X9T+cJjHdJFjXQ+oxAzjA@mail.gmail.com>

Check: https://freeswitch.org/confluence/display/FREESWITCH/ODBC+DSN

I think you have an error in the format used.

Example for MySQL:

<param name="core-db-dsn"
value="odbc://DRIVER=mysql;SERVER=a.b.c.d;UID=username;PWD=secretpassword;DATABASE=freeswitch;OPTION=67108864">

Most likely the "value" for your odbc-dsn has to follow that format...


On Sun, Jul 9, 2017 at 1:21 PM, Madovsky <infos at madovsky.org> wrote:

> Hi Folks,
>
> is it a normal behavior that mod_odbc_query
>
> accept only odbc-dsn like
>
> <param name="odbc-dsn" value="freeswitch:freeswitch:secret"/>
>
> and not
>
>
> <param name="odbc-dsn" value="pgsql://xxxxxx"/>
>
>
> ?
>
>
> thanks
>
>
> Franck
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Jul 10 01:00:17 2017
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 10 Jul 2017 02:00:17 +0100
Subject: [Freeswitch-users] mod_odbc_query
In-Reply-To: <0eb25e20-4155-4e2d-1b59-6b2e55a18eaa@madovsky.org>
References: <0eb25e20-4155-4e2d-1b59-6b2e55a18eaa@madovsky.org>
Message-ID: <CAFiqYumnNsY49dCmxH8WTX9LWjEw_5rF42T_jxfM7c70EEXnrQ@mail.gmail.com>

Looking at the code, yes it's been written to only support a ODBC DSN of
that format.



On 9 July 2017 at 21:21, Madovsky <infos at madovsky.org> wrote:

> Hi Folks,
>
> is it a normal behavior that mod_odbc_query
>
> accept only odbc-dsn like
>
> <param name="odbc-dsn" value="freeswitch:freeswitch:secret"/>
>
> and not
>
>
> <param name="odbc-dsn" value="pgsql://xxxxxx"/>
>
>
> ?
>
>
> thanks
>
>
> Franck
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From devang.nathwani31589 at gmail.com  Mon Jul 10 12:47:30 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Mon, 10 Jul 2017 18:17:30 +0530
Subject: [Freeswitch-users] call limit between multiple freewitchs using
	mod_db
Message-ID: <CAGOm0P+yc=jdQnFzsKWtoaXS-Yw3uE+kuqh7a0nAo1NEYg2FMg@mail.gmail.com>

I have two freeswitch servers load balancing is happening through opensips
at front

I am using mod_db of freeswitch to have centralized database for account
based call limit over both freeswitch

I have configured db.conf.xml settings with
<param name="odbc-dsn" value="freeswitch:umn:pwd"/>

However, when i set the limit per account freeswitch wont restrict call
attempt as it should be.

What could be the possible issue?
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From dig1234 at gmail.com  Mon Jul 10 15:43:38 2017
From: dig1234 at gmail.com (Daniel Greenwald)
Date: Mon, 10 Jul 2017 11:43:38 -0400
Subject: [Freeswitch-users] uuid_dump in dialplan
Message-ID: <CAOsHgtsH8Ou06V7=7nj-StbSbq+mm197EwyAPubegxDh2sYEbQ@mail.gmail.com>

Hello,
Looking for a way to dump all variables to the log at the end of the
dialplan? I know the info application can query specific variables but is
there a way to perform a full uuid_dump on the channel and send it to the
log just before or after hangup?
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From brian at freeswitch.org  Mon Jul 10 16:03:09 2017
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Jul 2017 11:03:09 -0500
Subject: [Freeswitch-users] uuid_dump in dialplan
In-Reply-To: <CAOsHgtsH8Ou06V7=7nj-StbSbq+mm197EwyAPubegxDh2sYEbQ@mail.gmail.com>
References: <CAOsHgtsH8Ou06V7=7nj-StbSbq+mm197EwyAPubegxDh2sYEbQ@mail.gmail.com>
Message-ID: <CAEJMVkCBioOLL7ytj9+9=QmYeqzRujiCu=MWs8nU1vRwtXdchg@mail.gmail.com>

info app.

/b


On Mon, Jul 10, 2017 at 10:43 AM, Daniel Greenwald <dig1234 at gmail.com>
wrote:

> Hello,
> Looking for a way to dump all variables to the log at the end of the
> dialplan? I know the info application can query specific variables but is
> there a way to perform a full uuid_dump on the channel and send it to the
> log just before or after hangup?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From devang.nathwani31589 at gmail.com  Mon Jul 10 17:31:58 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Mon, 10 Jul 2017 23:01:58 +0530
Subject: [Freeswitch-users] call limit between multiple freewitchs using
	mod_db
In-Reply-To: <CAGOm0P+yc=jdQnFzsKWtoaXS-Yw3uE+kuqh7a0nAo1NEYg2FMg@mail.gmail.com>
References: <CAGOm0P+yc=jdQnFzsKWtoaXS-Yw3uE+kuqh7a0nAo1NEYg2FMg@mail.gmail.com>
Message-ID: <CAGOm0PKbcriGiA5yFgJ2hRLHN9-sR9ZaOwKPoL7B679QtT8iNQ@mail.gmail.com>

https://wiki.freeswitch.org/wiki/Sofia.conf.xml#send-display-update
will this be helpful?

On Mon, Jul 10, 2017 at 6:17 PM, devang nathwani <
devang.nathwani31589 at gmail.com> wrote:

> I have two freeswitch servers load balancing is happening through opensips
> at front
>
> I am using mod_db of freeswitch to have centralized database for account
> based call limit over both freeswitch
>
> I have configured db.conf.xml settings with
> <param name="odbc-dsn" value="freeswitch:umn:pwd"/>
>
> However, when i set the limit per account freeswitch wont restrict call
> attempt as it should be.
>
> What could be the possible issue?
>
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From steveayre at gmail.com  Mon Jul 10 23:20:39 2017
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 11 Jul 2017 00:20:39 +0100
Subject: [Freeswitch-users] call limit between multiple freewitchs using
	mod_db
In-Reply-To: <CAGOm0P+yc=jdQnFzsKWtoaXS-Yw3uE+kuqh7a0nAo1NEYg2FMg@mail.gmail.com>
References: <CAGOm0P+yc=jdQnFzsKWtoaXS-Yw3uE+kuqh7a0nAo1NEYg2FMg@mail.gmail.com>
Message-ID: <CAFiqYuk-qo17Xi1R3x2WL1BCQGnC5i_yZkFivEnisgyceOQ5Xw@mail.gmail.com>

How are you invoking limit?

On 10 July 2017 at 13:47, devang nathwani <devang.nathwani31589 at gmail.com>
wrote:

> I have two freeswitch servers load balancing is happening through opensips
> at front
>
> I am using mod_db of freeswitch to have centralized database for account
> based call limit over both freeswitch
>
> I have configured db.conf.xml settings with
> <param name="odbc-dsn" value="freeswitch:umn:pwd"/>
>
> However, when i set the limit per account freeswitch wont restrict call
> attempt as it should be.
>
> What could be the possible 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
>
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From udy786 at gmail.com  Tue Jul 11 10:58:08 2017
From: udy786 at gmail.com (Uday kumar)
Date: Tue, 11 Jul 2017 16:28:08 +0530
Subject: [Freeswitch-users] PHP code in voicemail.tpl
Message-ID: <CAOfqg5bA9ekgJvyJcxgK4kKyXWHb9k+2PRqgPaY4jPH8KuiQrg@mail.gmail.com>

Hello All,

I am trying to use Google Transcribing to Transcribing Voicemail. Is
possible to add PHP code in voicemail.tpl or run php file like
voicemail.php on place of voicemail.tpl?

I tried to add PHP code in tpl file but didn't worked also tried with
voicemail.php after changing in voicemail.conf.xml but didn't worked. I am
using sendmail to send voicemail on email. Email going when extension
getting voicemail with recording. If I can run PHP file or php code in
template then i will implement google API and add response in email body.

I am using CURL for dialplan, directory and configuration. I
tried mailer-app=/usr/bin/php /usr/local/www/mailer_app.php but email not
working because not getting To email id. Also is possible to send
${voicemail_file_path}
to php mailer?

cat /tmp/voicemailtoemail.txt

To:
From: FreeSWITCH mod_voicemail <0001 at XX.XX.XX.XX>
Subject: Voicemail from +919377579349 +919377579349


Mailer Error: You must provide at least one recipient email address.

Please advice.

-- 
Thanks & Regard
Uday
Mobile:- +91-9377579349
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From david.villasmil.work at gmail.com  Tue Jul 11 11:58:03 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 11 Jul 2017 13:58:03 +0200
Subject: [Freeswitch-users] sip to webrtc - sdp invalid description
In-Reply-To: <CAMtXxQkB+cvj_HUF2p_ud8orbvTfPB-nL_=4se86WRoENkbc9A@mail.gmail.com>
References: <CAFGRPVoQB8mH3a8JYbY3wP1L=L3x-zdZF-kK1o+ocW5GzSYt-A@mail.gmail.com>
 <CAFGRPVrNFQ6rKYVehMX1bopaUqKqT0NQeDQkGigCGC5e5h7FUg@mail.gmail.com>
 <CAMtXxQm643PE7EJr11Tf7rkvoXuURicDfkR3GeuROQUSPkiECQ@mail.gmail.com>
 <CAMtXxQkB+cvj_HUF2p_ud8orbvTfPB-nL_=4se86WRoENkbc9A@mail.gmail.com>
Message-ID: <CAFGRPVpA522oxcn+hJBha6vms_2i8xUXTvz5gnNuotHEdTX5JA@mail.gmail.com>

many thanks!
ᐧ

Regards,

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

On Sun, Jul 9, 2017 at 7:35 AM, Joel Serrano <joel at gogii.net> wrote:

> Nevermind, I missed the part that FS is in the middle. I'm going to test
> that and see what I find.
>
> On Sat, Jul 8, 2017 at 19:56 Joel Serrano <joel at gogii.net> wrote:
>
>> Can it be that Zoiper is using SRTP-3DES instead of SRTP-DTLS?
>>
>> WebRTC requires SRTP-DTLS.
>>
>>
>>
>> On Fri, Jul 7, 2017 at 9:46 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Hello guys,
>>>
>>> Any help on this?
>>>
>>> Thanks
>>>
>>> David
>>> ᐧ
>>>
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>>
>>> On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Hello guys,
>>>>
>>>> I have this setup:
>>>>
>>>> Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
>>>>
>>>> Whenever webrtc clients call each other, calls are ok.
>>>> But when the zoiper (regular sip/tcp) calls, the browsers complaint
>>>> about:
>>>>
>>>> "no ice-ufrag" (firefox) or
>>>> "Failed to set remote offer sdp: Called with SDP without DTLS
>>>> fingerprint." (Chrome).
>>>>
>>>> I am setting the sdp in freeswitch as:
>>>>
>>>>       <action application="set" data="proxy_media=false"/>
>>>>       <action application="set" data="bypass_media=false"/>
>>>>       <action application="export" data="zrtp_secure_media=false"/>
>>>>       <action application="export" data="nolocal:rtp_secure_
>>>> media=true"/>
>>>>
>>>> and the actual sdp is:
>>>>
>>>> v=0
>>>> o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4
>>>> s=FreeSWITCH
>>>> c=IN IP4 1.2.3.4
>>>> t=0 0
>>>> m=audio 40954 RTP/SAVP 8 101 13
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=rtpmap:13 CN/8000
>>>> a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVrVfjA0Im
>>>> Wgu5WyLK0vtg60RTk6Koe8c0sRkjU
>>>> a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+
>>>> D5NVoKFFw06I0Xok/9KbRystQ
>>>> a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+
>>>> ETPuCOiOIxjFLRjbFbDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg
>>>> a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:
>>>> evU8MzAtiSHwKb95s4V9IAMpmok06kW9ZGDgH3/Lc3ZytVn2SR4
>>>> a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:
>>>> KKDkT0DssohSeKFsX6tbixRhwYdiIhE6r3u5CCVA
>>>> a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:
>>>> D3JgGuOlIxXfHGdqf7lKWqNDAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w
>>>> a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:
>>>> c12ebWWzZ1cqZN0v5C5uYzdvtfnw6AARU3+jGA0WzTSDlDd20vI
>>>> a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:
>>>> nPZku7S6hlz2OPAff8T9I8sNFzuZziNg64KuvfNS
>>>> a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3dhTqD0X
>>>> W8FAOLNtmP
>>>> a=ptime:20
>>>> a=nortpproxy:yes
>>>>
>>>>
>>>> I'm pretty new to the webrtc stuff, so any help is greatly appreciated!
>>>>
>>>> Thanks!
>>>>
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.work at gmail.com
>>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>>> ᐧ
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vma at vallimamod.org  Tue Jul 11 12:26:11 2017
From: vma at vallimamod.org (Vallimamod Abdullah)
Date: Tue, 11 Jul 2017 14:26:11 +0200
Subject: [Freeswitch-users] uuid_dump in dialplan
In-Reply-To: <CAOsHgtsH8Ou06V7=7nj-StbSbq+mm197EwyAPubegxDh2sYEbQ@mail.gmail.com>
References: <CAOsHgtsH8Ou06V7=7nj-StbSbq+mm197EwyAPubegxDh2sYEbQ@mail.gmail.com>
Message-ID: <A37FA9E5-204A-4745-8E3E-AB2A1C77A5E5@vallimamod.org>

Hi,

You can use mod_cdr_csv with debug=true. It will do the same thing as if you had called the info app after hangup.

Best Regards,
-- 
Vallimamod Abdullah
SIP Solutions
vma at sipsolutions.fr
tel: +33 6 62 60 68 97
.

> On 10 Jul 2017, at 17:43, Daniel Greenwald <dig1234 at gmail.com> wrote:
> 
> Hello,
> Looking for a way to dump all variables to the log at the end of the dialplan? I know the info application can query specific variables but is there a way to perform a full uuid_dump on the channel and send it to the log just before or after hangup? 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From devang.nathwani31589 at gmail.com  Tue Jul 11 13:57:32 2017
From: devang.nathwani31589 at gmail.com (devang nathwani)
Date: Tue, 11 Jul 2017 19:27:32 +0530
Subject: [Freeswitch-users] Got 603 Decline from freeswitch
Message-ID: <CAGOm0PJVrqVHEEJOvq_bG1k6P4CNCFYJcXDy2t_nUKpZo=Ed+Q@mail.gmail.com>

Hello,

Below is the fs_cli log

Asterisk
192.168.201.227

Opensips
192.168.201.245

Freeswitch
192.168.201.248

Here is the call flow,
Asterisk -> Opensips -> Freeswitch

https://pastebin.freeswitch.org/view/2cd0875b

What could be the issue?
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From mike at jerris.com  Tue Jul 11 14:16:09 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 Jul 2017 14:16:09 +0000
Subject: [Freeswitch-users] Got 603 Decline from freeswitch
In-Reply-To: <CAGOm0PJVrqVHEEJOvq_bG1k6P4CNCFYJcXDy2t_nUKpZo=Ed+Q@mail.gmail.com>
References: <CAGOm0PJVrqVHEEJOvq_bG1k6P4CNCFYJcXDy2t_nUKpZo=Ed+Q@mail.gmail.com>
Message-ID: <CAEwTsAhFV7SprramT-FOqqZtg9kABqL4fGP44_C7nEXmktmMUA@mail.gmail.com>

You do this in your dialplan


   1. ae48a366-6628-11e7-b187-0f41e0ba0a22 EXECUTE sofia/default/01617975151
   @192.168.201.227 hangup(CALL_REJECTED)


On Tue, Jul 11, 2017 at 9:58 AM devang nathwani <
devang.nathwani31589 at gmail.com> wrote:

> Hello,
>
> Below is the fs_cli log
>
> Asterisk
> 192.168.201.227
>
> Opensips
> 192.168.201.245
>
> Freeswitch
> 192.168.201.248
>
> Here is the call flow,
> Asterisk -> Opensips -> Freeswitch
>
> https://pastebin.freeswitch.org/view/2cd0875b
>
> What could be the 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
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From joelists at tm.net.uk  Tue Jul 11 19:32:09 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Tue, 11 Jul 2017 20:32:09 +0100
Subject: [Freeswitch-users] Set b-leg CallerID using from in a 300 Multiple
	Choices reply
Message-ID: <B5365ACE-09C6-4821-9BBC-D42C2DE24EC3@tm.net.uk>

Hi Guys

I am trying to use the From field in a 300 Multiple Choices message to set the outbound CallerID in the re-directed b-leg

So call comes into FreeSwitch, gets sent out to a SIP Redirect server which replies with a 300 Multiple choices.
I need to pull the number from the From field of this 300 Response and use it to set the From field, raid & Paid fields in the re-directed invite.

Any help would be very much appreciated.

Regards

From mike at jerris.com  Tue Jul 11 19:39:40 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 Jul 2017 15:39:40 -0400
Subject: [Freeswitch-users] ClueCon Hotel is filling up! ACT NOW and
 save $300
In-Reply-To: <CAKbxfG8DqKRDYxqpWbneViPyJEOQdOAr6YAr98ZZjpZwwsyGfA@mail.gmail.com>
References: <CAKbxfG8DqKRDYxqpWbneViPyJEOQdOAr6YAr98ZZjpZwwsyGfA@mail.gmail.com>
Message-ID: <7736AD67-4464-428B-BD6E-1F03F566626C@jerris.com>

This code is still good.  If you’ve been thinking about coming, now is the time to sign up.  If you are not sure if you want to come, give us a call and we can talk to you about all the great speakers and content.

877-7-4-A-CLUE

> Use the code CCJulySM2017 and save $300
> Help support the FS Community and have fun in the process!
> 
> 
> 
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From keith at rhizomatica.org  Wed Jul 12 06:48:07 2017
From: keith at rhizomatica.org (Keith Whyte)
Date: Wed, 12 Jul 2017 08:48:07 +0200
Subject: [Freeswitch-users] How to add path information to mod_sms message
Message-ID: <735c6768-f78c-4391-95ef-fb70d4d17520@rhizomatica.org>

Hi!

I'm registering clients to freeswitch via a Kamailio proxy.

Kamailio sends a Path: header to freeswitch, and calls are working fine,
but I'm struggling to get chats working.

I'm stuck trying to figure out how to send the path information back to
Kamailio in a MESSAGE with mod_sms.

Adding a "path" header to the event is ignored.

I have a python chatplan, I have found that if I do something like:

event = ESL.ESLevent("CUSTOM", "SMS::SEND_MESSAGE") 

event.addHeader("to", "extention at kamailio_server;received=sip:ip_addr:port")

Then Freeswitch will send the MESSAGE to the kamailio server with a To line:

To: <sip:extention at kamailio_server>;received=sip:ip_addr:port

I guess I could go about having Kamailio parse this, but is this the way
to go? It seems messy.

Any ideas?

Thanks!






From igorolhovskiy at gmail.com  Wed Jul 12 10:49:49 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Wed, 12 Jul 2017 13:49:49 +0300
Subject: [Freeswitch-users] NDLB-force-rport question
Message-ID: <96e620c4-6c5d-4a76-a096-f87769b8263e@Spark>

Hi!

I’m having strange issues on registering phones behind NAT on FreeSwitch.

Schema is Phone (port 5063)  -> Router (NAT) (port 1023) -> Freeswitch

Phone registering from port 5063. NAT is changing this port to 1023. Freeswitch receiving register message from port 1023, but still replying to port, that listed in Contact Header. So, Router can’t route this request back correctly.

NDLB-force-rport and aggressive-nat-detection is enabled on internal profile. What I got, this issue is when router changes port to 1024 or lower. If port is higher that 1024, than it’s working ok.
Also this problem with UDP only, with TCP all is ok.

Maybe I’m missing something in config?

Thanks.

Regards, Igor
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From infos at madovsky.org  Wed Jul 12 11:28:58 2017
From: infos at madovsky.org (Madovsky)
Date: Wed, 12 Jul 2017 04:28:58 -0700
Subject: [Freeswitch-users] compile FS with special lib dir
Message-ID: <e404b50d-aab4-38d7-01ea-319be7824b31@madovsky.org>

Hi,

is there any way to compile freeswitch with a different

library target path?

for example, I would like to compile mod_av

referring the libav libraries to a special folder (not /usr/local/lib)


thanks


Franck



From hunterj91 at hotmail.com  Wed Jul 12 12:04:10 2017
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Wed, 12 Jul 2017 12:04:10 +0000
Subject: [Freeswitch-users] Mod_Opus :Add fmtp maxplaybackrate=8000 in OPUS
 codec offer.Version 1.6.8-15-99de0ad~64bit
Message-ID: <DB4PR02MB033682EEF18D20C4AF0D1F39BDAF0@DB4PR02MB0336.eurprd02.prod.outlook.com>

Hi Guys,

We are currently testing OPUS and Im looking to lock down things to a maxplaybackrate of 8000HZ.

This works fine if FreeSWITCH is the receiver (Offer Answer), however if I then send an invite out from FreeSWITCH (bridge the same call)it doesnt seem to add the  fmtp maxplaybackrate=8000 paramater.

As when FreeSWITCH answers I see;

Media Attribute (a): fmtp:111 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; minptime=10

However when I send an Invite offer out I see;

 Media Attribute (a): rtpmap:111 opus/48000/2
 Media Attribute (a): fmtp:111 minptime=10;useinbandfec=1


The OPUS module document seems to suggest we can add this, and if so how can I achieve this, as I have my opus.conf set to;

<param name="use-vbr" value="1"/>
<param name="use-dtx" value="0"/>
<param name="complexity" value="10"/>
<param name="maxaveragebitrate" value="14400"/>
<param name="maxplaybackrate" value="8000"/>
<param name="packet-loss-percent" value="15"/>
<param name="keep-fec-enabled" value="1"/>
<param name="use-jb-lookahead" value="1"/>
<param name="advertise-useinbandfec" value="1"/>


And I also set the SIP profile codec list and absolute codec offer to OPUS at 8000h, and as I say works great when we answer an offer, but not when we originate.

Am I misconfiguring or is this not possible?

I know the RFC states the parameter "maxplaybackrate" is a unidirectional receive-only parameter that reflects limitations of the local receiver, however mod_opus documentation seems to suggest we can set it in the offer, so I should be able to add as;

maxplaybackrate=8000

Is this possible? I just want to ensure I get narrowband end to end.

Many thanks

Jon

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From david.villasmil.work at gmail.com  Wed Jul 12 12:10:21 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 12 Jul 2017 12:10:21 +0000
Subject: [Freeswitch-users] compile FS with special lib dir
In-Reply-To: <e404b50d-aab4-38d7-01ea-319be7824b31@madovsky.org>
References: <e404b50d-aab4-38d7-01ea-319be7824b31@madovsky.org>
Message-ID: <CAFGRPVpKxksVrW8Qk61No7ZFhnc5N0cgZ6JLhqabO-M5T92E7g@mail.gmail.com>

When you run configure --help you get a list of options
On Wed, Jul 12, 2017 at 1:29 PM Madovsky <infos at madovsky.org> wrote:

> Hi,
>
> is there any way to compile freeswitch with a different
>
> library target path?
>
> for example, I would like to compile mod_av
>
> referring the libav libraries to a special folder (not /usr/local/lib)
>
>
> thanks
>
>
> Franck
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From dragos.oancea at vonage.com  Wed Jul 12 12:14:41 2017
From: dragos.oancea at vonage.com (Oancea, Dragos)
Date: Wed, 12 Jul 2017 13:14:41 +0100
Subject: [Freeswitch-users] Mod_Opus :Add fmtp maxplaybackrate=8000 in
 OPUS codec offer.Version 1.6.8-15-99de0ad~64bit
In-Reply-To: <DB4PR02MB033682EEF18D20C4AF0D1F39BDAF0@DB4PR02MB0336.eurprd02.prod.outlook.com>
References: <DB4PR02MB033682EEF18D20C4AF0D1F39BDAF0@DB4PR02MB0336.eurprd02.prod.outlook.com>
Message-ID: <CAOsE5hjMDmhHbsVpsTk3T8n6FtnFkCy-p-t5+GLcW_vMfbSckg@mail.gmail.com>

Hi Jon,

If both sides follow the RFC , then you should be able to get narrowband
end to end.
How do you originate the call ?
Sounds like you originate a call from a device and then you bridge to the
callee, but you have inbound-late-negotiation on, so the OPUS config XML is
not loaded, hence fmtp cannot be changed according to config.

Regards,
Dragos


On Wed, Jul 12, 2017 at 1:04 PM, Jonathan Hunter <hunterj91 at hotmail.com>
wrote:

> Hi Guys,
>
> We are currently testing OPUS and Im looking to lock down things to a
> maxplaybackrate of 8000HZ.
>
> This works fine if FreeSWITCH is the receiver (Offer Answer), however if I
> then send an invite out from FreeSWITCH (bridge the same call)it doesnt
> seem to add the  fmtp maxplaybackrate=8000 paramater.
>
> As when FreeSWITCH answers I see;
>
> Media Attribute (a): fmtp:111 useinbandfec=1; maxaveragebitrate=14400;
> maxplaybackrate=8000; minptime=10
>
> However when I send an Invite offer out I see;
>
>  Media Attribute (a): rtpmap:111 opus/48000/2
>  Media Attribute (a): fmtp:111 minptime=10;useinbandfec=1
>
>
> The OPUS module document seems to suggest we can add this, and if so how
> can I achieve this, as I have my opus.conf set to;
>
> <param name="use-vbr" value="1"/>
> <param name="use-dtx" value="0"/>
> <param name="complexity" value="10"/>
> <param name="maxaveragebitrate" value="14400"/>
> <param name="maxplaybackrate" value="8000"/>
> <param name="packet-loss-percent" value="15"/>
> <param name="keep-fec-enabled" value="1"/>
> <param name="use-jb-lookahead" value="1"/>
> <param name="advertise-useinbandfec" value="1"/>
>
>
> And I also set the SIP profile codec list and absolute codec offer to
> OPUS at 8000h, and as I say works great when we answer an offer, but not
> when we originate.
>
> Am I misconfiguring or is this not possible?
>
> I know the RFC states the parameter "maxplaybackrate" is a unidirectional
> receive-only parameter that reflects limitations of the local receiver,
> however mod_opus documentation seems to suggest we can set it in the offer,
> so I should be able to add as;
>
> maxplaybackrate=8000
>
> Is this possible? I just want to ensure I get narrowband end to end.
>
> Many thanks
>
> Jon
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From hunterj91 at hotmail.com  Wed Jul 12 14:33:02 2017
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Wed, 12 Jul 2017 14:33:02 +0000
Subject: [Freeswitch-users] Mod_Opus :Add fmtp maxplaybackrate=8000 in
 OPUS codec offer.Version 1.6.8-15-99de0ad~64bit
In-Reply-To: <CAPQovySPLcDkfD=JpaE8Hjp2nH+zPARpvgAVayKHDMg4Gfp42Q@mail.gmail.com>
References: <DB4PR02MB033682EEF18D20C4AF0D1F39BDAF0@DB4PR02MB0336.eurprd02.prod.outlook.com>
 <CAOsE5hjMDmhHbsVpsTk3T8n6FtnFkCy-p-t5+GLcW_vMfbSckg@mail.gmail.com>,
 <CAPQovySPLcDkfD=JpaE8Hjp2nH+zPARpvgAVayKHDMg4Gfp42Q@mail.gmail.com>
Message-ID: <DB4PR02MB0336D982A0A2C4AEF6B6072EBDAF0@DB4PR02MB0336.eurprd02.prod.outlook.com>

Hi Dragos,


Thank you for the fast reply!


You are correct in terms of call flow.


I am originating from a WebRTC client on Janus, which then sends a SIP invite into FreeSWITCH , I then reply to that to set it to 8K, narrowband, this works perfectly ! 😊


However I then bridge the call out to a carrier, and I want to offer in the SDP the maxplaybackrate=8000, however I cant seem to do this, and this is with and without late-negotiation enabled.


Am I missing something or do you have any debug suggestions for me please?


Many  thanks


Jon

---------- Forwarded message ----------
From: Oancea, Dragos <dragos.oancea at vonage.com<mailto:dragos.oancea at vonage.com>>
Date: Wed, Jul 12, 2017 at 1:14 PM
Subject: Re: [Freeswitch-users] Mod_Opus :Add fmtp maxplaybackrate=8000 in OPUS codec offer.Version 1.6.8-15-99de0ad~64bit
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>


Hi Jon,

If both sides follow the RFC , then you should be able to get narrowband end to end.
How do you originate the call ?
Sounds like you originate a call from a device and then you bridge to the callee, but you have inbound-late-negotiation on, so the OPUS config XML is not loaded, hence fmtp cannot be changed according to config.

Regards,
Dragos


On Wed, Jul 12, 2017 at 1:04 PM, Jonathan Hunter <hunterj91 at hotmail.com<mailto:hunterj91 at hotmail.com>> wrote:

Hi Guys,

We are currently testing OPUS and Im looking to lock down things to a maxplaybackrate of 8000HZ.

This works fine if FreeSWITCH is the receiver (Offer Answer), however if I then send an invite out from FreeSWITCH (bridge the same call)it doesnt seem to add the  fmtp maxplaybackrate=8000 paramater.

As when FreeSWITCH answers I see;

Media Attribute (a): fmtp:111 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; minptime=10

However when I send an Invite offer out I see;

 Media Attribute (a): rtpmap:111 opus/48000/2
 Media Attribute (a): fmtp:111 minptime=10;useinbandfec=1


The OPUS module document seems to suggest we can add this, and if so how can I achieve this, as I have my opus.conf set to;

<param name="use-vbr" value="1"/>
<param name="use-dtx" value="0"/>
<param name="complexity" value="10"/>
<param name="maxaveragebitrate" value="14400"/>
<param name="maxplaybackrate" value="8000"/>
<param name="packet-loss-percent" value="15"/>
<param name="keep-fec-enabled" value="1"/>
<param name="use-jb-lookahead" value="1"/>
<param name="advertise-useinbandfec" value="1"/>


And I also set the SIP profile codec list and absolute codec offer to OPUS at 8000h, and as I say works great when we answer an offer, but not when we originate.

Am I misconfiguring or is this not possible?

I know the RFC states the parameter "maxplaybackrate" is a unidirectional receive-only parameter that reflects limitations of the local receiver, however mod_opus documentation seems to suggest we can set it in the offer, so I should be able to add as;

maxplaybackrate=8000

Is this possible? I just want to ensure I get narrowband end to end.

Many thanks

Jon


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

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

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


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

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

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



--
Jonathan Hunter
Technical Director /Telephony Developer
M:(+44) 7917 190 438
Email:jhunter at voxboxcoms.co.uk<mailto:Email%3Ajhunter at voxboxcoms.co.uk>

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From gmaruzz at gmail.com  Wed Jul 12 14:37:05 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 12 Jul 2017 16:37:05 +0200
Subject: [Freeswitch-users] NDLB-force-rport question
In-Reply-To: <96e620c4-6c5d-4a76-a096-f87769b8263e@Spark>
References: <96e620c4-6c5d-4a76-a096-f87769b8263e@Spark>
Message-ID: <CALXCt0p8k-cEmWHfrTE0L1FcSDTjEhYEQ6tCCQoKy-Ktzcpayw@mail.gmail.com>

What FS version and platform?

What phones?

Maybe is a problem with carrier? (Udp only...)

Have you tried the default configuration, the original one, without
altering or adding things?


sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT


On Jul 12, 2017 12:54 PM, "Igor Olhovskiy" <igorolhovskiy at gmail.com> wrote:

Hi!

I’m having strange issues on registering phones behind NAT on FreeSwitch.

Schema is Phone (port 5063)  -> Router (NAT) (port 1023) -> Freeswitch

Phone registering from port 5063. NAT is changing this port to 1023.
Freeswitch receiving register message from port 1023, but still replying to
port, that listed in Contact Header. So, Router can’t route this request
back correctly.

NDLB-force-rport and aggressive-nat-detection is enabled on internal
profile. What I got, this issue is when router changes port to 1024 or
lower. If port is higher that 1024, than it’s working ok.
Also this problem with UDP only, with TCP all is ok.

Maybe I’m missing something in config?

Thanks.

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
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From dragos.oancea at vonage.com  Wed Jul 12 15:14:36 2017
From: dragos.oancea at vonage.com (Oancea, Dragos)
Date: Wed, 12 Jul 2017 16:14:36 +0100
Subject: [Freeswitch-users] Mod_Opus :Add fmtp maxplaybackrate=8000 in
 OPUS codec offer.Version 1.6.8-15-99de0ad~64bit
In-Reply-To: <DB4PR02MB0336D982A0A2C4AEF6B6072EBDAF0@DB4PR02MB0336.eurprd02.prod.outlook.com>
References: <DB4PR02MB033682EEF18D20C4AF0D1F39BDAF0@DB4PR02MB0336.eurprd02.prod.outlook.com>
 <CAOsE5hjMDmhHbsVpsTk3T8n6FtnFkCy-p-t5+GLcW_vMfbSckg@mail.gmail.com>
 <CAPQovySPLcDkfD=JpaE8Hjp2nH+zPARpvgAVayKHDMg4Gfp42Q@mail.gmail.com>
 <DB4PR02MB0336D982A0A2C4AEF6B6072EBDAF0@DB4PR02MB0336.eurprd02.prod.outlook.com>
Message-ID: <CAOsE5hgGh3rzO2c-dfmzO5T3tkgybE6AtU9GJQqHhSnF=RYryg@mail.gmail.com>

Hi Jon ,

So the fmtp that goes to the carrier is the same as the one sent by Janus ?
Try : <param name="inbound-codec-negotiation" value="greedy"/> on the SIP
profile. I assume you have this set to "generous".
Also please use FS master or the latest release.
If it still does not work please open a FS jira.
You see the fmtp being changed with the XML values when you originate the
call through fs_cli, right ?

Cheers,
Dragos


On Wed, Jul 12, 2017 at 3:33 PM, Jonathan Hunter <hunterj91 at hotmail.com>
wrote:

> Hi Dragos,
>
>
> Thank you for the fast reply!
>
>
> You are correct in terms of call flow.
>
>
> I am originating from a WebRTC client on Janus, which then sends a SIP
> invite into FreeSWITCH , I then reply to that to set it to 8K, narrowband,
> this works perfectly ! 😊
>
>
> However I then bridge the call out to a carrier, and I want to offer in
> the SDP the maxplaybackrate=8000, however I cant seem to do this, and this
> is with and without late-negotiation enabled.
>
>
> Am I missing something or do you have any debug suggestions for me please?
>
>
> Many  thanks
>
>
> Jon
>
> ---------- Forwarded message ----------
> From: Oancea, Dragos <dragos.oancea at vonage.com>
> Date: Wed, Jul 12, 2017 at 1:14 PM
> Subject: Re: [Freeswitch-users] Mod_Opus :Add fmtp maxplaybackrate=8000 in
> OPUS codec offer.Version 1.6.8-15-99de0ad~64bit
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> Hi Jon,
>
> If both sides follow the RFC , then you should be able to get narrowband
> end to end.
> How do you originate the call ?
> Sounds like you originate a call from a device and then you bridge to the
> callee, but you have inbound-late-negotiation on, so the OPUS config XML is
> not loaded, hence fmtp cannot be changed according to config.
>
> Regards,
> Dragos
>
>
> On Wed, Jul 12, 2017 at 1:04 PM, Jonathan Hunter <hunterj91 at hotmail.com>
> wrote:
>
>> Hi Guys,
>>
>> We are currently testing OPUS and Im looking to lock down things to a
>> maxplaybackrate of 8000HZ.
>>
>> This works fine if FreeSWITCH is the receiver (Offer Answer), however if
>> I then send an invite out from FreeSWITCH (bridge the same call)it doesnt
>> seem to add the  fmtp maxplaybackrate=8000 paramater.
>>
>> As when FreeSWITCH answers I see;
>>
>> Media Attribute (a): fmtp:111 useinbandfec=1; maxaveragebitrate=14400;
>> maxplaybackrate=8000; minptime=10
>>
>> However when I send an Invite offer out I see;
>>
>>  Media Attribute (a): rtpmap:111 opus/48000/2
>>  Media Attribute (a): fmtp:111 minptime=10;useinbandfec=1
>>
>>
>> The OPUS module document seems to suggest we can add this, and if so how
>> can I achieve this, as I have my opus.conf set to;
>>
>> <param name="use-vbr" value="1"/>
>> <param name="use-dtx" value="0"/>
>> <param name="complexity" value="10"/>
>> <param name="maxaveragebitrate" value="14400"/>
>> <param name="maxplaybackrate" value="8000"/>
>> <param name="packet-loss-percent" value="15"/>
>> <param name="keep-fec-enabled" value="1"/>
>> <param name="use-jb-lookahead" value="1"/>
>> <param name="advertise-useinbandfec" value="1"/>
>>
>>
>> And I also set the SIP profile codec list and absolute codec offer to
>> OPUS at 8000h, and as I say works great when we answer an offer, but not
>> when we originate.
>>
>> Am I misconfiguring or is this not possible?
>>
>> I know the RFC states the parameter "maxplaybackrate" is a unidirectional
>> receive-only parameter that reflects limitations of the local receiver,
>> however mod_opus documentation seems to suggest we can set it in the offer,
>> so I should be able to add as;
>>
>> maxplaybackrate=8000
>>
>> Is this possible? I just want to ensure I get narrowband end to end.
>>
>> Many thanks
>>
>> Jon
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> --
> Jonathan Hunter
> Technical Director /Telephony Developer
> M:(+44) 7917 190 438 <+44%207917%20190438>
> Email:jhunter at voxboxcoms.co.uk
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gregor at infomedia.si  Wed Jul 12 17:40:19 2017
From: gregor at infomedia.si (Gregor Nanger)
Date: Wed, 12 Jul 2017 17:40:19 +0000
Subject: [Freeswitch-users] Bridge - user not registered - verto
Message-ID: <CAC7ttFEHL_kosYLyKweyKgry8AK7ErPJ7jPQG8tOBCMwAZ1FLA@mail.gmail.com>

When I make bridge to user/extension to verto user, I get user is not
registered. Call is transfered, but it looks like freeswitch tries to
transfer to verto and sip endpoint. And since sip endpoint is not
registered, I get User_not-registered. Everything works, but I get 2
calllogs because of this.

I also try to modify dial-string in user params, but same behavior

Is this by design?

Best regards, Gregor
-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485
• Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia
• www.infomedia.si
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From igorolhovskiy at gmail.com  Wed Jul 12 18:16:44 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Wed, 12 Jul 2017 21:16:44 +0300
Subject: [Freeswitch-users] NDLB-force-rport question
In-Reply-To: <CALXCt0p8k-cEmWHfrTE0L1FcSDTjEhYEQ6tCCQoKy-Ktzcpayw@mail.gmail.com>
References: <96e620c4-6c5d-4a76-a096-f87769b8263e@Spark>
 <CALXCt0p8k-cEmWHfrTE0L1FcSDTjEhYEQ6tCCQoKy-Ktzcpayw@mail.gmail.com>
Message-ID: <a2bab245-88fe-4e1a-a3ca-fed4a58d7870@Spark>

Freeswitch 1.6.18, Debian 8x64, tried with default config.
Phone - Yealink, Router - some Zyxel, I believe Kinetic.

Regards, Igor

On 12 июля 2017 г., 17:37 +0300, Giovanni Maruzzelli <gmaruzz at gmail.com>, wrote:
> What FS version and platform?
>
> What phones?
>
> Maybe is a problem with carrier? (Udp only...)
>
> Have you tried the default configuration, the original one, without altering or adding things?
>
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
>
>
> > On Jul 12, 2017 12:54 PM, "Igor Olhovskiy" <igorolhovskiy at gmail.com> wrote:
> > > Hi!
> > >
> > > I’m having strange issues on registering phones behind NAT on FreeSwitch.
> > >
> > > Schema is Phone (port 5063)  -> Router (NAT) (port 1023) -> Freeswitch
> > >
> > > Phone registering from port 5063. NAT is changing this port to 1023. Freeswitch receiving register message from port 1023, but still replying to port, that listed in Contact Header. So, Router can’t route this request back correctly.
> > >
> > > NDLB-force-rport and aggressive-nat-detection is enabled on internal profile. What I got, this issue is when router changes port to 1024 or lower. If port is higher that 1024, than it’s working ok.
> > > Also this problem with UDP only, with TCP all is ok.
> > >
> > > Maybe I’m missing something in config?
> > >
> > > Thanks.
> > >
> > > 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From joel at gogii.net  Thu Jul 13 03:24:50 2017
From: joel at gogii.net (Joel Serrano)
Date: Wed, 12 Jul 2017 20:24:50 -0700
Subject: [Freeswitch-users] sip to webrtc - sdp invalid description
In-Reply-To: <CAFGRPVpA522oxcn+hJBha6vms_2i8xUXTvz5gnNuotHEdTX5JA@mail.gmail.com>
References: <CAFGRPVoQB8mH3a8JYbY3wP1L=L3x-zdZF-kK1o+ocW5GzSYt-A@mail.gmail.com>
 <CAFGRPVrNFQ6rKYVehMX1bopaUqKqT0NQeDQkGigCGC5e5h7FUg@mail.gmail.com>
 <CAMtXxQm643PE7EJr11Tf7rkvoXuURicDfkR3GeuROQUSPkiECQ@mail.gmail.com>
 <CAMtXxQkB+cvj_HUF2p_ud8orbvTfPB-nL_=4se86WRoENkbc9A@mail.gmail.com>
 <CAFGRPVpA522oxcn+hJBha6vms_2i8xUXTvz5gnNuotHEdTX5JA@mail.gmail.com>
Message-ID: <CAMtXxQmu+D3UQSDE7xq3PV0UNS1=6T+YFhq3O9V-UoJZrD7ENA@mail.gmail.com>

Hi David,

I have:

[iOS/Android app with Linphone SIP stack] <-> Kamailio <-> FS <-> Kamailio
<-> [WebRTC]

And it works correctly in all combinations (both Kamailios are only
signalling, rtp goes from clients to FS directly).

Tomorrow i'll send you the combination of codec parameters we are using so
you can compare, we can do more tests from there with the traces.

Joel.

On Tue, Jul 11, 2017 at 4:58 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> many thanks!
> ᐧ
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337 <+34%20669%2044%2083%2037>
>
> On Sun, Jul 9, 2017 at 7:35 AM, Joel Serrano <joel at gogii.net> wrote:
>
>> Nevermind, I missed the part that FS is in the middle. I'm going to test
>> that and see what I find.
>>
>> On Sat, Jul 8, 2017 at 19:56 Joel Serrano <joel at gogii.net> wrote:
>>
>>> Can it be that Zoiper is using SRTP-3DES instead of SRTP-DTLS?
>>>
>>> WebRTC requires SRTP-DTLS.
>>>
>>>
>>>
>>> On Fri, Jul 7, 2017 at 9:46 AM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Hello guys,
>>>>
>>>> Any help on this?
>>>>
>>>> Thanks
>>>>
>>>> David
>>>> ᐧ
>>>>
>>>> Regards,
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.work at gmail.com
>>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>>>
>>>> On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> Hello guys,
>>>>>
>>>>> I have this setup:
>>>>>
>>>>> Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
>>>>>
>>>>> Whenever webrtc clients call each other, calls are ok.
>>>>> But when the zoiper (regular sip/tcp) calls, the browsers complaint
>>>>> about:
>>>>>
>>>>> "no ice-ufrag" (firefox) or
>>>>> "Failed to set remote offer sdp: Called with SDP without DTLS
>>>>> fingerprint." (Chrome).
>>>>>
>>>>> I am setting the sdp in freeswitch as:
>>>>>
>>>>>       <action application="set" data="proxy_media=false"/>
>>>>>       <action application="set" data="bypass_media=false"/>
>>>>>       <action application="export" data="zrtp_secure_media=false"/>
>>>>>       <action application="export" data="nolocal:rtp_secure_media
>>>>> =true"/>
>>>>>
>>>>> and the actual sdp is:
>>>>>
>>>>> v=0
>>>>> o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 1.2.3.4
>>>>> t=0 0
>>>>> m=audio 40954 RTP/SAVP 8 101 13
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=rtpmap:13 CN/8000
>>>>> a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVr
>>>>> VfjA0ImWgu5WyLK0vtg60RTk6Koe8c0sRkjU
>>>>> a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+D5NVoKFFw0
>>>>> 6I0Xok/9KbRystQ
>>>>> a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+ETPuCOiOIxjFLRjbF
>>>>> bDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg
>>>>> a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:evU8MzAtiSHwKb95s4V9IAM
>>>>> pmok06kW9ZGDgH3/Lc3ZytVn2SR4
>>>>> a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:KKDkT0DssohSeKFsX6tbixR
>>>>> hwYdiIhE6r3u5CCVA
>>>>> a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:D3JgGuOlIxXfHGdqf7lKWqN
>>>>> DAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w
>>>>> a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:c12ebWWzZ1cqZN0v5C5uYzd
>>>>> vtfnw6AARU3+jGA0WzTSDlDd20vI
>>>>> a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:nPZku7S6hlz2OPAff8T9I8s
>>>>> NFzuZziNg64KuvfNS
>>>>> a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3
>>>>> dhTqD0XW8FAOLNtmP
>>>>> a=ptime:20
>>>>> a=nortpproxy:yes
>>>>>
>>>>>
>>>>> I'm pretty new to the webrtc stuff, so any help is greatly appreciated!
>>>>>
>>>>> Thanks!
>>>>>
>>>>>
>>>>> David Villasmil
>>>>> email: david.villasmil.work at gmail.com
>>>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>>>> ᐧ
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Thu Jul 13 04:35:21 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Jul 2017 04:35:21 +0000
Subject: [Freeswitch-users] FreeSWITCH is Hiring a Software Developer!
Message-ID: <CAKbxfG-FpLTts=HkRk+VFp4t4XHHAQ_3n6fa5OX4yK5KHJ0U-w@mail.gmail.com>

Apply today:

https://freeswitch-solutions-llc.workable.com/jobs/523308



-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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

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

https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From joelists at tm.net.uk  Thu Jul 13 13:56:55 2017
From: joelists at tm.net.uk (Jospeh Waite)
Date: Thu, 13 Jul 2017 14:56:55 +0100
Subject: [Freeswitch-users] xml_radius only on bleg
Message-ID: <116730C0-C60C-437D-9EFC-76ADB5D6EA1D@tm.net.uk>

Hi Guys

I am trying to find a variable available to xml_radius to put in the regex section at the bottom of the config to only send a Star/Stop Radius accounting message for a successful connected b-leg.

I don’t want it to send any messages for the a-leg as Im only interested in accounting for CDR’s that actually connect.

Any help would be greatly appreciated.

Regards

From joel at gogii.net  Thu Jul 13 17:08:36 2017
From: joel at gogii.net (Joel Serrano)
Date: Thu, 13 Jul 2017 10:08:36 -0700
Subject: [Freeswitch-users] FreeSWITCH in debian stretch (9.0)
Message-ID: <CAMtXxQ=VEkJMEYc-70csHWGEWyDaaSernwGoBv=g1UUV13kb6Q@mail.gmail.com>

Hi all,

Does https://freeswitch.org/jira/browse/FS-9785 being fixed in v1.6.19 mean
that FS can now work on debian9?

Thanks,
Joel.
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From krice at freeswitch.org  Thu Jul 13 17:27:31 2017
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 13 Jul 2017 12:27:31 -0500
Subject: [Freeswitch-users] FreeSWITCH in debian stretch (9.0)
In-Reply-To: <CAMtXxQ=VEkJMEYc-70csHWGEWyDaaSernwGoBv=g1UUV13kb6Q@mail.gmail.com>
References: <CAMtXxQ=VEkJMEYc-70csHWGEWyDaaSernwGoBv=g1UUV13kb6Q@mail.gmail.com>
Message-ID: <092501d2fbfd$4f83efb0$ee8bcf10$@freeswitch.org>

We’re still working on this, but 1.6 will probably not have everything backported for stretch. The master branch is being worked on to address the remaining stretch issues watch for an announcement on that before long.

 

K

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joel Serrano
Sent: Thursday, July 13, 2017 12:09 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] FreeSWITCH in debian stretch (9.0)

 

Hi all, 

 

Does https://freeswitch.org/jira/browse/FS-9785 being fixed in v1.6.19 mean that FS can now work on debian9?

 

Thanks, 

Joel.

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From mario_fs at mgtech.com  Thu Jul 13 17:40:54 2017
From: mario_fs at mgtech.com (Mario G)
Date: Thu, 13 Jul 2017 10:40:54 -0700
Subject: [Freeswitch-users] macOS FreeSwitch Installer (macFI) Update
Message-ID: <CDB66F6E-A707-4276-812B-3A60A63FBAF4@mgtech.com>

For the macOS folks: macFI <https://freeswitch.org/confluence/display/FREESWITCH/macOS+macFI+Installation> was extensively updated and if you use it you should download the latest version.

Auto detemines the latest FS stable release, no need to update for new releases.

Can now download stable production, master development, and the current branch if needed

Other changes, see history <https://freeswitch.org/confluence/display/FREESWITCH/macOS+macFI+Installation#macOSmacFIInstallation-macFIChangeHistory>.
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From joel at gogii.net  Thu Jul 13 18:07:53 2017
From: joel at gogii.net (Joel Serrano)
Date: Thu, 13 Jul 2017 11:07:53 -0700
Subject: [Freeswitch-users] FreeSWITCH in debian stretch (9.0)
In-Reply-To: <092501d2fbfd$4f83efb0$ee8bcf10$@freeswitch.org>
References: <CAMtXxQ=VEkJMEYc-70csHWGEWyDaaSernwGoBv=g1UUV13kb6Q@mail.gmail.com>
 <092501d2fbfd$4f83efb0$ee8bcf10$@freeswitch.org>
Message-ID: <CAMtXxQ=ntpbeBJNN4V6J1wQ82pjLxnsoNRDjK0d1pQ18ETTtnQ@mail.gmail.com>

Will do! Thanks for the info Ken.

J.

On Thu, Jul 13, 2017 at 10:27 AM, Ken Rice <krice at freeswitch.org> wrote:

> We’re still working on this, but 1.6 will probably not have everything
> backported for stretch. The master branch is being worked on to address the
> remaining stretch issues watch for an announcement on that before long.
>
>
>
> K
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Joel Serrano
> *Sent:* Thursday, July 13, 2017 12:09 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] FreeSWITCH in debian stretch (9.0)
>
>
>
> Hi all,
>
>
>
> Does https://freeswitch.org/jira/browse/FS-9785 being fixed in v1.6.19
> mean that FS can now work on debian9?
>
>
>
> Thanks,
>
> Joel.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Thu Jul 13 18:48:00 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 13 Jul 2017 18:48:00 +0000
Subject: [Freeswitch-users] sip to webrtc - sdp invalid description
In-Reply-To: <CAMtXxQmu+D3UQSDE7xq3PV0UNS1=6T+YFhq3O9V-UoJZrD7ENA@mail.gmail.com>
References: <CAFGRPVoQB8mH3a8JYbY3wP1L=L3x-zdZF-kK1o+ocW5GzSYt-A@mail.gmail.com>
 <CAFGRPVrNFQ6rKYVehMX1bopaUqKqT0NQeDQkGigCGC5e5h7FUg@mail.gmail.com>
 <CAMtXxQm643PE7EJr11Tf7rkvoXuURicDfkR3GeuROQUSPkiECQ@mail.gmail.com>
 <CAMtXxQkB+cvj_HUF2p_ud8orbvTfPB-nL_=4se86WRoENkbc9A@mail.gmail.com>
 <CAFGRPVpA522oxcn+hJBha6vms_2i8xUXTvz5gnNuotHEdTX5JA@mail.gmail.com>
 <CAMtXxQmu+D3UQSDE7xq3PV0UNS1=6T+YFhq3O9V-UoJZrD7ENA@mail.gmail.com>
Message-ID: <CAFGRPVrzDri9YSO8BYPaARRA1akq=7eNO5Mk4my6ueb_8d1GWg@mail.gmail.com>

That would be amazing, thanks! And if you could share you kamailio config,
that would be awesome!
On Thu, Jul 13, 2017 at 5:26 AM Joel Serrano <joel at gogii.net> wrote:

> Hi David,
>
> I have:
>
> [iOS/Android app with Linphone SIP stack] <-> Kamailio <-> FS <-> Kamailio
> <-> [WebRTC]
>
> And it works correctly in all combinations (both Kamailios are only
> signalling, rtp goes from clients to FS directly).
>
> Tomorrow i'll send you the combination of codec parameters we are using so
> you can compare, we can do more tests from there with the traces.
>
> Joel.
>
> On Tue, Jul 11, 2017 at 4:58 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> many thanks!
>> ᐧ
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>
>> On Sun, Jul 9, 2017 at 7:35 AM, Joel Serrano <joel at gogii.net> wrote:
>>
>>> Nevermind, I missed the part that FS is in the middle. I'm going to test
>>> that and see what I find.
>>>
>>> On Sat, Jul 8, 2017 at 19:56 Joel Serrano <joel at gogii.net> wrote:
>>>
>>>> Can it be that Zoiper is using SRTP-3DES instead of SRTP-DTLS?
>>>>
>>>> WebRTC requires SRTP-DTLS.
>>>>
>>>>
>>>>
>>>> On Fri, Jul 7, 2017 at 9:46 AM, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> Hello guys,
>>>>>
>>>>> Any help on this?
>>>>>
>>>>> Thanks
>>>>>
>>>>> David
>>>>> ᐧ
>>>>>
>>>>> Regards,
>>>>>
>>>>> David Villasmil
>>>>> email: david.villasmil.work at gmail.com
>>>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>>>>
>>>>> On Thu, Jul 6, 2017 at 11:11 PM, David Villasmil <
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>>> Hello guys,
>>>>>>
>>>>>> I have this setup:
>>>>>>
>>>>>> Zoiper-->Kamailio->fs->kamailio->webrtc client(s)
>>>>>>
>>>>>> Whenever webrtc clients call each other, calls are ok.
>>>>>> But when the zoiper (regular sip/tcp) calls, the browsers complaint
>>>>>> about:
>>>>>>
>>>>>> "no ice-ufrag" (firefox) or
>>>>>> "Failed to set remote offer sdp: Called with SDP without DTLS
>>>>>> fingerprint." (Chrome).
>>>>>>
>>>>>> I am setting the sdp in freeswitch as:
>>>>>>
>>>>>>       <action application="set" data="proxy_media=false"/>
>>>>>>       <action application="set" data="bypass_media=false"/>
>>>>>>       <action application="export" data="zrtp_secure_media=false"/>
>>>>>>       <action application="export"
>>>>>> data="nolocal:rtp_secure_media=true"/>
>>>>>>
>>>>>> and the actual sdp is:
>>>>>>
>>>>>> v=0
>>>>>> o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4
>>>>>> s=FreeSWITCH
>>>>>> c=IN IP4 1.2.3.4
>>>>>> t=0 0
>>>>>> m=audio 40954 RTP/SAVP 8 101 13
>>>>>> a=rtpmap:8 PCMA/8000
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>> a=rtpmap:13 CN/8000
>>>>>> a=crypto:1 AEAD_AES_256_GCM_8
>>>>>> inline:9XidL0Z5VYb0L5CegRZaYVrVfjA0ImWgu5WyLK0vtg60RTk6Koe8c0sRkjU
>>>>>> a=crypto:2 AEAD_AES_128_GCM_8
>>>>>> inline:fLKE1lxhRoVw+D5NVoKFFw06I0Xok/9KbRystQ
>>>>>> a=crypto:3 AES_CM_256_HMAC_SHA1_80
>>>>>> inline:w/yDN+ETPuCOiOIxjFLRjbFbDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg
>>>>>> a=crypto:4 AES_CM_192_HMAC_SHA1_80
>>>>>> inline:evU8MzAtiSHwKb95s4V9IAMpmok06kW9ZGDgH3/Lc3ZytVn2SR4
>>>>>> a=crypto:5 AES_CM_128_HMAC_SHA1_80
>>>>>> inline:KKDkT0DssohSeKFsX6tbixRhwYdiIhE6r3u5CCVA
>>>>>> a=crypto:6 AES_CM_256_HMAC_SHA1_32
>>>>>> inline:D3JgGuOlIxXfHGdqf7lKWqNDAIiJrbOqOKb+erlhPQtBKF4wzomjbN0sBIiE4w
>>>>>> a=crypto:7 AES_CM_192_HMAC_SHA1_32
>>>>>> inline:c12ebWWzZ1cqZN0v5C5uYzdvtfnw6AARU3+jGA0WzTSDlDd20vI
>>>>>> a=crypto:8 AES_CM_128_HMAC_SHA1_32
>>>>>> inline:nPZku7S6hlz2OPAff8T9I8sNFzuZziNg64KuvfNS
>>>>>> a=crypto:9 AES_CM_128_NULL_AUTH
>>>>>> inline:9MNdj7xaingoGY14NUg8iS3dhTqD0XW8FAOLNtmP
>>>>>> a=ptime:20
>>>>>> a=nortpproxy:yes
>>>>>>
>>>>>>
>>>>>> I'm pretty new to the webrtc stuff, so any help is greatly
>>>>>> appreciated!
>>>>>>
>>>>>> Thanks!
>>>>>>
>>>>>>
>>>>>> David Villasmil
>>>>>> email: david.villasmil.work at gmail.com
>>>>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>>>>> ᐧ
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From nneul at mst.edu  Thu Jul 13 19:42:58 2017
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 13 Jul 2017 14:42:58 -0500
Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8"
	upgrade notes document?
Message-ID: <55259459-7390-1fe4-c6a1-d5ba03acac1b@mst.edu>

I remember moving from 1.4 to 1.6 there were several breaking behavior changes. Are you planning on doing anything to 
document specific known "you have to change X to Y" and "this behavior has changed" sort of document for sites upgrading 
from 1.6.x to 1.8.x?

-- Nathan

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


From andretodd at verizon.net  Thu Jul 13 20:06:19 2017
From: andretodd at verizon.net (Andre DeMattia)
Date: Thu, 13 Jul 2017 16:06:19 -0400
Subject: [Freeswitch-users] Channel Var Codec
Message-ID: <019701d2fc13$8d12e690$a738b3b0$@verizon.net>

 Hi what Channel variable should I use to see what Codec the A leg allows?

variable_rtp_use_codec_string or variable_ep_codec_string or something else?

In the CDR I see this
<rtp_use_codec_string>G722,PCMU,PCMA,GSM</rtp_use_codec_string>

This looks like what I'm looking for but I need it in the dialplan before the call goes out so I can direct the call where I need it to go.
Thanks



From joelists at tm.net.uk  Thu Jul 13 21:59:04 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Thu, 13 Jul 2017 22:59:04 +0100
Subject: [Freeswitch-users] Manipulating variables & doing a for loop in
	dialplan
Message-ID: <B783111F-4DAF-465A-A91F-D0A4CAD28642@tm.net.uk>

Hi Guys

Been googling but can’t really find anything on how to do what I need.

I receive the following in a variable sip_redirect_contact_<index>

<sip:441554555666 at 185.35.228.48:5060>;src_number=441554333444;q=1.00

I need to pull out the 441554333444 after src_number= and before th ;

How would I achieve this? 

Also is there any way to do a looping dial plan, that will execute a section of dial plan for each <index>

So I would have a block of xml that would run for each <index> from the sip_redirect_contact.

So for each sip_redirect_contact would execute a block of xml for each result until the bridge application connected a b-leg

Regards

From ljjimenez at gmail.com  Thu Jul 13 22:04:02 2017
From: ljjimenez at gmail.com (Luis Jimenez)
Date: Thu, 13 Jul 2017 18:04:02 -0400
Subject: [Freeswitch-users] Manipulating variables & doing a for loop in
	dialplan
In-Reply-To: <B783111F-4DAF-465A-A91F-D0A4CAD28642@tm.net.uk>
References: <B783111F-4DAF-465A-A91F-D0A4CAD28642@tm.net.uk>
Message-ID: <A8F2A583-67F2-4127-AB1A-D4A8E300FE90@gmail.com>

Regular expression, lua, mod_xml_curl


> On Jul 13, 2017, at 17:59, Joseph Waite <joelists at tm.net.uk> wrote:
> 
> Hi Guys
> 
> Been googling but can’t really find anything on how to do what I need.
> 
> I receive the following in a variable sip_redirect_contact_<index>
> 
> <sip:441554555666 at 185.35.228.48:5060>;src_number=441554333444;q=1.00
> 
> I need to pull out the 441554333444 after src_number= and before th ;
> 
> How would I achieve this? 
> 
> Also is there any way to do a looping dial plan, that will execute a section of dial plan for each <index>
> 
> So I would have a block of xml that would run for each <index> from the sip_redirect_contact.
> 
> So for each sip_redirect_contact would execute a block of xml for each result until the bridge application connected a b-leg
> 
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From matt.nichols at westtel.com  Thu Jul 13 16:25:51 2017
From: matt.nichols at westtel.com (Matthew Nichols)
Date: Thu, 13 Jul 2017 16:25:51 +0000
Subject: [Freeswitch-users] FreeSWITCH HA and BDR, wiki
Message-ID: <DM5PR1001MB21554F95B2AC0DFB15CC6AF883AC0@DM5PR1001MB2155.namprd10.prod.outlook.com>

After going through https://freeswitch.org/confluence/display/FREESWITCH/High+Availability, it appears out of date/misleading. Here are the things I have found:

Switchname needs to be different or they end up clobbering each other. Call recovery still works.

$${domain} needs to be set the same on both boxes for voicemails and registrations to work well after recovery.

As far as I can tell the entirety of /var/lib/freeswitch can be shared with something like GlusterFS (assuming sqlite is not being used).

Especially with PostgreSQL BDR auto-create-sql needs to be false and auto-clear-sql needs to be true, otherwise information is never cleared after a crash.

Also I see no documentation on the wiki on actually using FreeSWITCH with PostgreSQL BDR. I would be willing to start a page with what I have discovered so far. The only online documentation I have found is https://gist.github.com/DigiDaz/1cfe3d5d32080a8e3d75a20bb5bc4fb5, which doesn't have the primary keys set up quite right (calls and channels need to add hostname to the primary key or call recovery doesn't work quite right) and doesn't include voicemail.

I am very willing to setup the required edits but I don't have write access so I would need to go through one of you who does.

Matthew Nichols

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From joelists at tm.net.uk  Fri Jul 14 09:04:31 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Fri, 14 Jul 2017 10:04:31 +0100
Subject: [Freeswitch-users] FreeSWITCH HA and BDR, wiki
In-Reply-To: <DM5PR1001MB21554F95B2AC0DFB15CC6AF883AC0@DM5PR1001MB2155.namprd10.prod.outlook.com>
References: <DM5PR1001MB21554F95B2AC0DFB15CC6AF883AC0@DM5PR1001MB2155.namprd10.prod.outlook.com>
Message-ID: <9291D21A-A720-44FE-B95E-D09EDDCB12BD@tm.net.uk>

Hi Matthew

I'm about to attempt the same. And notes/corrected db schemes you got would be much appreciated!

Joe Waite

> On 13 Jul 2017, at 17:25, Matthew Nichols <matt.nichols at westtel.com> wrote:
> 
> After going through https://freeswitch.org/confluence/display/FREESWITCH/High+Availability, it appears out of date/misleading. Here are the things I have found:
>  
> Switchname needs to be different or they end up clobbering each other. Call recovery still works.
>  
> $${domain} needs to be set the same on both boxes for voicemails and registrations to work well after recovery.
>  
> As far as I can tell the entirety of /var/lib/freeswitch can be shared with something like GlusterFS (assuming sqlite is not being used).
>  
> Especially with PostgreSQL BDR auto-create-sql needs to be false and auto-clear-sql needs to be true, otherwise information is never cleared after a crash.
>  
> Also I see no documentation on the wiki on actually using FreeSWITCH with PostgreSQL BDR. I would be willing to start a page with what I have discovered so far. The only online documentation I have found is https://gist.github.com/DigiDaz/1cfe3d5d32080a8e3d75a20bb5bc4fb5, which doesn’t have the primary keys set up quite right (calls and channels need to add hostname to the primary key or call recovery doesn’t work quite right) and doesn’t include voicemail.
>  
> I am very willing to setup the required edits but I don’t have write access so I would need to go through one of you who does.
>  
> Matthew Nichols
>  
> <Matthew Nichols.vcf>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From joelists at tm.net.uk  Fri Jul 14 09:48:57 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Fri, 14 Jul 2017 10:48:57 +0100
Subject: [Freeswitch-users] xml_radius only on bleg
In-Reply-To: <116730C0-C60C-437D-9EFC-76ADB5D6EA1D@tm.net.uk>
References: <116730C0-C60C-437D-9EFC-76ADB5D6EA1D@tm.net.uk>
Message-ID: <486A8610-0563-4A1F-9BD3-1E045C0B214B@tm.net.uk>

Ok I have worked this out.

<param var="direction" regex="^outbound”/>

Just incase anyone in future needs this!
> On 13 Jul 2017, at 14:56, Jospeh Waite <joelists at tm.net.uk> wrote:
> 
> Hi Guys
> 
> I am trying to find a variable available to xml_radius to put in the regex section at the bottom of the config to only send a Star/Stop Radius accounting message for a successful connected b-leg.
> 
> I don’t want it to send any messages for the a-leg as Im only interested in accounting for CDR’s that actually connect.
> 
> Any help would be greatly appreciated.
> 
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From krice at freeswitch.org  Fri Jul 14 14:09:34 2017
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 14 Jul 2017 09:09:34 -0500
Subject: [Freeswitch-users] Are you planning on creating a "1.6 ->
	1.8"	upgrade notes document?
In-Reply-To: <55259459-7390-1fe4-c6a1-d5ba03acac1b@mst.edu>
References: <55259459-7390-1fe4-c6a1-d5ba03acac1b@mst.edu>
Message-ID: <0e2801d2fcaa$d3031480$79093d80$@freeswitch.org>

There will probably be a Wiki page for this, however, I doubt there will be much on it. In reality, I don’t think there's any config breaking changes from 1.6 to master at this point, and master is already being staged for early 1.8 releases. We'll prob see beta releases of 1.8 very soon. If you want to help us on this effort, please checkout the master branch (packages are already available see confluence) and test away.

Ken

-----Original Message-----
From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nathan Neulinger
Sent: Thursday, July 13, 2017 2:43 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8" upgrade notes document?

I remember moving from 1.4 to 1.6 there were several breaking behavior changes. Are you planning on doing anything to document specific known "you have to change X to Y" and "this behavior has changed" sort of document for sites upgrading from 1.6.x to 1.8.x?

-- Nathan

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

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

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

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



From nneul at mst.edu  Fri Jul 14 14:16:12 2017
From: nneul at mst.edu (Nathan Neulinger)
Date: Fri, 14 Jul 2017 09:16:12 -0500
Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8"
 upgrade notes document?
In-Reply-To: <0e2801d2fcaa$d3031480$79093d80$@freeswitch.org>
References: <55259459-7390-1fe4-c6a1-d5ba03acac1b@mst.edu>
 <0e2801d2fcaa$d3031480$79093d80$@freeswitch.org>
Message-ID: <40c92e39-896b-9427-9d3e-ec5695ecce21@mst.edu>

That's good.

I thought I remembered seeing something several months back about large structural changes in the config 
files/layout/etc. Maybe I misread something.

Will definitely give new branch a try soon.

-- Nathan

On 7/14/17 9:09 AM, Ken Rice wrote:
> There will probably be a Wiki page for this, however, I doubt there will be much on it. In reality, I don’t think there's any config breaking changes from 1.6 to master at this point, and master is already being staged for early 1.8 releases. We'll prob see beta releases of 1.8 very soon. If you want to help us on this effort, please checkout the master branch (packages are already available see confluence) and test away.
> 
> Ken
> 
> -----Original Message-----
> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nathan Neulinger
> Sent: Thursday, July 13, 2017 2:43 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8" upgrade notes document?
> 
> I remember moving from 1.4 to 1.6 there were several breaking behavior changes. Are you planning on doing anything to document specific known "you have to change X to Y" and "this behavior has changed" sort of document for sites upgrading from 1.6.x to 1.8.x?
> 
> -- Nathan
> 
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

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


From asilva at wirelessmundi.com  Fri Jul 14 14:25:02 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Fri, 14 Jul 2017 16:25:02 +0200
Subject: [Freeswitch-users] Error sending queries to pgsql
In-Reply-To: <CAKeZRhvm=NgcOxXQSBJkQQbOGvJEdwyhRunXEW8f6cskEa6d7g@mail.gmail.com>
References: <d2f4526f-a783-a216-405d-126c253e34fd@wirelessmundi.com>
 <31066AE1-BC31-4C3F-8809-6C5B93C3B0E8@jerris.com>
 <f76beadd-9014-d279-1fb8-ea9d0048b240@wirelessmundi.com>
 <CAKeZRhvyLu+uT4_c_ZSEQVxW5zYZgrgeuAf10Qxpqn_5B-XRTA@mail.gmail.com>
 <39b55634-01fd-afea-a08f-27094f2e237c@wirelessmundi.com>
 <1d4ae6f9-d857-4cba-c780-dc5ceaee004f@wirelessmundi.com>
 <CAKeZRhvm=NgcOxXQSBJkQQbOGvJEdwyhRunXEW8f6cskEa6d7g@mail.gmail.com>
Message-ID: <a48cdd8b-d2b1-6e33-b9ca-843531f8ef3a@wirelessmundi.com>

Hi Alexandr,

I've advance something in debugging this issue..

Do you use sip/wss or verto endpoints in your box?

Saludos / Regards / Cumprimentos,
António silva

On 07/06/2017 04:13 PM, Alexandr Popov wrote:
> i'm receiving two type of of error at FS are [FATAL:  invalid frontend 
> message type 21
> ]
> and
> This probably means the server terminated abnormally
> before or while processing the request.
> [Error sending query!]
>
>
> at postgers log i getting only -- invalid frontend message type 21
>
>
> 2017-07-06 15:26 GMT+03:00 Antonio Silva <asilva at wirelessmundi.com 
> <mailto:asilva at wirelessmundi.com>>:
>
>     Hi,
>
>     same results on old kernel..
>     i'm trying to reproduce it on isolate machine.. for now i put my
>     findings in jira https://freeswitch.org/jira/browse/FS-10474
>     <https://freeswitch.org/jira/browse/FS-10474>.
>
>
>
>     Alexandr:
>
>     What SO are you running? did you get some errors from postgres
>     before fs critical message?
>
>
>     Saludos / Regards / Cumprimentos,
>     António silva
>
>     On 07/04/2017 11:34 AM, Antonio Silva wrote:
>>     In this same box i also have data corruption with sqlite
>>     databases, special the ones that i use with mod_lua.. but I think
>>     that it could be something related with kernel.. i recently
>>     update to 4.9.x, and there was some changes in ext4.
>>
>>     I don't really know how to debug this, so today i'm reverting to
>>     the old kernel, 4.4.x and check it happens again..
>>
>>     If it happens again, my next move will try to reproduce this in FS..
>>
>>     Saludos / Regards / Cumprimentos,
>>     António silva
>>     On 07/04/2017 10:58 AM, Alexandr Popov wrote:
>>>     Seems its trouble with sockets. I have the same problem appeared
>>>     about a month ago.
>>>
>>>     2017-06-29 19:31 GMT+03:00 Antonio Silva
>>>     <asilva at wirelessmundi.com <mailto:asilva at wirelessmundi.com>>:
>>>
>>>         Hi Michael,
>>>
>>>         Yes, i'm trying to figure it out if is an issue in FS or
>>>         external.. but the message from PG i can't translate it.. i
>>>         just enable more logs to see if i got extra hints...
>>>
>>>
>>>         Thanks.
>>>
>>>         Saludos / Regards / Cumprimentos,
>>>         António silva
>>>
>>>         On 06/29/2017 06:22 PM, Michael Jerris wrote:
>>>
>>>             If you can figure out a reliable way to reproduce this
>>>             issue, please file a jira with details on what causes it.
>>>
>>>                 On Jun 29, 2017, at 7:22 AM, Antonio Silva
>>>                 <asilva at wirelessmundi.com
>>>                 <mailto:asilva at wirelessmundi.com>> wrote:
>>>
>>>                 Hi all,
>>>
>>>                 i use pgsql in core and from time to time i see
>>>                 critical messages like fail to send query, example:
>>>
>>>                 [CRIT] switch_pgsql.c:255 Failed to send query
>>>                 (update sip_authentication set
>>>                 expires='1498726568',last_nc=364 where
>>>                 nonce='3c20da2e-b663-46fe-8edc-c756f9166d07') to
>>>                 database: server closed the connection unexpectedly
>>>
>>>                 This was recently, i did an update to current master
>>>                 the previous version was from April, not sure if it
>>>                 could be an error on FS o some other issue on my box..
>>>
>>>
>>>                 PG is installed on the same server and the only
>>>                 thing i see from pg is "postgres[2236]: FATAL: 
>>>                 invalid frontend message type 21", PG is installed
>>>                 on the same server, running on /dev/shm with the
>>>                 same prio as FS and  the process never stopped.
>>>
>>>
>>>                 anyone has experience this error before? any idea
>>>                 what it could be the cause?
>>>
>>>
>>>             _________________________________________________________________________
>>>             Professional FreeSWITCH Consulting Services:
>>>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>             http://www.freeswitchsolutions.com
>>>             <http://www.freeswitchsolutions.com>
>>>
>>>             Official FreeSWITCH Sites
>>>             http://www.freeswitch.org
>>>             http://confluence.freeswitch.org
>>>             <http://confluence.freeswitch.org>
>>>             http://www.cluecon.com
>>>
>>>             FreeSWITCH-users mailing list
>>>             FreeSWITCH-users at lists.freeswitch.org
>>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>             <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>             <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>             http://www.freeswitch.org
>>>
>>>
>>>
>>>         _________________________________________________________________________
>>>         Professional FreeSWITCH Consulting Services:
>>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>         http://www.freeswitchsolutions.com
>>>         <http://www.freeswitchsolutions.com>
>>>
>>>         Official FreeSWITCH Sites
>>>         http://www.freeswitch.org
>>>         http://confluence.freeswitch.org
>>>         <http://confluence.freeswitch.org>
>>>         http://www.cluecon.com
>>>
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org
>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>         http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>     _________________________________________________________________________
>>>     Professional FreeSWITCH Consulting Services:
>>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>     http://www.freeswitchsolutions.com
>>>     <http://www.freeswitchsolutions.com>
>>>
>>>     Official FreeSWITCH Sites
>>>     http://www.freeswitch.org
>>>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>>>     http://www.cluecon.com
>>>
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>     http://www.freeswitch.org
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>     <http://www.freeswitchsolutions.com>
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>>     http://www.cluecon.com
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>     http://www.freeswitch.org
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com> Official FreeSWITCH Sites
>     http://www.freeswitch.org http://confluence.freeswitch.org
>     <http://confluence.freeswitch.org> http://www.cluecon.com
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>     http://www.freeswitch.org 
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mario_fs at mgtech.com  Fri Jul 14 15:44:02 2017
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 14 Jul 2017 08:44:02 -0700
Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8"
 upgrade notes document?
In-Reply-To: <40c92e39-896b-9427-9d3e-ec5695ecce21@mst.edu>
References: <55259459-7390-1fe4-c6a1-d5ba03acac1b@mst.edu>
 <0e2801d2fcaa$d3031480$79093d80$@freeswitch.org>
 <40c92e39-896b-9427-9d3e-ec5695ecce21@mst.edu>
Message-ID: <F3309FCC-A03A-4828-B4F0-35362DF52509@mgtech.com>

FYI, I had to test 1.8 a couple of weeks ago on the normal running system and for expediency I tried just dragging my 1.6 conf over to 1.8 and all worked fine for a week. However, when 1.8 comes out I go through every config line and make a new conf.

> On Jul 14, 2017, at 7:16 AM, Nathan Neulinger <nneul at mst.edu> wrote:
> 
> That's good.
> 
> I thought I remembered seeing something several months back about large structural changes in the config files/layout/etc. Maybe I misread something.
> 
> Will definitely give new branch a try soon.
> 
> -- Nathan
> 
> On 7/14/17 9:09 AM, Ken Rice wrote:
>> There will probably be a Wiki page for this, however, I doubt there will be much on it. In reality, I don’t think there's any config breaking changes from 1.6 to master at this point, and master is already being staged for early 1.8 releases. We'll prob see beta releases of 1.8 very soon. If you want to help us on this effort, please checkout the master branch (packages are already available see confluence) and test away.
>> Ken
>> -----Original Message-----
>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nathan Neulinger
>> Sent: Thursday, July 13, 2017 2:43 PM
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8" upgrade notes document?
>> I remember moving from 1.4 to 1.6 there were several breaking behavior changes. Are you planning on doing anything to document specific known "you have to change X to Y" and "this behavior has changed" sort of document for sites upgrading from 1.6.x to 1.8.x?
>> -- Nathan
>> ------------------------------------------------------------
>> Nathan Neulinger                       nneul at mst.edu
>> Missouri S&T Information Technology    (573) 612-1412
>> System Administrator - Architect
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> -- 
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jungleboogie0 at gmail.com  Fri Jul 14 16:57:29 2017
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Fri, 14 Jul 2017 09:57:29 -0700
Subject: [Freeswitch-users] macOS FreeSwitch Installer (macFI) Update
In-Reply-To: <CDB66F6E-A707-4276-812B-3A60A63FBAF4@mgtech.com>
References: <CDB66F6E-A707-4276-812B-3A60A63FBAF4@mgtech.com>
Message-ID: <CAKE2PDsJar+qfeJCGA+n1VRNkKfuOJRxBuVUZH=0m4FKS_HSEQ@mail.gmail.com>

Hi Mario,
On 13 July 2017 at 10:40, Mario G <mario_fs at mgtech.com> wrote:
> For the macOS folks: macFI was extensively updated and if you use it you
> should download the latest version.
>

Just curious, what hardware are you using to run freeswitch on with
the Mac operating system?
Does it have some gui that you're attracted to and that's why you're
running MacOS?


From krice at freeswitch.org  Fri Jul 14 17:01:47 2017
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 14 Jul 2017 12:01:47 -0500
Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8"
	upgrade notes document?
In-Reply-To: <40c92e39-896b-9427-9d3e-ec5695ecce21@mst.edu>
References: <55259459-7390-1fe4-c6a1-d5ba03acac1b@mst.edu>
 <0e2801d2fcaa$d3031480$79093d80$@freeswitch.org>
 <40c92e39-896b-9427-9d3e-ec5695ecce21@mst.edu>
Message-ID: <115901d2fcc2$e170b280$a4521780$@freeswitch.org>

The layout of the config files on the filesystem are not set in stone... The only thing set in stone is the base freeswitch.xml in your conf Directory.

This allows for flexibility in the actual config files in your config tree.

The changes Brian has been working on are based on just changing the config files. The config handling code in FreeSWITCH doesn’t require changes for this. As such, this does not affect someone copying their old configs into a master/1.8 install and just using them.

Any changes to actual config directives (which at this point I don’t think anything has actually changed) would be properly documented.

When it comes down to it the example configs we ship with FreeSWITCH, it's just an example of 1 way to do it.

Now back to the config changes Brian has been working on, As most of you are aware the FreeSWITCH team is quite small, and a project such as this requires many many person hours of work to maintain and continue adding new features. Brian put out a call for help from the community on this so that stake holders could help implement these changes and give feedback on them. However I don’t think he's received that much help on them and as such his other duties have slowed that project drastically. If you wish to help you can fork the git repo located at https://freeswitch.org/stash/projects/FS/repos/fs18configs/browse and propose pull requests.

-----Original Message-----
From: Nathan Neulinger [mailto:nneul at mst.edu] 
Sent: Friday, July 14, 2017 9:16 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>; Ken Rice <krice at freeswitch.org>
Subject: Re: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8" upgrade notes document?

That's good.

I thought I remembered seeing something several months back about large structural changes in the config files/layout/etc. Maybe I misread something.

Will definitely give new branch a try soon.

-- Nathan

On 7/14/17 9:09 AM, Ken Rice wrote:
> There will probably be a Wiki page for this, however, I doubt there will be much on it. In reality, I don’t think there's any config breaking changes from 1.6 to master at this point, and master is already being staged for early 1.8 releases. We'll prob see beta releases of 1.8 very soon. If you want to help us on this effort, please checkout the master branch (packages are already available see confluence) and test away.
> 
> Ken
> 
> -----Original Message-----
> From: FreeSWITCH-users 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> Nathan Neulinger
> Sent: Thursday, July 13, 2017 2:43 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8" upgrade notes document?
> 
> I remember moving from 1.4 to 1.6 there were several breaking behavior changes. Are you planning on doing anything to document specific known "you have to change X to Y" and "this behavior has changed" sort of document for sites upgrading from 1.6.x to 1.8.x?
> 
> -- Nathan
> 
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
> 
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
> 

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



From mario_fs at mgtech.com  Fri Jul 14 17:17:00 2017
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 14 Jul 2017 10:17:00 -0700
Subject: [Freeswitch-users] macOS FreeSwitch Installer (macFI) Update
In-Reply-To: <CAKE2PDsJar+qfeJCGA+n1VRNkKfuOJRxBuVUZH=0m4FKS_HSEQ@mail.gmail.com>
References: <CDB66F6E-A707-4276-812B-3A60A63FBAF4@mgtech.com>
 <CAKE2PDsJar+qfeJCGA+n1VRNkKfuOJRxBuVUZH=0m4FKS_HSEQ@mail.gmail.com>
Message-ID: <7BE10ABC-9FD5-48E0-B764-96117E17AF6F@mgtech.com>

I run the 24x7 system on a 2012 Mac Mini, it also serves up others things such as videos, etc. and fits nicely in the network rack in a closet. I run macOS because it’s the most secure system (I have been doing mainframe security 40 years), I have 2 Minis and 2 iMacs. On 2 of the machines, I can boot multiple versions of macOS or Linux, and run Windows easily. Can’t get a PC to do that.

Many years ago, when I had employees, I switched from Windows to Macs when I spent thousands of dollars sending them to Windows classes. We had one Mac used for desktop publishing. everyone used to do letters, etc. No training needed, everything was easier and faster. I, who was heavy into PCs and thought the Mac was a toy, also snuck using it! No more training classes, I switched and never looked back. I know a lot of people who spend days fixing Windows problems that don’t exist on a Mac. Just my 2 cents.


> On Jul 14, 2017, at 9:57 AM, jungle Boogie <jungleboogie0 at gmail.com> wrote:
> 
> Hi Mario,
> On 13 July 2017 at 10:40, Mario G <mario_fs at mgtech.com> wrote:
>> For the macOS folks: macFI was extensively updated and if you use it you
>> should download the latest version.
>> 
> 
> Just curious, what hardware are you using to run freeswitch on with
> the Mac operating system?
> Does it have some gui that you're attracted to and that's why you're
> running MacOS?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From hawkins at hawkinsegroup.com  Fri Jul 14 19:34:21 2017
From: hawkins at hawkinsegroup.com (Don Hawkins)
Date: Fri, 14 Jul 2017 14:34:21 -0500
Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8"
 upgrade notes document?
In-Reply-To: <115901d2fcc2$e170b280$a4521780$@freeswitch.org>
References: <55259459-7390-1fe4-c6a1-d5ba03acac1b@mst.edu>
 <0e2801d2fcaa$d3031480$79093d80$@freeswitch.org>
 <40c92e39-896b-9427-9d3e-ec5695ecce21@mst.edu>
 <115901d2fcc2$e170b280$a4521780$@freeswitch.org>
Message-ID: <CAO6gFMwGSf0mKeVyPSAJ0vUxHyYb37M602yUyHMFyo0Y-6hRUQ@mail.gmail.com>

Sorry if I'm off subject here a bit, where can I find the change log for
the latest version of FS?

On Fri, Jul 14, 2017 at 12:01 PM, Ken Rice <krice at freeswitch.org> wrote:

> The layout of the config files on the filesystem are not set in stone...
> The only thing set in stone is the base freeswitch.xml in your conf
> Directory.
>
> This allows for flexibility in the actual config files in your config tree.
>
> The changes Brian has been working on are based on just changing the
> config files. The config handling code in FreeSWITCH doesn’t require
> changes for this. As such, this does not affect someone copying their old
> configs into a master/1.8 install and just using them.
>
> Any changes to actual config directives (which at this point I don’t think
> anything has actually changed) would be properly documented.
>
> When it comes down to it the example configs we ship with FreeSWITCH, it's
> just an example of 1 way to do it.
>
> Now back to the config changes Brian has been working on, As most of you
> are aware the FreeSWITCH team is quite small, and a project such as this
> requires many many person hours of work to maintain and continue adding new
> features. Brian put out a call for help from the community on this so that
> stake holders could help implement these changes and give feedback on them.
> However I don’t think he's received that much help on them and as such his
> other duties have slowed that project drastically. If you wish to help you
> can fork the git repo located at https://freeswitch.org/stash/
> projects/FS/repos/fs18configs/browse and propose pull requests.
>
> -----Original Message-----
> From: Nathan Neulinger [mailto:nneul at mst.edu]
> Sent: Friday, July 14, 2017 9:16 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>; Ken
> Rice <krice at freeswitch.org>
> Subject: Re: [Freeswitch-users] Are you planning on creating a "1.6 ->
> 1.8" upgrade notes document?
>
> That's good.
>
> I thought I remembered seeing something several months back about large
> structural changes in the config files/layout/etc. Maybe I misread
> something.
>
> Will definitely give new branch a try soon.
>
> -- Nathan
>
> On 7/14/17 9:09 AM, Ken Rice wrote:
> > There will probably be a Wiki page for this, however, I doubt there will
> be much on it. In reality, I don’t think there's any config breaking
> changes from 1.6 to master at this point, and master is already being
> staged for early 1.8 releases. We'll prob see beta releases of 1.8 very
> soon. If you want to help us on this effort, please checkout the master
> branch (packages are already available see confluence) and test away.
> >
> > Ken
> >
> > -----Original Message-----
> > From: FreeSWITCH-users
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> > Nathan Neulinger
> > Sent: Thursday, July 13, 2017 2:43 PM
> > To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8"
> upgrade notes document?
> >
> > I remember moving from 1.4 to 1.6 there were several breaking behavior
> changes. Are you planning on doing anything to document specific known "you
> have to change X to Y" and "this behavior has changed" sort of document for
> sites upgrading from 1.6.x to 1.8.x?
> >
> > -- Nathan
> >
> > ------------------------------------------------------------
> > Nathan Neulinger                       nneul at mst.edu
> > Missouri S&T Information Technology    (573) 612-1412
> > System Administrator - Architect
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> >
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> >
>
> --
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 


*Sincerely,*
Don Hawkins
CEO
Hawkins Enterprise Group LLC
http://corporate.hawkinsegroup.com
Zello PTT <http://zello.com>: push2don
P: 469-214-5044
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From krice at freeswitch.org  Fri Jul 14 19:41:44 2017
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 14 Jul 2017 14:41:44 -0500
Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8"
	upgrade notes document?
In-Reply-To: <CAO6gFMwGSf0mKeVyPSAJ0vUxHyYb37M602yUyHMFyo0Y-6hRUQ@mail.gmail.com>
References: <55259459-7390-1fe4-c6a1-d5ba03acac1b@mst.edu>
 <0e2801d2fcaa$d3031480$79093d80$@freeswitch.org>
 <40c92e39-896b-9427-9d3e-ec5695ecce21@mst.edu>
 <115901d2fcc2$e170b280$a4521780$@freeswitch.org>
 <CAO6gFMwGSf0mKeVyPSAJ0vUxHyYb37M602yUyHMFyo0Y-6hRUQ@mail.gmail.com>
Message-ID: <11a501d2fcd9$39e94f60$adbbee20$@freeswitch.org>

They are published on the website every release... check the blog

 

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Don Hawkins
Sent: Friday, July 14, 2017 2:34 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8" upgrade notes document?

 

Sorry if I'm off subject here a bit, where can I find the change log for the latest version of FS?

 

On Fri, Jul 14, 2017 at 12:01 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> > wrote:

The layout of the config files on the filesystem are not set in stone... The only thing set in stone is the base freeswitch.xml in your conf Directory.

This allows for flexibility in the actual config files in your config tree.

The changes Brian has been working on are based on just changing the config files. The config handling code in FreeSWITCH doesn’t require changes for this. As such, this does not affect someone copying their old configs into a master/1.8 install and just using them.

Any changes to actual config directives (which at this point I don’t think anything has actually changed) would be properly documented.

When it comes down to it the example configs we ship with FreeSWITCH, it's just an example of 1 way to do it.

Now back to the config changes Brian has been working on, As most of you are aware the FreeSWITCH team is quite small, and a project such as this requires many many person hours of work to maintain and continue adding new features. Brian put out a call for help from the community on this so that stake holders could help implement these changes and give feedback on them. However I don’t think he's received that much help on them and as such his other duties have slowed that project drastically. If you wish to help you can fork the git repo located at https://freeswitch.org/stash/projects/FS/repos/fs18configs/browse and propose pull requests.


-----Original Message-----
From: Nathan Neulinger [mailto:nneul at mst.edu <mailto:nneul at mst.edu> ]
Sent: Friday, July 14, 2017 9:16 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >; Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >
Subject: Re: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8" upgrade notes document?

That's good.

I thought I remembered seeing something several months back about large structural changes in the config files/layout/etc. Maybe I misread something.

Will definitely give new branch a try soon.

-- Nathan

On 7/14/17 9:09 AM, Ken Rice wrote:
> There will probably be a Wiki page for this, however, I doubt there will be much on it. In reality, I don’t think there's any config breaking changes from 1.6 to master at this point, and master is already being staged for early 1.8 releases. We'll prob see beta releases of 1.8 very soon. If you want to help us on this effort, please checkout the master branch (packages are already available see confluence) and test away.
>
> Ken
>
> -----Original Message-----
> From: FreeSWITCH-users
> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of
> Nathan Neulinger
> Sent: Thursday, July 13, 2017 2:43 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
> Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8" upgrade notes document?
>
> I remember moving from 1.4 to 1.6 there were several breaking behavior changes. Are you planning on doing anything to document specific known "you have to change X to Y" and "this behavior has changed" sort of document for sites upgrading from 1.6.x to 1.8.x?
>
> -- Nathan
>
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu <mailto:nneul at mst.edu> 
> Missouri S&T Information Technology    (573) 612-1412 <tel:%28573%29%20612-1412> 
> System Administrator - Architect
>
> ______________________________________________________________________

> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org> 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs

> http://www.freeswitch.org
>

--
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu <mailto:nneul at mst.edu> 
Missouri S&T Information Technology    (573) 612-1412 <tel:%28573%29%20612-1412> 
System Administrator - Architect


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

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

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





 

-- 

  <http://hawkinsegroup.com/Corporate/img/logo_left_top.png> 

 

Sincerely,
Don Hawkins
CEO
Hawkins Enterprise Group LLC
http://corporate.hawkinsegroup.com
Zello PTT <http://zello.com> : push2don

P: 469-214-5044

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From mike at jerris.com  Fri Jul 14 19:48:25 2017
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Jul 2017 15:48:25 -0400
Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8"
 upgrade notes document?
In-Reply-To: <CAO6gFMwGSf0mKeVyPSAJ0vUxHyYb37M602yUyHMFyo0Y-6hRUQ@mail.gmail.com>
References: <55259459-7390-1fe4-c6a1-d5ba03acac1b@mst.edu>
 <0e2801d2fcaa$d3031480$79093d80$@freeswitch.org>
 <40c92e39-896b-9427-9d3e-ec5695ecce21@mst.edu>
 <115901d2fcc2$e170b280$a4521780$@freeswitch.org>
 <CAO6gFMwGSf0mKeVyPSAJ0vUxHyYb37M602yUyHMFyo0Y-6hRUQ@mail.gmail.com>
Message-ID: <6668C984-17C4-48A6-A155-39E375742C39@jerris.com>

there isn’t a changeling done for 1.8 yet (will have one when we release) …. for 1.6.19 its here:
https://freeswitch.org/the-freeswitch-1619-release-is-here/ <https://freeswitch.org/the-freeswitch-1619-release-is-here/>



> On Jul 14, 2017, at 3:34 PM, Don Hawkins <hawkins at hawkinsegroup.com> wrote:
> 
> Sorry if I'm off subject here a bit, where can I find the change log for the latest version of FS?
> 
> On Fri, Jul 14, 2017 at 12:01 PM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
> The layout of the config files on the filesystem are not set in stone... The only thing set in stone is the base freeswitch.xml in your conf Directory.
> 
> This allows for flexibility in the actual config files in your config tree.
> 
> The changes Brian has been working on are based on just changing the config files. The config handling code in FreeSWITCH doesn’t require changes for this. As such, this does not affect someone copying their old configs into a master/1.8 install and just using them.
> 
> Any changes to actual config directives (which at this point I don’t think anything has actually changed) would be properly documented.
> 
> When it comes down to it the example configs we ship with FreeSWITCH, it's just an example of 1 way to do it.
> 
> Now back to the config changes Brian has been working on, As most of you are aware the FreeSWITCH team is quite small, and a project such as this requires many many person hours of work to maintain and continue adding new features. Brian put out a call for help from the community on this so that stake holders could help implement these changes and give feedback on them. However I don’t think he's received that much help on them and as such his other duties have slowed that project drastically. If you wish to help you can fork the git repo located at https://freeswitch.org/stash/projects/FS/repos/fs18configs/browse <https://freeswitch.org/stash/projects/FS/repos/fs18configs/browse> and propose pull requests.
> 
> -----Original Message-----
> From: Nathan Neulinger [mailto:nneul at mst.edu <mailto:nneul at mst.edu>]
> Sent: Friday, July 14, 2017 9:16 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>; Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>>
> Subject: Re: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8" upgrade notes document?
> 
> That's good.
> 
> I thought I remembered seeing something several months back about large structural changes in the config files/layout/etc. Maybe I misread something.
> 
> Will definitely give new branch a try soon.
> 
> -- Nathan
> 
> On 7/14/17 9:09 AM, Ken Rice wrote:
> > There will probably be a Wiki page for this, however, I doubt there will be much on it. In reality, I don’t think there's any config breaking changes from 1.6 to master at this point, and master is already being staged for early 1.8 releases. We'll prob see beta releases of 1.8 very soon. If you want to help us on this effort, please checkout the master branch (packages are already available see confluence) and test away.
> >
> > Ken
> >
> > -----Original Message-----
> > From: FreeSWITCH-users
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of
> > Nathan Neulinger
> > Sent: Thursday, July 13, 2017 2:43 PM
> > To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> > Subject: [Freeswitch-users] Are you planning on creating a "1.6 -> 1.8" upgrade notes document?
> >
> > I remember moving from 1.4 to 1.6 there were several breaking behavior changes. Are you planning on doing anything to document specific known "you have to change X to Y" and "this behavior has changed" sort of document for sites upgrading from 1.6.x to 1.8.x?
> >
> > -- Nathan
> >
> > ------------------------------------------------------------
> > Nathan Neulinger                       nneul at mst.edu <mailto:nneul at mst.edu>
> > Missouri S&T Information Technology    (573) 612-1412 <tel:%28573%29%20612-1412>
> > System Administrator - Architect
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> > http://www.cluecon.com <http://www.cluecon.com/>
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use <http://lists.freeswitch.org/mailman/options/freeswitch-use>
> > rs
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> >
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> > http://www.cluecon.com <http://www.cluecon.com/>
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use <http://lists.freeswitch.org/mailman/options/freeswitch-use>
> > rs
> > http://www.freeswitch.org <http://www.freeswitch.org/>
> >
> 
> --
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu <mailto:nneul at mst.edu>
> Missouri S&T Information Technology    (573) 612-1412 <tel:%28573%29%20612-1412>
> System Administrator - Architect
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> -- 
> 
> 
> Sincerely,
> Don Hawkins
> CEO
> Hawkins Enterprise Group LLC
> http://corporate.hawkinsegroup.com <http://corporate.hawkinsegroup.com/>
> Zello PTT <http://zello.com/>: push2don
> P: 469-214-5044
>  <>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From matt.nichols at westtel.com  Fri Jul 14 15:33:44 2017
From: matt.nichols at westtel.com (Matthew Nichols)
Date: Fri, 14 Jul 2017 15:33:44 +0000
Subject: [Freeswitch-users] FreeSWITCH HA and BDR, wiki
In-Reply-To: <9291D21A-A720-44FE-B95E-D09EDDCB12BD@tm.net.uk>
References: <DM5PR1001MB21554F95B2AC0DFB15CC6AF883AC0@DM5PR1001MB2155.namprd10.prod.outlook.com>
 <9291D21A-A720-44FE-B95E-D09EDDCB12BD@tm.net.uk>
Message-ID: <DM5PR1001MB21553C6ED4E4D300232F78A283AD0@DM5PR1001MB2155.namprd10.prod.outlook.com>

This is most of what I’ve found out so far.

I will be doing much more thorough tests next week, so I’ll let you know.

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joseph Waite
Sent: Friday, July 14, 2017 3:05 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSWITCH HA and BDR, wiki

Hi Matthew

I'm about to attempt the same. And notes/corrected db schemes you got would be much appreciated!

Joe Waite

On 13 Jul 2017, at 17:25, Matthew Nichols <matt.nichols at westtel.com<mailto:matt.nichols at westtel.com>> wrote:
After going through https://freeswitch.org/confluence/display/FREESWITCH/High+Availability, it appears out of date/misleading. Here are the things I have found:

Switchname needs to be different or they end up clobbering each other. Call recovery still works.

$${domain} needs to be set the same on both boxes for voicemails and registrations to work well after recovery.

As far as I can tell the entirety of /var/lib/freeswitch can be shared with something like GlusterFS (assuming sqlite is not being used).

Especially with PostgreSQL BDR auto-create-sql needs to be false and auto-clear-sql needs to be true, otherwise information is never cleared after a crash.

Also I see no documentation on the wiki on actually using FreeSWITCH with PostgreSQL BDR. I would be willing to start a page with what I have discovered so far. The only online documentation I have found is https://gist.github.com/DigiDaz/1cfe3d5d32080a8e3d75a20bb5bc4fb5, which doesn’t have the primary keys set up quite right (calls and channels need to add hostname to the primary key or call recovery doesn’t work quite right) and doesn’t include voicemail.

I am very willing to setup the required edits but I don’t have write access so I would need to go through one of you who does.

Matthew Nichols

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users>
http://www.freeswitch.org
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From s.safarov at gmail.com  Sat Jul 15 09:02:15 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 15 Jul 2017 09:02:15 +0000
Subject: [Freeswitch-users] how to send simpe message
Message-ID: <CAHtxdDcfnZjOYop+XAJEsmoOPAHSsEy_E0vakEZ-x6beoH5b2A@mail.gmail.com>

Could you advice how to send simple message?
As I knows this may be done only from lua or esl script
https://freeswitch.org/confluence/display/FREESWITCH/mod_sms#mod_sms-SendingaMessagefromascript

But provided example can send only to local endpoin
When i try send message to remote host thne got error like this

2017-07-15 08:59:46.408247 [INFO] switch_cpp.cpp:1377 chat console
2017-07-15 08:59:46.408247 [ERR] sofia_presence.c:200 Chat proto [sip]
from [sip:1004 at my.local.host]
to [1019 at my.remote.host]
Hello from Seven Du! Have fun!
Invalid Profile my.remote.host

Please advice how to specify profile name in event?
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From gmaruzz at gmail.com  Sat Jul 15 09:14:59 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 15 Jul 2017 11:14:59 +0200
Subject: [Freeswitch-users] how to send simpe message
In-Reply-To: <CAHtxdDcfnZjOYop+XAJEsmoOPAHSsEy_E0vakEZ-x6beoH5b2A@mail.gmail.com>
References: <CAHtxdDcfnZjOYop+XAJEsmoOPAHSsEy_E0vakEZ-x6beoH5b2A@mail.gmail.com>
Message-ID: <CALXCt0rkuiPmqGXtaCyG66t1wjQ5AAsLhWD=_Zc0uJ07Pc3ONQ@mail.gmail.com>

Hello Sergey,

use the "chat" api command or dialplan app:

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



On 15 July 2017 at 11:02, Sergey Safarov <s.safarov at gmail.com> wrote:

> Could you advice how to send simple message?
> As I knows this may be done only from lua or esl script
> https://freeswitch.org/confluence/display/FREESWITCH/mod_
> sms#mod_sms-SendingaMessagefromascript
>
> But provided example can send only to local endpoin
> When i try send message to remote host thne got error like this
>
> 2017-07-15 08:59:46.408247 [INFO] switch_cpp.cpp:1377 chat console
> 2017-07-15 08:59:46.408247 [ERR] sofia_presence.c:200 Chat proto [sip]
> from [sip:1004 at my.local.host]
> to [1019 at my.remote.host]
> Hello from Seven Du! Have fun!
> Invalid Profile my.remote.host
>
> Please advice how to specify profile name in event?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From andretodd at verizon.net  Sat Jul 15 13:46:13 2017
From: andretodd at verizon.net (Andre Demattia)
Date: Sat, 15 Jul 2017 09:46:13 -0400
Subject: [Freeswitch-users] Channel Var Codec
In-Reply-To: <019701d2fc13$8d12e690$a738b3b0$@verizon.net>
References: <019701d2fc13$8d12e690$a738b3b0$@verizon.net>
Message-ID: <mailman.0.1500126384.1294.freeswitch-users@lists.freeswitch.org>

I see variable_switch_r_sdp has the codec values.
I just want to verify the correct one.

Thanks
Andre

Sent from my Windows 10 phone

From: Andre DeMattia
Sent: Thursday, July 13, 2017 4:06 PM
To: 'FreeSWITCH Users Help'
Subject: [Freeswitch-users] Channel Var Codec

 Hi what Channel variable should I use to see what Codec the A leg allows?

variable_rtp_use_codec_string or variable_ep_codec_string or something else?

In the CDR I see this
<rtp_use_codec_string>G722,PCMU,PCMA,GSM</rtp_use_codec_string>

This looks like what I'm looking for but I need it in the dialplan before the call goes out so I can direct the call where I need it to go.
Thanks


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

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

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

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From s.safarov at gmail.com  Sat Jul 15 19:33:17 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 15 Jul 2017 19:33:17 +0000
Subject: [Freeswitch-users] how to send simpe message
In-Reply-To: <CALXCt0rkuiPmqGXtaCyG66t1wjQ5AAsLhWD=_Zc0uJ07Pc3ONQ@mail.gmail.com>
References: <CAHtxdDcfnZjOYop+XAJEsmoOPAHSsEy_E0vakEZ-x6beoH5b2A@mail.gmail.com>
 <CALXCt0rkuiPmqGXtaCyG66t1wjQ5AAsLhWD=_Zc0uJ07Pc3ONQ@mail.gmail.com>
Message-ID: <CAHtxdDck7z+L6B3LZMkZ5dGHqti9=+z5gFwgdsqdC8Bs+acRpQ@mail.gmail.com>

Thank you Giovanni
I tested but got similar error

freeswitch at ip-10-0-0-245> chat sip|1001 at my.local.host|1002 at my.remote.host|Hello
chat via SIP!
Error! Message Not Sent
2017-07-15 19:31:46.048232 [ERR] sofia_presence.c:200 Chat proto [global]
from [1001 at my.local.host]
to [1002 at my.remote.host]
Hello chat via SIP!
Invalid Profile my.remote.host

Sergey

сб, 15 июл. 2017 г. в 12:20, Giovanni Maruzzelli <gmaruzz at gmail.com>:

> Hello Sergey,
>
> use the "chat" api command or dialplan app:
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+chat
>
>
>
> On 15 July 2017 at 11:02, Sergey Safarov <s.safarov at gmail.com> wrote:
>
>> Could you advice how to send simple message?
>> As I knows this may be done only from lua or esl script
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_sms#mod_sms-SendingaMessagefromascript
>>
>> But provided example can send only to local endpoin
>> When i try send message to remote host thne got error like this
>>
>> 2017-07-15 08:59:46.408247 [INFO] switch_cpp.cpp:1377 chat console
>> 2017-07-15 08:59:46.408247 [ERR] sofia_presence.c:200 Chat proto [sip]
>> from [sip:1004 at my.local.host]
>> to [1019 at my.remote.host]
>> Hello from Seven Du! Have fun!
>> Invalid Profile my.remote.host
>>
>> Please advice how to specify profile name in event?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From s.safarov at gmail.com  Sat Jul 15 19:54:05 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 15 Jul 2017 19:54:05 +0000
Subject: [Freeswitch-users] how to send simpe message
In-Reply-To: <CAHtxdDck7z+L6B3LZMkZ5dGHqti9=+z5gFwgdsqdC8Bs+acRpQ@mail.gmail.com>
References: <CAHtxdDcfnZjOYop+XAJEsmoOPAHSsEy_E0vakEZ-x6beoH5b2A@mail.gmail.com>
 <CALXCt0rkuiPmqGXtaCyG66t1wjQ5AAsLhWD=_Zc0uJ07Pc3ONQ@mail.gmail.com>
 <CAHtxdDck7z+L6B3LZMkZ5dGHqti9=+z5gFwgdsqdC8Bs+acRpQ@mail.gmail.com>
Message-ID: <CAHtxdDevP7tmVvC52tOXTY0x1gAByfp5HGtNcxYhE9q5abEJ4A@mail.gmail.com>

As i see profile added as in this example
https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/endpoints/mod_sofia/sofia_presence.c#4917

But here i see using chat API call i cannot specify event header.
https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/applications/mod_dptools/mod_dptools.c#4563-4569,4594
may be required update api call or directly create appropriate event

Sergey.

сб, 15 июл. 2017 г. в 22:33, Sergey Safarov <s.safarov at gmail.com>:

> Thank you Giovanni
> I tested but got similar error
>
> freeswitch at ip-10-0-0-245> chat sip|1001 at my.local.host|1002 at my.remote.host|Hello
> chat via SIP!
> Error! Message Not Sent
> 2017-07-15 19:31:46.048232 [ERR] sofia_presence.c:200 Chat proto [global]
> from [1001 at my.local.host]
> to [1002 at my.remote.host]
> Hello chat via SIP!
> Invalid Profile my.remote.host
>
> Sergey
>
> сб, 15 июл. 2017 г. в 12:20, Giovanni Maruzzelli <gmaruzz at gmail.com>:
>
>> Hello Sergey,
>>
>> use the "chat" api command or dialplan app:
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+chat
>>
>>
>>
>> On 15 July 2017 at 11:02, Sergey Safarov <s.safarov at gmail.com> wrote:
>>
>>> Could you advice how to send simple message?
>>> As I knows this may be done only from lua or esl script
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_sms#mod_sms-SendingaMessagefromascript
>>>
>>> But provided example can send only to local endpoin
>>> When i try send message to remote host thne got error like this
>>>
>>> 2017-07-15 08:59:46.408247 [INFO] switch_cpp.cpp:1377 chat console
>>> 2017-07-15 08:59:46.408247 [ERR] sofia_presence.c:200 Chat proto [sip]
>>> from [sip:1004 at my.local.host]
>>> to [1019 at my.remote.host]
>>> Hello from Seven Du! Have fun!
>>> Invalid Profile my.remote.host
>>>
>>> Please advice how to specify profile name in event?
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Sat Jul 15 20:14:48 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 15 Jul 2017 15:14:48 -0500
Subject: [Freeswitch-users] FreeSWITCH HA and BDR, wiki
In-Reply-To: <DM5PR1001MB21553C6ED4E4D300232F78A283AD0@DM5PR1001MB2155.namprd10.prod.outlook.com>
References: <DM5PR1001MB21554F95B2AC0DFB15CC6AF883AC0@DM5PR1001MB2155.namprd10.prod.outlook.com>
 <9291D21A-A720-44FE-B95E-D09EDDCB12BD@tm.net.uk>
 <DM5PR1001MB21553C6ED4E4D300232F78A283AD0@DM5PR1001MB2155.namprd10.prod.outlook.com>
Message-ID: <CAKbxfG_s8CSjKP0uxZntKnQ2tASb9ZGN7HZS8Ei3ZGn+_u5wNw@mail.gmail.com>

There is actually a docs team you could join.  There is a dedicated hipchat
room, and a mailing list for it.



On Fri, Jul 14, 2017 at 10:33 AM, Matthew Nichols <matt.nichols at westtel.com>
wrote:

> This is most of what I’ve found out so far.
>
>
>
> I will be doing much more thorough tests next week, so I’ll let you know.
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Joseph Waite
> *Sent:* Friday, July 14, 2017 3:05 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FreeSWITCH HA and BDR, wiki
>
>
>
> Hi Matthew
>
>
>
> I'm about to attempt the same. And notes/corrected db schemes you got
> would be much appreciated!
>
> Joe Waite
>
>
> On 13 Jul 2017, at 17:25, Matthew Nichols <matt.nichols at westtel.com>
> wrote:
>
> After going through https://freeswitch.org/confluence/display/FREESWITCH/
> High+Availability, it appears out of date/misleading. Here are the things
> I have found:
>
>
>
> Switchname needs to be different or they end up clobbering each other.
> Call recovery still works.
>
>
>
> $${domain} needs to be set the same on both boxes for voicemails and
> registrations to work well after recovery.
>
>
>
> As far as I can tell the entirety of /var/lib/freeswitch can be shared
> with something like GlusterFS (assuming sqlite is not being used).
>
>
>
> Especially with PostgreSQL BDR auto-create-sql needs to be false and
> auto-clear-sql needs to be true, otherwise information is never cleared
> after a crash.
>
>
>
> Also I see no documentation on the wiki on actually using FreeSWITCH with
> PostgreSQL BDR. I would be willing to start a page with what I have
> discovered so far. The only online documentation I have found is
> https://gist.github.com/DigiDaz/1cfe3d5d32080a8e3d75a20bb5bc4fb5, which
> doesn’t have the primary keys set up quite right (calls and channels need
> to add hostname to the primary key or call recovery doesn’t work quite
> right) and doesn’t include voicemail.
>
>
>
> I am very willing to setup the required edits but I don’t have write
> access so I would need to go through one of you who does.
>
>
>
> Matthew Nichols
>
>
>
> <Matthew Nichols.vcf>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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

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

https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From s.safarov at gmail.com  Sat Jul 15 20:15:38 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 15 Jul 2017 20:15:38 +0000
Subject: [Freeswitch-users] how to send simpe message
In-Reply-To: <CAHtxdDevP7tmVvC52tOXTY0x1gAByfp5HGtNcxYhE9q5abEJ4A@mail.gmail.com>
References: <CAHtxdDcfnZjOYop+XAJEsmoOPAHSsEy_E0vakEZ-x6beoH5b2A@mail.gmail.com>
 <CALXCt0rkuiPmqGXtaCyG66t1wjQ5AAsLhWD=_Zc0uJ07Pc3ONQ@mail.gmail.com>
 <CAHtxdDck7z+L6B3LZMkZ5dGHqti9=+z5gFwgdsqdC8Bs+acRpQ@mail.gmail.com>
 <CAHtxdDevP7tmVvC52tOXTY0x1gAByfp5HGtNcxYhE9q5abEJ4A@mail.gmail.com>
Message-ID: <CAHtxdDfwYE-88tHQYDy4LJSqe1JoRSXMaCfNh_MOa=yQsO1gFg@mail.gmail.com>

I added "event:addHeader("sip_profile", "external");" to lua script now
profile specified but FS still wants send message locally.

freeswitch at ip-10-0-0-245> luarun forward-message.lua
+OK
2017-07-15 20:13:44.978212 [INFO] switch_cpp.cpp:1377 chat console
2017-07-15 20:13:44.978212 [DEBUG] sofia_presence.c:225 Can't find
registered user 1002 at my.remote.host
2017-07-15 20:13:44.978212 [DEBUG] mod_sms.c:92 SMS Delivery assumed
successful due to being sent in non-blocking manner
freeswitch at ip-10-0-0-245>

Could you advice how specify send message to remote host?



сб, 15 июл. 2017 г. в 22:54, Sergey Safarov <s.safarov at gmail.com>:

> As i see profile added as in this example
>
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/endpoints/mod_sofia/sofia_presence.c#4917
>
> But here i see using chat API call i cannot specify event header.
>
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/applications/mod_dptools/mod_dptools.c#4563-4569,4594
> may be required update api call or directly create appropriate event
>
> Sergey.
>
> сб, 15 июл. 2017 г. в 22:33, Sergey Safarov <s.safarov at gmail.com>:
>
>> Thank you Giovanni
>> I tested but got similar error
>>
>> freeswitch at ip-10-0-0-245> chat sip|1001 at my.local.host|1002 at my.remote.host|Hello
>> chat via SIP!
>> Error! Message Not Sent
>> 2017-07-15 19:31:46.048232 [ERR] sofia_presence.c:200 Chat proto [global]
>> from [1001 at my.local.host]
>> to [1002 at my.remote.host]
>> Hello chat via SIP!
>> Invalid Profile my.remote.host
>>
>> Sergey
>>
>> сб, 15 июл. 2017 г. в 12:20, Giovanni Maruzzelli <gmaruzz at gmail.com>:
>>
>>> Hello Sergey,
>>>
>>> use the "chat" api command or dialplan app:
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+chat
>>>
>>>
>>>
>>> On 15 July 2017 at 11:02, Sergey Safarov <s.safarov at gmail.com> wrote:
>>>
>>>> Could you advice how to send simple message?
>>>> As I knows this may be done only from lua or esl script
>>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_sms#mod_sms-SendingaMessagefromascript
>>>>
>>>> But provided example can send only to local endpoin
>>>> When i try send message to remote host thne got error like this
>>>>
>>>> 2017-07-15 08:59:46.408247 [INFO] switch_cpp.cpp:1377 chat console
>>>> 2017-07-15 08:59:46.408247 [ERR] sofia_presence.c:200 Chat proto [sip]
>>>> from [sip:1004 at my.local.host]
>>>> to [1019 at my.remote.host]
>>>> Hello from Seven Du! Have fun!
>>>> Invalid Profile my.remote.host
>>>>
>>>> Please advice how to specify profile name in event?
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> cell: +39 347 266 56 18
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
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From david.villasmil.work at gmail.com  Sat Jul 15 20:28:47 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 15 Jul 2017 20:28:47 +0000
Subject: [Freeswitch-users] how to send simpe message
In-Reply-To: <CAHtxdDfwYE-88tHQYDy4LJSqe1JoRSXMaCfNh_MOa=yQsO1gFg@mail.gmail.com>
References: <CAHtxdDcfnZjOYop+XAJEsmoOPAHSsEy_E0vakEZ-x6beoH5b2A@mail.gmail.com>
 <CALXCt0rkuiPmqGXtaCyG66t1wjQ5AAsLhWD=_Zc0uJ07Pc3ONQ@mail.gmail.com>
 <CAHtxdDck7z+L6B3LZMkZ5dGHqti9=+z5gFwgdsqdC8Bs+acRpQ@mail.gmail.com>
 <CAHtxdDevP7tmVvC52tOXTY0x1gAByfp5HGtNcxYhE9q5abEJ4A@mail.gmail.com>
 <CAHtxdDfwYE-88tHQYDy4LJSqe1JoRSXMaCfNh_MOa=yQsO1gFg@mail.gmail.com>
Message-ID: <CAFGRPVrFMeP3mAJErd63DqXAimpCSYw6O9V_UFTWGFZ+GO_63A@mail.gmail.com>

Not sure, but i don't think you can send messages off-site. It designed to
send messages to registered contacts, afaik
On Sat, Jul 15, 2017 at 10:16 PM Sergey Safarov <s.safarov at gmail.com> wrote:

> I added "event:addHeader("sip_profile", "external");" to lua script now
> profile specified but FS still wants send message locally.
>
> freeswitch at ip-10-0-0-245> luarun forward-message.lua
> +OK
> 2017-07-15 20:13:44.978212 [INFO] switch_cpp.cpp:1377 chat console
> 2017-07-15 20:13:44.978212 [DEBUG] sofia_presence.c:225 Can't find
> registered user 1002 at my.remote.host
> 2017-07-15 20:13:44.978212 [DEBUG] mod_sms.c:92 SMS Delivery assumed
> successful due to being sent in non-blocking manner
> freeswitch at ip-10-0-0-245>
>
> Could you advice how specify send message to remote host?
>
>
>
> сб, 15 июл. 2017 г. в 22:54, Sergey Safarov <s.safarov at gmail.com>:
>
>> As i see profile added as in this example
>>
>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/endpoints/mod_sofia/sofia_presence.c#4917
>>
>> But here i see using chat API call i cannot specify event header.
>>
>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/applications/mod_dptools/mod_dptools.c#4563-4569,4594
>> may be required update api call or directly create appropriate event
>>
>> Sergey.
>>
>> сб, 15 июл. 2017 г. в 22:33, Sergey Safarov <s.safarov at gmail.com>:
>>
>>> Thank you Giovanni
>>> I tested but got similar error
>>>
>>> freeswitch at ip-10-0-0-245> chat sip|1001 at my.local.host
>>> |1002 at my.remote.host|Hello chat via SIP!
>>> Error! Message Not Sent
>>> 2017-07-15 19:31:46.048232 [ERR] sofia_presence.c:200 Chat proto [global]
>>> from [1001 at my.local.host]
>>> to [1002 at my.remote.host]
>>> Hello chat via SIP!
>>> Invalid Profile my.remote.host
>>>
>>> Sergey
>>>
>>> сб, 15 июл. 2017 г. в 12:20, Giovanni Maruzzelli <gmaruzz at gmail.com>:
>>>
>>>> Hello Sergey,
>>>>
>>>> use the "chat" api command or dialplan app:
>>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+chat
>>>>
>>>>
>>>>
>>>> On 15 July 2017 at 11:02, Sergey Safarov <s.safarov at gmail.com> wrote:
>>>>
>>>>> Could you advice how to send simple message?
>>>>> As I knows this may be done only from lua or esl script
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_sms#mod_sms-SendingaMessagefromascript
>>>>>
>>>>> But provided example can send only to local endpoin
>>>>> When i try send message to remote host thne got error like this
>>>>>
>>>>> 2017-07-15 08:59:46.408247 [INFO] switch_cpp.cpp:1377 chat console
>>>>> 2017-07-15 08:59:46.408247 [ERR] sofia_presence.c:200 Chat proto [sip]
>>>>> from [sip:1004 at my.local.host]
>>>>> to [1019 at my.remote.host]
>>>>> Hello from Seven Du! Have fun!
>>>>> Invalid Profile my.remote.host
>>>>>
>>>>> Please advice how to specify profile name in event?
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From s.safarov at gmail.com  Sat Jul 15 21:01:24 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 15 Jul 2017 21:01:24 +0000
Subject: [Freeswitch-users] how to send simpe message
In-Reply-To: <CAFGRPVrFMeP3mAJErd63DqXAimpCSYw6O9V_UFTWGFZ+GO_63A@mail.gmail.com>
References: <CAHtxdDcfnZjOYop+XAJEsmoOPAHSsEy_E0vakEZ-x6beoH5b2A@mail.gmail.com>
 <CALXCt0rkuiPmqGXtaCyG66t1wjQ5AAsLhWD=_Zc0uJ07Pc3ONQ@mail.gmail.com>
 <CAHtxdDck7z+L6B3LZMkZ5dGHqti9=+z5gFwgdsqdC8Bs+acRpQ@mail.gmail.com>
 <CAHtxdDevP7tmVvC52tOXTY0x1gAByfp5HGtNcxYhE9q5abEJ4A@mail.gmail.com>
 <CAHtxdDfwYE-88tHQYDy4LJSqe1JoRSXMaCfNh_MOa=yQsO1gFg@mail.gmail.com>
 <CAFGRPVrFMeP3mAJErd63DqXAimpCSYw6O9V_UFTWGFZ+GO_63A@mail.gmail.com>
Message-ID: <CAHtxdDf2N8NG1Adac90LSELzwW4PQZR3kjdpvTehOKSRxF=bzA@mail.gmail.com>

I found working arguments. Updated lua script
<https://freeswitch.org/confluence/display/FREESWITCH/mod_sms#mod_sms-SendingaMessagefromascript>

сб, 15 июл. 2017 г. в 23:31, David Villasmil <david.villasmil.work at gmail.com
>:

> Not sure, but i don't think you can send messages off-site. It designed to
> send messages to registered contacts, afaik
> On Sat, Jul 15, 2017 at 10:16 PM Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> I added "event:addHeader("sip_profile", "external");" to lua script now
>> profile specified but FS still wants send message locally.
>>
>> freeswitch at ip-10-0-0-245> luarun forward-message.lua
>> +OK
>> 2017-07-15 20:13:44.978212 [INFO] switch_cpp.cpp:1377 chat console
>> 2017-07-15 20:13:44.978212 [DEBUG] sofia_presence.c:225 Can't find
>> registered user 1002 at my.remote.host
>> 2017-07-15 20:13:44.978212 [DEBUG] mod_sms.c:92 SMS Delivery assumed
>> successful due to being sent in non-blocking manner
>> freeswitch at ip-10-0-0-245>
>>
>> Could you advice how specify send message to remote host?
>>
>>
>>
>> сб, 15 июл. 2017 г. в 22:54, Sergey Safarov <s.safarov at gmail.com>:
>>
>>> As i see profile added as in this example
>>>
>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/endpoints/mod_sofia/sofia_presence.c#4917
>>>
>>> But here i see using chat API call i cannot specify event header.
>>>
>>> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/applications/mod_dptools/mod_dptools.c#4563-4569,4594
>>> may be required update api call or directly create appropriate event
>>>
>>> Sergey.
>>>
>>> сб, 15 июл. 2017 г. в 22:33, Sergey Safarov <s.safarov at gmail.com>:
>>>
>>>> Thank you Giovanni
>>>> I tested but got similar error
>>>>
>>>> freeswitch at ip-10-0-0-245> chat sip|1001 at my.local.host
>>>> |1002 at my.remote.host|Hello chat via SIP!
>>>> Error! Message Not Sent
>>>> 2017-07-15 19:31:46.048232 [ERR] sofia_presence.c:200 Chat proto
>>>> [global]
>>>> from [1001 at my.local.host]
>>>> to [1002 at my.remote.host]
>>>> Hello chat via SIP!
>>>> Invalid Profile my.remote.host
>>>>
>>>> Sergey
>>>>
>>>> сб, 15 июл. 2017 г. в 12:20, Giovanni Maruzzelli <gmaruzz at gmail.com>:
>>>>
>>>>> Hello Sergey,
>>>>>
>>>>> use the "chat" api command or dialplan app:
>>>>>
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+chat
>>>>>
>>>>>
>>>>>
>>>>> On 15 July 2017 at 11:02, Sergey Safarov <s.safarov at gmail.com> wrote:
>>>>>
>>>>>> Could you advice how to send simple message?
>>>>>> As I knows this may be done only from lua or esl script
>>>>>>
>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_sms#mod_sms-SendingaMessagefromascript
>>>>>>
>>>>>> But provided example can send only to local endpoin
>>>>>> When i try send message to remote host thne got error like this
>>>>>>
>>>>>> 2017-07-15 08:59:46.408247 [INFO] switch_cpp.cpp:1377 chat console
>>>>>> 2017-07-15 08:59:46.408247 [ERR] sofia_presence.c:200 Chat proto [sip]
>>>>>> from [sip:1004 at my.local.host]
>>>>>> to [1019 at my.remote.host]
>>>>>> Hello from Seven Du! Have fun!
>>>>>> Invalid Profile my.remote.host
>>>>>>
>>>>>> Please advice how to specify profile name in event?
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> OpenTelecom.IT
>>>>> cell: +39 347 266 56 18
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From joelists at tm.net.uk  Sun Jul 16 15:15:58 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Sun, 16 Jul 2017 16:15:58 +0100
Subject: [Freeswitch-users] Freeswitch.org Website Down
Message-ID: <A379B6C8-2D73-48EA-8680-7C87BACF7987@tm.net.uk>

Is the freeswitch.org site including the wiki down for anyone else?

Regards


From rtreleaven at bunnykick.ca  Sun Jul 16 18:54:58 2017
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Sun, 16 Jul 2017 14:54:58 -0400
Subject: [Freeswitch-users] Freeswitch.org Website Down
In-Reply-To: <A379B6C8-2D73-48EA-8680-7C87BACF7987@tm.net.uk>
References: <A379B6C8-2D73-48EA-8680-7C87BACF7987@tm.net.uk>
Message-ID: <CAHzfiZ6mBWJR8-6aKTNSm5BwJxPcpwCFTUWgLywG5L4OsNiEqA@mail.gmail.com>

working for me

On Sun, Jul 16, 2017 at 11:15 AM, Joseph Waite <joelists at tm.net.uk> wrote:

> Is the freeswitch.org site including the wiki down for anyone else?
>
> Regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 
Sincerely,

Russell Treleaven
sip:rtreleaven at sip.bunnykick.ca
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From joel at gogii.net  Sun Jul 16 14:40:47 2017
From: joel at gogii.net (Joel Serrano)
Date: Sun, 16 Jul 2017 07:40:47 -0700
Subject: [Freeswitch-users] Can't connect to freeswitch.org, any issues??
Message-ID: <CAMtXxQmFhj4e1T+748Wv-gV92MTmZxHs3OQX4yJMgET_ygoB3w@mail.gmail.com>

Hi guys,

Don't know if it's just me, but I cannot reach freeswitch.org via ipv4/ipv6
or http/https. (I tried from multiple different sources, all fail)

Is there any maintenance or anything going on?

Thanks!
Joel.
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From joelists at tm.net.uk  Sun Jul 16 19:10:50 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Sun, 16 Jul 2017 20:10:50 +0100
Subject: [Freeswitch-users] Freeswitch.org Website Down
In-Reply-To: <CAHzfiZ6mBWJR8-6aKTNSm5BwJxPcpwCFTUWgLywG5L4OsNiEqA@mail.gmail.com>
References: <A379B6C8-2D73-48EA-8680-7C87BACF7987@tm.net.uk>
 <CAHzfiZ6mBWJR8-6aKTNSm5BwJxPcpwCFTUWgLywG5L4OsNiEqA@mail.gmail.com>
Message-ID: <E3F1D505-D01E-46C2-9132-C35D425792C1@tm.net.uk>

Its working for me now.

Interestingly, my email to the mailing list has only just appeared. I sent the email hours ago!

> On 16 Jul 2017, at 19:54, Russell Treleaven <rtreleaven at bunnykick.ca> wrote:
> 
> working for me
> 
> On Sun, Jul 16, 2017 at 11:15 AM, Joseph Waite <joelists at tm.net.uk <mailto:joelists at tm.net.uk>> wrote:
> Is the freeswitch.org <http://freeswitch.org/> site including the wiki down for anyone else?
> 
> Regards
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> -- 
> Sincerely,
> 
> Russell Treleaven
> sip:rtreleaven at sip.bunnykick.ca <mailto:sip%3Artreleaven at sip.bunnykick.ca>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From s.safarov at gmail.com  Sun Jul 16 19:16:17 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 16 Jul 2017 19:16:17 +0000
Subject: [Freeswitch-users] mod_sms: is possible to not forward message?
Message-ID: <CAHtxdDdOOnpt6UNJoUqb42j_zg7mPGr5ShELLdfkM-ybam-OLw@mail.gmail.com>

I created chatplan

<?xml version="1.0" encoding="utf-8"?>
<include>
  <context name="public">
    <extension name="demo">
      <condition field="to" expression="^(.*)$">
         <action application="stop" data=""/>
      </condition>
    </extension>
  </context>
</include>
Required simple accept message and do nothing. But really FreeSwitch send
forward message.
Tested current master and 1.5

Is it bug? Or i do some thing wrong?
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From mario_fs at mgtech.com  Sun Jul 16 19:10:00 2017
From: mario_fs at mgtech.com (Mario G)
Date: Sun, 16 Jul 2017 12:10:00 -0700
Subject: [Freeswitch-users] Freeswitch.org Website Down
In-Reply-To: <CAHzfiZ6mBWJR8-6aKTNSm5BwJxPcpwCFTUWgLywG5L4OsNiEqA@mail.gmail.com>
References: <A379B6C8-2D73-48EA-8680-7C87BACF7987@tm.net.uk>
 <CAHzfiZ6mBWJR8-6aKTNSm5BwJxPcpwCFTUWgLywG5L4OsNiEqA@mail.gmail.com>
Message-ID: <BB0260B1-F970-48F9-8E53-C121B2EFB584@mgtech.com>

Wiki, jira, etc. Was down this morning but is working fine now for me.

> On Jul 16, 2017, at 11:54 AM, Russell Treleaven <rtreleaven at bunnykick.ca> wrote:
> 
> working for me
> 
> On Sun, Jul 16, 2017 at 11:15 AM, Joseph Waite <joelists at tm.net.uk <mailto:joelists at tm.net.uk>> wrote:
> Is the freeswitch.org <http://freeswitch.org/> site including the wiki down for anyone else?
> 
> Regards
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> -- 
> Sincerely,
> 
> Russell Treleaven
> sip:rtreleaven at sip.bunnykick.ca <mailto:sip%3Artreleaven at sip.bunnykick.ca>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From davidwaf at gmail.com  Sun Jul 16 20:25:05 2017
From: davidwaf at gmail.com (David Wafula)
Date: Sun, 16 Jul 2017 22:25:05 +0200
Subject: [Freeswitch-users] Can't connect to freeswitch.org, any issues??
In-Reply-To: <CAMtXxQmFhj4e1T+748Wv-gV92MTmZxHs3OQX4yJMgET_ygoB3w@mail.gmail.com>
References: <CAMtXxQmFhj4e1T+748Wv-gV92MTmZxHs3OQX4yJMgET_ygoB3w@mail.gmail.com>
Message-ID: <CAK9Ws2eY+tYnxWAhEpYhZ+FQHq=CpECvp54dMA7H1sBwwsXE4w@mail.gmail.com>

Its all back online

On Sun, Jul 16, 2017 at 4:40 PM, Joel Serrano <joel at gogii.net> wrote:

> Hi guys,
>
> Don't know if it's just me, but I cannot reach freeswitch.org via
> ipv4/ipv6 or http/https. (I tried from multiple different sources, all fail)
>
> Is there any maintenance or anything going on?
>
> Thanks!
> Joel.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
David W
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From jungleboogie0 at gmail.com  Mon Jul 17 01:07:26 2017
From: jungleboogie0 at gmail.com (jungle boogie)
Date: Sun, 16 Jul 2017 18:07:26 -0700
Subject: [Freeswitch-users] secondary outbound gateway - how to prefix?
Message-ID: <83437836-5724-6f73-c174-508e89067b72@gmail.com>

Hi All,

I would like to have a secondary gateway added to freeswitch. To have it 
used for outbound calling, I need to prefix the dialstring with 
something like a 9 followed by the 10 digit number.

I think that's what this page is explaining:
https://freeswitch.org/confluence/display/FREESWITCH/Configuring+FreeSWITCH

<condition field="destination_number" expression="^9(1\d{10})$">

Will that use the gateway if the telephone number is prefixed with a 9?

Thanks!


From shaun.stokes at itec-support.co.uk  Mon Jul 17 05:22:08 2017
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Mon, 17 Jul 2017 05:22:08 +0000
Subject: [Freeswitch-users] secondary outbound gateway - how to prefix?
In-Reply-To: <83437836-5724-6f73-c174-508e89067b72@gmail.com>
References: <83437836-5724-6f73-c174-508e89067b72@gmail.com>
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E8681044@mbx-01.sysconfig.co.uk>

You've put a 1 before \d so the matched number should be 11 digits including the 1. This will match destination numbers prefixed with 9, the $1 variable will be the rest of the number after the 9 prefix.

-----Original Message-----
From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of jungle boogie
Sent: 17 July 2017 02:07
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] secondary outbound gateway - how to prefix?

Hi All,

I would like to have a secondary gateway added to freeswitch. To have it used for outbound calling, I need to prefix the dialstring with something like a 9 followed by the 10 digit number.

I think that's what this page is explaining:
https://freeswitch.org/confluence/display/FREESWITCH/Configuring+FreeSWITCH

<condition field="destination_number" expression="^9(1\d{10})$">

Will that use the gateway if the telephone number is prefixed with a 9?

Thanks!

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

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

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

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
______________________________________________________________________
[http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
Shaun Stokes - Infrastructure Analyst

T :     01453 700713
E :     shaun.stokes at itec-support.co.uk
W :     www.itec-support.co.uk

Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
Company No. 06908001

CONFIDENTIALITY NOTICE
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______________________________________________________________________
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______________________________________________________________________

From sean at missionlabs.co.uk  Mon Jul 17 09:33:53 2017
From: sean at missionlabs.co.uk (Sean Ingham)
Date: Mon, 17 Jul 2017 10:33:53 +0100
Subject: [Freeswitch-users] RTP Auto Switch behaviour on EC2
Message-ID: <CALhEHhUP7K26szNe2qxcQ98adOjgLp59whTG5Ogx8mjv2z4SWw@mail.gmail.com>

Hi,
I'm running FreeSwitch on an AWS EC2 with config as per the wiki EC2 page.

external_rtp_ip & bind_server_ip are both set to the box's public AWS
Elastic IP - as is ext_rtp_ip on all profiles.

I'm running a WebRTC gateway also in AWS. I can see that SDPs from
Freeswitch to the gateway all include Freeswitch's public IP as desired,
however when the call is answered Freeswitch always sends RTP traffic from
it's private IP. Then a second or 2 later Freeswitch sees incoming RTP from
the gateway to it's public IP address and RTP auto swtiching behaviour
kicks in, resulting in a line like this in FS logs:

switch_rtp.c:6954 Auto Changing audio port from 172.31.x.x:31196 to
52.17.x.x:31196

As it happens, the gateway is able to receive media from either private or
public IP as the box is in the same subnet, but in around 30% of cases at
the point this switching occurs the call audio drops. (I'm guessing the
gateway has issues stitching together the rtp streams from 2 different
sources.)

I've tried experimenting with the disable_rtp_auto_switch parameter, but
seems to me a cleaner solution would be to have Freeswitch send RTP from
it's public IP in the first place, and I can't understand why it's not
doing that given it's negotiated to use the public IP in it's SDP.

Can anyone provide any explanation for the behaviour I'm currently seeing,
or suggest how I can get FS to set the initial outgoing RTP port correctly?


Thanks,
Sean.
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From rfmundkowsky at yahoo.com  Mon Jul 17 15:18:30 2017
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Mon, 17 Jul 2017 15:18:30 +0000 (UTC)
Subject: [Freeswitch-users] real time configuration management (extensions)
References: <1099770109.2270192.1500304710981.ref@mail.yahoo.com>
Message-ID: <1099770109.2270192.1500304710981@mail.yahoo.com>


What are the best practice for real-time configurationmanagement of extensions in a live FreeSWTICH system without downtime?

 

We currently use FreeSWITCH XML configuration files andreload the dial plan via fs_cli.

 

Would using a database make configurations more dynamic?

 

If we move to using a database, I am guessing PostgreSQLis the way to go?  Or is MySQL possible?

 

Robert
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From rfmundkowsky at yahoo.com  Mon Jul 17 15:19:23 2017
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Mon, 17 Jul 2017 15:19:23 +0000 (UTC)
Subject: [Freeswitch-users] CDR - best practices
References: <224992776.2280023.1500304763949.ref@mail.yahoo.com>
Message-ID: <224992776.2280023.1500304763949@mail.yahoo.com>


What are the best practice for writing Call DetailRecords (CDRs) to a database?

 

We are using custom code to write CDR-like data to logfiles and then later loading logs into a database.  We have very low usage right now, so Isuspect that we could just write directly to the database.  But as we scale, is it a better idea to notwrite directly to the database?

 

Note I am not sure if we need custom CDRs or if FS' CDRmodule might be leveraged.  I have tolook into that.

 

 

Robert
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From rfmundkowsky at yahoo.com  Mon Jul 17 15:20:02 2017
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Mon, 17 Jul 2017 15:20:02 +0000 (UTC)
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
	and/or webRTC)?
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
Message-ID: <1685634783.2289866.1500304802144@mail.yahoo.com>


Anyone have suggestions on free automated calling toolsfor testing a system using SIP and/or webRTC?

 

Robert
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From rfmundkowsky at yahoo.com  Mon Jul 17 15:20:38 2017
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Mon, 17 Jul 2017 15:20:38 +0000 (UTC)
Subject: [Freeswitch-users] traffic and system monitoring
References: <1012898889.2268619.1500304838441.ref@mail.yahoo.com>
Message-ID: <1012898889.2268619.1500304838441@mail.yahoo.com>


 

Is Homer the best free option for traffic and systemmonitoring GUI?

               Or isCACTI better option?

 

Robert 
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From asilva at wirelessmundi.com  Mon Jul 17 16:36:33 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Mon, 17 Jul 2017 18:36:33 +0200
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
 and/or webRTC)?
In-Reply-To: <1685634783.2289866.1500304802144@mail.yahoo.com>
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
 <1685634783.2289866.1500304802144@mail.yahoo.com>
Message-ID: <3dd75f2d-f62e-090d-94f6-3ca8ee2fc3f7@wirelessmundi.com>

To test SIP you have sipp url: http://sipp.sourceforge.net/ , you can 
test lot of scenarios.

for webrtc i'm trying to find one too :)


Saludos / Regards / Cumprimentos,
António silva

On 07/17/2017 05:20 PM, robert mundkowsky wrote:
>
> Anyone have suggestions on free automated calling tools for testing a 
> system using SIP and/or webRTC?
>
> Robert
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Mon Jul 17 16:39:32 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 17 Jul 2017 12:39:32 -0400
Subject: [Freeswitch-users] secondary outbound gateway - how to prefix?
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E8681044@mbx-01.sysconfig.co.uk>
References: <83437836-5724-6f73-c174-508e89067b72@gmail.com>
 <6FD2F8B5BB72834E9939AEDF9FB802A901E8681044@mbx-01.sysconfig.co.uk>
Message-ID: <7C5197C1-8737-44B5-B48A-0F173CD6A85C@jerris.com>

Please note, as posted before, confidential messages are not permitted on the mailing list, and will not be removed upon request.

> On Jul 17, 2017, at 1:22 AM, Shaun Stokes <shaun.stokes at itec-support.co.uk> wrote:
> 
> You've put a 1 before \d so the matched number should be 11 digits including the 1. This will match destination numbers prefixed with 9, the $1 variable will be the rest of the number after the 9 prefix.
> 
> -----Original Message-----
> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of jungle boogie
> Sent: 17 July 2017 02:07
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] secondary outbound gateway - how to prefix?
> 
> Hi All,
> 
> I would like to have a secondary gateway added to freeswitch. To have it used for outbound calling, I need to prefix the dialstring with something like a 9 followed by the 10 digit number.
> 
> I think that's what this page is explaining:
> https://freeswitch.org/confluence/display/FREESWITCH/Configuring+FreeSWITCH
> 
> <condition field="destination_number" expression="^9(1\d{10})$">
> 
> Will that use the gateway if the telephone number is prefixed with a 9?
> 
> Thanks!
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
> ______________________________________________________________________
> [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg]
> Shaun Stokes - Infrastructure Analyst
> 
> T :     01453 700713
> E :     shaun.stokes at itec-support.co.uk
> W :     www.itec-support.co.uk
> 
> Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF
> Company No. 06908001
> 
> CONFIDENTIALITY NOTICE
> This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately.
> The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment.
> 
> ______________________________________________________________________
> This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
> ______________________________________________________________________
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Mon Jul 17 16:42:57 2017
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Jul 2017 11:42:57 -0500
Subject: [Freeswitch-users] RTP Auto Switch behaviour on EC2
In-Reply-To: <CALhEHhUP7K26szNe2qxcQ98adOjgLp59whTG5Ogx8mjv2z4SWw@mail.gmail.com>
References: <CALhEHhUP7K26szNe2qxcQ98adOjgLp59whTG5Ogx8mjv2z4SWw@mail.gmail.com>
Message-ID: <CAEJMVkA59nmydSvhD+VGRv11i7n6uAU3QRmmRyCb7Tznus9nyA@mail.gmail.com>

"Freeswitch always sends RTP traffic from it's private IP", Because thats
all it CAN do, the public IP isn't directly bound on the system so it can't
send from the public IP.

/b


On Mon, Jul 17, 2017 at 4:33 AM, Sean Ingham <sean at missionlabs.co.uk> wrote:

> Hi,
> I'm running FreeSwitch on an AWS EC2 with config as per the wiki EC2 page.
>
> external_rtp_ip & bind_server_ip are both set to the box's public AWS
> Elastic IP - as is ext_rtp_ip on all profiles.
>
> I'm running a WebRTC gateway also in AWS. I can see that SDPs from
> Freeswitch to the gateway all include Freeswitch's public IP as desired,
> however when the call is answered Freeswitch always sends RTP traffic from
> it's private IP. Then a second or 2 later Freeswitch sees incoming RTP from
> the gateway to it's public IP address and RTP auto swtiching behaviour
> kicks in, resulting in a line like this in FS logs:
>
> switch_rtp.c:6954 Auto Changing audio port from 172.31.x.x:31196 to
> 52.17.x.x:31196
>
> As it happens, the gateway is able to receive media from either private or
> public IP as the box is in the same subnet, but in around 30% of cases at
> the point this switching occurs the call audio drops. (I'm guessing the
> gateway has issues stitching together the rtp streams from 2 different
> sources.)
>
> I've tried experimenting with the disable_rtp_auto_switch parameter, but
> seems to me a cleaner solution would be to have Freeswitch send RTP from
> it's public IP in the first place, and I can't understand why it's not
> doing that given it's negotiated to use the public IP in it's SDP.
>
> Can anyone provide any explanation for the behaviour I'm currently seeing,
> or suggest how I can get FS to set the initial outgoing RTP port correctly?
>
>
> Thanks,
> Sean.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From rfmundkowsky at yahoo.com  Mon Jul 17 16:54:38 2017
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Mon, 17 Jul 2017 16:54:38 +0000 (UTC)
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
	and/or webRTC)?
In-Reply-To: <1685634783.2289866.1500304802144@mail.yahoo.com>
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
 <1685634783.2289866.1500304802144@mail.yahoo.com>
Message-ID: <2117846415.2342466.1500310478401@mail.yahoo.com>

Sliva, thanks for info on SIPP, I will check it out.
To all, I also meant RTP tools as well.



On Monday, July 17, 2017, 11:20:02 AM EDT, robert mundkowsky <rfmundkowsky at yahoo.com> wrote:


Anyone have suggestions on free automated calling toolsfor testing a system using SIP and/or webRTC?

 

Robert
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From chad at apartmentlines.com  Mon Jul 17 17:02:55 2017
From: chad at apartmentlines.com (Chad Phillips)
Date: Mon, 17 Jul 2017 10:02:55 -0700
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
 and/or webRTC)?
In-Reply-To: <3dd75f2d-f62e-090d-94f6-3ca8ee2fc3f7@wirelessmundi.com>
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
 <1685634783.2289866.1500304802144@mail.yahoo.com>
 <3dd75f2d-f62e-090d-94f6-3ca8ee2fc3f7@wirelessmundi.com>
Message-ID: <CAJ0sghTVNC+daNg8i+VK2Rx__X6t5V6pqgbrsg2g4Yz_RZ4V5w@mail.gmail.com>

I’ve set up a fairly robust WebRTC testing platform myself, using a
combination of existing FOSS tools, some of my own FOSS, and quite a bit of
elbow grease. Here’s a quick example of what I’m able to do with it:
https://www.youtube.com/watch?v=V4PBWXKi-WQ

Unfortunately, it would take quite a bit of time to fully explain
everything I did to build my system, much more than I’m willing to go into
in an email, but just know it’s possible!

I did submit a ClueCon talk proposal to show this system as a case study,
but don’t know if it will be accepted. I’d also be willing to go over it on
a ClueCon Weekly call at some point, if the talk falls through.

Chad

On Mon, Jul 17, 2017 at 9:36 AM, Antonio Silva <asilva at wirelessmundi.com>
wrote:

> To test SIP you have sipp url: http://sipp.sourceforge.net/ , you can
> test lot of scenarios.
>
> for webrtc i'm trying to find one too :)
>
>
> Saludos / Regards / Cumprimentos,
> António silva
>
> On 07/17/2017 05:20 PM, robert mundkowsky wrote:
>
> Anyone have suggestions on free automated calling tools for testing a
> system using SIP and/or webRTC?
>
>
>
> Robert
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sean at missionlabs.co.uk  Mon Jul 17 17:17:51 2017
From: sean at missionlabs.co.uk (Sean Ingham)
Date: Mon, 17 Jul 2017 18:17:51 +0100
Subject: [Freeswitch-users] RTP Auto Switch behaviour on EC2
Message-ID: <CALhEHhWZJNnHQUzdt_7reGn5DDPSvZ8m9QmCt_Q2vah3YNyCRA@mail.gmail.com>

Hi Brian,

That makes sense since the EC2 has no network interface for the public IP
that FS can bind to.
So what is RTP auto switching actually doing when it says:

Auto Changing audio port from 172.31.x.x:31196 to 52.17.x.x:31196

Is it just amending udp packet headers but not actually rerouting anything?

Thanks,
Sean

>"Freeswitch always sends RTP traffic from it's private IP", Because thats
>all it CAN do, the public IP isn't directly bound on the system so it
>can't send from the public IP.

>/b


On Mon, Jul 17, 2017 at 4:33 AM, Sean Ingham <sean at
missionlabs.co.uk
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>
wrote:

>* Hi,
*>* I'm running FreeSwitch on an AWS EC2 with config as per the wiki EC2 page.
*>>* external_rtp_ip & bind_server_ip are both set to the box's public AWS
*>* Elastic IP - as is ext_rtp_ip on all profiles.
*>>* I'm running a WebRTC gateway also in AWS. I can see that SDPs from
*>* Freeswitch to the gateway all include Freeswitch's public IP as desired,
*>* however when the call is answered Freeswitch always sends RTP traffic from
*>* it's private IP. Then a second or 2 later Freeswitch sees incoming RTP from
*>* the gateway to it's public IP address and RTP auto swtiching behaviour
*>* kicks in, resulting in a line like this in FS logs:
*>>* switch_rtp.c:6954 Auto Changing audio port from 172.31.x.x:31196 to
*>* 52.17.x.x:31196
*>>* As it happens, the gateway is able to receive media from either private or
*>* public IP as the box is in the same subnet, but in around 30% of cases at
*>* the point this switching occurs the call audio drops. (I'm guessing the
*>* gateway has issues stitching together the rtp streams from 2 different
*>* sources.)
*>>* I've tried experimenting with the disable_rtp_auto_switch parameter, but
*>* seems to me a cleaner solution would be to have Freeswitch send RTP from
*>* it's public IP in the first place, and I can't understand why it's not
*>* doing that given it's negotiated to use the public IP in it's SDP.
*>>* Can anyone provide any explanation for the behaviour I'm currently seeing,
*>* or suggest how I can get FS to set the initial outgoing RTP port correctly?
*>>>* Thanks,
*>* Sean.
*>>* _________________________________________________________________________
*>* Professional FreeSWITCH Consulting Services:
*>* consulting at freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
*>* http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
*>>* Official FreeSWITCH Sites
*>* http://www.freeswitch.org <http://www.freeswitch.org/>
*>* http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
*>* http://www.cluecon.com <http://www.cluecon.com/>
*>>* FreeSWITCH-users mailing list
*>* FreeSWITCH-users at lists.freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
*>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
*>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://lists.freeswitch.org/mailman/options/freeswitch-users>
*>* http://www.freeswitch.org <http://www.freeswitch.org/>
*>


-- 

*Brian West*brian at freeswitch.org
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>

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

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From chad at apartmentlines.com  Mon Jul 17 19:09:42 2017
From: chad at apartmentlines.com (Chad Phillips)
Date: Mon, 17 Jul 2017 12:09:42 -0700
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
 and/or webRTC)?
In-Reply-To: <3dd75f2d-f62e-090d-94f6-3ca8ee2fc3f7@wirelessmundi.com>
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
 <1685634783.2289866.1500304802144@mail.yahoo.com>
 <3dd75f2d-f62e-090d-94f6-3ca8ee2fc3f7@wirelessmundi.com>
Message-ID: <CAJ0sghRW+j5aNUFJ1jZCFMOvjb7bqjLfaBH2UaNouD=5diTiUw@mail.gmail.com>

OK, quick update: I *did* get accepted to do a lightning talk at this
year’s ClueCon on the topic of building a FOSS WebRTC testing framework, so
if you’re there you can get some info on this question :)

I’ll also at least make the slides available after the talk for those who
don’t attend.

On Mon, Jul 17, 2017 at 9:36 AM, Antonio Silva <asilva at wirelessmundi.com>
wrote:

> To test SIP you have sipp url: http://sipp.sourceforge.net/ , you can
> test lot of scenarios.
>
> for webrtc i'm trying to find one too :)
>
>
> Saludos / Regards / Cumprimentos,
> António silva
>
> On 07/17/2017 05:20 PM, robert mundkowsky wrote:
>
> Anyone have suggestions on free automated calling tools for testing a
> system using SIP and/or webRTC?
>
>
>
> Robert
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rfmundkowsky at yahoo.com  Mon Jul 17 19:18:29 2017
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Mon, 17 Jul 2017 19:18:29 +0000 (UTC)
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
 and/or webRTC)?
In-Reply-To: <CAJ0sghRW+j5aNUFJ1jZCFMOvjb7bqjLfaBH2UaNouD=5diTiUw@mail.gmail.com>
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
 <1685634783.2289866.1500304802144@mail.yahoo.com>
 <3dd75f2d-f62e-090d-94f6-3ca8ee2fc3f7@wirelessmundi.com>
 <CAJ0sghRW+j5aNUFJ1jZCFMOvjb7bqjLfaBH2UaNouD=5diTiUw@mail.gmail.com>
Message-ID: <1929064289.2499282.1500319109286@mail.yahoo.com>

looking forward to your talk and slides.



On Monday, July 17, 2017, 3:11:06 PM EDT, Chad Phillips <chad at apartmentlines.com> wrote:

OK, quick update: I *did* get accepted to do a lightning talk at this year’s ClueCon on the topic of building a FOSS WebRTC testing framework, so if you’re there you can get some info on this question :)
I’ll also at least make the slides available after the talk for those who don’t attend.
On Mon, Jul 17, 2017 at 9:36 AM, Antonio Silva <asilva at wirelessmundi.com> wrote:

  To test SIP you have sipp url: http://sipp.sourceforge.net/ , you can test lot of scenarios.
 
 for webrtc i'm trying to find one too :)
 
 
 Saludos / Regards / Cumprimentos,
António silva On 07/17/2017 05:20 PM, robert mundkowsky wrote:
  
   
Anyone have suggestions on free automated calling tools for testing a system using SIP and/or webRTC?
 
 
 
Robert
   
  
 ______________________________ ______________________________ _____________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www. freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists. freeswitch.org
http://lists.freeswitch.org/ mailman/listinfo/freeswitch- users
UNSUBSCRIBE:http://lists. freeswitch.org/mailman/ options/freeswitch-users
http://www.freeswitch.org 
 
 
______________________________ ______________________________ _____________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www. freeswitchsolutions.com

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

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


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From anthony.minessale at gmail.com  Mon Jul 17 20:19:32 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Jul 2017 15:19:32 -0500
Subject: [Freeswitch-users] ClueCon Hotel is filling up! ACT NOW and
	save $300
In-Reply-To: <7736AD67-4464-428B-BD6E-1F03F566626C@jerris.com>
References: <CAKbxfG8DqKRDYxqpWbneViPyJEOQdOAr6YAr98ZZjpZwwsyGfA@mail.gmail.com>
 <7736AD67-4464-428B-BD6E-1F03F566626C@jerris.com>
Message-ID: <CAKbxfG9YZE25xdDm4rJ-XT04tOg1jqaKMEmOhhMQSrQSOHtdUg@mail.gmail.com>

The code is only good a few more days, ACT NOW!

On Tue, Jul 11, 2017 at 2:39 PM, Michael Jerris <mike at jerris.com> wrote:

> This code is still good.  If you’ve been thinking about coming, now is the
> time to sign up.  If you are not sure if you want to come, give us a call
> and we can talk to you about all the great speakers and content.
>
> 877-7-4-A-CLUE
>
> Use the code CCJulySM2017 and save $300
> Help support the FS Community and have fun in the process!
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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

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

https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From ssinyagin at gmail.com  Mon Jul 17 21:34:05 2017
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 17 Jul 2017 23:34:05 +0200
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
 and/or webRTC)?
In-Reply-To: <1685634783.2289866.1500304802144@mail.yahoo.com>
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
 <1685634783.2289866.1500304802144@mail.yahoo.com>
Message-ID: <CACOYK=iRns03FXGKzMhadKhr7JZ35o2gGgLEs1fqkiiu8FZafg@mail.gmail.com>

I made this simple call generator:
https://github.com/voxserv/freeswitch-perf-dialer



On 17 Jul 2017 18:25, "robert mundkowsky" <rfmundkowsky at yahoo.com> wrote:

> Anyone have suggestions on free automated calling tools for testing a
> system using SIP and/or webRTC?
>
>
>
> Robert
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From grcamauer at gmail.com  Mon Jul 17 21:53:05 2017
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Mon, 17 Jul 2017 18:53:05 -0300
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
 and/or webRTC)?
In-Reply-To: <CACOYK=iRns03FXGKzMhadKhr7JZ35o2gGgLEs1fqkiiu8FZafg@mail.gmail.com>
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
 <1685634783.2289866.1500304802144@mail.yahoo.com>
 <CACOYK=iRns03FXGKzMhadKhr7JZ35o2gGgLEs1fqkiiu8FZafg@mail.gmail.com>
Message-ID: <CAPNTnnrFwUGV0TOR5fOrsoA7AhHvSryJvmrd-wTLku+b+BzETQ@mail.gmail.com>

You might want to look at StarTrinity.  It's commercial, but passive
monitoring is free.  They only do SIP and RTP for now, but WebRTC is in the
roadmap.



On Mon, Jul 17, 2017 at 6:34 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> I made this simple call generator:
> https://github.com/voxserv/freeswitch-perf-dialer
>
>
>
> On 17 Jul 2017 18:25, "robert mundkowsky" <rfmundkowsky at yahoo.com> wrote:
>
>> Anyone have suggestions on free automated calling tools for testing a
>> system using SIP and/or webRTC?
>>
>>
>>
>> Robert
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Guillermo Ruiz Camauer
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From rfmundkowsky at yahoo.com  Mon Jul 17 22:30:15 2017
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Mon, 17 Jul 2017 22:30:15 +0000 (UTC)
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
 and/or webRTC)?
In-Reply-To: <CACOYK=iRns03FXGKzMhadKhr7JZ35o2gGgLEs1fqkiiu8FZafg@mail.gmail.com>
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
 <1685634783.2289866.1500304802144@mail.yahoo.com>
 <CACOYK=iRns03FXGKzMhadKhr7JZ35o2gGgLEs1fqkiiu8FZafg@mail.gmail.com>
Message-ID: <1285318814.2647983.1500330615672@mail.yahoo.com>

Interesting, but are you using SIP/RTP?  I don't see the SIP and RTP ports in your options? I am guessing you are using FS's ESL instead?



On Monday, July 17, 2017, 5:35:12 PM EDT, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

I made this simple call generator:https://github.com/voxserv/freeswitch-perf-dialer



On 17 Jul 2017 18:25, "robert mundkowsky" <rfmundkowsky at yahoo.com> wrote:


Anyone have suggestions on free automated calling toolsfor testing a system using SIP and/or webRTC?

 

Robert

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

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

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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From rfmundkowsky at yahoo.com  Mon Jul 17 22:33:25 2017
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Mon, 17 Jul 2017 22:33:25 +0000 (UTC)
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
 and/or webRTC)?
In-Reply-To: <CAPNTnnrFwUGV0TOR5fOrsoA7AhHvSryJvmrd-wTLku+b+BzETQ@mail.gmail.com>
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
 <1685634783.2289866.1500304802144@mail.yahoo.com>
 <CACOYK=iRns03FXGKzMhadKhr7JZ35o2gGgLEs1fqkiiu8FZafg@mail.gmail.com>
 <CAPNTnnrFwUGV0TOR5fOrsoA7AhHvSryJvmrd-wTLku+b+BzETQ@mail.gmail.com>
Message-ID: <550876062.2637447.1500330805434@mail.yahoo.com>

Looks like this has a lot built-in.  Does "passive monitoring" refer to "listening" to interface and not generating traffic? 




On Monday, July 17, 2017, 5:54:07 PM EDT, Guillermo Ruiz Camauer <grcamauer at gmail.com> wrote:

You might want to look at StarTrinity.  It's commercial, but passive monitoring is free.  They only do SIP and RTP for now, but WebRTC is in the roadmap.


On Mon, Jul 17, 2017 at 6:34 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

I made this simple call generator:https://github.com/voxserv/ freeswitch-perf-dialer



On 17 Jul 2017 18:25, "robert mundkowsky" <rfmundkowsky at yahoo.com> wrote:


Anyone have suggestions on free automated calling toolsfor testing a system using SIP and/or webRTC?

 

Robert

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswi tch.org
http://lists.freeswitch.org/ma ilman/listinfo/freeswitch-user s
UNSUBSCRIBE:http://lists.frees witch.org/mailman/options/ freeswitch-users
http://www.freeswitch.org


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

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

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




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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From fs at voice2net.ca  Tue Jul 18 01:54:11 2017
From: fs at voice2net.ca (fs at voice2net.ca)
Date: Mon, 17 Jul 2017 21:54:11 -0400
Subject: [Freeswitch-users] Presence on debian 1.9
In-Reply-To: <BB0260B1-F970-48F9-8E53-C121B2EFB584@mgtech.com>
References: <A379B6C8-2D73-48EA-8680-7C87BACF7987@tm.net.uk>
 <CAHzfiZ6mBWJR8-6aKTNSm5BwJxPcpwCFTUWgLywG5L4OsNiEqA@mail.gmail.com>
 <BB0260B1-F970-48F9-8E53-C121B2EFB584@mgtech.com>
Message-ID: <01c701d2ff68$ce45faa0$6ad1efe0$@ca>

We are running freeswitch FreeSWITCH Version
1.9.0+git~20170119T195051Z~9c8d9cf120~32bit (git 9c8d9cf 2017-01-19
19:50:51Z 32bit) on raspbian.

 

IN the past we used presence to turn on and off buttons for Nite answer, and
various other information elements.  In the current release of Freeswitch I
cannot get a lamp to illuminate or even fire a presence request if there is
not device registered.  

 

Does anyone know of any changes made from presence that would cause this.
We send an event as below which does not fire anything in fs_cli however, if
I change 34Nite to a real extension it works.  I know this will eventually
go out, we have created a background task to keep the lights on or off
depending on conditions.

 

Any help would be appreciated.

 

 

sendevent PRESENCE_IN

proto: sip

from: 34Nite at 192.168.55.20

login: 34Nite at 192.168.55.20

event_type: presence

alt_event_type: dialog

Presence-Call-Direction: inbound

answer-state: confirmed

unique-id: foo

 

 

 

small_logo

Darcy Primrose

200 Prescott Street

Kemptville, ON

K0E 1T0

Company   613-713-1555

direct          613-713-3351

 

 

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From steveayre at gmail.com  Tue Jul 18 08:25:23 2017
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 18 Jul 2017 09:25:23 +0100
Subject: [Freeswitch-users] CDR - best practices
In-Reply-To: <224992776.2280023.1500304763949@mail.yahoo.com>
References: <224992776.2280023.1500304763949.ref@mail.yahoo.com>
 <224992776.2280023.1500304763949@mail.yahoo.com>
Message-ID: <CAFiqYu=kdf62=KC4yX-NzWdb2=YwywsuZL-4Mw9e0f7Zfi2fhw@mail.gmail.com>

Custom code is probably not the correct way to generate logs that you're
using as CDRs. It won't run in the CS_REPORTING state which is the best
place to generate CDRs.

As for writing direct to the database there are modules for that (eg
mod_odbc_cdr) but you can't rely on that alone, especially as you scale
up. The database might be unavailable, have reached the maximum number of
connections, or have locked tables. You need to be able to support spooling
CDRs that aren't written to the DB and retrying later. See csv-path-on-fail
for mod_odbc_cdr.

Personally I prefer mod_xml_cdr as it can post the CDRs to a web farm for
processing, and we can change their processing in that application if we
want extra information in future. That spools via err-log-dir.

I'd also suggest something like mod_xml_cdr or at least something like
mod_cdr_csv so that you can archive CDRs outside of the database. That way
if your database fails or gets corrupted you have a backup copy, or
something to reconcile against.

On 17 July 2017 at 16:19, robert mundkowsky <rfmundkowsky at yahoo.com> wrote:

> What are the best practice for writing Call Detail Records (CDRs) to a
> database?
>
>
>
> We are using custom code to write CDR-like data to log files and then
> later loading logs into a database.  We have very low usage right now, so
> I suspect that we could just write directly to the database.  But as we
> scale, is it a better idea to not write directly to the database?
>
>
>
> Note I am not sure if we need custom CDRs or if FS' CDR module might be
> leveraged.  I have to look into that.
>
>
>
>
>
> Robert
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at gmail.com  Tue Jul 18 13:43:21 2017
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Tue, 18 Jul 2017 15:43:21 +0200
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
 and/or webRTC)?
In-Reply-To: <1285318814.2647983.1500330615672@mail.yahoo.com>
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
 <1685634783.2289866.1500304802144@mail.yahoo.com>
 <CACOYK=iRns03FXGKzMhadKhr7JZ35o2gGgLEs1fqkiiu8FZafg@mail.gmail.com>
 <1285318814.2647983.1500330615672@mail.yahoo.com>
Message-ID: <CACOYK=iwroND=_KdHvG38Wet5XMJvFzBAwpYD9Ei7=fiApweWg@mail.gmail.com>

it's just a dialer that originates calls from a FreeSWITCH server via
ESL. It's up to you to tune FreeSWITCH configuration for your needs
(such as media transport, codecs, signaling, etc.)

You can use one instance of FS to send calls to another one that is under tests.



On Tue, Jul 18, 2017 at 12:30 AM, robert mundkowsky
<rfmundkowsky at yahoo.com> wrote:
> Interesting, but are you using SIP/RTP?  I don't see the SIP and RTP ports
> in your options? I am guessing you are using FS's ESL instead?
>
>
>
> ________________________________
> On Monday, July 17, 2017, 5:35:12 PM EDT, Stanislav Sinyagin
> <ssinyagin at gmail.com> wrote:
>
>
> I made this simple call generator:
> https://github.com/voxserv/freeswitch-perf-dialer
>
>
>
> On 17 Jul 2017 18:25, "robert mundkowsky" <rfmundkowsky at yahoo.com> wrote:
>
> Anyone have suggestions on free automated calling tools for testing a system
> using SIP and/or webRTC?
>
>
>
> Robert
>
>
> ______________________________ ______________________________ _____________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www. freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch. org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists. freeswitch.org
> http://lists.freeswitch.org/ mailman/listinfo/freeswitch- users
> UNSUBSCRIBE:http://lists. freeswitch.org/mailman/ options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From david.villasmil.work at gmail.com  Tue Jul 18 13:46:52 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 18 Jul 2017 13:46:52 +0000
Subject: [Freeswitch-users] free testing tool - automated calling (SIP
 and/or webRTC)?
In-Reply-To: <CACOYK=iwroND=_KdHvG38Wet5XMJvFzBAwpYD9Ei7=fiApweWg@mail.gmail.com>
References: <1685634783.2289866.1500304802144.ref@mail.yahoo.com>
 <1685634783.2289866.1500304802144@mail.yahoo.com>
 <CACOYK=iRns03FXGKzMhadKhr7JZ35o2gGgLEs1fqkiiu8FZafg@mail.gmail.com>
 <1285318814.2647983.1500330615672@mail.yahoo.com>
 <CACOYK=iwroND=_KdHvG38Wet5XMJvFzBAwpYD9Ei7=fiApweWg@mail.gmail.com>
Message-ID: <CAFGRPVog_TKrFF7BHfh+rBg9rZ6TCXm-+g5FsOS5hX-kojuHYQ@mail.gmail.com>

You might also want to look at

https://freeswitch.org/stash/users/davidcsi/repos/freeswitch/browse/src/mod/applications/mod_dialer/mod_dialer.c?at=refs%2Fheads%2Ffeature%2Fmod_dialer

It's a freeswitch module to generate outgoing calls and bridge them to
wherever you want, application, conference, transfer, whatever.
On Tue, Jul 18, 2017 at 3:44 PM Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> it's just a dialer that originates calls from a FreeSWITCH server via
> ESL. It's up to you to tune FreeSWITCH configuration for your needs
> (such as media transport, codecs, signaling, etc.)
>
> You can use one instance of FS to send calls to another one that is under
> tests.
>
>
>
> On Tue, Jul 18, 2017 at 12:30 AM, robert mundkowsky
> <rfmundkowsky at yahoo.com> wrote:
> > Interesting, but are you using SIP/RTP?  I don't see the SIP and RTP
> ports
> > in your options? I am guessing you are using FS's ESL instead?
> >
> >
> >
> > ________________________________
> > On Monday, July 17, 2017, 5:35:12 PM EDT, Stanislav Sinyagin
> > <ssinyagin at gmail.com> wrote:
> >
> >
> > I made this simple call generator:
> > https://github.com/voxserv/freeswitch-perf-dialer
> >
> >
> >
> > On 17 Jul 2017 18:25, "robert mundkowsky" <rfmundkowsky at yahoo.com>
> wrote:
> >
> > Anyone have suggestions on free automated calling tools for testing a
> system
> > using SIP and/or webRTC?
> >
> >
> >
> > Robert
> >
> >
> > ______________________________ ______________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www. freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch. org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists. freeswitch.org
> > http://lists.freeswitch.org/ mailman/listinfo/freeswitch- users
> > UNSUBSCRIBE:http://lists. freeswitch.org/mailman/
> options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From 3b1den at gmail.com  Tue Jul 18 07:31:56 2017
From: 3b1den at gmail.com (=?UTF-8?B?wqA=?=)
Date: Tue, 18 Jul 2017 10:31:56 +0300
Subject: [Freeswitch-users] Verto member name
Message-ID: <CA+ZG+uyuEHLu4MVp+nM--hYnrsEdn=uj9A8Hswc2aUtC-X6n3A@mail.gmail.com>

I`m trying to implement verto communicator into fusionpbx.
But it doesnt show member name, only caller number on test vm.
On working  vm (fs builded from source 1.6.17) code looks like "<h4
class="chat-members-name" ng-class="{ 'clickable': verto.data.confRole ==
'moderator' }" ng-click="toggleModMenu($index)">"
On test vm (fusionpbx autoinstaller debian jessie 1.6.19) "<h4
class="chat-members-name clickable" ng-class="{ 'clickable':
verto.data.confRole == 'moderator' }" ng-click="toggleModMenu($index)">
How to fix it?

https://prnt.sc/fx1pqu
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From jungleboogie0 at gmail.com  Tue Jul 18 15:24:36 2017
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Tue, 18 Jul 2017 08:24:36 -0700
Subject: [Freeswitch-users] CDR - best practices
In-Reply-To: <224992776.2280023.1500304763949@mail.yahoo.com>
References: <224992776.2280023.1500304763949.ref@mail.yahoo.com>
 <224992776.2280023.1500304763949@mail.yahoo.com>
Message-ID: <CAKE2PDvEMiBLV7PGY9hv22y3YZ6Nk=q8NCX7qNuQEVvPJVpytA@mail.gmail.com>

On 17 July 2017 at 08:19, robert mundkowsky <rfmundkowsky at yahoo.com> wrote:
> What are the best practice for writing Call Detail Records (CDRs) to a
> database?
>

Page 49 of Freeswitch Cookbook 1.6 reads in part:
Frequently, it is necessary to put CDR information into a database
such as PostgreSQL or
other SQL and NoSQL databases. FreeSWITCH has various modules for
writing CDRs directly
to many databases, but the preferred architecture is writing CDRs to
the disk or posting them
to a web server, and then processing them so that they can be inserted
into a database. Many
engineering reasons lead to this architecture (for example, avoiding
dependence on direct,
real-time interaction with the database), and most of them relate it
to integrity and resilience.

https://www.packtpub.com/networking-and-servers/freeswitch-16-cookbook

Following this section is mod_xml_cdr, which Steven mention in his email.


From jungleboogie0 at gmail.com  Tue Jul 18 15:27:20 2017
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Tue, 18 Jul 2017 08:27:20 -0700
Subject: [Freeswitch-users] secondary outbound gateway - how to prefix?
In-Reply-To: <7C5197C1-8737-44B5-B48A-0F173CD6A85C@jerris.com>
References: <83437836-5724-6f73-c174-508e89067b72@gmail.com>
 <6FD2F8B5BB72834E9939AEDF9FB802A901E8681044@mbx-01.sysconfig.co.uk>
 <7C5197C1-8737-44B5-B48A-0F173CD6A85C@jerris.com>
Message-ID: <CAKE2PDuMeP2WA06j7kgJ0DY=Eh6bYnWvqoBnz22JxwBNA0oqpA@mail.gmail.com>

On 17 July 2017 at 09:39, Michael Jerris <mike at jerris.com> wrote:
> Please note, as posted before, confidential messages are not permitted on the mailing list, and will not be removed upon request.
>

Sorry, was something confidential that I posted?


From nneul at mst.edu  Tue Jul 18 15:31:04 2017
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 18 Jul 2017 10:31:04 -0500
Subject: [Freeswitch-users] CDR - best practices
In-Reply-To: <CAKE2PDvEMiBLV7PGY9hv22y3YZ6Nk=q8NCX7qNuQEVvPJVpytA@mail.gmail.com>
References: <224992776.2280023.1500304763949.ref@mail.yahoo.com>
 <224992776.2280023.1500304763949@mail.yahoo.com>
 <CAKE2PDvEMiBLV7PGY9hv22y3YZ6Nk=q8NCX7qNuQEVvPJVpytA@mail.gmail.com>
Message-ID: <25afb919-7ad3-591c-472e-cee36090f49a@mst.edu>

There is also mod_format_cdr which covers both json and XML formats.

-- Nathan

On 7/18/17 10:24 AM, jungle Boogie wrote:
> On 17 July 2017 at 08:19, robert mundkowsky <rfmundkowsky at yahoo.com> wrote:
>> What are the best practice for writing Call Detail Records (CDRs) to a
>> database?
>>
> 
> Page 49 of Freeswitch Cookbook 1.6 reads in part:
> Frequently, it is necessary to put CDR information into a database
> such as PostgreSQL or
> other SQL and NoSQL databases. FreeSWITCH has various modules for
> writing CDRs directly
> to many databases, but the preferred architecture is writing CDRs to
> the disk or posting them
> to a web server, and then processing them so that they can be inserted
> into a database. Many
> engineering reasons lead to this architecture (for example, avoiding
> dependence on direct,
> real-time interaction with the database), and most of them relate it
> to integrity and resilience.
> 
> https://www.packtpub.com/networking-and-servers/freeswitch-16-cookbook
> 
> Following this section is mod_xml_cdr, which Steven mention in his email.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 

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


From tculjaga at gmail.com  Tue Jul 18 18:48:11 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 18 Jul 2017 20:48:11 +0200
Subject: [Freeswitch-users] dtmf_type under gateway
Message-ID: <68E0D437-9AF3-445C-AB1F-A40CEFE4B25C@gmail.com>

Hi, 

If i define a gateway in directory and have an external profile that scans gateways in directory...

Is it possible to overwrite dtmf-type using variable dtmf_type under gateway definition for that sip profile?

Example: sip_profile has set rfc2833 while the gw under that profile has the variable set to none (borh direction).


Please, advice.

Regards,
Tihomir.

Sent from my iPhone

From brian at freeswitch.org  Tue Jul 18 21:57:22 2017
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Jul 2017 16:57:22 -0500
Subject: [Freeswitch-users] Verto member name
In-Reply-To: <CA+ZG+uyuEHLu4MVp+nM--hYnrsEdn=uj9A8Hswc2aUtC-X6n3A@mail.gmail.com>
References: <CA+ZG+uyuEHLu4MVp+nM--hYnrsEdn=uj9A8Hswc2aUtC-X6n3A@mail.gmail.com>
Message-ID: <CAEJMVkDXq7qBryBX4y7Mkgu6bpRBAM+iY_nvXKxrUje=c2EhfQ@mail.gmail.com>

Are you able to assist in fixing the outstanding bugs in VC and adding all
the missing features that FS supports to VC?

/b


On Tue, Jul 18, 2017 at 2:31 AM, <3b1den at gmail.com> wrote:

> I`m trying to implement verto communicator into fusionpbx.
> But it doesnt show member name, only caller number on test vm.
> On working  vm (fs builded from source 1.6.17) code looks like "<h4
> class="chat-members-name" ng-class="{ 'clickable': verto.data.confRole ==
> 'moderator' }" ng-click="toggleModMenu($index)">"
> On test vm (fusionpbx autoinstaller debian jessie 1.6.19) "<h4
> class="chat-members-name clickable" ng-class="{ 'clickable':
> verto.data.confRole == 'moderator' }" ng-click="toggleModMenu($index)">
> How to fix it?
>
> https://prnt.sc/fx1pqu
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From italo at freeswitch.org  Wed Jul 19 00:24:37 2017
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Tue, 18 Jul 2017 21:24:37 -0300
Subject: [Freeswitch-users] Verto member name
In-Reply-To: <CAEJMVkDXq7qBryBX4y7Mkgu6bpRBAM+iY_nvXKxrUje=c2EhfQ@mail.gmail.com>
References: <CA+ZG+uyuEHLu4MVp+nM--hYnrsEdn=uj9A8Hswc2aUtC-X6n3A@mail.gmail.com>
 <CAEJMVkDXq7qBryBX4y7Mkgu6bpRBAM+iY_nvXKxrUje=c2EhfQ@mail.gmail.com>
Message-ID: <CABMX_w-LJXt+_ub+fqAmLhxqL4wKe-RGMB5PRA_+3CEtwnknkQ@mail.gmail.com>

I thought FusionPBX included VC on their interface.

The only difference there is a css class clickable. If you look at the
template you'll see that we need member.name to display the name, the code
is something like:

<div class="ellipsis members-name">{{ member.name }}</div>

which is probably read from variable caller_id_name.

On Tue, Jul 18, 2017 at 6:57 PM, Brian West <brian at freeswitch.org> wrote:

> Are you able to assist in fixing the outstanding bugs in VC and adding all
> the missing features that FS supports to VC?
>
> /b
>
>
> On Tue, Jul 18, 2017 at 2:31 AM, <3b1den at gmail.com> wrote:
>
>> I`m trying to implement verto communicator into fusionpbx.
>> But it doesnt show member name, only caller number on test vm.
>> On working  vm (fs builded from source 1.6.17) code looks like "<h4
>> class="chat-members-name" ng-class="{ 'clickable': verto.data.confRole ==
>> 'moderator' }" ng-click="toggleModMenu($index)">"
>> On test vm (fusionpbx autoinstaller debian jessie 1.6.19) "<h4
>> class="chat-members-name clickable" ng-class="{ 'clickable':
>> verto.data.confRole == 'moderator' }" ng-click="toggleModMenu($index)">
>> How to fix it?
>>
>> https://prnt.sc/fx1pqu
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Ítalo Rossi
italo at freeswitch.org
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From 3b1den at gmail.com  Tue Jul 18 17:22:11 2017
From: 3b1den at gmail.com (=?UTF-8?B?wqA=?=)
Date: Tue, 18 Jul 2017 20:22:11 +0300
Subject: [Freeswitch-users] Verto member name
Message-ID: <CA+ZG+uxNyT5iEq9jsU7j4VSFbavBH05AAAJhmwOLAGi2o0SsvQ@mail.gmail.com>

I`m trying to implement verto communicator into fusionpbx.
But it doesnt show member name, only caller number on test vm.
On working  vm (fs builded from source 1.6.17) code looks like "[ALERT]
mod_verto.c:5274 EVENT BROADCAST conference-liveArray.selector at domain {
        "data": {
                "action":       "add",
                "arrIndex":     0,
                "name": "selector",
                "hashKey":      "5cda1538-c6f2-f05b-2e3a-d92be05a2834",
                "wireSerno":    1,
                "data": ["0721", "admin at admin", "admin", "opus at 48000",
"{\"audio\":{\"muted\":true,\"deaf\":false,\"onHold\":false,\"talking\":false,\"floor\":false,\"energyScore\":0},\"video\":{\"visible\":false,\"videoOnly\":false,\"avatarPresented\":false,\"mediaFlow\":\"sendOnly\",\"muted\":true,\"floor\":false,\"reservationID\":null,\"videoLayerID\":-1},\"oldStatus\":\"MUTE
VIDEO (BLIND)\"}", {
                                "avatar":       "
http://gravatar.com/avatar/d41d8cd98f00b204e9800998ecf8427e.png?s=600"
                        }, null]
        },
        "eventChannel": "conference-liveArray.selector at domain"
}



On test vm (fusionpbx autoinstaller debian jessie 1.6.19) "[ALERT]
mod_verto.c:5322 EVENT BROADCAST conference-liveArray.selector at domain {
        "data": {
                "action":       "add",
                "arrIndex":     0,
                "name": "selector",
                "hashKey":      "807bde07-ed9d-673e-1fa1-6ad24379916b",
                "wireSerno":    1,
                "data": ["0006", "1808", "1808", "opus at 48000",
"{\"audio\":{\"muted\":true,\"deaf\":false,\"onHold\":false,\"talking\":false,\"floor\":false,\"energyScore\":0},\"video\":{\"visible\":false,\"videoOnly\":false,\"avatarPresented\":false,\"mediaFlow\":\"sendOnly\",\"muted\":true,\"floor\":false,\"reservationID\":null,\"videoLayerID\":-1},\"oldStatus\":\"MUTE
VIDEO (BLIND)\"}", {
                                "email":        "admin at admin",
                                "avatar":       "
http://gravatar.com/avatar/a3175a452c7a8fea80c62a198a40f6c9.png?s=600"
                        }, null]
        },
        "eventChannel": "conference-liveArray.selector at domain"
}


How to fix it?
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From 3b1den at gmail.com  Wed Jul 19 01:40:25 2017
From: 3b1den at gmail.com (=?UTF-8?B?wqA=?=)
Date: Wed, 19 Jul 2017 04:40:25 +0300
Subject: [Freeswitch-users] Verto member name
In-Reply-To: <CABMX_w-LJXt+_ub+fqAmLhxqL4wKe-RGMB5PRA_+3CEtwnknkQ@mail.gmail.com>
References: <CA+ZG+uyuEHLu4MVp+nM--hYnrsEdn=uj9A8Hswc2aUtC-X6n3A@mail.gmail.com>
 <CAEJMVkDXq7qBryBX4y7Mkgu6bpRBAM+iY_nvXKxrUje=c2EhfQ@mail.gmail.com>
 <CABMX_w-LJXt+_ub+fqAmLhxqL4wKe-RGMB5PRA_+3CEtwnknkQ@mail.gmail.com>
Message-ID: <CA+ZG+uyvs3Sjczt6NLq_OeysMtfRmOdnJgh7K=jThY6YhC-abg@mail.gmail.com>

I`m sorry, that was just clickable moderator controls.

On working  vm (fs builded from source 1.6.17) code looks like "[ALERT]
mod_verto.c:5274 EVENT BROADCAST conference-liveArray.selector at domain {
        "data": {
                "action":       "add",
                "arrIndex":     0,
                "name": "selector",
                "hashKey":      "5cda1538-c6f2-f05b-2e3a-d92be05a2834",
                "wireSerno":    1,
                "data": ["0721", "admin at admin", "admin", "opus at 48000",
"{\"audio\":{\"muted\":true,\"deaf\":false,\"onHold\":false,
\"talking\":false,\"floor\":false,\"energyScore\":0},\"
video\":{\"visible\":false,\"videoOnly\":false,\"avatarPresented\":false,\"
mediaFlow\":\"sendOnly\",\"muted\":true,\"floor\":false,\
"reservationID\":null,\"videoLayerID\":-1},\"oldStatus\":\"MUTE VIDEO
(BLIND)\"}", {
                                "avatar":       "http://gravatar.com/avatar/
d41d8cd98f00b204e9800998ecf8427e.png?s=600"
                        }, null]
        },
        "eventChannel": "conference-liveArray.selector at domain"
}



On test vm (fusionpbx autoinstaller debian jessie 1.6.19) "[ALERT]
mod_verto.c:5322 EVENT BROADCAST conference-liveArray.selector at domain {
        "data": {
                "action":       "add",
                "arrIndex":     0,
                "name": "selector",
                "hashKey":      "807bde07-ed9d-673e-1fa1-6ad24379916b",
                "wireSerno":    1,
                "data": ["0006", "1808", "1808", "opus at 48000",
"{\"audio\":{\"muted\":true,\"deaf\":false,\"onHold\":false,
\"talking\":false,\"floor\":false,\"energyScore\":0},\"
video\":{\"visible\":false,\"videoOnly\":false,\"avatarPresented\":false,\"
mediaFlow\":\"sendOnly\",\"muted\":true,\"floor\":false,\
"reservationID\":null,\"videoLayerID\":-1},\"oldStatus\":\"MUTE VIDEO
(BLIND)\"}", {
                                "email":        "admin at admin",
                                "avatar":       "http://gravatar.com/avatar/
a3175a452c7a8fea80c62a198a40f6c9.png?s=600"
                        }, null]
        },
        "eventChannel": "conference-liveArray.selector at domain"
}


2017-07-19 3:24 GMT+03:00 Ítalo Rossi <italo at freeswitch.org>:

> I thought FusionPBX included VC on their interface.
>
> The only difference there is a css class clickable. If you look at the
> template you'll see that we need member.name to display the name, the
> code is something like:
>
> <div class="ellipsis members-name">{{ member.name }}</div>
>
> which is probably read from variable caller_id_name.
>
> On Tue, Jul 18, 2017 at 6:57 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Are you able to assist in fixing the outstanding bugs in VC and adding
>> all the missing features that FS supports to VC?
>>
>> /b
>>
>>
>> On Tue, Jul 18, 2017 at 2:31 AM, <3b1den at gmail.com> wrote:
>>
>>> I`m trying to implement verto communicator into fusionpbx.
>>> But it doesnt show member name, only caller number on test vm.
>>> On working  vm (fs builded from source 1.6.17) code looks like "<h4
>>> class="chat-members-name" ng-class="{ 'clickable': verto.data.confRole ==
>>> 'moderator' }" ng-click="toggleModMenu($index)">"
>>> On test vm (fusionpbx autoinstaller debian jessie 1.6.19) "<h4
>>> class="chat-members-name clickable" ng-class="{ 'clickable':
>>> verto.data.confRole == 'moderator' }" ng-click="toggleModMenu($index)">
>>> How to fix it?
>>>
>>> https://prnt.sc/fx1pqu
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>>
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>
>> Allison prompts for FreeSWITCH:
>>
>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
>> | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Ítalo Rossi
> italo at freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From 3b1den at gmail.com  Wed Jul 19 01:42:37 2017
From: 3b1den at gmail.com (=?UTF-8?B?wqA=?=)
Date: Wed, 19 Jul 2017 04:42:37 +0300
Subject: [Freeswitch-users] Verto member name
In-Reply-To: <CAEJMVkDXq7qBryBX4y7Mkgu6bpRBAM+iY_nvXKxrUje=c2EhfQ@mail.gmail.com>
References: <CA+ZG+uyuEHLu4MVp+nM--hYnrsEdn=uj9A8Hswc2aUtC-X6n3A@mail.gmail.com>
 <CAEJMVkDXq7qBryBX4y7Mkgu6bpRBAM+iY_nvXKxrUje=c2EhfQ@mail.gmail.com>
Message-ID: <CA+ZG+ux-rcJHo_x2V-r0Jg56p2m9AATzBzsPaRC=KQBuYqUSew@mail.gmail.com>

I`m just a user, not enough skills

2017-07-19 0:57 GMT+03:00 Brian West <brian at freeswitch.org>:

> Are you able to assist in fixing the outstanding bugs in VC and adding all
> the missing features that FS supports to VC?
>
> /b
>
>
> On Tue, Jul 18, 2017 at 2:31 AM, <3b1den at gmail.com> wrote:
>
>> I`m trying to implement verto communicator into fusionpbx.
>> But it doesnt show member name, only caller number on test vm.
>> On working  vm (fs builded from source 1.6.17) code looks like "<h4
>> class="chat-members-name" ng-class="{ 'clickable': verto.data.confRole ==
>> 'moderator' }" ng-click="toggleModMenu($index)">"
>> On test vm (fusionpbx autoinstaller debian jessie 1.6.19) "<h4
>> class="chat-members-name clickable" ng-class="{ 'clickable':
>> verto.data.confRole == 'moderator' }" ng-click="toggleModMenu($index)">
>> How to fix it?
>>
>> https://prnt.sc/fx1pqu
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mark.melling at savageminds.com  Wed Jul 19 08:41:10 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Wed, 19 Jul 2017 08:41:10 +0000
Subject: [Freeswitch-users] Sharing information between an ESL client and
	Verto clients in a conference
Message-ID: <CAJp-f1PGn71oaM=X4jWig-LpZNETcKkQqEwd86d3e7x6dBQTbg@mail.gmail.com>

Hi,

I'd like to be able to share information between an Event Socket Library
client and Verto clients in a conference. This may also include information
that is shared before a Verto client has entered the conference, but they
will still get access to this information.

It looks like the way to achieve this is to use the Verto "live array"
functionality.

So on the assumption I'm correct how can an ESL client update the live
array of a conference?

And how can an ESL client receive any live array update events?

Thanks

Mark
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From mail at paulzillmann.de  Wed Jul 19 13:24:30 2017
From: mail at paulzillmann.de (Paul Zillmann)
Date: Wed, 19 Jul 2017 15:24:30 +0200
Subject: [Freeswitch-users] traffic and system monitoring
In-Reply-To: <1012898889.2268619.1500304838441@mail.yahoo.com>
References: <1012898889.2268619.1500304838441.ref@mail.yahoo.com>
 <1012898889.2268619.1500304838441@mail.yahoo.com>
Message-ID: <dd31c0e0-74b5-3a5a-9236-62bec7948049@paulzillmann.de>

Hey Robert,

I'm using check_mk / nagios for external monitoring and iftop and top /
htop for on site analysis.


Am 17.07.2017 um 17:20 schrieb robert mundkowsky:
>
>  
>
> Is Homer the best free option for traffic and system monitoring GUI?
>
>                Or is CACTI better option?
>
>  
>
> Robert
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From josedavid at zennio.com  Wed Jul 19 06:43:36 2017
From: josedavid at zennio.com (Jose David Jurado Alonso)
Date: Wed, 19 Jul 2017 08:43:36 +0200
Subject: [Freeswitch-users] Freeswitch SIP call not work using a static
	public IP
Message-ID: <CANeCd7qgLdOv8AOnZUGFfpn8DmXVEUDNUYRarfMmrfmwZurHbA@mail.gmail.com>

I've installed *freeswitch into a local environment* and I perform SIP
calls well using the server local IP.

This server has connected to ADSL network with a *static public IP* but
when I try to call using the public IP instead of the local IP the server
receive the call not work...

The clients can register well (or I think so). Others process like apache
work well.

I NAT forwarding in the firewall the next ports: 5060, 5080, 7443.

*$ netstat -putan*

Active Internet connections (servers and established)
Proto Recv-Q Send-Q Local Address           Foreign Address
State       PID/Program name
tcp        0      0 192.168.230.143:2855    0.0.0.0:*
LISTEN      15894/freeswitch
tcp        0      0 192.168.230.143:2856    0.0.0.0:*
LISTEN      15894/freeswitch
tcp        0      0 192.168.230.143:5066    0.0.0.0:*
LISTEN      15894/freeswitch
tcp        0      0 0.0.0.0:80              0.0.0.0:*
LISTEN      1166/nginx -g daemo
tcp        0      0 192.168.230.143:7443    0.0.0.0:*
LISTEN      15894/freeswitch
tcp        0      0 0.0.0.0:22              0.0.0.0:*
LISTEN      1079/sshd
tcp        0      0 192.168.230.143:5080    0.0.0.0:*
LISTEN      15894/freeswitch
tcp        0      0 0.0.0.0:443             0.0.0.0:*
LISTEN      1166/nginx -g daemo
tcp        0      0 192.168.230.143:5060    0.0.0.0:*
LISTEN      15894/freeswitch
tcp        0      0 192.168.230.143:22      192.168.230.167:60298
ESTABLISHED 14816/sshd: ubuntu
tcp        0      0 192.168.230.143:22      192.168.230.167:60296
ESTABLISHED 14744/sshd: ubuntu
tcp6       0      0 :::80                   :::*
LISTEN      1166/nginx -g daemo
tcp6       0      0 :::8021                 :::*
LISTEN      15894/freeswitch
tcp6       0      0 :::22                   :::*
LISTEN      1079/sshd
tcp6       0      0 ::1:5080                :::*
LISTEN      15894/freeswitch
tcp6       0      0 :::443                  :::*
LISTEN      1166/nginx -g daemo
tcp6       0      0 ::1:5060                :::*
LISTEN      15894/freeswitch
udp        0      0 192.168.230.143:5060    0.0.0.0:*
         15894/freeswitch
udp        0      0 192.168.230.143:5080    0.0.0.0:*
         15894/freeswitch
udp        0      0 0.0.0.0:68              0.0.0.0:*
         1019/dhclient
udp6       0      0 ::1:5060                :::*
         15894/freeswitch
udp6       0      0 ::1:5080                :::*
         15894/freeswitch

*$ fs_cli -x "sofia status"*

                     Name      Type
   Data  State
=================================================================================================
            external-ipv6   profile
sip:mod_sofia@[::1]:5080  RUNNING (0)
          192.168.230.143     alias
internal  ALIASED
                 external   profile
sip:mod_sofia at 192.168.230.143:5080  RUNNING (0)
    external::example.com   gateway
sip:joeuser at example.com  NOREG
            internal-ipv6   profile
sip:mod_sofia@[::1]:5060  RUNNING (0)
                 internal   profile
sip:mod_sofia at 192.168.230.143:5060  RUNNING (0)
=================================================================================================

*$ fs_cli -x "sofia status profile external"*

=================================================================================================
Name                external
Domain Name         N/A
Auto-NAT            false
DBName              sofia_reg_external
Pres Hosts
Dialplan            XML
Context             public
Challenge Realm     auto_to
RTP-IP              192.168.230.143
SIP-IP              192.168.230.143
URL                 sip:mod_sofia at 192.168.230.143:5080
BIND-URL            sip:mod_sofia at 192.168.230.143:5080;transport=udp,tcp
HOLD-MUSIC          local_stream://moh
OUTBOUND-PROXY      N/A
CODECS IN           OPUS,G722,PCMU,PCMA,VP8,H264
CODECS OUT          OPUS,G722,PCMU,PCMA,VP8,H264
TEL-EVENT           101
DTMF-MODE           rfc2833
CNG                 13
SESSION-TO          0
MAX-DIALOG          0
NOMEDIA             false
LATE-NEG            true
PROXY-MEDIA         false
ZRTP-PASSTHRU       true
AGGRESSIVENAT       false
CALLS-IN            0
FAILED-CALLS-IN     0
CALLS-OUT           0
FAILED-CALLS-OUT    0
REGISTRATIONS       0

*$ grep -r '"ext-' **

sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
sip_profiles/internal.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
sip_profiles/internal.xml:    <param name="ext-sip-ip" value="auto-nat"/>

When I perform a call the freeswitch.log file show:

df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939
[NOTICE] switch_channel.c:1104 New Channel
sofia/internal/1009 at 192.168.230.143:5060
[df895d9a-30f2-40b5-97bb-56e222c1c45f]
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939
[DEBUG] switch_core_state_machine.c:584
(sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_NEW
(Cur 1 Tot 7)
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939
[DEBUG] sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060
receiving invite from 88.33.111.22:57076 version: 1.9.0 -493-13f2f2a
64bit
2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10238 IP 88.33.111.22
Rejected by acl "domains". Falling back to Digest auth.
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939
[DEBUG] switch_core_state_machine.c:603
(sofia/internal/1009 at 192.168.230.143:5060) State NEW
2017-07-18 14:44:51.218939 [DEBUG] sofia.c:2405 detaching session
df895d9a-30f2-40b5-97bb-56e222c1c45f

and passed some seconds:

df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[WARNING] switch_core_state_machine.c:687
df895d9a-30f2-40b5-97bb-56e222c1c45f
sofia/internal/1009 at 192.168.230.143:5060 Abandoned
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[NOTICE] switch_core_state_machine.c:690 Hangup
sofia/internal/1009 at 192.168.230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:584
(sofia/internal/1009 at 192.168.230.143:5060) Running State Change
CS_HANGUP (Cur 1 Tot 7)
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:850
(sofia/internal/1009 at 192.168.230.143:5060) Callstate Change DOWN ->
HANGUP
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:852
(sofia/internal/1009 at 192.168.230.143:5060) State HANGUP
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] mod_sofia.c:449 Channel
sofia/internal/1009 at 192.168.230.143:5060 hanging up, cause:
WRONG_CALL_STATE
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:60
sofia/internal/1009 at 192.168.230.143:5060 Standard HANGUP, cause:
WRONG_CALL_STATE
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:852
(sofia/internal/1009 at 192.168.230.143:5060) State HANGUP going to sleep
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:619
(sofia/internal/1009 at 192.168.230.143:5060) State Change CS_HANGUP ->
CS_REPORTING
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:584
(sofia/internal/1009 at 192.168.230.143:5060) Running State Change
CS_REPORTING (Cur 1 Tot 7)
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:938
(sofia/internal/1009 at 192.168.230.143:5060) State REPORTING
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:174
sofia/internal/1009 at 192.168.230.143:5060 Standard REPORTING, cause:
WRONG_CALL_STATE
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:938
(sofia/internal/1009 at 192.168.230.143:5060) State REPORTING going to
sleep
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:610
(sofia/internal/1009 at 192.168.230.143:5060) State Change CS_REPORTING
-> CS_DESTROY
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_session.c:1713 Session 7
(sofia/internal/1009 at 192.168.230.143:5060) Locked, Waiting on external
entities
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[NOTICE] switch_core_session.c:1731 Session 7
(sofia/internal/1009 at 192.168.230.143:5060) Ended
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[NOTICE] switch_core_session.c:1735 Close Channel
sofia/internal/1009 at 192.168.230.143:5060 [CS_DESTROY]
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:741
(sofia/internal/1009 at 192.168.230.143:5060) Running State Change
CS_DESTROY (Cur 0 Tot 7)
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:751
(sofia/internal/1009 at 192.168.230.143:5060) State DESTROY
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA
DESTROY
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:181
sofia/internal/1009 at 192.168.230.143:5060 Standard DESTROY
df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940
[DEBUG] switch_core_state_machine.c:751
(sofia/internal/1009 at 192.168.230.143:5060) State DESTROY going to
sleep

Can it be an ACL or sip_profile bad configuration?? Router firewall
configuration maybe?
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From rahul.ultimate at gmail.com  Wed Jul 19 08:37:45 2017
From: rahul.ultimate at gmail.com (Rahul MathuR)
Date: Wed, 19 Jul 2017 14:07:45 +0530
Subject: [Freeswitch-users] Integration of Google Speech API V2
Message-ID: <CAM8wsqJ+Voj65OBhWfDjjzn3c9vhcU0MaN32eDE_dW5-vh50Dg@mail.gmail.com>

Hi,

I'm trying to integrate Google cloud speech recognition v2 in it. I can get
the audio recorded, have created Service key and API key but whenever I try
to access it, I just get 403 access denied. I am at my wits end here.

Has anybody tried it ? were you successful ? Could you please guide me how
to do it ?
I'll be grateful to you if this works !



-- 
Warm Regds.
MathuRahul
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From mateo.felipe05 at gmail.com  Wed Jul 19 11:23:05 2017
From: mateo.felipe05 at gmail.com (Felipe Mateo)
Date: Wed, 19 Jul 2017 07:23:05 -0400
Subject: [Freeswitch-users] Installed FreeSWITCH 1.6 / 1.9 on NetBSD
Message-ID: <CAAaM4gzDdK1H63S34be4J2TCsscUpKuVVv65fWQ4WvUQQcy36g@mail.gmail.com>

Hi, i have successfully installed freeswitch on NetBSD system, i am writing
some initial documentation about this.

-- 
*Felipe Mateo*.

*Cel:* *+18294302505 <https://onlinefm.changeip.org/>*
*Click for Webcall *
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From mike at jerris.com  Wed Jul 19 14:02:42 2017
From: mike at jerris.com (Michael Jerris)
Date: Wed, 19 Jul 2017 14:02:42 +0000
Subject: [Freeswitch-users] Integration of Google Speech API V2
In-Reply-To: <CAM8wsqJ+Voj65OBhWfDjjzn3c9vhcU0MaN32eDE_dW5-vh50Dg@mail.gmail.com>
References: <CAM8wsqJ+Voj65OBhWfDjjzn3c9vhcU0MaN32eDE_dW5-vh50Dg@mail.gmail.com>
Message-ID: <CAEwTsAh=16Jt44NnTHaKL5VGn4_bZULUUPs5d=M7wN0AXSaSSQ@mail.gmail.com>

in what?  i think unimrcp has support for it as if current master

On Wed, Jul 19, 2017 at 9:36 AM Rahul MathuR <rahul.ultimate at gmail.com>
wrote:

> Hi,
>
> I'm trying to integrate Google cloud speech recognition v2 in it. I can
> get the audio recorded, have created Service key and API key but whenever I
> try to access it, I just get 403 access denied. I am at my wits end here.
>
> Has anybody tried it ? were you successful ? Could you please guide me how
> to do it ?
> I'll be grateful to you if this works !
>
>
>
> --
> Warm Regds.
> MathuRahul
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From joel at gogii.net  Wed Jul 19 14:18:07 2017
From: joel at gogii.net (Joel Serrano)
Date: Wed, 19 Jul 2017 07:18:07 -0700
Subject: [Freeswitch-users] Freeswitch SIP call not work using a static
 public IP
In-Reply-To: <CANeCd7qgLdOv8AOnZUGFfpn8DmXVEUDNUYRarfMmrfmwZurHbA@mail.gmail.com>
References: <CANeCd7qgLdOv8AOnZUGFfpn8DmXVEUDNUYRarfMmrfmwZurHbA@mail.gmail.com>
Message-ID: <CAMtXxQm8oj9TfJVLfmukL7BrEEkQAA0GB3xrmxGOnn5fbLGE=Q@mail.gmail.com>

Double check your ACLs, yes. Also, I'm pretty sure you will also have to
forward the RTP port range so you can have audio... Otherwise signaling
will work, but your next problem will be one-way-audio.


On Tue, Jul 18, 2017 at 11:43 PM, Jose David Jurado Alonso <
josedavid at zennio.com> wrote:

>
> I've installed *freeswitch into a local environment* and I perform SIP
> calls well using the server local IP.
>
> This server has connected to ADSL network with a *static public IP* but
> when I try to call using the public IP instead of the local IP the server
> receive the call not work...
>
> The clients can register well (or I think so). Others process like apache
> work well.
>
> I NAT forwarding in the firewall the next ports: 5060, 5080, 7443.
>
> *$ netstat -putan*
>
> Active Internet connections (servers and established)
> Proto Recv-Q Send-Q Local Address           Foreign Address         State       PID/Program name
> tcp        0      0 192.168.230.143:2855    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 192.168.230.143:2856    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 192.168.230.143:5066    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN      1166/nginx -g daemo
> tcp        0      0 192.168.230.143:7443    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN      1079/sshd
> tcp        0      0 192.168.230.143:5080    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 0.0.0.0:443             0.0.0.0:*               LISTEN      1166/nginx -g daemo
> tcp        0      0 192.168.230.143:5060    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 192.168.230.143:22      192.168.230.167:60298   ESTABLISHED 14816/sshd: ubuntu
> tcp        0      0 192.168.230.143:22      192.168.230.167:60296   ESTABLISHED 14744/sshd: ubuntu
> tcp6       0      0 :::80                   :::*                    LISTEN      1166/nginx -g daemo
> tcp6       0      0 :::8021                 :::*                    LISTEN      15894/freeswitch
> tcp6       0      0 :::22                   :::*                    LISTEN      1079/sshd
> tcp6       0      0 ::1:5080                :::*                    LISTEN      15894/freeswitch
> tcp6       0      0 :::443                  :::*                    LISTEN      1166/nginx -g daemo
> tcp6       0      0 ::1:5060                :::*                    LISTEN      15894/freeswitch
> udp        0      0 192.168.230.143:5060    0.0.0.0:*                           15894/freeswitch
> udp        0      0 192.168.230.143:5080    0.0.0.0:*                           15894/freeswitch
> udp        0      0 0.0.0.0:68              0.0.0.0:*                           1019/dhclient
> udp6       0      0 ::1:5060                :::*                                15894/freeswitch
> udp6       0      0 ::1:5080                :::*                                15894/freeswitch
>
> *$ fs_cli -x "sofia status"*
>
>                      Name      Type                                       Data  State
> =================================================================================================
>             external-ipv6   profile                   sip:mod_sofia@[::1]:5080  RUNNING (0)
>           192.168.230.143     alias                                   internal  ALIASED
>                  external   profile         sip:mod_sofia at 192.168.230.143:5080  RUNNING (0)
>     external::example.com   gateway                    sip:joeuser at example.com  NOREG
>             internal-ipv6   profile                   sip:mod_sofia@[::1]:5060  RUNNING (0)
>                  internal   profile         sip:mod_sofia at 192.168.230.143:5060  RUNNING (0)
> =================================================================================================
>
> *$ fs_cli -x "sofia status profile external"*
>
> =================================================================================================
> Name                external
> Domain Name         N/A
> Auto-NAT            false
> DBName              sofia_reg_external
> Pres Hosts
> Dialplan            XML
> Context             public
> Challenge Realm     auto_to
> RTP-IP              192.168.230.143
> SIP-IP              192.168.230.143
> URL                 sip:mod_sofia at 192.168.230.143:5080
> BIND-URL            sip:mod_sofia at 192.168.230.143:5080;transport=udp,tcp
> HOLD-MUSIC          local_stream://moh
> OUTBOUND-PROXY      N/A
> CODECS IN           OPUS,G722,PCMU,PCMA,VP8,H264
> CODECS OUT          OPUS,G722,PCMU,PCMA,VP8,H264
> TEL-EVENT           101
> DTMF-MODE           rfc2833
> CNG                 13
> SESSION-TO          0
> MAX-DIALOG          0
> NOMEDIA             false
> LATE-NEG            true
> PROXY-MEDIA         false
> ZRTP-PASSTHRU       true
> AGGRESSIVENAT       false
> CALLS-IN            0
> FAILED-CALLS-IN     0
> CALLS-OUT           0
> FAILED-CALLS-OUT    0
> REGISTRATIONS       0
>
> *$ grep -r '"ext-' **
>
> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
> sip_profiles/internal.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
> sip_profiles/internal.xml:    <param name="ext-sip-ip" value="auto-nat"/>
>
> When I perform a call the freeswitch.log file show:
>
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1009 at 192.168.230.143:5060 [df895d9a-30f2-40b5-97bb-56e222c1c45f]
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_NEW (Cur 1 Tot 7)
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite from 88.33.111.22:57076 version: 1.9.0 -493-13f2f2a 64bit
> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10238 IP 88.33.111.22 Rejected by acl "domains". Falling back to Digest auth.
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.230.143:5060) State NEW
> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:2405 detaching session df895d9a-30f2-40b5-97bb-56e222c1c45f
>
> and passed some seconds:
>
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [WARNING] switch_core_state_machine.c:687 df895d9a-30f2-40b5-97bb-56e222c1c45f sofia/internal/1009 at 192.168.230.143:5060 Abandoned
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/1009 at 192.168.230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_HANGUP (Cur 1 Tot 7)
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.230.143:5060) Callstate Change DOWN -> HANGUP
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060 hanging up, cause: WRONG_CALL_STATE
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060 Standard HANGUP, cause: WRONG_CALL_STATE
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP going to sleep
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_HANGUP -> CS_REPORTING
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_REPORTING (Cur 1 Tot 7)
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060 Standard REPORTING, cause: WRONG_CALL_STATE
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING going to sleep
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_REPORTING -> CS_DESTROY
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_session.c:1713 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Locked, Waiting on external entities
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1731 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Ended
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1009 at 192.168.230.143:5060 [CS_DESTROY]
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_DESTROY (Cur 0 Tot 7)
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060 Standard DESTROY
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY going to sleep
>
> Can it be an ACL or sip_profile bad configuration?? Router firewall
> configuration maybe?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Wed Jul 19 14:27:25 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 19 Jul 2017 16:27:25 +0200
Subject: [Freeswitch-users] Freeswitch SIP call not work using a static
 public IP
In-Reply-To: <CANeCd7qgLdOv8AOnZUGFfpn8DmXVEUDNUYRarfMmrfmwZurHbA@mail.gmail.com>
References: <CANeCd7qgLdOv8AOnZUGFfpn8DmXVEUDNUYRarfMmrfmwZurHbA@mail.gmail.com>
Message-ID: <CAFGRPVq4dk_Fj5ev-AvySBSTG=jS-h3JkV35C7JiX44Y29oFSg@mail.gmail.com>

You should set ext-ip and ext-rtp to the actual public IPs.
ACl is rejecting the call, but it's falling back to digest, are you sure
the calling user is configured in the directory? Can the client register
properly?
ᐧ

Regards,

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

On Wed, Jul 19, 2017 at 8:43 AM, Jose David Jurado Alonso <
josedavid at zennio.com> wrote:

>
> I've installed *freeswitch into a local environment* and I perform SIP
> calls well using the server local IP.
>
> This server has connected to ADSL network with a *static public IP* but
> when I try to call using the public IP instead of the local IP the server
> receive the call not work...
>
> The clients can register well (or I think so). Others process like apache
> work well.
>
> I NAT forwarding in the firewall the next ports: 5060, 5080, 7443.
>
> *$ netstat -putan*
>
> Active Internet connections (servers and established)
> Proto Recv-Q Send-Q Local Address           Foreign Address         State       PID/Program name
> tcp        0      0 192.168.230.143:2855    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 192.168.230.143:2856    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 192.168.230.143:5066    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN      1166/nginx -g daemo
> tcp        0      0 192.168.230.143:7443    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN      1079/sshd
> tcp        0      0 192.168.230.143:5080    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 0.0.0.0:443             0.0.0.0:*               LISTEN      1166/nginx -g daemo
> tcp        0      0 192.168.230.143:5060    0.0.0.0:*               LISTEN      15894/freeswitch
> tcp        0      0 192.168.230.143:22      192.168.230.167:60298   ESTABLISHED 14816/sshd: ubuntu
> tcp        0      0 192.168.230.143:22      192.168.230.167:60296   ESTABLISHED 14744/sshd: ubuntu
> tcp6       0      0 :::80                   :::*                    LISTEN      1166/nginx -g daemo
> tcp6       0      0 :::8021                 :::*                    LISTEN      15894/freeswitch
> tcp6       0      0 :::22                   :::*                    LISTEN      1079/sshd
> tcp6       0      0 ::1:5080                :::*                    LISTEN      15894/freeswitch
> tcp6       0      0 :::443                  :::*                    LISTEN      1166/nginx -g daemo
> tcp6       0      0 ::1:5060                :::*                    LISTEN      15894/freeswitch
> udp        0      0 192.168.230.143:5060    0.0.0.0:*                           15894/freeswitch
> udp        0      0 192.168.230.143:5080    0.0.0.0:*                           15894/freeswitch
> udp        0      0 0.0.0.0:68              0.0.0.0:*                           1019/dhclient
> udp6       0      0 ::1:5060                :::*                                15894/freeswitch
> udp6       0      0 ::1:5080                :::*                                15894/freeswitch
>
> *$ fs_cli -x "sofia status"*
>
>                      Name      Type                                       Data  State
> =================================================================================================
>             external-ipv6   profile                   sip:mod_sofia@[::1]:5080  RUNNING (0)
>           192.168.230.143     alias                                   internal  ALIASED
>                  external   profile         sip:mod_sofia at 192.168.230.143:5080  RUNNING (0)
>     external::example.com   gateway                    sip:joeuser at example.com  NOREG
>             internal-ipv6   profile                   sip:mod_sofia@[::1]:5060  RUNNING (0)
>                  internal   profile         sip:mod_sofia at 192.168.230.143:5060  RUNNING (0)
> =================================================================================================
>
> *$ fs_cli -x "sofia status profile external"*
>
> =================================================================================================
> Name                external
> Domain Name         N/A
> Auto-NAT            false
> DBName              sofia_reg_external
> Pres Hosts
> Dialplan            XML
> Context             public
> Challenge Realm     auto_to
> RTP-IP              192.168.230.143
> SIP-IP              192.168.230.143
> URL                 sip:mod_sofia at 192.168.230.143:5080
> BIND-URL            sip:mod_sofia at 192.168.230.143:5080;transport=udp,tcp
> HOLD-MUSIC          local_stream://moh
> OUTBOUND-PROXY      N/A
> CODECS IN           OPUS,G722,PCMU,PCMA,VP8,H264
> CODECS OUT          OPUS,G722,PCMU,PCMA,VP8,H264
> TEL-EVENT           101
> DTMF-MODE           rfc2833
> CNG                 13
> SESSION-TO          0
> MAX-DIALOG          0
> NOMEDIA             false
> LATE-NEG            true
> PROXY-MEDIA         false
> ZRTP-PASSTHRU       true
> AGGRESSIVENAT       false
> CALLS-IN            0
> FAILED-CALLS-IN     0
> CALLS-OUT           0
> FAILED-CALLS-OUT    0
> REGISTRATIONS       0
>
> *$ grep -r '"ext-' **
>
> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
> sip_profiles/internal.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
> sip_profiles/internal.xml:    <param name="ext-sip-ip" value="auto-nat"/>
>
> When I perform a call the freeswitch.log file show:
>
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1009 at 192.168.230.143:5060 [df895d9a-30f2-40b5-97bb-56e222c1c45f]
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_NEW (Cur 1 Tot 7)
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite from 88.33.111.22:57076 version: 1.9.0 -493-13f2f2a 64bit
> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10238 IP 88.33.111.22 Rejected by acl "domains". Falling back to Digest auth.
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.230.143:5060) State NEW
> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:2405 detaching session df895d9a-30f2-40b5-97bb-56e222c1c45f
>
> and passed some seconds:
>
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [WARNING] switch_core_state_machine.c:687 df895d9a-30f2-40b5-97bb-56e222c1c45f sofia/internal/1009 at 192.168.230.143:5060 Abandoned
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/1009 at 192.168.230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_HANGUP (Cur 1 Tot 7)
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.230.143:5060) Callstate Change DOWN -> HANGUP
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060 hanging up, cause: WRONG_CALL_STATE
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060 Standard HANGUP, cause: WRONG_CALL_STATE
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP going to sleep
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_HANGUP -> CS_REPORTING
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_REPORTING (Cur 1 Tot 7)
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060 Standard REPORTING, cause: WRONG_CALL_STATE
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING going to sleep
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_REPORTING -> CS_DESTROY
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_session.c:1713 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Locked, Waiting on external entities
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1731 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Ended
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1009 at 192.168.230.143:5060 [CS_DESTROY]
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_DESTROY (Cur 0 Tot 7)
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060 Standard DESTROY
> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY going to sleep
>
> Can it be an ACL or sip_profile bad configuration?? Router firewall
> configuration maybe?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Wed Jul 19 14:30:37 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 19 Jul 2017 16:30:37 +0200
Subject: [Freeswitch-users] Freeswitch SIP call not work using a static
 public IP
In-Reply-To: <CAMtXxQm8oj9TfJVLfmukL7BrEEkQAA0GB3xrmxGOnn5fbLGE=Q@mail.gmail.com>
References: <CANeCd7qgLdOv8AOnZUGFfpn8DmXVEUDNUYRarfMmrfmwZurHbA@mail.gmail.com>
 <CAMtXxQm8oj9TfJVLfmukL7BrEEkQAA0GB3xrmxGOnn5fbLGE=Q@mail.gmail.com>
Message-ID: <CAFGRPVp2_=WpG9SWctEKVdYEn5_dwi1pE7gys0KXrHUg3OXSEQ@mail.gmail.com>

+1 about forwarding upds...
But looking at the logs, we can't really see whether the call was processed
or not, just that it was "abandoned" and that usually happens when
freeswitch responds with a 401/407 and the client never authenticates...

enter:
sofia profile internal siptrace on

to see all messages and make sure the client responds properly and paste
the signaling here.
ᐧ

Regards,

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

On Wed, Jul 19, 2017 at 4:18 PM, Joel Serrano <joel at gogii.net> wrote:

> Double check your ACLs, yes. Also, I'm pretty sure you will also have to
> forward the RTP port range so you can have audio... Otherwise signaling
> will work, but your next problem will be one-way-audio.
>
>
> On Tue, Jul 18, 2017 at 11:43 PM, Jose David Jurado Alonso <
> josedavid at zennio.com> wrote:
>
>>
>> I've installed *freeswitch into a local environment* and I perform SIP
>> calls well using the server local IP.
>>
>> This server has connected to ADSL network with a *static public IP* but
>> when I try to call using the public IP instead of the local IP the server
>> receive the call not work...
>>
>> The clients can register well (or I think so). Others process like apache
>> work well.
>>
>> I NAT forwarding in the firewall the next ports: 5060, 5080, 7443.
>>
>> *$ netstat -putan*
>>
>> Active Internet connections (servers and established)
>> Proto Recv-Q Send-Q Local Address           Foreign Address         State       PID/Program name
>> tcp        0      0 192.168.230.143:2855    0.0.0.0:*               LISTEN      15894/freeswitch
>> tcp        0      0 192.168.230.143:2856    0.0.0.0:*               LISTEN      15894/freeswitch
>> tcp        0      0 192.168.230.143:5066    0.0.0.0:*               LISTEN      15894/freeswitch
>> tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN      1166/nginx -g daemo
>> tcp        0      0 192.168.230.143:7443    0.0.0.0:*               LISTEN      15894/freeswitch
>> tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN      1079/sshd
>> tcp        0      0 192.168.230.143:5080    0.0.0.0:*               LISTEN      15894/freeswitch
>> tcp        0      0 0.0.0.0:443             0.0.0.0:*               LISTEN      1166/nginx -g daemo
>> tcp        0      0 192.168.230.143:5060    0.0.0.0:*               LISTEN      15894/freeswitch
>> tcp        0      0 192.168.230.143:22      192.168.230.167:60298   ESTABLISHED 14816/sshd: ubuntu
>> tcp        0      0 192.168.230.143:22      192.168.230.167:60296   ESTABLISHED 14744/sshd: ubuntu
>> tcp6       0      0 :::80                   :::*                    LISTEN      1166/nginx -g daemo
>> tcp6       0      0 :::8021                 :::*                    LISTEN      15894/freeswitch
>> tcp6       0      0 :::22                   :::*                    LISTEN      1079/sshd
>> tcp6       0      0 ::1:5080                :::*                    LISTEN      15894/freeswitch
>> tcp6       0      0 :::443                  :::*                    LISTEN      1166/nginx -g daemo
>> tcp6       0      0 ::1:5060                :::*                    LISTEN      15894/freeswitch
>> udp        0      0 192.168.230.143:5060    0.0.0.0:*                           15894/freeswitch
>> udp        0      0 192.168.230.143:5080    0.0.0.0:*                           15894/freeswitch
>> udp        0      0 0.0.0.0:68              0.0.0.0:*                           1019/dhclient
>> udp6       0      0 ::1:5060                :::*                                15894/freeswitch
>> udp6       0      0 ::1:5080                :::*                                15894/freeswitch
>>
>> *$ fs_cli -x "sofia status"*
>>
>>                      Name      Type                                       Data  State
>> =================================================================================================
>>             external-ipv6   profile                   sip:mod_sofia@[::1]:5080  RUNNING (0)
>>           192.168.230.143     alias                                   internal  ALIASED
>>                  external   profile         sip:mod_sofia at 192.168.230.143:5080  RUNNING (0)
>>     external::example.com   gateway                    sip:joeuser at example.com  NOREG
>>             internal-ipv6   profile                   sip:mod_sofia@[::1]:5060  RUNNING (0)
>>                  internal   profile         sip:mod_sofia at 192.168.230.143:5060  RUNNING (0)
>> =================================================================================================
>>
>> *$ fs_cli -x "sofia status profile external"*
>>
>> =================================================================================================
>> Name                external
>> Domain Name         N/A
>> Auto-NAT            false
>> DBName              sofia_reg_external
>> Pres Hosts
>> Dialplan            XML
>> Context             public
>> Challenge Realm     auto_to
>> RTP-IP              192.168.230.143
>> SIP-IP              192.168.230.143
>> URL                 sip:mod_sofia at 192.168.230.143:5080
>> BIND-URL            sip:mod_sofia at 192.168.230.143:5080;transport=udp,tcp
>> HOLD-MUSIC          local_stream://moh
>> OUTBOUND-PROXY      N/A
>> CODECS IN           OPUS,G722,PCMU,PCMA,VP8,H264
>> CODECS OUT          OPUS,G722,PCMU,PCMA,VP8,H264
>> TEL-EVENT           101
>> DTMF-MODE           rfc2833
>> CNG                 13
>> SESSION-TO          0
>> MAX-DIALOG          0
>> NOMEDIA             false
>> LATE-NEG            true
>> PROXY-MEDIA         false
>> ZRTP-PASSTHRU       true
>> AGGRESSIVENAT       false
>> CALLS-IN            0
>> FAILED-CALLS-IN     0
>> CALLS-OUT           0
>> FAILED-CALLS-OUT    0
>> REGISTRATIONS       0
>>
>> *$ grep -r '"ext-' **
>>
>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>> sip_profiles/internal.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>> sip_profiles/internal.xml:    <param name="ext-sip-ip" value="auto-nat"/>
>>
>> When I perform a call the freeswitch.log file show:
>>
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1009 at 192.168.230.143:5060 [df895d9a-30f2-40b5-97bb-56e222c1c45f]
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_NEW (Cur 1 Tot 7)
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite from 88.33.111.22:57076 version: 1.9.0 -493-13f2f2a 64bit
>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10238 IP 88.33.111.22 Rejected by acl "domains". Falling back to Digest auth.
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.230.143:5060) State NEW
>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:2405 detaching session df895d9a-30f2-40b5-97bb-56e222c1c45f
>>
>> and passed some seconds:
>>
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [WARNING] switch_core_state_machine.c:687 df895d9a-30f2-40b5-97bb-56e222c1c45f sofia/internal/1009 at 192.168.230.143:5060 Abandoned
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/1009 at 192.168.230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_HANGUP (Cur 1 Tot 7)
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.230.143:5060) Callstate Change DOWN -> HANGUP
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060 hanging up, cause: WRONG_CALL_STATE
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060 Standard HANGUP, cause: WRONG_CALL_STATE
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP going to sleep
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_HANGUP -> CS_REPORTING
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_REPORTING (Cur 1 Tot 7)
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060 Standard REPORTING, cause: WRONG_CALL_STATE
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING going to sleep
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_REPORTING -> CS_DESTROY
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_session.c:1713 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Locked, Waiting on external entities
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1731 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Ended
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1009 at 192.168.230.143:5060 [CS_DESTROY]
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_DESTROY (Cur 0 Tot 7)
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060 Standard DESTROY
>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY going to sleep
>>
>> Can it be an ACL or sip_profile bad configuration?? Router firewall
>> configuration maybe?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.villasmil.work at gmail.com  Wed Jul 19 14:46:45 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 19 Jul 2017 16:46:45 +0200
Subject: [Freeswitch-users] traffic and system monitoring
In-Reply-To: <dd31c0e0-74b5-3a5a-9236-62bec7948049@paulzillmann.de>
References: <1012898889.2268619.1500304838441.ref@mail.yahoo.com>
 <1012898889.2268619.1500304838441@mail.yahoo.com>
 <dd31c0e0-74b5-3a5a-9236-62bec7948049@paulzillmann.de>
Message-ID: <CAFGRPVqTMX57bsvWs9xM8ofLFdbs6Tfim3pbVQjC0MSB9CcbkQ@mail.gmail.com>

Although HOMER is not bad for monitoring things like ACD/ASR GLOBALLY, it
is not designed to be used as a monitoring tool. It is a tool to have the
signaling handy for troubleshooting (it can also keep logs, RTCP, and other
things), and for this, I don't know anything better that HOMER. HOMER is
specifically designed to troubleshoot VoIP.

I've never used CACTI, however, it doesn't look to be a SIP monitoring
tool, but a network monitoring tool, able to monitor CPU, MEMORY, ETC... It
won't monitor ACD, ASR, NER (Although i suppose you can write some
plugins)... which HOMER does GLOBALLY...

You could extend HOMER to monitor on a per-host level (I did a P/R just for
this, but it hasn't been merged yet, as there's no GUI to show that data
and I hate AngularJS)

In conclusion, Maybe you should think about using both, as they seem to
complement each other.

Hope this helps

Regards,

David
ᐧ

Regards,

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

On Wed, Jul 19, 2017 at 3:24 PM, Paul Zillmann <mail at paulzillmann.de> wrote:

> Hey Robert,
>
> I'm using check_mk / nagios for external monitoring and iftop and top /
> htop for on site analysis.
>
>
> Am 17.07.2017 um 17:20 schrieb robert mundkowsky:
>
>
>
> Is Homer the best free option for traffic and system monitoring GUI?
>
>                Or is CACTI better option?
>
>
>
> Robert
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rfmundkowsky at yahoo.com  Wed Jul 19 18:01:55 2017
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Wed, 19 Jul 2017 18:01:55 +0000 (UTC)
Subject: [Freeswitch-users] traffic and system monitoring
In-Reply-To: <dd31c0e0-74b5-3a5a-9236-62bec7948049@paulzillmann.de>
References: <1012898889.2268619.1500304838441.ref@mail.yahoo.com>
 <1012898889.2268619.1500304838441@mail.yahoo.com>
 <dd31c0e0-74b5-3a5a-9236-62bec7948049@paulzillmann.de>
Message-ID: <2016365814.617648.1500487315036@mail.yahoo.com>

Are check_mk / nagios free?

Looks like there is a SIP plugin. Is there a RTP plugin?



On Wednesday, July 19, 2017, 9:25:36 AM EDT, Paul Zillmann <mail at paulzillmann.de> wrote:

 Hey Robert,
 
 I'm using check_mk / nagios for external monitoring and iftop and top / htop for on site analysis.
 
 
 Am 17.07.2017 um 17:20 schrieb robert mundkowsky:
  
   
 
 
Is Homer the best free option for traffic and system monitoring GUI?
 
               Or is CACTI better option?
 
 
 
Robert 
   
  
 _________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 
 
 _________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From rfmundkowsky at yahoo.com  Wed Jul 19 18:02:10 2017
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Wed, 19 Jul 2017 18:02:10 +0000 (UTC)
Subject: [Freeswitch-users] traffic and system monitoring
In-Reply-To: <CAFGRPVqTMX57bsvWs9xM8ofLFdbs6Tfim3pbVQjC0MSB9CcbkQ@mail.gmail.com>
References: <1012898889.2268619.1500304838441.ref@mail.yahoo.com>
 <1012898889.2268619.1500304838441@mail.yahoo.com>
 <dd31c0e0-74b5-3a5a-9236-62bec7948049@paulzillmann.de>
 <CAFGRPVqTMX57bsvWs9xM8ofLFdbs6Tfim3pbVQjC0MSB9CcbkQ@mail.gmail.com>
Message-ID: <711495238.602117.1500487330688@mail.yahoo.com>

thanks.  This makes sense. 



On Wednesday, July 19, 2017, 10:47:54 AM EDT, David Villasmil <david.villasmil.work at gmail.com> wrote:

Although HOMER is not bad for monitoring things like ACD/ASR GLOBALLY, it is not designed to be used as a monitoring tool. It is a tool to have the signaling handy for troubleshooting (it can also keep logs, RTCP, and other things), and for this, I don't know anything better that HOMER. HOMER is specifically designed to troubleshoot VoIP.
I've never used CACTI, however, it doesn't look to be a SIP monitoring tool, but a network monitoring tool, able to monitor CPU, MEMORY, ETC... It won't monitor ACD, ASR, NER (Although i suppose you can write some plugins)... which HOMER does GLOBALLY...
You could extend HOMER to monitor on a per-host level (I did a P/R just for this, but it hasn't been merged yet, as there's no GUI to show that data and I hate AngularJS)
In conclusion, Maybe you should think about using both, as they seem to complement each other.
Hope this helps
Regards,
Davidᐧ
Regards,
David Villasmilemail: david.villasmil.work at gmail.comphone: +34669448337
On Wed, Jul 19, 2017 at 3:24 PM, Paul Zillmann <mail at paulzillmann.de> wrote:

  Hey Robert,
 
 I'm using check_mk / nagios for external monitoring and iftop and top / htop for on site analysis.
 
 
 Am 17.07.2017 um 17:20 schrieb robert mundkowsky:
  
   
 
 
Is Homer the best free option for traffic and system monitoring GUI?
 
               Or is CACTI better option?
 
 
 
Robert 
   
  
 ______________________________ ______________________________ _____________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www. freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists. freeswitch.org
http://lists.freeswitch.org/ mailman/listinfo/freeswitch- users
UNSUBSCRIBE:http://lists. freeswitch.org/mailman/ options/freeswitch-users
http://www.freeswitch.org 
 
 
______________________________ ______________________________ _____________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www. freeswitchsolutions.com

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

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


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From servtelar at gmail.com  Wed Jul 19 18:21:49 2017
From: servtelar at gmail.com (GM phy)
Date: Wed, 19 Jul 2017 15:21:49 -0300
Subject: [Freeswitch-users] traffic and system monitoring
In-Reply-To: <CAFGRPVqTMX57bsvWs9xM8ofLFdbs6Tfim3pbVQjC0MSB9CcbkQ@mail.gmail.com>
References: <1012898889.2268619.1500304838441.ref@mail.yahoo.com>
 <1012898889.2268619.1500304838441@mail.yahoo.com>
 <dd31c0e0-74b5-3a5a-9236-62bec7948049@paulzillmann.de>
 <CAFGRPVqTMX57bsvWs9xM8ofLFdbs6Tfim3pbVQjC0MSB9CcbkQ@mail.gmail.com>
Message-ID: <CAN8B5B=qb+Hr89AqaQ6cKkJobEH_MT1vHSAjSOK8YV6zShbJsw@mail.gmail.com>

Take a look on this:
https://www.slideshare.net/MoisesSilva6/freeswitch-monitoring

There are good ideas there.

Regards

Gustavo


On Wed, Jul 19, 2017 at 11:46 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Although HOMER is not bad for monitoring things like ACD/ASR GLOBALLY, it
> is not designed to be used as a monitoring tool. It is a tool to have the
> signaling handy for troubleshooting (it can also keep logs, RTCP, and other
> things), and for this, I don't know anything better that HOMER. HOMER is
> specifically designed to troubleshoot VoIP.
>
> I've never used CACTI, however, it doesn't look to be a SIP monitoring
> tool, but a network monitoring tool, able to monitor CPU, MEMORY, ETC... It
> won't monitor ACD, ASR, NER (Although i suppose you can write some
> plugins)... which HOMER does GLOBALLY...
>
> You could extend HOMER to monitor on a per-host level (I did a P/R just
> for this, but it hasn't been merged yet, as there's no GUI to show that
> data and I hate AngularJS)
>
> In conclusion, Maybe you should think about using both, as they seem to
> complement each other.
>
> Hope this helps
>
> Regards,
>
> David
> ᐧ
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
> On Wed, Jul 19, 2017 at 3:24 PM, Paul Zillmann <mail at paulzillmann.de>
> wrote:
>
>> Hey Robert,
>>
>> I'm using check_mk / nagios for external monitoring and iftop and top /
>> htop for on site analysis.
>>
>>
>> Am 17.07.2017 um 17:20 schrieb robert mundkowsky:
>>
>>
>>
>> Is Homer the best free option for traffic and system monitoring GUI?
>>
>>                Or is CACTI better option?
>>
>>
>>
>> Robert
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Jul 19 19:00:08 2017
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 Jul 2017 14:00:08 -0500
Subject: [Freeswitch-users] Freeswitch SIP call not work using a static
 public IP
In-Reply-To: <CAFGRPVp2_=WpG9SWctEKVdYEn5_dwi1pE7gys0KXrHUg3OXSEQ@mail.gmail.com>
References: <CANeCd7qgLdOv8AOnZUGFfpn8DmXVEUDNUYRarfMmrfmwZurHbA@mail.gmail.com>
 <CAMtXxQm8oj9TfJVLfmukL7BrEEkQAA0GB3xrmxGOnn5fbLGE=Q@mail.gmail.com>
 <CAFGRPVp2_=WpG9SWctEKVdYEn5_dwi1pE7gys0KXrHUg3OXSEQ@mail.gmail.com>
Message-ID: <CAEJMVkDh2vrGnxWagzKyWYp9+aR+Q5NaVsjtLvPHQVTs46Nmjw@mail.gmail.com>

1. set up your local-network-acl . (rfc1918.auto is good start)
2. setup your ext-rtp-ip and ext-sip-ip and MAKE SURE you prefix it with
autonat: so it activates the proper behavior for you.

This should be all you have to do so you can talk to devices inside and
outside the NAT at the same time.

/b


On Wed, Jul 19, 2017 at 9:30 AM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> +1 about forwarding upds...
> But looking at the logs, we can't really see whether the call was
> processed or not, just that it was "abandoned" and that usually happens
> when freeswitch responds with a 401/407 and the client never
> authenticates...
>
> enter:
> sofia profile internal siptrace on
>
> to see all messages and make sure the client responds properly and paste
> the signaling here.
> ᐧ
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337 <+34%20669%2044%2083%2037>
>
> On Wed, Jul 19, 2017 at 4:18 PM, Joel Serrano <joel at gogii.net> wrote:
>
>> Double check your ACLs, yes. Also, I'm pretty sure you will also have to
>> forward the RTP port range so you can have audio... Otherwise signaling
>> will work, but your next problem will be one-way-audio.
>>
>>
>> On Tue, Jul 18, 2017 at 11:43 PM, Jose David Jurado Alonso <
>> josedavid at zennio.com> wrote:
>>
>>>
>>> I've installed *freeswitch into a local environment* and I perform SIP
>>> calls well using the server local IP.
>>>
>>> This server has connected to ADSL network with a *static public IP* but
>>> when I try to call using the public IP instead of the local IP the server
>>> receive the call not work...
>>>
>>> The clients can register well (or I think so). Others process like
>>> apache work well.
>>>
>>> I NAT forwarding in the firewall the next ports: 5060, 5080, 7443.
>>>
>>> *$ netstat -putan*
>>>
>>> Active Internet connections (servers and established)
>>> Proto Recv-Q Send-Q Local Address           Foreign Address         State       PID/Program name
>>> tcp        0      0 192.168.230.143:2855    0.0.0.0:*               LISTEN      15894/freeswitch
>>> tcp        0      0 192.168.230.143:2856    0.0.0.0:*               LISTEN      15894/freeswitch
>>> tcp        0      0 192.168.230.143:5066    0.0.0.0:*               LISTEN      15894/freeswitch
>>> tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN      1166/nginx -g daemo
>>> tcp        0      0 192.168.230.143:7443    0.0.0.0:*               LISTEN      15894/freeswitch
>>> tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN      1079/sshd
>>> tcp        0      0 192.168.230.143:5080    0.0.0.0:*               LISTEN      15894/freeswitch
>>> tcp        0      0 0.0.0.0:443             0.0.0.0:*               LISTEN      1166/nginx -g daemo
>>> tcp        0      0 192.168.230.143:5060    0.0.0.0:*               LISTEN      15894/freeswitch
>>> tcp        0      0 192.168.230.143:22      192.168.230.167:60298   ESTABLISHED 14816/sshd: ubuntu
>>> tcp        0      0 192.168.230.143:22      192.168.230.167:60296   ESTABLISHED 14744/sshd: ubuntu
>>> tcp6       0      0 :::80                   :::*                    LISTEN      1166/nginx -g daemo
>>> tcp6       0      0 :::8021                 :::*                    LISTEN      15894/freeswitch
>>> tcp6       0      0 :::22                   :::*                    LISTEN      1079/sshd
>>> tcp6       0      0 ::1:5080                :::*                    LISTEN      15894/freeswitch
>>> tcp6       0      0 :::443                  :::*                    LISTEN      1166/nginx -g daemo
>>> tcp6       0      0 ::1:5060                :::*                    LISTEN      15894/freeswitch
>>> udp        0      0 192.168.230.143:5060    0.0.0.0:*                           15894/freeswitch
>>> udp        0      0 192.168.230.143:5080    0.0.0.0:*                           15894/freeswitch
>>> udp        0      0 0.0.0.0:68              0.0.0.0:*                           1019/dhclient
>>> udp6       0      0 ::1:5060                :::*                                15894/freeswitch
>>> udp6       0      0 ::1:5080                :::*                                15894/freeswitch
>>>
>>> *$ fs_cli -x "sofia status"*
>>>
>>>                      Name      Type                                       Data  State
>>> =================================================================================================
>>>             external-ipv6   profile                   sip:mod_sofia@[::1]:5080  RUNNING (0)
>>>           192.168.230.143     alias                                   internal  ALIASED
>>>                  external   profile         sip:mod_sofia at 192.168.230.143:5080  RUNNING (0)
>>>     external::example.com   gateway                    sip:joeuser at example.com  NOREG
>>>             internal-ipv6   profile                   sip:mod_sofia@[::1]:5060  RUNNING (0)
>>>                  internal   profile         sip:mod_sofia at 192.168.230.143:5060  RUNNING (0)
>>> =================================================================================================
>>>
>>> *$ fs_cli -x "sofia status profile external"*
>>>
>>> =================================================================================================
>>> Name                external
>>> Domain Name         N/A
>>> Auto-NAT            false
>>> DBName              sofia_reg_external
>>> Pres Hosts
>>> Dialplan            XML
>>> Context             public
>>> Challenge Realm     auto_to
>>> RTP-IP              192.168.230.143
>>> SIP-IP              192.168.230.143
>>> URL                 sip:mod_sofia at 192.168.230.143:5080
>>> BIND-URL            sip:mod_sofia at 192.168.230.143:5080;transport=udp,tcp
>>> HOLD-MUSIC          local_stream://moh
>>> OUTBOUND-PROXY      N/A
>>> CODECS IN           OPUS,G722,PCMU,PCMA,VP8,H264
>>> CODECS OUT          OPUS,G722,PCMU,PCMA,VP8,H264
>>> TEL-EVENT           101
>>> DTMF-MODE           rfc2833
>>> CNG                 13
>>> SESSION-TO          0
>>> MAX-DIALOG          0
>>> NOMEDIA             false
>>> LATE-NEG            true
>>> PROXY-MEDIA         false
>>> ZRTP-PASSTHRU       true
>>> AGGRESSIVENAT       false
>>> CALLS-IN            0
>>> FAILED-CALLS-IN     0
>>> CALLS-OUT           0
>>> FAILED-CALLS-OUT    0
>>> REGISTRATIONS       0
>>>
>>> *$ grep -r '"ext-' **
>>>
>>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>> sip_profiles/internal.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>> sip_profiles/internal.xml:    <param name="ext-sip-ip" value="auto-nat"/>
>>>
>>> When I perform a call the freeswitch.log file show:
>>>
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1009 at 192.168.230.143:5060 [df895d9a-30f2-40b5-97bb-56e222c1c45f]
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_NEW (Cur 1 Tot 7)
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite from 88.33.111.22:57076 version: 1.9.0 -493-13f2f2a 64bit
>>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10238 IP 88.33.111.22 Rejected by acl "domains". Falling back to Digest auth.
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.230.143:5060) State NEW
>>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:2405 detaching session df895d9a-30f2-40b5-97bb-56e222c1c45f
>>>
>>> and passed some seconds:
>>>
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [WARNING] switch_core_state_machine.c:687 df895d9a-30f2-40b5-97bb-56e222c1c45f sofia/internal/1009 at 192.168.230.143:5060 Abandoned
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/1009 at 192.168.230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_HANGUP (Cur 1 Tot 7)
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.230.143:5060) Callstate Change DOWN -> HANGUP
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060 hanging up, cause: WRONG_CALL_STATE
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060 Standard HANGUP, cause: WRONG_CALL_STATE
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP going to sleep
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_HANGUP -> CS_REPORTING
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_REPORTING (Cur 1 Tot 7)
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060 Standard REPORTING, cause: WRONG_CALL_STATE
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING going to sleep
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_REPORTING -> CS_DESTROY
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_session.c:1713 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Locked, Waiting on external entities
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1731 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Ended
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1009 at 192.168.230.143:5060 [CS_DESTROY]
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_DESTROY (Cur 0 Tot 7)
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060 Standard DESTROY
>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY going to sleep
>>>
>>> Can it be an ACL or sip_profile bad configuration?? Router firewall
>>> configuration maybe?
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From joel at gogii.net  Wed Jul 19 19:40:54 2017
From: joel at gogii.net (Joel Serrano)
Date: Wed, 19 Jul 2017 12:40:54 -0700
Subject: [Freeswitch-users] Adding video to audio call
In-Reply-To: <f01f26ec-d5da-b38c-9ad7-3174178c2685@xbipin.com>
References: <CACdNXfdsjhC5dbZGzNJxBsG4XnU1jtw6CpdTcqcwJmw2QGhnyw@mail.gmail.com>
 <CAHtxdDcns0e0oCaG8pHfKYYAeHhmbrihHg5EgbUWuatNpWnL5A@mail.gmail.com>
 <f01f26ec-d5da-b38c-9ad7-3174178c2685@xbipin.com>
Message-ID: <CAMtXxQnc5M8BCT9_c3-5kmfvOmCjeyy+zL-r8wEkZuYfb1jh1g@mail.gmail.com>

Hi,

Has anyone solved this in any way? We have the same situation, but we
cannot use proxy_media.

Can anyone explain a little what exactly "renegotiate-codec-on-reinvite"
does?




On Sat, Jun 17, 2017 at 11:02 PM, Bipin Patel <bipin at xbipin.com> wrote:

> hi,
>
> also i believe there is a bug request open on jira relating to this
>
>
> Regards,
> Bipin
>
>
> ------------------------------
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] Adding video to audio call
> From: Sergey Safarov <s.safarov at gmail.com> <s.safarov at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> <freeswitch-users at lists.freeswitch.org>
> Date: 6/17/2017, 10:39:25 PM
>
> Requred to enable proxy media.
>
> сб, 17 июня 2017 г., 19:20 Gauri Kshirsagar <ksh.sip at gmail.com>:
>
>> Hi,
>>
>> I am using Freeswitch version 1.9.0. I can make video calls. But when I
>> try to add video to audio call it does not work.
>>
>> A makes audio call to B. Call is established. A adds video .
>>
>> ReINVITE sent to freeswitch has  video added in SDP.Freeswitch sends 200
>> OK response for this INVITE to A which has video in SDP. But there is no
>> ReINVITE being sent to B.
>>
>> I tried enabling renegotiate-codec-on-reinvite in vars.xml and also
>> internal.xml
>>
>> Is this supported? If so is some other configuration required.
>>
>> Regards,
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joel at gogii.net  Wed Jul 19 19:42:08 2017
From: joel at gogii.net (Joel Serrano)
Date: Wed, 19 Jul 2017 12:42:08 -0700
Subject: [Freeswitch-users] Adding video to audio call
In-Reply-To: <CAMtXxQnc5M8BCT9_c3-5kmfvOmCjeyy+zL-r8wEkZuYfb1jh1g@mail.gmail.com>
References: <CACdNXfdsjhC5dbZGzNJxBsG4XnU1jtw6CpdTcqcwJmw2QGhnyw@mail.gmail.com>
 <CAHtxdDcns0e0oCaG8pHfKYYAeHhmbrihHg5EgbUWuatNpWnL5A@mail.gmail.com>
 <f01f26ec-d5da-b38c-9ad7-3174178c2685@xbipin.com>
 <CAMtXxQnc5M8BCT9_c3-5kmfvOmCjeyy+zL-r8wEkZuYfb1jh1g@mail.gmail.com>
Message-ID: <CAMtXxQmtBVs-ZoKc2XUF6JYjnEqKTTVA1tF3FAtLJbvv1WRjTA@mail.gmail.com>

BTW, the JIRA is: https://freeswitch.org/jira/browse/FS-7943



On Wed, Jul 19, 2017 at 12:40 PM, Joel Serrano <joel at gogii.net> wrote:

> Hi,
>
> Has anyone solved this in any way? We have the same situation, but we
> cannot use proxy_media.
>
> Can anyone explain a little what exactly "renegotiate-codec-on-reinvite"
> does?
>
>
>
>
> On Sat, Jun 17, 2017 at 11:02 PM, Bipin Patel <bipin at xbipin.com> wrote:
>
>> hi,
>>
>> also i believe there is a bug request open on jira relating to this
>>
>>
>> Regards,
>> Bipin
>>
>>
>> ------------------------------
>> -------- Original Message --------
>> Subject: Re: [Freeswitch-users] Adding video to audio call
>> From: Sergey Safarov <s.safarov at gmail.com> <s.safarov at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> Date: 6/17/2017, 10:39:25 PM
>>
>> Requred to enable proxy media.
>>
>> сб, 17 июня 2017 г., 19:20 Gauri Kshirsagar <ksh.sip at gmail.com>:
>>
>>> Hi,
>>>
>>> I am using Freeswitch version 1.9.0. I can make video calls. But when I
>>> try to add video to audio call it does not work.
>>>
>>> A makes audio call to B. Call is established. A adds video .
>>>
>>> ReINVITE sent to freeswitch has  video added in SDP.Freeswitch sends 200
>>> OK response for this INVITE to A which has video in SDP. But there is no
>>> ReINVITE being sent to B.
>>>
>>> I tried enabling renegotiate-codec-on-reinvite in vars.xml and also
>>> internal.xml
>>>
>>> Is this supported? If so is some other configuration required.
>>>
>>> Regards,
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From Alexander.Haugg at c4b.de  Thu Jul 20 06:55:03 2017
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 20 Jul 2017 06:55:03 +0000
Subject: [Freeswitch-users] Under buffering if i use ICE SDP
Message-ID: <83bbba3b51ef4770a0616291ddfa74fd@c4b.de>

Hi,

i have the problem with underbuffering if i use ICE SDP.
The good case:

-          ClientBria 4

-          Codec G711 ulaw

The bad case:

-          ICE Link API from Frozen Mountain

-          Codec G711 ulaw (no SRTP)

I have two Wireshark traces. In the bad case i can see and hear the simulated silent parts every (round about) 500 ms.
The bad case is independent of a codec. The same problem occured with G711 alaw and opus.

Is it a known problem, or have anybody an idea?
Is this a case for a Jira ticket?

Thanks a lot!

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From krice at freeswitch.org  Thu Jul 20 07:18:07 2017
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 20 Jul 2017 02:18:07 -0500
Subject: [Freeswitch-users] Under buffering if i use ICE SDP
In-Reply-To: <83bbba3b51ef4770a0616291ddfa74fd@c4b.de>
References: <83bbba3b51ef4770a0616291ddfa74fd@c4b.de>
Message-ID: <1A56CD20-19CE-4B02-A21D-E37E41EBF18D@freeswitch.org>

jira is always the place for bugs. 

have you tried using a webrtc based client like Verto? i have heard of no complaints there



Sent from my iPhone

> On Jul 20, 2017, at 01:55, Alexander Haugg <Alexander.Haugg at c4b.de> wrote:
> 
> Hi,
>  
> i have the problem with underbuffering if i use ICE SDP.
> The good case:
> -          ClientBria 4
> -          Codec G711 ulaw
>  
> The bad case:
> -          ICE Link API from Frozen Mountain
> -          Codec G711 ulaw (no SRTP)
>  
> I have two Wireshark traces. In the bad case i can see and hear the simulated silent parts every (round about) 500 ms.
> The bad case is independent of a codec. The same problem occured with G711 alaw and opus.
>  
> Is it a known problem, or have anybody an idea?
> Is this a case for a Jira ticket?
>  
> Thanks a lot!
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From Alexander.Haugg at c4b.de  Thu Jul 20 07:33:26 2017
From: Alexander.Haugg at c4b.de (Alexander Haugg)
Date: Thu, 20 Jul 2017 07:33:26 +0000
Subject: [Freeswitch-users] Under buffering if i use ICE SDP
In-Reply-To: <1A56CD20-19CE-4B02-A21D-E37E41EBF18D@freeswitch.org>
References: <83bbba3b51ef4770a0616291ddfa74fd@c4b.de>
 <1A56CD20-19CE-4B02-A21D-E37E41EBF18D@freeswitch.org>
Message-ID: <ea03d6a6de46446e95e9faacf1fcc485@c4b.de>

Not at the moment, but i think, the silent parts in the wireshark trace in the stream from Freeswitch to client could not be ok?!
I will open a Jira ticket.

Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Ken Rice
Gesendet: Donnerstag, 20. Juli 2017 09:18
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] Under buffering if i use ICE SDP

jira is always the place for bugs.

have you tried using a webrtc based client like Verto? i have heard of no complaints there



Sent from my iPhone

On Jul 20, 2017, at 01:55, Alexander Haugg <Alexander.Haugg at c4b.de<mailto:Alexander.Haugg at c4b.de>> wrote:
Hi,

i have the problem with underbuffering if i use ICE SDP.
The good case:

-          ClientBria 4

-          Codec G711 ulaw

The bad case:

-          ICE Link API from Frozen Mountain

-          Codec G711 ulaw (no SRTP)

I have two Wireshark traces. In the bad case i can see and hear the simulated silent parts every (round about) 500 ms.
The bad case is independent of a codec. The same problem occured with G711 alaw and opus.

Is it a known problem, or have anybody an idea?
Is this a case for a Jira ticket?

Thanks a lot!

_________________________________________________________________________
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From josedavid at zennio.com  Thu Jul 20 07:28:39 2017
From: josedavid at zennio.com (Jose David Jurado Alonso)
Date: Thu, 20 Jul 2017 09:28:39 +0200
Subject: [Freeswitch-users] Freeswitch SIP call not work using a static
 public IP
In-Reply-To: <CAEJMVkDh2vrGnxWagzKyWYp9+aR+Q5NaVsjtLvPHQVTs46Nmjw@mail.gmail.com>
References: <CANeCd7qgLdOv8AOnZUGFfpn8DmXVEUDNUYRarfMmrfmwZurHbA@mail.gmail.com>
 <CAMtXxQm8oj9TfJVLfmukL7BrEEkQAA0GB3xrmxGOnn5fbLGE=Q@mail.gmail.com>
 <CAFGRPVp2_=WpG9SWctEKVdYEn5_dwi1pE7gys0KXrHUg3OXSEQ@mail.gmail.com>
 <CAEJMVkDh2vrGnxWagzKyWYp9+aR+Q5NaVsjtLvPHQVTs46Nmjw@mail.gmail.com>
Message-ID: <CANeCd7oTVR1eTZRYiHRw0JaijGWjo-Qq=pd36iFh0Fed6kDjOw@mail.gmail.com>

Many thanks to all for the answers, I found the solution after writing the
problem but I did not have time to comment on it here.

The correct configuration is the one that Brian says although I had not set
"autonat:" as a public IP prefix. I've tried it now and it works too.

However, it does not work in all cases. The only case in which it works
correctly is the following:

- Video intercom device and FS server in the same network.
- Mobile client in other net.

But in many other cases it does NOT work properly:

- Video intercom device, FS server and mobile in the same network.
- Mobile to Mobile (mobiles in different networks and server in other)
- Mobile to Mobile (all in the same network)


My current configuration is the next:

- sip_profiles/external.xml

    <param name="local-network-acl" value="rfc1918.auto"/>
    <param name="ext-rtp-ip" value="autonat:88.2.243.70"/>
    <param name="ext-sip-ip" value="autonat:88.2.243.70"/>

- sip_profiles/internal.xml

    <param name="apply-nat-acl" value="rfc1918.auto"/>
    <param name="local-network-acl" value="rfc1918.auto"/>
    <param name="ext-rtp-ip" value="autonat:88.2.243.70"/>
    <param name="ext-sip-ip" value="autonat:88.2.243.70"/>


I test using "nat.auto" too. I can attach the entire file if needed.


The call is "abandoned" and show show "Rejected by acl "domains". Falling
back to Digest auth." among other things.

Source mobile as register as "1009 at 88.XX.YY.ZZ" and target mobile as
"1010 at 88.XX.YY.ZZ". The call is perform as "sip:1010 at 88.XX.YY.ZZ:5080"

The trace log of "Mobile to Mobile" call (I replace public IP to
88.XX.YY.ZZ):


recv 1401 bytes from udp/[213.143.51.43]:17334 at 09:13:06.058418:
   ------------------------------------------------------------------------
   INVITE sip:1010 at 88.XX.YY.ZZ:5080 SIP/2.0
   Via: SIP/2.0/UDP 10.78.108.57:51363;branch=z9hG4bK.BChE3dtFP;rport
   From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
   To: sip:1010 at 192.168.230.143:5060
   CSeq: 20 INVITE
   Call-ID: 5gQ3JhmBhp
   Max-Forwards: 70
   Supported: replaces, outbound
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO, UPDATE
   Content-Type: application/sdp
   Content-Length: 799
   Contact: <sip:1009 at 213.143.51.43:17334
;transport=udp>;+sip.instance="<urn:uuid:a0671e9c-6a12-4d32-88d9-43f873b85ef6>"
   User-Agent: LinphoneAndroid/3.2.7 (belle-sip/1.6.1)

   v=0
   o=1009 1906 519 IN IP4 10.78.108.57
   s=Talk
   c=IN IP4 10.78.108.57
   b=AS:512
   t=0 0
   a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
   m=audio 7076 RTP/AVP 96 97 98 99 0 8 18 101 100 102
   a=rtpmap:96 opus/48000/2
   a=fmtp:96 useinbandfec=1
   a=rtpmap:97 SILK/16000
   a=rtpmap:98 speex/16000
   a=fmtp:98 vbr=on
   a=rtpmap:99 speex/8000
   a=fmtp:99 vbr=on
   a=fmtp:18 annexb=yes
   a=rtpmap:101 telephone-event/48000
   a=rtpmap:100 telephone-event/16000
   a=rtpmap:102 telephone-event/8000
   a=rtcp-fb:* ccm tmmbr
   m=video 9078 RTP/AVP 96 97
   a=rtpmap:96 VP8/90000
   a=rtpmap:97 H264/90000
   a=fmtp:97 profile-level-id=42801F
   a=rtcp-fb:* ccm tmmbr
   a=rtcp-fb:96 nack pli
   a=rtcp-fb:96 nack sli
   a=rtcp-fb:96 ack rpsi
   a=rtcp-fb:96 ccm fir
   a=rtcp-fb:97 nack pli
   a=rtcp-fb:97 ccm fir
   ------------------------------------------------------------------------
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [NOTICE]
switch_channel.c:1104 New Channel sofia/internal/1009 at 192.168.230.143:5060
[367536c0-9e81-463d-9c5c-093cd85b292b]
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060)
Running State Change CS_NEW (Cur 1 Tot 1)
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite
from 213.143.51.43:17334 version: 1.9.0 -501-7c5d442 64bit
2017-07-20 09:13:06.047656 [DEBUG] sofia.c:10238 IP 213.143.51.43 Rejected
by acl "domains". Falling back to Digest auth.
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.230.143:5060)
State NEW
send 830 bytes to udp/[213.143.51.43]:17334 at 09:13:06.061889:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 10.78.108.57:51363
;branch=z9hG4bK.BChE3dtFP;rport=17334;received=213.143.51.43
   From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
   To: <sip:1010 at 192.168.230.143:5060>;tag=8Uc3QUtpgUjSm
   Call-ID: 5gQ3JhmBhp
   CSeq: 20 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.9.0-501-7c5d442~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event,
dialog, line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer
   Proxy-Authenticate: Digest realm="192.168.230.143",
nonce="1a8a328f-a982-4886-adae-1c5554c8d017", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
2017-07-20 09:13:06.047656 [DEBUG] sofia.c:2405 detaching session
367536c0-9e81-463d-9c5c-093cd85b292b
recv 389 bytes from udp/[213.143.51.43]:17334 at 09:13:06.184956:
   ------------------------------------------------------------------------
   ACK sip:1010 at 88.XX.YY.ZZ:5080 SIP/2.0
   Via: SIP/2.0/UDP 10.78.108.57:51363;branch=z9hG4bK.BChE3dtFP;rport
   Call-ID: 5gQ3JhmBhp
   From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
   To: <sip:1010 at 192.168.230.143:5060>;tag=8Uc3QUtpgUjSm
   Contact: <sip:1009 at 213.143.51.43:17334
;transport=udp>;+sip.instance="<urn:uuid:a0671e9c-6a12-4d32-88d9-43f873b85ef6>"
   Max-Forwards: 70
   CSeq: 20 ACK

   ------------------------------------------------------------------------
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [WARNING]
switch_core_state_machine.c:687 367536c0-9e81-463d-9c5c-093cd85b292b
sofia/internal/1009 at 192.168.230.143:5060 Abandoned
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [NOTICE]
switch_core_state_machine.c:690 Hangup sofia/internal/
1009 at 192.168.230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060)
Running State Change CS_HANGUP (Cur 1 Tot 1)
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.230.143:5060)
Callstate Change DOWN -> HANGUP
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060)
State HANGUP
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060 hanging
up, cause: WRONG_CALL_STATE
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060
Standard HANGUP, cause: WRONG_CALL_STATE
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060)
State HANGUP going to sleep
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.230.143:5060)
State Change CS_HANGUP -> CS_REPORTING
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060)
Running State Change CS_REPORTING (Cur 1 Tot 1)
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060)
State REPORTING
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060
Standard REPORTING, cause: WRONG_CALL_STATE
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060)
State REPORTING going to sleep
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.230.143:5060)
State Change CS_REPORTING -> CS_DESTROY
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_session.c:1713 Session 1 (sofia/internal/
1009 at 192.168.230.143:5060) Locked, Waiting on external entities
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [NOTICE]
switch_core_session.c:1731 Session 1 (sofia/internal/
1009 at 192.168.230.143:5060) Ended
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [NOTICE]
switch_core_session.c:1735 Close Channel sofia/internal/
1009 at 192.168.230.143:5060 [CS_DESTROY]
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.230.143:5060)
Running State Change CS_DESTROY (Cur 0 Tot 1)
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060)
State DESTROY
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060
Standard DESTROY
367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060)
State DESTROY going to sleep










José David Jurado Alonso <josedavid at zennio.com>

*Área de Desarrollo de Software de Alto Nivel*

*Dpto. Ingeniería*

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2017-07-19 21:00 GMT+02:00 Brian West <brian at freeswitch.org>:

> 1. set up your local-network-acl . (rfc1918.auto is good start)
> 2. setup your ext-rtp-ip and ext-sip-ip and MAKE SURE you prefix it with
> autonat: so it activates the proper behavior for you.
>
> This should be all you have to do so you can talk to devices inside and
> outside the NAT at the same time.
>
> /b
>
>
> On Wed, Jul 19, 2017 at 9:30 AM, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> +1 about forwarding upds...
>> But looking at the logs, we can't really see whether the call was
>> processed or not, just that it was "abandoned" and that usually happens
>> when freeswitch responds with a 401/407 and the client never
>> authenticates...
>>
>> enter:
>> sofia profile internal siptrace on
>>
>> to see all messages and make sure the client responds properly and paste
>> the signaling here.
>> ᐧ
>>
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>
>> On Wed, Jul 19, 2017 at 4:18 PM, Joel Serrano <joel at gogii.net> wrote:
>>
>>> Double check your ACLs, yes. Also, I'm pretty sure you will also have to
>>> forward the RTP port range so you can have audio... Otherwise signaling
>>> will work, but your next problem will be one-way-audio.
>>>
>>>
>>> On Tue, Jul 18, 2017 at 11:43 PM, Jose David Jurado Alonso <
>>> josedavid at zennio.com> wrote:
>>>
>>>>
>>>> I've installed *freeswitch into a local environment* and I perform SIP
>>>> calls well using the server local IP.
>>>>
>>>> This server has connected to ADSL network with a *static public IP*
>>>> but when I try to call using the public IP instead of the local IP the
>>>> server receive the call not work...
>>>>
>>>> The clients can register well (or I think so). Others process like
>>>> apache work well.
>>>>
>>>> I NAT forwarding in the firewall the next ports: 5060, 5080, 7443.
>>>>
>>>> *$ netstat -putan*
>>>>
>>>> Active Internet connections (servers and established)
>>>> Proto Recv-Q Send-Q Local Address           Foreign Address         State       PID/Program name
>>>> tcp        0      0 192.168.230.143:2855    0.0.0.0:*               LISTEN      15894/freeswitch
>>>> tcp        0      0 192.168.230.143:2856    0.0.0.0:*               LISTEN      15894/freeswitch
>>>> tcp        0      0 192.168.230.143:5066    0.0.0.0:*               LISTEN      15894/freeswitch
>>>> tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN      1166/nginx -g daemo
>>>> tcp        0      0 192.168.230.143:7443    0.0.0.0:*               LISTEN      15894/freeswitch
>>>> tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN      1079/sshd
>>>> tcp        0      0 192.168.230.143:5080    0.0.0.0:*               LISTEN      15894/freeswitch
>>>> tcp        0      0 0.0.0.0:443             0.0.0.0:*               LISTEN      1166/nginx -g daemo
>>>> tcp        0      0 192.168.230.143:5060    0.0.0.0:*               LISTEN      15894/freeswitch
>>>> tcp        0      0 192.168.230.143:22      192.168.230.167:60298   ESTABLISHED 14816/sshd: ubuntu
>>>> tcp        0      0 192.168.230.143:22      192.168.230.167:60296   ESTABLISHED 14744/sshd: ubuntu
>>>> tcp6       0      0 :::80                   :::*                    LISTEN      1166/nginx -g daemo
>>>> tcp6       0      0 :::8021                 :::*                    LISTEN      15894/freeswitch
>>>> tcp6       0      0 :::22                   :::*                    LISTEN      1079/sshd
>>>> tcp6       0      0 ::1:5080                :::*                    LISTEN      15894/freeswitch
>>>> tcp6       0      0 :::443                  :::*                    LISTEN      1166/nginx -g daemo
>>>> tcp6       0      0 ::1:5060                :::*                    LISTEN      15894/freeswitch
>>>> udp        0      0 192.168.230.143:5060    0.0.0.0:*                           15894/freeswitch
>>>> udp        0      0 192.168.230.143:5080    0.0.0.0:*                           15894/freeswitch
>>>> udp        0      0 0.0.0.0:68              0.0.0.0:*                           1019/dhclient
>>>> udp6       0      0 ::1:5060                :::*                                15894/freeswitch
>>>> udp6       0      0 ::1:5080                :::*                                15894/freeswitch
>>>>
>>>> *$ fs_cli -x "sofia status"*
>>>>
>>>>                      Name      Type                                       Data  State
>>>> =================================================================================================
>>>>             external-ipv6   profile                   sip:mod_sofia@[::1]:5080  RUNNING (0)
>>>>           192.168.230.143     alias                                   internal  ALIASED
>>>>                  external   profile         sip:mod_sofia at 192.168.230.143:5080  RUNNING (0)
>>>>     external::example.com   gateway                    sip:joeuser at example.com  NOREG
>>>>             internal-ipv6   profile                   sip:mod_sofia@[::1]:5060  RUNNING (0)
>>>>                  internal   profile         sip:mod_sofia at 192.168.230.143:5060  RUNNING (0)
>>>> =================================================================================================
>>>>
>>>> *$ fs_cli -x "sofia status profile external"*
>>>>
>>>> =================================================================================================
>>>> Name                external
>>>> Domain Name         N/A
>>>> Auto-NAT            false
>>>> DBName              sofia_reg_external
>>>> Pres Hosts
>>>> Dialplan            XML
>>>> Context             public
>>>> Challenge Realm     auto_to
>>>> RTP-IP              192.168.230.143
>>>> SIP-IP              192.168.230.143
>>>> URL                 sip:mod_sofia at 192.168.230.143:5080
>>>> BIND-URL            sip:mod_sofia at 192.168.230.143:5080;transport=udp,tcp
>>>> HOLD-MUSIC          local_stream://moh
>>>> OUTBOUND-PROXY      N/A
>>>> CODECS IN           OPUS,G722,PCMU,PCMA,VP8,H264
>>>> CODECS OUT          OPUS,G722,PCMU,PCMA,VP8,H264
>>>> TEL-EVENT           101
>>>> DTMF-MODE           rfc2833
>>>> CNG                 13
>>>> SESSION-TO          0
>>>> MAX-DIALOG          0
>>>> NOMEDIA             false
>>>> LATE-NEG            true
>>>> PROXY-MEDIA         false
>>>> ZRTP-PASSTHRU       true
>>>> AGGRESSIVENAT       false
>>>> CALLS-IN            0
>>>> FAILED-CALLS-IN     0
>>>> CALLS-OUT           0
>>>> FAILED-CALLS-OUT    0
>>>> REGISTRATIONS       0
>>>>
>>>> *$ grep -r '"ext-' **
>>>>
>>>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>> sip_profiles/internal.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>> sip_profiles/internal.xml:    <param name="ext-sip-ip" value="auto-nat"/>
>>>>
>>>> When I perform a call the freeswitch.log file show:
>>>>
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1009 at 192.168.230.143:5060 [df895d9a-30f2-40b5-97bb-56e222c1c45f]
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_NEW (Cur 1 Tot 7)
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite from 88.33.111.22:57076 version: 1.9.0 -493-13f2f2a 64bit
>>>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10238 IP 88.33.111.22 Rejected by acl "domains". Falling back to Digest auth.
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.230.143:5060) State NEW
>>>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:2405 detaching session df895d9a-30f2-40b5-97bb-56e222c1c45f
>>>>
>>>> and passed some seconds:
>>>>
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [WARNING] switch_core_state_machine.c:687 df895d9a-30f2-40b5-97bb-56e222c1c45f sofia/internal/1009 at 192.168.230.143:5060 Abandoned
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/1009 at 192.168.230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_HANGUP (Cur 1 Tot 7)
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.230.143:5060) Callstate Change DOWN -> HANGUP
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060 hanging up, cause: WRONG_CALL_STATE
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060 Standard HANGUP, cause: WRONG_CALL_STATE
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP going to sleep
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_HANGUP -> CS_REPORTING
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_REPORTING (Cur 1 Tot 7)
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060 Standard REPORTING, cause: WRONG_CALL_STATE
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING going to sleep
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_REPORTING -> CS_DESTROY
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_session.c:1713 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Locked, Waiting on external entities
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1731 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Ended
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1009 at 192.168.230.143:5060 [CS_DESTROY]
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_DESTROY (Cur 0 Tot 7)
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060 Standard DESTROY
>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY going to sleep
>>>>
>>>> Can it be an ACL or sip_profile bad configuration?? Router firewall
>>>> configuration maybe?
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From hunterj91 at hotmail.com  Thu Jul 20 16:51:19 2017
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Thu, 20 Jul 2017 16:51:19 +0000
Subject: [Freeswitch-users] group_call_function and record_session -
 Recording calls for a specific Extension within a call group
Message-ID: <DB4PR02MB03362FF459B1EFB5CC9C4BFFBDA70@DB4PR02MB0336.eurprd02.prod.outlook.com>

Hi All,


Hope everyone is well?


I have a quick question around recording calls for extensions that are contained within a huntgroup.


We use record_session to record calls when made outbound or inbound to specific extensions, however we do this using the bridge function.


Is it possible to record calls for a specific extension contained within a call group?


For example can dial-string be defined with record_session?


Has any body done this before?


Many thanks


Jon
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From brian at freeswitch.org  Thu Jul 20 20:16:51 2017
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Jul 2017 15:16:51 -0500
Subject: [Freeswitch-users] Freeswitch SIP call not work using a static
 public IP
In-Reply-To: <CANeCd7oTVR1eTZRYiHRw0JaijGWjo-Qq=pd36iFh0Fed6kDjOw@mail.gmail.com>
References: <CANeCd7qgLdOv8AOnZUGFfpn8DmXVEUDNUYRarfMmrfmwZurHbA@mail.gmail.com>
 <CAMtXxQm8oj9TfJVLfmukL7BrEEkQAA0GB3xrmxGOnn5fbLGE=Q@mail.gmail.com>
 <CAFGRPVp2_=WpG9SWctEKVdYEn5_dwi1pE7gys0KXrHUg3OXSEQ@mail.gmail.com>
 <CAEJMVkDh2vrGnxWagzKyWYp9+aR+Q5NaVsjtLvPHQVTs46Nmjw@mail.gmail.com>
 <CANeCd7oTVR1eTZRYiHRw0JaijGWjo-Qq=pd36iFh0Fed6kDjOw@mail.gmail.com>
Message-ID: <CAEJMVkDQZQe_aKJP89CBAT8G3FMxBKxs-LkSARephg+ipuwxzw@mail.gmail.com>

also make sure the clients aren't doing stun requests that are behind the
nat with FreeSWITCH....

/b


On Thu, Jul 20, 2017 at 2:28 AM, Jose David Jurado Alonso <
josedavid at zennio.com> wrote:

> Many thanks to all for the answers, I found the solution after writing the
> problem but I did not have time to comment on it here.
>
> The correct configuration is the one that Brian says although I had not
> set "autonat:" as a public IP prefix. I've tried it now and it works too.
>
> However, it does not work in all cases. The only case in which it works
> correctly is the following:
>
> - Video intercom device and FS server in the same network.
> - Mobile client in other net.
>
> But in many other cases it does NOT work properly:
>
> - Video intercom device, FS server and mobile in the same network.
> - Mobile to Mobile (mobiles in different networks and server in other)
> - Mobile to Mobile (all in the same network)
>
>
> My current configuration is the next:
>
> - sip_profiles/external.xml
>
>     <param name="local-network-acl" value="rfc1918.auto"/>
>     <param name="ext-rtp-ip" value="autonat:88.2.243.70"/>
>     <param name="ext-sip-ip" value="autonat:88.2.243.70"/>
>
> - sip_profiles/internal.xml
>
>     <param name="apply-nat-acl" value="rfc1918.auto"/>
>     <param name="local-network-acl" value="rfc1918.auto"/>
>     <param name="ext-rtp-ip" value="autonat:88.2.243.70"/>
>     <param name="ext-sip-ip" value="autonat:88.2.243.70"/>
>
>
> I test using "nat.auto" too. I can attach the entire file if needed.
>
>
> The call is "abandoned" and show show "Rejected by acl "domains". Falling
> back to Digest auth." among other things.
>
> Source mobile as register as "1009 at 88.XX.YY.ZZ" and target mobile as
> "1010 at 88.XX.YY.ZZ". The call is perform as "sip:1010 at 88.XX.YY.ZZ:5080"
>
> The trace log of "Mobile to Mobile" call (I replace public IP to
> 88.XX.YY.ZZ):
>
>
> recv 1401 bytes from udp/[213.143.51.43]:17334 at 09:13:06.058418:
>    -----------------------------------------------------------
> -------------
>    INVITE sip:1010 at 88.XX.YY.ZZ:5080 SIP/2.0
>    Via: SIP/2.0/UDP 10.78.108.57:51363;branch=z9hG4bK.BChE3dtFP;rport
>    From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
>    To: sip:1010 at 192.168.230.143:5060
>    CSeq: 20 INVITE
>    Call-ID: 5gQ3JhmBhp
>    Max-Forwards: 70
>    Supported: replaces, outbound
>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO, UPDATE
>    Content-Type: application/sdp
>    Content-Length: 799
>    Contact: <sip:1009 at 213.143.51.43:17334;transport=udp>;+sip.instance="
> <urn:uuid:a0671e9c-6a12-4d32-88d9-43f873b85ef6>"
>    User-Agent: LinphoneAndroid/3.2.7 (belle-sip/1.6.1)
>
>    v=0
>    o=1009 1906 519 IN IP4 10.78.108.57
>    s=Talk
>    c=IN IP4 10.78.108.57
>    b=AS:512
>    t=0 0
>    a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
>    m=audio 7076 RTP/AVP 96 97 98 99 0 8 18 101 100 102
>    a=rtpmap:96 opus/48000/2
>    a=fmtp:96 useinbandfec=1
>    a=rtpmap:97 SILK/16000
>    a=rtpmap:98 speex/16000
>    a=fmtp:98 vbr=on
>    a=rtpmap:99 speex/8000
>    a=fmtp:99 vbr=on
>    a=fmtp:18 annexb=yes
>    a=rtpmap:101 telephone-event/48000
>    a=rtpmap:100 telephone-event/16000
>    a=rtpmap:102 telephone-event/8000
>    a=rtcp-fb:* ccm tmmbr
>    m=video 9078 RTP/AVP 96 97
>    a=rtpmap:96 VP8/90000
>    a=rtpmap:97 H264/90000
>    a=fmtp:97 profile-level-id=42801F
>    a=rtcp-fb:* ccm tmmbr
>    a=rtcp-fb:96 nack pli
>    a=rtcp-fb:96 nack sli
>    a=rtcp-fb:96 ack rpsi
>    a=rtcp-fb:96 ccm fir
>    a=rtcp-fb:97 nack pli
>    a=rtcp-fb:97 ccm fir
>    -----------------------------------------------------------
> -------------
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [NOTICE]
> switch_channel.c:1104 New Channel sofia/internal/1009 at 192.168.230.143:5060
> [367536c0-9e81-463d-9c5c-093cd85b292b]
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
> switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060)
> Running State Change CS_NEW (Cur 1 Tot 1)
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
> sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite
> from 213.143.51.43:17334 version: 1.9.0 -501-7c5d442 64bit
> 2017-07-20 09:13:06.047656 [DEBUG] sofia.c:10238 IP 213.143.51.43 Rejected
> by acl "domains". Falling back to Digest auth.
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
> switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.230.143:5060)
> State NEW
> send 830 bytes to udp/[213.143.51.43]:17334 at 09:13:06.061889:
>    -----------------------------------------------------------
> -------------
>    SIP/2.0 407 Proxy Authentication Required
>    Via: SIP/2.0/UDP 10.78.108.57:51363;branch=
> z9hG4bK.BChE3dtFP;rport=17334;received=213.143.51.43
>    From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
>    To: <sip:1010 at 192.168.230.143:5060>;tag=8Uc3QUtpgUjSm
>    Call-ID: 5gQ3JhmBhp
>    CSeq: 20 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.9.0-501-7c5d442~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, presence, as-feature-event,
> dialog, line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer
>    Proxy-Authenticate: Digest realm="192.168.230.143",
> nonce="1a8a328f-a982-4886-adae-1c5554c8d017", algorithm=MD5, qop="auth"
>    Content-Length: 0
>
>    -----------------------------------------------------------
> -------------
> 2017-07-20 09:13:06.047656 [DEBUG] sofia.c:2405 detaching session
> 367536c0-9e81-463d-9c5c-093cd85b292b
> recv 389 bytes from udp/[213.143.51.43]:17334 at 09:13:06.184956:
>    -----------------------------------------------------------
> -------------
>    ACK sip:1010 at 88.XX.YY.ZZ:5080 SIP/2.0
>    Via: SIP/2.0/UDP 10.78.108.57:51363;branch=z9hG4bK.BChE3dtFP;rport
>    Call-ID: 5gQ3JhmBhp
>    From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
>    To: <sip:1010 at 192.168.230.143:5060>;tag=8Uc3QUtpgUjSm
>    Contact: <sip:1009 at 213.143.51.43:17334;transport=udp>;+sip.instance="
> <urn:uuid:a0671e9c-6a12-4d32-88d9-43f873b85ef6>"
>    Max-Forwards: 70
>    CSeq: 20 ACK
>
>    -----------------------------------------------------------
> -------------
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [WARNING]
> switch_core_state_machine.c:687 367536c0-9e81-463d-9c5c-093cd85b292b
> sofia/internal/1009 at 192.168.230.143:5060 Abandoned
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [NOTICE]
> switch_core_state_machine.c:690 Hangup sofia/internal/1009 at 192.168.
> 230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
> switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060)
> Running State Change CS_HANGUP (Cur 1 Tot 1)
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
> switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.230.143:5060)
> Callstate Change DOWN -> HANGUP
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
> switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060)
> State HANGUP
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
> mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060 hanging
> up, cause: WRONG_CALL_STATE
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060
> Standard HANGUP, cause: WRONG_CALL_STATE
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060)
> State HANGUP going to sleep
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.230.143:5060)
> State Change CS_HANGUP -> CS_REPORTING
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060)
> Running State Change CS_REPORTING (Cur 1 Tot 1)
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060)
> State REPORTING
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060
> Standard REPORTING, cause: WRONG_CALL_STATE
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060)
> State REPORTING going to sleep
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.230.143:5060)
> State Change CS_REPORTING -> CS_DESTROY
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_session.c:1713 Session 1 (sofia/internal/1009 at 192.168.
> 230.143:5060) Locked, Waiting on external entities
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [NOTICE]
> switch_core_session.c:1731 Session 1 (sofia/internal/1009 at 192.168.
> 230.143:5060) Ended
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [NOTICE]
> switch_core_session.c:1735 Close Channel sofia/internal/1009 at 192.168.
> 230.143:5060 [CS_DESTROY]
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.230.143:5060)
> Running State Change CS_DESTROY (Cur 0 Tot 1)
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060)
> State DESTROY
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060
> Standard DESTROY
> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
> switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060)
> State DESTROY going to sleep
>
>
>
>
>
>
>
>
>
>
> José David Jurado Alonso <josedavid at zennio.com>
>
> *Área de Desarrollo de Software de Alto Nivel*
>
> *Dpto. Ingeniería*
>
> [image: Descripción: cid:image010.jpg at 01D27631.DF7E30F0]
>
> Zennio Avance y Tecnología, S.L.
>
> C/ Rio Jarama,132. Nave P-8.11
>
> 45007 - Toledo (Spain)
>
> T: +34 925 232 002 <+34%20925%2023%2020%2002>
>
> www.zennio.com
>
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>
> 2017-07-19 21:00 GMT+02:00 Brian West <brian at freeswitch.org>:
>
>> 1. set up your local-network-acl . (rfc1918.auto is good start)
>> 2. setup your ext-rtp-ip and ext-sip-ip and MAKE SURE you prefix it with
>> autonat: so it activates the proper behavior for you.
>>
>> This should be all you have to do so you can talk to devices inside and
>> outside the NAT at the same time.
>>
>> /b
>>
>>
>> On Wed, Jul 19, 2017 at 9:30 AM, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> +1 about forwarding upds...
>>> But looking at the logs, we can't really see whether the call was
>>> processed or not, just that it was "abandoned" and that usually happens
>>> when freeswitch responds with a 401/407 and the client never
>>> authenticates...
>>>
>>> enter:
>>> sofia profile internal siptrace on
>>>
>>> to see all messages and make sure the client responds properly and paste
>>> the signaling here.
>>> ᐧ
>>>
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>>
>>> On Wed, Jul 19, 2017 at 4:18 PM, Joel Serrano <joel at gogii.net> wrote:
>>>
>>>> Double check your ACLs, yes. Also, I'm pretty sure you will also have
>>>> to forward the RTP port range so you can have audio... Otherwise signaling
>>>> will work, but your next problem will be one-way-audio.
>>>>
>>>>
>>>> On Tue, Jul 18, 2017 at 11:43 PM, Jose David Jurado Alonso <
>>>> josedavid at zennio.com> wrote:
>>>>
>>>>>
>>>>> I've installed *freeswitch into a local environment* and I perform
>>>>> SIP calls well using the server local IP.
>>>>>
>>>>> This server has connected to ADSL network with a *static public IP*
>>>>> but when I try to call using the public IP instead of the local IP the
>>>>> server receive the call not work...
>>>>>
>>>>> The clients can register well (or I think so). Others process like
>>>>> apache work well.
>>>>>
>>>>> I NAT forwarding in the firewall the next ports: 5060, 5080, 7443.
>>>>>
>>>>> *$ netstat -putan*
>>>>>
>>>>> Active Internet connections (servers and established)
>>>>> Proto Recv-Q Send-Q Local Address           Foreign Address         State       PID/Program name
>>>>> tcp        0      0 192.168.230.143:2855    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>> tcp        0      0 192.168.230.143:2856    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>> tcp        0      0 192.168.230.143:5066    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>> tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN      1166/nginx -g daemo
>>>>> tcp        0      0 192.168.230.143:7443    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>> tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN      1079/sshd
>>>>> tcp        0      0 192.168.230.143:5080    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>> tcp        0      0 0.0.0.0:443             0.0.0.0:*               LISTEN      1166/nginx -g daemo
>>>>> tcp        0      0 192.168.230.143:5060    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>> tcp        0      0 192.168.230.143:22      192.168.230.167:60298   ESTABLISHED 14816/sshd: ubuntu
>>>>> tcp        0      0 192.168.230.143:22      192.168.230.167:60296   ESTABLISHED 14744/sshd: ubuntu
>>>>> tcp6       0      0 :::80                   :::*                    LISTEN      1166/nginx -g daemo
>>>>> tcp6       0      0 :::8021                 :::*                    LISTEN      15894/freeswitch
>>>>> tcp6       0      0 :::22                   :::*                    LISTEN      1079/sshd
>>>>> tcp6       0      0 ::1:5080                :::*                    LISTEN      15894/freeswitch
>>>>> tcp6       0      0 :::443                  :::*                    LISTEN      1166/nginx -g daemo
>>>>> tcp6       0      0 ::1:5060                :::*                    LISTEN      15894/freeswitch
>>>>> udp        0      0 192.168.230.143:5060    0.0.0.0:*                           15894/freeswitch
>>>>> udp        0      0 192.168.230.143:5080    0.0.0.0:*                           15894/freeswitch
>>>>> udp        0      0 0.0.0.0:68              0.0.0.0:*                           1019/dhclient
>>>>> udp6       0      0 ::1:5060                :::*                                15894/freeswitch
>>>>> udp6       0      0 ::1:5080                :::*                                15894/freeswitch
>>>>>
>>>>> *$ fs_cli -x "sofia status"*
>>>>>
>>>>>                      Name      Type                                       Data  State
>>>>> =================================================================================================
>>>>>             external-ipv6   profile                   sip:mod_sofia@[::1]:5080  RUNNING (0)
>>>>>           192.168.230.143     alias                                   internal  ALIASED
>>>>>                  external   profile         sip:mod_sofia at 192.168.230.143:5080  RUNNING (0)
>>>>>     external::example.com   gateway                    sip:joeuser at example.com  NOREG
>>>>>             internal-ipv6   profile                   sip:mod_sofia@[::1]:5060  RUNNING (0)
>>>>>                  internal   profile         sip:mod_sofia at 192.168.230.143:5060  RUNNING (0)
>>>>> =================================================================================================
>>>>>
>>>>> *$ fs_cli -x "sofia status profile external"*
>>>>>
>>>>> =================================================================================================
>>>>> Name                external
>>>>> Domain Name         N/A
>>>>> Auto-NAT            false
>>>>> DBName              sofia_reg_external
>>>>> Pres Hosts
>>>>> Dialplan            XML
>>>>> Context             public
>>>>> Challenge Realm     auto_to
>>>>> RTP-IP              192.168.230.143
>>>>> SIP-IP              192.168.230.143
>>>>> URL                 sip:mod_sofia at 192.168.230.143:5080
>>>>> BIND-URL            sip:mod_sofia at 192.168.230.143:5080;transport=udp,tcp
>>>>> HOLD-MUSIC          local_stream://moh
>>>>> OUTBOUND-PROXY      N/A
>>>>> CODECS IN           OPUS,G722,PCMU,PCMA,VP8,H264
>>>>> CODECS OUT          OPUS,G722,PCMU,PCMA,VP8,H264
>>>>> TEL-EVENT           101
>>>>> DTMF-MODE           rfc2833
>>>>> CNG                 13
>>>>> SESSION-TO          0
>>>>> MAX-DIALOG          0
>>>>> NOMEDIA             false
>>>>> LATE-NEG            true
>>>>> PROXY-MEDIA         false
>>>>> ZRTP-PASSTHRU       true
>>>>> AGGRESSIVENAT       false
>>>>> CALLS-IN            0
>>>>> FAILED-CALLS-IN     0
>>>>> CALLS-OUT           0
>>>>> FAILED-CALLS-OUT    0
>>>>> REGISTRATIONS       0
>>>>>
>>>>> *$ grep -r '"ext-' **
>>>>>
>>>>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>>> sip_profiles/internal.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>>> sip_profiles/internal.xml:    <param name="ext-sip-ip" value="auto-nat"/>
>>>>>
>>>>> When I perform a call the freeswitch.log file show:
>>>>>
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1009 at 192.168.230.143:5060 [df895d9a-30f2-40b5-97bb-56e222c1c45f]
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_NEW (Cur 1 Tot 7)
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite from 88.33.111.22:57076 version: 1.9.0 -493-13f2f2a 64bit
>>>>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10238 IP 88.33.111.22 Rejected by acl "domains". Falling back to Digest auth.
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.230.143:5060) State NEW
>>>>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:2405 detaching session df895d9a-30f2-40b5-97bb-56e222c1c45f
>>>>>
>>>>> and passed some seconds:
>>>>>
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [WARNING] switch_core_state_machine.c:687 df895d9a-30f2-40b5-97bb-56e222c1c45f sofia/internal/1009 at 192.168.230.143:5060 Abandoned
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/1009 at 192.168.230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_HANGUP (Cur 1 Tot 7)
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.230.143:5060) Callstate Change DOWN -> HANGUP
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060 hanging up, cause: WRONG_CALL_STATE
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060 Standard HANGUP, cause: WRONG_CALL_STATE
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP going to sleep
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_HANGUP -> CS_REPORTING
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_REPORTING (Cur 1 Tot 7)
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060 Standard REPORTING, cause: WRONG_CALL_STATE
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING going to sleep
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_REPORTING -> CS_DESTROY
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_session.c:1713 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Locked, Waiting on external entities
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1731 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Ended
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1009 at 192.168.230.143:5060 [CS_DESTROY]
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_DESTROY (Cur 0 Tot 7)
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060 Standard DESTROY
>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY going to sleep
>>>>>
>>>>> Can it be an ACL or sip_profile bad configuration?? Router firewall
>>>>> configuration maybe?
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>>
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>
>> Allison prompts for FreeSWITCH:
>>
>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
>> | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From krc at retrospekt.dk  Thu Jul 20 20:36:47 2017
From: krc at retrospekt.dk (Kim Rostgaard Schomacker)
Date: Thu, 20 Jul 2017 22:36:47 +0200
Subject: [Freeswitch-users] CDR - best practices
In-Reply-To: <224992776.2280023.1500304763949@mail.yahoo.com>
References: <224992776.2280023.1500304763949.ref@mail.yahoo.com>
 <224992776.2280023.1500304763949@mail.yahoo.com>
Message-ID: <6e0d11bd-a6b2-ea24-d80f-a19f3ce9b6c7@retrospekt.dk>

Hi Robert


A while back we had similar considerations regarding CDR. What we ended 
up with, was to use the JSON CDR module for writing the records to disk. 
Then we used an custom tool[1] to upload the files to Google Cloud, 
where they may be indexed further.
The main idea was to store as much data as possible, as it may come in 
handy later. Also, the database module seemed to lack some of the 
information that we needed (eg. custom variables).
This worked very well for us, as we did not know from day 1 what 
information could be useful, and storing everything solved exactly that.

Have fun!
/Kim

[1] https://github.com/Bitstackers/openreception/tree/master/tools/cdrctl

On 2017-07-17 17:19, robert mundkowsky wrote:
>
> What are the best practice for writing Call Detail Records (CDRs) to a 
> database?
>
> We are using custom code to write CDR-like data to log files and then 
> later loading logs into a database.We have very low usage right now, 
> so I suspect that we could just write directly to the database.But as 
> we scale, is it a better idea to not write directly to the database?
>
> Note I am not sure if we need custom CDRs or if FS' CDR module might 
> be leveraged.I have to look into that.
>
> Robert
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kk-mailinglist at ednt.de  Thu Jul 20 20:38:33 2017
From: kk-mailinglist at ednt.de (Karlheinz Knapp)
Date: Thu, 20 Jul 2017 22:38:33 +0200
Subject: [Freeswitch-users] Call forward
Message-ID: <010A5B50-922C-4A56-AD1F-51B858EE9850@ednt.de>

i have extension which is forwarded to a mobile number


If I call this extension from a neighbor extension it works.

But if I call from a outside number to the extension

I don't get the voice on both sides.


I believe it is a issue with reinvite.


How I can fix this issue ?


-- 
Diese Nachricht wurde von meinem Android-Gerät mit K-9 Mail gesendet.
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From rfmundkowsky at yahoo.com  Thu Jul 20 21:52:01 2017
From: rfmundkowsky at yahoo.com (robert mundkowsky)
Date: Thu, 20 Jul 2017 21:52:01 +0000 (UTC)
Subject: [Freeswitch-users] traffic and system monitoring
In-Reply-To: <CAN8B5B=qb+Hr89AqaQ6cKkJobEH_MT1vHSAjSOK8YV6zShbJsw@mail.gmail.com>
References: <1012898889.2268619.1500304838441.ref@mail.yahoo.com>
 <1012898889.2268619.1500304838441@mail.yahoo.com>
 <dd31c0e0-74b5-3a5a-9236-62bec7948049@paulzillmann.de>
 <CAFGRPVqTMX57bsvWs9xM8ofLFdbs6Tfim3pbVQjC0MSB9CcbkQ@mail.gmail.com>
 <CAN8B5B=qb+Hr89AqaQ6cKkJobEH_MT1vHSAjSOK8YV6zShbJsw@mail.gmail.com>
Message-ID: <1019979566.1604480.1500587521638@mail.yahoo.com>

thanks, slides were very helpful.  I likely saw your presentation last year, but was on information overload from all the presentations from ClueCON.




On Wednesday, July 19, 2017, 2:22:45 PM EDT, GM phy <servtelar at gmail.com> wrote:

Take a look on this:https://www.slideshare.net/MoisesSilva6/freeswitch-monitoring

There are good ideas there.
Regards
Gustavo

On Wed, Jul 19, 2017 at 11:46 AM, David Villasmil <david.villasmil.work at gmail.com> wrote:

Although HOMER is not bad for monitoring things like ACD/ASR GLOBALLY, it is not designed to be used as a monitoring tool. It is a tool to have the signaling handy for troubleshooting (it can also keep logs, RTCP, and other things), and for this, I don't know anything better that HOMER. HOMER is specifically designed to troubleshoot VoIP.
I've never used CACTI, however, it doesn't look to be a SIP monitoring tool, but a network monitoring tool, able to monitor CPU, MEMORY, ETC... It won't monitor ACD, ASR, NER (Although i suppose you can write some plugins)... which HOMER does GLOBALLY...
You could extend HOMER to monitor on a per-host level (I did a P/R just for this, but it hasn't been merged yet, as there's no GUI to show that data and I hate AngularJS)
In conclusion, Maybe you should think about using both, as they seem to complement each other.
Hope this helps
Regards,
Davidᐧ
Regards,
David Villasmilemail: david.villasmil.work at gmail.comphone: +34669448337
On Wed, Jul 19, 2017 at 3:24 PM, Paul Zillmann <mail at paulzillmann.de> wrote:

  Hey Robert,
 
 I'm using check_mk / nagios for external monitoring and iftop and top / htop for on site analysis.
 
 
 Am 17.07.2017 um 17:20 schrieb robert mundkowsky:
  
   
 
 
Is Homer the best free option for traffic and system monitoring GUI?
 
               Or is CACTI better option?
 
 
 
Robert 
   
  
 ______________________________ ______________________________ _____________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions .com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswi tch.org
http://lists.freeswitch.org/ma ilman/listinfo/freeswitch-user s
UNSUBSCRIBE:http://lists.frees witch.org/mailman/options/ freeswitch-users
http://www.freeswitch.org 
 
 
______________________________ ______________________________ _____________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions .com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswi tch.org
http://lists.freeswitch.org/ma ilman/listinfo/freeswitch-user s
UNSUBSCRIBE:http://lists.frees witch.org/mailman/options/ freeswitch-users
http://www.freeswitch.org



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

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

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


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From udy786 at gmail.com  Fri Jul 21 07:06:48 2017
From: udy786 at gmail.com (Uday kumar)
Date: Fri, 21 Jul 2017 12:36:48 +0530
Subject: [Freeswitch-users] Integration of Google Speech API V2
In-Reply-To: <CAM8wsqJ+Voj65OBhWfDjjzn3c9vhcU0MaN32eDE_dW5-vh50Dg@mail.gmail.com>
References: <CAM8wsqJ+Voj65OBhWfDjjzn3c9vhcU0MaN32eDE_dW5-vh50Dg@mail.gmail.com>
Message-ID: <CAOfqg5b3q4TkKUHU7o+90nfXXvhhaWJX_yX1+qZLz1H13Znijw@mail.gmail.com>

I have implemented Google Speech on Freeswitch 1.6.16 Transcribing
Voicemail and its working really nice without any issue. When user get
voicemail then server send email to user with recording and text of
recording.

I am using SMTP to send mail. In switch.con.xml

<param name="mailer-app" value="/usr/bin/php
/var/www/html/PHPMailer/mailer_app.php"/>

Download mailer_app.php from freeswitch wiki and you can implement google
speech API and code in that file.

Thanks
Uday.

On Wed, Jul 19, 2017 at 2:07 PM, Rahul MathuR <rahul.ultimate at gmail.com>
wrote:

> Hi,
>
> I'm trying to integrate Google cloud speech recognition v2 in it. I can
> get the audio recorded, have created Service key and API key but whenever I
> try to access it, I just get 403 access denied. I am at my wits end here.
>
> Has anybody tried it ? were you successful ? Could you please guide me how
> to do it ?
> I'll be grateful to you if this works !
>
>
>
> --
> Warm Regds.
> MathuRahul
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regard
Uday
Site:- www.shareyourknowledge.in
Mobile:- +91-9377579349
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From josedavid at zennio.com  Fri Jul 21 05:40:04 2017
From: josedavid at zennio.com (Jose David Jurado Alonso)
Date: Fri, 21 Jul 2017 07:40:04 +0200
Subject: [Freeswitch-users] Freeswitch SIP call not work using a static
 public IP
In-Reply-To: <CAEJMVkDQZQe_aKJP89CBAT8G3FMxBKxs-LkSARephg+ipuwxzw@mail.gmail.com>
References: <CANeCd7qgLdOv8AOnZUGFfpn8DmXVEUDNUYRarfMmrfmwZurHbA@mail.gmail.com>
 <CAMtXxQm8oj9TfJVLfmukL7BrEEkQAA0GB3xrmxGOnn5fbLGE=Q@mail.gmail.com>
 <CAFGRPVp2_=WpG9SWctEKVdYEn5_dwi1pE7gys0KXrHUg3OXSEQ@mail.gmail.com>
 <CAEJMVkDh2vrGnxWagzKyWYp9+aR+Q5NaVsjtLvPHQVTs46Nmjw@mail.gmail.com>
 <CANeCd7oTVR1eTZRYiHRw0JaijGWjo-Qq=pd36iFh0Fed6kDjOw@mail.gmail.com>
 <CAEJMVkDQZQe_aKJP89CBAT8G3FMxBKxs-LkSARephg+ipuwxzw@mail.gmail.com>
Message-ID: <CANeCd7pYKp-kJOrtrb8y56TaEV3yCcYfrD2K1Y-RNn=YLuuYSA@mail.gmail.com>

Yes, both mobile clients use their own data network (4G) and access always
by the public IP..

José David Jurado Alonso <josedavid at zennio.com>

*Área de Desarrollo de Software de Alto Nivel*

*Dpto. Ingeniería*

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2017-07-20 22:16 GMT+02:00 Brian West <brian at freeswitch.org>:

> also make sure the clients aren't doing stun requests that are behind the
> nat with FreeSWITCH....
>
> /b
>
>
> On Thu, Jul 20, 2017 at 2:28 AM, Jose David Jurado Alonso <
> josedavid at zennio.com> wrote:
>
>> Many thanks to all for the answers, I found the solution after writing
>> the problem but I did not have time to comment on it here.
>>
>> The correct configuration is the one that Brian says although I had not
>> set "autonat:" as a public IP prefix. I've tried it now and it works too.
>>
>> However, it does not work in all cases. The only case in which it works
>> correctly is the following:
>>
>> - Video intercom device and FS server in the same network.
>> - Mobile client in other net.
>>
>> But in many other cases it does NOT work properly:
>>
>> - Video intercom device, FS server and mobile in the same network.
>> - Mobile to Mobile (mobiles in different networks and server in other)
>> - Mobile to Mobile (all in the same network)
>>
>>
>> My current configuration is the next:
>>
>> - sip_profiles/external.xml
>>
>>     <param name="local-network-acl" value="rfc1918.auto"/>
>>     <param name="ext-rtp-ip" value="autonat:88.2.243.70"/>
>>     <param name="ext-sip-ip" value="autonat:88.2.243.70"/>
>>
>> - sip_profiles/internal.xml
>>
>>     <param name="apply-nat-acl" value="rfc1918.auto"/>
>>     <param name="local-network-acl" value="rfc1918.auto"/>
>>     <param name="ext-rtp-ip" value="autonat:88.2.243.70"/>
>>     <param name="ext-sip-ip" value="autonat:88.2.243.70"/>
>>
>>
>> I test using "nat.auto" too. I can attach the entire file if needed.
>>
>>
>> The call is "abandoned" and show show "Rejected by acl "domains". Falling
>> back to Digest auth." among other things.
>>
>> Source mobile as register as "1009 at 88.XX.YY.ZZ" and target mobile as
>> "1010 at 88.XX.YY.ZZ". The call is perform as "sip:1010 at 88.XX.YY.ZZ:5080"
>>
>> The trace log of "Mobile to Mobile" call (I replace public IP to
>> 88.XX.YY.ZZ):
>>
>>
>> recv 1401 bytes from udp/[213.143.51.43]:17334 at 09:13:06.058418:
>>    -----------------------------------------------------------
>> -------------
>>    INVITE sip:1010 at 88.XX.YY.ZZ:5080 SIP/2.0
>>    Via: SIP/2.0/UDP 10.78.108.57:51363;branch=z9hG4bK.BChE3dtFP;rport
>>    From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
>>    To: sip:1010 at 192.168.230.143:5060
>>    CSeq: 20 INVITE
>>    Call-ID: 5gQ3JhmBhp
>>    Max-Forwards: 70
>>    Supported: replaces, outbound
>>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>> SUBSCRIBE, INFO, UPDATE
>>    Content-Type: application/sdp
>>    Content-Length: 799
>>    Contact: <sip:1009 at 213.143.51.43:17334;transport=udp>;+sip.instance="
>> <urn:uuid:a0671e9c-6a12-4d32-88d9-43f873b85ef6>"
>>    User-Agent: LinphoneAndroid/3.2.7 (belle-sip/1.6.1)
>>
>>    v=0
>>    o=1009 1906 519 IN IP4 10.78.108.57
>>    s=Talk
>>    c=IN IP4 10.78.108.57
>>    b=AS:512
>>    t=0 0
>>    a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL
>> voip-metrics
>>    m=audio 7076 RTP/AVP 96 97 98 99 0 8 18 101 100 102
>>    a=rtpmap:96 opus/48000/2
>>    a=fmtp:96 useinbandfec=1
>>    a=rtpmap:97 SILK/16000
>>    a=rtpmap:98 speex/16000
>>    a=fmtp:98 vbr=on
>>    a=rtpmap:99 speex/8000
>>    a=fmtp:99 vbr=on
>>    a=fmtp:18 annexb=yes
>>    a=rtpmap:101 telephone-event/48000
>>    a=rtpmap:100 telephone-event/16000
>>    a=rtpmap:102 telephone-event/8000
>>    a=rtcp-fb:* ccm tmmbr
>>    m=video 9078 RTP/AVP 96 97
>>    a=rtpmap:96 VP8/90000
>>    a=rtpmap:97 H264/90000
>>    a=fmtp:97 profile-level-id=42801F
>>    a=rtcp-fb:* ccm tmmbr
>>    a=rtcp-fb:96 nack pli
>>    a=rtcp-fb:96 nack sli
>>    a=rtcp-fb:96 ack rpsi
>>    a=rtcp-fb:96 ccm fir
>>    a=rtcp-fb:97 nack pli
>>    a=rtcp-fb:97 ccm fir
>>    -----------------------------------------------------------
>> -------------
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [NOTICE]
>> switch_channel.c:1104 New Channel sofia/internal/1009 at 192.168.23
>> 0.143:5060 [367536c0-9e81-463d-9c5c-093cd85b292b]
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
>> switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060)
>> Running State Change CS_NEW (Cur 1 Tot 1)
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
>> sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite
>> from 213.143.51.43:17334 version: 1.9.0 -501-7c5d442 64bit
>> 2017-07-20 09:13:06.047656 [DEBUG] sofia.c:10238 IP 213.143.51.43
>> Rejected by acl "domains". Falling back to Digest auth.
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
>> switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.230.143:5060)
>> State NEW
>> send 830 bytes to udp/[213.143.51.43]:17334 at 09:13:06.061889:
>>    -----------------------------------------------------------
>> -------------
>>    SIP/2.0 407 Proxy Authentication Required
>>    Via: SIP/2.0/UDP 10.78.108.57:51363;branch=z9hG
>> 4bK.BChE3dtFP;rport=17334;received=213.143.51.43
>>    From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
>>    To: <sip:1010 at 192.168.230.143:5060>;tag=8Uc3QUtpgUjSm
>>    Call-ID: 5gQ3JhmBhp
>>    CSeq: 20 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.9.0-501-7c5d442~64bit
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, path, replaces
>>    Allow-Events: talk, hold, conference, presence, as-feature-event,
>> dialog, line-seize, call-info, sla, include-session-description,
>> presence.winfo, message-summary, refer
>>    Proxy-Authenticate: Digest realm="192.168.230.143",
>> nonce="1a8a328f-a982-4886-adae-1c5554c8d017", algorithm=MD5, qop="auth"
>>    Content-Length: 0
>>
>>    -----------------------------------------------------------
>> -------------
>> 2017-07-20 09:13:06.047656 [DEBUG] sofia.c:2405 detaching session
>> 367536c0-9e81-463d-9c5c-093cd85b292b
>> recv 389 bytes from udp/[213.143.51.43]:17334 at 09:13:06.184956:
>>    -----------------------------------------------------------
>> -------------
>>    ACK sip:1010 at 88.XX.YY.ZZ:5080 SIP/2.0
>>    Via: SIP/2.0/UDP 10.78.108.57:51363;branch=z9hG4bK.BChE3dtFP;rport
>>    Call-ID: 5gQ3JhmBhp
>>    From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
>>    To: <sip:1010 at 192.168.230.143:5060>;tag=8Uc3QUtpgUjSm
>>    Contact: <sip:1009 at 213.143.51.43:17334;transport=udp>;+sip.instance="
>> <urn:uuid:a0671e9c-6a12-4d32-88d9-43f873b85ef6>"
>>    Max-Forwards: 70
>>    CSeq: 20 ACK
>>
>>    -----------------------------------------------------------
>> -------------
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657
>> [WARNING] switch_core_state_machine.c:687 367536c0-9e81-463d-9c5c-093cd85b292b
>> sofia/internal/1009 at 192.168.230.143:5060 Abandoned
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [NOTICE]
>> switch_core_state_machine.c:690 Hangup sofia/internal/1009 at 192.168.23
>> 0.143:5060 [CS_NEW] [WRONG_CALL_STATE]
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
>> switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060)
>> Running State Change CS_HANGUP (Cur 1 Tot 1)
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
>> switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.230.143:5060)
>> Callstate Change DOWN -> HANGUP
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
>> switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060)
>> State HANGUP
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
>> mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060 hanging
>> up, cause: WRONG_CALL_STATE
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060
>> Standard HANGUP, cause: WRONG_CALL_STATE
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060)
>> State HANGUP going to sleep
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.230.143:5060)
>> State Change CS_HANGUP -> CS_REPORTING
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060)
>> Running State Change CS_REPORTING (Cur 1 Tot 1)
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060)
>> State REPORTING
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060
>> Standard REPORTING, cause: WRONG_CALL_STATE
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060)
>> State REPORTING going to sleep
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.230.143:5060)
>> State Change CS_REPORTING -> CS_DESTROY
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_session.c:1713 Session 1 (sofia/internal/1009 at 192.168.2
>> 30.143:5060) Locked, Waiting on external entities
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [NOTICE]
>> switch_core_session.c:1731 Session 1 (sofia/internal/1009 at 192.168.2
>> 30.143:5060) Ended
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [NOTICE]
>> switch_core_session.c:1735 Close Channel sofia/internal/1009 at 192.168.23
>> 0.143:5060 [CS_DESTROY]
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.230.143:5060)
>> Running State Change CS_DESTROY (Cur 0 Tot 1)
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060)
>> State DESTROY
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060
>> Standard DESTROY
>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>> switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060)
>> State DESTROY going to sleep
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> José David Jurado Alonso <josedavid at zennio.com>
>>
>> *Área de Desarrollo de Software de Alto Nivel*
>>
>> *Dpto. Ingeniería*
>>
>> [image: Descripción: cid:image010.jpg at 01D27631.DF7E30F0]
>>
>> Zennio Avance y Tecnología, S.L.
>>
>> C/ Rio Jarama,132. Nave P-8.11
>>
>> 45007 - Toledo (Spain)
>>
>> T: +34 925 232 002 <+34%20925%2023%2020%2002>
>>
>> www.zennio.com
>>
>> [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\Z
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>> C:\Users\jjmanjarres\Desktop\twitter.jpg]
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>> C:\Users\jjmanjarres\Desktop\descarga.jpg]
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>> Descripción: C:\Users\jjmanjarres\Desktop\youtube.jpg]
>> <https://www.youtube.com/user/ZennioEnglish>
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>> la dirección de correo electrónico: info at zennio.com
>>
>> Please, consider the environment before printing this e-mail... Save
>> energy!
>>
>> Por favor, piensa en el medio ambiente antes de imprimir este e-mail... ¡Ahorra
>> energía!
>>
>> Disclaimer:
>> This message and any attached files transmitted with it, is confidential,
>> especially as regards personal data. It is intended solely for the use of
>> the individual or entity to whom it is addressed. If you are not the
>> intended recipient and have received this information in error or have
>> accessed it for any reason, please notify us of this fact by email reply
>> and then destroy or delete the message, refraining from any reproduction,
>> use, alteration, filing or communication to third parties of this message
>> and attached files on penalty of incurring legal responsibilities. The
>> opinions contained in this message and the attached archives, belong
>> exclusively to their sender and they do not represent the opinion of the
>> company unless it is said specifically and the sender is authorized for it.
>> The sender does not guarantee the integrity, the accuracy, the swift
>> delivery or the security of this email transmission, and assumes no
>> responsibility for any possible damage incurred through data capture, virus
>> incorporation or any manipulation carried out by third parties.
>>
>> Advertencia legal:
>> Este mensaje y, en su caso, los ficheros anexos son confidenciales,
>> especialmente en lo que respecta a los datos personales, y se dirigen
>> exclusivamente al destinatario referenciado. Si usted no lo es y lo ha
>> recibido por error o tiene conocimiento del mismo por cualquier motivo, le
>> rogamos que nos lo comunique por este medio y proceda a destruirlo o
>> borrarlo, y que en todo caso se abstenga de utilizar, reproducir, alterar,
>> archivar o comunicar a terceros el presente mensaje y ficheros anexos, todo
>> ello bajo pena de incurrir en responsabilidades legales. Las opiniones
>> contenidas en este mensaje y en los archivos adjuntos, pertenecen
>> exclusivamente a su remitente y no representan la opinión de la empresa
>> salvo que se diga expresamente y el remitente esté autorizado para ello. El
>> emisor no garantiza la integridad, rapidez o seguridad del presente correo,
>> ni se responsabiliza de posibles perjuicios derivados de la captura,
>> incorporaciones de virus o cualesquiera otras manipulaciones efectuadas por
>> terceros.
>>
>> 2017-07-19 21:00 GMT+02:00 Brian West <brian at freeswitch.org>:
>>
>>> 1. set up your local-network-acl . (rfc1918.auto is good start)
>>> 2. setup your ext-rtp-ip and ext-sip-ip and MAKE SURE you prefix it with
>>> autonat: so it activates the proper behavior for you.
>>>
>>> This should be all you have to do so you can talk to devices inside and
>>> outside the NAT at the same time.
>>>
>>> /b
>>>
>>>
>>> On Wed, Jul 19, 2017 at 9:30 AM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> +1 about forwarding upds...
>>>> But looking at the logs, we can't really see whether the call was
>>>> processed or not, just that it was "abandoned" and that usually happens
>>>> when freeswitch responds with a 401/407 and the client never
>>>> authenticates...
>>>>
>>>> enter:
>>>> sofia profile internal siptrace on
>>>>
>>>> to see all messages and make sure the client responds properly and
>>>> paste the signaling here.
>>>> ᐧ
>>>>
>>>> Regards,
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.work at gmail.com
>>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>>>
>>>> On Wed, Jul 19, 2017 at 4:18 PM, Joel Serrano <joel at gogii.net> wrote:
>>>>
>>>>> Double check your ACLs, yes. Also, I'm pretty sure you will also have
>>>>> to forward the RTP port range so you can have audio... Otherwise signaling
>>>>> will work, but your next problem will be one-way-audio.
>>>>>
>>>>>
>>>>> On Tue, Jul 18, 2017 at 11:43 PM, Jose David Jurado Alonso <
>>>>> josedavid at zennio.com> wrote:
>>>>>
>>>>>>
>>>>>> I've installed *freeswitch into a local environment* and I perform
>>>>>> SIP calls well using the server local IP.
>>>>>>
>>>>>> This server has connected to ADSL network with a *static public IP*
>>>>>> but when I try to call using the public IP instead of the local IP the
>>>>>> server receive the call not work...
>>>>>>
>>>>>> The clients can register well (or I think so). Others process like
>>>>>> apache work well.
>>>>>>
>>>>>> I NAT forwarding in the firewall the next ports: 5060, 5080, 7443.
>>>>>>
>>>>>> *$ netstat -putan*
>>>>>>
>>>>>> Active Internet connections (servers and established)
>>>>>> Proto Recv-Q Send-Q Local Address           Foreign Address         State       PID/Program name
>>>>>> tcp        0      0 192.168.230.143:2855    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>> tcp        0      0 192.168.230.143:2856    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>> tcp        0      0 192.168.230.143:5066    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>> tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN      1166/nginx -g daemo
>>>>>> tcp        0      0 192.168.230.143:7443    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>> tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN      1079/sshd
>>>>>> tcp        0      0 192.168.230.143:5080    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>> tcp        0      0 0.0.0.0:443             0.0.0.0:*               LISTEN      1166/nginx -g daemo
>>>>>> tcp        0      0 192.168.230.143:5060    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>> tcp        0      0 192.168.230.143:22      192.168.230.167:60298   ESTABLISHED 14816/sshd: ubuntu
>>>>>> tcp        0      0 192.168.230.143:22      192.168.230.167:60296   ESTABLISHED 14744/sshd: ubuntu
>>>>>> tcp6       0      0 :::80                   :::*                    LISTEN      1166/nginx -g daemo
>>>>>> tcp6       0      0 :::8021                 :::*                    LISTEN      15894/freeswitch
>>>>>> tcp6       0      0 :::22                   :::*                    LISTEN      1079/sshd
>>>>>> tcp6       0      0 ::1:5080                :::*                    LISTEN      15894/freeswitch
>>>>>> tcp6       0      0 :::443                  :::*                    LISTEN      1166/nginx -g daemo
>>>>>> tcp6       0      0 ::1:5060                :::*                    LISTEN      15894/freeswitch
>>>>>> udp        0      0 192.168.230.143:5060    0.0.0.0:*                           15894/freeswitch
>>>>>> udp        0      0 192.168.230.143:5080    0.0.0.0:*                           15894/freeswitch
>>>>>> udp        0      0 0.0.0.0:68              0.0.0.0:*                           1019/dhclient
>>>>>> udp6       0      0 ::1:5060                :::*                                15894/freeswitch
>>>>>> udp6       0      0 ::1:5080                :::*                                15894/freeswitch
>>>>>>
>>>>>> *$ fs_cli -x "sofia status"*
>>>>>>
>>>>>>                      Name      Type                                       Data  State
>>>>>> =================================================================================================
>>>>>>             external-ipv6   profile                   sip:mod_sofia@[::1]:5080  RUNNING (0)
>>>>>>           192.168.230.143     alias                                   internal  ALIASED
>>>>>>                  external   profile         sip:mod_sofia at 192.168.230.143:5080  RUNNING (0)
>>>>>>     external::example.com   gateway                    sip:joeuser at example.com  NOREG
>>>>>>             internal-ipv6   profile                   sip:mod_sofia@[::1]:5060  RUNNING (0)
>>>>>>                  internal   profile         sip:mod_sofia at 192.168.230.143:5060  RUNNING (0)
>>>>>> =================================================================================================
>>>>>>
>>>>>> *$ fs_cli -x "sofia status profile external"*
>>>>>>
>>>>>> =================================================================================================
>>>>>> Name                external
>>>>>> Domain Name         N/A
>>>>>> Auto-NAT            false
>>>>>> DBName              sofia_reg_external
>>>>>> Pres Hosts
>>>>>> Dialplan            XML
>>>>>> Context             public
>>>>>> Challenge Realm     auto_to
>>>>>> RTP-IP              192.168.230.143
>>>>>> SIP-IP              192.168.230.143
>>>>>> URL                 sip:mod_sofia at 192.168.230.143:5080
>>>>>> BIND-URL            sip:mod_sofia at 192.168.230.143:5080;transport=udp,tcp
>>>>>> HOLD-MUSIC          local_stream://moh
>>>>>> OUTBOUND-PROXY      N/A
>>>>>> CODECS IN           OPUS,G722,PCMU,PCMA,VP8,H264
>>>>>> CODECS OUT          OPUS,G722,PCMU,PCMA,VP8,H264
>>>>>> TEL-EVENT           101
>>>>>> DTMF-MODE           rfc2833
>>>>>> CNG                 13
>>>>>> SESSION-TO          0
>>>>>> MAX-DIALOG          0
>>>>>> NOMEDIA             false
>>>>>> LATE-NEG            true
>>>>>> PROXY-MEDIA         false
>>>>>> ZRTP-PASSTHRU       true
>>>>>> AGGRESSIVENAT       false
>>>>>> CALLS-IN            0
>>>>>> FAILED-CALLS-IN     0
>>>>>> CALLS-OUT           0
>>>>>> FAILED-CALLS-OUT    0
>>>>>> REGISTRATIONS       0
>>>>>>
>>>>>> *$ grep -r '"ext-' **
>>>>>>
>>>>>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>>>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>>>> sip_profiles/internal.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>>>> sip_profiles/internal.xml:    <param name="ext-sip-ip" value="auto-nat"/>
>>>>>>
>>>>>> When I perform a call the freeswitch.log file show:
>>>>>>
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1009 at 192.168.230.143:5060 [df895d9a-30f2-40b5-97bb-56e222c1c45f]
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_NEW (Cur 1 Tot 7)
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite from 88.33.111.22:57076 version: 1.9.0 -493-13f2f2a 64bit
>>>>>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10238 IP 88.33.111.22 Rejected by acl "domains". Falling back to Digest auth.
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.230.143:5060) State NEW
>>>>>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:2405 detaching session df895d9a-30f2-40b5-97bb-56e222c1c45f
>>>>>>
>>>>>> and passed some seconds:
>>>>>>
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [WARNING] switch_core_state_machine.c:687 df895d9a-30f2-40b5-97bb-56e222c1c45f sofia/internal/1009 at 192.168.230.143:5060 Abandoned
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/1009 at 192.168.230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_HANGUP (Cur 1 Tot 7)
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.230.143:5060) Callstate Change DOWN -> HANGUP
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060 hanging up, cause: WRONG_CALL_STATE
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060 Standard HANGUP, cause: WRONG_CALL_STATE
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP going to sleep
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_HANGUP -> CS_REPORTING
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_REPORTING (Cur 1 Tot 7)
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060 Standard REPORTING, cause: WRONG_CALL_STATE
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING going to sleep
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_REPORTING -> CS_DESTROY
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_session.c:1713 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Locked, Waiting on external entities
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1731 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Ended
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1009 at 192.168.230.143:5060 [CS_DESTROY]
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_DESTROY (Cur 0 Tot 7)
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060 Standard DESTROY
>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY going to sleep
>>>>>>
>>>>>> Can it be an ACL or sip_profile bad configuration?? Router firewall
>>>>>> configuration maybe?
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>>
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>>
>>> Allison prompts for FreeSWITCH:
>>>
>>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>>
>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>> *Skype:*briankwest
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fickledreams at yahoo.com  Fri Jul 21 12:04:10 2017
From: fickledreams at yahoo.com ('Yemi Obembe)
Date: Fri, 21 Jul 2017 12:04:10 +0000 (UTC)
Subject: [Freeswitch-users] Multiple endpoints bridge issue
References: <351917337.2364280.1500638650423.ref@mail.yahoo.com>
Message-ID: <351917337.2364280.1500638650423@mail.yahoo.com>

I am using bridge via mod_httapi to call multiple endpoints via an xml response like this:
<document type="text/freeswitch-httapi"><work><execute application="bridge" data="user/xxxxx1,user/xxxxx2,user/xxxxx3" /></work></document>

What I notice is that this works for one or two connected endpoints. But once there are three or four endpoints, other incoming calls drop. I changed to enterprise originate (i.e, :_:) and noticed that once a call is engaged, other calls don't come in till that one ends. Same if I separate users with pipe (user/xxxxx1|user/xxxxx2|user/xxxxx3). Even when I bridge just a user at time, <execute application="bridge" data="user/xxxxx1" /> the other calls drop once one is engaged.
What am I missing?

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From hemanth at advaitamtech.com  Fri Jul 21 12:30:34 2017
From: hemanth at advaitamtech.com (hemanth at advaitamtech.com)
Date: Fri, 21 Jul 2017 18:00:34 +0530 (IST)
Subject: [Freeswitch-users] Freeswitch call-ID configuration
Message-ID: <1500640234.26286570@apps.rackspace.com>


Hello,
 
I am using Freswitch as a media proxy and to handle SIP requests it is working as B2BUA,
 
Here i am not able handle 407 response from server, Because Freeswitch is sending different Call-ID and From tag in the Invite carrying proxy-authorization header,
 
Because of this behavior SIP server is sending 407 for Second Invite carrying Proxy-Authorization Header also,
 
Can any one please help me to configure freeswitch, So that it sends Same Call-ID and from tag in Both the Invite requests
 
Thanks & Regards
Hemanth
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From joelists at tm.net.uk  Fri Jul 21 12:44:08 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Fri, 21 Jul 2017 13:44:08 +0100
Subject: [Freeswitch-users] Multiple endpoints bridge issue
In-Reply-To: <351917337.2364280.1500638650423@mail.yahoo.com>
References: <351917337.2364280.1500638650423.ref@mail.yahoo.com>
 <351917337.2364280.1500638650423@mail.yahoo.com>
Message-ID: <FA109AFD-1F8D-446A-B1C0-B773D5041625@tm.net.uk>

Bridge is not a conferencing app.

Bridge is used, normally within a dial plan to link an incoming a-leg call to an outgoing b-leg which is created to the endpoint’s specified in the bridge command. 

So the behaviour your experiencing is correct when specifying multiple endpoints, as soon as an endpoint/user answers the call the bridge command cancels all others.


> On 21 Jul 2017, at 13:04, 'Yemi Obembe <fickledreams at yahoo.com> wrote:
> 
> I am using bridge via mod_httapi to call multiple endpoints via an xml response like this:
> 
> <document type="text/freeswitch-httapi"><work>
> <execute application="bridge" data="user/xxxxx1,user/xxxxx2,user/xxxxx3" />
> </work></document>
> 
> What I notice is that this works for one or two connected endpoints. But once there are three or four endpoints, other incoming calls drop. I changed to enterprise originate (i.e, :_:) and noticed that once a call is engaged, other calls don't come in till that one ends. Same if I separate users with pipe (user/xxxxx1|user/xxxxx2|user/xxxxx3). Even when I bridge just a user at time, <execute application="bridge" data="user/xxxxx1" /> the other calls drop once one is engaged.
> 
> What am I missing?
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mark.melling at savageminds.com  Fri Jul 21 13:06:57 2017
From: mark.melling at savageminds.com (Mark Melling)
Date: Fri, 21 Jul 2017 13:06:57 +0000
Subject: [Freeswitch-users] Understanding a dial string returned by
	group_call
Message-ID: <CAJp-f1MA-mT_CrC+SrRZJx4WicwPtBV==C7vQrPBK0_6VHiD7A@mail.gmail.com>

When I execute a group_call command I get responses like this

[^^:sip_invite_domain=example.com:presence_id=1010 at example.com
]sofia/external/sip:1010 at 192.168.56.1:59527
;ob;fs_nat=yes;fs_path=sip%3A1010%40192.168.56.1%3A59527%3Bob

Are there any decent resources that would help me understand this better?

As I understand it the [...] at the beginning means, apply variables to the
channel, but what does the ^^ mean?

And what does the bit after the address do/mean? i.e. in this specific case
the bit

;ob;fs_nat=yes;fs_path=sip%3A1010%40192.168.56.1%3A59527%3Bob

Thanks

Mark
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From gmaruzz at gmail.com  Fri Jul 21 15:10:42 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 21 Jul 2017 17:10:42 +0200
Subject: [Freeswitch-users] Freeswitch call-ID configuration
In-Reply-To: <1500640234.26286570@apps.rackspace.com>
References: <1500640234.26286570@apps.rackspace.com>
Message-ID: <CALXCt0rpbwwpbBkHH2YpwqJgMeNb8WxQC4r-WtzSpGjpPwSKEg@mail.gmail.com>

On 21 July 2017 at 14:30, hemanth at advaitamtech.com <hemanth at advaitamtech.com
> wrote:

> Hello,
>
>
>
> I am using Freswitch as a media proxy and to handle SIP requests it is
> working as B2BUA,
>
>
>
> Here i am not able handle 407 response from server, Because Freeswitch is
> sending different Call-ID and From tag in the Invite carrying
> proxy-authorization header,
>
>
>
> Because of this behavior SIP server is sending 407 for Second Invite
> carrying Proxy-Authorization Header also,
>
>
>
> Can any one please help me to configure freeswitch, So that it sends Same
> Call-ID and from tag in Both the Invite requests
>


check privacy presentation on http://freeswitch.org/confluence



>
>
> Thanks & Regards
>
> Hemanth
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From gmaruzz at gmail.com  Fri Jul 21 15:14:48 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 21 Jul 2017 17:14:48 +0200
Subject: [Freeswitch-users] Freeswitch call-ID configuration
In-Reply-To: <CALXCt0rpbwwpbBkHH2YpwqJgMeNb8WxQC4r-WtzSpGjpPwSKEg@mail.gmail.com>
References: <1500640234.26286570@apps.rackspace.com>
 <CALXCt0rpbwwpbBkHH2YpwqJgMeNb8WxQC4r-WtzSpGjpPwSKEg@mail.gmail.com>
Message-ID: <CALXCt0oUC3E6MiuUpoyQEcguBcCZUZCjt15HggWMh6Fcx9nJ4g@mail.gmail.com>

On 21 July 2017 at 17:10, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

>
>
> On 21 July 2017 at 14:30, hemanth at advaitamtech.com <
> hemanth at advaitamtech.com> wrote:
>
>> Hello,
>>
>>
>>
>> I am using Freswitch as a media proxy and to handle SIP requests it is
>> working as B2BUA,
>>
>>
>>
>> Here i am not able handle 407 response from server, Because Freeswitch is
>> sending different Call-ID and From tag in the Invite carrying
>> proxy-authorization header,
>>
>>
>>
>> Because of this behavior SIP server is sending 407 for Second Invite
>> carrying Proxy-Authorization Header also,
>>
>>
>>
>> Can any one please help me to configure freeswitch, So that it sends Same
>> Call-ID and from tag in Both the Invite requests
>>
>
>
> check privacy presentation on http://freeswitch.org/confluence
>
>
Oooops, my answer is totally wrong. Sorry I had not read correctly.






>
>
>>
>>
>> Thanks & Regards
>>
>> Hemanth
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From gmaruzz at gmail.com  Fri Jul 21 16:31:31 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 21 Jul 2017 18:31:31 +0200
Subject: [Freeswitch-users] SIM7100 LTE modem with USB audio
In-Reply-To: <CACOYK=jtpcNSyzveoWKVRssHad-_6CwJtZ8H72d=qnHNS5niVA@mail.gmail.com>
References: <CACOYK=jtpcNSyzveoWKVRssHad-_6CwJtZ8H72d=qnHNS5niVA@mail.gmail.com>
Message-ID: <CALXCt0rUcK8wck0UHu1u9Tqif6X+6PbNeEimZj3PXrC7t8C=Zw@mail.gmail.com>

Hello Stanislav,

have you had time to tinker with it?

How it goes?

Also, why you would prefer this one instead of the already supported ones?

-giovanni


On 8 July 2017 at 21:28, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> Simcom has recently released a new 4G/LTE modem, and it has USB audio
> support.
> You can find sim7100_usb_audio_application_note_v0.01.pdf   with
> details at the vendor site, or at techship.com  after registration.
>
> It transmits 8kHZ, 16-bit PCM audio via a USB UART simulation.
>
> So, in theory, gsmopen module may be adapted to it (or maybe a new
> module is worth starting).
>
> I ordered a sample, will check it out soon.
>
> cheers,
> stanislav
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From s.safarov at gmail.com  Fri Jul 21 16:54:11 2017
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 21 Jul 2017 16:54:11 +0000
Subject: [Freeswitch-users] Freeswitch call-ID configuration
In-Reply-To: <1500640234.26286570@apps.rackspace.com>
References: <1500640234.26286570@apps.rackspace.com>
Message-ID: <CAHtxdDf7WkKhbOA-PmDQQtW=QM5P3TT4CS12djep1oc7Hz2u9Q@mail.gmail.com>

Think such task need to solve using kamailio, may be with rtpproxy.

пт, 21 июля 2017 г., 15:37 hemanth at advaitamtech.com <
hemanth at advaitamtech.com>:

> Hello,
>
>
>
> I am using Freswitch as a media proxy and to handle SIP requests it is
> working as B2BUA,
>
>
>
> Here i am not able handle 407 response from server, Because Freeswitch is
> sending different Call-ID and From tag in the Invite carrying
> proxy-authorization header,
>
>
>
> Because of this behavior SIP server is sending 407 for Second Invite
> carrying Proxy-Authorization Header also,
>
>
>
> Can any one please help me to configure freeswitch, So that it sends Same
> Call-ID and from tag in Both the Invite requests
>
>
>
> Thanks & Regards
>
> Hemanth
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From rahul.ultimate at gmail.com  Sat Jul 22 00:30:32 2017
From: rahul.ultimate at gmail.com (Rahul MathuR)
Date: Sat, 22 Jul 2017 06:00:32 +0530
Subject: [Freeswitch-users] Integration of Google Speech API V2
In-Reply-To: <CAOfqg5b3q4TkKUHU7o+90nfXXvhhaWJX_yX1+qZLz1H13Znijw@mail.gmail.com>
References: <CAM8wsqJ+Voj65OBhWfDjjzn3c9vhcU0MaN32eDE_dW5-vh50Dg@mail.gmail.com>
 <CAOfqg5b3q4TkKUHU7o+90nfXXvhhaWJX_yX1+qZLz1H13Znijw@mail.gmail.com>
Message-ID: <CAM8wsqLToVcA9=a6wC3fOsU=VmsLV-R+YqQbPH6GxWL1wRzCfQ@mail.gmail.com>

Thanks Uday for sharing this info.

Could you please share the working config set. Please replace your key with
"UDAY_KEY".
I will be very grateful.


Thanks.


On Jul 21, 2017 12:38 PM, "Uday kumar" <udy786 at gmail.com> wrote:

I have implemented Google Speech on Freeswitch 1.6.16 Transcribing
Voicemail and its working really nice without any issue. When user get
voicemail then server send email to user with recording and text of
recording.

I am using SMTP to send mail. In switch.con.xml

<param name="mailer-app" value="/usr/bin/php /var/www/html/PHPMailer/
mailer_app.php"/>

Download mailer_app.php from freeswitch wiki and you can implement google
speech API and code in that file.

Thanks
Uday.

On Wed, Jul 19, 2017 at 2:07 PM, Rahul MathuR <rahul.ultimate at gmail.com>
wrote:

> Hi,
>
> I'm trying to integrate Google cloud speech recognition v2 in it. I can
> get the audio recorded, have created Service key and API key but whenever I
> try to access it, I just get 403 access denied. I am at my wits end here.
>
> Has anybody tried it ? were you successful ? Could you please guide me how
> to do it ?
> I'll be grateful to you if this works !
>
>
>
> --
> Warm Regds.
> MathuRahul
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regard
Uday
Site:- www.shareyourknowledge.in
Mobile:- +91-9377579349 <+91%2093775%2079349>

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From mail at paulzillmann.de  Sat Jul 22 00:36:01 2017
From: mail at paulzillmann.de (Paul Zillmann)
Date: Sat, 22 Jul 2017 02:36:01 +0200
Subject: [Freeswitch-users] traffic and system monitoring
In-Reply-To: <2016365814.617648.1500487315036@mail.yahoo.com>
References: <1012898889.2268619.1500304838441.ref@mail.yahoo.com>
 <1012898889.2268619.1500304838441@mail.yahoo.com>
 <dd31c0e0-74b5-3a5a-9236-62bec7948049@paulzillmann.de>
 <2016365814.617648.1500487315036@mail.yahoo.com>
Message-ID: <2344dcfc-2e41-45e5-876e-a38f9dba5ef2@paulzillmann.de>

Hey Robert,

sorry for my late reply. Your messages are flagged as spam here (Google
Mail)
Yes, check_mk and nagios are free and open source.
It is more likely a system health monitor (is port 5060 reachable and
what is the answer => active checks)
And it's reading from standard unix tools (like ps, free - CPU usage =>
passive checks)
It saves that data in a history (and builds fancy graphs) and sends me
mails / SMS messages when some service is down.

I've attached a screenshot from the webinterface.

So check_mk can't monitor RTP traffic, but you can write your own checks
in Python (at least the webview)

Am 19.07.2017 um 20:01 schrieb robert mundkowsky:
> Are check_mk / nagios free?
>
> Looks like there is a SIP plugin. Is there a RTP plugin?
>
>
>
> ------------------------------------------------------------------------
> On Wednesday, July 19, 2017, 9:25:36 AM EDT, Paul Zillmann
> <mail at paulzillmann.de> wrote:
>
>
> Hey Robert,
>
> I'm using check_mk / nagios for external monitoring and iftop and top
> / htop for on site analysis.
>
>
> Am 17.07.2017 um 17:20 schrieb robert mundkowsky:
>>
>>  
>>
>> Is Homer the best free option for traffic and system monitoring GUI?
>>
>>                Or is CACTI better option?
>>
>>  
>>
>> Robert
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From omortimer at gmail.com  Sat Jul 22 18:13:51 2017
From: omortimer at gmail.com (Oz Mortimer)
Date: Sat, 22 Jul 2017 19:13:51 +0100
Subject: [Freeswitch-users] Behaviour of blocking applications
Message-ID: <B723B014-65C8-42D8-B197-A923A68F4EF3@gmail.com>

Hi,

I know I can work around this issue, but was wondering if the behaviour is expected…

If I use an application that blocks, mod_http_cache for example and the web server is unavailable - the call will stay up until the connect-timeout is reached. Obviously I can set a lower connect-timeout, but this doesn’t really resolve the issue. 

So, say I set connect-timeout to 3 seconds and place a 1 second call via soft phone (caller hangup), the billable duration will be 3 seconds - should the billable duration not be 1 second to match the calling party? also, shouldn’t a hangup force any running applications to stop?

Again, obviously I can avoid this by being a bit smarter with the usage of http_cache, etc - but was really wondering if a) thats expected behaviour and the callee should count the duration based on the ACK from the BYE (which I suspect non do), or b) maybe there is a duration variable that I’ve not yet seen that contains the call duration until we receive the BYE from caller?. 

I would have thought (and Im probably wrong), billable duration s/u/m should resemble the SIP durations, i.e. endtime = time BYE was received?

The other byproduct of this symptom is that the channels will stack up until call-timeout is reached - if it was left to the default of 300 seconds - a dead web server could soon cause the switch to burn out. Again, easily resolvable by not using blocking applications, doing things in the background, etc but… 

Thoughts & postcards?
Thanks
Oz.






From ssinyagin at gmail.com  Sun Jul 23 08:17:55 2017
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Sun, 23 Jul 2017 10:17:55 +0200
Subject: [Freeswitch-users] SIM7100 LTE modem with USB audio
In-Reply-To: <CALXCt0rUcK8wck0UHu1u9Tqif6X+6PbNeEimZj3PXrC7t8C=Zw@mail.gmail.com>
References: <CACOYK=jtpcNSyzveoWKVRssHad-_6CwJtZ8H72d=qnHNS5niVA@mail.gmail.com>
 <CALXCt0rUcK8wck0UHu1u9Tqif6X+6PbNeEimZj3PXrC7t8C=Zw@mail.gmail.com>
Message-ID: <CACOYK=hn7ZNFmkeTe7-HX=9t70zc8vKyxbdrApU1Nf4MX7y3eQ@mail.gmail.com>

Hi Giovanni, I only had time to configure it for Internet access:

https://txlab.wordpress.com/2017/07/23/simcom-sim7100e-lte-modem/

The benefit is mPCIe interface, so it can be placed inside and enclosure.
No silly dongles any more :)

I'll tinker with voice some time later. I can also provide you SSH access
if you wish to work on gsmopen update.



On 21 Jul 2017 18:36, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

> Hello Stanislav,
>
> have you had time to tinker with it?
>
> How it goes?
>
> Also, why you would prefer this one instead of the already supported ones?
>
> -giovanni
>
>
> On 8 July 2017 at 21:28, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>
>> Simcom has recently released a new 4G/LTE modem, and it has USB audio
>> support.
>> You can find sim7100_usb_audio_application_note_v0.01.pdf   with
>> details at the vendor site, or at techship.com  after registration.
>>
>> It transmits 8kHZ, 16-bit PCM audio via a USB UART simulation.
>>
>> So, in theory, gsmopen module may be adapted to it (or maybe a new
>> module is worth starting).
>>
>> I ordered a sample, will check it out soon.
>>
>> cheers,
>> stanislav
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From iskren.hadzhinedev at ikiji.com  Sun Jul 23 09:26:39 2017
From: iskren.hadzhinedev at ikiji.com (Iskren Hadzhinedev)
Date: Sun, 23 Jul 2017 12:26:39 +0300
Subject: [Freeswitch-users] Path header handling
Message-ID: <6164d14f-b630-0d71-9d68-8231abf71226@ikiji.com>

Hi,

I have a kamailio proxy in front of FreeSWITCH. When a user registers, 
kamailio adds a path header like so:

Path: <sip:KAMAILIO_PUBLIC_IP;lr;received=sip:UAC_PUBLIC_IP:5063>

However, when FreeSWITCH sends the initial invite, it sends a Route 
header like this:

Route: <sip:KAMAILIO_PUBLIC_IP>;lr;received=sip:UAC_PUBLIC_IP:5063

Is it possible to instruct FreeSWITCH to add the lr;received parameters 
inside the angle brackets as a part of the URI and how? e.g. how to achieve

Route: <sip:KAMAILIO_PUBLIC_IP;lr;received=sip:UAC_PUBLIC_IP:5063>

Thank you!

Kind regards,
Iskren Hadzhinedev
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From gmaruzz at gmail.com  Sun Jul 23 10:24:55 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 23 Jul 2017 12:24:55 +0200
Subject: [Freeswitch-users] Path header handling
In-Reply-To: <6164d14f-b630-0d71-9d68-8231abf71226@ikiji.com>
References: <6164d14f-b630-0d71-9d68-8231abf71226@ikiji.com>
Message-ID: <CALXCt0qSqnKvaLSgTmpMaO-KK-ip6u4fN7J4d=E1Lypm+xzhMA@mail.gmail.com>

On 23 July 2017 at 11:26, Iskren Hadzhinedev <iskren.hadzhinedev at ikiji.com>
wrote:

> Hi,
>
> I have a kamailio proxy in front of FreeSWITCH. When a user registers,
> kamailio adds a path header like so:
>
> Path: <sip:KAMAILIO_PUBLIC_IP;lr;received=sip:UAC_PUBLIC_IP:5063>
>
> However, when FreeSWITCH sends the initial invite, it sends a Route header
> like this:
>
> Route: <sip:KAMAILIO_PUBLIC_IP>;lr;received=sip:UAC_PUBLIC_IP:5063
>
> Is it possible to instruct FreeSWITCH to add the lr;received parameters
> inside the angle brackets as a part of the URI and how? e.g. how to achieve
>
> Route: <sip:KAMAILIO_PUBLIC_IP;lr;received=sip:UAC_PUBLIC_IP:5063>
>

To my fellow FreeSWITCHers, this is the original  mail thread, where Daniel
seems to find a problem in FreeSWITCH:

https://lists.kamailio.org/pipermail/sr-users/2017-July/097893.html
<https://lists.kamailio.org/pipermail/sr-users/2017-July/097884.html>


-giovanni
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From fickledreams at yahoo.com  Sat Jul 22 02:26:22 2017
From: fickledreams at yahoo.com ('Yemi Obembe)
Date: Sat, 22 Jul 2017 02:26:22 +0000 (UTC)
Subject: [Freeswitch-users] Multiple endpoints bridge issue
In-Reply-To: <mailman.259.1500649915.1301.freeswitch-users@lists.freeswitch.org>
References: <mailman.259.1500649915.1301.freeswitch-users@lists.freeswitch.org>
Message-ID: <2145934767.3036179.1500690382461@mail.yahoo.com>

I totally understand that it should cancel all others. If Call A comes in and rings across user/xxxxx1,user/xxxxx2,user/xxxxx3 and xxxxx1 picks, I know ~2 and ~3 should be dropped. What I'd assume is that if Call B comes in (as Call A is engaged to xxxxx1), xxxxx2 and xxxxx3 should also be able to pick as they are available. As it is now, Call B will ring on them, but once they pick, as long as an existing call (A) exists, Call B drops. In other words, there can't be concurrent calls on the system.
(What I am trying to create is a call center system where multiple agents can be engaged with different calls at the same time).


Bridge is not a conferencing app.
Bridge is used, normally within a dial plan to link an incoming a-leg call to an outgoing b-leg which is created to the endpoint’s specified in the bridge command. 
So the behaviour your experiencing is correct when specifying multiple endpoints, as soon as an endpoint/user answers the call the bridge command cancels all others.


On 21 Jul 2017, at 13:04, 'Yemi Obembe <fickledreams at yahoo.com> wrote:




I am using bridge via mod_httapi to call multiple endpoints via an xml response like this:




<document type="text/freeswitch-httapi"><work>

<execute application="bridge" data="user/xxxxx1,user/xxxxx2,user/xxxxx3" />

</work></document>




What I notice is that this works for one or two connected endpoints. But once there are three or four endpoints, other incoming calls drop. I changed to enterprise originate (i.e, :_:) and noticed that once a call is engaged, other calls don't come in till that one ends. Same if I separate users with pipe (user/xxxxx1|user/xxxxx2|user/xxxxx3). Even when I bridge just a user at time, <execute application="bridge" data="user/xxxxx1" /> the other calls drop once one is engaged.




What am I missing?




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From lists at telefaks.de  Sun Jul 23 22:27:58 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Mon, 24 Jul 2017 00:27:58 +0200
Subject: [Freeswitch-users] traffic and system monitoring
In-Reply-To: <2344dcfc-2e41-45e5-876e-a38f9dba5ef2@paulzillmann.de>
References: <1012898889.2268619.1500304838441.ref@mail.yahoo.com>
 <1012898889.2268619.1500304838441@mail.yahoo.com>
 <dd31c0e0-74b5-3a5a-9236-62bec7948049@paulzillmann.de>
 <2016365814.617648.1500487315036@mail.yahoo.com>
 <2344dcfc-2e41-45e5-876e-a38f9dba5ef2@paulzillmann.de>
Message-ID: <597522EE.7030705@telefaks.de>

We have implemented Nagios/Icinga and Check_MK with Freeswitch. Both do
a good (passive) job, as Robert pointed out.
However we also use ESL to check data from hangup events and you may
also scan CDRs for RTP statistics.
We usually store those relevant data into Memcache, which is then
queried on demand by a Nagios/Icinga or Check_MK MRPE script.

Best regards
Peter


On 07/22/17 02:36, Paul Zillmann wrote:
> Hey Robert,
>
> sorry for my late reply. Your messages are flagged as spam here
> (Google Mail)
> Yes, check_mk and nagios are free and open source.
> It is more likely a system health monitor (is port 5060 reachable and
> what is the answer => active checks)
> And it's reading from standard unix tools (like ps, free - CPU usage
> => passive checks)
> It saves that data in a history (and builds fancy graphs) and sends me
> mails / SMS messages when some service is down.
>
> I've attached a screenshot from the webinterface.
>
> So check_mk can't monitor RTP traffic, but you can write your own
> checks in Python (at least the webview)
>
> Am 19.07.2017 um 20:01 schrieb robert mundkowsky:
>> Are check_mk / nagios free?
>>
>> Looks like there is a SIP plugin. Is there a RTP plugin?
>>
>>
>>
>> ------------------------------------------------------------------------
>> On Wednesday, July 19, 2017, 9:25:36 AM EDT, Paul Zillmann
>> <mail at paulzillmann.de> wrote:
>>
>>
>> Hey Robert,
>>
>> I'm using check_mk / nagios for external monitoring and iftop and top
>> / htop for on site analysis.
>>
>>
>> Am 17.07.2017 um 17:20 schrieb robert mundkowsky:
>>>
>>>  
>>>
>>> Is Homer the best free option for traffic and system monitoring GUI?
>>>
>>>                Or is CACTI better option?
>>>
>>>  
>>>
>>> Robert
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach 

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

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From andyf at andyit.com.au  Mon Jul 24 02:31:19 2017
From: andyf at andyit.com.au (Andy Farkas)
Date: Mon, 24 Jul 2017 12:31:19 +1000
Subject: [Freeswitch-users] RFC2543
Message-ID: <59755BF7.7080203@andyit.com.au>


Can anybody tell me what this log message means?

2017-07-24 12:10:59.006338 [WARNING] switch_core_media.c:3896 RFC2543 
from March 1999 called; They want their 0.0.0.0 hold method back.....

FreeSWITCH Version 1.6.8~64-bit ( 64bit)

running on FreeBSD 11.0/STABLE r315121

-andyf



From josedavid at zennio.com  Mon Jul 24 06:51:11 2017
From: josedavid at zennio.com (Jose David Jurado Alonso)
Date: Mon, 24 Jul 2017 08:51:11 +0200
Subject: [Freeswitch-users] Freeswitch SIP call not work using a static
 public IP
In-Reply-To: <CANeCd7pYKp-kJOrtrb8y56TaEV3yCcYfrD2K1Y-RNn=YLuuYSA@mail.gmail.com>
References: <CANeCd7qgLdOv8AOnZUGFfpn8DmXVEUDNUYRarfMmrfmwZurHbA@mail.gmail.com>
 <CAMtXxQm8oj9TfJVLfmukL7BrEEkQAA0GB3xrmxGOnn5fbLGE=Q@mail.gmail.com>
 <CAFGRPVp2_=WpG9SWctEKVdYEn5_dwi1pE7gys0KXrHUg3OXSEQ@mail.gmail.com>
 <CAEJMVkDh2vrGnxWagzKyWYp9+aR+Q5NaVsjtLvPHQVTs46Nmjw@mail.gmail.com>
 <CANeCd7oTVR1eTZRYiHRw0JaijGWjo-Qq=pd36iFh0Fed6kDjOw@mail.gmail.com>
 <CAEJMVkDQZQe_aKJP89CBAT8G3FMxBKxs-LkSARephg+ipuwxzw@mail.gmail.com>
 <CANeCd7pYKp-kJOrtrb8y56TaEV3yCcYfrD2K1Y-RNn=YLuuYSA@mail.gmail.com>
Message-ID: <CANeCd7rJ+ESes-ae5DWvjm7NqhUgQL+fs7QrKEz=u_O9HPiViw@mail.gmail.com>

I can not call between mobile devices outside the local network (where the
server is) yet. Using the local IP calls correctly.

MOBILE_1 (217.33.XX.ZZ)  call to MOBILE_2 (205.66.XX.ZZ) as follows: "
sip:1009 at 88.22.33.44:5080" being 88.22.33.44 the public IP of the server.

 Both mobile have register correctly using "88.22.33.44" and "5060" as SIP
register and "1010" default user to mobile 1 and "1009" default user to
mobile 2.

If I change the public IP for private on both devices, I connect the mobile
to the local network (where the server is) via wifi then if it works
correctly calling to ""sip:1009 at 192.168.230.143:5080" being 192.168.230.143
the private IP of the server.

In short, using private IP within a local network calls between mobile
phones work correctly but using different networks for mobile and server,
calls through public IP are not made.

The scenario I want is: Mobile 1 is anywhere connected to any wifi or
mobile network, mobile 2 same. Both must be able to call to another using
the FS server through a public IP or URL domain.



José David Jurado Alonso <josedavid at zennio.com>

*Área de Desarrollo de Software de Alto Nivel*

*Dpto. Ingeniería*

[image: Descripción: cid:image010.jpg at 01D27631.DF7E30F0]

Zennio Avance y Tecnología, S.L.

C/ Rio Jarama,132. Nave P-8.11

45007 - Toledo (Spain)

T: +34 925 232 002

www.zennio.com

[image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\Z zennio.jpg]
<http://www.zennio.com/> [image: Descripción: Descripción:
C:\Users\jjmanjarres\Desktop\twitter.jpg] <http://www.twitter.com/Zennio_en>
 [image: Descripción: Descripción:
C:\Users\jjmanjarres\Desktop\descarga.jpg]
<http://www.facebook.com/ZennioInternational> [image: Descripción:
Descripción: C:\Users\jjmanjarres\Desktop\linkedin.png]
<http://www.linkedin.com/company/zennio> [image: Descripción: Descripción:
C:\Users\jjmanjarres\Desktop\youtube.jpg]
<https://www.youtube.com/user/ZennioEnglish>

Zennio Avance y Tecnología S.L le informa de los siguientes extremos:
Los datos por usted suministrados pasarán a formar parte de un fichero cuyo
responsable es Zennio Avance y Tecnología S.L. Dicho fichero se encuentra
legalmente inscrito en el Registro General de Protección de Datos de la
Agencia Española de Protección de Datos. Los datos por usted suministrados
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adoptado las medidas de seguridad exigidas en función del nivel de los
datos suministrados, instalando las medidas técnicas y organizativas
necesarias, habida cuenta del estado de la tecnología, a fin de evitar su
pérdida, alteración, uso inadecuado o accesos no autorizados a los mismos.
Para el ejercicio de sus derechos de acceso, rectificación, cancelación y
oposición deberá dirigirse a la dirección del Responsable de Fichero Zennio
Avance y Tecnología S.L C/ Rio Jarama, 132. Nave P-8.11, 45007, TOLEDO o a
la dirección de correo electrónico: info at zennio.com

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Disclaimer:
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accessed it for any reason, please notify us of this fact by email reply
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opinions contained in this message and the attached archives, belong
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The sender does not guarantee the integrity, the accuracy, the swift
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contenidas en este mensaje y en los archivos adjuntos, pertenecen
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terceros.

2017-07-21 7:40 GMT+02:00 Jose David Jurado Alonso <josedavid at zennio.com>:

> Yes, both mobile clients use their own data network (4G) and access always
> by the public IP..
>
> José David Jurado Alonso <josedavid at zennio.com>
>
> *Área de Desarrollo de Software de Alto Nivel*
>
> *Dpto. Ingeniería*
>
> [image: Descripción: cid:image010.jpg at 01D27631.DF7E30F0]
>
> Zennio Avance y Tecnología, S.L.
>
> C/ Rio Jarama,132. Nave P-8.11
>
> 45007 - Toledo (Spain)
>
> T: +34 925 232 002 <+34%20925%2023%2020%2002>
>
> www.zennio.com
>
> [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\Z
> zennio.jpg] <http://www.zennio.com/> [image: Descripción: Descripción:
> C:\Users\jjmanjarres\Desktop\twitter.jpg]
> <http://www.twitter.com/Zennio_en> [image: Descripción: Descripción:
> C:\Users\jjmanjarres\Desktop\descarga.jpg]
> <http://www.facebook.com/ZennioInternational> [image: Descripción:
> Descripción: C:\Users\jjmanjarres\Desktop\linkedin.png]
> <http://www.linkedin.com/company/zennio> [image: Descripción:
> Descripción: C:\Users\jjmanjarres\Desktop\youtube.jpg]
> <https://www.youtube.com/user/ZennioEnglish>
>
> Zennio Avance y Tecnología S.L le informa de los siguientes extremos:
> Los datos por usted suministrados pasarán a formar parte de un fichero
> cuyo responsable es Zennio Avance y Tecnología S.L. Dicho fichero se
> encuentra legalmente inscrito en el Registro General de Protección de Datos
> de la Agencia Española de Protección de Datos. Los datos por usted
> suministrados serán empleados con fines de gestión, Zennio Avance y
> Tecnología S.L ha adoptado las medidas de seguridad exigidas en función del
> nivel de los datos suministrados, instalando las medidas técnicas y
> organizativas necesarias, habida cuenta del estado de la tecnología, a fin
> de evitar su pérdida, alteración, uso inadecuado o accesos no autorizados a
> los mismos.
> Para el ejercicio de sus derechos de acceso, rectificación, cancelación y
> oposición deberá dirigirse a la dirección del Responsable de Fichero Zennio
> Avance y Tecnología S.L C/ Rio Jarama, 132. Nave P-8.11, 45007, TOLEDO o a
> la dirección de correo electrónico: info at zennio.com
>
> Please, consider the environment before printing this e-mail... Save
> energy!
>
> Por favor, piensa en el medio ambiente antes de imprimir este e-mail... ¡Ahorra
> energía!
>
> Disclaimer:
> This message and any attached files transmitted with it, is confidential,
> especially as regards personal data. It is intended solely for the use of
> the individual or entity to whom it is addressed. If you are not the
> intended recipient and have received this information in error or have
> accessed it for any reason, please notify us of this fact by email reply
> and then destroy or delete the message, refraining from any reproduction,
> use, alteration, filing or communication to third parties of this message
> and attached files on penalty of incurring legal responsibilities. The
> opinions contained in this message and the attached archives, belong
> exclusively to their sender and they do not represent the opinion of the
> company unless it is said specifically and the sender is authorized for it.
> The sender does not guarantee the integrity, the accuracy, the swift
> delivery or the security of this email transmission, and assumes no
> responsibility for any possible damage incurred through data capture, virus
> incorporation or any manipulation carried out by third parties.
>
> Advertencia legal:
> Este mensaje y, en su caso, los ficheros anexos son confidenciales,
> especialmente en lo que respecta a los datos personales, y se dirigen
> exclusivamente al destinatario referenciado. Si usted no lo es y lo ha
> recibido por error o tiene conocimiento del mismo por cualquier motivo, le
> rogamos que nos lo comunique por este medio y proceda a destruirlo o
> borrarlo, y que en todo caso se abstenga de utilizar, reproducir, alterar,
> archivar o comunicar a terceros el presente mensaje y ficheros anexos, todo
> ello bajo pena de incurrir en responsabilidades legales. Las opiniones
> contenidas en este mensaje y en los archivos adjuntos, pertenecen
> exclusivamente a su remitente y no representan la opinión de la empresa
> salvo que se diga expresamente y el remitente esté autorizado para ello. El
> emisor no garantiza la integridad, rapidez o seguridad del presente correo,
> ni se responsabiliza de posibles perjuicios derivados de la captura,
> incorporaciones de virus o cualesquiera otras manipulaciones efectuadas por
> terceros.
>
> 2017-07-20 22:16 GMT+02:00 Brian West <brian at freeswitch.org>:
>
>> also make sure the clients aren't doing stun requests that are behind the
>> nat with FreeSWITCH....
>>
>> /b
>>
>>
>> On Thu, Jul 20, 2017 at 2:28 AM, Jose David Jurado Alonso <
>> josedavid at zennio.com> wrote:
>>
>>> Many thanks to all for the answers, I found the solution after writing
>>> the problem but I did not have time to comment on it here.
>>>
>>> The correct configuration is the one that Brian says although I had not
>>> set "autonat:" as a public IP prefix. I've tried it now and it works too.
>>>
>>> However, it does not work in all cases. The only case in which it works
>>> correctly is the following:
>>>
>>> - Video intercom device and FS server in the same network.
>>> - Mobile client in other net.
>>>
>>> But in many other cases it does NOT work properly:
>>>
>>> - Video intercom device, FS server and mobile in the same network.
>>> - Mobile to Mobile (mobiles in different networks and server in other)
>>> - Mobile to Mobile (all in the same network)
>>>
>>>
>>> My current configuration is the next:
>>>
>>> - sip_profiles/external.xml
>>>
>>>     <param name="local-network-acl" value="rfc1918.auto"/>
>>>     <param name="ext-rtp-ip" value="autonat:88.2.243.70"/>
>>>     <param name="ext-sip-ip" value="autonat:88.2.243.70"/>
>>>
>>> - sip_profiles/internal.xml
>>>
>>>     <param name="apply-nat-acl" value="rfc1918.auto"/>
>>>     <param name="local-network-acl" value="rfc1918.auto"/>
>>>     <param name="ext-rtp-ip" value="autonat:88.2.243.70"/>
>>>     <param name="ext-sip-ip" value="autonat:88.2.243.70"/>
>>>
>>>
>>> I test using "nat.auto" too. I can attach the entire file if needed.
>>>
>>>
>>> The call is "abandoned" and show show "Rejected by acl "domains".
>>> Falling back to Digest auth." among other things.
>>>
>>> Source mobile as register as "1009 at 88.XX.YY.ZZ" and target mobile as
>>> "1010 at 88.XX.YY.ZZ". The call is perform as "sip:1010 at 88.XX.YY.ZZ:5080"
>>>
>>> The trace log of "Mobile to Mobile" call (I replace public IP to
>>> 88.XX.YY.ZZ):
>>>
>>>
>>> recv 1401 bytes from udp/[213.143.51.43]:17334 at 09:13:06.058418:
>>>    -----------------------------------------------------------
>>> -------------
>>>    INVITE sip:1010 at 88.XX.YY.ZZ:5080 SIP/2.0
>>>    Via: SIP/2.0/UDP 10.78.108.57:51363;branch=z9hG4bK.BChE3dtFP;rport
>>>    From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
>>>    To: sip:1010 at 192.168.230.143:5060
>>>    CSeq: 20 INVITE
>>>    Call-ID: 5gQ3JhmBhp
>>>    Max-Forwards: 70
>>>    Supported: replaces, outbound
>>>    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
>>> SUBSCRIBE, INFO, UPDATE
>>>    Content-Type: application/sdp
>>>    Content-Length: 799
>>>    Contact: <sip:1009 at 213.143.51.43:17334;transport=udp>;+sip.instance="
>>> <urn:uuid:a0671e9c-6a12-4d32-88d9-43f873b85ef6>"
>>>    User-Agent: LinphoneAndroid/3.2.7 (belle-sip/1.6.1)
>>>
>>>    v=0
>>>    o=1009 1906 519 IN IP4 10.78.108.57
>>>    s=Talk
>>>    c=IN IP4 10.78.108.57
>>>    b=AS:512
>>>    t=0 0
>>>    a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL
>>> voip-metrics
>>>    m=audio 7076 RTP/AVP 96 97 98 99 0 8 18 101 100 102
>>>    a=rtpmap:96 opus/48000/2
>>>    a=fmtp:96 useinbandfec=1
>>>    a=rtpmap:97 SILK/16000
>>>    a=rtpmap:98 speex/16000
>>>    a=fmtp:98 vbr=on
>>>    a=rtpmap:99 speex/8000
>>>    a=fmtp:99 vbr=on
>>>    a=fmtp:18 annexb=yes
>>>    a=rtpmap:101 telephone-event/48000
>>>    a=rtpmap:100 telephone-event/16000
>>>    a=rtpmap:102 telephone-event/8000
>>>    a=rtcp-fb:* ccm tmmbr
>>>    m=video 9078 RTP/AVP 96 97
>>>    a=rtpmap:96 VP8/90000
>>>    a=rtpmap:97 H264/90000
>>>    a=fmtp:97 profile-level-id=42801F
>>>    a=rtcp-fb:* ccm tmmbr
>>>    a=rtcp-fb:96 nack pli
>>>    a=rtcp-fb:96 nack sli
>>>    a=rtcp-fb:96 ack rpsi
>>>    a=rtcp-fb:96 ccm fir
>>>    a=rtcp-fb:97 nack pli
>>>    a=rtcp-fb:97 ccm fir
>>>    -----------------------------------------------------------
>>> -------------
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656
>>> [NOTICE] switch_channel.c:1104 New Channel sofia/internal/
>>> 1009 at 192.168.230.143:5060 [367536c0-9e81-463d-9c5c-093cd85b292b]
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
>>> switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) Running State Change CS_NEW (Cur 1 Tot 1)
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
>>> sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite
>>> from 213.143.51.43:17334 version: 1.9.0 -501-7c5d442 64bit
>>> 2017-07-20 09:13:06.047656 [DEBUG] sofia.c:10238 IP 213.143.51.43
>>> Rejected by acl "domains". Falling back to Digest auth.
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:06.047656 [DEBUG]
>>> switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) State NEW
>>> send 830 bytes to udp/[213.143.51.43]:17334 at 09:13:06.061889:
>>>    -----------------------------------------------------------
>>> -------------
>>>    SIP/2.0 407 Proxy Authentication Required
>>>    Via: SIP/2.0/UDP 10.78.108.57:51363;branch=z9hG
>>> 4bK.BChE3dtFP;rport=17334;received=213.143.51.43
>>>    From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
>>>    To: <sip:1010 at 192.168.230.143:5060>;tag=8Uc3QUtpgUjSm
>>>    Call-ID: 5gQ3JhmBhp
>>>    CSeq: 20 INVITE
>>>    User-Agent: FreeSWITCH-mod_sofia/1.9.0-501-7c5d442~64bit
>>>    Accept: application/sdp
>>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>>    Supported: timer, path, replaces
>>>    Allow-Events: talk, hold, conference, presence, as-feature-event,
>>> dialog, line-seize, call-info, sla, include-session-description,
>>> presence.winfo, message-summary, refer
>>>    Proxy-Authenticate: Digest realm="192.168.230.143",
>>> nonce="1a8a328f-a982-4886-adae-1c5554c8d017", algorithm=MD5, qop="auth"
>>>    Content-Length: 0
>>>
>>>    -----------------------------------------------------------
>>> -------------
>>> 2017-07-20 09:13:06.047656 [DEBUG] sofia.c:2405 detaching session
>>> 367536c0-9e81-463d-9c5c-093cd85b292b
>>> recv 389 bytes from udp/[213.143.51.43]:17334 at 09:13:06.184956:
>>>    -----------------------------------------------------------
>>> -------------
>>>    ACK sip:1010 at 88.XX.YY.ZZ:5080 SIP/2.0
>>>    Via: SIP/2.0/UDP 10.78.108.57:51363;branch=z9hG4bK.BChE3dtFP;rport
>>>    Call-ID: 5gQ3JhmBhp
>>>    From: <sip:1009 at 192.168.230.143:5060>;tag=RV6JBclym
>>>    To: <sip:1010 at 192.168.230.143:5060>;tag=8Uc3QUtpgUjSm
>>>    Contact: <sip:1009 at 213.143.51.43:17334;transport=udp>;+sip.instance="
>>> <urn:uuid:a0671e9c-6a12-4d32-88d9-43f873b85ef6>"
>>>    Max-Forwards: 70
>>>    CSeq: 20 ACK
>>>
>>>    -----------------------------------------------------------
>>> -------------
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657
>>> [WARNING] switch_core_state_machine.c:687 367536c0-9e81-463d-9c5c-093cd85b292b
>>> sofia/internal/1009 at 192.168.230.143:5060 Abandoned
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657
>>> [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/
>>> 1009 at 192.168.230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
>>> switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) Running State Change CS_HANGUP (Cur 1 Tot 1)
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
>>> switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) Callstate Change DOWN -> HANGUP
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
>>> switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) State HANGUP
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.087657 [DEBUG]
>>> mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060
>>> hanging up, cause: WRONG_CALL_STATE
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060
>>> Standard HANGUP, cause: WRONG_CALL_STATE
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) State HANGUP going to sleep
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) State Change CS_HANGUP -> CS_REPORTING
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) Running State Change CS_REPORTING (Cur 1 Tot 1)
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) State REPORTING
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060
>>> Standard REPORTING, cause: WRONG_CALL_STATE
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) State REPORTING going to sleep
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) State Change CS_REPORTING -> CS_DESTROY
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_session.c:1713 Session 1 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) Locked, Waiting on external entities
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626
>>> [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/
>>> 1009 at 192.168.230.143:5060) Ended
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626
>>> [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/
>>> 1009 at 192.168.230.143:5060 [CS_DESTROY]
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) Running State Change CS_DESTROY (Cur 0 Tot 1)
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) State DESTROY
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060
>>> Standard DESTROY
>>> 367536c0-9e81-463d-9c5c-093cd85b292b 2017-07-20 09:13:16.107626 [DEBUG]
>>> switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.2
>>> 30.143:5060) State DESTROY going to sleep
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> José David Jurado Alonso <josedavid at zennio.com>
>>>
>>> *Área de Desarrollo de Software de Alto Nivel*
>>>
>>> *Dpto. Ingeniería*
>>>
>>> [image: Descripción: cid:image010.jpg at 01D27631.DF7E30F0]
>>>
>>> Zennio Avance y Tecnología, S.L.
>>>
>>> C/ Rio Jarama,132. Nave P-8.11
>>>
>>> 45007 - Toledo (Spain)
>>>
>>> T: +34 925 232 002 <+34%20925%2023%2020%2002>
>>>
>>> www.zennio.com
>>>
>>> [image: Descripción: Descripción: C:\Users\jjmanjarres\Desktop\Z
>>> zennio.jpg] <http://www.zennio.com/> [image: Descripción: Descripción:
>>> C:\Users\jjmanjarres\Desktop\twitter.jpg]
>>> <http://www.twitter.com/Zennio_en> [image: Descripción: Descripción:
>>> C:\Users\jjmanjarres\Desktop\descarga.jpg]
>>> <http://www.facebook.com/ZennioInternational> [image: Descripción:
>>> Descripción: C:\Users\jjmanjarres\Desktop\linkedin.png]
>>> <http://www.linkedin.com/company/zennio> [image: Descripción:
>>> Descripción: C:\Users\jjmanjarres\Desktop\youtube.jpg]
>>> <https://www.youtube.com/user/ZennioEnglish>
>>>
>>> Zennio Avance y Tecnología S.L le informa de los siguientes extremos:
>>> Los datos por usted suministrados pasarán a formar parte de un fichero
>>> cuyo responsable es Zennio Avance y Tecnología S.L. Dicho fichero se
>>> encuentra legalmente inscrito en el Registro General de Protección de Datos
>>> de la Agencia Española de Protección de Datos. Los datos por usted
>>> suministrados serán empleados con fines de gestión, Zennio Avance y
>>> Tecnología S.L ha adoptado las medidas de seguridad exigidas en función del
>>> nivel de los datos suministrados, instalando las medidas técnicas y
>>> organizativas necesarias, habida cuenta del estado de la tecnología, a fin
>>> de evitar su pérdida, alteración, uso inadecuado o accesos no autorizados a
>>> los mismos.
>>> Para el ejercicio de sus derechos de acceso, rectificación, cancelación
>>> y oposición deberá dirigirse a la dirección del Responsable de Fichero
>>> Zennio Avance y Tecnología S.L C/ Rio Jarama, 132. Nave P-8.11, 45007,
>>> TOLEDO o a la dirección de correo electrónico: info at zennio.com
>>>
>>> Please, consider the environment before printing this e-mail... Save
>>> energy!
>>>
>>> Por favor, piensa en el medio ambiente antes de imprimir este e-mail... ¡Ahorra
>>> energía!
>>>
>>> Disclaimer:
>>> This message and any attached files transmitted with it, is
>>> confidential, especially as regards personal data. It is intended solely
>>> for the use of the individual or entity to whom it is addressed. If you are
>>> not the intended recipient and have received this information in error or
>>> have accessed it for any reason, please notify us of this fact by email
>>> reply and then destroy or delete the message, refraining from any
>>> reproduction, use, alteration, filing or communication to third parties of
>>> this message and attached files on penalty of incurring legal
>>> responsibilities. The opinions contained in this message and the attached
>>> archives, belong exclusively to their sender and they do not represent the
>>> opinion of the company unless it is said specifically and the sender is
>>> authorized for it. The sender does not guarantee the integrity, the
>>> accuracy, the swift delivery or the security of this email transmission,
>>> and assumes no responsibility for any possible damage incurred through data
>>> capture, virus incorporation or any manipulation carried out by third
>>> parties.
>>>
>>> Advertencia legal:
>>> Este mensaje y, en su caso, los ficheros anexos son confidenciales,
>>> especialmente en lo que respecta a los datos personales, y se dirigen
>>> exclusivamente al destinatario referenciado. Si usted no lo es y lo ha
>>> recibido por error o tiene conocimiento del mismo por cualquier motivo, le
>>> rogamos que nos lo comunique por este medio y proceda a destruirlo o
>>> borrarlo, y que en todo caso se abstenga de utilizar, reproducir, alterar,
>>> archivar o comunicar a terceros el presente mensaje y ficheros anexos, todo
>>> ello bajo pena de incurrir en responsabilidades legales. Las opiniones
>>> contenidas en este mensaje y en los archivos adjuntos, pertenecen
>>> exclusivamente a su remitente y no representan la opinión de la empresa
>>> salvo que se diga expresamente y el remitente esté autorizado para ello. El
>>> emisor no garantiza la integridad, rapidez o seguridad del presente correo,
>>> ni se responsabiliza de posibles perjuicios derivados de la captura,
>>> incorporaciones de virus o cualesquiera otras manipulaciones efectuadas por
>>> terceros.
>>>
>>> 2017-07-19 21:00 GMT+02:00 Brian West <brian at freeswitch.org>:
>>>
>>>> 1. set up your local-network-acl . (rfc1918.auto is good start)
>>>> 2. setup your ext-rtp-ip and ext-sip-ip and MAKE SURE you prefix it
>>>> with autonat: so it activates the proper behavior for you.
>>>>
>>>> This should be all you have to do so you can talk to devices inside and
>>>> outside the NAT at the same time.
>>>>
>>>> /b
>>>>
>>>>
>>>> On Wed, Jul 19, 2017 at 9:30 AM, David Villasmil <
>>>> david.villasmil.work at gmail.com> wrote:
>>>>
>>>>> +1 about forwarding upds...
>>>>> But looking at the logs, we can't really see whether the call was
>>>>> processed or not, just that it was "abandoned" and that usually happens
>>>>> when freeswitch responds with a 401/407 and the client never
>>>>> authenticates...
>>>>>
>>>>> enter:
>>>>> sofia profile internal siptrace on
>>>>>
>>>>> to see all messages and make sure the client responds properly and
>>>>> paste the signaling here.
>>>>> ᐧ
>>>>>
>>>>> Regards,
>>>>>
>>>>> David Villasmil
>>>>> email: david.villasmil.work at gmail.com
>>>>> phone: +34669448337 <+34%20669%2044%2083%2037>
>>>>>
>>>>> On Wed, Jul 19, 2017 at 4:18 PM, Joel Serrano <joel at gogii.net> wrote:
>>>>>
>>>>>> Double check your ACLs, yes. Also, I'm pretty sure you will also have
>>>>>> to forward the RTP port range so you can have audio... Otherwise signaling
>>>>>> will work, but your next problem will be one-way-audio.
>>>>>>
>>>>>>
>>>>>> On Tue, Jul 18, 2017 at 11:43 PM, Jose David Jurado Alonso <
>>>>>> josedavid at zennio.com> wrote:
>>>>>>
>>>>>>>
>>>>>>> I've installed *freeswitch into a local environment* and I perform
>>>>>>> SIP calls well using the server local IP.
>>>>>>>
>>>>>>> This server has connected to ADSL network with a *static public IP*
>>>>>>> but when I try to call using the public IP instead of the local IP the
>>>>>>> server receive the call not work...
>>>>>>>
>>>>>>> The clients can register well (or I think so). Others process like
>>>>>>> apache work well.
>>>>>>>
>>>>>>> I NAT forwarding in the firewall the next ports: 5060, 5080, 7443.
>>>>>>>
>>>>>>> *$ netstat -putan*
>>>>>>>
>>>>>>> Active Internet connections (servers and established)
>>>>>>> Proto Recv-Q Send-Q Local Address           Foreign Address         State       PID/Program name
>>>>>>> tcp        0      0 192.168.230.143:2855    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>>> tcp        0      0 192.168.230.143:2856    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>>> tcp        0      0 192.168.230.143:5066    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>>> tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN      1166/nginx -g daemo
>>>>>>> tcp        0      0 192.168.230.143:7443    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>>> tcp        0      0 0.0.0.0:22              0.0.0.0:*               LISTEN      1079/sshd
>>>>>>> tcp        0      0 192.168.230.143:5080    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>>> tcp        0      0 0.0.0.0:443             0.0.0.0:*               LISTEN      1166/nginx -g daemo
>>>>>>> tcp        0      0 192.168.230.143:5060    0.0.0.0:*               LISTEN      15894/freeswitch
>>>>>>> tcp        0      0 192.168.230.143:22      192.168.230.167:60298   ESTABLISHED 14816/sshd: ubuntu
>>>>>>> tcp        0      0 192.168.230.143:22      192.168.230.167:60296   ESTABLISHED 14744/sshd: ubuntu
>>>>>>> tcp6       0      0 :::80                   :::*                    LISTEN      1166/nginx -g daemo
>>>>>>> tcp6       0      0 :::8021                 :::*                    LISTEN      15894/freeswitch
>>>>>>> tcp6       0      0 :::22                   :::*                    LISTEN      1079/sshd
>>>>>>> tcp6       0      0 ::1:5080                :::*                    LISTEN      15894/freeswitch
>>>>>>> tcp6       0      0 :::443                  :::*                    LISTEN      1166/nginx -g daemo
>>>>>>> tcp6       0      0 ::1:5060                :::*                    LISTEN      15894/freeswitch
>>>>>>> udp        0      0 192.168.230.143:5060    0.0.0.0:*                           15894/freeswitch
>>>>>>> udp        0      0 192.168.230.143:5080    0.0.0.0:*                           15894/freeswitch
>>>>>>> udp        0      0 0.0.0.0:68              0.0.0.0:*                           1019/dhclient
>>>>>>> udp6       0      0 ::1:5060                :::*                                15894/freeswitch
>>>>>>> udp6       0      0 ::1:5080                :::*                                15894/freeswitch
>>>>>>>
>>>>>>> *$ fs_cli -x "sofia status"*
>>>>>>>
>>>>>>>                      Name      Type                                       Data  State
>>>>>>> =================================================================================================
>>>>>>>             external-ipv6   profile                   sip:mod_sofia@[::1]:5080  RUNNING (0)
>>>>>>>           192.168.230.143     alias                                   internal  ALIASED
>>>>>>>                  external   profile         sip:mod_sofia at 192.168.230.143:5080  RUNNING (0)
>>>>>>>     external::example.com   gateway                    sip:joeuser at example.com  NOREG
>>>>>>>             internal-ipv6   profile                   sip:mod_sofia@[::1]:5060  RUNNING (0)
>>>>>>>                  internal   profile         sip:mod_sofia at 192.168.230.143:5060  RUNNING (0)
>>>>>>> =================================================================================================
>>>>>>>
>>>>>>> *$ fs_cli -x "sofia status profile external"*
>>>>>>>
>>>>>>> =================================================================================================
>>>>>>> Name                external
>>>>>>> Domain Name         N/A
>>>>>>> Auto-NAT            false
>>>>>>> DBName              sofia_reg_external
>>>>>>> Pres Hosts
>>>>>>> Dialplan            XML
>>>>>>> Context             public
>>>>>>> Challenge Realm     auto_to
>>>>>>> RTP-IP              192.168.230.143
>>>>>>> SIP-IP              192.168.230.143
>>>>>>> URL                 sip:mod_sofia at 192.168.230.143:5080
>>>>>>> BIND-URL            sip:mod_sofia at 192.168.230.143:5080;transport=udp,tcp
>>>>>>> HOLD-MUSIC          local_stream://moh
>>>>>>> OUTBOUND-PROXY      N/A
>>>>>>> CODECS IN           OPUS,G722,PCMU,PCMA,VP8,H264
>>>>>>> CODECS OUT          OPUS,G722,PCMU,PCMA,VP8,H264
>>>>>>> TEL-EVENT           101
>>>>>>> DTMF-MODE           rfc2833
>>>>>>> CNG                 13
>>>>>>> SESSION-TO          0
>>>>>>> MAX-DIALOG          0
>>>>>>> NOMEDIA             false
>>>>>>> LATE-NEG            true
>>>>>>> PROXY-MEDIA         false
>>>>>>> ZRTP-PASSTHRU       true
>>>>>>> AGGRESSIVENAT       false
>>>>>>> CALLS-IN            0
>>>>>>> FAILED-CALLS-IN     0
>>>>>>> CALLS-OUT           0
>>>>>>> FAILED-CALLS-OUT    0
>>>>>>> REGISTRATIONS       0
>>>>>>>
>>>>>>> *$ grep -r '"ext-' **
>>>>>>>
>>>>>>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>>>>> sip_profiles/external.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>>>>> sip_profiles/internal.xml:    <param name="ext-rtp-ip" value="auto-nat"/>
>>>>>>> sip_profiles/internal.xml:    <param name="ext-sip-ip" value="auto-nat"/>
>>>>>>>
>>>>>>> When I perform a call the freeswitch.log file show:
>>>>>>>
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1009 at 192.168.230.143:5060 [df895d9a-30f2-40b5-97bb-56e222c1c45f]
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_NEW (Cur 1 Tot 7)
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10067 sofia/internal/1009 at 192.168.230.143:5060 receiving invite from 88.33.111.22:57076 version: 1.9.0 -493-13f2f2a 64bit
>>>>>>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:10238 IP 88.33.111.22 Rejected by acl "domains". Falling back to Digest auth.
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:44:51.218939 [DEBUG] switch_core_state_machine.c:603 (sofia/internal/1009 at 192.168.230.143:5060) State NEW
>>>>>>> 2017-07-18 14:44:51.218939 [DEBUG] sofia.c:2405 detaching session df895d9a-30f2-40b5-97bb-56e222c1c45f
>>>>>>>
>>>>>>> and passed some seconds:
>>>>>>>
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [WARNING] switch_core_state_machine.c:687 df895d9a-30f2-40b5-97bb-56e222c1c45f sofia/internal/1009 at 192.168.230.143:5060 Abandoned
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_state_machine.c:690 Hangup sofia/internal/1009 at 192.168.230.143:5060 [CS_NEW] [WRONG_CALL_STATE]
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_HANGUP (Cur 1 Tot 7)
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/1009 at 192.168.230.143:5060) Callstate Change DOWN -> HANGUP
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:449 Channel sofia/internal/1009 at 192.168.230.143:5060 hanging up, cause: WRONG_CALL_STATE
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1009 at 192.168.230.143:5060 Standard HANGUP, cause: WRONG_CALL_STATE
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:852 (sofia/internal/1009 at 192.168.230.143:5060) State HANGUP going to sleep
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:619 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_HANGUP -> CS_REPORTING
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_REPORTING (Cur 1 Tot 7)
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:174 sofia/internal/1009 at 192.168.230.143:5060 Standard REPORTING, cause: WRONG_CALL_STATE
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:938 (sofia/internal/1009 at 192.168.230.143:5060) State REPORTING going to sleep
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:610 (sofia/internal/1009 at 192.168.230.143:5060) State Change CS_REPORTING -> CS_DESTROY
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_session.c:1713 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Locked, Waiting on external entities
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1731 Session 7 (sofia/internal/1009 at 192.168.230.143:5060) Ended
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1009 at 192.168.230.143:5060 [CS_DESTROY]
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1009 at 192.168.230.143:5060) Running State Change CS_DESTROY (Cur 0 Tot 7)
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] mod_sofia.c:354 sofia/internal/1009 at 192.168.230.143:5060 SOFIA DESTROY
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:181 sofia/internal/1009 at 192.168.230.143:5060 Standard DESTROY
>>>>>>> df895d9a-30f2-40b5-97bb-56e222c1c45f 2017-07-18 14:45:01.258940 [DEBUG] switch_core_state_machine.c:751 (sofia/internal/1009 at 192.168.230.143:5060) State DESTROY going to sleep
>>>>>>>
>>>>>>> Can it be an ACL or sip_profile bad configuration?? Router firewall
>>>>>>> configuration maybe?
>>>>>>>
>>>>>>> ____________________________________________________________
>>>>>>> _____________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>> switch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> ____________________________________________________________
>>>>>> _____________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>> switch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>>>
>>>> Allison prompts for FreeSWITCH:
>>>>
>>>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>>>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>>>
>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>
>>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>>> *Skype:*briankwest
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>>
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>
>> Allison prompts for FreeSWITCH:
>>
>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
>> | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From furi at vmtele.com  Mon Jul 24 08:31:27 2017
From: furi at vmtele.com (=?UTF-8?B?SsOhbiBGw7xyaQ==?=)
Date: Mon, 24 Jul 2017 10:31:27 +0200
Subject: [Freeswitch-users] g729 licenses
Message-ID: <b67e65de-2e62-59ce-52e4-1df8d42b4a26@vmtele.com>

Hi guys,
can somebody explain me how it is with g729 codecs now.
As far as I know the patent terms of most Licensed Patents under the 
G.729 Consortium have expired on January 2017.
Is that mean that I could use G729 for free on my freeswitches ?

Currently we buy g729 licenses from 
https://freeswitch.com/cart.php?gid=2 but it would be really cool to use 
g729 for free :)

Thx
Jan



From emb at get-voice.com  Mon Jul 24 08:46:15 2017
From: emb at get-voice.com (Ejal Breeman)
Date: Mon, 24 Jul 2017 08:46:15 +0000
Subject: [Freeswitch-users] g729 licenses
In-Reply-To: <b67e65de-2e62-59ce-52e4-1df8d42b4a26@vmtele.com>
References: <b67e65de-2e62-59ce-52e4-1df8d42b4a26@vmtele.com>
Message-ID: <CAGEyh_069hhfHbvakFjNoVW1i0ahT5NfRkduyYAZKfr9TeH31A@mail.gmail.com>

Hi Jan,

The patent has expired so it is free now. You can use the open source
version. You should be able to get it from here http://asterisk.hosting.lv/

Regards

Ejal

On Mon, 24 Jul 2017 at 11:35, Ján Füri <furi at vmtele.com> wrote:

> Hi guys,
> can somebody explain me how it is with g729 codecs now.
> As far as I know the patent terms of most Licensed Patents under the
> G.729 Consortium have expired on January 2017.
> Is that mean that I could use G729 for free on my freeswitches ?
>
> Currently we buy g729 licenses from
> https://freeswitch.com/cart.php?gid=2 but it would be really cool to use
> g729 for free :)
>
> Thx
> Jan
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From brians at iptel.co  Mon Jul 24 08:47:38 2017
From: brians at iptel.co (Brian :)
Date: Mon, 24 Jul 2017 09:47:38 +0100
Subject: [Freeswitch-users] RFC2543
In-Reply-To: <59755BF7.7080203@andyit.com.au>
References: <59755BF7.7080203@andyit.com.au>
Message-ID: <CAGPQfi9i6mWxtOYUKbhZ2NMzpNm82bT7rddmqst1Vys-Q02fSw@mail.gmail.com>

Hi Andy,

 the UA that is placing the call on hold sends a (Re)Invite with the
IP address for RTP in the SDP as 0.0.0.0 - its ancient method for
placing calls on hold from March 1999

Freeswitch still supports it but is making fun of the UA with that log message

Brian



On Mon, Jul 24, 2017 at 3:31 AM, Andy Farkas <andyf at andyit.com.au> wrote:
>
> Can anybody tell me what this log message means?
>
> 2017-07-24 12:10:59.006338 [WARNING] switch_core_media.c:3896 RFC2543 from
> March 1999 called; They want their 0.0.0.0 hold method back.....
>
> FreeSWITCH Version 1.6.8~64-bit ( 64bit)
>
> running on FreeBSD 11.0/STABLE r315121
>
> -andyf
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gb at cm.nl  Mon Jul 24 09:35:46 2017
From: gb at cm.nl (Grant Bagdasarian)
Date: Mon, 24 Jul 2017 09:35:46 +0000
Subject: [Freeswitch-users] Install mod_distributor Freeswitch for Windows
Message-ID: <35f6ee9fa0f04b7eb0fb7523c2ed4d5c@cm.nl>

Hi,

I just downloaded and installed Freeswitch 1.6.18 for Windows, but the mod_distributor.dll is missing.

How do I install the mod_distributor for Windows? I couldn't find an option during the installation wizard.

Regards,

Grant
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From udy786 at gmail.com  Mon Jul 24 09:44:03 2017
From: udy786 at gmail.com (Uday kumar)
Date: Mon, 24 Jul 2017 15:14:03 +0530
Subject: [Freeswitch-users] Integration of Google Speech API V2
In-Reply-To: <CAM8wsqLToVcA9=a6wC3fOsU=VmsLV-R+YqQbPH6GxWL1wRzCfQ@mail.gmail.com>
References: <CAM8wsqJ+Voj65OBhWfDjjzn3c9vhcU0MaN32eDE_dW5-vh50Dg@mail.gmail.com>
 <CAOfqg5b3q4TkKUHU7o+90nfXXvhhaWJX_yX1+qZLz1H13Znijw@mail.gmail.com>
 <CAM8wsqLToVcA9=a6wC3fOsU=VmsLV-R+YqQbPH6GxWL1wRzCfQ@mail.gmail.com>
Message-ID: <CAOfqg5Zf4D7NQbd7detJNWPp7KXuvK7tFrW91iVwHOYyPmLB+Q@mail.gmail.com>

Hi,

Please find attached file. You need to replace SMTP details and also key.
One line need to add in vociemail.tpl. Add *Path: ${voicemail_file_path} *below
subject on line 5th.

[image: Inline image 1]

 My code do that copy voicemail file from vociemail folder to web folder.
You need to change or create that location or you can change as per your
requirement.

Please let me know if you anything from my end.


Thanks
Uday.

On Sat, Jul 22, 2017 at 6:00 AM, Rahul MathuR <rahul.ultimate at gmail.com>
wrote:

> Thanks Uday for sharing this info.
>
> Could you please share the working config set. Please replace your key
> with "UDAY_KEY".
> I will be very grateful.
>
>
> Thanks.
>
>
> On Jul 21, 2017 12:38 PM, "Uday kumar" <udy786 at gmail.com> wrote:
>
> I have implemented Google Speech on Freeswitch 1.6.16 Transcribing
> Voicemail and its working really nice without any issue. When user get
> voicemail then server send email to user with recording and text of
> recording.
>
> I am using SMTP to send mail. In switch.con.xml
>
> <param name="mailer-app" value="/usr/bin/php /var/www/html/PHPMailer/mailer
> _app.php"/>
>
> Download mailer_app.php from freeswitch wiki and you can implement google
> speech API and code in that file.
>
> Thanks
> Uday.
>
> On Wed, Jul 19, 2017 at 2:07 PM, Rahul MathuR <rahul.ultimate at gmail.com>
> wrote:
>
>> Hi,
>>
>> I'm trying to integrate Google cloud speech recognition v2 in it. I can
>> get the audio recorded, have created Service key and API key but whenever I
>> try to access it, I just get 403 access denied. I am at my wits end here.
>>
>> Has anybody tried it ? were you successful ? Could you please guide me
>> how to do it ?
>> I'll be grateful to you if this works !
>>
>>
>>
>> --
>> Warm Regds.
>> MathuRahul
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Thanks & Regard
> Uday
> Site:- www.shareyourknowledge.in
> Mobile:- +91-9377579349 <+91%2093775%2079349>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regard
Uday
Site:- www.shareyourknowledge.in
Mobile:- +91-9377579349
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From asilva at wirelessmundi.com  Mon Jul 24 09:49:27 2017
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Mon, 24 Jul 2017 11:49:27 +0200
Subject: [Freeswitch-users] g729 licenses
In-Reply-To: <b67e65de-2e62-59ce-52e4-1df8d42b4a26@vmtele.com>
References: <b67e65de-2e62-59ce-52e4-1df8d42b4a26@vmtele.com>
Message-ID: <eca1e2c1-38c7-6bfc-9293-df1fdf8f149a@wirelessmundi.com>

Hi,

This was already discuses here in March, check:

http://lists.freeswitch.org/pipermail/freeswitch-users/2017-March/125296.html

For my opinion, buying licenses to FS is a way to contribute to FS 
project :)


Saludos / Regards / Cumprimentos,
António silva

On 07/24/2017 10:31 AM, Ján Füri wrote:
> Hi guys,
> can somebody explain me how it is with g729 codecs now.
> As far as I know the patent terms of most Licensed Patents under the 
> G.729 Consortium have expired on January 2017.
> Is that mean that I could use G729 for free on my freeswitches ?
>
> Currently we buy g729 licenses from 
> https://freeswitch.com/cart.php?gid=2 but it would be really cool to 
> use g729 for free :)
>
> Thx
> Jan
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From udy786 at gmail.com  Mon Jul 24 09:49:56 2017
From: udy786 at gmail.com (Uday kumar)
Date: Mon, 24 Jul 2017 15:19:56 +0530
Subject: [Freeswitch-users] g729 licenses
In-Reply-To: <CAGEyh_069hhfHbvakFjNoVW1i0ahT5NfRkduyYAZKfr9TeH31A@mail.gmail.com>
References: <b67e65de-2e62-59ce-52e4-1df8d42b4a26@vmtele.com>
 <CAGEyh_069hhfHbvakFjNoVW1i0ahT5NfRkduyYAZKfr9TeH31A@mail.gmail.com>
Message-ID: <CAOfqg5abHskZ+ttLAGuecenh-oPAWbayJJva=TBAtH=ROsqN_A@mail.gmail.com>

Given link is only for Asterisk. Its free for Asterisk. Is now free for
Freeswitch also?

On Mon, Jul 24, 2017 at 2:16 PM, Ejal Breeman <emb at get-voice.com> wrote:

> Hi Jan,
>
> The patent has expired so it is free now. You can use the open source
> version. You should be able to get it from here http://asterisk.hosting.
> lv/
>
> Regards
>
> Ejal
>
> On Mon, 24 Jul 2017 at 11:35, Ján Füri <furi at vmtele.com> wrote:
>
>> Hi guys,
>> can somebody explain me how it is with g729 codecs now.
>> As far as I know the patent terms of most Licensed Patents under the
>> G.729 Consortium have expired on January 2017.
>> Is that mean that I could use G729 for free on my freeswitches ?
>>
>> Currently we buy g729 licenses from
>> https://freeswitch.com/cart.php?gid=2 but it would be really cool to use
>> g729 for free :)
>>
>> Thx
>> Jan
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Thanks & Regard
Uday
Site:- www.shareyourknowledge.in
Mobile:- +91-9377579349
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From infos at madovsky.org  Mon Jul 24 10:07:36 2017
From: infos at madovsky.org (Madovsky)
Date: Mon, 24 Jul 2017 03:07:36 -0700
Subject: [Freeswitch-users] Integration of Google Speech API V2
In-Reply-To: <CAOfqg5Zf4D7NQbd7detJNWPp7KXuvK7tFrW91iVwHOYyPmLB+Q@mail.gmail.com>
References: <CAM8wsqJ+Voj65OBhWfDjjzn3c9vhcU0MaN32eDE_dW5-vh50Dg@mail.gmail.com>
 <CAOfqg5b3q4TkKUHU7o+90nfXXvhhaWJX_yX1+qZLz1H13Znijw@mail.gmail.com>
 <CAM8wsqLToVcA9=a6wC3fOsU=VmsLV-R+YqQbPH6GxWL1wRzCfQ@mail.gmail.com>
 <CAOfqg5Zf4D7NQbd7detJNWPp7KXuvK7tFrW91iVwHOYyPmLB+Q@mail.gmail.com>
Message-ID: <fb0765ed-492b-c10b-27fb-9224449b53de@madovsky.org>

The only problem to link your freeswitch with google (or any external 
service you don't control)

is at anytime can stop the service or block your server

and your voicemail won't be delivered.

I already had such kind of (not real cool) experience and I preferred to

not use any external service in all my project but some inevitable ones.


On 7/24/2017 2:44 AM, Uday kumar wrote:
> Hi,
>
> Please find attached file. You need to replace SMTP details and also 
> key. One line need to add in vociemail.tpl. Add *Path: 
> ${voicemail_file_path} *below subject on line 5th.
>
> Inline image 1
> *
> *
>  My code do that copy voicemail file from vociemail folder to web 
> folder. You need to change or create that location or you can change 
> as per your requirement.
>
> Please let me know if you anything from my end.
>
>
> Thanks
> Uday.
>
> On Sat, Jul 22, 2017 at 6:00 AM, Rahul MathuR 
> <rahul.ultimate at gmail.com <mailto:rahul.ultimate at gmail.com>> wrote:
>
>     Thanks Uday for sharing this info.
>
>     Could you please share the working config set. Please replace your
>     key with "UDAY_KEY".
>     I will be very grateful.
>
>
>     Thanks.
>
>
>     On Jul 21, 2017 12:38 PM, "Uday kumar" <udy786 at gmail.com
>     <mailto:udy786 at gmail.com>> wrote:
>
>         I have implemented Google Speech on
>         Freeswitch 1.6.16 Transcribing Voicemail and its working
>         really nice without any issue. When user get voicemail then
>         server send email to user with recording and text of recording.
>
>         I am using SMTP to send mail. In switch.con.xml
>
>         <param name="mailer-app" value="/usr/bin/php
>         /var/www/html/PHPMailer/mailer_app.php"/>
>
>         Download mailer_app.php from freeswitch wiki and you can
>         implement google speech API and code in that file.
>
>         Thanks
>         Uday.
>
>         On Wed, Jul 19, 2017 at 2:07 PM, Rahul MathuR
>         <rahul.ultimate at gmail.com <mailto:rahul.ultimate at gmail.com>>
>         wrote:
>
>             Hi,
>
>             I'm trying to integrate Google cloud speech recognition v2
>             in it. I can get the audio recorded, have created Service
>             key and API key but whenever I try to access it, I just
>             get 403 access denied. I am at my wits end here.
>
>             Has anybody tried it ? were you successful ? Could you
>             please guide me how to do it ?
>             I'll be grateful to you if this works !
>
>
>
>             -- 
>             Warm Regds.
>             MathuRahul
>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>             http://www.freeswitchsolutions.com
>             <http://www.freeswitchsolutions.com>
>
>             Official FreeSWITCH Sites
>             http://www.freeswitch.org
>             http://confluence.freeswitch.org
>             <http://confluence.freeswitch.org>
>             http://www.cluecon.com
>
>             FreeSWITCH-users mailing list
>             FreeSWITCH-users at lists.freeswitch.org
>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>             <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>             <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>             http://www.freeswitch.org
>
>
>
>
>         -- 
>         Thanks & Regard
>         Uday
>         Site:- www.shareyourknowledge.in
>         <http://www.shareyourknowledge.in/>
>         Mobile:- +91-9377579349 <tel:+91%2093775%2079349>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>         <http://www.freeswitchsolutions.com>
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://confluence.freeswitch.org
>         <http://confluence.freeswitch.org>
>         http://www.cluecon.com
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>         http://www.freeswitch.org
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Thanks & Regard
> Uday
> Site:- www.shareyourknowledge.in <http://www.shareyourknowledge.in/>
> Mobile:- +91-9377579349
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From rahul.ultimate at gmail.com  Mon Jul 24 10:08:39 2017
From: rahul.ultimate at gmail.com (Rahul MathuR)
Date: Mon, 24 Jul 2017 15:38:39 +0530
Subject: [Freeswitch-users] Integration of Google Speech API V2
In-Reply-To: <CAOfqg5Zf4D7NQbd7detJNWPp7KXuvK7tFrW91iVwHOYyPmLB+Q@mail.gmail.com>
References: <CAM8wsqJ+Voj65OBhWfDjjzn3c9vhcU0MaN32eDE_dW5-vh50Dg@mail.gmail.com>
 <CAOfqg5b3q4TkKUHU7o+90nfXXvhhaWJX_yX1+qZLz1H13Znijw@mail.gmail.com>
 <CAM8wsqLToVcA9=a6wC3fOsU=VmsLV-R+YqQbPH6GxWL1wRzCfQ@mail.gmail.com>
 <CAOfqg5Zf4D7NQbd7detJNWPp7KXuvK7tFrW91iVwHOYyPmLB+Q@mail.gmail.com>
Message-ID: <CAM8wsqK+82ybzvn6vX2eRC-dfz2Z_k688QOvyvFySUeA18O-qA@mail.gmail.com>

Thanks brother !



On Jul 24, 2017 3:15 PM, "Uday kumar" <udy786 at gmail.com> wrote:

> Hi,
>
> Please find attached file. You need to replace SMTP details and also key.
> One line need to add in vociemail.tpl. Add *Path: ${voicemail_file_path} *below
> subject on line 5th.
>
> [image: Inline image 1]
>
>  My code do that copy voicemail file from vociemail folder to web folder.
> You need to change or create that location or you can change as per your
> requirement.
>
> Please let me know if you anything from my end.
>
>
> Thanks
> Uday.
>
> On Sat, Jul 22, 2017 at 6:00 AM, Rahul MathuR <rahul.ultimate at gmail.com>
> wrote:
>
>> Thanks Uday for sharing this info.
>>
>> Could you please share the working config set. Please replace your key
>> with "UDAY_KEY".
>> I will be very grateful.
>>
>>
>> Thanks.
>>
>>
>> On Jul 21, 2017 12:38 PM, "Uday kumar" <udy786 at gmail.com> wrote:
>>
>> I have implemented Google Speech on Freeswitch 1.6.16 Transcribing
>> Voicemail and its working really nice without any issue. When user get
>> voicemail then server send email to user with recording and text of
>> recording.
>>
>> I am using SMTP to send mail. In switch.con.xml
>>
>> <param name="mailer-app" value="/usr/bin/php
>> /var/www/html/PHPMailer/mailer_app.php"/>
>>
>> Download mailer_app.php from freeswitch wiki and you can implement google
>> speech API and code in that file.
>>
>> Thanks
>> Uday.
>>
>> On Wed, Jul 19, 2017 at 2:07 PM, Rahul MathuR <rahul.ultimate at gmail.com>
>> wrote:
>>
>>> Hi,
>>>
>>> I'm trying to integrate Google cloud speech recognition v2 in it. I can
>>> get the audio recorded, have created Service key and API key but whenever I
>>> try to access it, I just get 403 access denied. I am at my wits end here.
>>>
>>> Has anybody tried it ? were you successful ? Could you please guide me
>>> how to do it ?
>>> I'll be grateful to you if this works !
>>>
>>>
>>>
>>> --
>>> Warm Regds.
>>> MathuRahul
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Thanks & Regard
>> Uday
>> Site:- www.shareyourknowledge.in
>> Mobile:- +91-9377579349 <+91%2093775%2079349>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Thanks & Regard
> Uday
> Site:- www.shareyourknowledge.in
> Mobile:- +91-9377579349 <+91%2093775%2079349>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From hai.bui at htklabs.com  Mon Jul 24 11:55:00 2017
From: hai.bui at htklabs.com (Hai Bui Duc Ha)
Date: Mon, 24 Jul 2017 18:25:00 +0630
Subject: [Freeswitch-users] Freeswitch crash after calling
Message-ID: <CA+Gtc=S-WK4SLBU1bf6f5o3JqecN_fAG_oDLWbuPtMMv4d0Unw@mail.gmail.com>

Dear all,

My Freeswitch crash after ending call.
I don't see any thing on log file, only one log crash on syslog:
*Jul 20 16:15:07 kazoo kernel: [3052190.684540] freeswitch[31078]: segfault
at 6270 ip 00007f7f76a5da55 sp 00007f7f4e003bd0 error 4 in
libfreeswitch.so.1.0.0[7f7f769ee000+419000]*

I also attached file core dump, but I don't understand what it show.
My Freeswitch version: *FreeSWITCH Version
1.6.17+git~20170525T170320Z~efbf7e5473~64bit (git efbf7e5 2017-05-25
17:03:20Z 64bit)*

You have any idea ?

-- 
Regards,
Hai Bui

Hai Bui
VoIP engineer, Cvoice team, HTK-HCM Office
+84-165-618-9876

<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon>
Virus-free.
www.avast.com
<https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
<#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
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From aqsyounas at gmail.com  Mon Jul 24 12:15:12 2017
From: aqsyounas at gmail.com (Aqs Younas)
Date: Mon, 24 Jul 2017 17:15:12 +0500
Subject: [Freeswitch-users] Freeswitch crash after calling
In-Reply-To: <CA+Gtc=S-WK4SLBU1bf6f5o3JqecN_fAG_oDLWbuPtMMv4d0Unw@mail.gmail.com>
References: <CA+Gtc=S-WK4SLBU1bf6f5o3JqecN_fAG_oDLWbuPtMMv4d0Unw@mail.gmail.com>
Message-ID: <CAGVOjVUNUXyuY39gXdryWGENq4TzE-z8kaZgV0ROJFasdUCVgg@mail.gmail.com>

Please file jira with informations you have.

On 24 Jul 2017 4:57 pm, "Hai Bui Duc Ha" <hai.bui at htklabs.com> wrote:

> Dear all,
>
> My Freeswitch crash after ending call.
> I don't see any thing on log file, only one log crash on syslog:
> *Jul 20 16:15:07 kazoo kernel: [3052190.684540] freeswitch[31078]:
> segfault at 6270 ip 00007f7f76a5da55 sp 00007f7f4e003bd0 error 4 in
> libfreeswitch.so.1.0.0[7f7f769ee000+419000]*
>
> I also attached file core dump, but I don't understand what it show.
> My Freeswitch version: *FreeSWITCH Version
> 1.6.17+git~20170525T170320Z~efbf7e5473~64bit (git efbf7e5 2017-05-25
> 17:03:20Z 64bit)*
>
> You have any idea ?
>
> --
> Regards,
> Hai Bui
>
> Hai Bui
> VoIP engineer, Cvoice team, HTK-HCM Office
> +84-165-618-9876 <+84%20165%20618%209876>
>
>
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=icon> Virus-free.
> www.avast.com
> <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campaign=sig-email&utm_content=webmail&utm_term=link>
> <#m_-5422577713978228088_DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Mon Jul 24 14:59:46 2017
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 24 Jul 2017 09:59:46 -0500
Subject: [Freeswitch-users] g729 licenses
In-Reply-To: <eca1e2c1-38c7-6bfc-9293-df1fdf8f149a@wirelessmundi.com>
References: <b67e65de-2e62-59ce-52e4-1df8d42b4a26@vmtele.com>
 <eca1e2c1-38c7-6bfc-9293-df1fdf8f149a@wirelessmundi.com>
Message-ID: <1cd701d3048d$7e32e1c0$7a98a540$@freeswitch.org>

This has been discussed ad nauseum....

http://lists.freeswitch.org/pipermail/freeswitch-users/2016-August/121715.html

-----Original Message-----
From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Antonio Silva
Sent: Monday, July 24, 2017 4:49 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] g729 licenses

Hi,

This was already discuses here in March, check:

http://lists.freeswitch.org/pipermail/freeswitch-users/2017-March/125296.html

For my opinion, buying licenses to FS is a way to contribute to FS project :)


Saludos / Regards / Cumprimentos,
António silva

On 07/24/2017 10:31 AM, Ján Füri wrote:
> Hi guys,
> can somebody explain me how it is with g729 codecs now.
> As far as I know the patent terms of most Licensed Patents under the 
> G.729 Consortium have expired on January 2017.
> Is that mean that I could use G729 for free on my freeswitches ?
>
> Currently we buy g729 licenses from 
> https://freeswitch.com/cart.php?gid=2 but it would be really cool to 
> use g729 for free :)
>
> Thx
> Jan
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

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

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



From brandon at cryy.com  Mon Jul 24 08:12:24 2017
From: brandon at cryy.com (Brandon Armstead)
Date: Mon, 24 Jul 2017 08:12:24 +0000
Subject: [Freeswitch-users] RFC2543
In-Reply-To: <59755BF7.7080203@andyit.com.au>
References: <59755BF7.7080203@andyit.com.au>
Message-ID: <CACr1DbBRyr9iLb8RtFbvjHE=DukEHpx2ikXhbALFA=6pdFCSLg@mail.gmail.com>

I think it means it's an old RFC defined/based method and or suggestion for
handling hold music in RTP SESSIONS by modifying the active RTP session and
setting c=media IP to 0.0.0.0, however there are bigger and better ways now
?

On Mon, Jul 24, 2017 at 1:06 AM Andy Farkas <andyf at andyit.com.au> wrote:

>
> Can anybody tell me what this log message means?
>
> 2017-07-24 12:10:59.006338 [WARNING] switch_core_media.c:3896 RFC2543
> from March 1999 called; They want their 0.0.0.0 hold method back.....
>
> FreeSWITCH Version 1.6.8~64-bit ( 64bit)
>
> running on FreeBSD 11.0/STABLE r315121
>
> -andyf
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Sent from Gmail Mobile
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From krice at tollfreegateway.com  Mon Jul 24 14:48:56 2017
From: krice at tollfreegateway.com (krice at tollfreegateway.com)
Date: Mon, 24 Jul 2017 09:48:56 -0500
Subject: [Freeswitch-users] g729 licenses
In-Reply-To: <b67e65de-2e62-59ce-52e4-1df8d42b4a26@vmtele.com>
References: <b67e65de-2e62-59ce-52e4-1df8d42b4a26@vmtele.com>
Message-ID: <1c7f01d3048b$fa533a40$eef9aec0$@tollfreegateway.com>

The Patents have expired... the Copyrights of the author have not... 

FreeSWITCH is for the most part OpenSource and Free... however, the developers still have to pay for housing, food, etc... The Core team still has expenses like hosting, dev hardware, power etc that need to be paid for...

G729 is one of the small ways to support the project....


-----Original Message-----
From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ján Füri
Sent: Monday, July 24, 2017 3:31 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] g729 licenses

Hi guys,
can somebody explain me how it is with g729 codecs now.
As far as I know the patent terms of most Licensed Patents under the 
G.729 Consortium have expired on January 2017.
Is that mean that I could use G729 for free on my freeswitches ?

Currently we buy g729 licenses from 
https://freeswitch.com/cart.php?gid=2 but it would be really cool to use 
g729 for free :)

Thx
Jan


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

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

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



From jungleboogie0 at gmail.com  Mon Jul 24 15:37:30 2017
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Mon, 24 Jul 2017 08:37:30 -0700
Subject: [Freeswitch-users] RFC2543
In-Reply-To: <CAGPQfi9i6mWxtOYUKbhZ2NMzpNm82bT7rddmqst1Vys-Q02fSw@mail.gmail.com>
References: <59755BF7.7080203@andyit.com.au>
 <CAGPQfi9i6mWxtOYUKbhZ2NMzpNm82bT7rddmqst1Vys-Q02fSw@mail.gmail.com>
Message-ID: <CAKE2PDtYA4=Ov6qr9h0RKrb_c4gq-W6G97Lw-pLaUDZiuywhOQ@mail.gmail.com>

On 24 July 2017 at 01:47, Brian : <brians at iptel.co> wrote:
> Hi Andy,
>
>  the UA that is placing the call on hold sends a (Re)Invite with the
> IP address for RTP in the SDP as 0.0.0.0 - its ancient method for
> placing calls on hold from March 1999
>

I see this with my polycom phones. Is there something I need to
configure within the phone webpage to prevent the 0.0.0.0 IP address?


> Freeswitch still supports it but is making fun of the UA with that log message
>
> Brian
>


From chris at gcjd.org  Mon Jul 24 15:53:54 2017
From: chris at gcjd.org (chris)
Date: Mon, 24 Jul 2017 17:53:54 +0200
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
	updating
Message-ID: <20170724175354895851.a7d89e7f@gcjd.org>

Hello List,

my problem is that the build after updating fails.

This is on a Mac Mini 2012 running OSX 10.11.6, XCode 7.3 and up to 
date brew.

I follow the instructions from 
<https://freeswitch.org/confluence/display/FREESWITCH/macOS+Manual+Installation>. 
The previous installed version is:
FreeSWITCH Version 1.6.17+git~20170421T211727Z~ab1f8eae62~64bit (git 
ab1f8ea 2017-04-21 21:17:27Z 64bit)

mini11:freeswitch chris$ pwd
/usr/local/src/freeswitch
mini11:freeswitch chris$ git pull && make current
...

config.status: libs/xmlrpc-c/xmlrpc_amconfig.h is unchanged
config.status: executing depfiles commands
config.status: executing libtool commands
cd src/mod/languages/mod_lua/lua && make clean
make[1]: *** No rule to make target `clean'.  Stop.
make: *** [lua-reconf] Error 2

mini11:freeswitch chris$ ls -la src/mod/languages/mod_lua/lua
total 0
drwxr-xr-x   3 chris  admin   102 21 Jul 00:14 .
drwxr-xr-x  19 chris  admin   646 21 Jul 00:15 ..
drwxr-xr-x  34 chris  admin  1156 25 Mär 22:18 .deps

Complete output from build is on 
<http://www.gcjd.org/temp0717/freeswitchbuild.txt>

I did an update before some time ago the same way and it worked 
flawless.

How can I proceed without loosing my config?

Thanks for any advice, Chris


From mario_fs at mgtech.com  Mon Jul 24 16:21:36 2017
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 24 Jul 2017 09:21:36 -0700
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <20170724175354895851.a7d89e7f@gcjd.org>
References: <20170724175354895851.a7d89e7f@gcjd.org>
Message-ID: <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>

I recommend:

1. Use macFI <https://freeswitch.org/confluence/display/FREESWITCH/macOS+macFI+Installation> to install 1.6.19, it will rename the old directories. You can do this manually but macFI is way faster.
2. Rename the default “conf” directory in the new “freeswitch” runtime folder.
3. Copy your old customized “conf” to the new “freeswitch” folder.

Start it up, you will be good to go, I just tested for the wiki yesterday, you can then to a make current from then on. Personally I like to just macFI and get the dirs fresh.
Mario G


> On Jul 24, 2017, at 8:53 AM, chris <chris at gcjd.org> wrote:
> 
> Hello List,
> 
> my problem is that the build after updating fails.
> 
> This is on a Mac Mini 2012 running OSX 10.11.6, XCode 7.3 and up to 
> date brew.
> 
> I follow the instructions from 
> <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Manual+Installation>. 
> The previous installed version is:
> FreeSWITCH Version 1.6.17+git~20170421T211727Z~ab1f8eae62~64bit (git 
> ab1f8ea 2017-04-21 21:17:27Z 64bit)
> 
> mini11:freeswitch chris$ pwd
> /usr/local/src/freeswitch
> mini11:freeswitch chris$ git pull && make current
> ...
> 
> config.status: libs/xmlrpc-c/xmlrpc_amconfig.h is unchanged
> config.status: executing depfiles commands
> config.status: executing libtool commands
> cd src/mod/languages/mod_lua/lua && make clean
> make[1]: *** No rule to make target `clean'.  Stop.
> make: *** [lua-reconf] Error 2
> 
> mini11:freeswitch chris$ ls -la src/mod/languages/mod_lua/lua
> total 0
> drwxr-xr-x   3 chris  admin   102 21 Jul 00:14 .
> drwxr-xr-x  19 chris  admin   646 21 Jul 00:15 ..
> drwxr-xr-x  34 chris  admin  1156 25 Mär 22:18 .deps
> 
> Complete output from build is on 
> <http://www.gcjd.org/temp0717/freeswitchbuild.txt>
> 
> I did an update before some time ago the same way and it worked 
> flawless.
> 
> How can I proceed without loosing my config?
> 
> Thanks for any advice, Chris
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Mon Jul 24 17:19:47 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 24 Jul 2017 13:19:47 -0400
Subject: [Freeswitch-users] Freeswitch crash after calling
In-Reply-To: <CA+Gtc=S-WK4SLBU1bf6f5o3JqecN_fAG_oDLWbuPtMMv4d0Unw@mail.gmail.com>
References: <CA+Gtc=S-WK4SLBU1bf6f5o3JqecN_fAG_oDLWbuPtMMv4d0Unw@mail.gmail.com>
Message-ID: <FB7B8B09-AEC9-444B-A868-C0B99EFD5965@jerris.com>

a decent number of crashes were found and fixed between 1.6.17 and the newest release 1.6.19.  I’d recommend moving to the latest release before you do anything else.

> On Jul 24, 2017, at 7:55 AM, Hai Bui Duc Ha <hai.bui at htklabs.com> wrote:
> 
> Dear all, 
> 
> My Freeswitch crash after ending call.
> I don't see any thing on log file, only one log crash on syslog:
> Jul 20 16:15:07 kazoo kernel: [3052190.684540] freeswitch[31078]: segfault at 6270 ip 00007f7f76a5da55 sp 00007f7f4e003bd0 error 4 in libfreeswitch.so.1.0.0[7f7f769ee000+419000]
> 
> I also attached file core dump, but I don't understand what it show.
> My Freeswitch version: FreeSWITCH Version 1.6.17+git~20170525T170320Z~efbf7e5473~64bit (git efbf7e5 2017-05-25 17:03:20Z 64bit)
> 
> You have any idea ?

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From brian at freeswitch.org  Mon Jul 24 20:21:54 2017
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Jul 2017 15:21:54 -0500
Subject: [Freeswitch-users] Freeswitch crash after calling
In-Reply-To: <FB7B8B09-AEC9-444B-A868-C0B99EFD5965@jerris.com>
References: <CA+Gtc=S-WK4SLBU1bf6f5o3JqecN_fAG_oDLWbuPtMMv4d0Unw@mail.gmail.com>
 <FB7B8B09-AEC9-444B-A868-C0B99EFD5965@jerris.com>
Message-ID: <CAEJMVkAW=9Q+vqocc6p5d=AXoHa7tRNtYztJGkKTwAfgh+sOuQ@mail.gmail.com>

I suspect it might be FS-10103, but not 100% sure, its the only JIRA that
rings a bell on crash with new call.

On Mon, Jul 24, 2017 at 12:19 PM, Michael Jerris <mike at jerris.com> wrote:

> a decent number of crashes were found and fixed between 1.6.17 and the
> newest release 1.6.19.  I’d recommend moving to the latest release before
> you do anything else.
>
> On Jul 24, 2017, at 7:55 AM, Hai Bui Duc Ha <hai.bui at htklabs.com> wrote:
>
> Dear all,
>
> My Freeswitch crash after ending call.
> I don't see any thing on log file, only one log crash on syslog:
> *Jul 20 16:15:07 kazoo kernel: [3052190.684540] freeswitch[31078]:
> segfault at 6270 ip 00007f7f76a5da55 sp 00007f7f4e003bd0 error 4 in
> libfreeswitch.so.1.0.0[7f7f769ee000+419000]*
>
> I also attached file core dump, but I don't understand what it show.
> My Freeswitch version: *FreeSWITCH Version
> 1.6.17+git~20170525T170320Z~efbf7e5473~64bit (git efbf7e5 2017-05-25
> 17:03:20Z 64bit)*
>
> You have any idea ?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From hai.bui at htklabs.com  Tue Jul 25 03:02:31 2017
From: hai.bui at htklabs.com (Hai Bui Duc Ha)
Date: Tue, 25 Jul 2017 10:02:31 +0700
Subject: [Freeswitch-users] Freeswitch crash after calling
In-Reply-To: <CAEJMVkAW=9Q+vqocc6p5d=AXoHa7tRNtYztJGkKTwAfgh+sOuQ@mail.gmail.com>
References: <CA+Gtc=S-WK4SLBU1bf6f5o3JqecN_fAG_oDLWbuPtMMv4d0Unw@mail.gmail.com>
 <FB7B8B09-AEC9-444B-A868-C0B99EFD5965@jerris.com>
 <CAEJMVkAW=9Q+vqocc6p5d=AXoHa7tRNtYztJGkKTwAfgh+sOuQ@mail.gmail.com>
Message-ID: <CA+Gtc=TUsD4VMXyWUdz0U7ToFW=9gVXwLnteeVVTLR6hzf5JoA@mail.gmail.com>

Thank for support.
I have already create the ticket on Jira:
https://freeswitch.org/jira/browse/FS-10537


On Tue, Jul 25, 2017 at 3:21 AM, Brian West <brian at freeswitch.org> wrote:

> I suspect it might be FS-10103, but not 100% sure, its the only JIRA that
> rings a bell on crash with new call.
>
> On Mon, Jul 24, 2017 at 12:19 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> a decent number of crashes were found and fixed between 1.6.17 and the
>> newest release 1.6.19.  I’d recommend moving to the latest release before
>> you do anything else.
>>
>> On Jul 24, 2017, at 7:55 AM, Hai Bui Duc Ha <hai.bui at htklabs.com> wrote:
>>
>> Dear all,
>>
>> My Freeswitch crash after ending call.
>> I don't see any thing on log file, only one log crash on syslog:
>> *Jul 20 16:15:07 kazoo kernel: [3052190.684540] freeswitch[31078]:
>> segfault at 6270 ip 00007f7f76a5da55 sp 00007f7f4e003bd0 error 4 in
>> libfreeswitch.so.1.0.0[7f7f769ee000+419000]*
>>
>> I also attached file core dump, but I don't understand what it show.
>> My Freeswitch version: *FreeSWITCH Version
>> 1.6.17+git~20170525T170320Z~efbf7e5473~64bit (git efbf7e5 2017-05-25
>> 17:03:20Z 64bit)*
>>
>> You have any idea ?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Hai Bui

Hai Bui
VoIP engineer, Cvoice team, HTK-HCM Office
+84-165-618-9876
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From emb at get-voice.com  Tue Jul 25 07:02:44 2017
From: emb at get-voice.com (Ejal Breeman)
Date: Tue, 25 Jul 2017 09:02:44 +0200
Subject: [Freeswitch-users] g729 licenses
In-Reply-To: <1c7f01d3048b$fa533a40$eef9aec0$@tollfreegateway.com>
References: <b67e65de-2e62-59ce-52e4-1df8d42b4a26@vmtele.com>
 <1c7f01d3048b$fa533a40$eef9aec0$@tollfreegateway.com>
Message-ID: <CAGEyh_3toioznWAQMu=UWm-p0SWZKbywu9vybPfvAsEy6x8dAw@mail.gmail.com>

I agree on point of view for supporting FreeSWITCH, you have all done an
amazing job guys!

Regards,

Ejal Breeman

Get-Voice

On Mon, Jul 24, 2017 at 4:48 PM, <krice at tollfreegateway.com> wrote:

> The Patents have expired... the Copyrights of the author have not...
>
> FreeSWITCH is for the most part OpenSource and Free... however, the
> developers still have to pay for housing, food, etc... The Core team still
> has expenses like hosting, dev hardware, power etc that need to be paid
> for...
>
> G729 is one of the small ways to support the project....
>
>
> -----Original Message-----
> From: FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] On Behalf Of Ján Füri
> Sent: Monday, July 24, 2017 3:31 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] g729 licenses
>
> Hi guys,
> can somebody explain me how it is with g729 codecs now.
> As far as I know the patent terms of most Licensed Patents under the
> G.729 Consortium have expired on January 2017.
> Is that mean that I could use G729 for free on my freeswitches ?
>
> Currently we buy g729 licenses from
> https://freeswitch.com/cart.php?gid=2 but it would be really cool to use
> g729 for free :)
>
> Thx
> Jan
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From royj at yandex.ru  Tue Jul 25 08:17:29 2017
From: royj at yandex.ru (roy j)
Date: Tue, 25 Jul 2017 11:17:29 +0300
Subject: [Freeswitch-users] automate fax checking
Message-ID: <3243731500970649@web8g.yandex.ru>

Hi everyone
Actually the question is not regarding FreeSWITCH but tools in general.
Does anybody use some cli tool just to check faxing including T.38. It would be sipp scenario with pcap to play, but real fax machine needs to negotiate T.30/T.38 params. FreeSWITCH or Asterisk originate is option, may be there is some other solution some of you use.
Thanks!


From mail at paulzillmann.de  Tue Jul 25 08:23:00 2017
From: mail at paulzillmann.de (Paul Zillmann)
Date: Tue, 25 Jul 2017 10:23:00 +0200
Subject: [Freeswitch-users] RFC2543
In-Reply-To: <59755BF7.7080203@andyit.com.au>
References: <59755BF7.7080203@andyit.com.au>
Message-ID: <2d0368db-f3a3-ea45-55ce-650423201398@paulzillmann.de>

Hello Andy,

currently there are two methods to set a call into the hold status.
One of your clients still uses the older and deprecated method - so
freeswitch throws that funny message.

Am 24.07.2017 um 04:31 schrieb Andy Farkas:
>
> Can anybody tell me what this log message means?
>
> 2017-07-24 12:10:59.006338 [WARNING] switch_core_media.c:3896 RFC2543
> from March 1999 called; They want their 0.0.0.0 hold method back.....
>
> FreeSWITCH Version 1.6.8~64-bit ( 64bit)
>
> running on FreeBSD 11.0/STABLE r315121
>
> -andyf
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gmaruzz at gmail.com  Tue Jul 25 13:05:02 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 25 Jul 2017 15:05:02 +0200
Subject: [Freeswitch-users] SIM7100 LTE modem with USB audio
In-Reply-To: <CACOYK=hn7ZNFmkeTe7-HX=9t70zc8vKyxbdrApU1Nf4MX7y3eQ@mail.gmail.com>
References: <CACOYK=jtpcNSyzveoWKVRssHad-_6CwJtZ8H72d=qnHNS5niVA@mail.gmail.com>
 <CALXCt0rUcK8wck0UHu1u9Tqif6X+6PbNeEimZj3PXrC7t8C=Zw@mail.gmail.com>
 <CACOYK=hn7ZNFmkeTe7-HX=9t70zc8vKyxbdrApU1Nf4MX7y3eQ@mail.gmail.com>
Message-ID: <CALXCt0pgDzc7rYE8zFsSTdr4RQL_wY0ocbYLDX4AdDxsWL72zw@mail.gmail.com>

On 23 July 2017 at 10:17, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> Hi Giovanni, I only had time to configure it for Internet access:
>
> https://txlab.wordpress.com/2017/07/23/simcom-sim7100e-lte-modem/
>
> The benefit is mPCIe interface, so it can be placed inside and enclosure.
> No silly dongles any more :)
>
> I'll tinker with voice some time later. I can also provide you SSH access
> if you wish to work on gsmopen update.
>

Please, let me know if any progress or so.

Will not be so easy to interface audio to it via USB, will need to create a
kernel driver, or use a userspace USB lib and create a user space driver.

Happy hacking!

-giovanni




>
>
>
> On 21 Jul 2017 18:36, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>
>> Hello Stanislav,
>>
>> have you had time to tinker with it?
>>
>> How it goes?
>>
>> Also, why you would prefer this one instead of the already supported ones?
>>
>> -giovanni
>>
>>
>> On 8 July 2017 at 21:28, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>
>>> Simcom has recently released a new 4G/LTE modem, and it has USB audio
>>> support.
>>> You can find sim7100_usb_audio_application_note_v0.01.pdf   with
>>> details at the vendor site, or at techship.com  after registration.
>>>
>>> It transmits 8kHZ, 16-bit PCM audio via a USB UART simulation.
>>>
>>> So, in theory, gsmopen module may be adapted to it (or maybe a new
>>> module is worth starting).
>>>
>>> I ordered a sample, will check it out soon.
>>>
>>> cheers,
>>> stanislav
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From pkelly at gmail.com  Tue Jul 25 15:07:33 2017
From: pkelly at gmail.com (Pete Kelly)
Date: Tue, 25 Jul 2017 17:07:33 +0200
Subject: [Freeswitch-users] Accessing sip_reply_host / sip_reply_port when
	using a redirect_context
Message-ID: <CAEWH9M_5DuAT3yifXC1XpOxfyfbSCGFhwo=CZr9rnuVokqsCog@mail.gmail.com>

Hi

I am use a reply_context during a bridge in order to handle a 302 Redirect
myself within the FS dialplan.

One of the things I need to know is which host and port that the 302 was
received from.

Looking through sofia.c, this information is being set in the form of:

  switch_channel_set_variable(channel, "sip_reply_host", network_ip);
  switch_channel_set_variable_printf(channel, "sip_reply_port", "%d",
network_port);

However the code then goes on to perform a transfer to the new context:

   switch_ivr_session_transfer(a_session, p_contact->m_url->url_user,
sip_redirect_dialplan, sip_redirect_context);

And within the XML dialplan in that context, the variables sip_reply_host,
sip_reply_port (and the contact_uris from the 302) are all missing,.


Is is possible to get this information somehow from within a
redirect_context? Am I missing a trick somewhere?

Thanks

Pete
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From mike at jerris.com  Tue Jul 25 15:11:37 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Jul 2017 11:11:37 -0400
Subject: [Freeswitch-users] automate fax checking
In-Reply-To: <3243731500970649@web8g.yandex.ru>
References: <3243731500970649@web8g.yandex.ru>
Message-ID: <AF9E29F2-6FA7-44A6-BC26-E6F8A952D8FF@jerris.com>

FreeSWITCH is a perfect tool to do this.  You can’t do this with pcap playback as it requires 2 way communication modifies what is sent.

> On Jul 25, 2017, at 4:17 AM, roy j <royj at yandex.ru> wrote:
> 
> Hi everyone
> Actually the question is not regarding FreeSWITCH but tools in general.
> Does anybody use some cli tool just to check faxing including T.38. It would be sipp scenario with pcap to play, but real fax machine needs to negotiate T.30/T.38 params. FreeSWITCH or Asterisk originate is option, may be there is some other solution some of you use.
> Thanks!
> 



From chris at gcjd.org  Tue Jul 25 10:48:45 2017
From: chris at gcjd.org (chris)
Date: Tue, 25 Jul 2017 12:48:45 +0200
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
Message-ID: <20170725124845350260.78b7d7d1@gcjd.org>

On Mon, 24 Jul 2017 09:21:36 -0700, Mario G wrote:
> I recommend:
> 
> 1. Use macFI 
> 
<https://freeswitch.org/confluence/display/FREESWITCH/macOS+macFI+Installation> 
> to install 1.6.19, it will rename the old directories. You can do 
> this manually but macFI is way faster.

I tried this using macFI but the installation fails with a different 
problem. From the log folder I see that the configure step failed, so 
nothing is build afterwards.

...
checking for inflateReset in -lz... no
configure: error: no usable zlib; please install zlib devel package or 
equivalent
Elf:freeswitch chris$ 

<http://www.gcjd.org/temp0717/FreeSWITCHconfigure.txt>

This should not happen? As far as I know zlib is available in OSX 
standard configuration.

Greetings, Chris


From findmeinwland at gmail.com  Tue Jul 25 15:47:30 2017
From: findmeinwland at gmail.com (Artur Mega)
Date: Tue, 25 Jul 2017 20:47:30 +0500
Subject: [Freeswitch-users] XML_radius module doesn't send request to radius
	server
Message-ID: <CAOLEQ7cRjiOG7QcnKBr2cHqSJK=N7ZyhpS=yovDbWJNrOHh0kw@mail.gmail.com>

Good day,
I see in the logs this error. What does it mean? How to handle it?
2017-07-25 20:44:21.129627 [ERR] mod_xml_radius.c:930 Didn't
match: 168.9.82.191 == ^168.9.82.191

-- 
​Regards, Arthur​
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From mike at jerris.com  Tue Jul 25 16:00:28 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Jul 2017 12:00:28 -0400
Subject: [Freeswitch-users] XML_radius module doesn't send request to
 radius server
In-Reply-To: <CAOLEQ7cRjiOG7QcnKBr2cHqSJK=N7ZyhpS=yovDbWJNrOHh0kw@mail.gmail.com>
References: <CAOLEQ7cRjiOG7QcnKBr2cHqSJK=N7ZyhpS=yovDbWJNrOHh0kw@mail.gmail.com>
Message-ID: <817363BD-C17D-4FBC-9323-8FA70A294961@jerris.com>

do you have “anti” set in your config?

> On Jul 25, 2017, at 11:47 AM, Artur Mega <findmeinwland at gmail.com> wrote:
> 
> Good day, 
> I see in the logs this error. What does it mean? How to handle it?
> 2017-07-25 20:44:21.129627 [ERR] mod_xml_radius.c:930 Didn't match: 168.9.82.191 == ^168.9.82.191
> 
> -- 
> ​Regards, Arthur​

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From findmeinwland at gmail.com  Tue Jul 25 16:06:19 2017
From: findmeinwland at gmail.com (Artur Mega)
Date: Tue, 25 Jul 2017 21:06:19 +0500
Subject: [Freeswitch-users] XML_radius module doesn't send request to
 radius server
In-Reply-To: <817363BD-C17D-4FBC-9323-8FA70A294961@jerris.com>
References: <CAOLEQ7cRjiOG7QcnKBr2cHqSJK=N7ZyhpS=yovDbWJNrOHh0kw@mail.gmail.com>
 <817363BD-C17D-4FBC-9323-8FA70A294961@jerris.com>
Message-ID: <CAOLEQ7fkqvPtYTb3QykFHf0UJ+eK5iVZ-5jB=6cfAEskowADig@mail.gmail.com>

Yes, anti="true". When I set it to "false", same error raises

2017-07-25 21:00 GMT+05:00 Michael Jerris <mike at jerris.com>:

> do you have “anti” set in your config?
>
> On Jul 25, 2017, at 11:47 AM, Artur Mega <findmeinwland at gmail.com> wrote:
>
> Good day,
> I see in the logs this error. What does it mean? How to handle it?
> 2017-07-25 20:44:21.129627 [ERR] mod_xml_radius.c:930 Didn't
> match: 168.9.82.191 == ^168.9.82.191
>
> --
> ​Regards, Arthur​
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
​С уважением, ​
Артур
​Regards, Arthur​
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From mario_fs at mgtech.com  Tue Jul 25 16:07:43 2017
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 25 Jul 2017 09:07:43 -0700
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <20170725124845350260.78b7d7d1@gcjd.org>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
Message-ID: <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>

I used macFI yesterday and it installed both master and stable fine. Are you running macFI from 20170713? Also, are you installing production stable? You may want to backup your config folder, trash everything in /usr/local (not local itself) and run macFI from July 13.

If you look at the test status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation#macOSInstallation-TestingStatus> you will see they all work, and I use macFI for all testing to save time.

> On Jul 25, 2017, at 3:48 AM, chris <chris at gcjd.org> wrote:
> 
> On Mon, 24 Jul 2017 09:21:36 -0700, Mario G wrote:
>> I recommend:
>> 
>> 1. Use macFI 
>> 
> <https://freeswitch.org/confluence/display/FREESWITCH/macOS+macFI+Installation> 
>> to install 1.6.19, it will rename the old directories. You can do 
>> this manually but macFI is way faster.
> 
> I tried this using macFI but the installation fails with a different 
> problem. From the log folder I see that the configure step failed, so 
> nothing is build afterwards.
> 
> ...
> checking for inflateReset in -lz... no
> configure: error: no usable zlib; please install zlib devel package or 
> equivalent
> Elf:freeswitch chris$ 
> 
> <http://www.gcjd.org/temp0717/FreeSWITCHconfigure.txt>
> 
> This should not happen? As far as I know zlib is available in OSX 
> standard configuration.
> 
> Greetings, Chris
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Jul 25 17:13:34 2017
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Jul 2017 13:13:34 -0400
Subject: [Freeswitch-users] XML_radius module doesn't send request to
 radius server
In-Reply-To: <CAOLEQ7fkqvPtYTb3QykFHf0UJ+eK5iVZ-5jB=6cfAEskowADig@mail.gmail.com>
References: <CAOLEQ7cRjiOG7QcnKBr2cHqSJK=N7ZyhpS=yovDbWJNrOHh0kw@mail.gmail.com>
 <817363BD-C17D-4FBC-9323-8FA70A294961@jerris.com>
 <CAOLEQ7fkqvPtYTb3QykFHf0UJ+eK5iVZ-5jB=6cfAEskowADig@mail.gmail.com>
Message-ID: <5578AF39-F709-4C00-98AD-45EFC70230A1@jerris.com>

anti being set at all is what is triggering your issue, if you remove it, it should work.

> On Jul 25, 2017, at 12:06 PM, Artur Mega <findmeinwland at gmail.com> wrote:
> 
> Yes, anti="true". When I set it to "false", same error raises
> 
> 2017-07-25 21:00 GMT+05:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> do you have “anti” set in your config?
> 
>> On Jul 25, 2017, at 11:47 AM, Artur Mega <findmeinwland at gmail.com <mailto:findmeinwland at gmail.com>> wrote:
>> 
>> Good day, 
>> I see in the logs this error. What does it mean? How to handle it?
>> 2017-07-25 20:44:21.129627 [ERR] mod_xml_radius.c:930 Didn't match: 168.9.82.191 == ^168.9.82.191
>> 
>> -- 
>> ​Regards, Arthur​
> 

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From mario_fs at mgtech.com  Tue Jul 25 17:31:13 2017
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 25 Jul 2017 10:31:13 -0700
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
 <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
Message-ID: <1AAF4B84-36EB-4AF7-85F8-88AC708DFE3E@mgtech.com>

On 10.11.6 and Xcode 7.3.1, I just trashed everything in /usr/local and used macFI to install stable production and master, it went perfect. zlib headers are part of the install from Xcode placed into various subfolders in "applications/Xcode.app/Contents/Developer/Platforms/…”. My suggestions:

1. Make sure you opened Xcode at least once to allow it to install final components.

2. sudo find /* -name “*zlib*"   to find all files on your system with lib in the name. zlib.h should be there. Like:
Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.11.sdk/usr/include/zlib.h

3. Backup/move your freeswitch/conf folder.

4. Move everything in /use/local to trash

5. Use macFI from July 13 to install stable production. Copy/move your old conf folder to the freeswitch runtime.

6. If all good then empty trash.

Otherwise you have some kind of Xcode problem. But since you previously installed that’s puzzling how that could happen.


If you look at the test status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation#macOSInstallation-TestingStatus> you will see they all work, and I use macFI for all testing to save time.
> On Jul 25, 2017, at 9:07 AM, Mario G <mario_fs at mgtech.com> wrote:
> 
> I used macFI yesterday and it installed both master and stable fine. Are you running macFI from 20170713? Also, are you installing production stable? You may want to backup your config folder, trash everything in /usr/local (not local itself) and run macFI from July 13.
> 
> If you look at the test status page <https://freeswitch.org/confluence/display/FREESWITCH/macOS+Installation#macOSInstallation-TestingStatus> you will see they all work, and I use macFI for all testing to save time.

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From chris at gcjd.org  Tue Jul 25 19:00:58 2017
From: chris at gcjd.org (chris)
Date: Tue, 25 Jul 2017 21:00:58 +0200
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
 <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
Message-ID: <20170725210058133466.a7e8adc4@gcjd.org>

On Tue, 25 Jul 2017 09:07:43 -0700, Mario G wrote:
> I used macFI yesterday and it installed both master and stable fine. 
> Are you running macFI from 20170713? Also, are you installing 
> production stable? You may want to backup your config folder, trash 
> everything in /usr/local (not local itself) and run macFI from July 
> 13.

I cannot trash usr/local as there is too much installed. Please do not 
assume a clean usr/local; there is brew of course but also TeX and 
others. That is not an option.

But I found the reason for the zlib problem, which is not a zlib 
problem at all.
The reason is that there is a gcc installed on my test system. So I 
have to do:

TerminalCommandLog("export CC=clang; export CXX=clang++; ./configure", 
3, "FreeSWITCH configure")

for the configure step or else configure will get confused.
(I admit Google helped me to find this out)

While I am typing this mail macFI is building fine for latest 
production 1.6.19.

Do I understand correct that I cannot update the production install 
afterwards via macFI ore manual? Choosing "Update prerequisites and 
FreeSwitch" results in:

Elf:~ chris$ cd /usr/local/src/freeswitch && git pull && make current 
fatal: Not a git repository (or any of the parent directories): .git

Greetings, Chris




From findmeinwland at gmail.com  Tue Jul 25 20:59:31 2017
From: findmeinwland at gmail.com (Artur Mega)
Date: Wed, 26 Jul 2017 01:59:31 +0500
Subject: [Freeswitch-users] XML_radius module doesn't send request to
 radius server
In-Reply-To: <5578AF39-F709-4C00-98AD-45EFC70230A1@jerris.com>
References: <CAOLEQ7cRjiOG7QcnKBr2cHqSJK=N7ZyhpS=yovDbWJNrOHh0kw@mail.gmail.com>
 <817363BD-C17D-4FBC-9323-8FA70A294961@jerris.com>
 <CAOLEQ7fkqvPtYTb3QykFHf0UJ+eK5iVZ-5jB=6cfAEskowADig@mail.gmail.com>
 <5578AF39-F709-4C00-98AD-45EFC70230A1@jerris.com>
Message-ID: <CAOLEQ7e=zsX7qoTqmJJhFPx_Zi_XVKzi+YS9H0u9ToDBwndPbA@mail.gmail.com>

Thanks for reply, now I have

2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:933 Result of (null)
match: 168.9.82.191 == ^168.9.82.191
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
param 'acctserver' with value '6.7.32.422:1813:dsfdssfww'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
param 'radius_timeout' with value '10'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
param 'radius_retries' with value '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
param 'radius_deadtime' with value '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
param 'dictionary' with value '/etc/radiusclient-ng/dictionary'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
param 'seqfile' with value '/var/run/radius.seq'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Acct-Session-Id' value '44' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: 63C50CB0-5977AFEC000E8E1D-D59CF700
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'h323-call-origin' value '589850' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: answer
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'h323-conf-id' value '589848' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: 63C50CB0-5977AFEC000E8E1D-D59CF700
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Cisco-AVPair' value '589825' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: h323-call-id=63C50CB0-5977AFEC000E8E1D-D59CF700
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Cisco-AVPair' value '589825' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: src-gw-ip=130.211.110.78
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Cisco-AVPair' value '589825' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: src-gw-name=arturtrunk
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Cisco-AVPair' value '589825' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: src-number-in=arturtrunk
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Cisco-AVPair' value '589825' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: src-number-out=arturtrunk
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Calling-Station-Id' value '31' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: arturtrunk
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Cisco-AVPair' value '589825' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: dst-gw-ip=168.9.82.191
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Cisco-AVPair' value '589825' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: dst-gw-name=79273004050
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Cisco-AVPair' value '589825' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: dst-number-in=79273004050
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Cisco-AVPair' value '589825' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: dst-number-out=79273004050
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'Called-Station-Id' value '30' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
value: 79273004050
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
attr 'h323-setup-time' value '589849' type '0'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
vend name 'Cisco' vendorpec '9'
2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:437 mod_xml_radius:
value: 01:57:02.218 UTC Wed Jul 26 2017
2017-07-26 01:57:02.218338 [INFO] mod_xml_radius.c:986 mod_xml_radius:
 Accounting Start success
2017-07-26 01:57:02.218338 [DEBUG] switch_core_state_machine.c:166
sofia/internal/arturtrunk at 168.9.82.191 Standard ROUTING
2017-07-26 01:57:02.218338 [INFO] mod_dialplan_xml.c:637 Processing 301
<74950000011>->79273004050 in context internal

Why this?

2017-07-25 22:13 GMT+05:00 Michael Jerris <mike at jerris.com>:

> anti being set at all is what is triggering your issue, if you remove it,
> it should work.
>
> On Jul 25, 2017, at 12:06 PM, Artur Mega <findmeinwland at gmail.com> wrote:
>
> Yes, anti="true". When I set it to "false", same error raises
>
> 2017-07-25 21:00 GMT+05:00 Michael Jerris <mike at jerris.com>:
>
>> do you have “anti” set in your config?
>>
>> On Jul 25, 2017, at 11:47 AM, Artur Mega <findmeinwland at gmail.com> wrote:
>>
>> Good day,
>> I see in the logs this error. What does it mean? How to handle it?
>> 2017-07-25 20:44:21.129627 [ERR] mod_xml_radius.c:930 Didn't
>> match: 168.9.82.191 == ^168.9.82.191
>>
>> --
>> ​Regards, Arthur​
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
​С уважением, ​
Артур
​Regards, Arthur​
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From mario_fs at mgtech.com  Tue Jul 25 22:16:44 2017
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 25 Jul 2017 15:16:44 -0700
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <20170725210058133466.a7e8adc4@gcjd.org>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
 <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
 <20170725210058133466.a7e8adc4@gcjd.org>
Message-ID: <28076722-F29C-4D23-9EC1-2FD1BAAA2294@mgtech.com>

You cannot update stable production using git update, that is only for master development and current branch which are installed via git. Stable does not use git, it is a simple download. For a stable install, you simply reinstall the next maintenance level, macFI will automatically show 1.6.20 when it shows up. I should have asked what you had installed, another gcc messes things up good, been there done that. BTW, you’re original FS was from April, if you look at the wiki manual download instructions you will see I updated them after April for a stable production download. That is the same line macFI uses now for stable download. Glad you’re working now.

> On Jul 25, 2017, at 12:00 PM, chris <chris at gcjd.org> wrote:
> 
> On Tue, 25 Jul 2017 09:07:43 -0700, Mario G wrote:
>> I used macFI yesterday and it installed both master and stable fine. 
>> Are you running macFI from 20170713? Also, are you installing 
>> production stable? You may want to backup your config folder, trash 
>> everything in /usr/local (not local itself) and run macFI from July 
>> 13.
> 
> I cannot trash usr/local as there is too much installed. Please do not 
> assume a clean usr/local; there is brew of course but also TeX and 
> others. That is not an option.
> 
> But I found the reason for the zlib problem, which is not a zlib 
> problem at all.
> The reason is that there is a gcc installed on my test system. So I 
> have to do:
> 
> TerminalCommandLog("export CC=clang; export CXX=clang++; ./configure", 
> 3, "FreeSWITCH configure")
> 
> for the configure step or else configure will get confused.
> (I admit Google helped me to find this out)
> 
> While I am typing this mail macFI is building fine for latest 
> production 1.6.19.
> 
> Do I understand correct that I cannot update the production install 
> afterwards via macFI ore manual? Choosing "Update prerequisites and 
> FreeSwitch" results in:
> 
> Elf:~ chris$ cd /usr/local/src/freeswitch && git pull && make current 
> fatal: Not a git repository (or any of the parent directories): .git
> 
> Greetings, Chris
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From ssinyagin at gmail.com  Tue Jul 25 23:54:14 2017
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 26 Jul 2017 01:54:14 +0200
Subject: [Freeswitch-users] SIM7100 LTE modem with USB audio
In-Reply-To: <CALXCt0pgDzc7rYE8zFsSTdr4RQL_wY0ocbYLDX4AdDxsWL72zw@mail.gmail.com>
References: <CACOYK=jtpcNSyzveoWKVRssHad-_6CwJtZ8H72d=qnHNS5niVA@mail.gmail.com>
 <CALXCt0rUcK8wck0UHu1u9Tqif6X+6PbNeEimZj3PXrC7t8C=Zw@mail.gmail.com>
 <CACOYK=hn7ZNFmkeTe7-HX=9t70zc8vKyxbdrApU1Nf4MX7y3eQ@mail.gmail.com>
 <CALXCt0pgDzc7rYE8zFsSTdr4RQL_wY0ocbYLDX4AdDxsWL72zw@mail.gmail.com>
Message-ID: <CACOYK=haemjXigfs_FJZ8q9yX5FRRqjudu+m0XOuNa0GVqXBOQ@mail.gmail.com>

I assembled the lab, and will start digging the code tomorrow.

The modem sends PCM audio via an UART USB device, just like Huawei dongles
do. So, it looks like only a matter of accommodating to vendor and product
codes and initialization strings.



On 25 Jul 2017 15:10, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

>
>
> On 23 July 2017 at 10:17, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>
>> Hi Giovanni, I only had time to configure it for Internet access:
>>
>> https://txlab.wordpress.com/2017/07/23/simcom-sim7100e-lte-modem/
>>
>> The benefit is mPCIe interface, so it can be placed inside and enclosure.
>> No silly dongles any more :)
>>
>> I'll tinker with voice some time later. I can also provide you SSH access
>> if you wish to work on gsmopen update.
>>
>
> Please, let me know if any progress or so.
>
> Will not be so easy to interface audio to it via USB, will need to create
> a kernel driver, or use a userspace USB lib and create a user space driver.
>
> Happy hacking!
>
> -giovanni
>
>
>
>
>>
>>
>>
>> On 21 Jul 2017 18:36, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>>
>>> Hello Stanislav,
>>>
>>> have you had time to tinker with it?
>>>
>>> How it goes?
>>>
>>> Also, why you would prefer this one instead of the already supported
>>> ones?
>>>
>>> -giovanni
>>>
>>>
>>> On 8 July 2017 at 21:28, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>>
>>>> Simcom has recently released a new 4G/LTE modem, and it has USB audio
>>>> support.
>>>> You can find sim7100_usb_audio_application_note_v0.01.pdf   with
>>>> details at the vendor site, or at techship.com  after registration.
>>>>
>>>> It transmits 8kHZ, 16-bit PCM audio via a USB UART simulation.
>>>>
>>>> So, in theory, gsmopen module may be adapted to it (or maybe a new
>>>> module is worth starting).
>>>>
>>>> I ordered a sample, will check it out soon.
>>>>
>>>> cheers,
>>>> stanislav
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> cell: +39 347 266 56 18
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yu at yu-boot.ru  Wed Jul 26 06:20:49 2017
From: yu at yu-boot.ru (Yu Boot)
Date: Wed, 26 Jul 2017 09:20:49 +0300
Subject: [Freeswitch-users] 180/183 messages timeout
Message-ID: <1ff80335-602d-5be5-3d0e-d21f8863bab9@yu-boot.ru>

Hello.

How to set timeout for progress/ringing messages before "to give up" 
current "bridge" and proceed to next "bridge" rule?



From gmaruzz at gmail.com  Wed Jul 26 07:30:46 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 26 Jul 2017 09:30:46 +0200
Subject: [Freeswitch-users] SIM7100 LTE modem with USB audio
In-Reply-To: <CACOYK=haemjXigfs_FJZ8q9yX5FRRqjudu+m0XOuNa0GVqXBOQ@mail.gmail.com>
References: <CACOYK=jtpcNSyzveoWKVRssHad-_6CwJtZ8H72d=qnHNS5niVA@mail.gmail.com>
 <CALXCt0rUcK8wck0UHu1u9Tqif6X+6PbNeEimZj3PXrC7t8C=Zw@mail.gmail.com>
 <CACOYK=hn7ZNFmkeTe7-HX=9t70zc8vKyxbdrApU1Nf4MX7y3eQ@mail.gmail.com>
 <CALXCt0pgDzc7rYE8zFsSTdr4RQL_wY0ocbYLDX4AdDxsWL72zw@mail.gmail.com>
 <CACOYK=haemjXigfs_FJZ8q9yX5FRRqjudu+m0XOuNa0GVqXBOQ@mail.gmail.com>
Message-ID: <CALXCt0pxVWKHDDEvgmqy199fOVsw3rmsk-stSS=ESANxLuCvaA@mail.gmail.com>

On 26 July 2017 at 01:54, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> I assembled the lab, and will start digging the code tomorrow.
>
> The modem sends PCM audio via an UART USB device, just like Huawei dongles
> do. So, it looks like only a matter of accommodating to vendor and product
> codes and initialization strings.
>

Nice!!!

-giovanni
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From igorolhovskiy at gmail.com  Wed Jul 26 09:27:08 2017
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Wed, 26 Jul 2017 12:27:08 +0300
Subject: [Freeswitch-users] Freeswitch 1.8 - ?
Message-ID: <79b0445a-3e32-42fb-b820-61ba5de64780@Spark>

Seems, release is close?

https://www.packtpub.com/networking-and-servers/freeswitch-18

Regards, Igor
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From findmeinwland at gmail.com  Wed Jul 26 09:47:42 2017
From: findmeinwland at gmail.com (Artur Mega)
Date: Wed, 26 Jul 2017 14:47:42 +0500
Subject: [Freeswitch-users] XML_radius module doesn't send request to
 radius server
In-Reply-To: <CAOLEQ7e=zsX7qoTqmJJhFPx_Zi_XVKzi+YS9H0u9ToDBwndPbA@mail.gmail.com>
References: <CAOLEQ7cRjiOG7QcnKBr2cHqSJK=N7ZyhpS=yovDbWJNrOHh0kw@mail.gmail.com>
 <817363BD-C17D-4FBC-9323-8FA70A294961@jerris.com>
 <CAOLEQ7fkqvPtYTb3QykFHf0UJ+eK5iVZ-5jB=6cfAEskowADig@mail.gmail.com>
 <5578AF39-F709-4C00-98AD-45EFC70230A1@jerris.com>
 <CAOLEQ7e=zsX7qoTqmJJhFPx_Zi_XVKzi+YS9H0u9ToDBwndPbA@mail.gmail.com>
Message-ID: <CAOLEQ7d_3xpbFORBY+J4ibUKLpVuFun4RRM2UoTecFXEcK7z+Q@mail.gmail.com>

My mistake, I used not correct xml config file.

2017-07-26 1:59 GMT+05:00 Artur Mega <findmeinwland at gmail.com>:

> Thanks for reply, now I have
>
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:933 Result of (null)
> match: 168.9.82.191 == ^168.9.82.191
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
> param 'acctserver' with value '6.7.32.422:1813:dsfdssfww'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
> param 'radius_timeout' with value '10'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
> param 'radius_retries' with value '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
> param 'radius_deadtime' with value '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
> param 'dictionary' with value '/etc/radiusclient-ng/dictionary'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
> param 'seqfile' with value '/var/run/radius.seq'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Acct-Session-Id' value '44' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: 63C50CB0-5977AFEC000E8E1D-D59CF700
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'h323-call-origin' value '589850' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: answer
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'h323-conf-id' value '589848' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: 63C50CB0-5977AFEC000E8E1D-D59CF700
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Cisco-AVPair' value '589825' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: h323-call-id=63C50CB0-5977AFEC000E8E1D-D59CF700
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Cisco-AVPair' value '589825' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: src-gw-ip=130.211.110.78
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Cisco-AVPair' value '589825' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: src-gw-name=arturtrunk
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Cisco-AVPair' value '589825' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: src-number-in=arturtrunk
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Cisco-AVPair' value '589825' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: src-number-out=arturtrunk
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Calling-Station-Id' value '31' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: arturtrunk
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Cisco-AVPair' value '589825' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: dst-gw-ip=168.9.82.191
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Cisco-AVPair' value '589825' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: dst-gw-name=79273004050
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Cisco-AVPair' value '589825' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: dst-number-in=79273004050
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Cisco-AVPair' value '589825' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: dst-number-out=79273004050
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'Called-Station-Id' value '30' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
> value: 79273004050
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict
> attr 'h323-setup-time' value '589849' type '0'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict
> vend name 'Cisco' vendorpec '9'
> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:437 mod_xml_radius:
> value: 01:57:02.218 UTC Wed Jul 26 2017
> 2017-07-26 01:57:02.218338 [INFO] mod_xml_radius.c:986 mod_xml_radius:
>  Accounting Start success
> 2017-07-26 01:57:02.218338 [DEBUG] switch_core_state_machine.c:166
> sofia/internal/arturtrunk at 168.9.82.191 Standard ROUTING
> 2017-07-26 01:57:02.218338 [INFO] mod_dialplan_xml.c:637 Processing 301
> <74950000011>->79273004050 in context internal
>
> Why this?
>
> 2017-07-25 22:13 GMT+05:00 Michael Jerris <mike at jerris.com>:
>
>> anti being set at all is what is triggering your issue, if you remove it,
>> it should work.
>>
>> On Jul 25, 2017, at 12:06 PM, Artur Mega <findmeinwland at gmail.com> wrote:
>>
>> Yes, anti="true". When I set it to "false", same error raises
>>
>> 2017-07-25 21:00 GMT+05:00 Michael Jerris <mike at jerris.com>:
>>
>>> do you have “anti” set in your config?
>>>
>>> On Jul 25, 2017, at 11:47 AM, Artur Mega <findmeinwland at gmail.com>
>>> wrote:
>>>
>>> Good day,
>>> I see in the logs this error. What does it mean? How to handle it?
>>> 2017-07-25 20:44:21.129627 [ERR] mod_xml_radius.c:930 Didn't
>>> match: 168.9.82.191 == ^168.9.82.191
>>>
>>> --
>>> ​Regards, Arthur​
>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ​С уважением, ​
> Артур
> ​Regards, Arthur​
>



-- 
​С уважением, ​
Артур
​Regards, Arthur​
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From david.villasmil.work at gmail.com  Wed Jul 26 11:12:07 2017
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 26 Jul 2017 13:12:07 +0200
Subject: [Freeswitch-users] 180/183 messages timeout
In-Reply-To: <1ff80335-602d-5be5-3d0e-d21f8863bab9@yu-boot.ru>
References: <1ff80335-602d-5be5-3d0e-d21f8863bab9@yu-boot.ru>
Message-ID: <CAFGRPVqw3TK_SxBkwXLNEzu5eEJ3o_=LkapkXG-jicAem5TW1A@mail.gmail.com>

https://wiki.freeswitch.org/wiki/Variable_call_timeout

ᐧ

Regards,

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

On Wed, Jul 26, 2017 at 8:20 AM, Yu Boot <yu at yu-boot.ru> wrote:

> Hello.
>
> How to set timeout for progress/ringing messages before "to give up"
> current "bridge" and proceed to next "bridge" rule?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From joelists at tm.net.uk  Wed Jul 26 11:50:35 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Wed, 26 Jul 2017 12:50:35 +0100
Subject: [Freeswitch-users] XML_radius module doesn't send request to
	radius server
In-Reply-To: <CAOLEQ7d_3xpbFORBY+J4ibUKLpVuFun4RRM2UoTecFXEcK7z+Q@mail.gmail.com>
References: <CAOLEQ7cRjiOG7QcnKBr2cHqSJK=N7ZyhpS=yovDbWJNrOHh0kw@mail.gmail.com>
 <817363BD-C17D-4FBC-9323-8FA70A294961@jerris.com>
 <CAOLEQ7fkqvPtYTb3QykFHf0UJ+eK5iVZ-5jB=6cfAEskowADig@mail.gmail.com>
 <5578AF39-F709-4C00-98AD-45EFC70230A1@jerris.com>
 <CAOLEQ7e=zsX7qoTqmJJhFPx_Zi_XVKzi+YS9H0u9ToDBwndPbA@mail.gmail.com>
 <CAOLEQ7d_3xpbFORBY+J4ibUKLpVuFun4RRM2UoTecFXEcK7z+Q@mail.gmail.com>
Message-ID: <2FF66026-9218-4F17-9686-0DFC4C8771BA@tm.net.uk>

Out of interest, what radius server are you using?

Joe Waite

> On 26 Jul 2017, at 10:47, Artur Mega <findmeinwland at gmail.com> wrote:
> 
> My mistake, I used not correct xml config file.
> 
> 2017-07-26 1:59 GMT+05:00 Artur Mega <findmeinwland at gmail.com>:
>> Thanks for reply, now I have
>> 
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:933 Result of (null) match: 168.9.82.191 == ^168.9.82.191 
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add param 'acctserver' with value '6.7.32.422:1813:dsfdssfww' 
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add param 'radius_timeout' with value '10' 
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add param 'radius_retries' with value '0' 
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add param 'radius_deadtime' with value '0' 
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add param 'dictionary' with value '/etc/radiusclient-ng/dictionary' 
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add param 'seqfile' with value '/var/run/radius.seq' 
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Acct-Session-Id' value '44' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: 63C50CB0-5977AFEC000E8E1D-D59CF700
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'h323-call-origin' value '589850' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: answer
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'h323-conf-id' value '589848' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: 63C50CB0-5977AFEC000E8E1D-D59CF700
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: h323-call-id=63C50CB0-5977AFEC000E8E1D-D59CF700
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: src-gw-ip=130.211.110.78
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: src-gw-name=arturtrunk
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: src-number-in=arturtrunk
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: src-number-out=arturtrunk
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Calling-Station-Id' value '31' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: arturtrunk
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: dst-gw-ip=168.9.82.191
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: dst-gw-name=79273004050
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: dst-number-in=79273004050
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: dst-number-out=79273004050
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'Called-Station-Id' value '30' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius: value: 79273004050
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius: dict attr 'h323-setup-time' value '589849' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius: dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:437 mod_xml_radius: value: 01:57:02.218 UTC Wed Jul 26 2017
>> 2017-07-26 01:57:02.218338 [INFO] mod_xml_radius.c:986 mod_xml_radius:  Accounting Start success
>> 2017-07-26 01:57:02.218338 [DEBUG] switch_core_state_machine.c:166 sofia/internal/arturtrunk at 168.9.82.191 Standard ROUTING
>> 2017-07-26 01:57:02.218338 [INFO] mod_dialplan_xml.c:637 Processing 301 <74950000011>->79273004050 in context internal
>> 
>> Why this?
>> 
>> 2017-07-25 22:13 GMT+05:00 Michael Jerris <mike at jerris.com>:
>>> anti being set at all is what is triggering your issue, if you remove it, it should work.
>>> 
>>>> On Jul 25, 2017, at 12:06 PM, Artur Mega <findmeinwland at gmail.com> wrote:
>>>> 
>>>> Yes, anti="true". When I set it to "false", same error raises
>>>> 
>>>> 2017-07-25 21:00 GMT+05:00 Michael Jerris <mike at jerris.com>:
>>>>> do you have “anti” set in your config?
>>>>> 
>>>>>> On Jul 25, 2017, at 11:47 AM, Artur Mega <findmeinwland at gmail.com> wrote:
>>>>>> 
>>>>>> Good day, 
>>>>>> I see in the logs this error. What does it mean? How to handle it?
>>>>>> 2017-07-25 20:44:21.129627 [ERR] mod_xml_radius.c:930 Didn't match: 168.9.82.191 == ^168.9.82.191
>>>>>> 
>>>>>> -- 
>>>>>> ​Regards, Arthur​
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> -- 
>> ​С уважением, ​Артур
>> ​Regards, Arthur​
> 
> 
> 
> -- 
> ​С уважением, ​Артур
> ​Regards, Arthur​
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gmaruzz at gmail.com  Wed Jul 26 12:14:48 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 26 Jul 2017 14:14:48 +0200
Subject: [Freeswitch-users] Freeswitch 1.8 - ?
In-Reply-To: <79b0445a-3e32-42fb-b820-61ba5de64780@Spark>
References: <79b0445a-3e32-42fb-b820-61ba5de64780@Spark>
Message-ID: <CALXCt0o3w6De-iu2zUU-iVUGhJ=Z07UbZAx5Sqy7Q7ns+-MjxQ@mail.gmail.com>

Very close :)

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT

On Jul 26, 2017 11:28 AM, "Igor Olhovskiy" <igorolhovskiy at gmail.com> wrote:

> Seems, release is close?
>
> https://www.packtpub.com/networking-and-servers/freeswitch-18
>
> 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
>
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From brian at freeswitch.org  Wed Jul 26 14:15:35 2017
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Jul 2017 09:15:35 -0500
Subject: [Freeswitch-users] Freeswitch 1.8 - ?
In-Reply-To: <CALXCt0o3w6De-iu2zUU-iVUGhJ=Z07UbZAx5Sqy7Q7ns+-MjxQ@mail.gmail.com>
References: <79b0445a-3e32-42fb-b820-61ba5de64780@Spark>
 <CALXCt0o3w6De-iu2zUU-iVUGhJ=Z07UbZAx5Sqy7Q7ns+-MjxQ@mail.gmail.com>
Message-ID: <CAEJMVkBd6pu-hf0FHkgJS7KbTH+shZ5zJ8C5m+jKwD+NO57m1g@mail.gmail.com>

But you MUST attend ClueCon 2017 to get early access! ;)

/b


On Wed, Jul 26, 2017 at 7:14 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> Very close :)
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
>
> On Jul 26, 2017 11:28 AM, "Igor Olhovskiy" <igorolhovskiy at gmail.com>
> wrote:
>
>> Seems, release is close?
>>
>> https://www.packtpub.com/networking-and-servers/freeswitch-18
>>
>> 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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From findmeinwland at gmail.com  Wed Jul 26 16:23:56 2017
From: findmeinwland at gmail.com (Artur Mega)
Date: Wed, 26 Jul 2017 21:23:56 +0500
Subject: [Freeswitch-users] XML_radius module doesn't send request to
 radius server
In-Reply-To: <2FF66026-9218-4F17-9686-0DFC4C8771BA@tm.net.uk>
References: <CAOLEQ7cRjiOG7QcnKBr2cHqSJK=N7ZyhpS=yovDbWJNrOHh0kw@mail.gmail.com>
 <817363BD-C17D-4FBC-9323-8FA70A294961@jerris.com>
 <CAOLEQ7fkqvPtYTb3QykFHf0UJ+eK5iVZ-5jB=6cfAEskowADig@mail.gmail.com>
 <5578AF39-F709-4C00-98AD-45EFC70230A1@jerris.com>
 <CAOLEQ7e=zsX7qoTqmJJhFPx_Zi_XVKzi+YS9H0u9ToDBwndPbA@mail.gmail.com>
 <CAOLEQ7d_3xpbFORBY+J4ibUKLpVuFun4RRM2UoTecFXEcK7z+Q@mail.gmail.com>
 <2FF66026-9218-4F17-9686-0DFC4C8771BA@tm.net.uk>
Message-ID: <CAOLEQ7d35CL-L=yvy+1Q2EKQYZoG+GD0k7mzBhyvnTj0fpaJFA@mail.gmail.com>

​It is an radius server of one billing system, currently I'm trying to
setup it

2017-07-26 16:50 GMT+05:00 Joseph Waite <joelists at tm.net.uk>:

> Out of interest, what radius server are you using?
>
> Joe Waite
>
> On 26 Jul 2017, at 10:47, Artur Mega <findmeinwland at gmail.com> wrote:
>
> My mistake, I used not correct xml config file.
>
> 2017-07-26 1:59 GMT+05:00 Artur Mega <findmeinwland at gmail.com>:
>
>> Thanks for reply, now I have
>>
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:933 Result of (null)
>> match: 168.9.82.191 == ^168.9.82.191
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
>> param 'acctserver' with value '6.7.32.422:1813:dsfdssfww'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
>> param 'radius_timeout' with value '10'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
>> param 'radius_retries' with value '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
>> param 'radius_deadtime' with value '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
>> param 'dictionary' with value '/etc/radiusclient-ng/dictionary'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:94 Attempting to add
>> param 'seqfile' with value '/var/run/radius.seq'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Acct-Session-Id' value '44' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: 63C50CB0-5977AFEC000E8E1D-D59CF700
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'h323-call-origin' value '589850' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: answer
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'h323-conf-id' value '589848' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: 63C50CB0-5977AFEC000E8E1D-D59CF700
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: h323-call-id=63C50CB0-5977AFEC000E8E1D-D59CF700
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: src-gw-ip=130.211.110.78
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: src-gw-name=arturtrunk
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: src-number-in=arturtrunk
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: src-number-out=arturtrunk
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Calling-Station-Id' value '31' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: arturtrunk
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: dst-gw-ip=168.9.82.191
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: dst-gw-name=79273004050
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: dst-number-in=79273004050
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Cisco-AVPair' value '589825' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: dst-number-out=79273004050
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'Called-Station-Id' value '30' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:557 mod_xml_radius:
>> value: 79273004050
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:378 mod_xml_radius:
>> dict attr 'h323-setup-time' value '589849' type '0'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:394 mod_xml_radius:
>> dict vend name 'Cisco' vendorpec '9'
>> 2017-07-26 01:57:02.218338 [ERR] mod_xml_radius.c:437 mod_xml_radius:
>> value: 01:57:02.218 UTC Wed Jul 26 2017
>> 2017-07-26 01:57:02.218338 [INFO] mod_xml_radius.c:986 mod_xml_radius:
>>  Accounting Start success
>> 2017-07-26 01:57:02.218338 [DEBUG] switch_core_state_machine.c:166
>> sofia/internal/arturtrunk at 168.9.82.191 Standard ROUTING
>> 2017-07-26 01:57:02.218338 [INFO] mod_dialplan_xml.c:637 Processing 301
>> <74950000011>->79273004050 in context internal
>>
>> Why this?
>>
>> 2017-07-25 22:13 GMT+05:00 Michael Jerris <mike at jerris.com>:
>>
>>> anti being set at all is what is triggering your issue, if you remove
>>> it, it should work.
>>>
>>> On Jul 25, 2017, at 12:06 PM, Artur Mega <findmeinwland at gmail.com>
>>> wrote:
>>>
>>> Yes, anti="true". When I set it to "false", same error raises
>>>
>>> 2017-07-25 21:00 GMT+05:00 Michael Jerris <mike at jerris.com>:
>>>
>>>> do you have “anti” set in your config?
>>>>
>>>> On Jul 25, 2017, at 11:47 AM, Artur Mega <findmeinwland at gmail.com>
>>>> wrote:
>>>>
>>>> Good day,
>>>> I see in the logs this error. What does it mean? How to handle it?
>>>> 2017-07-25 20:44:21.129627 [ERR] mod_xml_radius.c:930 Didn't
>>>> match: 168.9.82.191 == ^168.9.82.191
>>>>
>>>> --
>>>> ​Regards, Arthur​
>>>>
>>>>
>>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> ​С уважением, ​
>> Артур
>> ​Regards, Arthur​
>>
>
>
>
> --
> ​С уважением, ​
> Артур
> ​Regards, Arthur​
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
​С уважением, ​
Артур
​Regards, Arthur​
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From sdame at 207me.com  Wed Jul 26 18:06:25 2017
From: sdame at 207me.com (Stephen Dame)
Date: Wed, 26 Jul 2017 14:06:25 -0400
Subject: [Freeswitch-users] Freeswitch 1.8 - ?
In-Reply-To: <CAEJMVkBd6pu-hf0FHkgJS7KbTH+shZ5zJ8C5m+jKwD+NO57m1g@mail.gmail.com>
References: <79b0445a-3e32-42fb-b820-61ba5de64780@Spark>
 <CALXCt0o3w6De-iu2zUU-iVUGhJ=Z07UbZAx5Sqy7Q7ns+-MjxQ@mail.gmail.com>
 <CAEJMVkBd6pu-hf0FHkgJS7KbTH+shZ5zJ8C5m+jKwD+NO57m1g@mail.gmail.com>
Message-ID: <132601d30639$e5f0c360$b1d24a20$@207me.com>

I was able to just download as a https://mapt.io subscription member.  Lots of goods stuff to read.

 

Regards,

Stephen

 

From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Wednesday, July 26, 2017 10:16 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch 1.8 - ?

 

But you MUST attend ClueCon 2017 to get early access! ;)

 

/b

 

 

On Wed, Jul 26, 2017 at 7:14 AM, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com> > wrote:

Very close :)

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT

 

On Jul 26, 2017 11:28 AM, "Igor Olhovskiy" <igorolhovskiy at gmail.com <mailto:igorolhovskiy at gmail.com> > wrote:

Seems, release is close? 

 

https://www.packtpub.com/networking-and-servers/freeswitch-18


Regards, Igor

 

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

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

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


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

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

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





 

-- 

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

Twitter: @FreeSWITCH , @briankwest

http://www.freeswitchbook.com 
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment> 

Allison prompts for FreeSWITCH:


 <https://www.gofundme.com/allison-prompts-for-freeswitch> https://www.gofundme.com/allison-prompts-for-freeswitch

Got Bugs? Report them  <https://freeswitch.org/jira> here! | Reddit:  <https://www.reddit.com/r/freeswitch> /r/freeswitch

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
Skype:briankwest

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From yu at yu-boot.ru  Wed Jul 26 12:27:21 2017
From: yu at yu-boot.ru (Yu Boot)
Date: Wed, 26 Jul 2017 15:27:21 +0300
Subject: [Freeswitch-users] 180/183 messages timeout
In-Reply-To: <CAFGRPVqw3TK_SxBkwXLNEzu5eEJ3o_=LkapkXG-jicAem5TW1A@mail.gmail.com>
References: <1ff80335-602d-5be5-3d0e-d21f8863bab9@yu-boot.ru>
 <CAFGRPVqw3TK_SxBkwXLNEzu5eEJ3o_=LkapkXG-jicAem5TW1A@mail.gmail.com>
Message-ID: <a69ca6f5-3b29-c525-5108-0f7218da261a@yu-boot.ru>

Both call_timeout and originate_timeout drop a call on timeout even if a 
call is in ringing/progress state.


26.07.2017 14:12, David Villasmil пишет:
> https://wiki.freeswitch.org/wiki/Variable_call_timeout
>
> ᐧ
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com 
> <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
>
> On Wed, Jul 26, 2017 at 8:20 AM, Yu Boot <yu at yu-boot.ru 
> <mailto:yu at yu-boot.ru>> wrote:
>
>     Hello.
>
>     How to set timeout for progress/ringing messages before "to give
>     up" current "bridge" and proceed to next "bridge" rule?
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>     http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Wed Jul 26 19:06:36 2017
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Jul 2017 14:06:36 -0500
Subject: [Freeswitch-users] 180/183 messages timeout
In-Reply-To: <a69ca6f5-3b29-c525-5108-0f7218da261a@yu-boot.ru>
References: <1ff80335-602d-5be5-3d0e-d21f8863bab9@yu-boot.ru>
 <CAFGRPVqw3TK_SxBkwXLNEzu5eEJ3o_=LkapkXG-jicAem5TW1A@mail.gmail.com>
 <a69ca6f5-3b29-c525-5108-0f7218da261a@yu-boot.ru>
Message-ID: <CAEJMVkAjw=uCcsZ88Q4LTtK_TmfWpX=aAs6gP8=dd9Jjs3oE7g@mail.gmail.com>

never use call_timeout, use leg_timeout inside {} on each target.

/b


On Wed, Jul 26, 2017 at 7:27 AM, Yu Boot <yu at yu-boot.ru> wrote:

> Both call_timeout and originate_timeout drop a call on timeout even if a
> call is in ringing/progress state.
>
> 26.07.2017 14:12, David Villasmil пишет:
>
> https://wiki.freeswitch.org/wiki/Variable_call_timeout
>
> ᐧ
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337 <+34%20669%2044%2083%2037>
>
> On Wed, Jul 26, 2017 at 8:20 AM, Yu Boot <yu at yu-boot.ru> wrote:
>
>> Hello.
>>
>> How to set timeout for progress/ringing messages before "to give up"
>> current "bridge" and proceed to next "bridge" rule?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From dig1234 at gmail.com  Wed Jul 26 19:55:56 2017
From: dig1234 at gmail.com (Daniel Greenwald)
Date: Wed, 26 Jul 2017 15:55:56 -0400
Subject: [Freeswitch-users] Txfax not hanging up after FAX complete
Message-ID: <CAOsHgts4n5szrdpT=qRiZobuftpzmBeHBuZtRCbDRuHeY4QF=w@mail.gmail.com>

We are seeing cases where FS doesn't hang up after the FAX exchange
complete.

Version is: 1.6.18+git-20170621T132243Z~6e79667c0a~64bit (git 6e79667
2017-06-21 13:22:43Z 64bit)

Any thoughts?
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From brian at freeswitch.org  Wed Jul 26 21:53:37 2017
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Jul 2017 16:53:37 -0500
Subject: [Freeswitch-users] Txfax not hanging up after FAX complete
In-Reply-To: <CAOsHgts4n5szrdpT=qRiZobuftpzmBeHBuZtRCbDRuHeY4QF=w@mail.gmail.com>
References: <CAOsHgts4n5szrdpT=qRiZobuftpzmBeHBuZtRCbDRuHeY4QF=w@mail.gmail.com>
Message-ID: <CAEJMVkBcND28MUvYTXUKM2R8u=RUgfjbSvVAnTbsgKUisANBOw@mail.gmail.com>

Try Master, if the issue persists, and is present in 1.6.19, File a JIRA at
freeswitch.org/jira.

Thanks,
Have a Blessed Day

On Wed, Jul 26, 2017 at 2:55 PM, Daniel Greenwald <dig1234 at gmail.com> wrote:

>
> We are seeing cases where FS doesn't hang up after the FAX exchange
> complete.
>
> Version is: 1.6.18+git-20170621T132243Z~6e79667c0a~64bit (git 6e79667
> 2017-06-21 13:22:43Z 64bit)
>
> Any thoughts?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From brian at freeswitch.org  Wed Jul 26 21:50:13 2017
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Jul 2017 16:50:13 -0500
Subject: [Freeswitch-users] Freeswitch 1.8 - ?
In-Reply-To: <132601d30639$e5f0c360$b1d24a20$@207me.com>
References: <79b0445a-3e32-42fb-b820-61ba5de64780@Spark>
 <CALXCt0o3w6De-iu2zUU-iVUGhJ=Z07UbZAx5Sqy7Q7ns+-MjxQ@mail.gmail.com>
 <CAEJMVkBd6pu-hf0FHkgJS7KbTH+shZ5zJ8C5m+jKwD+NO57m1g@mail.gmail.com>
 <132601d30639$e5f0c360$b1d24a20$@207me.com>
Message-ID: <CAEJMVkAn23AArBrJazS+ymZuYpGekC7sSeviCNQCx+0z-K9sug@mail.gmail.com>

Looks SPAMtastic.

./b


On Wed, Jul 26, 2017 at 1:06 PM, Stephen Dame <sdame at 207me.com> wrote:

> I was able to just download as a https://mapt.io subscription member.
> Lots of goods stuff to read.
>
>
>
> Regards,
>
> Stephen
>
>
>
> *From:* FreeSWITCH-users [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Wednesday, July 26, 2017 10:16 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Freeswitch 1.8 - ?
>
>
>
> But you MUST attend ClueCon 2017 to get early access! ;)
>
>
>
> /b
>
>
>
>
>
> On Wed, Jul 26, 2017 at 7:14 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
> Very close :)
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
>
>
>
> On Jul 26, 2017 11:28 AM, "Igor Olhovskiy" <igorolhovskiy at gmail.com>
> wrote:
>
> Seems, release is close?
>
>
>
> https://www.packtpub.com/networking-and-servers/freeswitch-18
>
>
> 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002>
> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From ssinyagin at gmail.com  Wed Jul 26 22:41:39 2017
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 27 Jul 2017 00:41:39 +0200
Subject: [Freeswitch-users] SIM7100 LTE modem with USB audio
In-Reply-To: <CACOYK=haemjXigfs_FJZ8q9yX5FRRqjudu+m0XOuNa0GVqXBOQ@mail.gmail.com>
References: <CACOYK=jtpcNSyzveoWKVRssHad-_6CwJtZ8H72d=qnHNS5niVA@mail.gmail.com>
 <CALXCt0rUcK8wck0UHu1u9Tqif6X+6PbNeEimZj3PXrC7t8C=Zw@mail.gmail.com>
 <CACOYK=hn7ZNFmkeTe7-HX=9t70zc8vKyxbdrApU1Nf4MX7y3eQ@mail.gmail.com>
 <CALXCt0pgDzc7rYE8zFsSTdr4RQL_wY0ocbYLDX4AdDxsWL72zw@mail.gmail.com>
 <CACOYK=haemjXigfs_FJZ8q9yX5FRRqjudu+m0XOuNa0GVqXBOQ@mail.gmail.com>
Message-ID: <CACOYK=gV3SGLjOwVujxtThOPANo1iLxYLJRr1sLpcfj+YFE2uA@mail.gmail.com>

I looked inside gsmopen_protocol.cpp and my eyes hurt. It has so many
hardcoded vendor-specific commands, and it does not know the actual
vendor that it speaks to. I really don't like adding more bad code
here.

I'm inclining to have a Perl daemon working the AT commands via the
control UART ports, and telling FreeSWITCH take the audio channel from
the audio UART port when it's ready. I think mod_portaudio should do
the job after minor modifications.

Then the AT command handler can be written in a modern and modular
fashion, with independent modules for different hardware.

Also this way we would open the door for new hardware solutions which
are able to send/receive PCM via ttyUSB ports. Then, for example, one
may build a hardware phone with it :)





On Wed, Jul 26, 2017 at 1:54 AM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> I assembled the lab, and will start digging the code tomorrow.
>
> The modem sends PCM audio via an UART USB device, just like Huawei dongles
> do. So, it looks like only a matter of accommodating to vendor and product
> codes and initialization strings.
>
>
>
> On 25 Jul 2017 15:10, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>>
>>
>>
>> On 23 July 2017 at 10:17, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>>
>>> Hi Giovanni, I only had time to configure it for Internet access:
>>>
>>> https://txlab.wordpress.com/2017/07/23/simcom-sim7100e-lte-modem/
>>>
>>> The benefit is mPCIe interface, so it can be placed inside and enclosure.
>>> No silly dongles any more :)
>>>
>>> I'll tinker with voice some time later. I can also provide you SSH access
>>> if you wish to work on gsmopen update.
>>
>>
>> Please, let me know if any progress or so.
>>
>> Will not be so easy to interface audio to it via USB, will need to create
>> a kernel driver, or use a userspace USB lib and create a user space driver.
>>
>> Happy hacking!
>>
>> -giovanni
>>
>>
>>
>>>
>>>
>>>
>>>
>>> On 21 Jul 2017 18:36, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>>>>
>>>> Hello Stanislav,
>>>>
>>>> have you had time to tinker with it?
>>>>
>>>> How it goes?
>>>>
>>>> Also, why you would prefer this one instead of the already supported
>>>> ones?
>>>>
>>>> -giovanni
>>>>
>>>>
>>>> On 8 July 2017 at 21:28, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>>>>
>>>>> Simcom has recently released a new 4G/LTE modem, and it has USB audio
>>>>> support.
>>>>> You can find sim7100_usb_audio_application_note_v0.01.pdf   with
>>>>> details at the vendor site, or at techship.com  after registration.
>>>>>
>>>>> It transmits 8kHZ, 16-bit PCM audio via a USB UART simulation.
>>>>>
>>>>> So, in theory, gsmopen module may be adapted to it (or maybe a new
>>>>> module is worth starting).
>>>>>
>>>>> I ordered a sample, will check it out soon.
>>>>>
>>>>> cheers,
>>>>> stanislav
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> cell: +39 347 266 56 18
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> cell: +39 347 266 56 18
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org


From stefan.kainz at 1012.at  Thu Jul 27 10:42:45 2017
From: stefan.kainz at 1012.at (Stefan Kainz 1012)
Date: Thu, 27 Jul 2017 12:42:45 +0200
Subject: [Freeswitch-users] Reinvite Issue / no SDP
Message-ID: <bc5e1b32-8b55-82ee-bd5a-9199460ca2d8@1012.at>

Hello,

We're using Freeswitch 1.4.18 on Debian 8.
We ran into a problem with internal call transfers, when the PBX sends a
re-invite without SDP offer.

The Problem is as follows:

1) PSTN -> Freeswitch -> PBX -> Call is answered on extension 100, Audio
in both directions
2) Extension 100 does an unattended transfer to extension 101
2.1) PBX sends a re-invite without SDP. Freeswitch offers SDP in OK message
3) PBX acknoledges the OK message
4) Freeswitch is sending media as announced in SDP Offer, but PBX doesnt
"receive" the audio. Also, no media from the PBX.

Since we're a little desperate, we already tried the following settings:

1) 3pcc-enable true/false/proxy
2) inbound-late-negotiation is set to true
3) renegotiate-codec-on-reinvite true / false
4) disable-transfer true / false
5) enable-soa true / false ( it acutally does work when set to false,
but the call gets hangup after a few seconds of audio - ack timeout )

We also tried to enable bypass_media, so that freeswitch doesnt mess
with SDP.
So, the SDP Offer is generated by a Sonus Media Gateway - still no luck.

Any of you had some experience with such behavior?
We would be very thankful for every advice!

have a nice day!

Stefan Kainz



From joelists at tm.net.uk  Fri Jul 28 00:52:58 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Fri, 28 Jul 2017 01:52:58 +0100
Subject: [Freeswitch-users] Sofia Recover
Message-ID: <C86AF05B-53BB-4AC1-B523-68E55ECD669C@tm.net.uk>

Hi Guys

I am setting up Freeswitch HA and have it working, and freeswithc is performing recover when I issue sofia recover

I am using a shared IP so the IP doesn’t change, however FreeSwitch issues a re-invite when it performs the recovery.

This is not needed in a shared IP situation and breaks some clients/carriers.

Is there a setting to tell freeswitch to just re-start media and not send the re-invite.

Regards

From cmrienzo at gmail.com  Fri Jul 28 01:33:38 2017
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Thu, 27 Jul 2017 21:33:38 -0400
Subject: [Freeswitch-users] Sofia Recover
In-Reply-To: <C86AF05B-53BB-4AC1-B523-68E55ECD669C@tm.net.uk>
References: <C86AF05B-53BB-4AC1-B523-68E55ECD669C@tm.net.uk>
Message-ID: <A2181484-C695-4EAF-889C-7CFFECD8D77C@gmail.com>

No


> On Jul 27, 2017, at 20:52, Joseph Waite <joelists at tm.net.uk> wrote:
> 
> Hi Guys
> 
> I am setting up Freeswitch HA and have it working, and freeswithc is performing recover when I issue sofia recover
> 
> I am using a shared IP so the IP doesn’t change, however FreeSwitch issues a re-invite when it performs the recovery.
> 
> This is not needed in a shared IP situation and breaks some clients/carriers.
> 
> Is there a setting to tell freeswitch to just re-start media and not send the re-invite.
> 
> Regards
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From joel at gogii.net  Fri Jul 28 01:37:09 2017
From: joel at gogii.net (Joel Serrano)
Date: Thu, 27 Jul 2017 18:37:09 -0700
Subject: [Freeswitch-users] Sofia Recover
In-Reply-To: <A2181484-C695-4EAF-889C-7CFFECD8D77C@gmail.com>
References: <C86AF05B-53BB-4AC1-B523-68E55ECD669C@tm.net.uk>
 <A2181484-C695-4EAF-889C-7CFFECD8D77C@gmail.com>
Message-ID: <CAMtXxQ=VheqKqbmb70mQc9n0AXhDMii5NneCTTvzyE-qrZN9gQ@mail.gmail.com>

Just curious... why is the re-INVITE needed? (I just would like to
understand, nothing else :D)



On Thu, Jul 27, 2017 at 6:33 PM, <cmrienzo at gmail.com> wrote:

> No
>
>
> > On Jul 27, 2017, at 20:52, Joseph Waite <joelists at tm.net.uk> wrote:
> >
> > Hi Guys
> >
> > I am setting up Freeswitch HA and have it working, and freeswithc is
> performing recover when I issue sofia recover
> >
> > I am using a shared IP so the IP doesn’t change, however FreeSwitch
> issues a re-invite when it performs the recovery.
> >
> > This is not needed in a shared IP situation and breaks some
> clients/carriers.
> >
> > Is there a setting to tell freeswitch to just re-start media and not
> send the re-invite.
> >
> > Regards
> > ____________________________________________________________
> _____________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nandy1925 at gmail.com  Fri Jul 28 01:55:31 2017
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 28 Jul 2017 01:55:31 +0000
Subject: [Freeswitch-users] TURN Configuration under CGN (RFC6598)
Message-ID: <CAHKqMb6uw9fBDu=8_UwQ8LOhLxJKyjs2qwhdvyiLq_hxnPAv+Q@mail.gmail.com>

Hi folks,

I just learned that our Internet provider implemented CGN when I'm about to
experiment WebRTC/mod_verto.  I already signed up a free TURN server
account at http://numb.viagenie.ca/ to proceed.   Where do I place turn:
numb.viagenie.ca in the config files? Is this correct?

sip_profiles/external.xml:
<param name="ext-rtp-ip" value="turn:numb.viagenie.ca"/>
<param name="ext-sip-ip" value="turn:numb.viagenie.ca"/>

/Nandy
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From mirkobrankovic at gmail.com  Fri Jul 28 08:06:20 2017
From: mirkobrankovic at gmail.com (Mirko Brankovic)
Date: Fri, 28 Jul 2017 10:06:20 +0200
Subject: [Freeswitch-users] Sofia Recover
In-Reply-To: <CAMtXxQ=VheqKqbmb70mQc9n0AXhDMii5NneCTTvzyE-qrZN9gQ@mail.gmail.com>
References: <C86AF05B-53BB-4AC1-B523-68E55ECD669C@tm.net.uk>
 <A2181484-C695-4EAF-889C-7CFFECD8D77C@gmail.com>
 <CAMtXxQ=VheqKqbmb70mQc9n0AXhDMii5NneCTTvzyE-qrZN9gQ@mail.gmail.com>
Message-ID: <CAND18T0DK2D9=OjFmybUuNj6Y_dgrQcbVOPqzk_=RNeZXXf09g@mail.gmail.com>

I'm guessing that FS box that is taking ovr the recover, can't open same
udp ports as old one, so that part needs to go trough SDP re-negotiation /
re-invite

On Fri, Jul 28, 2017 at 3:37 AM, Joel Serrano <joel at gogii.net> wrote:

> Just curious... why is the re-INVITE needed? (I just would like to
> understand, nothing else :D)
>
>
>
> On Thu, Jul 27, 2017 at 6:33 PM, <cmrienzo at gmail.com> wrote:
>
>> No
>>
>>
>> > On Jul 27, 2017, at 20:52, Joseph Waite <joelists at tm.net.uk> wrote:
>> >
>> > Hi Guys
>> >
>> > I am setting up Freeswitch HA and have it working, and freeswithc is
>> performing recover when I issue sofia recover
>> >
>> > I am using a shared IP so the IP doesn’t change, however FreeSwitch
>> issues a re-invite when it performs the recovery.
>> >
>> > This is not needed in a shared IP situation and breaks some
>> clients/carriers.
>> >
>> > Is there a setting to tell freeswitch to just re-start media and not
>> send the re-invite.
>> >
>> > Regards
>> > ____________________________________________________________
>> _____________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>> freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Regards,
Mirko
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From joelists at tm.net.uk  Fri Jul 28 08:13:01 2017
From: joelists at tm.net.uk (Joseph Waite)
Date: Fri, 28 Jul 2017 09:13:01 +0100
Subject: [Freeswitch-users] Sofia Recover
In-Reply-To: <CAMtXxQ=VheqKqbmb70mQc9n0AXhDMii5NneCTTvzyE-qrZN9gQ@mail.gmail.com>
References: <C86AF05B-53BB-4AC1-B523-68E55ECD669C@tm.net.uk>
 <A2181484-C695-4EAF-889C-7CFFECD8D77C@gmail.com>
 <CAMtXxQ=VheqKqbmb70mQc9n0AXhDMii5NneCTTvzyE-qrZN9gQ@mail.gmail.com>
Message-ID: <552437AD-BF74-435C-A3CA-53699781C29D@tm.net.uk>

I assume because it is able to work from a server with a different IP. So in that use case it would need the re-invite. The only issue there is that it would not work on anything behind a Nat I would think as the new IP wouldn't have access.

Joe Waite

> On 28 Jul 2017, at 02:37, Joel Serrano <joel at gogii.net> wrote:
> 
> Just curious... why is the re-INVITE needed? (I just would like to understand, nothing else :D)
> 
> 
> 
>> On Thu, Jul 27, 2017 at 6:33 PM, <cmrienzo at gmail.com> wrote:
>> No
>> 
>> 
>> > On Jul 27, 2017, at 20:52, Joseph Waite <joelists at tm.net.uk> wrote:
>> >
>> > Hi Guys
>> >
>> > I am setting up Freeswitch HA and have it working, and freeswithc is performing recover when I issue sofia recover
>> >
>> > I am using a shared IP so the IP doesn’t change, however FreeSwitch issues a re-invite when it performs the recovery.
>> >
>> > This is not needed in a shared IP situation and breaks some clients/carriers.
>> >
>> > Is there a setting to tell freeswitch to just re-start media and not send the re-invite.
>> >
>> > Regards
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From chris at gcjd.org  Fri Jul 28 10:12:09 2017
From: chris at gcjd.org (chris)
Date: Fri, 28 Jul 2017 12:12:09 +0200
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <28076722-F29C-4D23-9EC1-2FD1BAAA2294@mgtech.com>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
 <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
 <20170725210058133466.a7e8adc4@gcjd.org>
 <28076722-F29C-4D23-9EC1-2FD1BAAA2294@mgtech.com>
Message-ID: <20170728121209206595.824ac4d3@gcjd.org>

On Tue, 25 Jul 2017 15:16:44 -0700, Mario G wrote:
> You cannot update stable production using git update, that is only 
> for master development and current branch which are installed via 
> git. Stable does not use git, it is a simple download. For a stable 
> install, you simply reinstall the next maintenance level, macFI will 
> automatically show 1.6.20 when it shows up. I should have asked what 
> you had installed, another gcc messes things up good, been there done 
> that. BTW, you’re original FS was from April, if you look at the wiki 
> manual download instructions you will see I updated them after April 
> for a stable production download. That is the same line macFI uses 
> now for stable download. Glad you’re working now.

Thanks for your help. I have a working 1.6.19 now.

But I still see a problem with the build system. I did a fresh install 
from git to be able to do incremental updates via make current:

FreeSWITCH Version 1.9.0+git~20170725T233142Z~41130001b0~64bit (git 
4113000 2017-07-25 23:31:42Z 64bit)

and that works ok. But if I then do:

git pull && make current

I get  a similar problem like the first time.

config.status: executing depfiles commands
config.status: executing libtool commands
cd src/mod/languages/mod_lua/lua && make clean
/bin/sh: line 0: cd: src/mod/languages/mod_lua/lua: No such file or 
directory
make: *** [lua-reconf] Error 1

There is no src/mod/languages/mod_lua/lua directory yet the makefile 
refers to this directory. See also 
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/Makefile.am>
at line 780 this directory is used.

What's going on here?

Greetings, chris

From brian at freeswitch.org  Fri Jul 28 13:52:47 2017
From: brian at freeswitch.org (Brian West)
Date: Fri, 28 Jul 2017 08:52:47 -0500
Subject: [Freeswitch-users] Sofia Recover
In-Reply-To: <552437AD-BF74-435C-A3CA-53699781C29D@tm.net.uk>
References: <C86AF05B-53BB-4AC1-B523-68E55ECD669C@tm.net.uk>
 <A2181484-C695-4EAF-889C-7CFFECD8D77C@gmail.com>
 <CAMtXxQ=VheqKqbmb70mQc9n0AXhDMii5NneCTTvzyE-qrZN9gQ@mail.gmail.com>
 <552437AD-BF74-435C-A3CA-53699781C29D@tm.net.uk>
Message-ID: <CAEJMVkBsHpcNE5MscR8C6Uzxp5ghTJqRDH5hWTF9Q74+RBZ0PQ@mail.gmail.com>

The reinvite is needed to re-establish the dialog in the sip stack, without
there is no way to even get what you have right now without modifying sofia
to allow you to bring a dialog back to life without having to send a
reinvite.

/b


On Fri, Jul 28, 2017 at 3:13 AM, Joseph Waite <joelists at tm.net.uk> wrote:

> I assume because it is able to work from a server with a different IP. So
> in that use case it would need the re-invite. The only issue there is that
> it would not work on anything behind a Nat I would think as the new IP
> wouldn't have access.
>
> Joe Waite
>
> On 28 Jul 2017, at 02:37, Joel Serrano <joel at gogii.net> wrote:
>
> Just curious... why is the re-INVITE needed? (I just would like to
> understand, nothing else :D)
>
>
>
> On Thu, Jul 27, 2017 at 6:33 PM, <cmrienzo at gmail.com> wrote:
>
>> No
>>
>>
>> > On Jul 27, 2017, at 20:52, Joseph Waite <joelists at tm.net.uk> wrote:
>> >
>> > Hi Guys
>> >
>> > I am setting up Freeswitch HA and have it working, and freeswithc is
>> performing recover when I issue sofia recover
>> >
>> > I am using a shared IP so the IP doesn’t change, however FreeSwitch
>> issues a re-invite when it performs the recovery.
>> >
>> > This is not needed in a shared IP situation and breaks some
>> clients/carriers.
>> >
>> > Is there a setting to tell freeswitch to just re-start media and not
>> send the re-invite.
>> >
>> > Regards
>> > ____________________________________________________________
>> _____________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>> freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From brian at freeswitch.org  Fri Jul 28 13:54:25 2017
From: brian at freeswitch.org (Brian West)
Date: Fri, 28 Jul 2017 08:54:25 -0500
Subject: [Freeswitch-users] TURN Configuration under CGN (RFC6598)
In-Reply-To: <CAHKqMb6uw9fBDu=8_UwQ8LOhLxJKyjs2qwhdvyiLq_hxnPAv+Q@mail.gmail.com>
References: <CAHKqMb6uw9fBDu=8_UwQ8LOhLxJKyjs2qwhdvyiLq_hxnPAv+Q@mail.gmail.com>
Message-ID: <CAEJMVkCo=z+jgBPaOpDeOFqiNXdup77QDb6H_egE0bx9aRChbw@mail.gmail.com>

TURN is not needed with FreeSWITCH, plus we do not support it.

/b


On Thu, Jul 27, 2017 at 8:55 PM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> Hi folks,
>
> I just learned that our Internet provider implemented CGN when I'm about
> to experiment WebRTC/mod_verto.  I already signed up a free TURN server
> account at http://numb.viagenie.ca/ to proceed.   Where do I place turn:
> numb.viagenie.ca in the config files? Is this correct?
>
> sip_profiles/external.xml:
> <param name="ext-rtp-ip" value="turn:numb.viagenie.ca"/>
> <param name="ext-sip-ip" value="turn:numb.viagenie.ca"/>
>
> /Nandy
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From brian at freeswitch.org  Fri Jul 28 13:55:58 2017
From: brian at freeswitch.org (Brian West)
Date: Fri, 28 Jul 2017 08:55:58 -0500
Subject: [Freeswitch-users] Reinvite Issue / no SDP
In-Reply-To: <bc5e1b32-8b55-82ee-bd5a-9199460ca2d8@1012.at>
References: <bc5e1b32-8b55-82ee-bd5a-9199460ca2d8@1012.at>
Message-ID: <CAEJMVkCzAxv5PENJfcP3YNrQfccJ-5ueg0Us3cYxtGeP5VGusA@mail.gmail.com>

1.4 has been EOL for quite some time, you should try the last 1.4 release
and you should test on the latest 1.6 release, If the issue persists
collect the data per the reporting guidelines and file a JIRA.

Thanks,
/b


On Thu, Jul 27, 2017 at 5:42 AM, Stefan Kainz 1012 <stefan.kainz at 1012.at>
wrote:

> Hello,
>
> We're using Freeswitch 1.4.18 on Debian 8.
> We ran into a problem with internal call transfers, when the PBX sends a
> re-invite without SDP offer.
>
> The Problem is as follows:
>
> 1) PSTN -> Freeswitch -> PBX -> Call is answered on extension 100, Audio
> in both directions
> 2) Extension 100 does an unattended transfer to extension 101
> 2.1) PBX sends a re-invite without SDP. Freeswitch offers SDP in OK message
> 3) PBX acknoledges the OK message
> 4) Freeswitch is sending media as announced in SDP Offer, but PBX doesnt
> "receive" the audio. Also, no media from the PBX.
>
> Since we're a little desperate, we already tried the following settings:
>
> 1) 3pcc-enable true/false/proxy
> 2) inbound-late-negotiation is set to true
> 3) renegotiate-codec-on-reinvite true / false
> 4) disable-transfer true / false
> 5) enable-soa true / false ( it acutally does work when set to false,
> but the call gets hangup after a few seconds of audio - ack timeout )
>
> We also tried to enable bypass_media, so that freeswitch doesnt mess
> with SDP.
> So, the SDP Offer is generated by a Sonus Media Gateway - still no luck.
>
> Any of you had some experience with such behavior?
> We would be very thankful for every advice!
>
> have a nice day!
>
> Stefan Kainz
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From tchen61 at gmail.com  Thu Jul 27 23:35:02 2017
From: tchen61 at gmail.com (Thomas Chen)
Date: Thu, 27 Jul 2017 19:35:02 -0400
Subject: [Freeswitch-users] Double NAT
Message-ID: <4c27e39a-69d6-1d16-bda1-af4a55287036@gmail.com>

i am trying this set up  (double NATs)

SIP CLIENT (10.1.2.x) ===( NAT1 )===>[  FS1 ]===>( NAT2 )===>[  FS2 ]

i am trying to make a call from SIP CLIENT...  to FS2 (twilio)

for FS1  configuration...    $${domain}  is set to IP address on NAT1  
(10.1.2.1)   and

ext-rtp-ip and ext-sip-ip  are set to  IP address  on NAT2 (192.168.0.x) 
side...

the "sofia global siptrace on"   is showing that requests are sent from 
FS1 sent to

FS2, and FS1 is not getting any response...


is there any info regarding double NATs ???

thanks



From josedavid at zennio.com  Fri Jul 28 11:21:34 2017
From: josedavid at zennio.com (Jose David Jurado Alonso)
Date: Fri, 28 Jul 2017 13:21:34 +0200
Subject: [Freeswitch-users] FS AWS does not call between mobile phones
	connected to the same WiFi
Message-ID: <CANeCd7rnLuNQChX=iUN6_2i-+13XSeQzXnQ0aJ_U36trt9F-vQ@mail.gmail.com>

Hi,

I have an FS server running on an AWS instance and it does not call between
mobiles connected to the same WiFi. The call is successful if each of the
mobiles is connected to different Wifi or by mobile network (4G).

Obviously FS is always in a different network and I use a domain/URL to
routing the calls.

I also do not see anything strange in the logs. I think that call does not
occur because the server does not find the destination number as registered
and hangup/abandoned the call by timeout.

It always works except when the two mobiles are connected to the same wifi.

My current configuration is the next:

vars.xml:

  <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:mydomain.us"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:mydomain.us"/>
  <!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
  <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
  <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>
  <!-- External SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
  <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/>


internal.xml:

    <param name="apply-nat-acl" value="rfc1918.auto"/>
    <param name="local-network-acl" value="rfc1918.auto"/>
    <param name="ext-rtp-ip" value="autonat:$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="autonat:$${external_sip_ip}"/>

internal-ipv6.xml:

    <!-- Shouldn't set these on IPv6 -->
    <!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
    <!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->


external.xml:

    <param name="local-network-acl" value="rfc1918.auto"/>
    <param name="ext-rtp-ip" value="autonat:$${external_rtp_ip}"/>
    <param name="ext-sip-ip" value="autonat:$${external_sip_ip}"/>

external-ipv6.xml:

    <param name="local-network-acl" value="rfc1918.auto"/>
    <!-- Shouldn't set these on IPv6 -->
    <!--<param name="ext-rtp-ip" value="auto-nat"/>-->
    <!--<param name="ext-sip-ip" value="auto-nat"/>-->


One call example is "sip:1009 at mydomain.us:5080"
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From mario_fs at mgtech.com  Fri Jul 28 15:23:38 2017
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 28 Jul 2017 08:23:38 -0700
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <20170728121209206595.824ac4d3@gcjd.org>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
 <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
 <20170725210058133466.a7e8adc4@gcjd.org>
 <28076722-F29C-4D23-9EC1-2FD1BAAA2294@mgtech.com>
 <20170728121209206595.824ac4d3@gcjd.org>
Message-ID: <5D59A706-38AA-4BCB-82B5-36FD46C99EB2@mgtech.com>

I just tested and yes it just happened here.I just opened Jira FS-10556 <https://freeswitch.org/jira/browse/FS-10556>.

> On Jul 28, 2017, at 3:12 AM, chris <chris at gcjd.org> wrote:
> 
> On Tue, 25 Jul 2017 15:16:44 -0700, Mario G wrote:
>> You cannot update stable production using git update, that is only 
>> for master development and current branch which are installed via 
>> git. Stable does not use git, it is a simple download. For a stable 
>> install, you simply reinstall the next maintenance level, macFI will 
>> automatically show 1.6.20 when it shows up. I should have asked what 
>> you had installed, another gcc messes things up good, been there done 
>> that. BTW, you’re original FS was from April, if you look at the wiki 
>> manual download instructions you will see I updated them after April 
>> for a stable production download. That is the same line macFI uses 
>> now for stable download. Glad you’re working now.
> 
> Thanks for your help. I have a working 1.6.19 now.
> 
> But I still see a problem with the build system. I did a fresh install 
> from git to be able to do incremental updates via make current:
> 
> FreeSWITCH Version 1.9.0+git~20170725T233142Z~41130001b0~64bit (git 
> 4113000 2017-07-25 23:31:42Z 64bit)
> 
> and that works ok. But if I then do:
> 
> git pull && make current
> 
> I get  a similar problem like the first time.
> 
> config.status: executing depfiles commands
> config.status: executing libtool commands
> cd src/mod/languages/mod_lua/lua && make clean
> /bin/sh: line 0: cd: src/mod/languages/mod_lua/lua: No such file or 
> directory
> make: *** [lua-reconf] Error 1
> 
> There is no src/mod/languages/mod_lua/lua directory yet the makefile 
> refers to this directory. See also 
> <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/Makefile.am>
> at line 780 this directory is used.
> 
> What's going on here?
> 
> Greetings, chris
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Jul 28 15:31:35 2017
From: mike at jerris.com (Michael Jerris)
Date: Fri, 28 Jul 2017 11:31:35 -0400
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <5D59A706-38AA-4BCB-82B5-36FD46C99EB2@mgtech.com>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
 <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
 <20170725210058133466.a7e8adc4@gcjd.org>
 <28076722-F29C-4D23-9EC1-2FD1BAAA2294@mgtech.com>
 <20170728121209206595.824ac4d3@gcjd.org>
 <5D59A706-38AA-4BCB-82B5-36FD46C99EB2@mgtech.com>
Message-ID: <A0EE80CA-DBB1-47CA-9F33-92E098F8FDF2@jerris.com>

fixed in master.

> On Jul 28, 2017, at 11:23 AM, Mario G <mario_fs at mgtech.com> wrote:
> 
> I just tested and yes it just happened here.I just opened Jira FS-10556 <https://freeswitch.org/jira/browse/FS-10556>.
> 
>> On Jul 28, 2017, at 3:12 AM, chris <chris at gcjd.org <mailto:chris at gcjd.org>> wrote:
>> 
>> On Tue, 25 Jul 2017 15:16:44 -0700, Mario G wrote:
>>> You cannot update stable production using git update, that is only 
>>> for master development and current branch which are installed via 
>>> git. Stable does not use git, it is a simple download. For a stable 
>>> install, you simply reinstall the next maintenance level, macFI will 
>>> automatically show 1.6.20 when it shows up. I should have asked what 
>>> you had installed, another gcc messes things up good, been there done 
>>> that. BTW, you’re original FS was from April, if you look at the wiki 
>>> manual download instructions you will see I updated them after April 
>>> for a stable production download. That is the same line macFI uses 
>>> now for stable download. Glad you’re working now.
>> 
>> Thanks for your help. I have a working 1.6.19 now.
>> 
>> But I still see a problem with the build system. I did a fresh install 
>> from git to be able to do incremental updates via make current:
>> 
>> FreeSWITCH Version 1.9.0+git~20170725T233142Z~41130001b0~64bit (git 
>> 4113000 2017-07-25 23:31:42Z 64bit)
>> 
>> and that works ok. But if I then do:
>> 
>> git pull && make current
>> 
>> I get  a similar problem like the first time.
>> 
>> config.status: executing depfiles commands
>> config.status: executing libtool commands
>> cd src/mod/languages/mod_lua/lua && make clean
>> /bin/sh: line 0: cd: src/mod/languages/mod_lua/lua: No such file or 
>> directory
>> make: *** [lua-reconf] Error 1
>> 
>> There is no src/mod/languages/mod_lua/lua directory yet the makefile 
>> refers to this directory. See also 
>> <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/Makefile.am <https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/Makefile.am>>
>> at line 780 this directory is used.
>> 
>> What's going on here?
>> 
>> Greetings, chris

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From jungleboogie0 at gmail.com  Fri Jul 28 15:44:17 2017
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Fri, 28 Jul 2017 08:44:17 -0700
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
	updating
In-Reply-To: <20170728121209206595.824ac4d3@gcjd.org>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
 <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
 <20170725210058133466.a7e8adc4@gcjd.org>
 <28076722-F29C-4D23-9EC1-2FD1BAAA2294@mgtech.com>
 <20170728121209206595.824ac4d3@gcjd.org>
Message-ID: <CAKE2PDsw0YXJ+s8O08oN+=iGM4DkV9y2mm8ytOKyUtn5p3DfZA@mail.gmail.com>

On 28 July 2017 at 03:12, chris <chris at gcjd.org> wrote:
> On Tue, 25 Jul 2017 15:16:44 -0700, Mario G wrote:
>> You cannot update stable production using git update, that is only
>> for master development and current branch which are installed via
>> git. Stable does not use git, it is a simple download. For a stable
>> install, you simply reinstall the next maintenance level, macFI will
>> automatically show 1.6.20 when it shows up. I should have asked what
>> you had installed, another gcc messes things up good, been there done
>> that. BTW, you’re original FS was from April, if you look at the wiki
>> manual download instructions you will see I updated them after April
>> for a stable production download. That is the same line macFI uses
>> now for stable download. Glad you’re working now.
>
> Thanks for your help. I have a working 1.6.19 now.
>
> But I still see a problem with the build system. I did a fresh install
> from git to be able to do incremental updates via make current:
>
> FreeSWITCH Version 1.9.0+git~20170725T233142Z~41130001b0~64bit (git
> 4113000 2017-07-25 23:31:42Z 64bit)
>
> and that works ok. But if I then do:
>
> git pull && make current
>

Mike did fix previous lua changes a couple months ago. I always
thought this was failing because of my install.
I've been using "make" and make install to circumvent the problem.

This wasn't OS specific as I'm not using mac.

Thanks to Mike for fixing this in master!!

> I get  a similar problem like the first time.


From gmaruzz at gmail.com  Fri Jul 28 16:05:15 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 29 Jul 2017 01:05:15 +0900
Subject: [Freeswitch-users] FS AWS does not call between mobile phones
 connected to the same WiFi
In-Reply-To: <CANeCd7rnLuNQChX=iUN6_2i-+13XSeQzXnQ0aJ_U36trt9F-vQ@mail.gmail.com>
References: <CANeCd7rnLuNQChX=iUN6_2i-+13XSeQzXnQ0aJ_U36trt9F-vQ@mail.gmail.com>
Message-ID: <CALXCt0peLF_nxPfZUVBn2LUsRGyduKkOXm9J4otAD=rqQ49Bhw@mail.gmail.com>

On 28 July 2017 at 20:21, Jose David Jurado Alonso <josedavid at zennio.com>
wrote:

> My current configuration is the next:
>
> vars.xml:
>
>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:mydomain.us"/>
>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:mydomain.us"/>
>


try making it:

<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=135.44.55.123"/>
<X-PRE-PROCESS cmd="set" data="external_sip_ip=135.44.55.123"/>

Stop and start FreeSeWITCH and see if it helps

Also, maybe something is blocked by some ALG in the clients' router? Try to
connect via TLS...

Hope this help,

-giovanni
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From mario_fs at mgtech.com  Fri Jul 28 16:12:44 2017
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 28 Jul 2017 09:12:44 -0700
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <CAKE2PDsw0YXJ+s8O08oN+=iGM4DkV9y2mm8ytOKyUtn5p3DfZA@mail.gmail.com>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
 <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
 <20170725210058133466.a7e8adc4@gcjd.org>
 <28076722-F29C-4D23-9EC1-2FD1BAAA2294@mgtech.com>
 <20170728121209206595.824ac4d3@gcjd.org>
 <CAKE2PDsw0YXJ+s8O08oN+=iGM4DkV9y2mm8ytOKyUtn5p3DfZA@mail.gmail.com>
Message-ID: <E007E38A-A353-4D3C-AA06-A5215BC83283@mgtech.com>

That looks fixed but I just got failure for libtiff so I reopened jira. Please follow the jira to see the final results.

> On Jul 28, 2017, at 8:44 AM, jungle Boogie <jungleboogie0 at gmail.com> wrote:
> 
> On 28 July 2017 at 03:12, chris <chris at gcjd.org <mailto:chris at gcjd.org>> wrote:
>> On Tue, 25 Jul 2017 15:16:44 -0700, Mario G wrote:
>>> You cannot update stable production using git update, that is only
>>> for master development and current branch which are installed via
>>> git. Stable does not use git, it is a simple download. For a stable
>>> install, you simply reinstall the next maintenance level, macFI will
>>> automatically show 1.6.20 when it shows up. I should have asked what
>>> you had installed, another gcc messes things up good, been there done
>>> that. BTW, you’re original FS was from April, if you look at the wiki
>>> manual download instructions you will see I updated them after April
>>> for a stable production download. That is the same line macFI uses
>>> now for stable download. Glad you’re working now.
>> 
>> Thanks for your help. I have a working 1.6.19 now.
>> 
>> But I still see a problem with the build system. I did a fresh install
>> from git to be able to do incremental updates via make current:
>> 
>> FreeSWITCH Version 1.9.0+git~20170725T233142Z~41130001b0~64bit (git
>> 4113000 2017-07-25 23:31:42Z 64bit)
>> 
>> and that works ok. But if I then do:
>> 
>> git pull && make current
>> 
> 
> Mike did fix previous lua changes a couple months ago. I always
> thought this was failing because of my install.
> I've been using "make" and make install to circumvent the problem.
> 
> This wasn't OS specific as I'm not using mac.
> 
> Thanks to Mike for fixing this in master!!
> 
>> I get  a similar problem like the first time.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Jul 28 18:04:22 2017
From: mike at jerris.com (Michael Jerris)
Date: Fri, 28 Jul 2017 18:04:22 +0000
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
	updating
In-Reply-To: <E007E38A-A353-4D3C-AA06-A5215BC83283@mgtech.com>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
 <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
 <20170725210058133466.a7e8adc4@gcjd.org>
 <28076722-F29C-4D23-9EC1-2FD1BAAA2294@mgtech.com>
 <20170728121209206595.824ac4d3@gcjd.org>
 <CAKE2PDsw0YXJ+s8O08oN+=iGM4DkV9y2mm8ytOKyUtn5p3DfZA@mail.gmail.com>
 <E007E38A-A353-4D3C-AA06-A5215BC83283@mgtech.com>
Message-ID: <CAEwTsAgJEQ1qcGwjCXLnwqG8RA5i_-Swcr4wfsUom05bsUBG8g@mail.gmail.com>

the libtiff error is correct and not a bug.  Libtiff is a dependency you'll
need to install.

On Fri, Jul 28, 2017 at 12:16 PM Mario G <mario_fs at mgtech.com> wrote:

> That looks fixed but I just got failure for libtiff so I reopened jira.
> Please follow the jira to see the final results.
>
> On Jul 28, 2017, at 8:44 AM, jungle Boogie <jungleboogie0 at gmail.com>
> wrote:
>
> On 28 July 2017 at 03:12, chris <chris at gcjd.org> wrote:
>
> On Tue, 25 Jul 2017 15:16:44 -0700, Mario G wrote:
>
> You cannot update stable production using git update, that is only
> for master development and current branch which are installed via
> git. Stable does not use git, it is a simple download. For a stable
> install, you simply reinstall the next maintenance level, macFI will
> automatically show 1.6.20 when it shows up. I should have asked what
> you had installed, another gcc messes things up good, been there done
> that. BTW, you’re original FS was from April, if you look at the wiki
> manual download instructions you will see I updated them after April
> for a stable production download. That is the same line macFI uses
> now for stable download. Glad you’re working now.
>
>
> Thanks for your help. I have a working 1.6.19 now.
>
> But I still see a problem with the build system. I did a fresh install
> from git to be able to do incremental updates via make current:
>
> FreeSWITCH Version 1.9.0+git~20170725T233142Z~41130001b0~64bit (git
> 4113000 2017-07-25 23:31:42Z 64bit)
>
> and that works ok. But if I then do:
>
> git pull && make current
>
>
> Mike did fix previous lua changes a couple months ago. I always
> thought this was failing because of my install.
> I've been using "make" and make install to circumvent the problem.
>
> This wasn't OS specific as I'm not using mac.
>
> Thanks to Mike for fixing this in master!!
>
> I get  a similar problem like the first time.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From nbhatti at gmail.com  Fri Jul 28 19:24:18 2017
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Fri, 28 Jul 2017 21:24:18 +0200
Subject: [Freeswitch-users] Is there Sampling rate conversion CPU penalty?
Message-ID: <CACak_yygwObEYje5j5oFKr9cZOuXZjziu0rGmCQZoEK7kZN_og@mail.gmail.com>

Hi, I am streaming a wav file who’s sampling rate is 44100 and FreeSWITCH
says sample rate 44100 doesn't match requested rate 8000. So I think it’s
doing the conversion here. If so, is there any CPU performance penalty
while doing the conversion? What’s the best practice here should I change
the sampling rate of the file or just leave it like that?

-- 

Sent with Airmail
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From tayeb.meftah at gmail.com  Fri Jul 28 19:53:06 2017
From: tayeb.meftah at gmail.com (Tayeb Meftah)
Date: Fri, 28 Jul 2017 20:53:06 +0100
Subject: [Freeswitch-users] Is there Sampling rate conversion CPU
	penalty?
In-Reply-To: <CACak_yygwObEYje5j5oFKr9cZOuXZjziu0rGmCQZoEK7kZN_og@mail.gmail.com>
References: <CACak_yygwObEYje5j5oFKr9cZOuXZjziu0rGmCQZoEK7kZN_og@mail.gmail.com>
Message-ID: <B4E0F3CC-C7EF-480E-8034-CF64692E852C@gmail.com>

Hey,
The best practice is to transcode your wave files to the diferent rates your clients use
If your clients are only on pstn side, 8khz should be enough
If your clients is on sip side, better to generate 8, 16,32, and 48khz (mono) files
Converting rate is a cpu intencive task and its part of transcoding, so its not recomanded...
Thanks.

Envoyé de mon iPad

> Le 28 juil. 2017 à 20:24, Muhammad Naseer Bhatti <nbhatti at gmail.com> a écrit :
> 
> Hi, I am streaming a wav file who’s sampling rate is 44100 and FreeSWITCH says sample rate 44100 doesn't match requested rate 8000. So I think it’s doing the conversion here. If so, is there any CPU performance penalty while doing the conversion? What’s the best practice here should I change the sampling rate of the file or just leave it like that? 
> 
> -- 
> 
> Sent with Airmail
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Fri Jul 28 20:00:25 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Jul 2017 15:00:25 -0500
Subject: [Freeswitch-users] Is there Sampling rate conversion CPU
	penalty?
In-Reply-To: <B4E0F3CC-C7EF-480E-8034-CF64692E852C@gmail.com>
References: <CACak_yygwObEYje5j5oFKr9cZOuXZjziu0rGmCQZoEK7kZN_og@mail.gmail.com>
 <B4E0F3CC-C7EF-480E-8034-CF64692E852C@gmail.com>
Message-ID: <CAKbxfG9df=LGdK97TpbCcyahi4ben2Gpu+a_PfcFq=xNYZbDog@mail.gmail.com>

Pro tip:

If the file path is /var/sounds/example.wav

you can create

/var/sounds/8000/example.wav
/var/sounds/16000/example.wav
/var/sounds/32000/example.wav
/var/sounds/48000/example.wav

And it will auto-select the correct one that matches the sample rate of
that call.





On Fri, Jul 28, 2017 at 2:53 PM, Tayeb Meftah <tayeb.meftah at gmail.com>
wrote:

> Hey,
> The best practice is to transcode your wave files to the diferent rates
> your clients use
> If your clients are only on pstn side, 8khz should be enough
> If your clients is on sip side, better to generate 8, 16,32, and 48khz
> (mono) files
> Converting rate is a cpu intencive task and its part of transcoding, so
> its not recomanded...
> Thanks.
>
> Envoyé de mon iPad
>
> Le 28 juil. 2017 à 20:24, Muhammad Naseer Bhatti <nbhatti at gmail.com> a
> écrit :
>
> Hi, I am streaming a wav file who’s sampling rate is 44100 and FreeSWITCH
> says sample rate 44100 doesn't match requested rate 8000. So I think it’s
> doing the conversion here. If so, is there any CPU performance penalty
> while doing the conversion? What’s the best practice here should I change
> the sampling rate of the file or just leave it like that?
>
> --
>
> Sent with Airmail
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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

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

https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From nbhatti at gmail.com  Fri Jul 28 21:45:22 2017
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Fri, 28 Jul 2017 23:45:22 +0200
Subject: [Freeswitch-users] Is there Sampling rate conversion CPU
	penalty?
In-Reply-To: <CAKbxfG9df=LGdK97TpbCcyahi4ben2Gpu+a_PfcFq=xNYZbDog@mail.gmail.com>
References: <CACak_yygwObEYje5j5oFKr9cZOuXZjziu0rGmCQZoEK7kZN_og@mail.gmail.com>
 <B4E0F3CC-C7EF-480E-8034-CF64692E852C@gmail.com>
 <CAKbxfG9df=LGdK97TpbCcyahi4ben2Gpu+a_PfcFq=xNYZbDog@mail.gmail.com>
Message-ID: <CACak_yy0frhn-SMZOCjXxxgLc+gzytwcJ+4v-hJ9fpWL7hx=FA@mail.gmail.com>

Ok Awesome. So the transcoding from one sampling rate to other is expensive
and does takes some CPU cycles, correct?

-- 

Sent with Airmail

From: Anthony Minessale <anthony.minessale at gmail.com>
<anthony.minessale at gmail.com>
Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
<freeswitch-users at lists.freeswitch.org>
Date: July 28, 2017 at 11:03:18 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
<freeswitch-users at lists.freeswitch.org>
Subject:  Re: [Freeswitch-users] Is there Sampling rate conversion CPU
penalty?

Pro tip:

If the file path is /var/sounds/example.wav

you can create

/var/sounds/8000/example.wav
/var/sounds/16000/example.wav
/var/sounds/32000/example.wav
/var/sounds/48000/example.wav

And it will auto-select the correct one that matches the sample rate of
that call.





On Fri, Jul 28, 2017 at 2:53 PM, Tayeb Meftah <tayeb.meftah at gmail.com>
wrote:

> Hey,
> The best practice is to transcode your wave files to the diferent rates
> your clients use
> If your clients are only on pstn side, 8khz should be enough
> If your clients is on sip side, better to generate 8, 16,32, and 48khz
> (mono) files
> Converting rate is a cpu intencive task and its part of transcoding, so
> its not recomanded...
> Thanks.
>
> Envoyé de mon iPad
>
> Le 28 juil. 2017 à 20:24, Muhammad Naseer Bhatti <nbhatti at gmail.com> a
> écrit :
>
> Hi, I am streaming a wav file who’s sampling rate is 44100 and FreeSWITCH
> says sample rate 44100 doesn't match requested rate 8000. So I think it’s
> doing the conversion here. If so, is there any CPU performance penalty
> while doing the conversion? What’s the best practice here should I change
> the sampling rate of the file or just leave it like that?
>
> --
>
> Sent with Airmail
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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

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

https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From jungleboogie0 at gmail.com  Fri Jul 28 21:53:05 2017
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Fri, 28 Jul 2017 14:53:05 -0700
Subject: [Freeswitch-users] Is there Sampling rate conversion CPU
	penalty?
In-Reply-To: <CAKbxfG9df=LGdK97TpbCcyahi4ben2Gpu+a_PfcFq=xNYZbDog@mail.gmail.com>
References: <CACak_yygwObEYje5j5oFKr9cZOuXZjziu0rGmCQZoEK7kZN_og@mail.gmail.com>
 <B4E0F3CC-C7EF-480E-8034-CF64692E852C@gmail.com>
 <CAKbxfG9df=LGdK97TpbCcyahi4ben2Gpu+a_PfcFq=xNYZbDog@mail.gmail.com>
Message-ID: <CAKE2PDsQFErnE24_OwSKdPdhz27e+_OeZAzj_MwMXKk+hY7Nrg@mail.gmail.com>

On 28 July 2017 at 13:00, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
> Pro tip:
>
> If the file path is /var/sounds/example.wav
>
> you can create
>
> /var/sounds/8000/example.wav
> /var/sounds/16000/example.wav
> /var/sounds/32000/example.wav
> /var/sounds/48000/example.wav
>
> And it will auto-select the correct one that matches the sample rate of that call.
>
>

Yes, a great tip!

Also, you might need to normalize the audio. Michael pointed me in the
right direction on this earlier in the week and it made a big
difference on some of my audio files.

This is this script freeswitch uses for MOH and the audio files:
https://freeswitch.org/stash/projects/FS/repos/freeswitch-sounds/browse/dist.pl

Check out these lines:
https://freeswitch.org/stash/projects/FS/repos/freeswitch-sounds/browse/dist.pl#82
https://freeswitch.org/stash/projects/FS/repos/freeswitch-sounds/browse/dist.pl#112


From mike at jerris.com  Fri Jul 28 22:07:01 2017
From: mike at jerris.com (Michael Jerris)
Date: Fri, 28 Jul 2017 18:07:01 -0400
Subject: [Freeswitch-users] Is there Sampling rate conversion CPU
	penalty?
In-Reply-To: <CACak_yy0frhn-SMZOCjXxxgLc+gzytwcJ+4v-hJ9fpWL7hx=FA@mail.gmail.com>
References: <CACak_yygwObEYje5j5oFKr9cZOuXZjziu0rGmCQZoEK7kZN_og@mail.gmail.com>
 <B4E0F3CC-C7EF-480E-8034-CF64692E852C@gmail.com>
 <CAKbxfG9df=LGdK97TpbCcyahi4ben2Gpu+a_PfcFq=xNYZbDog@mail.gmail.com>
 <CACak_yy0frhn-SMZOCjXxxgLc+gzytwcJ+4v-hJ9fpWL7hx=FA@mail.gmail.com>
Message-ID: <8188E3D3-A7D0-47D0-BD91-E3FB86B71685@jerris.com>

Compared to not having to, its certainly more cpu intensive, not so much compared to video transcoding.  Depending on the rates involved its probably less expensive than g729 or opus transcoding, more so than g711 transcoding.


> On Jul 28, 2017, at 5:45 PM, Muhammad Naseer Bhatti <nbhatti at gmail.com> wrote:
> 
> 
> Ok Awesome. So the transcoding from one sampling rate to other is expensive and does takes some CPU cycles, correct?
> 
> From: Anthony Minessale <anthony.minessale at gmail.com> <mailto:anthony.minessale at gmail.com>
> Reply: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> <mailto:freeswitch-users at lists.freeswitch.org>
> Date: July 28, 2017 at 11:03:18 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> <mailto:freeswitch-users at lists.freeswitch.org>
> Subject:  Re: [Freeswitch-users] Is there Sampling rate conversion CPU penalty? 
> 
>> Pro tip:
>> 
>> If the file path is /var/sounds/example.wav
>> 
>> you can create 
>> 
>> /var/sounds/8000/example.wav
>> /var/sounds/16000/example.wav
>> /var/sounds/32000/example.wav
>> /var/sounds/48000/example.wav
>> 
>> And it will auto-select the correct one that matches the sample rate of that call.
>> 
>> 
>> On Fri, Jul 28, 2017 at 2:53 PM, Tayeb Meftah <tayeb.meftah at gmail.com <mailto:tayeb.meftah at gmail.com>> wrote:
>> Hey,
>> The best practice is to transcode your wave files to the diferent rates your clients use
>> If your clients are only on pstn side, 8khz should be enough
>> If your clients is on sip side, better to generate 8, 16,32, and 48khz (mono) files
>> Converting rate is a cpu intencive task and its part of transcoding, so its not recomanded...
>> Thanks.
>> 
>> Envoyé de mon iPad
>> 
>> Le 28 juil. 2017 à 20:24, Muhammad Naseer Bhatti <nbhatti at gmail.com <mailto:nbhatti at gmail.com>> a écrit :
>> 
>>> Hi, I am streaming a wav file who’s sampling rate is 44100 and FreeSWITCH says sample rate 44100 doesn't match requested rate 8000. So I think it’s doing the conversion here. If so, is there any CPU performance penalty while doing the conversion? What’s the best practice here should I change the sampling rate of the file or just leave it like that? 
>>> 
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From mike at jerris.com  Fri Jul 28 22:07:51 2017
From: mike at jerris.com (Michael Jerris)
Date: Fri, 28 Jul 2017 18:07:51 -0400
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <CAEwTsAgJEQ1qcGwjCXLnwqG8RA5i_-Swcr4wfsUom05bsUBG8g@mail.gmail.com>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
 <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
 <20170725210058133466.a7e8adc4@gcjd.org>
 <28076722-F29C-4D23-9EC1-2FD1BAAA2294@mgtech.com>
 <20170728121209206595.824ac4d3@gcjd.org>
 <CAKE2PDsw0YXJ+s8O08oN+=iGM4DkV9y2mm8ytOKyUtn5p3DfZA@mail.gmail.com>
 <E007E38A-A353-4D3C-AA06-A5215BC83283@mgtech.com>
 <CAEwTsAgJEQ1qcGwjCXLnwqG8RA5i_-Swcr4wfsUom05bsUBG8g@mail.gmail.com>
Message-ID: <DD04A17C-6D81-46BD-91A1-F8DA1DE2E0D6@jerris.com>

Found the issue in master with make current and libtiff.  I think this is fixed now.

> On Jul 28, 2017, at 2:04 PM, Michael Jerris <mike at jerris.com> wrote:
> 
> the libtiff error is correct and not a bug.  Libtiff is a dependency you'll need to install.
> 
> On Fri, Jul 28, 2017 at 12:16 PM Mario G <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
> That looks fixed but I just got failure for libtiff so I reopened jira. Please follow the jira to see the final results.
> 
>> On Jul 28, 2017, at 8:44 AM, jungle Boogie <jungleboogie0 at gmail.com <mailto:jungleboogie0 at gmail.com>> wrote:
>> 
>> On 28 July 2017 at 03:12, chris <chris at gcjd.org <mailto:chris at gcjd.org>> wrote:
>>> On Tue, 25 Jul 2017 15:16:44 -0700, Mario G wrote:
>>>> You cannot update stable production using git update, that is only
>>>> for master development and current branch which are installed via
>>>> git. Stable does not use git, it is a simple download. For a stable
>>>> install, you simply reinstall the next maintenance level, macFI will
>>>> automatically show 1.6.20 when it shows up. I should have asked what
>>>> you had installed, another gcc messes things up good, been there done
>>>> that. BTW, you’re original FS was from April, if you look at the wiki
>>>> manual download instructions you will see I updated them after April
>>>> for a stable production download. That is the same line macFI uses
>>>> now for stable download. Glad you’re working now.
>>> 
>>> Thanks for your help. I have a working 1.6.19 now.
>>> 
>>> But I still see a problem with the build system. I did a fresh install
>>> from git to be able to do incremental updates via make current:
>>> 
>>> FreeSWITCH Version 1.9.0+git~20170725T233142Z~41130001b0~64bit (git
>>> 4113000 2017-07-25 23:31:42Z 64bit)
>>> 
>>> and that works ok. But if I then do:
>>> 
>>> git pull && make current
>>> 
>> 
>> Mike did fix previous lua changes a couple months ago. I always
>> thought this was failing because of my install.
>> I've been using "make" and make install to circumvent the problem.
>> 
>> This wasn't OS specific as I'm not using mac.
>> 
>> Thanks to Mike for fixing this in master!!
>> 
>>> I get  a similar problem like the first time.

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From grcamauer at gmail.com  Fri Jul 28 22:13:40 2017
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Fri, 28 Jul 2017 19:13:40 -0300
Subject: [Freeswitch-users] Is there Sampling rate conversion CPU
	penalty?
In-Reply-To: <CAKE2PDsQFErnE24_OwSKdPdhz27e+_OeZAzj_MwMXKk+hY7Nrg@mail.gmail.com>
References: <CACak_yygwObEYje5j5oFKr9cZOuXZjziu0rGmCQZoEK7kZN_og@mail.gmail.com>
 <B4E0F3CC-C7EF-480E-8034-CF64692E852C@gmail.com>
 <CAKbxfG9df=LGdK97TpbCcyahi4ben2Gpu+a_PfcFq=xNYZbDog@mail.gmail.com>
 <CAKE2PDsQFErnE24_OwSKdPdhz27e+_OeZAzj_MwMXKk+hY7Nrg@mail.gmail.com>
Message-ID: <CAPNTnnpW0PRiYbW0RFCPP7HAaNv+b8nFHjSiE6ZcMznu8ZjcrQ@mail.gmail.com>

Isn't this what mod_native_file does?

On Fri, Jul 28, 2017 at 6:53 PM, jungle Boogie <jungleboogie0 at gmail.com>
wrote:

> On 28 July 2017 at 13:00, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
> >
> > Pro tip:
> >
> > If the file path is /var/sounds/example.wav
> >
> > you can create
> >
> > /var/sounds/8000/example.wav
> > /var/sounds/16000/example.wav
> > /var/sounds/32000/example.wav
> > /var/sounds/48000/example.wav
> >
> > And it will auto-select the correct one that matches the sample rate of
> that call.
> >
> >
>
> Yes, a great tip!
>
> Also, you might need to normalize the audio. Michael pointed me in the
> right direction on this earlier in the week and it made a big
> difference on some of my audio files.
>
> This is this script freeswitch uses for MOH and the audio files:
> https://freeswitch.org/stash/projects/FS/repos/freeswitch-
> sounds/browse/dist.pl
>
> Check out these lines:
> https://freeswitch.org/stash/projects/FS/repos/freeswitch-
> sounds/browse/dist.pl#82
> https://freeswitch.org/stash/projects/FS/repos/freeswitch-
> sounds/browse/dist.pl#112
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Guillermo Ruiz Camauer
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From mario_fs at mgtech.com  Fri Jul 28 22:26:06 2017
From: mario_fs at mgtech.com (Mario G)
Date: Fri, 28 Jul 2017 15:26:06 -0700
Subject: [Freeswitch-users] Freeswitch 1.6.19: make current fails after
 updating
In-Reply-To: <DD04A17C-6D81-46BD-91A1-F8DA1DE2E0D6@jerris.com>
References: <20170724175354895851.a7d89e7f@gcjd.org>
 <F560EA52-ADE3-47A9-A709-AE62585EF7F6@mgtech.com>
 <20170725124845350260.78b7d7d1@gcjd.org>
 <B39567EC-5BFB-44BE-B01E-289774EE5313@mgtech.com>
 <20170725210058133466.a7e8adc4@gcjd.org>
 <28076722-F29C-4D23-9EC1-2FD1BAAA2294@mgtech.com>
 <20170728121209206595.824ac4d3@gcjd.org>
 <CAKE2PDsw0YXJ+s8O08oN+=iGM4DkV9y2mm8ytOKyUtn5p3DfZA@mail.gmail.com>
 <E007E38A-A353-4D3C-AA06-A5215BC83283@mgtech.com>
 <CAEwTsAgJEQ1qcGwjCXLnwqG8RA5i_-Swcr4wfsUom05bsUBG8g@mail.gmail.com>
 <DD04A17C-6D81-46BD-91A1-F8DA1DE2E0D6@jerris.com>
Message-ID: <C4B4F612-F59E-4F2D-B8F7-CABE3B9B7044@mgtech.com>

Chris, one of the developers fixed this <https://freeswitch.org/jira/browse/FS-10556> and this <https://freeswitch.org/jira/browse/FS-10558> very quickly today. If you continue using FreeSWITCH regularly please consider contributing here <https://www.gofundme.com/allison-prompts-for-freeswitch> and/or here <https://www.freeswitch.org/>.

> On Jul 28, 2017, at 3:07 PM, Michael Jerris <mike at jerris.com> wrote:
> 
> Found the issue in master with make current and libtiff.  I think this is fixed now.
> 
>> On Jul 28, 2017, at 2:04 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> 
>> the libtiff error is correct and not a bug.  Libtiff is a dependency you'll need to install.
>> 
>> On Fri, Jul 28, 2017 at 12:16 PM Mario G <mario_fs at mgtech.com <mailto:mario_fs at mgtech.com>> wrote:
>> That looks fixed but I just got failure for libtiff so I reopened jira. Please follow the jira to see the final results.
>> 
>>> On Jul 28, 2017, at 8:44 AM, jungle Boogie <jungleboogie0 at gmail.com <mailto:jungleboogie0 at gmail.com>> wrote:
>>> 
>>> On 28 July 2017 at 03:12, chris <chris at gcjd.org <mailto:chris at gcjd.org>> wrote:
>>>> On Tue, 25 Jul 2017 15:16:44 -0700, Mario G wrote:
>>>>> You cannot update stable production using git update, that is only
>>>>> for master development and current branch which are installed via
>>>>> git. Stable does not use git, it is a simple download. For a stable
>>>>> install, you simply reinstall the next maintenance level, macFI will
>>>>> automatically show 1.6.20 when it shows up. I should have asked what
>>>>> you had installed, another gcc messes things up good, been there done
>>>>> that. BTW, you’re original FS was from April, if you look at the wiki
>>>>> manual download instructions you will see I updated them after April
>>>>> for a stable production download. That is the same line macFI uses
>>>>> now for stable download. Glad you’re working now.
>>>> 
>>>> Thanks for your help. I have a working 1.6.19 now.
>>>> 
>>>> But I still see a problem with the build system. I did a fresh install
>>>> from git to be able to do incremental updates via make current:
>>>> 
>>>> FreeSWITCH Version 1.9.0+git~20170725T233142Z~41130001b0~64bit (git
>>>> 4113000 2017-07-25 23:31:42Z 64bit)
>>>> 
>>>> and that works ok. But if I then do:
>>>> 
>>>> git pull && make current
>>>> 
>>> 
>>> Mike did fix previous lua changes a couple months ago. I always
>>> thought this was failing because of my install.
>>> I've been using "make" and make install to circumvent the problem.
>>> 
>>> This wasn't OS specific as I'm not using mac.
>>> 
>>> Thanks to Mike for fixing this in master!!
>>> 
>>>> I get  a similar problem like the first time.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Sat Jul 29 03:50:47 2017
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Jul 2017 22:50:47 -0500
Subject: [Freeswitch-users] Is there Sampling rate conversion CPU
	penalty?
In-Reply-To: <CAPNTnnpW0PRiYbW0RFCPP7HAaNv+b8nFHjSiE6ZcMznu8ZjcrQ@mail.gmail.com>
References: <CACak_yygwObEYje5j5oFKr9cZOuXZjziu0rGmCQZoEK7kZN_og@mail.gmail.com>
 <B4E0F3CC-C7EF-480E-8034-CF64692E852C@gmail.com>
 <CAKbxfG9df=LGdK97TpbCcyahi4ben2Gpu+a_PfcFq=xNYZbDog@mail.gmail.com>
 <CAKE2PDsQFErnE24_OwSKdPdhz27e+_OeZAzj_MwMXKk+hY7Nrg@mail.gmail.com>
 <CAPNTnnpW0PRiYbW0RFCPP7HAaNv+b8nFHjSiE6ZcMznu8ZjcrQ@mail.gmail.com>
Message-ID: <CAKbxfG_Rxe535mwAG32uUAYUJfmrL6jesBpW0w7SnyE4FL3bsg@mail.gmail.com>

mod_native file is different.  It only supports raw formats of non VBR
codes.
If you specify a path with no extension, the codec in use will be
substituted, and if it doesn't exist, it will not work.



On Fri, Jul 28, 2017 at 5:13 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com
> wrote:

> Isn't this what mod_native_file does?
>
> On Fri, Jul 28, 2017 at 6:53 PM, jungle Boogie <jungleboogie0 at gmail.com>
> wrote:
>
>> On 28 July 2017 at 13:00, Anthony Minessale <anthony.minessale at gmail.com>
>> wrote:
>> >
>> > Pro tip:
>> >
>> > If the file path is /var/sounds/example.wav
>> >
>> > you can create
>> >
>> > /var/sounds/8000/example.wav
>> > /var/sounds/16000/example.wav
>> > /var/sounds/32000/example.wav
>> > /var/sounds/48000/example.wav
>> >
>> > And it will auto-select the correct one that matches the sample rate of
>> that call.
>> >
>> >
>>
>> Yes, a great tip!
>>
>> Also, you might need to normalize the audio. Michael pointed me in the
>> right direction on this earlier in the week and it made a big
>> difference on some of my audio files.
>>
>> This is this script freeswitch uses for MOH and the audio files:
>> https://freeswitch.org/stash/projects/FS/repos/freeswitch-so
>> unds/browse/dist.pl
>>
>> Check out these lines:
>> https://freeswitch.org/stash/projects/FS/repos/freeswitch-so
>> unds/browse/dist.pl#82
>> https://freeswitch.org/stash/projects/FS/repos/freeswitch-so
>> unds/browse/dist.pl#112
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Guillermo Ruiz Camauer
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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

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

https://www.youtube.com/watch?v=oAxXgyx5jUw
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From tculjaga at gmail.com  Sat Jul 29 06:41:42 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 29 Jul 2017 08:41:42 +0200
Subject: [Freeswitch-users] TURN Configuration under CGN (RFC6598)
In-Reply-To: <CAEJMVkCo=z+jgBPaOpDeOFqiNXdup77QDb6H_egE0bx9aRChbw@mail.gmail.com>
References: <CAHKqMb6uw9fBDu=8_UwQ8LOhLxJKyjs2qwhdvyiLq_hxnPAv+Q@mail.gmail.com>
 <CAEJMVkCo=z+jgBPaOpDeOFqiNXdup77QDb6H_egE0bx9aRChbw@mail.gmail.com>
Message-ID: <CAJoKKdW9k1VTtrf+mznmQ7j5cwtUwuq4-NdebkQ8YUfAUY27Tw@mail.gmail.com>

FS doesn't use turn... but verto client can use it if you pass the
parameter in init function ... like that:

   iceServers: [
      {url: 'turn:numb.viagenie.ca',credential: 'pass', username: 'user'},
      {url: 'stun:ripslinger.undo.it',},
      {url: 'stun:stun.schlund.de',},
      {url: 'stun:stun.ekiga.net',},
      {url: 'stun:stun01.sipphone.com',},
   ],




On 28 July 2017 at 15:54, Brian West <brian at freeswitch.org> wrote:

> TURN is not needed with FreeSWITCH, plus we do not support it.
>
> /b
>
>
> On Thu, Jul 27, 2017 at 8:55 PM, Nandy Dagondon <nandy1925 at gmail.com>
> wrote:
>
>> Hi folks,
>>
>> I just learned that our Internet provider implemented CGN when I'm about
>> to experiment WebRTC/mod_verto.  I already signed up a free TURN server
>> account at http://numb.viagenie.ca/ to proceed.   Where do I place turn:
>> numb.viagenie.ca in the config files? Is this correct?
>>
>> sip_profiles/external.xml:
>> <param name="ext-rtp-ip" value="turn:numb.viagenie.ca"/>
>> <param name="ext-sip-ip" value="turn:numb.viagenie.ca"/>
>>
>> /Nandy
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jungleboogie0 at gmail.com  Sun Jul 30 05:05:56 2017
From: jungleboogie0 at gmail.com (jungle boogie)
Date: Sat, 29 Jul 2017 22:05:56 -0700
Subject: [Freeswitch-users] simonics & freeswitch inbound call immediately
 disconnects when answered
Message-ID: <069e642c-3e4d-cce5-71a0-09d9df5a5905@gmail.com>

Hello All,

I want to have my google voice telephone number registered with 
freeswitch via simonics.com. I have my account already setup and I've 
used it with sip credentials without freeswitch so I know it's setup 
correctly.

The problem I'm having is that as soon as the call is answered, it's 
immediately disconnects.


/usr/local/freeswitch/conf/sip_profiles/external/googlev.xml
<include>
   <gateway name="GVGW">
   <param name="username" value="GV12132935740"/>
   <param name="from-domain" value="gvgw.simonics.com"/>
   <param name="password" value="thepassword"/>
   <param name="extension" value="12130000000"/>
   <param name="proxy" value="gvgw.simonics.com"/>
   <param name="register" value="true"/>
   <param name="register-transport" value="tcp"/>
   </gateway>
</include>

/usr/local/freeswitch/conf/dialplan/public/04_inbound_did.xml
<include>
   <extension name="gv_public_did">
     <condition field="destination_number" expression="^12130000000$">
       <action application="set" data="domain_name=$${domain}"/>
       <action application="transfer" data="2001 XML default"/>
     </condition>
   </extension>
</include>


Here's a link to the call log:
https://pastebin.freeswitch.org/view/79a0e071

Anyone notice something wrong as to why the call would immediately 
disconnect when answered?

Thanks!


From nandy1925 at gmail.com  Sun Jul 30 05:19:09 2017
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sun, 30 Jul 2017 05:19:09 +0000
Subject: [Freeswitch-users] TURN Configuration under CGN (RFC6598)
In-Reply-To: <CAJoKKdW9k1VTtrf+mznmQ7j5cwtUwuq4-NdebkQ8YUfAUY27Tw@mail.gmail.com>
References: <CAHKqMb6uw9fBDu=8_UwQ8LOhLxJKyjs2qwhdvyiLq_hxnPAv+Q@mail.gmail.com>
 <CAEJMVkCo=z+jgBPaOpDeOFqiNXdup77QDb6H_egE0bx9aRChbw@mail.gmail.com>
 <CAJoKKdW9k1VTtrf+mznmQ7j5cwtUwuq4-NdebkQ8YUfAUY27Tw@mail.gmail.com>
Message-ID: <CAHKqMb6kLxjUdoi8a26=0KBgkHyUiOW9=Hv_tn31ZQM1yFv6Vg@mail.gmail.com>

I think this is what I need. Thanks for the feedback. You, too. BKW.



On Sat, Jul 29, 2017 at 6:41 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
> FS doesn't use turn... but verto client can use it if you pass the
> parameter in init function ... like that:
>
>    iceServers: [
>       {url: 'turn:numb.viagenie.ca',credential: 'pass', username: 'user'},
>       {url: 'stun:ripslinger.undo.it',},
>       {url: 'stun:stun.schlund.de',},
>       {url: 'stun:stun.ekiga.net',},
>       {url: 'stun:stun01.sipphone.com',},
>    ],
>
>
>
>
> On 28 July 2017 at 15:54, Brian West <brian at freeswitch.org> wrote:
>
>> TURN is not needed with FreeSWITCH, plus we do not support it.
>>
>> /b
>>
>>
>> On Thu, Jul 27, 2017 at 8:55 PM, Nandy Dagondon <nandy1925 at gmail.com>
>> wrote:
>>
>>> Hi folks,
>>>
>>> I just learned that our Internet provider implemented CGN when I'm about
>>> to experiment WebRTC/mod_verto.  I already signed up a free TURN server
>>> account at http://numb.viagenie.ca/ to proceed.   Where do I place turn:
>>> numb.viagenie.ca in the config files? Is this correct?
>>>
>>> sip_profiles/external.xml:
>>> <param name="ext-rtp-ip" value="turn:numb.viagenie.ca"/>
>>> <param name="ext-sip-ip" value="turn:numb.viagenie.ca"/>
>>>
>>> /Nandy
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>>
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>
>> Allison prompts for FreeSWITCH:
>>
>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
>> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jungleboogie0 at gmail.com  Sun Jul 30 05:22:53 2017
From: jungleboogie0 at gmail.com (jungle boogie)
Date: Sat, 29 Jul 2017 22:22:53 -0700
Subject: [Freeswitch-users] simonics & freeswitch inbound call
 immediately disconnects when answered
In-Reply-To: <069e642c-3e4d-cce5-71a0-09d9df5a5905@gmail.com>
References: <069e642c-3e4d-cce5-71a0-09d9df5a5905@gmail.com>
Message-ID: <5beabe7e-dcf4-f8df-9caf-0e8176714cdb@gmail.com>

Thus said Jungle Boogie on Sat, 29 Jul 2017 22:05:56 -0700
> 
> Here's a link to the call log:
> https://pastebin.freeswitch.org/view/79a0e071
> 

Here's a call going directly to an extension, not to the call group:
https://pastebin.freeswitch.org/view/65e5a179

Also this has global sip trace enabled

> Anyone notice something wrong as to why the call would immediately 
> disconnect when answered?
> 
> Thanks!






From tculjaga at gmail.com  Sun Jul 30 16:08:22 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sun, 30 Jul 2017 18:08:22 +0200
Subject: [Freeswitch-users] SIM7100 LTE modem with USB audio
In-Reply-To: <CACOYK=gV3SGLjOwVujxtThOPANo1iLxYLJRr1sLpcfj+YFE2uA@mail.gmail.com>
References: <CACOYK=jtpcNSyzveoWKVRssHad-_6CwJtZ8H72d=qnHNS5niVA@mail.gmail.com>
 <CALXCt0rUcK8wck0UHu1u9Tqif6X+6PbNeEimZj3PXrC7t8C=Zw@mail.gmail.com>
 <CACOYK=hn7ZNFmkeTe7-HX=9t70zc8vKyxbdrApU1Nf4MX7y3eQ@mail.gmail.com>
 <CALXCt0pgDzc7rYE8zFsSTdr4RQL_wY0ocbYLDX4AdDxsWL72zw@mail.gmail.com>
 <CACOYK=haemjXigfs_FJZ8q9yX5FRRqjudu+m0XOuNa0GVqXBOQ@mail.gmail.com>
 <CACOYK=gV3SGLjOwVujxtThOPANo1iLxYLJRr1sLpcfj+YFE2uA@mail.gmail.com>
Message-ID: <CAJoKKdW5UOGime-vFoadvfNztVJUXMm6+Q5TJ1w-mbKAmP6GOA@mail.gmail.com>

Hi Stanislav, I like the idea!

i'd prefer to have a separate module (not some perl daemon)  tho.

you can instruct udev to run an external program to name devices ( a simple
rpc to stdout would to great ) . This way you can specify dongles
(vendorID, productID, interfaceNumber, dongleSerial...) in your module
config and let your module name the ttys accordingly.

The module should have AT command mappings for preInit, Init, postInit,
dongle_events ( incoming call, ring,...), basic commands ( answer,
hangup...) exposed all in config...

This is where it gets ugly (and a bit less secure), but to keep the module
code sane, its better to expose AT mappings in config.

Once the "signaling" is done, the module should notify mod_portaudio ( a
fork of ) what audio tty it should sit on for the given call


well in theory :=)
i see a lot of work ahead..



On 27 July 2017 at 00:41, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> I looked inside gsmopen_protocol.cpp and my eyes hurt. It has so many
> hardcoded vendor-specific commands, and it does not know the actual
> vendor that it speaks to. I really don't like adding more bad code
> here.
>
> I'm inclining to have a Perl daemon working the AT commands via the
> control UART ports, and telling FreeSWITCH take the audio channel from
> the audio UART port when it's ready. I think mod_portaudio should do
> the job after minor modifications.
>
> Then the AT command handler can be written in a modern and modular
> fashion, with independent modules for different hardware.
>
> Also this way we would open the door for new hardware solutions which
> are able to send/receive PCM via ttyUSB ports. Then, for example, one
> may build a hardware phone with it :)
>
>
>
>
>
> On Wed, Jul 26, 2017 at 1:54 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
> > I assembled the lab, and will start digging the code tomorrow.
> >
> > The modem sends PCM audio via an UART USB device, just like Huawei
> dongles
> > do. So, it looks like only a matter of accommodating to vendor and
> product
> > codes and initialization strings.
> >
> >
> >
> > On 25 Jul 2017 15:10, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
> >>
> >>
> >>
> >> On 23 July 2017 at 10:17, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
> >>>
> >>> Hi Giovanni, I only had time to configure it for Internet access:
> >>>
> >>> https://txlab.wordpress.com/2017/07/23/simcom-sim7100e-lte-modem/
> >>>
> >>> The benefit is mPCIe interface, so it can be placed inside and
> enclosure.
> >>> No silly dongles any more :)
> >>>
> >>> I'll tinker with voice some time later. I can also provide you SSH
> access
> >>> if you wish to work on gsmopen update.
> >>
> >>
> >> Please, let me know if any progress or so.
> >>
> >> Will not be so easy to interface audio to it via USB, will need to
> create
> >> a kernel driver, or use a userspace USB lib and create a user space
> driver.
> >>
> >> Happy hacking!
> >>
> >> -giovanni
> >>
> >>
> >>
> >>>
> >>>
> >>>
> >>>
> >>> On 21 Jul 2017 18:36, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
> >>>>
> >>>> Hello Stanislav,
> >>>>
> >>>> have you had time to tinker with it?
> >>>>
> >>>> How it goes?
> >>>>
> >>>> Also, why you would prefer this one instead of the already supported
> >>>> ones?
> >>>>
> >>>> -giovanni
> >>>>
> >>>>
> >>>> On 8 July 2017 at 21:28, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
> >>>>>
> >>>>> Simcom has recently released a new 4G/LTE modem, and it has USB audio
> >>>>> support.
> >>>>> You can find sim7100_usb_audio_application_note_v0.01.pdf   with
> >>>>> details at the vendor site, or at techship.com  after registration.
> >>>>>
> >>>>> It transmits 8kHZ, 16-bit PCM audio via a USB UART simulation.
> >>>>>
> >>>>> So, in theory, gsmopen module may be adapted to it (or maybe a new
> >>>>> module is worth starting).
> >>>>>
> >>>>> I ordered a sample, will check it out soon.
> >>>>>
> >>>>> cheers,
> >>>>> stanislav
> >>>>>
> >>>>>
> >>>>> ____________________________________________________________
> _____________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://confluence.freeswitch.org
> >>>>> http://www.cluecon.com
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
> options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>>
> >>>> Sincerely,
> >>>>
> >>>> Giovanni Maruzzelli
> >>>> OpenTelecom.IT
> >>>> cell: +39 347 266 56 18
> >>>>
> >>>>
> >>>> ____________________________________________________________
> _____________
> >>>> Professional FreeSWITCH Consulting Services:
> >>>> consulting at freeswitch.org
> >>>> http://www.freeswitchsolutions.com
> >>>>
> >>>> Official FreeSWITCH Sites
> >>>> http://www.freeswitch.org
> >>>> http://confluence.freeswitch.org
> >>>> http://www.cluecon.com
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
> options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>> ____________________________________________________________
> _____________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://confluence.freeswitch.org
> >>> http://www.cluecon.com
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
> options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >>
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> OpenTelecom.IT
> >> cell: +39 347 266 56 18
> >>
> >> ____________________________________________________________
> _____________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
> options/freeswitch-users
> >> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From lists at telefaks.de  Sun Jul 30 21:28:06 2017
From: lists at telefaks.de (Peter Steinbach)
Date: Sun, 30 Jul 2017 23:28:06 +0200
Subject: [Freeswitch-users] Working example for presence_in
Message-ID: <597E4F66.8060804@telefaks.de>

Hallo,

I want to make a BLF lamp on the phone turn red/green/flash based on
day/night info

According to
https://freeswitch.org/confluence/display/FREESWITCH/PRESENCE+IN+event+example
I am sending here a presence info for 200 (ringing)
sendevent PRESENCE_IN
proto: sip
from: 200 at my.domain.com
login: 200 at my.domain.com
event_type: presence
alt_event_type: dialog
Presence-Call-Direction: outbound
answer-state: early

via ESL an get a reply
+OK 74a91ebc-7aaa-4f04-8fd1-5158836f594d

Fs_cli shows
freeswitch at my.domain.com> /events plain PRESENCE_IN
+OK event listener enabled plain
RECV EVENT
Command: sendevent PRESENCE_IN
proto: sip
from: 200 at my.domain.com
login: 200 at my.domain.com
event_type: presence
alt_event_type: dialog
Presence-Call-Direction: outbound
answer-state: early
Event-UUID: 7a320e1e-8455-4804-b878-66c83914976e
Event-Name: PRESENCE_IN
Core-UUID: 81230f28-a930-4777-bdc5-d3df670c62ff
FreeSWITCH-Hostname: my.domain.com
FreeSWITCH-Switchname: my.domain.com
FreeSWITCH-IPv4: 192.168.1.9
FreeSWITCH-IPv6: ::1
Event-Date-Local: 2017-07-30 23:23:11
Event-Date-GMT: Sun, 30 Jul 2017 21:23:11 GMT
Event-Date-Timestamp: 1501449791174168
Event-Calling-File: mod_event_socket.c
Event-Calling-Function: parse_command
Event-Calling-Line-Number: 2256
Event-Sequence: 88094

I also see a Notify message to 209 (as 209 was subscribing to 200)
NOTIFY sip:209 at 192.168.178.121:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.178.9;rport;branch=z9hG4bKXKDD7j4B9SBeD.
Max-Forwards: 70.
From: <sip:200 at my.domain.com;user=phone>;tag=svXxoq1TseLS.
To: <sip:209 at my.domain.com;user=phone>;tag=1852934565.
Call-ID: 470672824-5060-5 at 192.168.178.121.
CSeq: 911343401 NOTIFY.
Contact: <sip:200 at 192.168.178.9:5060>.
User-Agent: Freeswitch.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Supported: timer, path, replaces.
Event: dialog.
Allow-Events: talk, hold, conference, presence, as-feature-event,
dialog, line-seize, call-info, sla, include-session-description,
presence.win$
Subscription-State: active;expires=3214.
Content-Type: application/dialog-info+xml.
Content-Length: 156.
.
<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="123"
state="full" entity="sip:200 at my.domain.com">
</dialog-info>
 
But the message sent is always the same (state=full), and does not
reflect the answer state in the message (early, confirmed, terminated).

Anybody has a clue, what's wrong here or has a better working example?

-- 
With kind regards
Peter Steinbach 

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



From michael at mailworks.org  Sun Jul 30 23:17:33 2017
From: michael at mailworks.org (Michael Avers)
Date: Sun, 30 Jul 2017 16:17:33 -0700
Subject: [Freeswitch-users] Greenswitch ESL for Python
Message-ID: <1501456653.311283.1057571504.5FE3D6C2@webmail.messagingengine.com>

Hello,

I'm looking at greenswitch as an alternative Python inbound ESL connector. Other than Italo's company (thanks for releasing this BTW), anyone else has seen success using it in a moderate to heavy call environment?

Project is at https://github.com/EvoluxBR/greenswitch

Thanks
Mike


From ssinyagin at gmail.com  Mon Jul 31 00:43:56 2017
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 31 Jul 2017 02:43:56 +0200
Subject: [Freeswitch-users] SIM7100 LTE modem with USB audio
In-Reply-To: <CAJoKKdW5UOGime-vFoadvfNztVJUXMm6+Q5TJ1w-mbKAmP6GOA@mail.gmail.com>
References: <CACOYK=jtpcNSyzveoWKVRssHad-_6CwJtZ8H72d=qnHNS5niVA@mail.gmail.com>
 <CALXCt0rUcK8wck0UHu1u9Tqif6X+6PbNeEimZj3PXrC7t8C=Zw@mail.gmail.com>
 <CACOYK=hn7ZNFmkeTe7-HX=9t70zc8vKyxbdrApU1Nf4MX7y3eQ@mail.gmail.com>
 <CALXCt0pgDzc7rYE8zFsSTdr4RQL_wY0ocbYLDX4AdDxsWL72zw@mail.gmail.com>
 <CACOYK=haemjXigfs_FJZ8q9yX5FRRqjudu+m0XOuNa0GVqXBOQ@mail.gmail.com>
 <CACOYK=gV3SGLjOwVujxtThOPANo1iLxYLJRr1sLpcfj+YFE2uA@mail.gmail.com>
 <CAJoKKdW5UOGime-vFoadvfNztVJUXMm6+Q5TJ1w-mbKAmP6GOA@mail.gmail.com>
Message-ID: <CACOYK=iW30wNWLjCacsabBb8AgLwGGtky9gy=pPyHaFnaSF5-A@mail.gmail.com>

Greetings,

I started slowly outlining a new module, mod_ttyusb_audio. It will do
a simple task by bridging a channel with a specified /dev/ttyUSBx PCM
flow using a specified codec (All modems that I know use the same L16
at 8kHZ, but who knows what other codecs there might be). As I didn't
yet find a sponsor for this development, it goes quite slowly.

Once it's done, the signaling could be done either within FreeSWITCH,
or in an outside ESL daemon. An external daemon would allow more
flexibility, but there's a burden of running one more process.

In regards to udev rules -- you actually don't need them. The logic of
finding the modem and matching it against IMEI and IMSI is already in
mod_gsmopen (was that Giovanni who implemented it?). It just needs a
bit of improvement in making it more vendor-independent. Also I would
do matching on IMEI OR IMSI, not the AND condition that is in the
module currently.

I actually do have a set of udev rules, they are in fact needed for
data connection:
https://github.com/ssinyagin/wwan_udev_rules
But they fail if you have more than one modem, and here's an explanation:
https://txlab.wordpress.com/2017/05/19/two-lte-modems-with-pc-engines-apu3/

cheers,
stanislav








On Sun, Jul 30, 2017 at 6:08 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> Hi Stanislav, I like the idea!
>
> i'd prefer to have a separate module (not some perl daemon)  tho.
>
> you can instruct udev to run an external program to name devices ( a simple
> rpc to stdout would to great ) . This way you can specify dongles (vendorID,
> productID, interfaceNumber, dongleSerial...) in your module config and let
> your module name the ttys accordingly.
>
> The module should have AT command mappings for preInit, Init, postInit,
> dongle_events ( incoming call, ring,...), basic commands ( answer,
> hangup...) exposed all in config...
>
> This is where it gets ugly (and a bit less secure), but to keep the module
> code sane, its better to expose AT mappings in config.
>
> Once the "signaling" is done, the module should notify mod_portaudio ( a
> fork of ) what audio tty it should sit on for the given call
>
>
> well in theory :=)
> i see a lot of work ahead..
>
>
>
> On 27 July 2017 at 00:41, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>>
>> I looked inside gsmopen_protocol.cpp and my eyes hurt. It has so many
>> hardcoded vendor-specific commands, and it does not know the actual
>> vendor that it speaks to. I really don't like adding more bad code
>> here.
>>
>> I'm inclining to have a Perl daemon working the AT commands via the
>> control UART ports, and telling FreeSWITCH take the audio channel from
>> the audio UART port when it's ready. I think mod_portaudio should do
>> the job after minor modifications.
>>
>> Then the AT command handler can be written in a modern and modular
>> fashion, with independent modules for different hardware.
>>
>> Also this way we would open the door for new hardware solutions which
>> are able to send/receive PCM via ttyUSB ports. Then, for example, one
>> may build a hardware phone with it :)
>>
>>
>>
>>
>>
>> On Wed, Jul 26, 2017 at 1:54 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>> > I assembled the lab, and will start digging the code tomorrow.
>> >
>> > The modem sends PCM audio via an UART USB device, just like Huawei
>> > dongles
>> > do. So, it looks like only a matter of accommodating to vendor and
>> > product
>> > codes and initialization strings.
>> >
>> >
>> >
>> > On 25 Jul 2017 15:10, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>> >>
>> >>
>> >>
>> >> On 23 July 2017 at 10:17, Stanislav Sinyagin <ssinyagin at gmail.com>
>> >> wrote:
>> >>>
>> >>> Hi Giovanni, I only had time to configure it for Internet access:
>> >>>
>> >>> https://txlab.wordpress.com/2017/07/23/simcom-sim7100e-lte-modem/
>> >>>
>> >>> The benefit is mPCIe interface, so it can be placed inside and
>> >>> enclosure.
>> >>> No silly dongles any more :)
>> >>>
>> >>> I'll tinker with voice some time later. I can also provide you SSH
>> >>> access
>> >>> if you wish to work on gsmopen update.
>> >>
>> >>
>> >> Please, let me know if any progress or so.
>> >>
>> >> Will not be so easy to interface audio to it via USB, will need to
>> >> create
>> >> a kernel driver, or use a userspace USB lib and create a user space
>> >> driver.
>> >>
>> >> Happy hacking!
>> >>
>> >> -giovanni
>> >>
>> >>
>> >>
>> >>>
>> >>>
>> >>>
>> >>>
>> >>> On 21 Jul 2017 18:36, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>> >>>>
>> >>>> Hello Stanislav,
>> >>>>
>> >>>> have you had time to tinker with it?
>> >>>>
>> >>>> How it goes?
>> >>>>
>> >>>> Also, why you would prefer this one instead of the already supported
>> >>>> ones?
>> >>>>
>> >>>> -giovanni
>> >>>>
>> >>>>
>> >>>> On 8 July 2017 at 21:28, Stanislav Sinyagin <ssinyagin at gmail.com>
>> >>>> wrote:
>> >>>>>
>> >>>>> Simcom has recently released a new 4G/LTE modem, and it has USB
>> >>>>> audio
>> >>>>> support.
>> >>>>> You can find sim7100_usb_audio_application_note_v0.01.pdf   with
>> >>>>> details at the vendor site, or at techship.com  after registration.
>> >>>>>
>> >>>>> It transmits 8kHZ, 16-bit PCM audio via a USB UART simulation.
>> >>>>>
>> >>>>> So, in theory, gsmopen module may be adapted to it (or maybe a new
>> >>>>> module is worth starting).
>> >>>>>
>> >>>>> I ordered a sample, will check it out soon.
>> >>>>>
>> >>>>> cheers,
>> >>>>> stanislav
>> >>>>>
>> >>>>>
>> >>>>>
>> >>>>> _________________________________________________________________________
>> >>>>> Professional FreeSWITCH Consulting Services:
>> >>>>> consulting at freeswitch.org
>> >>>>> http://www.freeswitchsolutions.com
>> >>>>>
>> >>>>> Official FreeSWITCH Sites
>> >>>>> http://www.freeswitch.org
>> >>>>> http://confluence.freeswitch.org
>> >>>>> http://www.cluecon.com
>> >>>>>
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>
>> >>>>>
>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> http://www.freeswitch.org
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>> --
>> >>>>
>> >>>> Sincerely,
>> >>>>
>> >>>> Giovanni Maruzzelli
>> >>>> OpenTelecom.IT
>> >>>> cell: +39 347 266 56 18
>> >>>>
>> >>>>
>> >>>>
>> >>>> _________________________________________________________________________
>> >>>> Professional FreeSWITCH Consulting Services:
>> >>>> consulting at freeswitch.org
>> >>>> http://www.freeswitchsolutions.com
>> >>>>
>> >>>> Official FreeSWITCH Sites
>> >>>> http://www.freeswitch.org
>> >>>> http://confluence.freeswitch.org
>> >>>> http://www.cluecon.com
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>
>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>>
>> >>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://confluence.freeswitch.org
>> >>> http://www.cluecon.com
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> >> --
>> >>
>> >> Sincerely,
>> >>
>> >> Giovanni Maruzzelli
>> >> OpenTelecom.IT
>> >> cell: +39 347 266 56 18
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From brian at freeswitch.org  Mon Jul 31 00:44:44 2017
From: brian at freeswitch.org (Brian West)
Date: Mon, 31 Jul 2017 00:44:44 +0000
Subject: [Freeswitch-users] TURN Configuration under CGN (RFC6598)
In-Reply-To: <CAJoKKdW9k1VTtrf+mznmQ7j5cwtUwuq4-NdebkQ8YUfAUY27Tw@mail.gmail.com>
References: <CAHKqMb6uw9fBDu=8_UwQ8LOhLxJKyjs2qwhdvyiLq_hxnPAv+Q@mail.gmail.com>
 <CAEJMVkCo=z+jgBPaOpDeOFqiNXdup77QDb6H_egE0bx9aRChbw@mail.gmail.com>
 <CAJoKKdW9k1VTtrf+mznmQ7j5cwtUwuq4-NdebkQ8YUfAUY27Tw@mail.gmail.com>
Message-ID: <CAEJMVkDKBGoBn8C9c1ny0x9yVfcD1w7XvmDC5x6Qq43E+xzOUg@mail.gmail.com>

What issue are you trying to solve with turn?

On Sat, Jul 29, 2017 at 1:42 AM Tihomir Culjaga <tculjaga at gmail.com> wrote:

>
> FS doesn't use turn... but verto client can use it if you pass the
> parameter in init function ... like that:
>
>    iceServers: [
>       {url: 'turn:numb.viagenie.ca',credential: 'pass', username: 'user'},
>       {url: 'stun:ripslinger.undo.it',},
>       {url: 'stun:stun.schlund.de',},
>       {url: 'stun:stun.ekiga.net',},
>       {url: 'stun:stun01.sipphone.com',},
>    ],
>
>
>
>
> On 28 July 2017 at 15:54, Brian West <brian at freeswitch.org> wrote:
>
>> TURN is not needed with FreeSWITCH, plus we do not support it.
>>
>> /b
>>
>>
>> On Thu, Jul 27, 2017 at 8:55 PM, Nandy Dagondon <nandy1925 at gmail.com>
>> wrote:
>>
>>> Hi folks,
>>>
>>> I just learned that our Internet provider implemented CGN when I'm about
>>> to experiment WebRTC/mod_verto.  I already signed up a free TURN server
>>> account at http://numb.viagenie.ca/ to proceed.   Where do I place turn:
>>> numb.viagenie.ca in the config files? Is this correct?
>>>
>>> sip_profiles/external.xml:
>>> <param name="ext-rtp-ip" value="turn:numb.viagenie.ca"/>
>>> <param name="ext-sip-ip" value="turn:numb.viagenie.ca"/>
>>>
>>> /Nandy
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>>
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>
>> Allison prompts for FreeSWITCH:
>>
>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
>> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 

*Brian West*
brian at freeswitch.org

*Twitter: @FreeSWITCH , @briankwest*

http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

Book a phone call (CST) <https://freeswitch.com/appointment>

Allison prompts for FreeSWITCH:

*https://www.gofundme.com/allison-prompts-for-freeswitch*
<https://www.gofundme.com/allison-prompts-for-freeswitch>

Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
/r/freeswitch <https://www.reddit.com/r/freeswitch>

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*Skype:*briankwest
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From joel at gogii.net  Mon Jul 31 06:42:35 2017
From: joel at gogii.net (Joel Serrano)
Date: Sun, 30 Jul 2017 23:42:35 -0700
Subject: [Freeswitch-users] Greenswitch ESL for Python
In-Reply-To: <1501456653.311283.1057571504.5FE3D6C2@webmail.messagingengine.com>
References: <1501456653.311283.1057571504.5FE3D6C2@webmail.messagingengine.com>
Message-ID: <CAMtXxQnhSeY7HCcxuZ0P+akYWa68qHjNj3d33tKiVkZGss+Pxw@mail.gmail.com>

I'm actually in the process of testing https://github.com/sjthomason/PyESL from
Spencer Thomason, he shared it with the list around a year ago:

http://lists.freeswitch.org/pipermail/freeswitch-users/2016-August/122148.html

I didn't know about GreenSWITCH so I'll check it out too.

Joel.







On Sun, Jul 30, 2017 at 4:17 PM, Michael Avers <michael at mailworks.org>
wrote:

> Hello,
>
> I'm looking at greenswitch as an alternative Python inbound ESL connector.
> Other than Italo's company (thanks for releasing this BTW), anyone else has
> seen success using it in a moderate to heavy call environment?
>
> Project is at https://github.com/EvoluxBR/greenswitch
>
> Thanks
> Mike
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From michael at mailworks.org  Mon Jul 31 09:04:16 2017
From: michael at mailworks.org (Michael Avers)
Date: Mon, 31 Jul 2017 02:04:16 -0700
Subject: [Freeswitch-users] Greenswitch ESL for Python
In-Reply-To: <CAMtXxQnhSeY7HCcxuZ0P+akYWa68qHjNj3d33tKiVkZGss+Pxw@mail.gmail.com>
References: <1501456653.311283.1057571504.5FE3D6C2@webmail.messagingengine.com>
 <CAMtXxQnhSeY7HCcxuZ0P+akYWa68qHjNj3d33tKiVkZGss+Pxw@mail.gmail.com>
Message-ID: <1501491856.437006.1057954392.6A534CBA@webmail.messagingengine.com>

Code-wise they both work in a similar fashion, but I think where Greenswitch shines is in its use of Gevent.
Mike


On Sun, Jul 30, 2017, at 11:42 PM, Joel Serrano wrote:
> I'm actually in the process of testing https://github.com/sjthomason/PyESL from Spencer Thomason, he shared it with the list around a year ago:> 
> http://lists.freeswitch.org/pipermail/freeswitch-users/2016-August/122148.html> 
> I didn't know about GreenSWITCH so I'll check it out too.
> 
> Joel.
> 
> 
> 
> 
> 
> 
> 
> On Sun, Jul 30, 2017 at 4:17 PM, Michael Avers <michael at mailworks.org> wrote:>> Hello,
>> 
>>  I'm looking at greenswitch as an alternative Python inbound ESL connector. Other than Italo's company (thanks for releasing this BTW), anyone else has seen success using it in a moderate to heavy call environment?>> 
>>  Project is at https://github.com/EvoluxBR/greenswitch
>> 
>>  Thanks
>>  Mike
>> 
>>  _________________________________________________________________________>>  Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>>  Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>>  FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users>> http://www.freeswitch.org
> ___________________________________________________________________________> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org

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From stefan.kainz at 1012.at  Mon Jul 31 09:28:00 2017
From: stefan.kainz at 1012.at (Stefan Kainz 1012)
Date: Mon, 31 Jul 2017 11:28:00 +0200
Subject: [Freeswitch-users] Reinvite Issue / no SDP
In-Reply-To: <CAEJMVkCzAxv5PENJfcP3YNrQfccJ-5ueg0Us3cYxtGeP5VGusA@mail.gmail.com>
References: <bc5e1b32-8b55-82ee-bd5a-9199460ca2d8@1012.at>
 <CAEJMVkCzAxv5PENJfcP3YNrQfccJ-5ueg0Us3cYxtGeP5VGusA@mail.gmail.com>
Message-ID: <3402e82d-fae6-6303-40c0-afef4ed30467@1012.at>

Thank you for your answer!

after investigating a few different cases, i came to the following 
conclusion:
Some PBX's seem to not to accept the SDP Offer in the OK Message when 
the Session Version ID is not increased from the SDP Version ID in the 
initial Call Setup.

Ever heard of such a behavior?
What would you suggest to get by this issue?

I also testet it on Version 1.6 and 1.6.17. Same issue here.

Thank you in advance!

regards,
Am 28.07.2017 um 15:55 schrieb Brian West:
> 1.4 has been EOL for quite some time, you should try the last 1.4 
> release and you should test on the latest 1.6 release, If the issue 
> persists collect the data per the reporting guidelines and file a JIRA.
>
> Thanks,
> /b
>
>
> On Thu, Jul 27, 2017 at 5:42 AM, Stefan Kainz 1012 
> <stefan.kainz at 1012.at <mailto:stefan.kainz at 1012.at>> wrote:
>
>     Hello,
>
>     We're using Freeswitch 1.4.18 on Debian 8.
>     We ran into a problem with internal call transfers, when the PBX
>     sends a
>     re-invite without SDP offer.
>
>     The Problem is as follows:
>
>     1) PSTN -> Freeswitch -> PBX -> Call is answered on extension 100,
>     Audio
>     in both directions
>     2) Extension 100 does an unattended transfer to extension 101
>     2.1) PBX sends a re-invite without SDP. Freeswitch offers SDP in
>     OK message
>     3) PBX acknoledges the OK message
>     4) Freeswitch is sending media as announced in SDP Offer, but PBX
>     doesnt
>     "receive" the audio. Also, no media from the PBX.
>
>     Since we're a little desperate, we already tried the following
>     settings:
>
>     1) 3pcc-enable true/false/proxy
>     2) inbound-late-negotiation is set to true
>     3) renegotiate-codec-on-reinvite true / false
>     4) disable-transfer true / false
>     5) enable-soa true / false ( it acutally does work when set to false,
>     but the call gets hangup after a few seconds of audio - ack timeout )
>
>     We also tried to enable bypass_media, so that freeswitch doesnt mess
>     with SDP.
>     So, the SDP Offer is generated by a Sonus Media Gateway - still no
>     luck.
>
>     Any of you had some experience with such behavior?
>     We would be very thankful for every advice!
>
>     have a nice day!
>
>     Stefan Kainz
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org <http://confluence.freeswitch.org>
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>     http://www.freeswitch.org
>
>
>
>
> -- 
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
> */Twitter: @FreeSWITCH , @briankwest/*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: 
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>

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From kkothari157 at gmail.com  Mon Jul 31 12:16:19 2017
From: kkothari157 at gmail.com (Ketan Kothari)
Date: Mon, 31 Jul 2017 17:46:19 +0530
Subject: [Freeswitch-users] DTMF not working from yealink
Message-ID: <CAGqqW9_Bf45_fZi09dAmkNkn_9D9B2oLQD_r04a8-o8LfFoOHQ@mail.gmail.com>

Hello All,

I have installed freeswitch 1.6.19 and trying to login in voicemail box
7777 that time DTMF is not working from my yealink and zoiper phone.

In same freeswitch when i called from Grandstrem phone its DTMF working
fine.

Please give me suggesiton to resolve DTMF issue from yealink.


Working_DTMF_Grandstream_logs
https://pastebin.com/DMnDKjrR


Not working_DTMF_Yealink_logs
https://pastebin.com/iYYi6Kta
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From gmaruzz at gmail.com  Mon Jul 31 12:27:04 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 31 Jul 2017 14:27:04 +0200
Subject: [Freeswitch-users] DTMF not working from yealink
In-Reply-To: <CAGqqW9_Bf45_fZi09dAmkNkn_9D9B2oLQD_r04a8-o8LfFoOHQ@mail.gmail.com>
References: <CAGqqW9_Bf45_fZi09dAmkNkn_9D9B2oLQD_r04a8-o8LfFoOHQ@mail.gmail.com>
Message-ID: <CALXCt0qrkKAeVvoYmzVHA514HtK5cHxPjp=c6bxUtfsWZBY7Og@mail.gmail.com>

check the dtmf settings into yealink and zoiper, and make them the same
like in the grandstream.

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT

On Jul 31, 2017 14:17, "Ketan Kothari" <kkothari157 at gmail.com> wrote:

> Hello All,
>
> I have installed freeswitch 1.6.19 and trying to login in voicemail box
> 7777 that time DTMF is not working from my yealink and zoiper phone.
>
> In same freeswitch when i called from Grandstrem phone its DTMF working
> fine.
>
> Please give me suggesiton to resolve DTMF issue from yealink.
>
>
> Working_DTMF_Grandstream_logs
> https://pastebin.com/DMnDKjrR
>
>
> Not working_DTMF_Yealink_logs
> https://pastebin.com/iYYi6Kta
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at gmail.com  Mon Jul 31 12:31:53 2017
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 31 Jul 2017 14:31:53 +0200
Subject: [Freeswitch-users] DTMF not working from yealink
In-Reply-To: <CALXCt0qrkKAeVvoYmzVHA514HtK5cHxPjp=c6bxUtfsWZBY7Og@mail.gmail.com>
References: <CAGqqW9_Bf45_fZi09dAmkNkn_9D9B2oLQD_r04a8-o8LfFoOHQ@mail.gmail.com>
 <CALXCt0qrkKAeVvoYmzVHA514HtK5cHxPjp=c6bxUtfsWZBY7Og@mail.gmail.com>
Message-ID: <CALXCt0owJ-ouz3kEHrJMPKxx77BVovE7DdfXjTbYRPbtOuDR4g@mail.gmail.com>

I looked at pastebin.

It has nothing to do with dtmf, your yealink is not registering.

Check user/password/domainname in yealink.

It is not accepted by FreeSWITCH


sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT

On Jul 31, 2017 2:27 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

> check the dtmf settings into yealink and zoiper, and make them the same
> like in the grandstream.
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
>
> On Jul 31, 2017 14:17, "Ketan Kothari" <kkothari157 at gmail.com> wrote:
>
>> Hello All,
>>
>> I have installed freeswitch 1.6.19 and trying to login in voicemail box
>> 7777 that time DTMF is not working from my yealink and zoiper phone.
>>
>> In same freeswitch when i called from Grandstrem phone its DTMF working
>> fine.
>>
>> Please give me suggesiton to resolve DTMF issue from yealink.
>>
>>
>> Working_DTMF_Grandstream_logs
>> https://pastebin.com/DMnDKjrR
>>
>>
>> Not working_DTMF_Yealink_logs
>> https://pastebin.com/iYYi6Kta
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
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From joelists at tm.net.uk  Mon Jul 31 14:15:40 2017
From: joelists at tm.net.uk (Jospeh Waite)
Date: Mon, 31 Jul 2017 15:15:40 +0100
Subject: [Freeswitch-users] Question about DNS handling on FS
Message-ID: <1021F9DF-3861-40EE-AFDD-5D914E860111@tm.net.uk>

Hi Guys

Got a question, I’m wondering if anyone can answer.

I am wondering how FS handles DNS resolution.

I assume that if I set the TTL on the DNS entry to say 30 seconds that FS will accept that value and do a DNS lookup every 30 seconds? Is that correct.

My second question is one a cll is in motion to a DNS defined endpoint, i.e. call bridged to a SIP endpoint by domain name, would it keep the IP for the call or the domain name? So would it re-DNS query when it comes to sending the BYE message at end of call if it was over 30 seconds, assuming the above 30sec TTL or would it keep the name/IP mapping for the whole call?

Regards

From mike at jerris.com  Mon Jul 31 15:10:40 2017
From: mike at jerris.com (Michael Jerris)
Date: Mon, 31 Jul 2017 15:10:40 +0000
Subject: [Freeswitch-users] Reinvite Issue / no SDP
In-Reply-To: <3402e82d-fae6-6303-40c0-afef4ed30467@1012.at>
References: <bc5e1b32-8b55-82ee-bd5a-9199460ca2d8@1012.at>
 <CAEJMVkCzAxv5PENJfcP3YNrQfccJ-5ueg0Us3cYxtGeP5VGusA@mail.gmail.com>
 <3402e82d-fae6-6303-40c0-afef4ed30467@1012.at>
Message-ID: <CAEwTsAi0pyksm8nGJrw5c7tEL=Xk1o1oitgF-rBLQABGx149aQ@mail.gmail.com>

does the sdp change but not change the version or is it 100% identical to
the previous we sent?

On Mon, Jul 31, 2017 at 5:28 AM Stefan Kainz 1012 <stefan.kainz at 1012.at>
wrote:

> Thank you for your answer!
>
> after investigating a few different cases, i came to the following
> conclusion:
> Some PBX's seem to not to accept the SDP Offer in the OK Message when the
> Session Version ID is not increased from the SDP Version ID in the initial
> Call Setup.
>
> Ever heard of such a behavior?
> What would you suggest to get by this issue?
>
> I also testet it on Version 1.6 and 1.6.17. Same issue here.
>
> Thank you in advance!
> regards,
> Am 28.07.2017 um 15:55 schrieb Brian West:
>
> 1.4 has been EOL for quite some time, you should try the last 1.4 release
> and you should test on the latest 1.6 release, If the issue persists
> collect the data per the reporting guidelines and file a JIRA.
>
> Thanks,
> /b
>
>
> On Thu, Jul 27, 2017 at 5:42 AM, Stefan Kainz 1012 <stefan.kainz at 1012.at>
> wrote:
>
>> Hello,
>>
>> We're using Freeswitch 1.4.18 on Debian 8.
>> We ran into a problem with internal call transfers, when the PBX sends a
>> re-invite without SDP offer.
>>
>> The Problem is as follows:
>>
>> 1) PSTN -> Freeswitch -> PBX -> Call is answered on extension 100, Audio
>> in both directions
>> 2) Extension 100 does an unattended transfer to extension 101
>> 2.1) PBX sends a re-invite without SDP. Freeswitch offers SDP in OK
>> message
>> 3) PBX acknoledges the OK message
>> 4) Freeswitch is sending media as announced in SDP Offer, but PBX doesnt
>> "receive" the audio. Also, no media from the PBX.
>>
>> Since we're a little desperate, we already tried the following settings:
>>
>> 1) 3pcc-enable true/false/proxy
>> 2) inbound-late-negotiation is set to true
>> 3) renegotiate-codec-on-reinvite true / false
>> 4) disable-transfer true / false
>> 5) enable-soa true / false ( it acutally does work when set to false,
>> but the call gets hangup after a few seconds of audio - ack timeout )
>>
>> We also tried to enable bypass_media, so that freeswitch doesnt mess
>> with SDP.
>> So, the SDP Offer is generated by a Sonus Media Gateway - still no luck.
>>
>> Any of you had some experience with such behavior?
>> We would be very thankful for every advice!
>>
>> have a nice day!
>>
>> Stefan Kainz
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From tculjaga at gmail.com  Mon Jul 31 15:42:48 2017
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 31 Jul 2017 17:42:48 +0200
Subject: [Freeswitch-users] TURN Configuration under CGN (RFC6598)
In-Reply-To: <CAEJMVkDKBGoBn8C9c1ny0x9yVfcD1w7XvmDC5x6Qq43E+xzOUg@mail.gmail.com>
References: <CAHKqMb6uw9fBDu=8_UwQ8LOhLxJKyjs2qwhdvyiLq_hxnPAv+Q@mail.gmail.com>
 <CAEJMVkCo=z+jgBPaOpDeOFqiNXdup77QDb6H_egE0bx9aRChbw@mail.gmail.com>
 <CAJoKKdW9k1VTtrf+mznmQ7j5cwtUwuq4-NdebkQ8YUfAUY27Tw@mail.gmail.com>
 <CAEJMVkDKBGoBn8C9c1ny0x9yVfcD1w7XvmDC5x6Qq43E+xzOUg@mail.gmail.com>
Message-ID: <CAJoKKdWp4YOqJqGpKHi86jQ9uY5+8+o6SL721A9kxqRDxfX4MQ@mail.gmail.com>

TURN helps in Symmetric NAT scenarios only

you should always try to get a direct connection first ( using STUN ) and
if everything else fails, you can use TURN.

to check your NAT type, you can use stunclient program from stunprotocol.org

e.g.:  /stunclient --mode full stun.ekiga.net 3479

On 31 July 2017 at 02:44, Brian West <brian at freeswitch.org> wrote:

> What issue are you trying to solve with turn?
>
> On Sat, Jul 29, 2017 at 1:42 AM Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
>
>>
>> FS doesn't use turn... but verto client can use it if you pass the
>> parameter in init function ... like that:
>>
>>    iceServers: [
>>       {url: 'turn:numb.viagenie.ca',credential: 'pass', username:
>> 'user'},
>>       {url: 'stun:ripslinger.undo.it',},
>>       {url: 'stun:stun.schlund.de',},
>>       {url: 'stun:stun.ekiga.net',},
>>       {url: 'stun:stun01.sipphone.com',},
>>    ],
>>
>>
>>
>>
>> On 28 July 2017 at 15:54, Brian West <brian at freeswitch.org> wrote:
>>
>>> TURN is not needed with FreeSWITCH, plus we do not support it.
>>>
>>> /b
>>>
>>>
>>> On Thu, Jul 27, 2017 at 8:55 PM, Nandy Dagondon <nandy1925 at gmail.com>
>>> wrote:
>>>
>>>> Hi folks,
>>>>
>>>> I just learned that our Internet provider implemented CGN when I'm
>>>> about to experiment WebRTC/mod_verto.  I already signed up a free TURN
>>>> server account at http://numb.viagenie.ca/ to proceed.   Where do I
>>>> place turn:numb.viagenie.ca in the config files? Is this correct?
>>>>
>>>> sip_profiles/external.xml:
>>>> <param name="ext-rtp-ip" value="turn:numb.viagenie.ca"/>
>>>> <param name="ext-sip-ip" value="turn:numb.viagenie.ca"/>
>>>>
>>>> /Nandy
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>> options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>> *Twitter: @FreeSWITCH , @briankwest*
>>>
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>>
>>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>>
>>> Allison prompts for FreeSWITCH:
>>>
>>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>>
>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>
>>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
>>> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
>>> *Skype:*briankwest
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
> *Twitter: @FreeSWITCH , @briankwest*
>
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> Book a phone call (CST) <https://freeswitch.com/appointment>
>
> Allison prompts for FreeSWITCH:
>
> *https://www.gofundme.com/allison-prompts-for-freeswitch*
> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
> *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Jul 31 17:42:16 2017
From: brian at freeswitch.org (Brian West)
Date: Mon, 31 Jul 2017 12:42:16 -0500
Subject: [Freeswitch-users] TURN Configuration under CGN (RFC6598)
In-Reply-To: <CAJoKKdWp4YOqJqGpKHi86jQ9uY5+8+o6SL721A9kxqRDxfX4MQ@mail.gmail.com>
References: <CAHKqMb6uw9fBDu=8_UwQ8LOhLxJKyjs2qwhdvyiLq_hxnPAv+Q@mail.gmail.com>
 <CAEJMVkCo=z+jgBPaOpDeOFqiNXdup77QDb6H_egE0bx9aRChbw@mail.gmail.com>
 <CAJoKKdW9k1VTtrf+mznmQ7j5cwtUwuq4-NdebkQ8YUfAUY27Tw@mail.gmail.com>
 <CAEJMVkDKBGoBn8C9c1ny0x9yVfcD1w7XvmDC5x6Qq43E+xzOUg@mail.gmail.com>
 <CAJoKKdWp4YOqJqGpKHi86jQ9uY5+8+o6SL721A9kxqRDxfX4MQ@mail.gmail.com>
Message-ID: <CAEJMVkCq3Y5xQP1p-gEFR8EvW_RFc_xHcX-2Yjn3Zvt34ceMFg@mail.gmail.com>

Still makes no sense with FreeSWITCH we already have sufficient NAT busting
capabilities.

On Mon, Jul 31, 2017 at 10:42 AM, Tihomir Culjaga <tculjaga at gmail.com>
wrote:

> TURN helps in Symmetric NAT scenarios only
>
> you should always try to get a direct connection first ( using STUN ) and
> if everything else fails, you can use TURN.
>
> to check your NAT type, you can use stunclient program from
> stunprotocol.org
>
> e.g.:  /stunclient --mode full stun.ekiga.net 3479
>
> On 31 July 2017 at 02:44, Brian West <brian at freeswitch.org> wrote:
>
>> What issue are you trying to solve with turn?
>>
>> On Sat, Jul 29, 2017 at 1:42 AM Tihomir Culjaga <tculjaga at gmail.com>
>> wrote:
>>
>>>
>>> FS doesn't use turn... but verto client can use it if you pass the
>>> parameter in init function ... like that:
>>>
>>>    iceServers: [
>>>       {url: 'turn:numb.viagenie.ca',credential: 'pass', username:
>>> 'user'},
>>>       {url: 'stun:ripslinger.undo.it',},
>>>       {url: 'stun:stun.schlund.de',},
>>>       {url: 'stun:stun.ekiga.net',},
>>>       {url: 'stun:stun01.sipphone.com',},
>>>    ],
>>>
>>>
>>>
>>>
>>> On 28 July 2017 at 15:54, Brian West <brian at freeswitch.org> wrote:
>>>
>>>> TURN is not needed with FreeSWITCH, plus we do not support it.
>>>>
>>>> /b
>>>>
>>>>
>>>> On Thu, Jul 27, 2017 at 8:55 PM, Nandy Dagondon <nandy1925 at gmail.com>
>>>> wrote:
>>>>
>>>>> Hi folks,
>>>>>
>>>>> I just learned that our Internet provider implemented CGN when I'm
>>>>> about to experiment WebRTC/mod_verto.  I already signed up a free TURN
>>>>> server account at http://numb.viagenie.ca/ to proceed.   Where do I
>>>>> place turn:numb.viagenie.ca in the config files? Is this correct?
>>>>>
>>>>> sip_profiles/external.xml:
>>>>> <param name="ext-rtp-ip" value="turn:numb.viagenie.ca"/>
>>>>> <param name="ext-sip-ip" value="turn:numb.viagenie.ca"/>
>>>>>
>>>>> /Nandy
>>>>>
>>>>> ____________________________________________________________
>>>>> _____________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>>> freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Brian West*
>>>> brian at freeswitch.org
>>>>
>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>
>>>> http://www.freeswitchbook.com
>>>> http://www.freeswitchcookbook.com
>>>>
>>>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>>>
>>>> Allison prompts for FreeSWITCH:
>>>>
>>>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>>>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>>>
>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>>>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>
>>>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
>>>> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
>>>> *Skype:*briankwest
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> ____________________________________________________________
>>> _____________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>>
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> Book a phone call (CST) <https://freeswitch.com/appointment>
>>
>> Allison prompts for FreeSWITCH:
>>
>> *https://www.gofundme.com/allison-prompts-for-freeswitch*
>> <https://www.gofundme.com/allison-prompts-for-freeswitch>
>>
>> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
>> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>
>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002
>> <+1%20918-420-9002> | *M:*+1918424WEST (9378)
>> *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
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From josedavid at zennio.com  Mon Jul 31 06:36:45 2017
From: josedavid at zennio.com (Jose David Jurado Alonso)
Date: Mon, 31 Jul 2017 08:36:45 +0200
Subject: [Freeswitch-users] FS AWS does not call between mobile phones
 connected to the same WiFi
In-Reply-To: <CALXCt0peLF_nxPfZUVBn2LUsRGyduKkOXm9J4otAD=rqQ49Bhw@mail.gmail.com>
References: <CANeCd7rnLuNQChX=iUN6_2i-+13XSeQzXnQ0aJ_U36trt9F-vQ@mail.gmail.com>
 <CALXCt0peLF_nxPfZUVBn2LUsRGyduKkOXm9J4otAD=rqQ49Bhw@mail.gmail.com>
Message-ID: <CANeCd7riu76oyQq4Q5Q1puXNtMj9CcAffPb=FM8U14TXrLV=qQ@mail.gmail.com>

Hi Giovanni,

I have tried with the change you mention, changing the domain by the public
IP, but it still does not work.

Both devices are connected to the same WiFi, on a TP-Link Archer C1200
router, and in the ALG section I do not see the SIP option.

By connecting one of the devices to this router and the other to another
network, the call is successful. The problem is when the two are connected
to the same network (I have tested with several different networks).

It's as if FS does not find the destination number and stays waiting. Both
devices are correctly registered and can be called or called by any device
as long as it is not on the same network.





2017-07-28 18:05 GMT+02:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:

> On 28 July 2017 at 20:21, Jose David Jurado Alonso <josedavid at zennio.com>
> wrote:
>
>> My current configuration is the next:
>>
>> vars.xml:
>>
>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:mydomain.us"/>
>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:mydomain.us"/>
>>
>
>
> try making it:
>
> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=135.44.55.123"/>
> <X-PRE-PROCESS cmd="set" data="external_sip_ip=135.44.55.123"/>
>
> Stop and start FreeSeWITCH and see if it helps
>
> Also, maybe something is blocked by some ALG in the clients' router? Try
> to connect via TLS...
>
> Hope this help,
>
> -giovanni
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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> FreeSWITCH-users at lists.freeswitch.org
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