From mike at jerris.com  Fri Apr  1 00:01:31 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 31 Mar 2016 16:01:31 -0400
Subject: [Freeswitch-users] codec changed
In-Reply-To: <CAFETSWgdUxQeeFA++NrkciPUP_+d312O1qa-Npv1kJfRBtng3Q@mail.gmail.com>
References: <CAFETSWig3dQ3viXCo9g+sXp_FzMrajH3G1nY7nO+AsBbumCNBw@mail.gmail.com>
	<CAEwTsAgrTDQpZghDfoKvf=EhF=7GJXtn2V5n33nMazEL-cU4dw@mail.gmail.com>
	<CAFETSWgdUxQeeFA++NrkciPUP_+d312O1qa-Npv1kJfRBtng3Q@mail.gmail.com>
Message-ID: <68C49814-E6AD-4B14-810E-4097EBC03E86@jerris.com>

https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation <https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation>

> On Mar 31, 2016, at 10:31 AM, amani mansour <amani.mansour2 at gmail.com> wrote:
> 
> sir i will call a FS(extension 600) from my sip phone(extension 423)  i used a gateway  when the call is ringing i will hear a short msg ,
> this msg  is a pcap file  ,when i changed it to wav file automatically in freeswitch it has  changed to the first codec selected in my soft phone :
> 
> my pcap file has us codec g711
> my phoner lite i have selected 1/opus 2/ g711A 3./.........
> 
> but when i play this wav  i have in tshark this line:
> 8533 960.577294  192.168.3.5 -> 192.168.3.1  RTP 109 PT=opus, SSRC=0x57A054EC, Seq=2363, Time=424320
> 
> my dialplan/default.xml:
> <extension name="600">
>        <condition field="caller_id_number" expression="^(10[01][0-9]|1500|423|400|1011)$">
>             <action application="set" data="continue_on_fail=true"/>
>             <action application="ring_ready"/>
>             <action application="set" data="ringback=${us-ring}"/>
>             <action application="set" data="transfer_ringback=$${hold_music}"/>
>             <action application="set" data="ignore_early_media=false"/>
>             <action application="set" data="hangup_after_bridge=true"/>
>             <action application="bridge"  data="[leg_timeout=20,absolute_codec_string=PCMA ]sofia/gateway/gateway_to_600/600"/>
>             <action application="answer"/>
> 
>       </condition>
>   </extension>
> 
>      <extension name="600">
>        <condition field="destination_number" expression="^(600)$">
>              <action application="export" data="nolocal:codec_string=PCMA"/> 
>             <action application="answer"/>
>            <action application="info"/>
>             <action application="sleep" data="1000"/>
>             <action application="playback" data="/home/amani/projet/PFE/gg711.pcap.0xd2bd4e3e.wav"/>
>          </condition>
> </extension>
> 
> 
> 
> i have us tcpdump :
> 14:59:31.659246 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: INVITE sip:600 at 192.168.3.5 <mailto:sip%3A600 at 192.168.3.5> SIP/2.0
> 14:59:31.659773 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 100 Trying
> 14:59:31.661679 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 407 Proxy Authentication Required
> 14:59:31.662289 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: ACK sip:600 at 192.168.3.5 <mailto:sip%3A600 at 192.168.3.5> SIP/2.0
> 14:59:31.662726 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: INVITE sip:600 at 192.168.3.5 <mailto:sip%3A600 at 192.168.3.5> SIP/2.0
> 14:59:31.663177 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 100 Trying
> 14:59:38.508070 IP 192.168.3.5.sip > 192.168.3.1.35200: Flags [.], ack 2002, win 789, length 0
> 14:59:38.508188 IP 192.168.3.1.35200 > 192.168.3.5.sip: Flags [.], ack 2260, win 253, length 0
> 14:59:41.823972 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 180 Ringing
> 14:59:52.073381 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 183 Session Progress
> 15:00:02.027890 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: SIP/2.0 200 OK
> 15:00:02.028681 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: ACK sip:600 at 192.168.3.5:5060;transport=udp SIP/2.0
> 15:00:02.034884 IP 192.168.3.5.sip > 192.168.3.1.sip: SIP: BYE sip:423 at 192.168.3.1:5060 <http://sip:423 at 192.168.3.1:5060/> SIP/2.0
> 15:00:02.035367 IP 192.168.3.1.sip > 192.168.3.5.sip: SIP: SIP/2.0 200 OK
> 
> my supervisor tel me that when i change the file to .wav  nrmally FS code and decode the file using unknown codec
> can you help me  please sir ???
> 
> 
> 
> best regards 
> 
> amani 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vfclists at gmail.com  Fri Apr  1 00:20:35 2016
From: vfclists at gmail.com (vfclists .)
Date: Thu, 31 Mar 2016 21:20:35 +0100
Subject: [Freeswitch-users] How to use $0 in dialplan?
In-Reply-To: <CAOrA2Ub_Dx3-9TDcMY3-+2ormd_u61OeckUpy5q3yOfMfTtOAQ@mail.gmail.com>
References: <CAE3mkNDmqegSGQsbpFbgeXQACvs9RN-KP0EbGJQgBwkv5Fn9Sg@mail.gmail.com>
	<CAOrA2Ub_Dx3-9TDcMY3-+2ormd_u61OeckUpy5q3yOfMfTtOAQ@mail.gmail.com>
Message-ID: <CAE3mkNAAmeAOLiWoeQGNdup39gTtMGntfydHSYCQbSHYKmOGiA@mail.gmail.com>

On 28 March 2016 at 12:26, Jurijs Ivolga <jurijs.ivolga at gmail.com> wrote:

> Hi,
>
> As far as I understood you need to prefix exact number with 240 and send
> it out to provider.
>
> Please try following:
>
> <!-- route 1234 numbers through trunkxxx ->
>     <condition field="destination_number"  break="on-true"
>           expression="^(1234567890$)">
>           <action application="bridge"
> data="sofia/gateway/trunkxxx/240$1"/>
>     </condition>
>
> In this case only 1234567890 numbers will satisfy this condition and you
> will sand out to provider 240123456790
>
> With kind regards,
>
> Jurijs
>
>
Thanks, it works

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



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From vfclists at gmail.com  Fri Apr  1 01:38:08 2016
From: vfclists at gmail.com (vfclists .)
Date: Thu, 31 Mar 2016 22:38:08 +0100
Subject: [Freeswitch-users] Can mod_cdr_csv write to multiple files?
Message-ID: <CAE3mkNAmj4upASYehd4YioKtQGJyniri2YnTScCPCO4Y4Lmn-A@mail.gmail.com>

The cdr_csv.conf uses the default-template option to select the output
format for the csv file. <param name="default-template" value="example"/>

Is there some option for multiple files, such as
generating output for Postgres and say Linksys?

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From nneul at mst.edu  Fri Apr  1 01:41:22 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 31 Mar 2016 16:41:22 -0500
Subject: [Freeswitch-users] Can mod_cdr_csv write to multiple files?
In-Reply-To: <CAE3mkNAmj4upASYehd4YioKtQGJyniri2YnTScCPCO4Y4Lmn-A@mail.gmail.com>
References: <CAE3mkNAmj4upASYehd4YioKtQGJyniri2YnTScCPCO4Y4Lmn-A@mail.gmail.com>
Message-ID: <56FD9982.9040104@mst.edu>

Unknown on the csv module, but the mod_format_cdr module allows for multiple profiles to be defined sending to different 
files or http endpoints.

-- Nathan

On 03/31/2016 04:38 PM, vfclists . wrote:
> The cdr_csv.conf uses the default-template option to select the output
> format for the csv file. <param name="default-template" value="example"/>
>
> Is there some option for multiple files, such as
> generating output for Postgres and say Linksys?
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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


From anthony.minessale at gmail.com  Fri Apr  1 01:54:01 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 31 Mar 2016 16:54:01 -0500
Subject: [Freeswitch-users] Can mod_cdr_csv write to multiple files?
In-Reply-To: <56FD9982.9040104@mst.edu>
References: <CAE3mkNAmj4upASYehd4YioKtQGJyniri2YnTScCPCO4Y4Lmn-A@mail.gmail.com>
	<56FD9982.9040104@mst.edu>
Message-ID: <CAKbxfG-KO7Kp86_jbwWtd5_XxniUMbTEisWNn18vudgmNGFSjQ@mail.gmail.com>

if you set the channel variable accountcode then it will create a matching
seperate file with that name.
If you name the accountcode exactly after one of the loaded template names
it will use that template AND write it into a matching file.






On Thu, Mar 31, 2016 at 4:41 PM, Nathan Neulinger <nneul at mst.edu> wrote:

> Unknown on the csv module, but the mod_format_cdr module allows for
> multiple profiles to be defined sending to different
> files or http endpoints.
>
> -- Nathan
>
> On 03/31/2016 04:38 PM, vfclists . wrote:
> > The cdr_csv.conf uses the default-template option to select the output
> > format for the csv file. <param name="default-template" value="example"/>
> >
> > Is there some option for multiple files, such as
> > generating output for Postgres and say Linksys?
> >
> > --
> > Frank Church
> >
> > =======================
> > http://devblog.brahmancreations.com
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> --
> ------------------------------------------------------------
> 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
>



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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From mervyn.yeo at gmail.com  Fri Apr  1 09:42:12 2016
From: mervyn.yeo at gmail.com (Mervyn Yeo)
Date: Fri, 1 Apr 2016 13:42:12 +0800
Subject: [Freeswitch-users] Google Voice outbound calls fail with
	DESTINATION_OUT_OF_ORDER
Message-ID: <CA+cDheNL0ZwrKgjLEFXLfm2O+ioLZA82ks7XFQjkSoTOm1jmeg@mail.gmail.com>

Hi all,

I got my GV (Google Voice) setup following the wiki and inbound calls works
fine. It's the outbound calls that dies immediately after dialing. I've
enabled console and dingaling logging and attached the logs here. I've also
attached various configurations related to GV setup. If you see something
misconfigured or know what could be the cause of the problem please do let
me know.

client.xml - jingle_profiles/client.xml
default - outbound dialplan. part of default.xml
dingaling.conf.xml - as the name implies
inbound.xml - inbound dialplan
gv.log - console log

Thanks,
Mervyn
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From mervyn.yeo at gmail.com  Fri Apr  1 11:48:45 2016
From: mervyn.yeo at gmail.com (Mervyn Yeo)
Date: Fri, 1 Apr 2016 15:48:45 +0800
Subject: [Freeswitch-users] Google Voice outbound calls fail with
	DESTINATION_OUT_OF_ORDER
In-Reply-To: <CA+cDheNL0ZwrKgjLEFXLfm2O+ioLZA82ks7XFQjkSoTOm1jmeg@mail.gmail.com>
References: <CA+cDheNL0ZwrKgjLEFXLfm2O+ioLZA82ks7XFQjkSoTOm1jmeg@mail.gmail.com>
Message-ID: <CA+cDhePcdP3A4K_i3u8qZUUDENKysMP7nOiQZWz8bXaom5gB9Q@mail.gmail.com>

I forgot to mention, I'm running freeswitch branch 1.6 build from source.

On Fri, Apr 1, 2016 at 1:42 PM, Mervyn Yeo <mervyn.yeo at gmail.com> wrote:

> Hi all,
>
> I got my GV (Google Voice) setup following the wiki and inbound calls
> works fine. It's the outbound calls that dies immediately after dialing.
> I've enabled console and dingaling logging and attached the logs here. I've
> also attached various configurations related to GV setup. If you see
> something misconfigured or know what could be the cause of the problem
> please do let me know.
>
> client.xml - jingle_profiles/client.xml
> default - outbound dialplan. part of default.xml
> dingaling.conf.xml - as the name implies
> inbound.xml - inbound dialplan
> gv.log - console log
>
> Thanks,
> Mervyn
>
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From rutu.patel at inextrix.com  Fri Apr  1 13:54:50 2016
From: rutu.patel at inextrix.com (Rutu Patel)
Date: Fri, 1 Apr 2016 15:24:50 +0530
Subject: [Freeswitch-users] 480 Temporarily
	Unavailable[MANDATORY_IE_MISSING]
In-Reply-To: <CAOrA2Ub=NFpdsQZxsDX2BhmoX-8DLtMFHBBQeJXQtrKSeq6m8A@mail.gmail.com>
References: <CAPQet1zRj3sogx3csggUCe9zkeCLN2X=99fr=z73WkLvD5=bPA@mail.gmail.com>
	<CAOrA2UYZTKucCxUC3XbB=yG_Z9ZOTOZpRH_MvrgLHhqoHnVc8g@mail.gmail.com>
	<CAKbxfG_BacH6FQqEPba7w6UTHbKhQsw+7UX9=pCgvr7mo1t=XA@mail.gmail.com>
	<CAPQet1zPPUVHSpNyUVJVvj_df7qfq+ty_yYatuBoTPboaQ0F+Q@mail.gmail.com>
	<CAKzWOxXUCNjRXGy=UPDNSQyZUx8i5eJ-iw9Xs4gYKdJR5RA3aA@mail.gmail.com>
	<CAPQet1wsOwEC3hWFqKns0-0gTOkMEs94B6D8mxp8HwyGsWqPHg@mail.gmail.com>
	<CAOrA2Ub=NFpdsQZxsDX2BhmoX-8DLtMFHBBQeJXQtrKSeq6m8A@mail.gmail.com>
Message-ID: <CAPQet1z+=0rw=VO9Q-b95n4B_Bk-R6VEZvrM5VE5sMd-u4dTgQ@mail.gmail.com>

Hi Juris,

Yes we added peer in asterisk sip.conf.

Are we still missing any configurations?

On Thursday, March 31, 2016, Jurijs Ivolga <jurijs.ivolga at gmail.com> wrote:

> Hi Rutu,
>
> Just a shot in a dark. :)
>
> Did you added sip peer record in sip.conf, something similar to:
>
> [freeswitch]
> type=peer
> host=IP_ADDRESS_OF_FREESWITCH_SERVER
> username=HOSTNAME.DOMAIN.COM
> port=5080
> fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER
> secret=BOOTH_WAY_PASSWORD
>
> With kind regards,
>
>
> Jurijs
>
> On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <rutu.patel at inextrix.com
> <javascript:_e(%7B%7D,'cvml','rutu.patel at inextrix.com');>> wrote:
>
>> Hi Michael
>>
>> Thank you for your assistance.
>>
>> We are trying to make call between two freeswitch users.
>>
>> One user is registered in softphone and other is registered in Asterisk
>> by register string but asterisk user cannot authenticate properly and
>> freeswitch directly send 480 Temporary Unavailable.
>>
>> In sip profile we set challenge-realm= auto_to also tried with auto_from
>> but still not getting success.
>>
>> I am attaching some more logs if anyone can guide in right direction.
>>
>> --
>> Thanks,
>> Rutu Patel
>>
>>
>>
>> On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','msc at freeswitch.org');>> wrote:
>>
>>> Try a gateway. Look in conf/sip_profiles/external/example.xml for a
>>> heavily-commented sample. For more detailed discussion see this nice doc:
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration
>>>
>>> -MSC
>>>
>>> On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel at inextrix.com
>>> <javascript:_e(%7B%7D,'cvml','rutu.patel at inextrix.com');>> wrote:
>>>
>>>> Hi,
>>>>
>>>> Thank you for reply.
>>>>
>>>> We have registered extension of Freeswitch in Asterisk as below
>>>>
>>>> register=>test-ext:secret at 192.168.1.64/test-ext
>>>>
>>>> ?We want to send call from Freeswitch to Asterisk?.
>>>>
>>>> What authentication parameters we need to set in Freeswitch ?
>>>>
>>>> --
>>>> Thanks,
>>>> Rutu Patel
>>>> On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com
>>>> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>> wrote:
>>>>
>>>>> You need to supply auth credentials to Asterisk because its set to
>>>>> authenticate calls.
>>>>> The best thing to do would be to make a gateway pointing to Asterisk
>>>>> with all the credentials and call it using that.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga <
>>>>> jurijs.ivolga at gmail.com
>>>>> <javascript:_e(%7B%7D,'cvml','jurijs.ivolga at gmail.com');>> wrote:
>>>>>
>>>>>> Hi Rutu,
>>>>>>
>>>>>> In logs I can see that user-agent Asterisk is sending INVITE to
>>>>>> Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This
>>>>>> a bit differs from what you explained in your mail...
>>>>>>
>>>>>> First think what pop up in my mind is that you need to configure
>>>>>> Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my
>>>>>> first guess...
>>>>>>
>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
>>>>>>
>>>>>> With kind regards,
>>>>>>
>>>>>> Jurijs
>>>>>>
>>>>>> On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel at inextrix.com
>>>>>> <javascript:_e(%7B%7D,'cvml','rutu.patel at inextrix.com');>> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> We are sending call to asterisk PBX but we are getting 480
>>>>>>> Temporarily Unavailable immediately. from freeswitch server.
>>>>>>>
>>>>>>> Call flow is as below:
>>>>>>>
>>>>>>> User => opensips => FS => PBX,
>>>>>>>
>>>>>>> but call not sending out from FS and drop by FS.
>>>>>>>
>>>>>>> Here I am attaching both sip logs and freeswitch logs.
>>>>>>>
>>>>>>> Can anyone please help to sort out the issue.
>>>>>>> --
>>>>>>> Thanks,
>>>>>>> Rutu Patel
>>>>>>> <http://www.inextrix.com>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> 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
>>>>> <javascript:_e(%7B%7D,'cvml','sip:888 at conference.freeswitch.org');>
>>>>>  ? +19193869900
>>>>>
>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>

-- 
--
Thanks,
Rutu Patel
iNextrix Technologies Pvt. Ltd.
www.inextrix.com
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From jurijs.ivolga at gmail.com  Fri Apr  1 14:26:33 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Fri, 1 Apr 2016 13:26:33 +0300
Subject: [Freeswitch-users] 480 Temporarily
	Unavailable[MANDATORY_IE_MISSING]
In-Reply-To: <CAPQet1z+=0rw=VO9Q-b95n4B_Bk-R6VEZvrM5VE5sMd-u4dTgQ@mail.gmail.com>
References: <CAPQet1zRj3sogx3csggUCe9zkeCLN2X=99fr=z73WkLvD5=bPA@mail.gmail.com>
	<CAOrA2UYZTKucCxUC3XbB=yG_Z9ZOTOZpRH_MvrgLHhqoHnVc8g@mail.gmail.com>
	<CAKbxfG_BacH6FQqEPba7w6UTHbKhQsw+7UX9=pCgvr7mo1t=XA@mail.gmail.com>
	<CAPQet1zPPUVHSpNyUVJVvj_df7qfq+ty_yYatuBoTPboaQ0F+Q@mail.gmail.com>
	<CAKzWOxXUCNjRXGy=UPDNSQyZUx8i5eJ-iw9Xs4gYKdJR5RA3aA@mail.gmail.com>
	<CAPQet1wsOwEC3hWFqKns0-0gTOkMEs94B6D8mxp8HwyGsWqPHg@mail.gmail.com>
	<CAOrA2Ub=NFpdsQZxsDX2BhmoX-8DLtMFHBBQeJXQtrKSeq6m8A@mail.gmail.com>
	<CAPQet1z+=0rw=VO9Q-b95n4B_Bk-R6VEZvrM5VE5sMd-u4dTgQ@mail.gmail.com>
Message-ID: <CAOrA2UZEYXX_bdThsjcqjdUXqtR9oHMjkr2k-FT0WAU-O9Y=9g@mail.gmail.com>

Hi,

I'm not sure about your environment, but on this point I will recommend you
to isolate issue. So on this point you should understand if your sip trunk
between Asterisk and Freeswitch configured correctly, without involving
Opensips and etc. To test SIP trunk you can try to send call directly from
freeswitch to Asterisk and see if it works, you can send call just directly
from freeswitch console, something similar to:

originate sofia/example/300 at foo.com 8600

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

If you call will reach Asterisk without any issue, then this means that sip
trunk created correctly and issue is somewhere on Opensips side or any
other part of your set-up.

Second option will be to try to configure sip trunk without password and
check if it helps

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

With kind regards,

Jurijs

On Fri, Apr 1, 2016 at 12:54 PM, Rutu Patel <rutu.patel at inextrix.com> wrote:

> Hi Juris,
>
> Yes we added peer in asterisk sip.conf.
>
> Are we still missing any configurations?
>
>
> On Thursday, March 31, 2016, Jurijs Ivolga <jurijs.ivolga at gmail.com>
> wrote:
>
>> Hi Rutu,
>>
>> Just a shot in a dark. :)
>>
>> Did you added sip peer record in sip.conf, something similar to:
>>
>> [freeswitch]
>> type=peer
>> host=IP_ADDRESS_OF_FREESWITCH_SERVER
>> username=HOSTNAME.DOMAIN.COM
>> port=5080
>> fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER
>> secret=BOOTH_WAY_PASSWORD
>>
>> With kind regards,
>>
>>
>> Jurijs
>>
>> On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <rutu.patel at inextrix.com>
>> wrote:
>>
>>> Hi Michael
>>>
>>> Thank you for your assistance.
>>>
>>> We are trying to make call between two freeswitch users.
>>>
>>> One user is registered in softphone and other is registered in Asterisk
>>> by register string but asterisk user cannot authenticate properly and
>>> freeswitch directly send 480 Temporary Unavailable.
>>>
>>> In sip profile we set challenge-realm= auto_to also tried with auto_from
>>> but still not getting success.
>>>
>>> I am attaching some more logs if anyone can guide in right direction.
>>>
>>> --
>>> Thanks,
>>> Rutu Patel
>>>
>>>
>>>
>>> On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>>
>>>> Try a gateway. Look in conf/sip_profiles/external/example.xml for a
>>>> heavily-commented sample. For more detailed discussion see this nice doc:
>>>>
>>>> https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration
>>>>
>>>> -MSC
>>>>
>>>> On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel at inextrix.com>
>>>> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> Thank you for reply.
>>>>>
>>>>> We have registered extension of Freeswitch in Asterisk as below
>>>>>
>>>>> register=>test-ext:secret at 192.168.1.64/test-ext
>>>>>
>>>>> ?We want to send call from Freeswitch to Asterisk?.
>>>>>
>>>>> What authentication parameters we need to set in Freeswitch ?
>>>>>
>>>>> --
>>>>> Thanks,
>>>>> Rutu Patel
>>>>> On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale <
>>>>> anthony.minessale at gmail.com> wrote:
>>>>>
>>>>>> You need to supply auth credentials to Asterisk because its set to
>>>>>> authenticate calls.
>>>>>> The best thing to do would be to make a gateway pointing to Asterisk
>>>>>> with all the credentials and call it using that.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga <
>>>>>> jurijs.ivolga at gmail.com> wrote:
>>>>>>
>>>>>>> Hi Rutu,
>>>>>>>
>>>>>>> In logs I can see that user-agent Asterisk is sending INVITE to
>>>>>>> Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This
>>>>>>> a bit differs from what you explained in your mail...
>>>>>>>
>>>>>>> First think what pop up in my mind is that you need to configure
>>>>>>> Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my
>>>>>>> first guess...
>>>>>>>
>>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
>>>>>>>
>>>>>>> With kind regards,
>>>>>>>
>>>>>>> Jurijs
>>>>>>>
>>>>>>> On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <rutu.patel at inextrix.com
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>> We are sending call to asterisk PBX but we are getting 480
>>>>>>>> Temporarily Unavailable immediately. from freeswitch server.
>>>>>>>>
>>>>>>>> Call flow is as below:
>>>>>>>>
>>>>>>>> User => opensips => FS => PBX,
>>>>>>>>
>>>>>>>> but call not sending out from FS and drop by FS.
>>>>>>>>
>>>>>>>> Here I am attaching both sip logs and freeswitch logs.
>>>>>>>>
>>>>>>>> Can anyone please help to sort out the issue.
>>>>>>>> --
>>>>>>>> Thanks,
>>>>>>>> Rutu Patel
>>>>>>>> <http://www.inextrix.com>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>>>>
>>>>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>>>>> http://twitter.com/FreeSWITCH
>>>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>>>>>> <http://freeswitch.org/g+>*
>>>>>>
>>>>>> ClueCon Weekly Development Call
>>>>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>>>>
>>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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,
> Rutu Patel
> iNextrix Technologies Pvt. Ltd.
> www.inextrix.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rajib.dey at mindteck.com  Fri Apr  1 12:50:03 2016
From: rajib.dey at mindteck.com (Rajib Dey)
Date: Fri, 01 Apr 2016 14:20:03 +0530
Subject: [Freeswitch-users] How to handle invite session from pjsua
Message-ID: <000001d18bf3$7a69e890$6f3db9b0$@dey@mindteck.com>

Hello,

 

I am using pjsua sample application and I want to handle the invite sessions
from my own logic. I need access/modify remote and local sdp sessions,
endpoint and socket information. Is there any API available in pjsua that I
can use to achieve this?

 

Thanks & Regards,

Rajib Dey

 

 

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From saurabhkv01 at gmail.com  Fri Apr  1 17:48:39 2016
From: saurabhkv01 at gmail.com (saurabh verrma)
Date: Fri, 1 Apr 2016 19:18:39 +0530
Subject: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
Message-ID: <CACot2ntTB+7OXvomm3sxMzxnNEnT6vHtQb=T=m8MTOFT0=zFZg@mail.gmail.com>

Hi All,

I want to use FreeSWITCH library for DTLS/SRTP support. I want to know in
freeswitch which library has the support of these features(SRTP/DTLS) ?
Is there any application available based on freeswitch library ?

Any help would be appreciable.

-- 
*With Warm Regards:*
*Saurabh Kumar Verma*
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From mike at jerris.com  Fri Apr  1 17:59:27 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 1 Apr 2016 09:59:27 -0400
Subject: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
In-Reply-To: <CACot2ntTB+7OXvomm3sxMzxnNEnT6vHtQb=T=m8MTOFT0=zFZg@mail.gmail.com>
References: <CACot2ntTB+7OXvomm3sxMzxnNEnT6vHtQb=T=m8MTOFT0=zFZg@mail.gmail.com>
Message-ID: <E93C0E00-0EEE-44E1-81F2-1E8CF374057B@jerris.com>

we have full support for webrtc media profile which would include these features  There are not a ton of people who use freeswitch as a library, but it is built that way so that you can control it and host it in another application instead of stand alone.  If you are trying to accomplish something I'd try to handle it standalone first so you can learn all the different ways you might control it before architecting a solution

> On Apr 1, 2016, at 9:48 AM, saurabh verrma <saurabhkv01 at gmail.com> wrote:
> 
> Hi All, 
> 
> I want to use FreeSWITCH library for DTLS/SRTP support. I want to know in freeswitch which library has the support of these features(SRTP/DTLS) ?
> Is there any application available based on freeswitch library ?
> 
> Any help would be appreciable.
> 
> -- 
> With Warm Regards:
> Saurabh Kumar Verma
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  1 18:00:04 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 1 Apr 2016 10:00:04 -0400
Subject: [Freeswitch-users] How to handle invite session from pjsua
In-Reply-To: <56fe7e07.aa4f320a.c4cfb.ffffc137SMTPIN_ADDED_BROKEN@mx.google.com>
References: <56fe7e07.aa4f320a.c4cfb.ffffc137SMTPIN_ADDED_BROKEN@mx.google.com>
Message-ID: <A5C65112-6ED0-4921-9034-7AEF37A78B3C@jerris.com>

maybe better to ask them on their mailing list?

> On Apr 1, 2016, at 4:50 AM, Rajib Dey <rajib.dey at mindteck.com> wrote:
> 
> Hello,
>  
> I am using pjsua sample application and I want to handle the invite sessions from my own logic. I need access/modify remote and local sdp sessions, endpoint and socket information. Is there any API available in pjsua that I can use to achieve this?
>  
> Thanks & Regards,
> Rajib Dey
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From vfclists at gmail.com  Fri Apr  1 23:08:26 2016
From: vfclists at gmail.com (Frank Church)
Date: Fri, 1 Apr 2016 20:08:26 +0100
Subject: [Freeswitch-users] Google Voice outbound calls fail with
 DESTINATION_OUT_OF_ORDER
In-Reply-To: <CA+cDheNL0ZwrKgjLEFXLfm2O+ioLZA82ks7XFQjkSoTOm1jmeg@mail.gmail.com>
References: <CA+cDheNL0ZwrKgjLEFXLfm2O+ioLZA82ks7XFQjkSoTOm1jmeg@mail.gmail.com>
Message-ID: <56FEC72A.2040101@gmail.com>

On 01/04/2016 06:42, Mervyn Yeo wrote:
> Hi all,
>
> I got my GV (Google Voice) setup following the wiki and inbound calls 
> works fine. It's the outbound calls that dies immediately after 
> dialing. I've enabled console and dingaling logging and attached the 
> logs here. I've also attached various configurations related to GV 
> setup. If you see something misconfigured or know what could be the 
> cause of the problem please do let me know.
>
> client.xml - jingle_profiles/client.xml
> default - outbound dialplan. part of default.xml
> dingaling.conf.xml - as the name implies
> inbound.xml - inbound dialplan
> gv.log - console log
>
> Thanks,
> Mervyn
>
>
I  think the preferred method here is put the files in pastebin and link 
to it so others don't have to go through the process of downloading and 
extracting before they can view them. You must sanitize if before.
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From rtreleaven at bunnykick.ca  Fri Apr  1 23:34:16 2016
From: rtreleaven at bunnykick.ca (Russell Treleaven)
Date: Fri, 1 Apr 2016 15:34:16 -0400
Subject: [Freeswitch-users] Google Voice outbound calls fail with
	DESTINATION_OUT_OF_ORDER
In-Reply-To: <56FEC72A.2040101@gmail.com>
References: <CA+cDheNL0ZwrKgjLEFXLfm2O+ioLZA82ks7XFQjkSoTOm1jmeg@mail.gmail.com>
	<56FEC72A.2040101@gmail.com>
Message-ID: <CAHzfiZ5D_Bw_Fj6oCGcH1ANGgZneYC8V+_vkcyqRBvA-698x+Q@mail.gmail.com>

Here is an alternative.

https://simonics.com/services/



On Fri, Apr 1, 2016 at 3:08 PM, Frank Church <vfclists at gmail.com> wrote:

> On 01/04/2016 06:42, Mervyn Yeo wrote:
>
> Hi all,
>
> I got my GV (Google Voice) setup following the wiki and inbound calls
> works fine. It's the outbound calls that dies immediately after dialing.
> I've enabled console and dingaling logging and attached the logs here. I've
> also attached various configurations related to GV setup. If you see
> something misconfigured or know what could be the cause of the problem
> please do let me know.
>
> client.xml - jingle_profiles/client.xml
> default - outbound dialplan. part of default.xml
> dingaling.conf.xml - as the name implies
> inbound.xml - inbound dialplan
> gv.log - console log
>
> Thanks,
> Mervyn
>
>
> I  think the preferred method here is put the files in pastebin and link
> to it so others don't have to go through the process of downloading and
> extracting before they can view them. You must sanitize if before.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dave.may at tresta.com  Sat Apr  2 00:42:56 2016
From: dave.may at tresta.com (dave.may at tresta.com)
Date: Fri, 1 Apr 2016 16:42:56 -0400
Subject: [Freeswitch-users] T38->g711 gateway issues with latest master and
	v1.6
Message-ID: <B89AB0014204884D8E5E9D3EA1409E7C1294D1EE09@mail1.patlive.local>

Is anybody else having trouble with T38 -> g711 gateway with either v1.6 or latest master?  Initially I thought my problems might have been limited to calls bridging to mod_freetdm (i.e. T38 in -> PRI out), but today I was able to reproduce the problem using SIP only.

I entered a Jira on the issue yesterday, but was hoping that more eyeballs on the problem might reveal something.

https://freeswitch.org/jira/browse/FS-9008

Not sure if everyone can read Jira tickets, so here's the gist of what I can see happening:

- FS1 txfax to FS2
- FS2 sets the following parameters, then bridges call to FS3
** fax_enable_t38_request=true
** execute_on_answer=t38_gateway self
- FS3 answers, then calls rxfax
- bidirectional g711 between FS1/FS2 and FS2/FS3
- FS1 sends T38 reinvite to FS2
- FS2 responds 100 Trying
- FS2 starts sending T38 packets to FS1 (unexpected), while still sending g711 to FS3 (expected)
- 39 seconds later FS2 sends BYE to FS1, but using a new CSeq/branch not seen elsewhere in either leg (?)
- FS2 finally responds to T38 request from FS1, but with 486 Busy Here instead of 200 OK
- FS2 sends repeated BYE to FS1 (e.g. at 1,2,4 seconds)

Thanks!

Dave.
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From anthony.minessale at gmail.com  Sat Apr  2 00:54:55 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 1 Apr 2016 15:54:55 -0500
Subject: [Freeswitch-users] T38->g711 gateway issues with latest master
 and v1.6
In-Reply-To: <B89AB0014204884D8E5E9D3EA1409E7C1294D1EE09@mail1.patlive.local>
References: <B89AB0014204884D8E5E9D3EA1409E7C1294D1EE09@mail1.patlive.local>
Message-ID: <CAKbxfG_8_4OduYNgVRqHO+Se-V77Jb3NyXATyuQdo5TeO9x6EQ@mail.gmail.com>

Same eyeballs mostly =D

On Fri, Apr 1, 2016 at 3:42 PM, dave.may at tresta.com <dave.may at tresta.com>
wrote:

> Is anybody else having trouble with T38 -> g711 gateway with either v1.6
> or latest master?  Initially I thought my problems might have been limited
> to calls bridging to mod_freetdm (i.e. T38 in -> PRI out), but today I was
> able to reproduce the problem using SIP only.
>
>
>
> I entered a Jira on the issue yesterday, but was hoping that more eyeballs
> on the problem might reveal something.
>
>
>
> https://freeswitch.org/jira/browse/FS-9008
>
>
>
> Not sure if everyone can read Jira tickets, so here's the gist of what I
> can see happening:
>
>
>
> - FS1 txfax to FS2
>
> - FS2 sets the following parameters, then bridges call to FS3
>
> ** fax_enable_t38_request=true
>
> ** execute_on_answer=t38_gateway self
>
> - FS3 answers, then calls rxfax
>
> - bidirectional g711 between FS1/FS2 and FS2/FS3
>
> - FS1 sends T38 reinvite to FS2
>
> - FS2 responds 100 Trying
>
> - FS2 starts sending T38 packets to FS1 (unexpected), while still sending
> g711 to FS3 (expected)
>
> - 39 seconds later FS2 sends BYE to FS1, but using a new CSeq/branch not
> seen elsewhere in either leg (?)
>
> - FS2 finally responds to T38 request from FS1, but with 486 Busy Here
> instead of 200 OK
>
> - FS2 sends repeated BYE to FS1 (e.g. at 1,2,4 seconds)
>
>
>
> Thanks!
>
>
>
> Dave.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From brian at freeswitch.org  Sat Apr  2 00:57:01 2016
From: brian at freeswitch.org (Brian West)
Date: Fri, 1 Apr 2016 15:57:01 -0500
Subject: [Freeswitch-users] T38->g711 gateway issues with latest master
 and v1.6
In-Reply-To: <CAKbxfG_8_4OduYNgVRqHO+Se-V77Jb3NyXATyuQdo5TeO9x6EQ@mail.gmail.com>
References: <B89AB0014204884D8E5E9D3EA1409E7C1294D1EE09@mail1.patlive.local>
	<CAKbxfG_8_4OduYNgVRqHO+Se-V77Jb3NyXATyuQdo5TeO9x6EQ@mail.gmail.com>
Message-ID: <CAEJMVkD6AJSih2_eOMin1TEbEMR81r7V66Y7oukaUHs=YdzQSg@mail.gmail.com>

Look at

*conf/testing/dialplan/default/0026_fax.xml*

If you can create a self contained test case, similar to how I have in that
file for t.38 passthru then we should be able to make progress faster.

On Fri, Apr 1, 2016 at 3:54 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Same eyeballs mostly =D
>
> On Fri, Apr 1, 2016 at 3:42 PM, dave.may at tresta.com <dave.may at tresta.com>
> wrote:
>
>> Is anybody else having trouble with T38 -> g711 gateway with either v1.6
>> or latest master?  Initially I thought my problems might have been limited
>> to calls bridging to mod_freetdm (i.e. T38 in -> PRI out), but today I was
>> able to reproduce the problem using SIP only.
>>
>>
>>
>> I entered a Jira on the issue yesterday, but was hoping that more
>> eyeballs on the problem might reveal something.
>>
>>
>>
>> https://freeswitch.org/jira/browse/FS-9008
>>
>>
>>
>> Not sure if everyone can read Jira tickets, so here's the gist of what I
>> can see happening:
>>
>>
>>
>> - FS1 txfax to FS2
>>
>> - FS2 sets the following parameters, then bridges call to FS3
>>
>> ** fax_enable_t38_request=true
>>
>> ** execute_on_answer=t38_gateway self
>>
>> - FS3 answers, then calls rxfax
>>
>> - bidirectional g711 between FS1/FS2 and FS2/FS3
>>
>> - FS1 sends T38 reinvite to FS2
>>
>> - FS2 responds 100 Trying
>>
>> - FS2 starts sending T38 packets to FS1 (unexpected), while still sending
>> g711 to FS3 (expected)
>>
>> - 39 seconds later FS2 sends BYE to FS1, but using a new CSeq/branch not
>> seen elsewhere in either leg (?)
>>
>> - FS2 finally responds to T38 request from FS1, but with 486 Busy Here
>> instead of 200 OK
>>
>> - FS2 sends repeated BYE to FS1 (e.g. at 1,2,4 seconds)
>>
>>
>>
>> Thanks!
>>
>>
>>
>> Dave.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From dave.may at tresta.com  Sat Apr  2 01:10:45 2016
From: dave.may at tresta.com (dave.may at tresta.com)
Date: Fri, 1 Apr 2016 17:10:45 -0400
Subject: [Freeswitch-users] T38->g711 gateway issues with latest master
 and v1.6
In-Reply-To: <CAEJMVkD6AJSih2_eOMin1TEbEMR81r7V66Y7oukaUHs=YdzQSg@mail.gmail.com>
References: <B89AB0014204884D8E5E9D3EA1409E7C1294D1EE09@mail1.patlive.local>
	<CAKbxfG_8_4OduYNgVRqHO+Se-V77Jb3NyXATyuQdo5TeO9x6EQ@mail.gmail.com>
	<CAEJMVkD6AJSih2_eOMin1TEbEMR81r7V66Y7oukaUHs=YdzQSg@mail.gmail.com>
Message-ID: <B89AB0014204884D8E5E9D3EA1409E7C1294D1EE0C@mail1.patlive.local>

Will do -- thanks for the quick response Brian!

Dave.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Friday, April 1, 2016 4:57 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] T38->g711 gateway issues with latest master and v1.6

Look at

conf/testing/dialplan/default/0026_fax.xml

If you can create a self contained test case, similar to how I have in that file for t.38 passthru then we should be able to make progress faster.

On Fri, Apr 1, 2016 at 3:54 PM, Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
Same eyeballs mostly =D

On Fri, Apr 1, 2016 at 3:42 PM, dave.may at tresta.com<mailto:dave.may at tresta.com> <dave.may at tresta.com<mailto:dave.may at tresta.com>> wrote:
Is anybody else having trouble with T38 -> g711 gateway with either v1.6 or latest master?  Initially I thought my problems might have been limited to calls bridging to mod_freetdm (i.e. T38 in -> PRI out), but today I was able to reproduce the problem using SIP only.

I entered a Jira on the issue yesterday, but was hoping that more eyeballs on the problem might reveal something.

https://freeswitch.org/jira/browse/FS-9008

Not sure if everyone can read Jira tickets, so here's the gist of what I can see happening:

- FS1 txfax to FS2
- FS2 sets the following parameters, then bridges call to FS3
** fax_enable_t38_request=true
** execute_on_answer=t38_gateway self
- FS3 answers, then calls rxfax
- bidirectional g711 between FS1/FS2 and FS2/FS3
- FS1 sends T38 reinvite to FS2
- FS2 responds 100 Trying
- FS2 starts sending T38 packets to FS1 (unexpected), while still sending g711 to FS3 (expected)
- 39 seconds later FS2 sends BYE to FS1, but using a new CSeq/branch not seen elsewhere in either leg (?)
- FS2 finally responds to T38 request from FS1, but with 486 Busy Here instead of 200 OK
- FS2 sends repeated BYE to FS1 (e.g. at 1,2,4 seconds)

Thanks!

Dave.

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

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

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



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

? http://freeswitch.org/  ? http://cluecon.com/  ? http://twitter.com/FreeSWITCH
? irc.freenode.net<http://irc.freenode.net> #freeswitch ? http://freeswitch.org/g+
ClueCon Weekly Development Call
? sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900<tel:%2B19193869900>

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA

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

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

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



--

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

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

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

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

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
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From gregor at infomedia.si  Sat Apr  2 13:12:20 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sat, 2 Apr 2016 11:12:20 +0200
Subject: [Freeswitch-users] Playback from remote server
Message-ID: <CAC7ttFGPXvS95zqkKBPQUyDX0q_R7oeTsYXhWKa=CUhjQFTWQQ@mail.gmail.com>

Hi!

Need some help.

I am trying to playback file that is on remote server, but getting error:

[ERR] switch_core_file.c:253 Invalid file format [http] for [
www.myserver.com/file.wav]

File is accessible via browser and mod_sndfile is loaded.

Any hint what could be wrong?

Best regards, Gregor
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From blasterjr at gmail.com  Sat Apr  2 13:22:19 2016
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Sat, 2 Apr 2016 03:22:19 -0600
Subject: [Freeswitch-users] Playback from remote server
In-Reply-To: <CAC7ttFGPXvS95zqkKBPQUyDX0q_R7oeTsYXhWKa=CUhjQFTWQQ@mail.gmail.com>
References: <CAC7ttFGPXvS95zqkKBPQUyDX0q_R7oeTsYXhWKa=CUhjQFTWQQ@mail.gmail.com>
Message-ID: <CAO3hyzFbDs0N0XuYOO_7ear8YWrS4NVPiKAL0bf7xmP22WYBYA@mail.gmail.com>

In order to use files from an http server you need to use mod_http_cache (
https://freeswitch.org/confluence/display/FREESWITCH/mod_http_cache) or
mod_httpapi (https://freeswitch.org/confluence/display/FREESWITCH/mod_httapi
)

On Sat, Apr 2, 2016 at 3:12 AM, Gregor Nanger <gregor at infomedia.si> wrote:

> Hi!
>
> Need some help.
>
> I am trying to playback file that is on remote server, but getting error:
>
> [ERR] switch_core_file.c:253 Invalid file format [http] for [
> www.myserver.com/file.wav]
>
> File is accessible via browser and mod_sndfile is loaded.
>
> Any hint what could be wrong?
>
> Best regards, Gregor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gregor at infomedia.si  Sat Apr  2 16:56:41 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sat, 2 Apr 2016 14:56:41 +0200
Subject: [Freeswitch-users] Playback from remote server
In-Reply-To: <CAO3hyzFbDs0N0XuYOO_7ear8YWrS4NVPiKAL0bf7xmP22WYBYA@mail.gmail.com>
References: <CAC7ttFGPXvS95zqkKBPQUyDX0q_R7oeTsYXhWKa=CUhjQFTWQQ@mail.gmail.com>
	<CAO3hyzFbDs0N0XuYOO_7ear8YWrS4NVPiKAL0bf7xmP22WYBYA@mail.gmail.com>
Message-ID: <CAC7ttFFhh-trr-id6uDFu6cV0pN4EzEWbn+bhGHjjTBiMkQgvQ@mail.gmail.com>

Thank you!

That was it :-)

2016-04-02 11:22 GMT+02:00 Chris Tunbridge <blasterjr at gmail.com>:

> In order to use files from an http server you need to use mod_http_cache (
> https://freeswitch.org/confluence/display/FREESWITCH/mod_http_cache) or
> mod_httpapi (
> https://freeswitch.org/confluence/display/FREESWITCH/mod_httapi)
>
> On Sat, Apr 2, 2016 at 3:12 AM, Gregor Nanger <gregor at infomedia.si> wrote:
>
>> Hi!
>>
>> Need some help.
>>
>> I am trying to playback file that is on remote server, but getting error:
>>
>> [ERR] switch_core_file.c:253 Invalid file format [http] for [
>> www.myserver.com/file.wav]
>>
>> File is accessible via browser and mod_sndfile is loaded.
>>
>> Any hint what could be wrong?
>>
>> Best regards, Gregor
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 saurabhkv01 at gmail.com  Sat Apr  2 10:54:17 2016
From: saurabhkv01 at gmail.com (saurabh verrma)
Date: Sat, 2 Apr 2016 12:24:17 +0530
Subject: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
In-Reply-To: <E93C0E00-0EEE-44E1-81F2-1E8CF374057B@jerris.com>
References: <CACot2ntTB+7OXvomm3sxMzxnNEnT6vHtQb=T=m8MTOFT0=zFZg@mail.gmail.com>
	<E93C0E00-0EEE-44E1-81F2-1E8CF374057B@jerris.com>
Message-ID: <CACot2ntmtZzcXs4bWqd3oW9i5URXMpqaouu8EXKWeN2thDJG4A@mail.gmail.com>

Thanks Michael,

Basically we're writing a PJSIP based application & PJSIP doesn't have DTLS
support. So we're thinking to use FreeSWITCH library for DTLS/SRTP.

On Fri, Apr 1, 2016 at 7:29 PM, Michael Jerris <mike at jerris.com> wrote:

> we have full support for webrtc media profile which would include these
> features  There are not a ton of people who use freeswitch as a library,
> but it is built that way so that you can control it and host it in another
> application instead of stand alone.  If you are trying to accomplish
> something I'd try to handle it standalone first so you can learn all the
> different ways you might control it before architecting a solution
>
> On Apr 1, 2016, at 9:48 AM, saurabh verrma <saurabhkv01 at gmail.com> wrote:
>
> Hi All,
>
> I want to use FreeSWITCH library for DTLS/SRTP support. I want to know in
> freeswitch which library has the support of these features(SRTP/DTLS) ?
> Is there any application available based on freeswitch library ?
>
> Any help would be appreciable.
>
> --
> *With Warm Regards:*
> *Saurabh Kumar Verma*
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Warm Regards:*
*Saurabh Kumar Verma*
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From mandra at gmail.com  Sun Apr  3 04:29:37 2016
From: mandra at gmail.com (Chris Mandra)
Date: Sat, 2 Apr 2016 20:29:37 -0400
Subject: [Freeswitch-users] Module initialization failure status
Message-ID: <CAJdRzN4XOaj=-F6=CX0tOLv9FDMFDkiVi8ETgAhPh-RHEzONsg@mail.gmail.com>

Hi Guys - Happy Saturday.
Quick question: what is  the proper status to return if a module fails on
initialization?
That is: should it always return SWITCH_STATUS_SUCCESS ?
What if something goes wrong?  What status should it return?
Thanks!chris
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From gregor at infomedia.si  Sun Apr  3 14:49:00 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 03 Apr 2016 10:49:00 +0000
Subject: [Freeswitch-users] Auto answer via Esl
Message-ID: <CAC7ttFHm20w9f7W7bYZ0xiOszS7saui7_rH5155NP6yYPMA6KA@mail.gmail.com>

For example, if I have sip phone that supports auto answer feature, how can
I send variable or sip header via Esl to uuid?

Any hints?
-- 
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 vincent.gire at gmail.com  Sun Apr  3 20:13:53 2016
From: vincent.gire at gmail.com (Vincent Gire)
Date: Sun, 3 Apr 2016 18:13:53 +0200
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
Message-ID: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>

Hi all,

Thank you to all contributing to FreeSWITCH !

I'm working on a IVR project where logic is implemented on a HTTP server.
We are leaving Twilio because we now need to record and live stream the
session to the HTTP server in WAV format (chunked transfer encoding).

*mod_httapi* looks great (HT TAPI very similar to Twilio's) but it seems
that the records are first saved to disk before there are sent to the
server as chunked data.
We need the transfer to start as soon as the recording starts.

*mod_shout* does start the request almost as the records starts but it does
not support WAV file and shout:// is not exactly a HTTP request (SOURCE
method instead of PUT).

Is there a way to use these modules to achieve our goal ?

If not, we are willing to author a specific module or rather contribute to
the existing ones.

We've identified two approaches:

   1. From *mod_httapi* Modify mod_httapi to directly stream the record
   instead of completely saving it to disk before the HTTP chunked transfer
   starts.
   This seems the most logical but with more than 3000 lines, mod_httapi
   does not seem to be the easiest module to build upon for newcomers!

   2. From *mod_shout*
      1. Modify libshoot to replace the custom SOURCE method with standard
      HTTP PUT method
      2. Modify mod_shout to support wav files
      3. Implement our IVR in script (javascript/lua)

What do you think ?

Thank you for your help.

Cheers,
Vincent
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From s.safarov at gmail.com  Sun Apr  3 21:30:41 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 03 Apr 2016 17:30:41 +0000
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
Message-ID: <CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>

Please look at webdav mounted filesystem.

On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com> wrote:

> Hi all,
>
> Thank you to all contributing to FreeSWITCH !
>
> I'm working on a IVR project where logic is implemented on a HTTP server.
> We are leaving Twilio because we now need to record and live stream the
> session to the HTTP server in WAV format (chunked transfer encoding).
>
> *mod_httapi* looks great (HT TAPI very similar to Twilio's) but it seems
> that the records are first saved to disk before there are sent to the
> server as chunked data.
> We need the transfer to start as soon as the recording starts.
>
> *mod_shout* does start the request almost as the records starts but it
> does not support WAV file and shout:// is not exactly a HTTP request
> (SOURCE method instead of PUT).
>
> Is there a way to use these modules to achieve our goal ?
>
> If not, we are willing to author a specific module or rather contribute to
> the existing ones.
>
> We've identified two approaches:
>
>    1. From *mod_httapi* Modify mod_httapi to directly stream the record
>    instead of completely saving it to disk before the HTTP chunked transfer
>    starts.
>    This seems the most logical but with more than 3000 lines, mod_httapi
>    does not seem to be the easiest module to build upon for newcomers!
>
>    2. From *mod_shout*
>       1. Modify libshoot to replace the custom SOURCE method with
>       standard HTTP PUT method
>       2. Modify mod_shout to support wav files
>       3. Implement our IVR in script (javascript/lua)
>
> What do you think ?
>
> Thank you for your help.
>
> Cheers,
> Vincent
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From gregor at infomedia.si  Sun Apr  3 23:44:38 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 3 Apr 2016 21:44:38 +0200
Subject: [Freeswitch-users] Auto answer via Esl
In-Reply-To: <CAC7ttFHm20w9f7W7bYZ0xiOszS7saui7_rH5155NP6yYPMA6KA@mail.gmail.com>
References: <CAC7ttFHm20w9f7W7bYZ0xiOszS7saui7_rH5155NP6yYPMA6KA@mail.gmail.com>
Message-ID: <CAC7ttFF895o6ax7FGuVcRXuVTHisOZnQhrUrhn-Qt1AgFTQJHg@mail.gmail.com>

Maybe there is another way to accomplish what I want.

I have my app connected via ESL to Freeswitch. On incoming call voip phone
rings. I want user to click in my app in gui accept call, not on voip
phone. Voip phone does support auto answer feature.

I was thinking if there is a way to send SIP message (sip_h_Call-Info)
?or set variable on channel (I have channel uuid in my code) and send ?
to phone via ESL
??
, after phone already rings.
?

Best regards, Gregor

2016-04-03 12:49 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:

> For example, if I have sip phone that supports auto answer feature, how
> can I send variable or sip header via Esl to uuid?
>
> Any hints?
> --
> 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
>



-- 
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 brian at freeswitch.org  Mon Apr  4 01:24:17 2016
From: brian at freeswitch.org (Brian West)
Date: Sun, 3 Apr 2016 16:24:17 -0500
Subject: [Freeswitch-users] Auto answer via Esl
In-Reply-To: <CAC7ttFF895o6ax7FGuVcRXuVTHisOZnQhrUrhn-Qt1AgFTQJHg@mail.gmail.com>
References: <CAC7ttFHm20w9f7W7bYZ0xiOszS7saui7_rH5155NP6yYPMA6KA@mail.gmail.com>
	<CAC7ttFF895o6ax7FGuVcRXuVTHisOZnQhrUrhn-Qt1AgFTQJHg@mail.gmail.com>
Message-ID: <CAEJMVkC8UWYUQhfDauOhC3xn8ny9zr=rMZ7bNEzQLHyu2HMpvA@mail.gmail.com>

Set sip_auto_answer=true

On Sunday, April 3, 2016, Gregor Nanger <gregor at infomedia.si> wrote:

> Maybe there is another way to accomplish what I want.
>
> I have my app connected via ESL to Freeswitch. On incoming call voip phone
> rings. I want user to click in my app in gui accept call, not on voip
> phone. Voip phone does support auto answer feature.
>
> I was thinking if there is a way to send SIP message (sip_h_Call-Info)
> ?or set variable on channel (I have channel uuid in my code) and send ?
> to phone via ESL
> ??
> , after phone already rings.
> ?
>
> Best regards, Gregor
>
> 2016-04-03 12:49 GMT+02:00 Gregor Nanger <gregor at infomedia.si
> <javascript:_e(%7B%7D,'cvml','gregor at infomedia.si');>>:
>
>> For example, if I have sip phone that supports auto answer feature, how
>> can I send variable or sip header via Esl to uuid?
>>
>> Any hints?
>> --
>> 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
>>
>
>
>
> --
> 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
>


-- 

*Brian West*
brian at freeswitch.org


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

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From gregor at infomedia.si  Mon Apr  4 01:38:10 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sun, 3 Apr 2016 23:38:10 +0200
Subject: [Freeswitch-users] Auto answer via Esl
In-Reply-To: <CAEJMVkC8UWYUQhfDauOhC3xn8ny9zr=rMZ7bNEzQLHyu2HMpvA@mail.gmail.com>
References: <CAC7ttFHm20w9f7W7bYZ0xiOszS7saui7_rH5155NP6yYPMA6KA@mail.gmail.com>
	<CAC7ttFF895o6ax7FGuVcRXuVTHisOZnQhrUrhn-Qt1AgFTQJHg@mail.gmail.com>
	<CAEJMVkC8UWYUQhfDauOhC3xn8ny9zr=rMZ7bNEzQLHyu2HMpvA@mail.gmail.com>
Message-ID: <CAC7ttFGRZRVLkoPz7nPNgxvRaYRTXJgdrmB5VktmasEkGp9fng@mail.gmail.com>

Yes, I saw this feature and I can set it in dialplan, but can I set this
variable on active channel via ESL if I know uuid of channel.

It is that I want to set variable on channel that is already in RINGING
state and when user click in my app Accept Call.


2016-04-03 23:24 GMT+02:00 Brian West <brian at freeswitch.org>:

> Set sip_auto_answer=true
>
>
> On Sunday, April 3, 2016, Gregor Nanger <gregor at infomedia.si> wrote:
>
>> Maybe there is another way to accomplish what I want.
>>
>> I have my app connected via ESL to Freeswitch. On incoming call voip
>> phone rings. I want user to click in my app in gui accept call, not on voip
>> phone. Voip phone does support auto answer feature.
>>
>> I was thinking if there is a way to send SIP message (sip_h_Call-Info)
>> ?or set variable on channel (I have channel uuid in my code) and send ?
>> to phone via ESL
>> ??
>> , after phone already rings.
>> ?
>>
>> Best regards, Gregor
>>
>> 2016-04-03 12:49 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:
>>
>>> For example, if I have sip phone that supports auto answer feature, how
>>> can I send variable or sip header via Esl to uuid?
>>>
>>> Any hints?
>>> --
>>> 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
>>>
>>
>>
>>
>> --
>> 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
>>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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 mandra at gmail.com  Mon Apr  4 02:16:48 2016
From: mandra at gmail.com (Chris Mandra)
Date: Sun, 3 Apr 2016 18:16:48 -0400
Subject: [Freeswitch-users] Module initialization failure status
In-Reply-To: <CAJdRzN4XOaj=-F6=CX0tOLv9FDMFDkiVi8ETgAhPh-RHEzONsg@mail.gmail.com>
References: <CAJdRzN4XOaj=-F6=CX0tOLv9FDMFDkiVi8ETgAhPh-RHEzONsg@mail.gmail.com>
Message-ID: <CAJdRzN4Ni6dg6_cx5VsPWpe1+EY8n3559qcqDvgAumpF4Pw79g@mail.gmail.com>

Anything?

On Saturday, April 2, 2016, Chris Mandra <mandra at gmail.com> wrote:

> Hi Guys - Happy Saturday.
> Quick question: what is  the proper status to return if a module fails on
> initialization?
> That is: should it always return SWITCH_STATUS_SUCCESS ?
> What if something goes wrong?  What status should it return?
> Thanks!chris
>
>
>

-- 
mandra
c:410.258.5281
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From ahabiba at gmail.com  Mon Apr  4 09:18:55 2016
From: ahabiba at gmail.com (Ahmed habiba)
Date: Mon, 4 Apr 2016 08:18:55 +0300
Subject: [Freeswitch-users] Module initialization failure status
In-Reply-To: <mailman.41.1459721839.8223.freeswitch-users@lists.freeswitch.org>
References: <mailman.41.1459721839.8223.freeswitch-users@lists.freeswitch.org>
Message-ID: <CE0F0E89-A33F-4CF6-8BE3-0D71F3A53C9C@gmail.com>

Any other return status would be considered as failed i.e ?SWITCH_STATUS_FALSE <http://docs.freeswitch.org/switch__types_8h.html#fedb3923fc3ae51937cef145c5499dc11e51a33abaa215d06337d2b38f1f4c53>? but you should print a reasonable debug message to allow the user of your module the exact failure reason.

> 
> From: Chris Mandra <mandra at gmail.com>
> Subject: Re: [Freeswitch-users] Module initialization failure status
> Date: April 4, 2016 at 1:16:48 AM GMT+3
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Anything?
> 
> On Saturday, April 2, 2016, Chris Mandra <mandra at gmail.com <mailto:mandra at gmail.com>> wrote:
> Hi Guys - Happy Saturday. 
> 
> Quick question: what is  the proper status to return if a module fails on initialization?
> That is: should it always return SWITCH_STATUS_SUCCESS ?
> What if something goes wrong?  What status should it return?
> 
> Thanks!
> chris
> 
> 
> 
> 
> -- 
> mandra
> c:410.258.5281
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From amani.mansour2 at gmail.com  Mon Apr  4 12:22:20 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Mon, 4 Apr 2016 09:22:20 +0100
Subject: [Freeswitch-users] Configure SDP
Message-ID: <CAFETSWiskTaDA3fFw4qbBo7h7CsXKjiFcY7EXF5CYLk917YQ0Q@mail.gmail.com>

Good morning ,

I need to configure a number to send invite without SDP , can anyone help
me please ?


best regards
amani
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From hunterj91 at hotmail.com  Mon Apr  4 15:55:44 2016
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Mon, 4 Apr 2016 11:55:44 +0000
Subject: [Freeswitch-users] Hash wav file instead of pound for # digit UK
Message-ID: <DUB112-W1396D72A031B061777D73EBD9D0@phx.gbl>

Hi All,
Is there a .wav file which says 'hash' instead of pound which can be applied for users in the UK/GB region?
Looking through the ascii files  I see 35 is 'pound' but ideally I need 'hash'.
Is one available or do I need to create my own? 
Many thanks
J 		 	   		  
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From ovoshlook at gmail.com  Sun Apr  3 23:07:47 2016
From: ovoshlook at gmail.com (Yuriy Gorlichenko)
Date: Sun, 3 Apr 2016 22:07:47 +0300
Subject: [Freeswitch-users] FS not see invite
Message-ID: <CABSP_VfiEzaP58j0jSA8dd8D6psubk9QGvpZ=i46xBVqNVE=Dg@mail.gmail.com>

Hello. I try to make a call through another SIP instalse to freeswitch.
I added extension and use FS as proxy to rpovider.
I successfully registered on FS as user and try to make a call.

I see at FS cli that invite arrives and FS answers 407, next my instanse
Sends ACK and thn INVITE with a digest auth.

I see thins invite at the TCPDUMP on the FS machine but not see it at the FS
Firewall is disabled for tests....
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From jelena at misticnabica.hr  Mon Apr  4 02:25:36 2016
From: jelena at misticnabica.hr (jelena at misticnabica.hr)
Date: Sun, 3 Apr 2016 22:25:36 GMT
Subject: [Freeswitch-users] Module initialization failure status
Message-ID: <mailman.0.1459774842.8220.freeswitch-users@lists.freeswitch.org>



From mandra at gmail.com  Mon Apr  4 17:24:28 2016
From: mandra at gmail.com (Chris Mandra)
Date: Mon, 4 Apr 2016 09:24:28 -0400
Subject: [Freeswitch-users] Module initialization failure status
In-Reply-To: <57026603.2289320a.4e3b3.7016SMTPIN_ADDED_MISSING@mx.google.com>
References: <57026603.2289320a.4e3b3.7016SMTPIN_ADDED_MISSING@mx.google.com>
Message-ID: <CAJdRzN4exy8YMeQiWemZ0AomZY3q6CYEmtOnY-n7wmwCfASfKQ@mail.gmail.com>

Thank you Ahmed - Jelena, is your email missing content, or are you
suggesting the consulting services for this question?

On Sun, Apr 3, 2016 at 6:25 PM, <jelena at misticnabica.hr> wrote:

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



-- 
mandra
c:410.258.5281
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From mike at jerris.com  Mon Apr  4 18:44:27 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Apr 2016 10:44:27 -0400
Subject: [Freeswitch-users] Module initialization failure status
In-Reply-To: <CAJdRzN4Ni6dg6_cx5VsPWpe1+EY8n3559qcqDvgAumpF4Pw79g@mail.gmail.com>
References: <CAJdRzN4XOaj=-F6=CX0tOLv9FDMFDkiVi8ETgAhPh-RHEzONsg@mail.gmail.com>
	<CAJdRzN4Ni6dg6_cx5VsPWpe1+EY8n3559qcqDvgAumpF4Pw79g@mail.gmail.com>
Message-ID: <16D92427-15DA-49F7-9D7D-B4CF86CC43B2@jerris.com>

anything but

SWITCH_STATUS_SUCCESS
SWITCH_STATUS_NOUNLOAD

is considered an error.


> On Apr 3, 2016, at 6:16 PM, Chris Mandra <mandra at gmail.com> wrote:
> 
> Anything?
> 
> On Saturday, April 2, 2016, Chris Mandra <mandra at gmail.com <mailto:mandra at gmail.com>> wrote:
> Hi Guys - Happy Saturday. 
> 
> Quick question: what is  the proper status to return if a module fails on initialization?
> That is: should it always return SWITCH_STATUS_SUCCESS ?
> What if something goes wrong?  What status should it return?
> 
> Thanks!
> chris
> 
> 
> 
> 
> -- 
> mandra
> c:410.258.5281
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Mon Apr  4 18:44:59 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Apr 2016 10:44:59 -0400
Subject: [Freeswitch-users] Configure SDP
In-Reply-To: <CAFETSWiskTaDA3fFw4qbBo7h7CsXKjiFcY7EXF5CYLk917YQ0Q@mail.gmail.com>
References: <CAFETSWiskTaDA3fFw4qbBo7h7CsXKjiFcY7EXF5CYLk917YQ0Q@mail.gmail.com>
Message-ID: <3BD6E704-4227-4FB7-A12E-02AB326679F7@jerris.com>

you are looking for 3pcc

> On Apr 4, 2016, at 4:22 AM, amani mansour <amani.mansour2 at gmail.com> wrote:
> 
> Good morning ,
> 
> I need to configure a number to send invite without SDP , can anyone help me please ? 



From mike at jerris.com  Mon Apr  4 18:45:55 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Apr 2016 10:45:55 -0400
Subject: [Freeswitch-users] Hash wav file instead of pound for # digit UK
In-Reply-To: <DUB112-W1396D72A031B061777D73EBD9D0@phx.gbl>
References: <DUB112-W1396D72A031B061777D73EBD9D0@phx.gbl>
Message-ID: <1A022EDB-7C1B-4916-BFD2-C5AFDED3869D@jerris.com>

we don't have a uk english sounds set, you would need to create your own.


> On Apr 4, 2016, at 7:55 AM, Jonathan Hunter <hunterj91 at hotmail.com> wrote:
> 
> Hi All,
> 
> Is there a .wav file which says 'hash' instead of pound which can be applied for users in the UK/GB region?
> 
> Looking through the ascii files  I see 35 is 'pound' but ideally I need 'hash'.
> 
> Is one available or do I need to create my own? 
> 
> Many thanks
> 
> J
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From mike at jerris.com  Mon Apr  4 18:48:02 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Apr 2016 10:48:02 -0400
Subject: [Freeswitch-users] FS not see invite
In-Reply-To: <CABSP_VfiEzaP58j0jSA8dd8D6psubk9QGvpZ=i46xBVqNVE=Dg@mail.gmail.com>
References: <CABSP_VfiEzaP58j0jSA8dd8D6psubk9QGvpZ=i46xBVqNVE=Dg@mail.gmail.com>
Message-ID: <B07B3220-0A4E-4264-9D24-9FD9F75942DE@jerris.com>

are you SURE the firewall is turned off?  Is the sip going to the right iP/port that FreeSWITCH is listening on?

> On Apr 3, 2016, at 3:07 PM, Yuriy Gorlichenko <ovoshlook at gmail.com> wrote:
> 
> Hello. I try to make a call through another SIP instalse to freeswitch. 
> I added extension and use FS as proxy to rpovider.
> I successfully registered on FS as user and try to make a call.
> 
> I see at FS cli that invite arrives and FS answers 407, next my instanse Sends ACK and thn INVITE with a digest auth.
> 
> I see thins invite at the TCPDUMP on the FS machine but not see it at the FS
> Firewall is disabled for tests....

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From andrew at cassidywebservices.co.uk  Mon Apr  4 19:18:08 2016
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Mon, 4 Apr 2016 16:18:08 +0100
Subject: [Freeswitch-users] Hash wav file instead of pound for # digit UK
In-Reply-To: <1A022EDB-7C1B-4916-BFD2-C5AFDED3869D@jerris.com>
References: <DUB112-W1396D72A031B061777D73EBD9D0@phx.gbl>
	<1A022EDB-7C1B-4916-BFD2-C5AFDED3869D@jerris.com>
Message-ID: <CAMbW9o+znPJ-P59qR-3oPCFv5mXzWeokoP1ftgny37PQa3t0Gw@mail.gmail.com>

There are commercial British English packs, but a community owned one would
be great.

On 4 April 2016 at 15:45, Michael Jerris <mike at jerris.com> wrote:

> we don't have a uk english sounds set, you would need to create your own.
>
>
> On Apr 4, 2016, at 7:55 AM, Jonathan Hunter <hunterj91 at hotmail.com> wrote:
>
> Hi All,
>
> Is there a .wav file which says 'hash' instead of pound which can be
> applied for users in the UK/GB region?
>
> Looking through the ascii files  I see 35 is 'pound' but ideally I need
> 'hash'.
>
> Is one available or do I need to create my own?
>
> Many thanks
>
> J
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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

03303 880 960 andrew at cassidyweb.co.uk <andrew at cassidywebservices.co.uk>
www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>
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From mike at jerris.com  Mon Apr  4 19:24:25 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Apr 2016 11:24:25 -0400
Subject: [Freeswitch-users] Hash wav file instead of pound for # digit UK
In-Reply-To: <CAMbW9o+znPJ-P59qR-3oPCFv5mXzWeokoP1ftgny37PQa3t0Gw@mail.gmail.com>
References: <DUB112-W1396D72A031B061777D73EBD9D0@phx.gbl>
	<1A022EDB-7C1B-4916-BFD2-C5AFDED3869D@jerris.com>
	<CAMbW9o+znPJ-P59qR-3oPCFv5mXzWeokoP1ftgny37PQa3t0Gw@mail.gmail.com>
Message-ID: <96F59C65-678A-41BF-A062-370F244961B8@jerris.com>

If we can crowd fund that, or if someone has one to donate, we'd be happy to host it.

> On Apr 4, 2016, at 11:18 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk> wrote:
> 
> There are commercial British English packs, but a community owned one would be great.
> 
> On 4 April 2016 at 15:45, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> we don't have a uk english sounds set, you would need to create your own.
> 
> 
>> On Apr 4, 2016, at 7:55 AM, Jonathan Hunter <hunterj91 at hotmail.com <mailto:hunterj91 at hotmail.com>> wrote:
>> 
>> Hi All,
>> 
>> Is there a .wav file which says 'hash' instead of pound which can be applied for users in the UK/GB region?
>> 
>> Looking through the ascii files  I see 35 is 'pound' but ideally I need 'hash'.
>> 
>> Is one available or do I need to create my own? 
>> 
>> Many thanks
>> 
>> J
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Andrew Cassidy BSc (Hons) MBCS SSCA
> Managing Director
> 
> 03303 880 960 andrew at cassidyweb.co.uk <mailto:andrew at cassidywebservices.co.uk> www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nneul at mst.edu  Mon Apr  4 19:37:14 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Mon, 4 Apr 2016 10:37:14 -0500
Subject: [Freeswitch-users] Hash wav file instead of pound for # digit UK
In-Reply-To: <96F59C65-678A-41BF-A062-370F244961B8@jerris.com>
References: <DUB112-W1396D72A031B061777D73EBD9D0@phx.gbl>
	<1A022EDB-7C1B-4916-BFD2-C5AFDED3869D@jerris.com>
	<CAMbW9o+znPJ-P59qR-3oPCFv5mXzWeokoP1ftgny37PQa3t0Gw@mail.gmail.com>
	<96F59C65-678A-41BF-A062-370F244961B8@jerris.com>
Message-ID: <57028A2A.5040407@mst.edu>

It sounds like in this case though a full language pack isn't strictly required - just an additional word/sound from 
Callie. Is there a queue/process/etc. for requesting (and funding?) additional words/phrases the next time phrases are 
ordered?

I had previously put in a jira in requesting/suggesting a couple of phrases (FS-8733) related to conferencing. I could 
see this being a good list to add:

	"this"
	"conference"
	"is"
	"being"
	"recorded"
	"are"
	"transferred"
	"hash"

	"is being recorded"

-- Nathan

On 04/04/2016 10:24 AM, Michael Jerris wrote:
> If we can crowd fund that, or if someone has one to donate, we'd be happy to host it.
>
>> On Apr 4, 2016, at 11:18 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>>
>> wrote:
>>
>> There are commercial British English packs, but a community owned one would be great.
>>
>> On 4 April 2016 at 15:45, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>
>>     we don't have a uk english sounds set, you would need to create your own.
>>
>>
>>>     On Apr 4, 2016, at 7:55 AM, Jonathan Hunter <hunterj91 at hotmail.com <mailto:hunterj91 at hotmail.com>> wrote:
>>>
>>>     Hi All,
>>>
>>>     Is there a .wav file which says 'hash' instead of pound which can be applied for users in the UK/GB region?
>>>
>>>     Looking through the ascii files  I see 35 is 'pound' but ideally I need 'hash'.
>>>
>>>     Is one available or do I need to create my own?
>>>
>>>     Many thanks
>>>
>>>     J
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>     http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>     http://www.cluecon.com <http://www.cluecon.com/>
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>>
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>> 03303 880 960 andrew at cassidyweb.co.uk <mailto:andrew at cassidywebservices.co.uk> www.cassidyweb.co.uk
>> <http://www.cassidywebservices.co.uk/>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 fdelawarde at wirelessmundi.com  Mon Apr  4 19:46:40 2016
From: fdelawarde at wirelessmundi.com (=?UTF-8?B?RnJhbsOnb2lz?=)
Date: Mon, 4 Apr 2016 17:46:40 +0200
Subject: [Freeswitch-users] Specific ring-device for verto communicator (or
	webrtc in general)
Message-ID: <57028C60.9030306@wirelessmundi.com>

Hi,

Would it be possible to for a webrtc/verto application to use a 
different audio device for "ringing" purposes (like most softphones do)?

Thanks,
F


From mike at jerris.com  Mon Apr  4 19:58:30 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Apr 2016 11:58:30 -0400
Subject: [Freeswitch-users] Hash wav file instead of pound for # digit UK
In-Reply-To: <57028A2A.5040407@mst.edu>
References: <DUB112-W1396D72A031B061777D73EBD9D0@phx.gbl>
	<1A022EDB-7C1B-4916-BFD2-C5AFDED3869D@jerris.com>
	<CAMbW9o+znPJ-P59qR-3oPCFv5mXzWeokoP1ftgny37PQa3t0Gw@mail.gmail.com>
	<96F59C65-678A-41BF-A062-370F244961B8@jerris.com>
	<57028A2A.5040407@mst.edu>
Message-ID: <D663B890-B3A1-40F1-8113-259CDF354E4D@jerris.com>

We can do a recording session to get a handful recorded... Is anyone out there able to fund it?  If so, lets collect up a list.


> On Apr 4, 2016, at 11:37 AM, Nathan Neulinger <nneul at mst.edu> wrote:
> 
> It sounds like in this case though a full language pack isn't strictly required - just an additional word/sound from 
> Callie. Is there a queue/process/etc. for requesting (and funding?) additional words/phrases the next time phrases are 
> ordered?
> 
> I had previously put in a jira in requesting/suggesting a couple of phrases (FS-8733) related to conferencing. I could 
> see this being a good list to add:
> 
> 	"this"
> 	"conference"
> 	"is"
> 	"being"
> 	"recorded"
> 	"are"
> 	"transferred"
> 	"hash"
> 
> 	"is being recorded"
> 
> -- Nathan
> 
> On 04/04/2016 10:24 AM, Michael Jerris wrote:
>> If we can crowd fund that, or if someone has one to donate, we'd be happy to host it.
>> 
>>> On Apr 4, 2016, at 11:18 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk> <mailto:andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>>>
>>> wrote:
>>> 
>>> There are commercial British English packs, but a community owned one would be great.
>>> 
>>> On 4 April 2016 at 15:45, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com> <mailto:mike at jerris.com <mailto:mike at jerris.com>>> wrote:
>>> 
>>>    we don't have a uk english sounds set, you would need to create your own.
>>> 
>>> 
>>>>    On Apr 4, 2016, at 7:55 AM, Jonathan Hunter <hunterj91 at hotmail.com <mailto:hunterj91 at hotmail.com> <mailto:hunterj91 at hotmail.com <mailto:hunterj91 at hotmail.com>>> wrote:
>>>> 
>>>>    Hi All,
>>>> 
>>>>    Is there a .wav file which says 'hash' instead of pound which can be applied for users in the UK/GB region?
>>>> 
>>>>    Looking through the ascii files  I see 35 is 'pound' but ideally I need 'hash'.
>>>> 
>>>>    Is one available or do I need to create my own?
>>>> 
>>>>    Many thanks
>>>> 
>>>>    J

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From hunterj91 at hotmail.com  Mon Apr  4 20:40:15 2016
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Mon, 4 Apr 2016 16:40:15 +0000
Subject: [Freeswitch-users] Hash wav file instead of pound for # digit UK
In-Reply-To: <CAPQovyQorkHCdxxuj10bzG3nQywG3QPvzudAdLsuHXJ2h-8p6g@mail.gmail.com>
References: <DUB112-W1396D72A031B061777D73EBD9D0@phx.gbl>,
	<1A022EDB-7C1B-4916-BFD2-C5AFDED3869D@jerris.com>,
	<CAMbW9o+znPJ-P59qR-3oPCFv5mXzWeokoP1ftgny37PQa3t0Gw@mail.gmail.com>,
	<96F59C65-678A-41BF-A062-370F244961B8@jerris.com>,
	<57028A2A.5040407@mst.edu>,
	<D663B890-B3A1-40F1-8113-259CDF354E4D@jerris.com>,
	<CAPQovyQorkHCdxxuj10bzG3nQywG3QPvzudAdLsuHXJ2h-8p6g@mail.gmail.com>
Message-ID: <DUB112-W668664F48F62ED5D853F80BD9D0@phx.gbl>

Hi Guys,
Thank you so much for the responses.
Michael, how much would it cost, do you have a ball-park figure? And also time scale?
Many thanks
Jon

From: Michael Jerris <mike at jerris.com>
Date: Mon, Apr 4, 2016 at 4:58 PM
Subject: Re: [Freeswitch-users] Hash wav file instead of pound for # digit UK
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>


We can do a recording session to get a handful recorded... Is anyone out there able to fund it?  If so, lets collect up a list.

On Apr 4, 2016, at 11:37 AM, Nathan Neulinger <nneul at mst.edu> wrote:
It sounds like in this case though a full language pack isn't strictly required - just an additional word/sound from Callie. Is there a queue/process/etc. for requesting (and funding?) additional words/phrases the next time phrases are ordered?I had previously put in a jira in requesting/suggesting a couple of phrases (FS-8733) related to conferencing. I could see this being a good list to add:	"this"	"conference"	"is"	"being"	"recorded"	"are"	"transferred"	"hash"	"is being recorded"-- NathanOn 04/04/2016 10:24 AM, Michael Jerris wrote:If we can crowd fund that, or if someone has one to donate, we'd be happy to host it.

On Apr 4, 2016, at 11:18 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>>
wrote:

There are commercial British English packs, but a community owned one would be great.

On 4 April 2016 at 15:45, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:

   we don't have a uk english sounds set, you would need to create your own.


   On Apr 4, 2016, at 7:55 AM, Jonathan Hunter <hunterj91 at hotmail.com <mailto:hunterj91 at hotmail.com>> wrote:

   Hi All,

   Is there a .wav file which says 'hash' instead of pound which can be applied for users in the UK/GB region?

   Looking through the ascii files  I see 35 is 'pound' but ideally I need 'hash'.

   Is one available or do I need to create my own?

   Many thanks

   J


_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

http://www.cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at 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 HunterTechnical Director /Telephony DeveloperM:(+44) 7917 190 438Email:jhunter at voxboxcoms.co.uk

 		 	   		  
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From brian at freeswitch.org  Mon Apr  4 20:47:28 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 4 Apr 2016 11:47:28 -0500
Subject: [Freeswitch-users] Hash wav file instead of pound for # digit UK
In-Reply-To: <DUB112-W668664F48F62ED5D853F80BD9D0@phx.gbl>
References: <DUB112-W1396D72A031B061777D73EBD9D0@phx.gbl>
	<1A022EDB-7C1B-4916-BFD2-C5AFDED3869D@jerris.com>
	<CAMbW9o+znPJ-P59qR-3oPCFv5mXzWeokoP1ftgny37PQa3t0Gw@mail.gmail.com>
	<96F59C65-678A-41BF-A062-370F244961B8@jerris.com>
	<57028A2A.5040407@mst.edu>
	<D663B890-B3A1-40F1-8113-259CDF354E4D@jerris.com>
	<CAPQovyQorkHCdxxuj10bzG3nQywG3QPvzudAdLsuHXJ2h-8p6g@mail.gmail.com>
	<DUB112-W668664F48F62ED5D853F80BD9D0@phx.gbl>
Message-ID: <CAEJMVkDqz8rc_vmgh_0TMtUx3VMP6wezT2trmcpFN7CCQj3cVw@mail.gmail.com>

Its probably a 200-300 US to do it.

On Mon, Apr 4, 2016 at 11:40 AM, Jonathan Hunter <hunterj91 at hotmail.com>
wrote:

> Hi Guys,
>
> Thank you so much for the responses.
>
> Michael, how much would it cost, do you have a ball-park figure? And also
> time scale?
>
> Many thanks
>
> Jon
>
>
> From: *Michael Jerris* <mike at jerris.com>
> Date: Mon, Apr 4, 2016 at 4:58 PM
> Subject: Re: [Freeswitch-users] Hash wav file instead of pound for # digit
> UK
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
>
> We can do a recording session to get a handful recorded... Is anyone out
> there able to fund it?  If so, lets collect up a list.
>
>
> On Apr 4, 2016, at 11:37 AM, Nathan Neulinger <nneul at mst.edu> wrote:
>
> It sounds like in this case though a full language pack isn't strictly
> required - just an additional word/sound from
> Callie. Is there a queue/process/etc. for requesting (and funding?)
> additional words/phrases the next time phrases are
> ordered?
>
> I had previously put in a jira in requesting/suggesting a couple of
> phrases (FS-8733) related to conferencing. I could
> see this being a good list to add:
>
> "this"
> "conference"
> "is"
> "being"
> "recorded"
> "are"
> "transferred"
> "hash"
>
> "is being recorded"
>
> -- Nathan
>
> On 04/04/2016 10:24 AM, Michael Jerris wrote:
>
> If we can crowd fund that, or if someone has one to donate, we'd be happy
> to host it.
>
> On Apr 4, 2016, at 11:18 AM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk
> <andrew at cassidywebservices.co.uk>>>
> wrote:
>
> There are commercial British English packs, but a community owned one
> would be great.
>
> On 4 April 2016 at 15:45, Michael Jerris <mike at jerris.com <
> mailto:mike at jerris.com <mike at jerris.com>>> wrote:
>
>    we don't have a uk english sounds set, you would need to create your
> own.
>
>
>    On Apr 4, 2016, at 7:55 AM, Jonathan Hunter <hunterj91 at hotmail.com <
> mailto:hunterj91 at hotmail.com <hunterj91 at hotmail.com>>> wrote:
>
>    Hi All,
>
>    Is there a .wav file which says 'hash' instead of pound which can be
> applied for users in the UK/GB region?
>
>    Looking through the ascii files  I see 35 is 'pound' but ideally I need
> 'hash'.
>
>    Is one available or do I need to create my own?
>
>    Many thanks
>
>    J
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
https://www.gofundme.com/freeswitch_ubuntu

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From hunterj91 at hotmail.com  Mon Apr  4 21:32:01 2016
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Mon, 4 Apr 2016 17:32:01 +0000
Subject: [Freeswitch-users] Hash wav file instead of pound for # digit UK
In-Reply-To: <CAPQovyRtwKCt92m=dMYqT3Ox3UvmOUB3uN6+PT8AoQAL7XK5rw@mail.gmail.com>
References: <DUB112-W1396D72A031B061777D73EBD9D0@phx.gbl>,
	<1A022EDB-7C1B-4916-BFD2-C5AFDED3869D@jerris.com>,
	<CAMbW9o+znPJ-P59qR-3oPCFv5mXzWeokoP1ftgny37PQa3t0Gw@mail.gmail.com>,
	<96F59C65-678A-41BF-A062-370F244961B8@jerris.com>,
	<57028A2A.5040407@mst.edu>,
	<D663B890-B3A1-40F1-8113-259CDF354E4D@jerris.com>,
	<CAPQovyQorkHCdxxuj10bzG3nQywG3QPvzudAdLsuHXJ2h-8p6g@mail.gmail.com>,
	<DUB112-W668664F48F62ED5D853F80BD9D0@phx.gbl>,
	<CAEJMVkDqz8rc_vmgh_0TMtUx3VMP6wezT2trmcpFN7CCQj3cVw@mail.gmail.com>,
	<CAPQovyRtwKCt92m=dMYqT3Ox3UvmOUB3uN6+PT8AoQAL7XK5rw@mail.gmail.com>
Message-ID: <DUB112-W46B91A79BEB9D28657D1E6BD9D0@phx.gbl>

Hi Brian,
I am happy to contribute/pay for that to give something back no problem.
Thanks
Jon

---------- Forwarded message ----------
From: Brian West <brian at freeswitch.org>
Date: Mon, Apr 4, 2016 at 5:47 PM
Subject: Re: [Freeswitch-users] Hash wav file instead of pound for # digit UK
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>


Its probably a 200-300 US to do it.
On Mon, Apr 4, 2016 at 11:40 AM, Jonathan Hunter <hunterj91 at hotmail.com> wrote:



Hi Guys,
Thank you so much for the responses.
Michael, how much would it cost, do you have a ball-park figure? And also time scale?
Many thanks
Jon

From: Michael Jerris <mike at jerris.com>
Date: Mon, Apr 4, 2016 at 4:58 PM
Subject: Re: [Freeswitch-users] Hash wav file instead of pound for # digit UK
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>


We can do a recording session to get a handful recorded... Is anyone out there able to fund it?  If so, lets collect up a list.

On Apr 4, 2016, at 11:37 AM, Nathan Neulinger <nneul at mst.edu> wrote:
It sounds like in this case though a full language pack isn't strictly required - just an additional word/sound from Callie. Is there a queue/process/etc. for requesting (and funding?) additional words/phrases the next time phrases are ordered?I had previously put in a jira in requesting/suggesting a couple of phrases (FS-8733) related to conferencing. I could see this being a good list to add:	"this"	"conference"	"is"	"being"	"recorded"	"are"	"transferred"	"hash"	"is being recorded"-- NathanOn 04/04/2016 10:24 AM, Michael Jerris wrote:If we can crowd fund that, or if someone has one to donate, we'd be happy to host it.

On Apr 4, 2016, at 11:18 AM, Andrew Cassidy <andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>>
wrote:

There are commercial British English packs, but a community owned one would be great.

On 4 April 2016 at 15:45, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:

   we don't have a uk english sounds set, you would need to create your own.


   On Apr 4, 2016, at 7:55 AM, Jonathan Hunter <hunterj91 at hotmail.com <mailto:hunterj91 at hotmail.com>> wrote:

   Hi All,

   Is there a .wav file which says 'hash' instead of pound which can be applied for users in the UK/GB region?

   Looking through the ascii files  I see 35 is 'pound' but ideally I need 'hash'.

   Is one available or do I need to create my own?

   Many thanks

   J


_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

http://www.cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at 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 HunterTechnical Director /Telephony DeveloperM:(+44) 7917 190 438Email: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


-- 








Brian West
brian at freeswitch.org



Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com
https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here! | Reddit: /r/freeswitch

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest



_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

http://www.cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org


-- 
Jonathan HunterTechnical Director /Telephony DeveloperM:(+44) 7917 190 438Email:jhunter at voxboxcoms.co.uk

 		 	   		  
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From vincent.gire at gmail.com  Mon Apr  4 22:09:24 2016
From: vincent.gire at gmail.com (Vincent Gire)
Date: Mon, 4 Apr 2016 20:09:24 +0200
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
Message-ID: <CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>

Hello Sergey,

Thank you for your answer.
I've looked into webdav mounted filesystem.

Unfortunately, most WebDav clients (especially davfs2 on debian) do a lot
of buffering, caching and even lock-null requests (lock a non existent
resource before writing to it). I also suspect that they wait for the end
of the write operation.
The result is a latency of a few seconds witch is not much better than what
I achieve with mod_shout if I transcode the MP3.

Any other idea ?

Thank you !

Best regards

Vincent

On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Please look at webdav mounted filesystem.
>
> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com> wrote:
>
>> Hi all,
>>
>> Thank you to all contributing to FreeSWITCH !
>>
>> I'm working on a IVR project where logic is implemented on a HTTP server.
>> We are leaving Twilio because we now need to record and live stream the
>> session to the HTTP server in WAV format (chunked transfer encoding).
>>
>> *mod_httapi* looks great (HT TAPI very similar to Twilio's) but it seems
>> that the records are first saved to disk before there are sent to the
>> server as chunked data.
>> We need the transfer to start as soon as the recording starts.
>>
>> *mod_shout* does start the request almost as the records starts but it
>> does not support WAV file and shout:// is not exactly a HTTP request
>> (SOURCE method instead of PUT).
>>
>> Is there a way to use these modules to achieve our goal ?
>>
>> If not, we are willing to author a specific module or rather contribute
>> to the existing ones.
>>
>> We've identified two approaches:
>>
>>    1. From *mod_httapi* Modify mod_httapi to directly stream the record
>>    instead of completely saving it to disk before the HTTP chunked transfer
>>    starts.
>>    This seems the most logical but with more than 3000 lines, mod_httapi
>>    does not seem to be the easiest module to build upon for newcomers!
>>
>>    2. From *mod_shout*
>>       1. Modify libshoot to replace the custom SOURCE method with
>>       standard HTTP PUT method
>>       2. Modify mod_shout to support wav files
>>       3. Implement our IVR in script (javascript/lua)
>>
>> What do you think ?
>>
>> Thank you for your help.
>>
>> Cheers,
>> Vincent
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Vincent Gire
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From tom at tomlynn.com  Mon Apr  4 22:20:25 2016
From: tom at tomlynn.com (Tom Lynn)
Date: Mon, 4 Apr 2016 11:20:25 -0700
Subject: [Freeswitch-users] Unable to conmplie mod_shout
In-Reply-To: <CAEwTsAgOapNFK9vKuLSz++4+W9oGJrgkpUt4HagVD7QDkPfDgQ@mail.gmail.com>
References: <957799883.93209.1444803762558.JavaMail.yahoo@mail.yahoo.com>
	<CAEwTsAgOapNFK9vKuLSz++4+W9oGJrgkpUt4HagVD7QDkPfDgQ@mail.gmail.com>
Message-ID: <CAERdjiY+g+3woe3+F8WtC3Jb=xX99SSqWeGuL_ephyu_Efi1Jg@mail.gmail.com>

I had a similar problem under Raspbian Jessie, FreeSWITCH Version
1.7.0+git~20160402T205739Z~4e96a9e6bb~32bit (git 4e96a9e 2016-04-02
20:57:39Z 32bit)  I'd installed using directions from Algis Salys, recently
prominently displayed on the Freeswitch homepage.

make would not recognize libmpg123-dev until in went back to the configure
step and left off the -C option.  Likewise, I left off the -j3 options when
using make and then all proceeded normally.

On Wed, Oct 14, 2015 at 6:27 AM, Michael Jerris <mike at jerris.com> wrote:

> This means configure did not detect it, maybe you installed after you ran
> configure or it's an error in the Ubuntu package ?
>
>
> On Wednesday, October 14, 2015, sunil more <sunil.more64 at yahoo.in> wrote:
>
>> Hello All,
>> I am trying to compile freeswitch latest commit on Ubuntu Server and I am
>> getting stuck while compiling mod_shout. It asked me for adding dependency
>> libmpg123-dev. I have this lib package already installed on my machine.
>> However i still get stuck over this error. Please help me.
>> Operating system:
>> Distributor ID: Ubuntu
>> Description: Ubuntu 14.04.3 LTS
>> Release: 14.04
>> Codename: trusty
>>
>> Error:
>> making all mod_shoutmake[4]: Entering directory
>> `/usr/local/src/freeswitch/src/mod/formats/mod_shout'
>> Makefile:895: *** You must install libmpg123-dev to build mod_shout.
>> Stop.
>> make[4]: Leaving directory
>> `/usr/local/src/freeswitch/src/mod/formats/mod_shout'
>> make[3]: *** [mod_shout-all] Error 1
>> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>> make[2]: *** [all-recursive] Error 1
>> make[2]: Leaving directory `/usr/local/src/freeswitch/src'
>> make[1]: *** [all-recursive] Error 1
>> make[1]: Leaving directory `/usr/local/src/freeswitch'
>> make: *** [all] Error 2
>>
>> Regards,
>> Sunil More
>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ovoshlook at gmail.com  Mon Apr  4 22:48:35 2016
From: ovoshlook at gmail.com (Yuriy Gorlichenko)
Date: Mon, 4 Apr 2016 21:48:35 +0300
Subject: [Freeswitch-users] FS not see invite
In-Reply-To: <B07B3220-0A4E-4264-9D24-9FD9F75942DE@jerris.com>
References: <CABSP_VfiEzaP58j0jSA8dd8D6psubk9QGvpZ=i46xBVqNVE=Dg@mail.gmail.com>
	<B07B3220-0A4E-4264-9D24-9FD9F75942DE@jerris.com>
Message-ID: <CABSP_VdFOJY_+cs1hsEVrAyY4GpynhYT7DBDVpVpvUSMgpdqXA@mail.gmail.com>

Hi yes. Im shure. Also yes . right ip and port.
It is looks like fw issue but it is turned off. I see invite with auth at
TCPDUMP but not see it at the fs_cli

Can it be some troubles with internal FS blacklist or thomething like this?
I just a beginner with FS. Previously worked with Asterisk.


2016-04-04 17:48 GMT+03:00 Michael Jerris <mike at jerris.com>:

> are you SURE the firewall is turned off?  Is the sip going to the right
> iP/port that FreeSWITCH is listening on?
>
> On Apr 3, 2016, at 3:07 PM, Yuriy Gorlichenko <ovoshlook at gmail.com> wrote:
>
> Hello. I try to make a call through another SIP instalse to freeswitch.
> I added extension and use FS as proxy to rpovider.
> I successfully registered on FS as user and try to make a call.
>
> I see at FS cli that invite arrives and FS answers 407, next my instanse
> Sends ACK and thn INVITE with a digest auth.
>
> I see thins invite at the TCPDUMP on the FS machine but not see it at the
> FS
> Firewall is disabled for tests....
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From s.safarov at gmail.com  Mon Apr  4 22:50:02 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 04 Apr 2016 18:50:02 +0000
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
Message-ID: <CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>

Input/output latency is not problem.  I use Kazoo on my servers and call
recording is stored to database during 5 seconds after hangup.
What is broken in your case if save file using webdav or http_cache?

On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com> wrote:

> Hello Sergey,
>
> Thank you for your answer.
> I've looked into webdav mounted filesystem.
>
> Unfortunately, most WebDav clients (especially davfs2 on debian) do a lot
> of buffering, caching and even lock-null requests (lock a non existent
> resource before writing to it). I also suspect that they wait for the end
> of the write operation.
> The result is a latency of a few seconds witch is not much better than
> what I achieve with mod_shout if I transcode the MP3.
>
> Any other idea ?
>
> Thank you !
>
> Best regards
>
> Vincent
>
> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Please look at webdav mounted filesystem.
>>
>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com> wrote:
>>
>>> Hi all,
>>>
>>> Thank you to all contributing to FreeSWITCH !
>>>
>>> I'm working on a IVR project where logic is implemented on a HTTP server.
>>> We are leaving Twilio because we now need to record and live stream the
>>> session to the HTTP server in WAV format (chunked transfer encoding).
>>>
>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's) but it
>>> seems that the records are first saved to disk before there are sent to the
>>> server as chunked data.
>>> We need the transfer to start as soon as the recording starts.
>>>
>>> *mod_shout* does start the request almost as the records starts but it
>>> does not support WAV file and shout:// is not exactly a HTTP request
>>> (SOURCE method instead of PUT).
>>>
>>> Is there a way to use these modules to achieve our goal ?
>>>
>>> If not, we are willing to author a specific module or rather contribute
>>> to the existing ones.
>>>
>>> We've identified two approaches:
>>>
>>>    1. From *mod_httapi* Modify mod_httapi to directly stream the record
>>>    instead of completely saving it to disk before the HTTP chunked transfer
>>>    starts.
>>>    This seems the most logical but with more than 3000 lines,
>>>    mod_httapi does not seem to be the easiest module to build upon for
>>>    newcomers!
>>>
>>>    2. From *mod_shout*
>>>       1. Modify libshoot to replace the custom SOURCE method with
>>>       standard HTTP PUT method
>>>       2. Modify mod_shout to support wav files
>>>       3. Implement our IVR in script (javascript/lua)
>>>
>>> What do you think ?
>>>
>>> Thank you for your help.
>>>
>>> Cheers,
>>> Vincent
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Vincent Gire
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Mon Apr  4 22:50:58 2016
From: omortimer at gmail.com (Oz Mortimer)
Date: Mon, 4 Apr 2016 19:50:58 +0100
Subject: [Freeswitch-users] FS not see invite
In-Reply-To: <CABSP_VdFOJY_+cs1hsEVrAyY4GpynhYT7DBDVpVpvUSMgpdqXA@mail.gmail.com>
References: <CABSP_VfiEzaP58j0jSA8dd8D6psubk9QGvpZ=i46xBVqNVE=Dg@mail.gmail.com>
	<B07B3220-0A4E-4264-9D24-9FD9F75942DE@jerris.com>
	<CABSP_VdFOJY_+cs1hsEVrAyY4GpynhYT7DBDVpVpvUSMgpdqXA@mail.gmail.com>
Message-ID: <CAKOBZXOhfE9tXVsmx8feLYf1dv0xx7JBQTDpA5E6xXmOntztpA@mail.gmail.com>

have you checked that the sofia profile is running (sofia status)? might be
that its not listening, or listening on incorrect IP.

On 4 April 2016 at 19:48, Yuriy Gorlichenko <ovoshlook at gmail.com> wrote:

> Hi yes. Im shure. Also yes . right ip and port.
> It is looks like fw issue but it is turned off. I see invite with auth at
> TCPDUMP but not see it at the fs_cli
>
> Can it be some troubles with internal FS blacklist or thomething like
> this? I just a beginner with FS. Previously worked with Asterisk.
>
>
> 2016-04-04 17:48 GMT+03:00 Michael Jerris <mike at jerris.com>:
>
>> are you SURE the firewall is turned off?  Is the sip going to the right
>> iP/port that FreeSWITCH is listening on?
>>
>> On Apr 3, 2016, at 3:07 PM, Yuriy Gorlichenko <ovoshlook at gmail.com>
>> wrote:
>>
>> Hello. I try to make a call through another SIP instalse to freeswitch.
>> I added extension and use FS as proxy to rpovider.
>> I successfully registered on FS as user and try to make a call.
>>
>> I see at FS cli that invite arrives and FS answers 407, next my instanse
>> Sends ACK and thn INVITE with a digest auth.
>>
>> I see thins invite at the TCPDUMP on the FS machine but not see it at the
>> FS
>> Firewall is disabled for tests....
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  4 22:53:39 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Apr 2016 14:53:39 -0400
Subject: [Freeswitch-users] FS not see invite
In-Reply-To: <CABSP_VdFOJY_+cs1hsEVrAyY4GpynhYT7DBDVpVpvUSMgpdqXA@mail.gmail.com>
References: <CABSP_VfiEzaP58j0jSA8dd8D6psubk9QGvpZ=i46xBVqNVE=Dg@mail.gmail.com>
	<B07B3220-0A4E-4264-9D24-9FD9F75942DE@jerris.com>
	<CABSP_VdFOJY_+cs1hsEVrAyY4GpynhYT7DBDVpVpvUSMgpdqXA@mail.gmail.com>
Message-ID: <21DA600E-2ABF-4E0F-864F-D9B8DEEBA485@jerris.com>

If you have the sip trace enabled in freeswitch and you still don't see it, either you are sending to the wrong Ip/port, or firewall is dropping it.


> On Apr 4, 2016, at 2:48 PM, Yuriy Gorlichenko <ovoshlook at gmail.com> wrote:
> 
> Hi yes. Im shure. Also yes . right ip and port.
> It is looks like fw issue but it is turned off. I see invite with auth at TCPDUMP but not see it at the fs_cli
> 
> Can it be some troubles with internal FS blacklist or thomething like this? I just a beginner with FS. Previously worked with Asterisk.
> 
> 
> 2016-04-04 17:48 GMT+03:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> are you SURE the firewall is turned off?  Is the sip going to the right iP/port that FreeSWITCH is listening on?
> 
>> On Apr 3, 2016, at 3:07 PM, Yuriy Gorlichenko <ovoshlook at gmail.com <mailto:ovoshlook at gmail.com>> wrote:
>> 
>> Hello. I try to make a call through another SIP instalse to freeswitch. 
>> I added extension and use FS as proxy to rpovider.
>> I successfully registered on FS as user and try to make a call.
>> 
>> I see at FS cli that invite arrives and FS answers 407, next my instanse Sends ACK and thn INVITE with a digest auth.
>> 
>> I see thins invite at the TCPDUMP on the FS machine but not see it at the FS
>> Firewall is disabled for tests....
> 

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From ahabiba at gmail.com  Mon Apr  4 23:37:46 2016
From: ahabiba at gmail.com (Ahmed habiba)
Date: Mon, 4 Apr 2016 22:37:46 +0300
Subject: [Freeswitch-users] IMT Codec Enforcement
Message-ID: <19A18F2D-ECB7-4B04-BC5C-9DC2733056A8@gmail.com>

Dears,

kindly I have the below scenario:

[SoftPhone1]<??Channel1??>[FreeSwitch1]<??Channel2??>[FreeSwitch2]<??Chanel3??>[SoftPhone2]

Channel1 codec G722

Channel3 codec G722

I want to enforce Channel 3 between FreeSwitch instances to be GSM, I change the Dial plane of freeSwitch1 to and the same for FS2

                  <action application="set" data=?absolute_codec_string=GSM"/>

                  <action application="bridge" data="sofia/internal/$1 at 10.8.0.22"/>

However when I run call from FS2 to FS1, FS1 reply with the below putting ?a=rtpmap:106 opus/48000/2? as the only available codec as below, your kind help will be appreciated



recv 1275 bytes from udp/[10.8.0.22]:5060 at 15:22:35.560845:
   ------------------------------------------------------------------------
   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP 10.8.0.29;rport=5060;branch=z9hG4bKSr1v2mgcctX5H
   From: "Extension 1000" <sip:1000 at 10.8.0.29>;tag=U82NjmSa9S9Kg
   To: <sip:1000 at 10.8.0.22>;tag=g0eNeprr8e4Fg
   Call-ID: 6d5dbd7f-753d-1234-d78c-bc305bd3840c
   CSeq: 89562813 INVITE
   Contact: <sip:1000 at 10.8.0.22:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.6.6+git~20160111T201612Z~d2d0b3283a~64bit
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 261
   X-FS-Display-Name: 1000
   X-FS-Display-Number: sip:1000 at 10.8.0.22
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "1000" <sip:1000 at 10.8.0.22>;party=calling;privacy=off;screen=no
   
   v=0
   o=FreeSWITCH 1459771930 1459771931 IN IP4 10.8.0.22
   s=FreeSWITCH
   c=IN IP4 10.8.0.22
   t=0 0
   m=audio 25836 RTP/AVP 106 101
   a=rtpmap:106 opus/48000/2
   a=fmtp:106 useinbandfec=1; minptime=20
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20

Thanks,

Ahmed Habiba.
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From ehsanminachi at gmail.com  Tue Apr  5 02:34:18 2016
From: ehsanminachi at gmail.com (Ehsan Minachi)
Date: Mon, 4 Apr 2016 15:34:18 -0700
Subject: [Freeswitch-users] Using lib sofia-sip outside of FreeSWITCH
Message-ID: <CAPDpkrBUsQJ1dcKeqM3s2aQr8sUAyQLRYLYWuEvXC7o71L+jEg@mail.gmail.com>

Hello everyone,

I know FreeSWITCH maintains Sofia library internally. I'm building a
distributed presence server using sofia, however I'm having hard time
linking it against current supplied version of sofia lib under libs
directory.

I would very much prefer to link my app against the FreeSWITCH maintained
lib vs. the one found over source forge.

Any hints on how I should call ./configure on libs/sofia-sip is highly
appreciated.

Thank you very much.

Best regards,

- Ehsan
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From mike at jerris.com  Tue Apr  5 03:09:44 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 4 Apr 2016 19:09:44 -0400
Subject: [Freeswitch-users] Using lib sofia-sip outside of FreeSWITCH
In-Reply-To: <CAPDpkrBUsQJ1dcKeqM3s2aQr8sUAyQLRYLYWuEvXC7o71L+jEg@mail.gmail.com>
References: <CAPDpkrBUsQJ1dcKeqM3s2aQr8sUAyQLRYLYWuEvXC7o71L+jEg@mail.gmail.com>
Message-ID: <CEC5200A-0009-4A10-AC70-7966C603179C@jerris.com>

you probably want to do this if you want an installable library:

iff --git a/libs/sofia-sip/libsofia-sip-ua/Makefile.am b/libs/sofia-sip/libsofia-sip-ua/Makefile.am
index d2855be..576a1dd 100644
--- a/libs/sofia-sip/libsofia-sip-ua/Makefile.am
+++ b/libs/sofia-sip/libsofia-sip-ua/Makefile.am
@@ -35,7 +35,7 @@ DIST_SUBDIRS = su features bnf sresolv ipt sdp url msg sip http \
 
 DOXYGEN = doxygen
 
-noinst_LTLIBRARIES = libsofia-sip-ua.la
+lib_LTLIBRARIES = libsofia-sip-ua.la
 
 libsofia_sip_ua_la_SOURCES =
 libsofia_sip_ua_la_LIBADD =    bnf/libbnf.la \


If you are still having issues, I'd need more details of whats going on.

> On Apr 4, 2016, at 6:34 PM, Ehsan Minachi <ehsanminachi at gmail.com> wrote:
> 
> Hello everyone,
> 
> I know FreeSWITCH maintains Sofia library internally. I'm building a distributed presence server using sofia, however I'm having hard time linking it against current supplied version of sofia lib under libs directory.
> 
> I would very much prefer to link my app against the FreeSWITCH maintained lib vs. the one found over source forge.
> 
> Any hints on how I should call ./configure on libs/sofia-sip is highly appreciated.
> 
> Thank you very much.
> 
> Best regards,
> 
> - Ehsan
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ehsanminachi at gmail.com  Tue Apr  5 05:23:53 2016
From: ehsanminachi at gmail.com (Ehsan Minachi)
Date: Mon, 4 Apr 2016 18:23:53 -0700
Subject: [Freeswitch-users] Using lib sofia-sip outside of FreeSWITCH
In-Reply-To: <CEC5200A-0009-4A10-AC70-7966C603179C@jerris.com>
References: <CAPDpkrBUsQJ1dcKeqM3s2aQr8sUAyQLRYLYWuEvXC7o71L+jEg@mail.gmail.com>
	<CEC5200A-0009-4A10-AC70-7966C603179C@jerris.com>
Message-ID: <CAPDpkrBNzgG6e0fxWguQgGqpy2BVVdvvTFXsm-9RFNzXjrtntw@mail.gmail.com>

Awesome! Thank you so much Michael, that did the trick.

On Mon, Apr 4, 2016 at 4:09 PM, Michael Jerris <mike at jerris.com> wrote:

> you probably want to do this if you want an installable library:
>
> iff --git a/libs/sofia-sip/libsofia-sip-ua/Makefile.am
> b/libs/sofia-sip/libsofia-sip-ua/Makefile.am
> index d2855be..576a1dd 100644
> --- a/libs/sofia-sip/libsofia-sip-ua/Makefile.am
> +++ b/libs/sofia-sip/libsofia-sip-ua/Makefile.am
> @@ -35,7 +35,7 @@ DIST_SUBDIRS = su features bnf sresolv ipt sdp url msg
> sip http \
>
>
>  DOXYGEN = doxygen
>
>
> -noinst_LTLIBRARIES = libsofia-sip-ua.la
> +lib_LTLIBRARIES = libsofia-sip-ua.la
>
>
>  libsofia_sip_ua_la_SOURCES =
>  libsofia_sip_ua_la_LIBADD =    bnf/libbnf.la \
>
>
> If you are still having issues, I'd need more details of whats going on.
>
> On Apr 4, 2016, at 6:34 PM, Ehsan Minachi <ehsanminachi at gmail.com> wrote:
>
> Hello everyone,
>
> I know FreeSWITCH maintains Sofia library internally. I'm building a
> distributed presence server using sofia, however I'm having hard time
> linking it against current supplied version of sofia lib under libs
> directory.
>
> I would very much prefer to link my app against the FreeSWITCH maintained
> lib vs. the one found over source forge.
>
> Any hints on how I should call ./configure on libs/sofia-sip is highly
> appreciated.
>
> Thank you very much.
>
> Best regards,
>
> - Ehsan
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From quanhs at stee.stengg.com  Tue Apr  5 06:32:33 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Tue, 5 Apr 2016 02:32:33 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION
	ERROR. SDP
Message-ID: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>

Hi All;

I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min),just following the information in book "FreeSWITCH 1.6 Cookbook".
If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
But if disabling media bypass mode, call is rejected by FS.

FS log shows "[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP". Attachment is detailed FS log file.
Chrome uses opus 111, FS uses opus 116.
Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().

Help is needed to troubleshoot this issue.

Thanks advance.
Smile.


[This e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
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---ST Electronics Group---
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From ovoshlook at gmail.com  Tue Apr  5 07:59:07 2016
From: ovoshlook at gmail.com (Yuriy Gorlichenko)
Date: Tue, 5 Apr 2016 06:59:07 +0300
Subject: [Freeswitch-users] FS not see invite
In-Reply-To: <21DA600E-2ABF-4E0F-864F-D9B8DEEBA485@jerris.com>
References: <CABSP_VfiEzaP58j0jSA8dd8D6psubk9QGvpZ=i46xBVqNVE=Dg@mail.gmail.com>
	<B07B3220-0A4E-4264-9D24-9FD9F75942DE@jerris.com>
	<CABSP_VdFOJY_+cs1hsEVrAyY4GpynhYT7DBDVpVpvUSMgpdqXA@mail.gmail.com>
	<21DA600E-2ABF-4E0F-864F-D9B8DEEBA485@jerris.com>
Message-ID: <CABSP_VeCQgR_rTUg2=SwmKGHA9Em8E7y+bySYtdgrfevx5AtuQ@mail.gmail.com>

I making registrations from Microsip client and all works fine so sofia
Mode works. It happens only with another instanse that i want to connect to
FS. firewall disabed. Also I see invites without auth from instance. I can
not see only IVITE with auth params. And only at the fs_cli.

I cnow it sounds strandge but it is a fact... Thats why am asking about
inernal blaqcklists at the fs.

2016-04-04 21:53 GMT+03:00 Michael Jerris <mike at jerris.com>:

> If you have the sip trace enabled in freeswitch and you still don't see
> it, either you are sending to the wrong Ip/port, or firewall is dropping it.
>
>
> On Apr 4, 2016, at 2:48 PM, Yuriy Gorlichenko <ovoshlook at gmail.com> wrote:
>
> Hi yes. Im shure. Also yes . right ip and port.
> It is looks like fw issue but it is turned off. I see invite with auth at
> TCPDUMP but not see it at the fs_cli
>
> Can it be some troubles with internal FS blacklist or thomething like
> this? I just a beginner with FS. Previously worked with Asterisk.
>
>
> 2016-04-04 17:48 GMT+03:00 Michael Jerris <mike at jerris.com>:
>
>> are you SURE the firewall is turned off?  Is the sip going to the right
>> iP/port that FreeSWITCH is listening on?
>>
>> On Apr 3, 2016, at 3:07 PM, Yuriy Gorlichenko <ovoshlook at gmail.com>
>> wrote:
>>
>> Hello. I try to make a call through another SIP instalse to freeswitch.
>> I added extension and use FS as proxy to rpovider.
>> I successfully registered on FS as user and try to make a call.
>>
>> I see at FS cli that invite arrives and FS answers 407, next my instanse
>> Sends ACK and thn INVITE with a digest auth.
>>
>> I see thins invite at the TCPDUMP on the FS machine but not see it at the
>> FS
>> Firewall is disabled for tests....
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From rutu.patel at inextrix.com  Tue Apr  5 09:16:52 2016
From: rutu.patel at inextrix.com (Rutu Patel)
Date: Tue, 5 Apr 2016 10:46:52 +0530
Subject: [Freeswitch-users] 480 Temporarily
	Unavailable[MANDATORY_IE_MISSING]
In-Reply-To: <CAOrA2UZEYXX_bdThsjcqjdUXqtR9oHMjkr2k-FT0WAU-O9Y=9g@mail.gmail.com>
References: <CAPQet1zRj3sogx3csggUCe9zkeCLN2X=99fr=z73WkLvD5=bPA@mail.gmail.com>
	<CAOrA2UYZTKucCxUC3XbB=yG_Z9ZOTOZpRH_MvrgLHhqoHnVc8g@mail.gmail.com>
	<CAKbxfG_BacH6FQqEPba7w6UTHbKhQsw+7UX9=pCgvr7mo1t=XA@mail.gmail.com>
	<CAPQet1zPPUVHSpNyUVJVvj_df7qfq+ty_yYatuBoTPboaQ0F+Q@mail.gmail.com>
	<CAKzWOxXUCNjRXGy=UPDNSQyZUx8i5eJ-iw9Xs4gYKdJR5RA3aA@mail.gmail.com>
	<CAPQet1wsOwEC3hWFqKns0-0gTOkMEs94B6D8mxp8HwyGsWqPHg@mail.gmail.com>
	<CAOrA2Ub=NFpdsQZxsDX2BhmoX-8DLtMFHBBQeJXQtrKSeq6m8A@mail.gmail.com>
	<CAPQet1z+=0rw=VO9Q-b95n4B_Bk-R6VEZvrM5VE5sMd-u4dTgQ@mail.gmail.com>
	<CAOrA2UZEYXX_bdThsjcqjdUXqtR9oHMjkr2k-FT0WAU-O9Y=9g@mail.gmail.com>
Message-ID: <CAPQet1ye8rw18g16DiLyNNSCuBbNd0X=yn5YuT7t2ZdJ-N65hw@mail.gmail.com>

Hi,

Thank you for your help to debug the issue.

We are now trying without opensips but still facing authentication issue.

We created two users of freeswitch in asterisk as peer.

For example, First peer in sip.conf is 456789 and second is 123456.

[123456]
type=friend
host=x.x.x.45
username=123456
secret=abcdef
disallow=all
allow=g729
trustrpid=yes
canreinvite=yes
sendrpid=yes
context=extensions-test
qualify=yes
directmedia=yes

Now when we are trying to reach 123456, the request reach to asterisk but
not authenticate proper user(123456). It authenticate first peer(456789)
defined in sip.conf.

If we define 123456 before 456789 then it works but we have multiple peers
in asterisk so how can we manage that ?

We also tried by sending user credential direct in bridge,but still it is
authenticate the first peer defined in sip.conf .

{sip_auth_username=123456,sip_auth_password=abcdef,sip_contact_user=123456}sofia/default/NUMBER at IP

?Can anyone please explain how to authenticate specific peer ?

Please let me know if you want any other details.?

--
Thanks,
Rutu Patel
??

<http://www.inextrix.com>

On Fri, Apr 1, 2016 at 3:56 PM, Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:

> Hi,
>
> I'm not sure about your environment, but on this point I will recommend
> you to isolate issue. So on this point you should understand if your sip
> trunk between Asterisk and Freeswitch configured correctly, without
> involving Opensips and etc. To test SIP trunk you can try to send call
> directly from freeswitch to Asterisk and see if it works, you can send call
> just directly from freeswitch console, something similar to:
>
> originate sofia/example/300 at foo.com 8600
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_commands
>
> If you call will reach Asterisk without any issue, then this means that
> sip trunk created correctly and issue is somewhere on Opensips side or any
> other part of your set-up.
>
> Second option will be to try to configure sip trunk without password and
> check if it helps
>
> https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
>
> With kind regards,
>
> Jurijs
>
> On Fri, Apr 1, 2016 at 12:54 PM, Rutu Patel <rutu.patel at inextrix.com>
> wrote:
>
>> Hi Juris,
>>
>> Yes we added peer in asterisk sip.conf.
>>
>> Are we still missing any configurations?
>>
>>
>> On Thursday, March 31, 2016, Jurijs Ivolga <jurijs.ivolga at gmail.com>
>> wrote:
>>
>>> Hi Rutu,
>>>
>>> Just a shot in a dark. :)
>>>
>>> Did you added sip peer record in sip.conf, something similar to:
>>>
>>> [freeswitch]
>>> type=peer
>>> host=IP_ADDRESS_OF_FREESWITCH_SERVER
>>> username=HOSTNAME.DOMAIN.COM
>>> port=5080
>>> fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER
>>> secret=BOOTH_WAY_PASSWORD
>>>
>>> With kind regards,
>>>
>>>
>>> Jurijs
>>>
>>> On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <rutu.patel at inextrix.com>
>>> wrote:
>>>
>>>> Hi Michael
>>>>
>>>> Thank you for your assistance.
>>>>
>>>> We are trying to make call between two freeswitch users.
>>>>
>>>> One user is registered in softphone and other is registered in Asterisk
>>>> by register string but asterisk user cannot authenticate properly and
>>>> freeswitch directly send 480 Temporary Unavailable.
>>>>
>>>> In sip profile we set challenge-realm= auto_to also tried with
>>>> auto_from but still not getting success.
>>>>
>>>> I am attaching some more logs if anyone can guide in right direction.
>>>>
>>>> --
>>>> Thanks,
>>>> Rutu Patel
>>>>
>>>>
>>>>
>>>> On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc at freeswitch.org>
>>>> wrote:
>>>>
>>>>> Try a gateway. Look in conf/sip_profiles/external/example.xml for a
>>>>> heavily-commented sample. For more detailed discussion see this nice doc:
>>>>>
>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration
>>>>>
>>>>> -MSC
>>>>>
>>>>> On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel at inextrix.com>
>>>>> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> Thank you for reply.
>>>>>>
>>>>>> We have registered extension of Freeswitch in Asterisk as below
>>>>>>
>>>>>> register=>test-ext:secret at 192.168.1.64/test-ext
>>>>>>
>>>>>> ?We want to send call from Freeswitch to Asterisk?.
>>>>>>
>>>>>> What authentication parameters we need to set in Freeswitch ?
>>>>>>
>>>>>> --
>>>>>> Thanks,
>>>>>> Rutu Patel
>>>>>> On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale <
>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>
>>>>>>> You need to supply auth credentials to Asterisk because its set to
>>>>>>> authenticate calls.
>>>>>>> The best thing to do would be to make a gateway pointing to Asterisk
>>>>>>> with all the credentials and call it using that.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga <
>>>>>>> jurijs.ivolga at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hi Rutu,
>>>>>>>>
>>>>>>>> In logs I can see that user-agent Asterisk is sending INVITE to
>>>>>>>> Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This
>>>>>>>> a bit differs from what you explained in your mail...
>>>>>>>>
>>>>>>>> First think what pop up in my mind is that you need to configure
>>>>>>>> Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my
>>>>>>>> first guess...
>>>>>>>>
>>>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
>>>>>>>>
>>>>>>>> With kind regards,
>>>>>>>>
>>>>>>>> Jurijs
>>>>>>>>
>>>>>>>> On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <
>>>>>>>> rutu.patel at inextrix.com> wrote:
>>>>>>>>
>>>>>>>>> Hi,
>>>>>>>>>
>>>>>>>>> We are sending call to asterisk PBX but we are getting 480
>>>>>>>>> Temporarily Unavailable immediately. from freeswitch server.
>>>>>>>>>
>>>>>>>>> Call flow is as below:
>>>>>>>>>
>>>>>>>>> User => opensips => FS => PBX,
>>>>>>>>>
>>>>>>>>> but call not sending out from FS and drop by FS.
>>>>>>>>>
>>>>>>>>> Here I am attaching both sip logs and freeswitch logs.
>>>>>>>>>
>>>>>>>>> Can anyone please help to sort out the issue.
>>>>>>>>> --
>>>>>>>>> Thanks,
>>>>>>>>> Rutu Patel
>>>>>>>>> <http://www.inextrix.com>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>>>>>
>>>>>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>>>>>> http://twitter.com/FreeSWITCH
>>>>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>>>>>>> <http://freeswitch.org/g+>*
>>>>>>>
>>>>>>> ClueCon Weekly Development Call
>>>>>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>>>>>
>>>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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,
>> Rutu Patel
>> iNextrix Technologies Pvt. Ltd.
>> www.inextrix.com
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vincent.gire at gmail.com  Tue Apr  5 11:30:12 2016
From: vincent.gire at gmail.com (Vincent Gire)
Date: Tue, 5 Apr 2016 09:30:12 +0200
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
Message-ID: <CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>

webdav, mod_http_cache or mod_httapi all results in sending the recording
only *after* it is complete.
They all write the recording to a file, wait for the recording to complete
and the file to close and then send it over HTTP.

I would like to start sending the recording to the remove server as soon as
it starts (max 1 sec latency).
mod_http_cache or mod_httapi would be perfect if they were streaming the
recording like mod_shout.


On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Input/output latency is not problem.  I use Kazoo on my servers and call
> recording is stored to database during 5 seconds after hangup.
> What is broken in your case if save file using webdav or http_cache?
>
> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com> wrote:
>
>> Hello Sergey,
>>
>> Thank you for your answer.
>> I've looked into webdav mounted filesystem.
>>
>> Unfortunately, most WebDav clients (especially davfs2 on debian) do a lot
>> of buffering, caching and even lock-null requests (lock a non existent
>> resource before writing to it). I also suspect that they wait for the end
>> of the write operation.
>> The result is a latency of a few seconds witch is not much better than
>> what I achieve with mod_shout if I transcode the MP3.
>>
>> Any other idea ?
>>
>> Thank you !
>>
>> Best regards
>>
>> Vincent
>>
>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> Please look at webdav mounted filesystem.
>>>
>>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com> wrote:
>>>
>>>> Hi all,
>>>>
>>>> Thank you to all contributing to FreeSWITCH !
>>>>
>>>> I'm working on a IVR project where logic is implemented on a HTTP
>>>> server.
>>>> We are leaving Twilio because we now need to record and live stream the
>>>> session to the HTTP server in WAV format (chunked transfer encoding).
>>>>
>>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's) but it
>>>> seems that the records are first saved to disk before there are sent to the
>>>> server as chunked data.
>>>> We need the transfer to start as soon as the recording starts.
>>>>
>>>> *mod_shout* does start the request almost as the records starts but it
>>>> does not support WAV file and shout:// is not exactly a HTTP request
>>>> (SOURCE method instead of PUT).
>>>>
>>>> Is there a way to use these modules to achieve our goal ?
>>>>
>>>> If not, we are willing to author a specific module or rather contribute
>>>> to the existing ones.
>>>>
>>>> We've identified two approaches:
>>>>
>>>>    1. From *mod_httapi* Modify mod_httapi to directly stream the
>>>>    record instead of completely saving it to disk before the HTTP chunked
>>>>    transfer starts.
>>>>    This seems the most logical but with more than 3000 lines,
>>>>    mod_httapi does not seem to be the easiest module to build upon for
>>>>    newcomers!
>>>>
>>>>    2. From *mod_shout*
>>>>       1. Modify libshoot to replace the custom SOURCE method with
>>>>       standard HTTP PUT method
>>>>       2. Modify mod_shout to support wav files
>>>>       3. Implement our IVR in script (javascript/lua)
>>>>
>>>> What do you think ?
>>>>
>>>> Thank you for your help.
>>>>
>>>> Cheers,
>>>> Vincent
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Vincent Gire
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Vincent Gire
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From s.safarov at gmail.com  Tue Apr  5 12:04:09 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 05 Apr 2016 08:04:09 +0000
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
	<CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
Message-ID: <CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>

Think is requred streaming feature of freeswitch.
Look at mod_esf
<https://freeswitch.org/confluence/display/FREESWITCH/mod_esf> and mod_vlc
<https://wiki.freeswitch.org/wiki/Mod_vlc>
Instruction to compile mod_vlc on provided link is to old but helpfull to
undestand how to stream media to http server.

For compiling mod_vlc please use vlc repo
<https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos
instruction
<https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>
.
After you intall vlc, then you can enable mod_vlc module in freeswitch
sources(SPEC file) and compile freeswitch.

Sergey


??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com>:

> webdav, mod_http_cache or mod_httapi all results in sending the recording
> only *after* it is complete.
> They all write the recording to a file, wait for the recording to
> complete and the file to close and then send it over HTTP.
>
> I would like to start sending the recording to the remove server as soon
> as it starts (max 1 sec latency).
> mod_http_cache or mod_httapi would be perfect if they were streaming the
> recording like mod_shout.
>
>
> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Input/output latency is not problem.  I use Kazoo on my servers and call
>> recording is stored to database during 5 seconds after hangup.
>> What is broken in your case if save file using webdav or http_cache?
>>
>> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com> wrote:
>>
>>> Hello Sergey,
>>>
>>> Thank you for your answer.
>>> I've looked into webdav mounted filesystem.
>>>
>>> Unfortunately, most WebDav clients (especially davfs2 on debian) do a
>>> lot of buffering, caching and even lock-null requests (lock a non existent
>>> resource before writing to it). I also suspect that they wait for the end
>>> of the write operation.
>>> The result is a latency of a few seconds witch is not much better than
>>> what I achieve with mod_shout if I transcode the MP3.
>>>
>>> Any other idea ?
>>>
>>> Thank you !
>>>
>>> Best regards
>>>
>>> Vincent
>>>
>>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <s.safarov at gmail.com>
>>> wrote:
>>>
>>>> Please look at webdav mounted filesystem.
>>>>
>>>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com> wrote:
>>>>
>>>>> Hi all,
>>>>>
>>>>> Thank you to all contributing to FreeSWITCH !
>>>>>
>>>>> I'm working on a IVR project where logic is implemented on a HTTP
>>>>> server.
>>>>> We are leaving Twilio because we now need to record and live stream
>>>>> the session to the HTTP server in WAV format (chunked transfer encoding).
>>>>>
>>>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's) but it
>>>>> seems that the records are first saved to disk before there are sent to the
>>>>> server as chunked data.
>>>>> We need the transfer to start as soon as the recording starts.
>>>>>
>>>>> *mod_shout* does start the request almost as the records starts but
>>>>> it does not support WAV file and shout:// is not exactly a HTTP request
>>>>> (SOURCE method instead of PUT).
>>>>>
>>>>> Is there a way to use these modules to achieve our goal ?
>>>>>
>>>>> If not, we are willing to author a specific module or rather
>>>>> contribute to the existing ones.
>>>>>
>>>>> We've identified two approaches:
>>>>>
>>>>>    1. From *mod_httapi* Modify mod_httapi to directly stream the
>>>>>    record instead of completely saving it to disk before the HTTP chunked
>>>>>    transfer starts.
>>>>>    This seems the most logical but with more than 3000 lines,
>>>>>    mod_httapi does not seem to be the easiest module to build upon for
>>>>>    newcomers!
>>>>>
>>>>>    2. From *mod_shout*
>>>>>       1. Modify libshoot to replace the custom SOURCE method with
>>>>>       standard HTTP PUT method
>>>>>       2. Modify mod_shout to support wav files
>>>>>       3. Implement our IVR in script (javascript/lua)
>>>>>
>>>>> What do you think ?
>>>>>
>>>>> Thank you for your help.
>>>>>
>>>>> Cheers,
>>>>> Vincent
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Vincent Gire
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Vincent Gire
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From vincent.gire at gmail.com  Tue Apr  5 13:36:10 2016
From: vincent.gire at gmail.com (Vincent Gire)
Date: Tue, 5 Apr 2016 11:36:10 +0200
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
	<CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
	<CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>
Message-ID: <CAKsJLpa83aWDBMFuZaYeMFVR6XeN=2uQqwcuTXAtqTY3szckSg@mail.gmail.com>

Ok thanks.
It looks promising !
I'll dig into mod_vlc.

Best

Vincent

On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Think is requred streaming feature of freeswitch.
> Look at mod_esf
> <https://freeswitch.org/confluence/display/FREESWITCH/mod_esf> and mod_vlc
> <https://wiki.freeswitch.org/wiki/Mod_vlc>
> Instruction to compile mod_vlc on provided link is to old but helpfull to
> undestand how to stream media to http server.
>
> For compiling mod_vlc please use vlc repo
> <https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos
> instruction
> <https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>
> .
> After you intall vlc, then you can enable mod_vlc module in freeswitch
> sources(SPEC file) and compile freeswitch.
>
> Sergey
>
>
> ??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com>:
>
>> webdav, mod_http_cache or mod_httapi all results in sending the recording
>> only *after* it is complete.
>> They all write the recording to a file, wait for the recording to
>> complete and the file to close and then send it over HTTP.
>>
>> I would like to start sending the recording to the remove server as soon
>> as it starts (max 1 sec latency).
>> mod_http_cache or mod_httapi would be perfect if they were streaming the
>> recording like mod_shout.
>>
>>
>> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> Input/output latency is not problem.  I use Kazoo on my servers and call
>>> recording is stored to database during 5 seconds after hangup.
>>> What is broken in your case if save file using webdav or http_cache?
>>>
>>> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com> wrote:
>>>
>>>> Hello Sergey,
>>>>
>>>> Thank you for your answer.
>>>> I've looked into webdav mounted filesystem.
>>>>
>>>> Unfortunately, most WebDav clients (especially davfs2 on debian) do a
>>>> lot of buffering, caching and even lock-null requests (lock a non existent
>>>> resource before writing to it). I also suspect that they wait for the end
>>>> of the write operation.
>>>> The result is a latency of a few seconds witch is not much better than
>>>> what I achieve with mod_shout if I transcode the MP3.
>>>>
>>>> Any other idea ?
>>>>
>>>> Thank you !
>>>>
>>>> Best regards
>>>>
>>>> Vincent
>>>>
>>>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <s.safarov at gmail.com>
>>>> wrote:
>>>>
>>>>> Please look at webdav mounted filesystem.
>>>>>
>>>>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi all,
>>>>>>
>>>>>> Thank you to all contributing to FreeSWITCH !
>>>>>>
>>>>>> I'm working on a IVR project where logic is implemented on a HTTP
>>>>>> server.
>>>>>> We are leaving Twilio because we now need to record and live stream
>>>>>> the session to the HTTP server in WAV format (chunked transfer encoding).
>>>>>>
>>>>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's) but it
>>>>>> seems that the records are first saved to disk before there are sent to the
>>>>>> server as chunked data.
>>>>>> We need the transfer to start as soon as the recording starts.
>>>>>>
>>>>>> *mod_shout* does start the request almost as the records starts but
>>>>>> it does not support WAV file and shout:// is not exactly a HTTP request
>>>>>> (SOURCE method instead of PUT).
>>>>>>
>>>>>> Is there a way to use these modules to achieve our goal ?
>>>>>>
>>>>>> If not, we are willing to author a specific module or rather
>>>>>> contribute to the existing ones.
>>>>>>
>>>>>> We've identified two approaches:
>>>>>>
>>>>>>    1. From *mod_httapi* Modify mod_httapi to directly stream the
>>>>>>    record instead of completely saving it to disk before the HTTP chunked
>>>>>>    transfer starts.
>>>>>>    This seems the most logical but with more than 3000 lines,
>>>>>>    mod_httapi does not seem to be the easiest module to build upon for
>>>>>>    newcomers!
>>>>>>
>>>>>>    2. From *mod_shout*
>>>>>>       1. Modify libshoot to replace the custom SOURCE method with
>>>>>>       standard HTTP PUT method
>>>>>>       2. Modify mod_shout to support wav files
>>>>>>       3. Implement our IVR in script (javascript/lua)
>>>>>>
>>>>>> What do you think ?
>>>>>>
>>>>>> Thank you for your help.
>>>>>>
>>>>>> Cheers,
>>>>>> Vincent
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Vincent Gire
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Vincent Gire
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Vincent Gire
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From italo at freeswitch.org  Tue Apr  5 16:58:29 2016
From: italo at freeswitch.org (=?utf-8?q?=C3=8Dtalo_Rossi?=)
Date: Tue, 05 Apr 2016 05:58:29 -0700 (PDT)
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION	ERROR. SDP
In-Reply-To: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
Message-ID: <cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>

Set media_mix_inbound_outbound_codecs=true  
  
?talo Rossi

italo at freeswitch.org

IRC chat.freenode.net #freeswitch #freeswitch-dev

Bugs? https://freeswitch.org/jira

Docs? https://freeswitch.org/jira

Chat? https://hipchat.freeswitch.org/gUdAgy0m6

  

  

On Apr 4 2016, at 11:45 pm, Quan Huo Sheng &lt;quanhs at stee.stengg.com&gt;
wrote:  

> Hi All;

>

>  
>

> I want to use freeswitch to set up a WebRTC POC (Audio only,
udp/tls/rtp/savp).  Setup is Anonymous
(192.168.199.216/chrome/sipjs:sip-0.7.0.min) -&gt;freeswitch (192.168.199.22)
-&gt;1010 (192.168.199.216/chrome/sip-0.7.0.min),just following the
information in book ?FreeSWITCH 1.6 Cookbook?.

>

> If using media bypass mode (inbound-bypass-media == true), all works fine,
caller and called can hear each other.

>

> But if disabling media bypass mode, call is rejected by FS.

>

>  
>

> FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?.
Attachment is detailed FS log file.

>

> Chrome uses opus 111, FS uses opus 116.

>

> Mod_sofia.c::sofia_receive_message() --&gt;
sofia_media.c::sofia_media_negotiate_sdp() --&gt;
sofia_core_media.c::sofia_core_media_negotiate_sdp().

>

>  
>

> Help is needed to troubleshoot this issue.

>

>  
>

> Thanks advance.

>

> Smile.

>

>  
>

>  
[This e-mail is confidential and may be privileged. If you are not the  
intended recipient, please kindly notify us immediately and delete the message  
from your system; please do not copy or use it for any purpose, nor disclose  
its contents to any other person. Thank you.]  
\---ST Electronics Group---

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From italo at freeswitch.org  Tue Apr  5 17:01:01 2016
From: italo at freeswitch.org (=?utf-8?q?=C3=8Dtalo_Rossi?=)
Date: Tue, 05 Apr 2016 06:01:01 -0700 (PDT)
Subject: [Freeswitch-users] Specific ring-device for verto communicator
 (or	webrtc in general)
In-Reply-To: <57028C60.9030306@wirelessmundi.com>
References: <57028C60.9030306@wirelessmundi.com>
Message-ID: <1vzv55159fqgs0xpxejgucjc0-0@mailer.nylas.com>

Yes, you can have as many audio tags you need and each one attached to a
different output device  
  
?talo Rossi

italo at freeswitch.org

IRC chat.freenode.net #freeswitch #freeswitch-dev

Bugs? https://freeswitch.org/jira

Docs? https://freeswitch.org/jira

Chat? https://hipchat.freeswitch.org/gUdAgy0m6

  

  

On Apr 4 2016, at 12:48 pm, Fran?ois &lt;fdelawarde at wirelessmundi.com&gt;
wrote:  

> Hi,

>

> Would it be possible to for a webrtc/verto application to use a  
different audio device for "ringing" purposes (like most softphones do)?

>

> Thanks,  
F

>

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

>

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

>

> FreeSWITCH-users mailing list  
FreeSWITCH-users at lists.freeswitch.org  
<http://lists.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  Tue Apr  5 17:01:49 2016
From: italo at freeswitch.org (=?utf-8?q?=C3=8Dtalo_Rossi?=)
Date: Tue, 05 Apr 2016 06:01:49 -0700 (PDT)
Subject: [Freeswitch-users] Thursday FreeSWITCH Bug Hunt
Message-ID: <9kt3y5sxvjf91iqz3mc97slt5-0@mailer.nylas.com>

FreeSWITCHers,

  

Join us thursday 2PM CST for the Thursday FreeSWITCH Bug Hunt!

  

Where? [conference.freeswitch.org/vc/#/?autocall=888](https://conference.frees
witch.org/vc/#/?autocall=888
"https://conference.freeswitch.org/vc/#/?autocall=888" )

  

Chat? <https://hipchat.freeswitch.org/gYzPMg3Oc>

  

What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support!

  

Help us help you, Join the Bug Hunt!

?talo Rossi

italo at freeswitch.org

IRC chat.freenode.net #freeswitch #freeswitch-dev

Bugs? https://freeswitch.org/jira

Docs? https://freeswitch.org/jira

Chat? https://hipchat.freeswitch.org/gUdAgy0m6

  

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From ahabiba at gmail.com  Tue Apr  5 17:11:27 2016
From: ahabiba at gmail.com (Ahmed habiba)
Date: Tue, 5 Apr 2016 16:11:27 +0300
Subject: [Freeswitch-users] IMT Codec Enforcement
In-Reply-To: <mailman.87.1459811416.8223.freeswitch-users@lists.freeswitch.org>
References: <mailman.87.1459811416.8223.freeswitch-users@lists.freeswitch.org>
Message-ID: <EE04EE06-9026-4861-BF6F-01329FA200BF@gmail.com>

Your kind feedback will be appreciated.


> 
> 
> From: Ahmed habiba <ahabiba at gmail.com>
> Subject: [Freeswitch-users] IMT Codec Enforcement
> Date: April 4, 2016 at 10:37:46 PM GMT+3
> To: freeswitch-users at lists.freeswitch.org
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Dears,
> 
> kindly I have the below scenario:
> 
> [SoftPhone1]<??Channel1??>[FreeSwitch1]<??Channel2??>[FreeSwitch2]<??Chanel3??>[SoftPhone2]
> 
> Channel1 codec G722
> 
> Channel3 codec G722
> 
> I want to enforce Channel 3 between FreeSwitch instances to be GSM, I change the Dial plane of freeSwitch1 to and the same for FS2
> 
>                   <action application="set" data=?absolute_codec_string=GSM"/>
> 
>                   <action application="bridge" data="sofia/internal/$1 at 10.8.0.22 <mailto:sofia/internal/$1 at 10.8.0.22>"/>
> 
> However when I run call from FS2 to FS1, FS1 reply with the below putting ?a=rtpmap:106 opus/48000/2? as the only available codec as below, your kind help will be appreciated
> 
> 
> 
> recv 1275 bytes from udp/[10.8.0.22]:5060 at 15:22:35.560845:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 10.8.0.29;rport=5060;branch=z9hG4bKSr1v2mgcctX5H
>    From: "Extension 1000" <sip:1000 at 10.8.0.29 <sip:1000 at 10.8.0.29>>;tag=U82NjmSa9S9Kg
>    To: <sip:1000 at 10.8.0.22 <sip:1000 at 10.8.0.22>>;tag=g0eNeprr8e4Fg
>    Call-ID: 6d5dbd7f-753d-1234-d78c-bc305bd3840c
>    CSeq: 89562813 INVITE
>    Contact: <sip:1000 at 10.8.0.22:5060;transport=udp <sip:1000 at 10.8.0.22:5060;transport=udp>>
>    User-Agent: FreeSWITCH-mod_sofia/1.6.6+git~20160111T201612Z~d2d0b3283a~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 261
>    X-FS-Display-Name: 1000
>    X-FS-Display-Number: sip:1000 at 10.8.0.22 <sip:1000 at 10.8.0.22>
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "1000" <sip:1000 at 10.8.0.22 <sip:1000 at 10.8.0.22>>;party=calling;privacy=off;screen=no
>    
>    v=0
>    o=FreeSWITCH 1459771930 1459771931 IN IP4 10.8.0.22
>    s=FreeSWITCH
>    c=IN IP4 10.8.0.22
>    t=0 0
>    m=audio 25836 RTP/AVP 106 101
>    a=rtpmap:106 opus/48000/2
>    a=fmtp:106 useinbandfec=1; minptime=20
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
> 
> Thanks,
> 
> Ahmed Habiba.
> 

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From gregor at infomedia.si  Tue Apr  5 17:34:06 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 5 Apr 2016 15:34:06 +0200
Subject: [Freeswitch-users] Format_cdr modul
Message-ID: <CAC7ttFEf1-4nP=vj+yn5AA+stAQD45m2j1FLNuUvwB-JEDCwrQ@mail.gmail.com>

Hi!

I would like to use format_cdr module to post json cdr. I can see there is
format_cdr.conf file, but module is not defined in modules.conf.

How can I enable this module?

Best regards, Gergor
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From gregor at infomedia.si  Tue Apr  5 17:37:32 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 5 Apr 2016 15:37:32 +0200
Subject: [Freeswitch-users] Freeswitch 1.67. is out
Message-ID: <CAC7ttFHFOBTcmxtg27jZO7DZ_e0meAHN2sUF5gtN9Q9C5nLN7A@mail.gmail.com>

This could be silly question for someone, but anyway. We all have to learn
something new .

Can someone please explain me how code is managed in such large projects.

I saw that FS 1.6.7 is out and has a lot of mod_verto features added.

Are those features already in latest master branch? Is master 1.7 version?
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From jurijs.ivolga at gmail.com  Tue Apr  5 17:52:30 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Tue, 5 Apr 2016 16:52:30 +0300
Subject: [Freeswitch-users] Format_cdr modul
In-Reply-To: <CAC7ttFEf1-4nP=vj+yn5AA+stAQD45m2j1FLNuUvwB-JEDCwrQ@mail.gmail.com>
References: <CAC7ttFEf1-4nP=vj+yn5AA+stAQD45m2j1FLNuUvwB-JEDCwrQ@mail.gmail.com>
Message-ID: <CAOrA2UZcpiHp7Qgwfv9_G6aKVbP5syie9TVZajvBJihUtveBmQ@mail.gmail.com>

Hi,

You can enable it by adding following line in modules.conf.xml:

<load module="mod_format_cdr"/>

With kind regards,

Jurijs

On Tue, Apr 5, 2016 at 4:34 PM, Gregor Nanger <gregor at infomedia.si> wrote:

> Hi!
>
> I would like to use format_cdr module to post json cdr. I can see there is
> format_cdr.conf file, but module is not defined in modules.conf.
>
> How can I enable this module?
>
> Best regards, Gergor
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From amani.mansour2 at gmail.com  Tue Apr  5 18:03:07 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Tue, 5 Apr 2016 15:03:07 +0100
Subject: [Freeswitch-users] IMT Codec Enforcement
In-Reply-To: <EE04EE06-9026-4861-BF6F-01329FA200BF@gmail.com>
References: <mailman.87.1459811416.8223.freeswitch-users@lists.freeswitch.org>
	<EE04EE06-9026-4861-BF6F-01329FA200BF@gmail.com>
Message-ID: <CAFETSWgO0LQUadbM3j8GHu2WNiUAwR23T-N4-jE_Xh_pzDxixQ@mail.gmail.com>

Hi ,
Try Mr ahmed to replace
 <action application="set" data=?absolute_codec_string=GSM"/>
with:
<action  application="set" data="originator_codec=GSM")


with best regards
amani

2016-04-05 14:11 GMT+01:00 Ahmed habiba <ahabiba at gmail.com>:

> Your kind feedback will be appreciated.
>
>
>
>
> *From: *Ahmed habiba <ahabiba at gmail.com>
> *Subject: **[Freeswitch-users] IMT Codec Enforcement*
> *Date: *April 4, 2016 at 10:37:46 PM GMT+3
> *To: *freeswitch-users at lists.freeswitch.org
> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> Dears,
>
> kindly I have the below scenario:
>
> [SoftPhone1]<??*Channel1*??>[FreeSwitch1]<??*Channel2*??>[FreeSwitch2]<??
> *Chanel3*??>[SoftPhone2]
>
> Channel1 codec G722
>
> Channel3 codec G722
>
> I want to enforce Channel 3 between FreeSwitch instances to be GSM, I
> change the Dial plane of freeSwitch1 to and the same for FS2
>
>                   <action application="set"
> data=?absolute_codec_string=GSM"/>
>
>                   <action application="bridge" data="
> sofia/internal/$1 at 10.8.0.22"/>
>
> However when I run call from FS2 to FS1, *FS1* reply with the below
> putting ?a=rtpmap:106 opus/48000/2? as the only available codec as below,
> your kind help will be appreciated
>
>
>
> recv 1275 bytes from udp/[10.8.0.22]:5060 at 15:22:35.560845:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 10.8.0.29;rport=5060;branch=z9hG4bKSr1v2mgcctX5H
>    From: "Extension 1000" <sip:1000 at 10.8.0.29>;tag=U82NjmSa9S9Kg
>    To: <sip:1000 at 10.8.0.22>;tag=g0eNeprr8e4Fg
>    Call-ID: 6d5dbd7f-753d-1234-d78c-bc305bd3840c
>    CSeq: 89562813 INVITE
>    Contact: <sip:1000 at 10.8.0.22:5060;transport=udp>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.6.6+git~20160111T201612Z~d2d0b3283a~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, presence, as-feature-event,
> dialog, line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 261
>    X-FS-Display-Name: 1000
>    X-FS-Display-Number: sip:1000 at 10.8.0.22
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "1000" <sip:1000 at 10.8.0.22
> >;party=calling;privacy=off;screen=no
>
>    v=0
>    o=FreeSWITCH 1459771930 1459771931 IN IP4 10.8.0.22
>    s=FreeSWITCH
>    c=IN IP4 10.8.0.22
>    t=0 0
>    m=audio 25836 RTP/AVP 106 101
> *   a=rtpmap:106 opus/48000/2*
>    a=fmtp:106 useinbandfec=1; minptime=20
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>
> Thanks,
>
> Ahmed Habiba.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From m.hubert at hexanet.fr  Tue Apr  5 18:04:41 2016
From: m.hubert at hexanet.fr (Mickael Hubert)
Date: Tue, 5 Apr 2016 16:04:41 +0200
Subject: [Freeswitch-users] Codec negotiation issue
Message-ID: <CAFSwG-vUMZ=FEET4Vwr20UVWGH4T-d4pGjcnwWkH8rZ+cWbpKQ@mail.gmail.com>

Hi list,
I have an issue with the negociation codec in my Freeswitch.

I followed this doc: https://wiki.freeswitch.org/wiki/Codec_Negotiation

The call flow:

1) INVITE: UAC -- (G729, PCMA) --> FS -- (G729, PCMA) --> SVI Asterisk
2) 200OK UAC <-- (G729) -- FS <-- (PCMA) -- SVI Asterisk (HOMER)

I have inherit_codec=true in my dialplan and inbound-late-negotiation true
in sip-profile.
But freeswitch do not force codec learned from leg B to leg A.

*LEG B (200OK):*



*2016-04-01 11:28:16.453784 [DEBUG] switch_core_media.c:3194 Audio Codec
Compare [PCMA:8:8000:150:64000]/[G729:18:8000:20:8000]2016-04-01
11:28:16.453784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare
[PCMA:8:8000:150:64000]/[PCMA:8:8000:20:64000]2016-04-01 11:28:16.453784
[DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMA:8:8000:20:64000]
++++ is saved as a match2016-04-01 11:28:16.453784 [DEBUG]
switch_core_codec.c:111 sofia/internal/06********@HOMER Original read codec
set to PCMA:8*

LEG B is in G711A, OK, next:

2016-04-01 11:28:16.453784 [NOTICE] sofia.c:6727 Channel
[sofia/internal/06*******@HOMER] has been answered
2016-04-01 11:28:16.453784 [DEBUG] switch_channel.c:3686
(sofia/internal/06*********@HOMER) Callstate Change DOWN -> ACTIVE
2016-04-01 11:28:16.473813 [DEBUG] switch_ivr_originate.c:412 *Setting
codec string on sofia/external/0326793005 at 1.1.1.1 <0326793005 at 1.1.1.1> to
PCMA at 8000h@20i*

We can see, FS set the correct codec to LEG A; good ! , next...






*2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:3194 Audio Codec
Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000]2016-04-01
11:28:16.473813 [DEBUG] switch_core_media.c:3248 Audio Codec Compare
[G729:18:8000:20:8000] ++++ is saved as a match2016-04-01 11:28:16.473813
[DEBUG] switch_core_media.c:3194 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]2016-04-01 11:28:16.473813
[DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMA:8:8000:20:64000]
++++ is saved as a match2016-04-01 11:28:16.473813 [DEBUG]
switch_core_media.c:2139 Set Codec sofia/external/0326793005 at 1.1.1.1
<0326793005 at 1.1.1.1> G729/8000 20 ms 160 samples 8000 bits*
Why FS compare and re set a new codec (G729) ?

Next:


*2016-04-01 11:28:16.613776 [ERR] mod_g729.c:145 This codec is only usable
in passthrough mode!2016-04-01 11:28:16.613776 [ERR] switch_core_io.c:1245
Codec G.729 decoder error!*

normal ....

*For informations:*
- FreeSWITCH Version 1.5.8b+git~20140214T000311Z~fe2a4d6d47~64bit (git
fe2a4d6 2014-02-14 00:03:11Z 64bit)

Is it a bug ? an configuration error ?

Thanks in advance


PS:

*My dialplan:*




*<condition field="${destination_number}" expression="^0([1-79][0-9]{8})$"
break="on-true"><action application="export"
data="codec_string=${ep_codec_string}"/><action application="set"
data="inherit_codec=true"/><action application="bridge"
data="sofia/internal/${destination_number}@HOMER"/></condition>*



-- 
Cordialement

HUBERT Micka?l
Ing?nieur VOIP - Hexanet

-- 

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From gregor at infomedia.si  Tue Apr  5 18:08:30 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 5 Apr 2016 16:08:30 +0200
Subject: [Freeswitch-users] Format_cdr modul
In-Reply-To: <CAOrA2UZcpiHp7Qgwfv9_G6aKVbP5syie9TVZajvBJihUtveBmQ@mail.gmail.com>
References: <CAC7ttFEf1-4nP=vj+yn5AA+stAQD45m2j1FLNuUvwB-JEDCwrQ@mail.gmail.com>
	<CAOrA2UZcpiHp7Qgwfv9_G6aKVbP5syie9TVZajvBJihUtveBmQ@mail.gmail.com>
Message-ID: <CAC7ttFGL5y51X5frqhK1nJcdZtTOe0iDzDbFSF5SEGdk66v_bA@mail.gmail.com>

Thank you!

but I am on windows and it looks like this module is not precompiled,
because I get error that it cannot find dll :-(

2016-04-05 15:52 GMT+02:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:

> Hi,
>
> You can enable it by adding following line in modules.conf.xml:
>
> <load module="mod_format_cdr"/>
>
> With kind regards,
>
> Jurijs
>
> On Tue, Apr 5, 2016 at 4:34 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>
>> Hi!
>>
>> I would like to use format_cdr module to post json cdr. I can see there
>> is format_cdr.conf file, but module is not defined in modules.conf.
>>
>> How can I enable this module?
>>
>> Best regards, Gergor
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 max at nysolutions.com  Tue Apr  5 18:40:07 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Tue, 5 Apr 2016 14:40:07 +0000
Subject: [Freeswitch-users] Freeswitch 1.67. is out
In-Reply-To: <CAC7ttFHFOBTcmxtg27jZO7DZ_e0meAHN2sUF5gtN9Q9C5nLN7A@mail.gmail.com>
References: <CAC7ttFHFOBTcmxtg27jZO7DZ_e0meAHN2sUF5gtN9Q9C5nLN7A@mail.gmail.com>
Message-ID: <1c2e685aa850454698eb69e790ad6d63@nysolutions.com>

Yes, 1.7 is Master. Master is odd numbers, release is Even numbers.
Release is a resync of Master.

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com<mailto:support at nysolutions.com>
[cid:image001.jpg at 01C72F94.9EE45D60]<http://www.nysolutions.com/>
Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gregor Nanger
Sent: Tuesday, April 5, 2016 9:38 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Freeswitch 1.67. is out

This could be silly question for someone, but anyway. We all have to learn something new .

Can someone please explain me how code is managed in such large projects.

I saw that FS 1.6.7 is out and has a lot of mod_verto features added.

Are those features already in latest master branch? Is master 1.7 version?

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From gregor at infomedia.si  Tue Apr  5 18:44:16 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 5 Apr 2016 16:44:16 +0200
Subject: [Freeswitch-users] Freeswitch 1.67. is out
In-Reply-To: <1c2e685aa850454698eb69e790ad6d63@nysolutions.com>
References: <CAC7ttFHFOBTcmxtg27jZO7DZ_e0meAHN2sUF5gtN9Q9C5nLN7A@mail.gmail.com>
	<1c2e685aa850454698eb69e790ad6d63@nysolutions.com>
Message-ID: <CAC7ttFFvQgPgVvmUqmMtYyxsHNTOR9884F5wEfRHXiT193FYGw@mail.gmail.com>

Oh, I get it :-) Thank you :-)

2016-04-05 16:40 GMT+02:00 Moishe Grunstein <max at nysolutions.com>:

> Yes, 1.7 is Master. Master is odd numbers, release is Even numbers.
>
> Release is a resync of Master.
>
>
>
> Thanks,
>
>
>
> Moishe Grunstein
>
> Tornado Computer Systems, Inc.
>
> 212.400.7650 888.IPPBX.US
> *Service Request Email: support at nysolutions.com <support at nysolutions.com> *
>
> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>
> Computer Networking * Managed Services * IP Video Surveillance * Network
> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
> Security * Site Surveys * CMS
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
> Nanger
> *Sent:* Tuesday, April 5, 2016 9:38 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Freeswitch 1.67. is out
>
>
>
> This could be silly question for someone, but anyway. We all have to learn
> something new .
>
>
>
> Can someone please explain me how code is managed in such large projects.
>
>
>
> I saw that FS 1.6.7 is out and has a lot of mod_verto features added.
>
>
>
> Are those features already in latest master branch? Is master 1.7 version?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
? www.infomedia.si
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From gregor at infomedia.si  Tue Apr  5 18:55:34 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 5 Apr 2016 16:55:34 +0200
Subject: [Freeswitch-users] Freeswitch 1.67. is out
In-Reply-To: <CAC7ttFFvQgPgVvmUqmMtYyxsHNTOR9884F5wEfRHXiT193FYGw@mail.gmail.com>
References: <CAC7ttFHFOBTcmxtg27jZO7DZ_e0meAHN2sUF5gtN9Q9C5nLN7A@mail.gmail.com>
	<1c2e685aa850454698eb69e790ad6d63@nysolutions.com>
	<CAC7ttFFvQgPgVvmUqmMtYyxsHNTOR9884F5wEfRHXiT193FYGw@mail.gmail.com>
Message-ID: <CAC7ttFHaYeCotkQyhue0wdhAqRGzOmbDde08ByjQA560jZ5pVw@mail.gmail.com>

If I want to clone source to build it, I can issue:

git clone ?b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git
freeswitch.git

Will this clone version 1.67 or should I specify v1.67?


Best regards, Gregor

2016-04-05 16:44 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:

> Oh, I get it :-) Thank you :-)
>
> 2016-04-05 16:40 GMT+02:00 Moishe Grunstein <max at nysolutions.com>:
>
>> Yes, 1.7 is Master. Master is odd numbers, release is Even numbers.
>>
>> Release is a resync of Master.
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Moishe Grunstein
>>
>> Tornado Computer Systems, Inc.
>>
>> 212.400.7650 888.IPPBX.US
>> *Service Request Email: support at nysolutions.com <support at nysolutions.com>
>> *
>>
>> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>>
>> Computer Networking * Managed Services * IP Video Surveillance * Network
>> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
>> Security * Site Surveys * CMS
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
>> Nanger
>> *Sent:* Tuesday, April 5, 2016 9:38 AM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] Freeswitch 1.67. is out
>>
>>
>>
>> This could be silly question for someone, but anyway. We all have to
>> learn something new .
>>
>>
>>
>> Can someone please explain me how code is managed in such large projects.
>>
>>
>>
>> I saw that FS 1.6.7 is out and has a lot of mod_verto features added.
>>
>>
>>
>> Are those features already in latest master branch? Is master 1.7 version?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
? www.infomedia.si
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From krice at freeswitch.org  Tue Apr  5 19:04:48 2016
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 5 Apr 2016 10:04:48 -0500
Subject: [Freeswitch-users] Freeswitch 1.67. is out
In-Reply-To: <CAC7ttFHaYeCotkQyhue0wdhAqRGzOmbDde08ByjQA560jZ5pVw@mail.gmail.com>
References: <CAC7ttFHFOBTcmxtg27jZO7DZ_e0meAHN2sUF5gtN9Q9C5nLN7A@mail.gmail.com>	<1c2e685aa850454698eb69e790ad6d63@nysolutions.com>	<CAC7ttFFvQgPgVvmUqmMtYyxsHNTOR9884F5wEfRHXiT193FYGw@mail.gmail.com>
	<CAC7ttFHaYeCotkQyhue0wdhAqRGzOmbDde08ByjQA560jZ5pVw@mail.gmail.com>
Message-ID: <7d1a01d18f4c$7fe055f0$7fa101d0$@freeswitch.org>

If you clone the v1.6 branch you will get the latest release plus any addition patches we have already staged for the next release? and note. There is no 1.67 yet the latest release is 1.6.7

 

K

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gregor Nanger
Sent: Tuesday, April 5, 2016 9:56 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch 1.67. is out

 

If I want to clone source to build it, I can issue:

git clone ?b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch.git

Will this clone version 1.67 or should I specify v1.67?


Best regards, Gregor  <http://t.sidekickopen04.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v5dbp0W3MPnzY4XrbSMW8q-f_43LvrVvW59BB4r1k1H6H0?si=5036397111279616&pi=2dfee9cb-4a20-43a0-b5d0-3362ccd9705d>   <http://t.sidekickopen04.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v5dbp0W3MPnzY4XrbSMW8q-f_43LvrVvW59BB4r1k1H6H0?si=5036397111279616&pi=1b7be28a-ca40-4af8-967d-1403a9eff72d> 

 

2016-04-05 16:44 GMT+02:00 Gregor Nanger <gregor at infomedia.si <mailto:gregor at infomedia.si> >:

Oh, I get it :-) Thank you :-)

  <http://t.sidekickopen04.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v5dbp0W3MPnzY4XrbSMW8q-f_43LvrVvW59BB4r1k1H6H0?si=5036397111279616&pi=61e514e6-b2be-413f-cec0-02b04dd3fbe9> 

 

2016-04-05 16:40 GMT+02:00 Moishe Grunstein <max at nysolutions.com <mailto:max at nysolutions.com> >:

Yes, 1.7 is Master. Master is odd numbers, release is Even numbers.

Release is a resync of Master.

 

Thanks,

 

Moishe Grunstein

Tornado Computer Systems, Inc.

212.400.7650 888.IPPBX.US <http://888.IPPBX.US> 
Service Request Email: support at nysolutions.com <mailto:support at nysolutions.com>  

 <http://www.nysolutions.com/> 

Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Gregor Nanger
Sent: Tuesday, April 5, 2016 9:38 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: [Freeswitch-users] Freeswitch 1.67. is out

 

This could be silly question for someone, but anyway. We all have to learn something new .

 

Can someone please explain me how code is managed in such large projects.

 

I saw that FS 1.6.7 is out and has a lot of mod_verto features added. 

 

Are those features already in latest master branch? Is master 1.7 version?

  <http://t.sidekickopen04.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v5dbp0W3MPnzY4XrbSMW8q-f_43LvrVvW59BB4r1k1H6H0?si=5036397111279616&pi=9c1c9cb2-2d16-4a7b-84ca-11f11f56b72c> 

 

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

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

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





 

-- 


Gregor Nanger

 

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





 

-- 


Gregor Nanger

 

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

 

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From gregor at infomedia.si  Tue Apr  5 19:14:05 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 5 Apr 2016 17:14:05 +0200
Subject: [Freeswitch-users] Freeswitch 1.67. is out
In-Reply-To: <7d1a01d18f4c$7fe055f0$7fa101d0$@freeswitch.org>
References: <CAC7ttFHFOBTcmxtg27jZO7DZ_e0meAHN2sUF5gtN9Q9C5nLN7A@mail.gmail.com>
	<1c2e685aa850454698eb69e790ad6d63@nysolutions.com>
	<CAC7ttFFvQgPgVvmUqmMtYyxsHNTOR9884F5wEfRHXiT193FYGw@mail.gmail.com>
	<CAC7ttFHaYeCotkQyhue0wdhAqRGzOmbDde08ByjQA560jZ5pVw@mail.gmail.com>
	<7d1a01d18f4c$7fe055f0$7fa101d0$@freeswitch.org>
Message-ID: <CAC7ttFGNRRUs+cPOGhqHoFZKZgh3-Ac+RaXoZi0Ps-_NsMH4Tw@mail.gmail.com>

Thank you, Ken!

Please, just explain me what is difference between

git clone ?b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git
git clone ?b v1.6.7 https://freeswitch.org/stash/scm/fs/freeswitch.git

If latest release is 1.6.7, then clone v1.6 would pull this release?

2016-04-05 17:04 GMT+02:00 Ken Rice <krice at freeswitch.org>:

> If you clone the v1.6 branch you will get the latest release plus any
> addition patches we have already staged for the next release? and note.
> There is no 1.67 yet the latest release is 1.6.7
>
>
>
> K
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
> Nanger
> *Sent:* Tuesday, April 5, 2016 9:56 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Freeswitch 1.67. is out
>
>
>
> If I want to clone source to build it, I can issue:
>
> git clone ?b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git
> freeswitch.git
>
> Will this clone version 1.67 or should I specify v1.67?
>
>
> Best regards, Gregor
>
>
>
> 2016-04-05 16:44 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:
>
> Oh, I get it :-) Thank you :-)
>
>
>
> 2016-04-05 16:40 GMT+02:00 Moishe Grunstein <max at nysolutions.com>:
>
> Yes, 1.7 is Master. Master is odd numbers, release is Even numbers.
>
> Release is a resync of Master.
>
>
>
> Thanks,
>
>
>
> Moishe Grunstein
>
> Tornado Computer Systems, Inc.
>
> 212.400.7650 888.IPPBX.US
> *Service Request Email: support at nysolutions.com <support at nysolutions.com> *
>
> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>
> Computer Networking * Managed Services * IP Video Surveillance * Network
> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
> Security * Site Surveys * CMS
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
> Nanger
> *Sent:* Tuesday, April 5, 2016 9:38 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Freeswitch 1.67. is out
>
>
>
> This could be silly question for someone, but anyway. We all have to learn
> something new .
>
>
>
> Can someone please explain me how code is managed in such large projects.
>
>
>
> I saw that FS 1.6.7 is out and has a lot of mod_verto features added.
>
>
>
> Are those features already in latest master branch? Is master 1.7 version?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
>
>
>
>
> --
>
> *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
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
? www.infomedia.si
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From krice at freeswitch.org  Tue Apr  5 19:23:41 2016
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 5 Apr 2016 10:23:41 -0500
Subject: [Freeswitch-users] Freeswitch 1.67. is out
In-Reply-To: <CAC7ttFGNRRUs+cPOGhqHoFZKZgh3-Ac+RaXoZi0Ps-_NsMH4Tw@mail.gmail.com>
References: <CAC7ttFHFOBTcmxtg27jZO7DZ_e0meAHN2sUF5gtN9Q9C5nLN7A@mail.gmail.com>	<1c2e685aa850454698eb69e790ad6d63@nysolutions.com>	<CAC7ttFFvQgPgVvmUqmMtYyxsHNTOR9884F5wEfRHXiT193FYGw@mail.gmail.com>	<CAC7ttFHaYeCotkQyhue0wdhAqRGzOmbDde08ByjQA560jZ5pVw@mail.gmail.com>	<7d1a01d18f4c$7fe055f0$7fa101d0$@freeswitch.org>
	<CAC7ttFGNRRUs+cPOGhqHoFZKZgh3-Ac+RaXoZi0Ps-_NsMH4Tw@mail.gmail.com>
Message-ID: <7d2b01d18f4f$231d34c0$69579e40$@freeswitch.org>

1.6.7 is a tag on the v1.6 branch for a specific release?  

 

git clone -b v1.6.7 is asking for a branch called v1.6.7 which does not exist?

 

git clone -b v1.6 ?

git checkout v1.6.7

that is the most correct way to get the tag (imho)

 

will the -b v1.6.7 work? Maybe, I?m not sure but the -b flag on a clone is asking for a branch?

 

when you git clone v1.6 you will get whatever the latest 1.6 release is plus any additional patches we have merged? 

example if we have found a bug or an issue in the last release (in this case 1.6.7) and we have already rolled the patch from master to the v1.6 branch, you will get 1.6.7+ additional patches to resolve some issue(s)

 

K

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gregor Nanger
Sent: Tuesday, April 5, 2016 10:14 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch 1.67. is out

 

Thank you, Ken!

 

Please, just explain me what is difference between

 

git clone ?b v1.6  <https://freeswitch.org/stash/scm/fs/freeswitch.git> https://freeswitch.org/stash/scm/fs/freeswitch.git

git clone ?b v1.6.7  <https://freeswitch.org/stash/scm/fs/freeswitch.git> https://freeswitch.org/stash/scm/fs/freeswitch.git

 

If latest release is 1.6.7, then clone v1.6 would pull this release?

  <http://t.sidekickopen04.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v5dbp0W3MPnzY4XrbSMW8q-f_43LvrVvW59BB4r1k1H6H0?si=5036397111279616&pi=4461e948-3741-4593-bce3-994236ca0422> 

 

2016-04-05 17:04 GMT+02:00 Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >:

If you clone the v1.6 branch you will get the latest release plus any addition patches we have already staged for the next release? and note. There is no 1.67 yet the latest release is 1.6.7

 

K

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Gregor Nanger
Sent: Tuesday, April 5, 2016 9:56 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] Freeswitch 1.67. is out

 

If I want to clone source to build it, I can issue:

git clone ?b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git freeswitch.git

Will this clone version 1.67 or should I specify v1.67?


Best regards, Gregor  <http://t.sidekickopen04.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v5dbp0W3MPnzY4XrbSMW8q-f_43LvrVvW59BB4r1k1H6H0?si=5036397111279616&pi=2dfee9cb-4a20-43a0-b5d0-3362ccd9705d>   <http://t.sidekickopen04.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v5dbp0W3MPnzY4XrbSMW8q-f_43LvrVvW59BB4r1k1H6H0?si=5036397111279616&pi=1b7be28a-ca40-4af8-967d-1403a9eff72d> 

 

2016-04-05 16:44 GMT+02:00 Gregor Nanger <gregor at infomedia.si <mailto:gregor at infomedia.si> >:

Oh, I get it :-) Thank you :-)

  <http://t.sidekickopen04.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v5dbp0W3MPnzY4XrbSMW8q-f_43LvrVvW59BB4r1k1H6H0?si=5036397111279616&pi=61e514e6-b2be-413f-cec0-02b04dd3fbe9> 

 

2016-04-05 16:40 GMT+02:00 Moishe Grunstein <max at nysolutions.com <mailto:max at nysolutions.com> >:

Yes, 1.7 is Master. Master is odd numbers, release is Even numbers.

Release is a resync of Master.

 

Thanks,

 

Moishe Grunstein

Tornado Computer Systems, Inc.

212.400.7650 888.IPPBX.US <http://888.IPPBX.US> 
Service Request Email: support at nysolutions.com <mailto:support at nysolutions.com>  

 <http://www.nysolutions.com/> 

Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Gregor Nanger
Sent: Tuesday, April 5, 2016 9:38 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: [Freeswitch-users] Freeswitch 1.67. is out

 

This could be silly question for someone, but anyway. We all have to learn something new .

 

Can someone please explain me how code is managed in such large projects.

 

I saw that FS 1.6.7 is out and has a lot of mod_verto features added. 

 

Are those features already in latest master branch? Is master 1.7 version?

  <http://t.sidekickopen04.com/e1t/o/5/f18dQhb0S7ks8dDMPbW2n0x6l2B9gXrN7sKj6v5dbp0W3MPnzY4XrbSMW8q-f_43LvrVvW59BB4r1k1H6H0?si=5036397111279616&pi=9c1c9cb2-2d16-4a7b-84ca-11f11f56b72c> 

 

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

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

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





 

-- 


Gregor Nanger

 

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





 

-- 


Gregor Nanger

 

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

 


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

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

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





 

-- 


Gregor Nanger

 

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

 

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From gregor at infomedia.si  Tue Apr  5 19:28:17 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 5 Apr 2016 17:28:17 +0200
Subject: [Freeswitch-users] Freeswitch 1.67. is out
In-Reply-To: <7d2b01d18f4f$231d34c0$69579e40$@freeswitch.org>
References: <CAC7ttFHFOBTcmxtg27jZO7DZ_e0meAHN2sUF5gtN9Q9C5nLN7A@mail.gmail.com>
	<1c2e685aa850454698eb69e790ad6d63@nysolutions.com>
	<CAC7ttFFvQgPgVvmUqmMtYyxsHNTOR9884F5wEfRHXiT193FYGw@mail.gmail.com>
	<CAC7ttFHaYeCotkQyhue0wdhAqRGzOmbDde08ByjQA560jZ5pVw@mail.gmail.com>
	<7d1a01d18f4c$7fe055f0$7fa101d0$@freeswitch.org>
	<CAC7ttFGNRRUs+cPOGhqHoFZKZgh3-Ac+RaXoZi0Ps-_NsMH4Tw@mail.gmail.com>
	<7d2b01d18f4f$231d34c0$69579e40$@freeswitch.org>
Message-ID: <CAC7ttFGY3Bs7yy-whq7h_nQ3jN+4zrn6g6NA=O+uHe5f+LRvZA@mail.gmail.com>

Thanks...

2016-04-05 17:23 GMT+02:00 Ken Rice <krice at freeswitch.org>:

> 1.6.7 is a tag on the v1.6 branch for a specific release?
>
>
>
> git clone -b v1.6.7 is asking for a branch called v1.6.7 which does not
> exist?
>
>
>
> git clone -b v1.6 ?
>
> git checkout v1.6.7
>
> that is the most correct way to get the tag (imho)
>
>
>
> will the -b v1.6.7 work? Maybe, I?m not sure but the -b flag on a clone is
> asking for a branch?
>
>
>
> when you git clone v1.6 you will get whatever the latest 1.6 release is
> plus any additional patches we have merged?
>
> example if we have found a bug or an issue in the last release (in this
> case 1.6.7) and we have already rolled the patch from master to the v1.6
> branch, you will get 1.6.7+ additional patches to resolve some issue(s)
>
>
>
> K
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
> Nanger
> *Sent:* Tuesday, April 5, 2016 10:14 AM
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Freeswitch 1.67. is out
>
>
>
> Thank you, Ken!
>
>
>
> Please, just explain me what is difference between
>
>
>
> git clone ?b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git
>
> git clone ?b v1.6.7 https://freeswitch.org/stash/scm/fs/freeswitch.git
>
>
>
> If latest release is 1.6.7, then clone v1.6 would pull this release?
>
>
>
> 2016-04-05 17:04 GMT+02:00 Ken Rice <krice at freeswitch.org>:
>
> If you clone the v1.6 branch you will get the latest release plus any
> addition patches we have already staged for the next release? and note.
> There is no 1.67 yet the latest release is 1.6.7
>
>
>
> K
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
> Nanger
> *Sent:* Tuesday, April 5, 2016 9:56 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Freeswitch 1.67. is out
>
>
>
> If I want to clone source to build it, I can issue:
>
> git clone ?b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git
> freeswitch.git
>
> Will this clone version 1.67 or should I specify v1.67?
>
>
> Best regards, Gregor
>
>
>
> 2016-04-05 16:44 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:
>
> Oh, I get it :-) Thank you :-)
>
>
>
> 2016-04-05 16:40 GMT+02:00 Moishe Grunstein <max at nysolutions.com>:
>
> Yes, 1.7 is Master. Master is odd numbers, release is Even numbers.
>
> Release is a resync of Master.
>
>
>
> Thanks,
>
>
>
> Moishe Grunstein
>
> Tornado Computer Systems, Inc.
>
> 212.400.7650 888.IPPBX.US
> *Service Request Email: support at nysolutions.com <support at nysolutions.com> *
>
> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>
> Computer Networking * Managed Services * IP Video Surveillance * Network
> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
> Security * Site Surveys * CMS
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
> Nanger
> *Sent:* Tuesday, April 5, 2016 9:38 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Freeswitch 1.67. is out
>
>
>
> This could be silly question for someone, but anyway. We all have to learn
> something new .
>
>
>
> Can someone please explain me how code is managed in such large projects.
>
>
>
> I saw that FS 1.6.7 is out and has a lot of mod_verto features added.
>
>
>
> Are those features already in latest master branch? Is master 1.7 version?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
>
>
>
>
> --
>
> *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
>
>
>
>
>
> --
>
> *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
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
? www.infomedia.si
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From gregor at infomedia.si  Tue Apr  5 19:31:03 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 5 Apr 2016 17:31:03 +0200
Subject: [Freeswitch-users] Freeswitch 1.67. is out
In-Reply-To: <7d2b01d18f4f$231d34c0$69579e40$@freeswitch.org>
References: <CAC7ttFHFOBTcmxtg27jZO7DZ_e0meAHN2sUF5gtN9Q9C5nLN7A@mail.gmail.com>
	<1c2e685aa850454698eb69e790ad6d63@nysolutions.com>
	<CAC7ttFFvQgPgVvmUqmMtYyxsHNTOR9884F5wEfRHXiT193FYGw@mail.gmail.com>
	<CAC7ttFHaYeCotkQyhue0wdhAqRGzOmbDde08ByjQA560jZ5pVw@mail.gmail.com>
	<7d1a01d18f4c$7fe055f0$7fa101d0$@freeswitch.org>
	<CAC7ttFGNRRUs+cPOGhqHoFZKZgh3-Ac+RaXoZi0Ps-_NsMH4Tw@mail.gmail.com>
	<7d2b01d18f4f$231d34c0$69579e40$@freeswitch.org>
Message-ID: <CAC7ttFG7YcQWbVZ_+Yc4cOgPooRrEM4xJcCMN96ttr75B9X02A@mail.gmail.com>

Thanks...

2016-04-05 17:23 GMT+02:00 Ken Rice <krice at freeswitch.org>:

> 1.6.7 is a tag on the v1.6 branch for a specific release?
>
>
>
> git clone -b v1.6.7 is asking for a branch called v1.6.7 which does not
> exist?
>
>
>
> git clone -b v1.6 ?
>
> git checkout v1.6.7
>
> that is the most correct way to get the tag (imho)
>
>
>
> will the -b v1.6.7 work? Maybe, I?m not sure but the -b flag on a clone is
> asking for a branch?
>
>
>
> when you git clone v1.6 you will get whatever the latest 1.6 release is
> plus any additional patches we have merged?
>
> example if we have found a bug or an issue in the last release (in this
> case 1.6.7) and we have already rolled the patch from master to the v1.6
> branch, you will get 1.6.7+ additional patches to resolve some issue(s)
>
>
>
> K
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
> Nanger
> *Sent:* Tuesday, April 5, 2016 10:14 AM
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Freeswitch 1.67. is out
>
>
>
> Thank you, Ken!
>
>
>
> Please, just explain me what is difference between
>
>
>
> git clone ?b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git
>
> git clone ?b v1.6.7 https://freeswitch.org/stash/scm/fs/freeswitch.git
>
>
>
> If latest release is 1.6.7, then clone v1.6 would pull this release?
>
>
>
> 2016-04-05 17:04 GMT+02:00 Ken Rice <krice at freeswitch.org>:
>
> If you clone the v1.6 branch you will get the latest release plus any
> addition patches we have already staged for the next release? and note.
> There is no 1.67 yet the latest release is 1.6.7
>
>
>
> K
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
> Nanger
> *Sent:* Tuesday, April 5, 2016 9:56 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Freeswitch 1.67. is out
>
>
>
> If I want to clone source to build it, I can issue:
>
> git clone ?b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git
> freeswitch.git
>
> Will this clone version 1.67 or should I specify v1.67?
>
>
> Best regards, Gregor
>
>
>
> 2016-04-05 16:44 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:
>
> Oh, I get it :-) Thank you :-)
>
>
>
> 2016-04-05 16:40 GMT+02:00 Moishe Grunstein <max at nysolutions.com>:
>
> Yes, 1.7 is Master. Master is odd numbers, release is Even numbers.
>
> Release is a resync of Master.
>
>
>
> Thanks,
>
>
>
> Moishe Grunstein
>
> Tornado Computer Systems, Inc.
>
> 212.400.7650 888.IPPBX.US
> *Service Request Email: support at nysolutions.com <support at nysolutions.com> *
>
> [image: cid:image001.jpg at 01C72F94.9EE45D60] <http://www.nysolutions.com/>
>
> Computer Networking * Managed Services * IP Video Surveillance * Network
> Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network
> Security * Site Surveys * CMS
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Gregor
> Nanger
> *Sent:* Tuesday, April 5, 2016 9:38 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Freeswitch 1.67. is out
>
>
>
> This could be silly question for someone, but anyway. We all have to learn
> something new .
>
>
>
> Can someone please explain me how code is managed in such large projects.
>
>
>
> I saw that FS 1.6.7 is out and has a lot of mod_verto features added.
>
>
>
> Are those features already in latest master branch? Is master 1.7 version?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
>
>
>
>
> --
>
> *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
>
>
>
>
>
> --
>
> *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
>



-- 
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 mike at jerris.com  Tue Apr  5 19:37:17 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 5 Apr 2016 11:37:17 -0400
Subject: [Freeswitch-users] FS not see invite
In-Reply-To: <CABSP_VeCQgR_rTUg2=SwmKGHA9Em8E7y+bySYtdgrfevx5AtuQ@mail.gmail.com>
References: <CABSP_VfiEzaP58j0jSA8dd8D6psubk9QGvpZ=i46xBVqNVE=Dg@mail.gmail.com>
	<B07B3220-0A4E-4264-9D24-9FD9F75942DE@jerris.com>
	<CABSP_VdFOJY_+cs1hsEVrAyY4GpynhYT7DBDVpVpvUSMgpdqXA@mail.gmail.com>
	<21DA600E-2ABF-4E0F-864F-D9B8DEEBA485@jerris.com>
	<CABSP_VeCQgR_rTUg2=SwmKGHA9Em8E7y+bySYtdgrfevx5AtuQ@mail.gmail.com>
Message-ID: <9AC9F152-91C3-42E3-9D44-459A0CCAC254@jerris.com>

Any internal ACL in freeswitch would happen after the tport_log that is used to display sip packets.  If you enable sip trace and debug logs you will still see things from those packets.

> On Apr 4, 2016, at 11:59 PM, Yuriy Gorlichenko <ovoshlook at gmail.com> wrote:
> 
> I making registrations from Microsip client and all works fine so sofia Mode works. It happens only with another instanse that i want to connect to FS. firewall disabed. Also I see invites without auth from instance. I can not see only IVITE with auth params. And only at the fs_cli.
> 
> I cnow it sounds strandge but it is a fact... Thats why am asking about inernal blaqcklists at the fs.
> 
> 2016-04-04 21:53 GMT+03:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> If you have the sip trace enabled in freeswitch and you still don't see it, either you are sending to the wrong Ip/port, or firewall is dropping it.
> 
> 
>> On Apr 4, 2016, at 2:48 PM, Yuriy Gorlichenko <ovoshlook at gmail.com <mailto:ovoshlook at gmail.com>> wrote:
>> 
>> Hi yes. Im shure. Also yes . right ip and port.
>> It is looks like fw issue but it is turned off. I see invite with auth at TCPDUMP but not see it at the fs_cli
>> 
>> Can it be some troubles with internal FS blacklist or thomething like this? I just a beginner with FS. Previously worked with Asterisk.
>> 
>> 
>> 2016-04-04 17:48 GMT+03:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
>> are you SURE the firewall is turned off?  Is the sip going to the right iP/port that FreeSWITCH is listening on?
>> 
>>> On Apr 3, 2016, at 3:07 PM, Yuriy Gorlichenko <ovoshlook at gmail.com <mailto:ovoshlook at gmail.com>> wrote:
>>> 
>>> Hello. I try to make a call through another SIP instalse to freeswitch. 
>>> I added extension and use FS as proxy to rpovider.
>>> I successfully registered on FS as user and try to make a call.
>>> 
>>> I see at FS cli that invite arrives and FS answers 407, next my instanse Sends ACK and thn INVITE with a digest auth.
>>> 
>>> I see thins invite at the TCPDUMP on the FS machine but not see it at the FS
>>> Firewall is disabled for tests....
>> 
> 

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From mike at jerris.com  Tue Apr  5 19:39:28 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 5 Apr 2016 11:39:28 -0400
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAKsJLpa83aWDBMFuZaYeMFVR6XeN=2uQqwcuTXAtqTY3szckSg@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
	<CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
	<CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>
	<CAKsJLpa83aWDBMFuZaYeMFVR6XeN=2uQqwcuTXAtqTY3szckSg@mail.gmail.com>
Message-ID: <E6948933-09AC-4A48-A0C9-70E206EED453@jerris.com>

I would stay away from mod_vlc.  Its audio portions with recording have known issues.  We do use the rtmp streaming in mod_av heavily but thats obviously not wav.  Can you explain a bit more why you have this requirement?

> On Apr 5, 2016, at 5:36 AM, Vincent Gire <vincent.gire at gmail.com> wrote:
> 
> Ok thanks.
> It looks promising !
> I'll dig into mod_vlc.
> 
> Best 
> 
> Vincent
> 
> On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
> Think is requred streaming feature of freeswitch.
> Look at mod_esf <https://freeswitch.org/confluence/display/FREESWITCH/mod_esf> and mod_vlc <https://wiki.freeswitch.org/wiki/Mod_vlc>
> Instruction to compile mod_vlc on provided link is to old but helpfull to undestand how to stream media to http server.
> 
> For compiling mod_vlc please use vlc repo <https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos instruction <https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>.
> After you intall vlc, then you can enable mod_vlc module in freeswitch sources(SPEC file) and compile freeswitch.
> 
> Sergey
> 
> 
> ??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com <mailto:vincent.gire at gmail.com>>:
> webdav, mod_http_cache or mod_httapi all results in sending the recording only after it is complete.
> They all write the recording to a file, wait for the recording to complete and the file to close and then send it over HTTP.
> 
> I would like to start sending the recording to the remove server as soon as it starts (max 1 sec latency).
> mod_http_cache or mod_httapi would be perfect if they were streaming the recording like mod_shout.
> 
> 
> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
> Input/output latency is not problem.  I use Kazoo on my servers and call recording is stored to database during 5 seconds after hangup.
> What is broken in your case if save file using webdav or http_cache?
> 
> 
> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com <mailto:vincent.gire at gmail.com>> wrote:
> Hello Sergey,
> 
> Thank you for your answer.
> I've looked into webdav mounted filesystem.
> 
> Unfortunately, most WebDav clients (especially davfs2 on debian) do a lot of buffering, caching and even lock-null requests (lock a non existent resource before writing to it). I also suspect that they wait for the end of the write operation.
> The result is a latency of a few seconds witch is not much better than what I achieve with mod_shout if I transcode the MP3.
> 
> Any other idea ?
> 
> Thank you !
> 
> Best regards
> 
> Vincent
> 
> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
> Please look at webdav mounted filesystem.
> 
> 
> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com <mailto:vincent.gire at gmail.com>> wrote:
> Hi all,
> 
> Thank you to all contributing to FreeSWITCH !
> 
> I'm working on a IVR project where logic is implemented on a HTTP server.
> We are leaving Twilio because we now need to record and live stream the session to the HTTP server in WAV format (chunked transfer encoding).
> 
> mod_httapi looks great (HT TAPI very similar to Twilio's) but it seems that the records are first saved to disk before there are sent to the server as chunked data. 
> We need the transfer to start as soon as the recording starts.
> 
> mod_shout does start the request almost as the records starts but it does not support WAV file and shout:// is not exactly a HTTP request (SOURCE method instead of PUT).
> 
> Is there a way to use these modules to achieve our goal ?
> 
> If not, we are willing to author a specific module or rather contribute to the existing ones.
> 
> We've identified two approaches:
> From mod_httapi
> Modify mod_httapi to directly stream the record instead of completely saving it to disk before the HTTP chunked transfer starts.
> This seems the most logical but with more than 3000 lines, mod_httapi does not seem to be the easiest module to build upon for newcomers!
> 
> From mod_shout
> Modify libshoot to replace the custom SOURCE method with standard HTTP PUT method
> Modify mod_shout to support wav files
> Implement our IVR in script (javascript/lua)
> What do you think ?
> 
> Thank you for your help.
> 
> Cheers,
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From mike at jerris.com  Tue Apr  5 19:43:31 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 5 Apr 2016 11:43:31 -0400
Subject: [Freeswitch-users] Format_cdr modul
In-Reply-To: <CAC7ttFGL5y51X5frqhK1nJcdZtTOe0iDzDbFSF5SEGdk66v_bA@mail.gmail.com>
References: <CAC7ttFEf1-4nP=vj+yn5AA+stAQD45m2j1FLNuUvwB-JEDCwrQ@mail.gmail.com>
	<CAOrA2UZcpiHp7Qgwfv9_G6aKVbP5syie9TVZajvBJihUtveBmQ@mail.gmail.com>
	<CAC7ttFGL5y51X5frqhK1nJcdZtTOe0iDzDbFSF5SEGdk66v_bA@mail.gmail.com>
Message-ID: <0990E2F7-EC17-41BF-898B-5F36A08A13BB@jerris.com>

It looks like we don't have build files for that module, but if you have visual studio 2015, it shouldn't be hard to add.

> On Apr 5, 2016, at 10:08 AM, Gregor Nanger <gregor at infomedia.si> wrote:
> 
> Thank you!
> 
> but I am on windows and it looks like this module is not precompiled, because I get error that it cannot find dll :-(
> 
> 2016-04-05 15:52 GMT+02:00 Jurijs Ivolga <jurijs.ivolga at gmail.com <mailto:jurijs.ivolga at gmail.com>>:
> Hi,
> 
> You can enable it by adding following line in modules.conf.xml:
> 
> <load module="mod_format_cdr"/>
> 
> With kind regards,
> 
> Jurijs
> 
> On Tue, Apr 5, 2016 at 4:34 PM, Gregor Nanger <gregor at infomedia.si <mailto:gregor at infomedia.si>> wrote:
> Hi!
> 
> I would like to use format_cdr module to post json cdr. I can see there is format_cdr.conf file, but module is not defined in modules.conf.
> 
> How can I enable this module?
> 
> Best regards, Gergor
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Gregor Nanger
>  
> CTO
> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 
> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia 
> ? www.infomedia.si <http://www.infomedia.si/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From fdelawarde at wirelessmundi.com  Tue Apr  5 19:55:37 2016
From: fdelawarde at wirelessmundi.com (=?UTF-8?B?RnJhbsOnb2lz?=)
Date: Tue, 5 Apr 2016 17:55:37 +0200
Subject: [Freeswitch-users] Specific ring-device for verto communicator
 (or webrtc in general)
In-Reply-To: <1vzv55159fqgs0xpxejgucjc0-0@mailer.nylas.com>
References: <57028C60.9030306@wirelessmundi.com>
	<1vzv55159fqgs0xpxejgucjc0-0@mailer.nylas.com>
Message-ID: <5703DFF9.1030306@wirelessmundi.com>

Nice! Will verto communicator support this maybe at some point?

Thanks,
F

On 2016-04-05 15:01, ?talo Rossi wrote:
> Yes, you can have as many audio tags you need and each one attached to 
> a different output device
>
> ?talo Rossi
> italo at freeswitch.org
> IRC chat.freenode.net #freeswitch #freeswitch-dev
> Bugs? https://freeswitch.org/jira
> Docs? https://freeswitch.org/jira
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6
>
>
> On Apr 4 2016, at 12:48 pm, Fran?ois <fdelawarde at wirelessmundi.com> 
> wrote:
>
>     Hi,
>
>     Would it be possible to for a webrtc/verto application to use a
>     different audio device for "ringing" purposes (like most
>     softphones do)?
>
>     Thanks,
>     F
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From adahary at gmail.com  Tue Apr  5 19:57:37 2016
From: adahary at gmail.com (Assaf Dahary)
Date: Tue, 05 Apr 2016 18:57:37 +0300
Subject: [Freeswitch-users] extended FS as ZRTP client
In-Reply-To: <F486B3C0-9B54-44C8-AA46-2E8C8513ECFC@jerris.com>
References: <001401d188eb$6dbba080$4932e180$@gmail.com>
	<F486B3C0-9B54-44C8-AA46-2E8C8513ECFC@jerris.com>
Message-ID: <3700c283-701b-4c03-af56-b5783539203f@typeapp.com>

Following Mike's advise, I've compiled the latest FS version 1.6.7 on a Raspi-2 device with the zrtp libs and setup:
./configure --enable-zrtp.
I've enabled zrtp in vars.XML and in dialplan and tried to make a regular zrtp call with several zrtp clients but failed to make it happen.

The fs_CLI log shows the zrtp hash and there is a statement that zrtp is detected BUT no zrtp session is starting with HELLO or anything else. It looks like the ZRTP module is not active.
>From fs_CLI I get zrtp_enabled=false.

I've also recompiled with ver 1.7 with same results.

When I'm recompiling back with ver 1.4 then zrtp is working.

Is there any issue with fs 1.6.7 on raspi-2 using zrtp?

Regards

Assaf



On Mar 28, 2016, 18:13, at 18:13, Michael Jerris <mike at jerris.com> wrote:
>did you try and see if this is the case with a current release instead
>of the old one?
>
>> On Mar 28, 2016, at 8:14 AM, Assaf Dahary <adahary at gmail.com> wrote:
>> 
>> Hi,
>>  
>> I would like to use FS (multiple) as a ZRTP client register on a the
>main FS.
>>  
>> I have already managed to setup a Gateway with user/pass on the FS
>client and register it on the main FS for regular incoming/outgoing
>calls (without ZRTP).
>>  
>> To enable ZRTP calls I setup the FS client/main as follow:
>> None ZRTP SIP phone -> FS client -> NAT Internet -> FS main ->
>CSipSimple ZRTP enabled).
>>  
>> FS Client (ver 1.4) :
>> ZRTP enabled globally in VARS and in dialplan.
>> Media proxy disabled on both internal and external profiles and in
>dialplan.
>>  
>> FS Main (ver 1.4):
>> Media proxy enabled ? including late negotiation.
>>  
>> I forced the FS client and the CSipSimple to use only PCMU codec to
>avoid transcoding.
>>  
>> The problem is that on a call from the SIP phone via the FS client
>there are always CRC errors on the ZRTP log.
>> Only if the FS main is set to disable media proxy then there are no
>CRC errors ? but then it becomes a MITM with incompatible SASs.
>>  
>> From reading other posts about FS and ZRTP CRC errors I assume that
>it happens because the FS client is not creating a zrtp hash in the
>invite SDP.
>>  
>> So my question is how to make the FS client to generate the zrtp hash
>in the invite SDP to act as real ZRTP enabled client? 
>> I've already tried to set the FS client internal/external/dialplan
>with several zrtp configs with no success.
>>  
>> I would appreciate any tip to resolve this issue.
>>  
>> Regards
>>  
>> Assaf  
>>  
>>  
>> The client FS 
>>  
>>
>_________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
><http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
><mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
><http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
><http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>
>
>------------------------------------------------------------------------
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services: 
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://confluence.freeswitch.org
>http://www.cluecon.com
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
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From mike at jerris.com  Tue Apr  5 20:02:44 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 5 Apr 2016 12:02:44 -0400
Subject: [Freeswitch-users] Specific ring-device for verto communicator
	(or webrtc in general)
In-Reply-To: <5703DFF9.1030306@wirelessmundi.com>
References: <57028C60.9030306@wirelessmundi.com>
	<1vzv55159fqgs0xpxejgucjc0-0@mailer.nylas.com>
	<5703DFF9.1030306@wirelessmundi.com>
Message-ID: <D4812591-8B5F-40E4-9FD1-9163C16196EC@jerris.com>

Patches welcome.

> On Apr 5, 2016, at 11:55 AM, Fran?ois <fdelawarde at wirelessmundi.com> wrote:
> 
> Nice! Will verto communicator support this maybe at some point?
> 
> Thanks,
> F
> 
> On 2016-04-05 15:01, ?talo Rossi wrote:
>> Yes, you can have as many audio tags you need and each one attached to a different output device
>> 
>> ?talo Rossi
>> italo at freeswitch.org <mailto:italo at freeswitch.org>
>> IRC chat.freenode.net #freeswitch #freeswitch-dev
>> Bugs? https://freeswitch.org/jira <https://freeswitch.org/jira>
>> Docs? https://freeswitch.org/jira <https://freeswitch.org/jira>
>> Chat? https://hipchat.freeswitch.org/gUdAgy0m6 <https://hipchat.freeswitch.org/gUdAgy0m6>
>> 
>> 
>> On Apr 4 2016, at 12:48 pm, Fran?ois <fdelawarde at wirelessmundi.com> <mailto:fdelawarde at wirelessmundi.com> wrote: 
>> Hi,
>> 
>> Would it be possible to for a webrtc/verto application to use a 
>> different audio device for "ringing" purposes (like most softphones do)?
>> 
>> Thanks,
>> F
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vincent.gire at gmail.com  Tue Apr  5 20:37:29 2016
From: vincent.gire at gmail.com (Vincent Gire)
Date: Tue, 5 Apr 2016 18:37:29 +0200
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <E6948933-09AC-4A48-A0C9-70E206EED453@jerris.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
	<CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
	<CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>
	<CAKsJLpa83aWDBMFuZaYeMFVR6XeN=2uQqwcuTXAtqTY3szckSg@mail.gmail.com>
	<E6948933-09AC-4A48-A0C9-70E206EED453@jerris.com>
Message-ID: <CAKsJLpavsm2N7odd2xw6CP6zuBxhLR_QEy3acEPukujVAPU5HA@mail.gmail.com>

We are building an IVR completely driven by ASR.
ASR is performed in a distant location by a HTTP service (supporting
chunked transfer) and adds an incompressible latency. We would like to
stream the record to the ASR service as soon as it starts to reduce the
overall latency before response.
Does it make sense ?


On Tue, Apr 5, 2016 at 5:39 PM, Michael Jerris <mike at jerris.com> wrote:

> I would stay away from mod_vlc.  Its audio portions with recording have
> known issues.  We do use the rtmp streaming in mod_av heavily but thats
> obviously not wav.  Can you explain a bit more why you have this
> requirement?
>
> On Apr 5, 2016, at 5:36 AM, Vincent Gire <vincent.gire at gmail.com> wrote:
>
> Ok thanks.
> It looks promising !
> I'll dig into mod_vlc.
>
> Best
>
> Vincent
>
> On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Think is requred streaming feature of freeswitch.
>> Look at mod_esf
>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_esf> and
>> mod_vlc <https://wiki.freeswitch.org/wiki/Mod_vlc>
>> Instruction to compile mod_vlc on provided link is to old but helpfull to
>> undestand how to stream media to http server.
>>
>> For compiling mod_vlc please use vlc repo
>> <https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos
>> instruction
>> <https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>
>> .
>> After you intall vlc, then you can enable mod_vlc module in freeswitch
>> sources(SPEC file) and compile freeswitch.
>>
>> Sergey
>>
>>
>> ??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com>:
>>
>>> webdav, mod_http_cache or mod_httapi all results in sending the
>>> recording only *after* it is complete.
>>> They all write the recording to a file, wait for the recording to
>>> complete and the file to close and then send it over HTTP.
>>>
>>> I would like to start sending the recording to the remove server as soon
>>> as it starts (max 1 sec latency).
>>> mod_http_cache or mod_httapi would be perfect if they were streaming the
>>> recording like mod_shout.
>>>
>>>
>>> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <s.safarov at gmail.com>
>>> wrote:
>>>
>>>> Input/output latency is not problem.  I use Kazoo on my servers and
>>>> call recording is stored to database during 5 seconds after hangup.
>>>> What is broken in your case if save file using webdav or http_cache?
>>>>
>>>> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com> wrote:
>>>>
>>>>> Hello Sergey,
>>>>>
>>>>> Thank you for your answer.
>>>>> I've looked into webdav mounted filesystem.
>>>>>
>>>>> Unfortunately, most WebDav clients (especially davfs2 on debian) do a
>>>>> lot of buffering, caching and even lock-null requests (lock a non existent
>>>>> resource before writing to it). I also suspect that they wait for the end
>>>>> of the write operation.
>>>>> The result is a latency of a few seconds witch is not much better than
>>>>> what I achieve with mod_shout if I transcode the MP3.
>>>>>
>>>>> Any other idea ?
>>>>>
>>>>> Thank you !
>>>>>
>>>>> Best regards
>>>>>
>>>>> Vincent
>>>>>
>>>>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <s.safarov at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Please look at webdav mounted filesystem.
>>>>>>
>>>>>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Hi all,
>>>>>>>
>>>>>>> Thank you to all contributing to FreeSWITCH !
>>>>>>>
>>>>>>> I'm working on a IVR project where logic is implemented on a HTTP
>>>>>>> server.
>>>>>>> We are leaving Twilio because we now need to record and live stream
>>>>>>> the session to the HTTP server in WAV format (chunked transfer encoding).
>>>>>>>
>>>>>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's) but it
>>>>>>> seems that the records are first saved to disk before there are sent to the
>>>>>>> server as chunked data.
>>>>>>> We need the transfer to start as soon as the recording starts.
>>>>>>>
>>>>>>> *mod_shout* does start the request almost as the records starts but
>>>>>>> it does not support WAV file and shout:// is not exactly a HTTP request
>>>>>>> (SOURCE method instead of PUT).
>>>>>>>
>>>>>>> Is there a way to use these modules to achieve our goal ?
>>>>>>>
>>>>>>> If not, we are willing to author a specific module or rather
>>>>>>> contribute to the existing ones.
>>>>>>>
>>>>>>> We've identified two approaches:
>>>>>>>
>>>>>>>    1. From *mod_httapi* Modify mod_httapi to directly stream the
>>>>>>>    record instead of completely saving it to disk before the HTTP chunked
>>>>>>>    transfer starts.
>>>>>>>    This seems the most logical but with more than 3000 lines,
>>>>>>>    mod_httapi does not seem to be the easiest module to build upon for
>>>>>>>    newcomers!
>>>>>>>
>>>>>>>    2. From *mod_shout*
>>>>>>>       1. Modify libshoot to replace the custom SOURCE method with
>>>>>>>       standard HTTP PUT method
>>>>>>>       2. Modify mod_shout to support wav files
>>>>>>>       3. Implement our IVR in script (javascript/lua)
>>>>>>>
>>>>>>> What do you think ?
>>>>>>>
>>>>>>> Thank you for your help.
>>>>>>>
>>>>>>> Cheers,
>>>>>>>
>>>>>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Vincent Gire
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From s.safarov at gmail.com  Tue Apr  5 21:04:47 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 05 Apr 2016 17:04:47 +0000
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAKsJLpavsm2N7odd2xw6CP6zuBxhLR_QEy3acEPukujVAPU5HA@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
	<CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
	<CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>
	<CAKsJLpa83aWDBMFuZaYeMFVR6XeN=2uQqwcuTXAtqTY3szckSg@mail.gmail.com>
	<E6948933-09AC-4A48-A0C9-70E206EED453@jerris.com>
	<CAKsJLpavsm2N7odd2xw6CP6zuBxhLR_QEy3acEPukujVAPU5HA@mail.gmail.com>
Message-ID: <CAHtxdDfQGXWAUopsmOUO3hTV2A91i9t=9ss5P3+GtnxPtwwsfg@mail.gmail.com>

Look at http://www.unimrcp.org/ and
https://wiki.freeswitch.org/wiki/Mod_unimrcp

On Tue, Apr 5, 2016, 19:38 Vincent Gire <vincent.gire at gmail.com> wrote:

> We are building an IVR completely driven by ASR.
> ASR is performed in a distant location by a HTTP service (supporting
> chunked transfer) and adds an incompressible latency. We would like to
> stream the record to the ASR service as soon as it starts to reduce the
> overall latency before response.
> Does it make sense ?
>
>
> On Tue, Apr 5, 2016 at 5:39 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I would stay away from mod_vlc.  Its audio portions with recording have
>> known issues.  We do use the rtmp streaming in mod_av heavily but thats
>> obviously not wav.  Can you explain a bit more why you have this
>> requirement?
>>
>> On Apr 5, 2016, at 5:36 AM, Vincent Gire <vincent.gire at gmail.com> wrote:
>>
>> Ok thanks.
>> It looks promising !
>> I'll dig into mod_vlc.
>>
>> Best
>>
>> Vincent
>>
>> On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> Think is requred streaming feature of freeswitch.
>>> Look at mod_esf
>>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_esf> and
>>> mod_vlc <https://wiki.freeswitch.org/wiki/Mod_vlc>
>>> Instruction to compile mod_vlc on provided link is to old but helpfull
>>> to undestand how to stream media to http server.
>>>
>>> For compiling mod_vlc please use vlc repo
>>> <https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos
>>> instruction
>>> <https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>
>>> .
>>> After you intall vlc, then you can enable mod_vlc module in freeswitch
>>> sources(SPEC file) and compile freeswitch.
>>>
>>> Sergey
>>>
>>>
>>> ??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com>:
>>>
>>>> webdav, mod_http_cache or mod_httapi all results in sending the
>>>> recording only *after* it is complete.
>>>> They all write the recording to a file, wait for the recording to
>>>> complete and the file to close and then send it over HTTP.
>>>>
>>>> I would like to start sending the recording to the remove server as
>>>> soon as it starts (max 1 sec latency).
>>>> mod_http_cache or mod_httapi would be perfect if they were streaming
>>>> the recording like mod_shout.
>>>>
>>>>
>>>> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <s.safarov at gmail.com>
>>>> wrote:
>>>>
>>>>> Input/output latency is not problem.  I use Kazoo on my servers and
>>>>> call recording is stored to database during 5 seconds after hangup.
>>>>> What is broken in your case if save file using webdav or http_cache?
>>>>>
>>>>> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hello Sergey,
>>>>>>
>>>>>> Thank you for your answer.
>>>>>> I've looked into webdav mounted filesystem.
>>>>>>
>>>>>> Unfortunately, most WebDav clients (especially davfs2 on debian) do a
>>>>>> lot of buffering, caching and even lock-null requests (lock a non existent
>>>>>> resource before writing to it). I also suspect that they wait for the end
>>>>>> of the write operation.
>>>>>> The result is a latency of a few seconds witch is not much better
>>>>>> than what I achieve with mod_shout if I transcode the MP3.
>>>>>>
>>>>>> Any other idea ?
>>>>>>
>>>>>> Thank you !
>>>>>>
>>>>>> Best regards
>>>>>>
>>>>>> Vincent
>>>>>>
>>>>>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <s.safarov at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Please look at webdav mounted filesystem.
>>>>>>>
>>>>>>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Hi all,
>>>>>>>>
>>>>>>>> Thank you to all contributing to FreeSWITCH !
>>>>>>>>
>>>>>>>> I'm working on a IVR project where logic is implemented on a HTTP
>>>>>>>> server.
>>>>>>>> We are leaving Twilio because we now need to record and live stream
>>>>>>>> the session to the HTTP server in WAV format (chunked transfer encoding).
>>>>>>>>
>>>>>>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's) but it
>>>>>>>> seems that the records are first saved to disk before there are sent to the
>>>>>>>> server as chunked data.
>>>>>>>> We need the transfer to start as soon as the recording starts.
>>>>>>>>
>>>>>>>> *mod_shout* does start the request almost as the records starts
>>>>>>>> but it does not support WAV file and shout:// is not exactly a HTTP request
>>>>>>>> (SOURCE method instead of PUT).
>>>>>>>>
>>>>>>>> Is there a way to use these modules to achieve our goal ?
>>>>>>>>
>>>>>>>> If not, we are willing to author a specific module or rather
>>>>>>>> contribute to the existing ones.
>>>>>>>>
>>>>>>>> We've identified two approaches:
>>>>>>>>
>>>>>>>>    1. From *mod_httapi* Modify mod_httapi to directly stream the
>>>>>>>>    record instead of completely saving it to disk before the HTTP chunked
>>>>>>>>    transfer starts.
>>>>>>>>    This seems the most logical but with more than 3000 lines,
>>>>>>>>    mod_httapi does not seem to be the easiest module to build upon for
>>>>>>>>    newcomers!
>>>>>>>>
>>>>>>>>    2. From *mod_shout*
>>>>>>>>       1. Modify libshoot to replace the custom SOURCE method with
>>>>>>>>       standard HTTP PUT method
>>>>>>>>       2. Modify mod_shout to support wav files
>>>>>>>>       3. Implement our IVR in script (javascript/lua)
>>>>>>>>
>>>>>>>> What do you think ?
>>>>>>>>
>>>>>>>> Thank you for your help.
>>>>>>>>
>>>>>>>> Cheers,
>>>>>>>>
>>>>>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Vincent Gire
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  5 21:14:50 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 5 Apr 2016 13:14:50 -0400
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAHtxdDfQGXWAUopsmOUO3hTV2A91i9t=9ss5P3+GtnxPtwwsfg@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
	<CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
	<CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>
	<CAKsJLpa83aWDBMFuZaYeMFVR6XeN=2uQqwcuTXAtqTY3szckSg@mail.gmail.com>
	<E6948933-09AC-4A48-A0C9-70E206EED453@jerris.com>
	<CAKsJLpavsm2N7odd2xw6CP6zuBxhLR_QEy3acEPukujVAPU5HA@mail.gmail.com>
	<CAHtxdDfQGXWAUopsmOUO3hTV2A91i9t=9ss5P3+GtnxPtwwsfg@mail.gmail.com>
Message-ID: <CAEwTsAid_21ysmiu3E+iUdvazMGpLpb0KOpVCOxK6WaurmnWEA@mail.gmail.com>

Unimrcp isn't going to provide the interface he's talking about.  Can I ask
which engine this is?

On Tuesday, April 5, 2016, Sergey Safarov <s.safarov at gmail.com> wrote:

> Look at http://www.unimrcp.org/ and
> https://wiki.freeswitch.org/wiki/Mod_unimrcp
>
> On Tue, Apr 5, 2016, 19:38 Vincent Gire <vincent.gire at gmail.com
> <javascript:_e(%7B%7D,'cvml','vincent.gire at gmail.com');>> wrote:
>
>> We are building an IVR completely driven by ASR.
>> ASR is performed in a distant location by a HTTP service (supporting
>> chunked transfer) and adds an incompressible latency. We would like to
>> stream the record to the ASR service as soon as it starts to reduce the
>> overall latency before response.
>> Does it make sense ?
>>
>>
>> On Tue, Apr 5, 2016 at 5:39 PM, Michael Jerris <mike at jerris.com
>> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>>
>>> I would stay away from mod_vlc.  Its audio portions with recording have
>>> known issues.  We do use the rtmp streaming in mod_av heavily but thats
>>> obviously not wav.  Can you explain a bit more why you have this
>>> requirement?
>>>
>>> On Apr 5, 2016, at 5:36 AM, Vincent Gire <vincent.gire at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','vincent.gire at gmail.com');>> wrote:
>>>
>>> Ok thanks.
>>> It looks promising !
>>> I'll dig into mod_vlc.
>>>
>>> Best
>>>
>>> Vincent
>>>
>>> On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <s.safarov at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','s.safarov at gmail.com');>> wrote:
>>>
>>>> Think is requred streaming feature of freeswitch.
>>>> Look at mod_esf
>>>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_esf> and
>>>> mod_vlc <https://wiki.freeswitch.org/wiki/Mod_vlc>
>>>> Instruction to compile mod_vlc on provided link is to old but helpfull
>>>> to undestand how to stream media to http server.
>>>>
>>>> For compiling mod_vlc please use vlc repo
>>>> <https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos
>>>> instruction
>>>> <https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>
>>>> .
>>>> After you intall vlc, then you can enable mod_vlc module in freeswitch
>>>> sources(SPEC file) and compile freeswitch.
>>>>
>>>> Sergey
>>>>
>>>>
>>>> ??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com
>>>> <javascript:_e(%7B%7D,'cvml','vincent.gire at gmail.com');>>:
>>>>
>>>>> webdav, mod_http_cache or mod_httapi all results in sending the
>>>>> recording only *after* it is complete.
>>>>> They all write the recording to a file, wait for the recording to
>>>>> complete and the file to close and then send it over HTTP.
>>>>>
>>>>> I would like to start sending the recording to the remove server as
>>>>> soon as it starts (max 1 sec latency).
>>>>> mod_http_cache or mod_httapi would be perfect if they were streaming
>>>>> the recording like mod_shout.
>>>>>
>>>>>
>>>>> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <s.safarov at gmail.com
>>>>> <javascript:_e(%7B%7D,'cvml','s.safarov at gmail.com');>> wrote:
>>>>>
>>>>>> Input/output latency is not problem.  I use Kazoo on my servers and
>>>>>> call recording is stored to database during 5 seconds after hangup.
>>>>>> What is broken in your case if save file using webdav or http_cache?
>>>>>>
>>>>>> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com
>>>>>> <javascript:_e(%7B%7D,'cvml','vincent.gire at gmail.com');>> wrote:
>>>>>>
>>>>>>> Hello Sergey,
>>>>>>>
>>>>>>> Thank you for your answer.
>>>>>>> I've looked into webdav mounted filesystem.
>>>>>>>
>>>>>>> Unfortunately, most WebDav clients (especially davfs2 on debian) do
>>>>>>> a lot of buffering, caching and even lock-null requests (lock a non
>>>>>>> existent resource before writing to it). I also suspect that they wait for
>>>>>>> the end of the write operation.
>>>>>>> The result is a latency of a few seconds witch is not much better
>>>>>>> than what I achieve with mod_shout if I transcode the MP3.
>>>>>>>
>>>>>>> Any other idea ?
>>>>>>>
>>>>>>> Thank you !
>>>>>>>
>>>>>>> Best regards
>>>>>>>
>>>>>>> Vincent
>>>>>>>
>>>>>>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <s.safarov at gmail.com
>>>>>>> <javascript:_e(%7B%7D,'cvml','s.safarov at gmail.com');>> wrote:
>>>>>>>
>>>>>>>> Please look at webdav mounted filesystem.
>>>>>>>>
>>>>>>>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com
>>>>>>>> <javascript:_e(%7B%7D,'cvml','vincent.gire at gmail.com');>> wrote:
>>>>>>>>
>>>>>>>>> Hi all,
>>>>>>>>>
>>>>>>>>> Thank you to all contributing to FreeSWITCH !
>>>>>>>>>
>>>>>>>>> I'm working on a IVR project where logic is implemented on a HTTP
>>>>>>>>> server.
>>>>>>>>> We are leaving Twilio because we now need to record and live
>>>>>>>>> stream the session to the HTTP server in WAV format (chunked transfer
>>>>>>>>> encoding).
>>>>>>>>>
>>>>>>>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's) but
>>>>>>>>> it seems that the records are first saved to disk before there are sent to
>>>>>>>>> the server as chunked data.
>>>>>>>>> We need the transfer to start as soon as the recording starts.
>>>>>>>>>
>>>>>>>>> *mod_shout* does start the request almost as the records starts
>>>>>>>>> but it does not support WAV file and shout:// is not exactly a HTTP request
>>>>>>>>> (SOURCE method instead of PUT).
>>>>>>>>>
>>>>>>>>> Is there a way to use these modules to achieve our goal ?
>>>>>>>>>
>>>>>>>>> If not, we are willing to author a specific module or rather
>>>>>>>>> contribute to the existing ones.
>>>>>>>>>
>>>>>>>>> We've identified two approaches:
>>>>>>>>>
>>>>>>>>>    1. From *mod_httapi* Modify mod_httapi to directly stream the
>>>>>>>>>    record instead of completely saving it to disk before the HTTP chunked
>>>>>>>>>    transfer starts.
>>>>>>>>>    This seems the most logical but with more than 3000 lines,
>>>>>>>>>    mod_httapi does not seem to be the easiest module to build upon for
>>>>>>>>>    newcomers!
>>>>>>>>>
>>>>>>>>>    2. From *mod_shout*
>>>>>>>>>       1. Modify libshoot to replace the custom SOURCE method with
>>>>>>>>>       standard HTTP PUT method
>>>>>>>>>       2. Modify mod_shout to support wav files
>>>>>>>>>       3. Implement our IVR in script (javascript/lua)
>>>>>>>>>
>>>>>>>>> What do you think ?
>>>>>>>>>
>>>>>>>>> Thank you for your help.
>>>>>>>>>
>>>>>>>>> Cheers,
>>>>>>>>>
>>>>>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Vincent Gire
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From vincent.gire at gmail.com  Tue Apr  5 21:24:28 2016
From: vincent.gire at gmail.com (Vincent Gire)
Date: Tue, 5 Apr 2016 19:24:28 +0200
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAEwTsAid_21ysmiu3E+iUdvazMGpLpb0KOpVCOxK6WaurmnWEA@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
	<CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
	<CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>
	<CAKsJLpa83aWDBMFuZaYeMFVR6XeN=2uQqwcuTXAtqTY3szckSg@mail.gmail.com>
	<E6948933-09AC-4A48-A0C9-70E206EED453@jerris.com>
	<CAKsJLpavsm2N7odd2xw6CP6zuBxhLR_QEy3acEPukujVAPU5HA@mail.gmail.com>
	<CAHtxdDfQGXWAUopsmOUO3hTV2A91i9t=9ss5P3+GtnxPtwwsfg@mail.gmail.com>
	<CAEwTsAid_21ysmiu3E+iUdvazMGpLpb0KOpVCOxK6WaurmnWEA@mail.gmail.com>
Message-ID: <CAKsJLpYAR333mU6VuD2AjxepbPJ4quyzGKpW2ziOmMLds6rgsg@mail.gmail.com>

We use multiple ones to compare confidence.
They all support chunk transfer and provide a HTTP transaction similar to :
http://developer.att.com/apis/speech/docs/v3
Look at 3/ Make API Calls

On Tue, Apr 5, 2016 at 7:14 PM, Michael Jerris <mike at jerris.com> wrote:

> Unimrcp isn't going to provide the interface he's talking about.  Can I
> ask which engine this is?
>
>
> On Tuesday, April 5, 2016, Sergey Safarov <s.safarov at gmail.com> wrote:
>
>> Look at http://www.unimrcp.org/ and
>> https://wiki.freeswitch.org/wiki/Mod_unimrcp
>>
>> On Tue, Apr 5, 2016, 19:38 Vincent Gire <vincent.gire at gmail.com> wrote:
>>
>>> We are building an IVR completely driven by ASR.
>>> ASR is performed in a distant location by a HTTP service (supporting
>>> chunked transfer) and adds an incompressible latency. We would like to
>>> stream the record to the ASR service as soon as it starts to reduce the
>>> overall latency before response.
>>> Does it make sense ?
>>>
>>>
>>> On Tue, Apr 5, 2016 at 5:39 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> I would stay away from mod_vlc.  Its audio portions with recording have
>>>> known issues.  We do use the rtmp streaming in mod_av heavily but thats
>>>> obviously not wav.  Can you explain a bit more why you have this
>>>> requirement?
>>>>
>>>> On Apr 5, 2016, at 5:36 AM, Vincent Gire <vincent.gire at gmail.com>
>>>> wrote:
>>>>
>>>> Ok thanks.
>>>> It looks promising !
>>>> I'll dig into mod_vlc.
>>>>
>>>> Best
>>>>
>>>> Vincent
>>>>
>>>> On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <s.safarov at gmail.com>
>>>> wrote:
>>>>
>>>>> Think is requred streaming feature of freeswitch.
>>>>> Look at mod_esf
>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_esf> and
>>>>> mod_vlc <https://wiki.freeswitch.org/wiki/Mod_vlc>
>>>>> Instruction to compile mod_vlc on provided link is to old but helpfull
>>>>> to undestand how to stream media to http server.
>>>>>
>>>>> For compiling mod_vlc please use vlc repo
>>>>> <https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos
>>>>> instruction
>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>
>>>>> .
>>>>> After you intall vlc, then you can enable mod_vlc module in freeswitch
>>>>> sources(SPEC file) and compile freeswitch.
>>>>>
>>>>> Sergey
>>>>>
>>>>>
>>>>> ??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com>:
>>>>>
>>>>>> webdav, mod_http_cache or mod_httapi all results in sending the
>>>>>> recording only *after* it is complete.
>>>>>> They all write the recording to a file, wait for the recording to
>>>>>> complete and the file to close and then send it over HTTP.
>>>>>>
>>>>>> I would like to start sending the recording to the remove server as
>>>>>> soon as it starts (max 1 sec latency).
>>>>>> mod_http_cache or mod_httapi would be perfect if they were streaming
>>>>>> the recording like mod_shout.
>>>>>>
>>>>>>
>>>>>> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <s.safarov at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Input/output latency is not problem.  I use Kazoo on my servers and
>>>>>>> call recording is stored to database during 5 seconds after hangup.
>>>>>>> What is broken in your case if save file using webdav or http_cache?
>>>>>>>
>>>>>>> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Hello Sergey,
>>>>>>>>
>>>>>>>> Thank you for your answer.
>>>>>>>> I've looked into webdav mounted filesystem.
>>>>>>>>
>>>>>>>> Unfortunately, most WebDav clients (especially davfs2 on debian) do
>>>>>>>> a lot of buffering, caching and even lock-null requests (lock a non
>>>>>>>> existent resource before writing to it). I also suspect that they wait for
>>>>>>>> the end of the write operation.
>>>>>>>> The result is a latency of a few seconds witch is not much better
>>>>>>>> than what I achieve with mod_shout if I transcode the MP3.
>>>>>>>>
>>>>>>>> Any other idea ?
>>>>>>>>
>>>>>>>> Thank you !
>>>>>>>>
>>>>>>>> Best regards
>>>>>>>>
>>>>>>>> Vincent
>>>>>>>>
>>>>>>>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <s.safarov at gmail.com
>>>>>>>> > wrote:
>>>>>>>>
>>>>>>>>> Please look at webdav mounted filesystem.
>>>>>>>>>
>>>>>>>>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> Hi all,
>>>>>>>>>>
>>>>>>>>>> Thank you to all contributing to FreeSWITCH !
>>>>>>>>>>
>>>>>>>>>> I'm working on a IVR project where logic is implemented on a HTTP
>>>>>>>>>> server.
>>>>>>>>>> We are leaving Twilio because we now need to record and live
>>>>>>>>>> stream the session to the HTTP server in WAV format (chunked transfer
>>>>>>>>>> encoding).
>>>>>>>>>>
>>>>>>>>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's) but
>>>>>>>>>> it seems that the records are first saved to disk before there are sent to
>>>>>>>>>> the server as chunked data.
>>>>>>>>>> We need the transfer to start as soon as the recording starts.
>>>>>>>>>>
>>>>>>>>>> *mod_shout* does start the request almost as the records starts
>>>>>>>>>> but it does not support WAV file and shout:// is not exactly a HTTP request
>>>>>>>>>> (SOURCE method instead of PUT).
>>>>>>>>>>
>>>>>>>>>> Is there a way to use these modules to achieve our goal ?
>>>>>>>>>>
>>>>>>>>>> If not, we are willing to author a specific module or rather
>>>>>>>>>> contribute to the existing ones.
>>>>>>>>>>
>>>>>>>>>> We've identified two approaches:
>>>>>>>>>>
>>>>>>>>>>    1. From *mod_httapi* Modify mod_httapi to directly stream the
>>>>>>>>>>    record instead of completely saving it to disk before the HTTP chunked
>>>>>>>>>>    transfer starts.
>>>>>>>>>>    This seems the most logical but with more than 3000 lines,
>>>>>>>>>>    mod_httapi does not seem to be the easiest module to build upon for
>>>>>>>>>>    newcomers!
>>>>>>>>>>
>>>>>>>>>>    2. From *mod_shout*
>>>>>>>>>>       1. Modify libshoot to replace the custom SOURCE method
>>>>>>>>>>       with standard HTTP PUT method
>>>>>>>>>>       2. Modify mod_shout to support wav files
>>>>>>>>>>       3. Implement our IVR in script (javascript/lua)
>>>>>>>>>>
>>>>>>>>>> What do you think ?
>>>>>>>>>>
>>>>>>>>>> Thank you for your help.
>>>>>>>>>>
>>>>>>>>>> Cheers,
>>>>>>>>>>
>>>>>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Vincent Gire
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Vincent Gire
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From mike at jerris.com  Wed Apr  6 02:22:37 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 5 Apr 2016 18:22:37 -0400
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAKsJLpYAR333mU6VuD2AjxepbPJ4quyzGKpW2ziOmMLds6rgsg@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
	<CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
	<CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>
	<CAKsJLpa83aWDBMFuZaYeMFVR6XeN=2uQqwcuTXAtqTY3szckSg@mail.gmail.com>
	<E6948933-09AC-4A48-A0C9-70E206EED453@jerris.com>
	<CAKsJLpavsm2N7odd2xw6CP6zuBxhLR_QEy3acEPukujVAPU5HA@mail.gmail.com>
	<CAHtxdDfQGXWAUopsmOUO3hTV2A91i9t=9ss5P3+GtnxPtwwsfg@mail.gmail.com>
	<CAEwTsAid_21ysmiu3E+iUdvazMGpLpb0KOpVCOxK6WaurmnWEA@mail.gmail.com>
	<CAKsJLpYAR333mU6VuD2AjxepbPJ4quyzGKpW2ziOmMLds6rgsg@mail.gmail.com>
Message-ID: <731C9BFA-0101-42F3-B368-17F8AEFD7580@jerris.com>

The best way to handle this is probably to write a custom module against the speech interface that sends the streams like you are describing, and supports multiple providers.  How exactly do you go about sending to multiple at the same time and combining the results, as this might actually be the trickiest part as youll need to integrate that into a module that uses the speech interface to be able to at all sanely handle that in freeswitch

> On Apr 5, 2016, at 1:24 PM, Vincent Gire <vincent.gire at gmail.com> wrote:
> 
> We use multiple ones to compare confidence.
> They all support chunk transfer and provide a HTTP transaction similar to :
> http://developer.att.com/apis/speech/docs/v3 <http://developer.att.com/apis/speech/docs/v3>
> Look at 3/ Make API Calls
> 
> On Tue, Apr 5, 2016 at 7:14 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Unimrcp isn't going to provide the interface he's talking about.  Can I ask which engine this is?
> 
> 
> On Tuesday, April 5, 2016, Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>> wrote:
> Look at http://www.unimrcp.org/ <http://www.unimrcp.org/> and https://wiki.freeswitch.org/wiki/Mod_unimrcp <https://wiki.freeswitch.org/wiki/Mod_unimrcp>
> On Tue, Apr 5, 2016, 19:38 Vincent Gire <vincent.gire at gmail.com <>> wrote:
> We are building an IVR completely driven by ASR.
> ASR is performed in a distant location by a HTTP service (supporting chunked transfer) and adds an incompressible latency. We would like to stream the record to the ASR service as soon as it starts to reduce the overall latency before response.
> Does it make sense ?
> 
> 
> On Tue, Apr 5, 2016 at 5:39 PM, Michael Jerris <mike at jerris.com <>> wrote:
> I would stay away from mod_vlc.  Its audio portions with recording have known issues.  We do use the rtmp streaming in mod_av heavily but thats obviously not wav.  Can you explain a bit more why you have this requirement?
> 
>> On Apr 5, 2016, at 5:36 AM, Vincent Gire <vincent.gire at gmail.com <>> wrote:
>> 
>> Ok thanks.
>> It looks promising !
>> I'll dig into mod_vlc.
>> 
>> Best 
>> 
>> Vincent
>> 
>> On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <s.safarov at gmail.com <>> wrote:
>> Think is requred streaming feature of freeswitch.
>> Look at mod_esf <https://freeswitch.org/confluence/display/FREESWITCH/mod_esf> and mod_vlc <https://wiki.freeswitch.org/wiki/Mod_vlc>
>> Instruction to compile mod_vlc on provided link is to old but helpfull to undestand how to stream media to http server.
>> 
>> For compiling mod_vlc please use vlc repo <https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos instruction <https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>.
>> After you intall vlc, then you can enable mod_vlc module in freeswitch sources(SPEC file) and compile freeswitch.
>> 
>> Sergey
>> 
>> 
>> ??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com <>>:
>> webdav, mod_http_cache or mod_httapi all results in sending the recording only after it is complete.
>> They all write the recording to a file, wait for the recording to complete and the file to close and then send it over HTTP.
>> 
>> I would like to start sending the recording to the remove server as soon as it starts (max 1 sec latency).
>> mod_http_cache or mod_httapi would be perfect if they were streaming the recording like mod_shout.
>> 
>> 
>> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <s.safarov at gmail.com <>> wrote:
>> Input/output latency is not problem.  I use Kazoo on my servers and call recording is stored to database during 5 seconds after hangup.
>> What is broken in your case if save file using webdav or http_cache?
>> 
>> 
>> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com <>> wrote:
>> Hello Sergey,
>> 
>> Thank you for your answer.
>> I've looked into webdav mounted filesystem.
>> 
>> Unfortunately, most WebDav clients (especially davfs2 on debian) do a lot of buffering, caching and even lock-null requests (lock a non existent resource before writing to it). I also suspect that they wait for the end of the write operation.
>> The result is a latency of a few seconds witch is not much better than what I achieve with mod_shout if I transcode the MP3.
>> 
>> Any other idea ?
>> 
>> Thank you !
>> 
>> Best regards
>> 
>> Vincent
>> 
>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <s.safarov at gmail.com <>> wrote:
>> Please look at webdav mounted filesystem.
>> 
>> 
>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com <>> wrote:
>> Hi all,
>> 
>> Thank you to all contributing to FreeSWITCH !
>> 
>> I'm working on a IVR project where logic is implemented on a HTTP server.
>> We are leaving Twilio because we now need to record and live stream the session to the HTTP server in WAV format (chunked transfer encoding).
>> 
>> mod_httapi looks great (HT TAPI very similar to Twilio's) but it seems that the records are first saved to disk before there are sent to the server as chunked data. 
>> We need the transfer to start as soon as the recording starts.
>> 
>> mod_shout does start the request almost as the records starts but it does not support WAV file and shout:// is not exactly a HTTP request (SOURCE method instead of PUT).
>> 
>> Is there a way to use these modules to achieve our goal ?
>> 
>> If not, we are willing to author a specific module or rather contribute to the existing ones.
>> 
>> We've identified two approaches:
>> From mod_httapi
>> Modify mod_httapi to directly stream the record instead of completely saving it to disk before the HTTP chunked transfer starts.
>> This seems the most logical but with more than 3000 lines, mod_httapi does not seem to be the easiest module to build upon for newcomers!
>> 
>> From mod_shout
>> Modify libshoot to replace the custom SOURCE method with standard HTTP PUT method
>> Modify mod_shout to support wav files
>> Implement our IVR in script (javascript/lua)
>> What do you think ?
>> 
>> Thank you for your help.
>> 
>> Cheers,
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Vincent Gire
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Vincent Gire
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
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From jelena at misticnabica.hr  Wed Apr  6 02:33:54 2016
From: jelena at misticnabica.hr (jelena at misticnabica.hr)
Date: Tue, 5 Apr 2016 22:33:54 GMT
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
Message-ID: <mailman.1.1459895668.8220.freeswitch-users@lists.freeswitch.org>



From gregor at infomedia.si  Wed Apr  6 02:38:04 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Wed, 6 Apr 2016 00:38:04 +0200
Subject: [Freeswitch-users] Windows build
In-Reply-To: <CAHtxdDdY2JRqVuw-W0b+UF0Z_ZUGbCj_iUFUOXi=nrxDDzKb9A@mail.gmail.com>
References: <CAC7ttFFYf+dt3NfgRQy0R5PbNut0=-=mmxv6+gTvUYFB1RsKBA@mail.gmail.com>
	<CAPEdYThObr1ha7ffrFNijAj3H30Z6Zk41TETEgdoeZWxLu6igg@mail.gmail.com>
	<CAC7ttFE98byMZX+uNYUcLJ-A8+WnQwcwXBqqPO-g2fBu3XRfvA@mail.gmail.com>
	<CAPEdYTghm8J20E5COERvG1TidbYNUqcVsjD1TngTyH-zwWE52w@mail.gmail.com>
	<CAC7ttFGRJQAFn0vC3sCC1tguvWmZ==BpKR4i-=UWjDPRgGx-hA@mail.gmail.com>
	<9589f6e5d757412ba5d620cf8e2bf26c@imladris.sermotec.local>
	<56E10D5D.1000303@zg.t-com.hr>
	<38494249596647c6a08591eca41e3000@imladris.sermotec.local>
	<CAC7ttFFbv9YAxXNg7zjc1B9cxLS74v6f6HfBpSt09sULor6EoA@mail.gmail.com>
	<CAHtxdDdY2JRqVuw-W0b+UF0Z_ZUGbCj_iUFUOXi=nrxDDzKb9A@mail.gmail.com>
Message-ID: <CAC7ttFErJO-gTLBQTHy_6zCcRWBxOYgsXtJXqMC0U8Tv+79AOg@mail.gmail.com>

Just bumping this post if someone has any hints.

I am trying to build Freeswitch with Visual studio 2015, but keep getting
error:
Error MSB4057 The target "v8:Rebuild" does not exist in the project.
[C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
? ?
C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
?"


Would realy like to successfuly build Freeswitch in windows. I tried with
1.6 branch.

Best regards, Gregor

2016-03-17 5:43 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:

> One week ago I successfully compiled mod_V8 on CentOS 7.
> May be switch to Linux?
>
> On Thu, Mar 17, 2016 at 12:49 AM, Gregor Nanger <gregor at infomedia.si>
> wrote:
>
>> Thank you Harald.
>>
>> I tried with latest branch 1.7 and x64 and got errors regarding mod_V8. I
>> tried what Peter suggested, but I am more in c# and web projects and do not
>> have experience in building  C++ projects. There is already prebuilt setup
>> on freeswitch site, so someone successfully build it :-)) It is not so
>> important for us at this point to make own build, so will try again later.
>>
>>
>> 2016-03-11 7:32 GMT+01:00 Harald Petrovitsch <
>> Harald.Petrovitsch at sermotec.at>:
>>
>>> Hi Gregor,
>>>
>>>
>>>
>>> I only do a
>>>
>>> Git.exe clone ?bv1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git
>>> .
>>>
>>>
>>>
>>> Loaded the solution into vs2015, set configuration Win32 / Release and
>>> press f7 (need to do it two times)
>>>
>>>
>>>
>>> The build ended with
>>>
>>> ========== Build: 20 succeeded, 0 failed, 157 up-to-date, 15 skipped
>>> ==========
>>>
>>>
>>>
>>> I?ve attached a list of the generated mod folder
>>>
>>>
>>>
>>> Regards
>>>
>>> Harald
>>>
>>>
>>>
>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Shishko
>>> *Gesendet:* Donnerstag, 10. M?rz 2016 07:00
>>> *An:* freeswitch-users at lists.freeswitch.org
>>>
>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>
>>>
>>>
>>> Hi Harald,
>>>
>>> what did you do to build libv8 and mod_v8? I tried with VS2015 Update 1,
>>> branch 1.6, but to no avail.
>>>
>>> Thanks
>>>
>>> On 03/07/2016 08:20 AM, Harald Petrovitsch wrote:
>>>
>>> Hi Gregor !
>>>
>>>
>>>
>>> V8 libs and mod builds fine here (visual Studio 2015 Sp1, 1.6 branch,
>>> used tortoiseGit to download it))
>>>
>>>
>>>
>>> Regards
>>>
>>> Harald
>>>
>>>
>>>
>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [
>>> mailto:freeswitch-users-bounces at lists.freeswitch.org
>>> <freeswitch-users-bounces at lists.freeswitch.org>] *Im Auftrag v**on *Gregor
>>> Nanger
>>> *Gesendet:* Montag, 07. M?rz 2016 00:19
>>> *An:* FreeSWITCH Users Help
>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>
>>>
>>>
>>> Thank you, H
>>>
>>> ?a?
>>>
>>> rald. This works: "AFAIK, for the ?'lame/lame.h'? you have to change the
>>> include line to  only ?lame.h?"
>>>
>>> B
>>>
>>> ?ut for v8 stil do  not have solution. I do not want to exclude mod_v8,
>>> since this module runs javascript. But, can you please confirm me that is
>>> not yet compatible, to stop trying to solve it.
>>>
>>>
>>> Any other suggestion what does this mean:
>>> "
>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>
>>> ? ?
>>>
>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>
>>> ?"
>>>
>>>
>>>
>>> Best regards, Gregor?
>>>
>>>
>>>
>>> 2016-03-06 16:39 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>
>>> Remove mod_v8 from the build. I don't think it's compatible with VS2015
>>> for now. However, all other modules should be ok.
>>>
>>>
>>>
>>> /Peter
>>>
>>>
>>>
>>> 2016-03-06 12:31 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>
>>> ?This helped a lot, thank you. Now I have only few errors. Any hint?
>>>
>>>
>>>
>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln]
>>> libv8 C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>
>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>
>>> Error C1083 Cannot open include file: 'lame/lame.h': No such file or
>>> directory mod_shout C:\Git\freeswitch\src\mod\formats\mod_shout\mod_shout.c
>>> 38
>>>
>>> Error LNK1181 cannot open input file 'icui18n.lib' mod_v8
>>> C:\Git\freeswitch\src\mod\languages\mod_v8\LINK 1
>>>
>>>
>>>
>>> ?
>>>
>>>
>>>
>>> 2016-03-06 7:53 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>
>>> One common mistake is that you allow Git to modify line endings. Make
>>> sure autocrlf is turned off - then clone the repository again from scratch.
>>>
>>>
>>>
>>> Also, I'm not sure if it will work in VS2015, but give it a try.
>>>
>>>
>>>
>>> /Peter
>>>
>>>
>>>
>>> 2016-03-06 2:06 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>
>>> Hi!
>>>
>>>
>>>
>>> I want to build Freeswitch on windows with visual studio 2015.
>>>
>>>
>>>
>>> Where should I start if I get 600 errors when try to Rebuild All. I
>>> opened solution and start Rebuild All, but I get so many errors that I
>>> belive that I am doing something wrong.
>>>
>>>
>>>
>>> Mainl yre errors regarding:
>>>
>>>
>>>
>>> Cannot open source file....
>>>
>>>
>>>
>>>
>>>
>>> Best regards, Gregor
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Gregor Nanger*
>>>
>>>
>>>
>>> *CTO*
>>> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
>>> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
>>> ? www.infomedia.si
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>
>>> --
>>>
>>> *Gregor Nanger*
>>>
>>>
>>>
>>> *CTO*
>>> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
>>> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
>>> ? www.infomedia.si
>>>
>>>
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>>
>>> Professional FreeSWITCH Consulting Services:
>>>
>>> consulting at freeswitch.org
>>>
>>> http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>> Official FreeSWITCH Sites
>>>
>>> http://www.freeswitch.org
>>>
>>> http://confluence.freeswitch.org
>>>
>>> http://www.cluecon.com
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Gregor Nanger
>>
>> *CTO*
>> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
>> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
>> ? www.infomedia.si
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
? www.infomedia.si
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From abaci64 at gmail.com  Wed Apr  6 02:43:59 2016
From: abaci64 at gmail.com (Abaci B)
Date: Tue, 5 Apr 2016 18:43:59 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <CAEwTsAi_008hpvESZk2wz19K4NqYJ9pnab21dvZ5zr6c+1w9gQ@mail.gmail.com>
References: <CAFmZFiP63rmSHjoMT=9=pST0imY2mOxJ=j6EwTkdUJsii47jSw@mail.gmail.com>
	<CAO3obxbD0Fu-Ou2AP6UDafuc5qLS72Ma0U8rtB3QAHzRgYBwxw@mail.gmail.com>
	<3025F31C-1FCC-4299-A375-97141EF1861E@440hz.fr>
	<CAO3obxZ3yzsmyRFhGQzSdiZ06sQ3H8cuMQvodeX-0YFQgDATmQ@mail.gmail.com>
	<CAKbxfG_ATKoaJ+5UnmXaxLq9f9Tp=yCwXBiag7tNRgqNFgjkAw@mail.gmail.com>
	<CAO3obxZk3d5h3-0ayQ9RFZ--vCBURuUkj5fTs4i-m4xxAUiUwA@mail.gmail.com>
	<CAFmZFiO3qUezw3oRvqMpr41e6rM5Lw8c4bQFmYjU1oCrrhGZ-g@mail.gmail.com>
	<CAO3obxZcRwfyr2zxVk7Co9SV9Z3VJap-TdBXOHJ0TkRwPiq2nA@mail.gmail.com>
	<CAO3obxbQP=Q26JBeKFpc5CE5uJ=vp-GpeXVHpqFkwGt+3K7NvA@mail.gmail.com>
	<CAFmZFiPbU04_7vvhJnP_Aeu5pYtep9-NR5etNdKk2Tj_3m7sdQ@mail.gmail.com>
	<CAEwTsAi_008hpvESZk2wz19K4NqYJ9pnab21dvZ5zr6c+1w9gQ@mail.gmail.com>
Message-ID: <CAFmZFiPv7sbEtHECKWEn8X93Cs6Q7E7oOTmoo+gpS+nKL=OFNg@mail.gmail.com>

Just tested mod_avmd on master and mod_avmd was able to detect the beep of
a Verizon voicemail. Will do more testing and report back.

On Wed, Mar 30, 2016 at 8:39 PM, Michael Jerris <mike at jerris.com> wrote:

> Just in master so far. It will hit 1.6 branch when we do the next release
>
>
> On Wednesday, March 30, 2016, Abaci B <abaci64 at gmail.com> wrote:
>
>> was it merged into 1.6 or just master?
>>
>> On Wed, Mar 30, 2016 at 11:09 AM, Piotr Gregor <piotrek.gregor at gmail.com>
>> wrote:
>>
>>> Hi Abaci,
>>>
>>> after FS-8875 was merged yesterday in PR 775
>>> I have tested avmd on your Verizon voicemail (contains 1000hz beep)
>>> and on another Voipfone voicemail (contains beep of lower frequency,
>>> 724.1914 is detected).
>>> Detection of tone works just fine while not giving many false positives
>>> in the same time.
>>> But you should note the avmd module is in beta stage so not everything
>>> is working as expected yet. In case of Verizon sometimes it gives false
>>> alarm on silence, this can be handled at this moment by adjusting avmd
>>> session parameters
>>>
>>> AVMD_SAMLPE_TO_SKIP_N  increase
>>> BEEP_TIME increase
>>> SAMPLES_CONSECUTIVE_STREAK increase
>>> VARIANCE_THRESHOLD may need to decrease
>>>
>>> or starting avmd after 3 - 4 first ring tones. Regardless of this it
>>> will be
>>> treated in more generic way - by adding of estimation of amplitude.
>>> Related to this is a need for adjusting parameters on a basis of expected
>>> characteristics of incoming signal, e.g. SNR. And it should be possible
>>> to be set the settings on a per call basis.
>>>
>>> So you may wish to checkout master branch,
>>> build, test, and let us know the results.
>>>
>>> cheers,
>>> Piotr
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Apr  6 03:08:20 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 5 Apr 2016 19:08:20 -0400
Subject: [Freeswitch-users] Windows build
In-Reply-To: <CAC7ttFErJO-gTLBQTHy_6zCcRWBxOYgsXtJXqMC0U8Tv+79AOg@mail.gmail.com>
References: <CAC7ttFFYf+dt3NfgRQy0R5PbNut0=-=mmxv6+gTvUYFB1RsKBA@mail.gmail.com>
	<CAPEdYThObr1ha7ffrFNijAj3H30Z6Zk41TETEgdoeZWxLu6igg@mail.gmail.com>
	<CAC7ttFE98byMZX+uNYUcLJ-A8+WnQwcwXBqqPO-g2fBu3XRfvA@mail.gmail.com>
	<CAPEdYTghm8J20E5COERvG1TidbYNUqcVsjD1TngTyH-zwWE52w@mail.gmail.com>
	<CAC7ttFGRJQAFn0vC3sCC1tguvWmZ==BpKR4i-=UWjDPRgGx-hA@mail.gmail.com>
	<9589f6e5d757412ba5d620cf8e2bf26c@imladris.sermotec.local>
	<56E10D5D.1000303@zg.t-com.hr>
	<38494249596647c6a08591eca41e3000@imladris.sermotec.local>
	<CAC7ttFFbv9YAxXNg7zjc1B9cxLS74v6f6HfBpSt09sULor6EoA@mail.gmail.com>
	<CAHtxdDdY2JRqVuw-W0b+UF0Z_ZUGbCj_iUFUOXi=nrxDDzKb9A@mail.gmail.com>
	<CAC7ttFErJO-gTLBQTHy_6zCcRWBxOYgsXtJXqMC0U8Tv+79AOg@mail.gmail.com>
Message-ID: <3ED4D03E-7C8C-4DE6-9DC1-1C6E039C5355@jerris.com>

I just built windows fine yesterday.  It may be unhappy with rebuild, but I built fine. (this should cover master).... mod_avmd might have been failing for a few commits, but I pushed the fix for that module yesterday, all the others in default configuration built fine.

> On Apr 5, 2016, at 6:38 PM, Gregor Nanger <gregor at infomedia.si> wrote:
> 
> Just bumping this post if someone has any hints.
> 
> I am trying to build Freeswitch with Visual studio 2015, but keep getting error:
> Error MSB4057 The target "v8:Rebuild" does not exist in the project. [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8? ?C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1?"
> 
> Would realy like to successfuly build Freeswitch in windows. I tried with 1.6 branch.
> 
> Best regards, Gregor
> 
> 2016-03-17 5:43 GMT+01:00 Sergey Safarov <s.safarov at gmail.com <mailto:s.safarov at gmail.com>>:
> One week ago I successfully compiled mod_V8 on CentOS 7.
> May be switch to Linux?
> 
> On Thu, Mar 17, 2016 at 12:49 AM, Gregor Nanger <gregor at infomedia.si <mailto:gregor at infomedia.si>> wrote:
> Thank you Harald.
> 
> I tried with latest branch 1.7 and x64 and got errors regarding mod_V8. I tried what Peter suggested, but I am more in c# and web projects and do not have experience in building  C++ projects. There is already prebuilt setup on freeswitch site, so someone successfully build it :-)) It is not so important for us at this point to make own build, so will try again later.
> 
> 
> 2016-03-11 7:32 GMT+01:00 Harald Petrovitsch <Harald.Petrovitsch at sermotec.at <mailto:Harald.Petrovitsch at sermotec.at>>:
> Hi Gregor,
> 
>  
> 
> I only do a
> 
> Git.exe clone ?bv1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git <https://freeswitch.org/stash/scm/fs/freeswitch.git> .
> 
>  
> 
> Loaded the solution into vs2015, set configuration Win32 / Release and press f7 (need to do it two times)
> 
>  
> 
> The build ended with
> 
> ========== Build: 20 succeeded, 0 failed, 157 up-to-date, 15 skipped ==========
> 
>  
> 
> I?ve attached a list of the generated mod folder
> 
>  
> 
> Regards
> 
> Harald
> 
>  
> 
> Von: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Shishko
> Gesendet: Donnerstag, 10. M?rz 2016 07:00
> An: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>
> 
> Betreff: Re: [Freeswitch-users] Windows build
> 
>  
> 
> Hi Harald,
> 
> what did you do to build libv8 and mod_v8? I tried with VS2015 Update 1, branch 1.6, but to no avail.
> 
> Thanks
> 
> 
> On 03/07/2016 08:20 AM, Harald Petrovitsch wrote:
> 
> Hi Gregor !
> 
>  
> 
> V8 libs and mod builds fine here (visual Studio 2015 Sp1, 1.6 branch, used tortoiseGit to download it))
> 
>  
> 
> Regards
> 
> Harald
> 
>  
> 
> Von: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Gregor Nanger
> Gesendet: Montag, 07. M?rz 2016 00:19
> An: FreeSWITCH Users Help
> Betreff: Re: [Freeswitch-users] Windows build
> 
>  
> 
> Thank you, H
> 
> ?a?
> 
> rald. This works: "AFAIK, for the ?'lame/lame.h'? you have to change the include line to  only ?lame.h?"
> 
> B
> 
> ?ut for v8 stil do  not have solution. I do not want to exclude mod_v8, since this module runs javascript. But, can you please confirm me that is not yet compatible, to stop trying to solve it.
> 
> 
> Any other suggestion what does this mean:
> "
> Error MSB4057 The target "v8:Rebuild" does not exist in the project. [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
> 
> ? ?
> 
> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
> 
> ?"
> 
>  
> 
> Best regards, Gregor?
> 
> 
> 
>  
> 
> 2016-03-06 16:39 GMT+01:00 Peter Olsson <peter at olssononline.se <mailto:peter at olssononline.se>>:
> 
> Remove mod_v8 from the build. I don't think it's compatible with VS2015 for now. However, all other modules should be ok.
> 
>  
> 
> /Peter
> 
>  
> 
> 2016-03-06 12:31 GMT+01:00 Gregor Nanger <gregor at infomedia.si <mailto:gregor at infomedia.si>>:
> 
> ?This helped a lot, thank you. Now I have only few errors. Any hint?
> 
>  
> 
> Error MSB4057 The target "v8:Rebuild" does not exist in the project. [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8 C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
> 
> Error MSB4057 The target "v8:Rebuild" does not exist in the project. [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8 C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
> 
> Error C1083 Cannot open include file: 'lame/lame.h': No such file or directory mod_shout C:\Git\freeswitch\src\mod\formats\mod_shout\mod_shout.c 38
> 
> Error LNK1181 cannot open input file 'icui18n.lib' mod_v8 C:\Git\freeswitch\src\mod\languages\mod_v8\LINK 1
> 
>  
> 
> ?
> 
> 
> 
>  
> 
> 2016-03-06 7:53 GMT+01:00 Peter Olsson <peter at olssononline.se <mailto:peter at olssononline.se>>:
> 
> One common mistake is that you allow Git to modify line endings. Make sure autocrlf is turned off - then clone the repository again from scratch.
> 
>  
> 
> Also, I'm not sure if it will work in VS2015, but give it a try.
> 
>  
> 
> /Peter
> 
>  
> 
> 2016-03-06 2:06 GMT+01:00 Gregor Nanger <gregor at infomedia.si <mailto:gregor at infomedia.si>>:
> 
> Hi!
> 
>  
> 
> I want to build Freeswitch on windows with visual studio 2015. 
> 
>  
> 
> Where should I start if I get 600 errors when try to Rebuild All. I opened solution and start Rebuild All, but I get so many errors that I belive that I am doing something wrong.
> 
>  
> 
> Mainl yre errors regarding:
> 
>  
> 
> Cannot open source file....
> 
>  
> 
>  
> 
> Best regards, Gregor
> 
> 
> 
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
>  
> 
> --
> 
> Gregor Nanger
> 
>  
> 
> CTO
> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 
> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia 
> ? www.infomedia.si <http://www.infomedia.si/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
>  
> 
> --
> 
> Gregor Nanger
> 
>  
> 
> CTO
> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 
> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia 
> ? www.infomedia.si <http://www.infomedia.si/>
>  
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
>  
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Gregor Nanger
>  
> CTO
> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 
> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia 
> ? www.infomedia.si <http://www.infomedia.si/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Gregor Nanger
>  
> CTO
> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485 
> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia 
> ? www.infomedia.si <http://www.infomedia.si/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From amani.mansour2 at gmail.com  Wed Apr  6 03:50:13 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Wed, 6 Apr 2016 00:50:13 +0100
Subject: [Freeswitch-users] comand convert
Message-ID: <CAFETSWgZWFw1Nm6bw=1Zq-c35gu_EJ4uPsNt4eCcfode2paxqA@mail.gmail.com>

Hi ,
i need to convert this comand shell to python :


sudo tshark -n -r $pcap -Y "rtp && rtp.ssrc == $ssrc" -T fields -e
rtp.payload | sed "s/:/ /g" | perl -ne 's/([0-9a-f]{2})/print chr hex
$1/gie' >>  $pcap.$ssrc.raw

i do this :
p2 = sub.Popen(('sudo', 'tshark','-n','-r',pcap,'-Y','rtp and
rtp.ssrc=='+pa[n],'-T','fields','-e', 'rtp.payload','|','
sed','s','/',':','/',' /','g' ,'|',' perl',' -ne',
's','/,','([0-9a-f]{2}),','/','print',' chr',' hex',law,'/gie','
>>',pcap+'.'+pa[n]+'.raw'),stdout=sub.PIPE)


is there some one who can help me please ?
 best regards
amani
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From quanhs at stee.stengg.com  Wed Apr  6 07:35:39 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Wed, 6 Apr 2016 03:35:39 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION	ERROR. SDP
In-Reply-To: <cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
Message-ID: <c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>

Hi Itola;

Sorry, same error.

Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?

Finding when FS runs in media mode:
codec causes caller side ?488 not acceptable here| incompatible destination ?
                callee side: ?cancel |user not registered?

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Set media_mix_inbound_outbound_codecs=true

?talo Rossi
italo at freeswitch.org<mailto:italo at freeswitch.org>
IRC chat.freenode.net #freeswitch #freeswitch-dev
Bugs? https://freeswitch.org/jira
Docs? https://freeswitch.org/jira
Chat? https://hipchat.freeswitch.org/gUdAgy0m6



On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

Hi All;



I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min),just following the information in book ?FreeSWITCH 1.6 Cookbook?.

If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.

But if disabling media bypass mode, call is rejected by FS.



FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.

Chrome uses opus 111, FS uses opus 116.

Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().



Help is needed to troubleshoot this issue.



Thanks advance.

Smile.



[This e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
from your system; please do not copy or use it for any purpose, nor disclose
its contents to any other person. Thank you.]
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From gregor at infomedia.si  Wed Apr  6 10:49:21 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Wed, 6 Apr 2016 08:49:21 +0200
Subject: [Freeswitch-users] Windows build
In-Reply-To: <3ED4D03E-7C8C-4DE6-9DC1-1C6E039C5355@jerris.com>
References: <CAC7ttFFYf+dt3NfgRQy0R5PbNut0=-=mmxv6+gTvUYFB1RsKBA@mail.gmail.com>
	<CAPEdYThObr1ha7ffrFNijAj3H30Z6Zk41TETEgdoeZWxLu6igg@mail.gmail.com>
	<CAC7ttFE98byMZX+uNYUcLJ-A8+WnQwcwXBqqPO-g2fBu3XRfvA@mail.gmail.com>
	<CAPEdYTghm8J20E5COERvG1TidbYNUqcVsjD1TngTyH-zwWE52w@mail.gmail.com>
	<CAC7ttFGRJQAFn0vC3sCC1tguvWmZ==BpKR4i-=UWjDPRgGx-hA@mail.gmail.com>
	<9589f6e5d757412ba5d620cf8e2bf26c@imladris.sermotec.local>
	<56E10D5D.1000303@zg.t-com.hr>
	<38494249596647c6a08591eca41e3000@imladris.sermotec.local>
	<CAC7ttFFbv9YAxXNg7zjc1B9cxLS74v6f6HfBpSt09sULor6EoA@mail.gmail.com>
	<CAHtxdDdY2JRqVuw-W0b+UF0Z_ZUGbCj_iUFUOXi=nrxDDzKb9A@mail.gmail.com>
	<CAC7ttFErJO-gTLBQTHy_6zCcRWBxOYgsXtJXqMC0U8Tv+79AOg@mail.gmail.com>
	<3ED4D03E-7C8C-4DE6-9DC1-1C6E039C5355@jerris.com>
Message-ID: <CAC7ttFFfGFD+mzKRZWvwNy3BYoKK7FXY1pBgPGUMbj79qYeMmA@mail.gmail.com>

Thank you Michael!

Maybe it is something with my configuration of VS...

Can you copy your build also on
http://files.freeswitch.org/windows/installer/x64/?

2016-04-06 1:08 GMT+02:00 Michael Jerris <mike at jerris.com>:

> I just built windows fine yesterday.  It may be unhappy with rebuild, but
> I built fine. (this should cover master).... mod_avmd might have been
> failing for a few commits, but I pushed the fix for that module yesterday,
> all the others in default configuration built fine.
>
> On Apr 5, 2016, at 6:38 PM, Gregor Nanger <gregor at infomedia.si> wrote:
>
> Just bumping this post if someone has any hints.
>
> I am trying to build Freeswitch with Visual studio 2015, but keep getting
> error:
> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
> ? ?
> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
> ?"
>
>
> Would realy like to successfuly build Freeswitch in windows. I tried with
> 1.6 branch.
>
> Best regards, Gregor
>
> 2016-03-17 5:43 GMT+01:00 Sergey Safarov <s.safarov at gmail.com>:
>
>> One week ago I successfully compiled mod_V8 on CentOS 7.
>> May be switch to Linux?
>>
>> On Thu, Mar 17, 2016 at 12:49 AM, Gregor Nanger <gregor at infomedia.si>
>> wrote:
>>
>>> Thank you Harald.
>>>
>>> I tried with latest branch 1.7 and x64 and got errors regarding mod_V8.
>>> I tried what Peter suggested, but I am more in c# and web projects and do
>>> not have experience in building  C++ projects. There is already prebuilt
>>> setup on freeswitch site, so someone successfully build it :-)) It is not
>>> so important for us at this point to make own build, so will try again
>>> later.
>>>
>>>
>>> 2016-03-11 7:32 GMT+01:00 Harald Petrovitsch <
>>> Harald.Petrovitsch at sermotec.at>:
>>>
>>>> Hi Gregor,
>>>>
>>>>
>>>>
>>>> I only do a
>>>>
>>>> Git.exe clone ?bv1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git
>>>> .
>>>>
>>>>
>>>>
>>>> Loaded the solution into vs2015, set configuration Win32 / Release and
>>>> press f7 (need to do it two times)
>>>>
>>>>
>>>>
>>>> The build ended with
>>>>
>>>> ========== Build: 20 succeeded, 0 failed, 157 up-to-date, 15 skipped
>>>> ==========
>>>>
>>>>
>>>>
>>>> I?ve attached a list of the generated mod folder
>>>>
>>>>
>>>>
>>>> Regards
>>>>
>>>> Harald
>>>>
>>>>
>>>>
>>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Shishko
>>>> *Gesendet:* Donnerstag, 10. M?rz 2016 07:00
>>>> *An:* freeswitch-users at lists.freeswitch.org
>>>>
>>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>>
>>>>
>>>>
>>>> Hi Harald,
>>>>
>>>> what did you do to build libv8 and mod_v8? I tried with VS2015 Update
>>>> 1, branch 1.6, but to no avail.
>>>>
>>>> Thanks
>>>>
>>>> On 03/07/2016 08:20 AM, Harald Petrovitsch wrote:
>>>>
>>>> Hi Gregor !
>>>>
>>>>
>>>>
>>>> V8 libs and mod builds fine here (visual Studio 2015 Sp1, 1.6 branch,
>>>> used tortoiseGit to download it))
>>>>
>>>>
>>>>
>>>> Regards
>>>>
>>>> Harald
>>>>
>>>>
>>>>
>>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [
>>>> mailto:freeswitch-users-bounces at lists.freeswitch.org
>>>> <freeswitch-users-bounces at lists.freeswitch.org>] *Im Auftrag v**on *Gregor
>>>> Nanger
>>>> *Gesendet:* Montag, 07. M?rz 2016 00:19
>>>> *An:* FreeSWITCH Users Help
>>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>>
>>>>
>>>>
>>>> Thank you, H
>>>>
>>>> ?a?
>>>>
>>>> rald. This works: "AFAIK, for the ?'lame/lame.h'? you have to change
>>>> the include line to  only ?lame.h?"
>>>>
>>>> B
>>>>
>>>> ?ut for v8 stil do  not have solution. I do not want to exclude
>>>> mod_v8, since this module runs javascript. But, can you please confirm me
>>>> that is not yet compatible, to stop trying to solve it.
>>>>
>>>>
>>>> Any other suggestion what does this mean:
>>>> "
>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>>
>>>> ? ?
>>>>
>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>
>>>> ?"
>>>>
>>>>
>>>>
>>>> Best regards, Gregor?
>>>>
>>>>
>>>>
>>>> 2016-03-06 16:39 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>>
>>>> Remove mod_v8 from the build. I don't think it's compatible with VS2015
>>>> for now. However, all other modules should be ok.
>>>>
>>>>
>>>>
>>>> /Peter
>>>>
>>>>
>>>>
>>>> 2016-03-06 12:31 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>>
>>>> ?This helped a lot, thank you. Now I have only few errors. Any hint?
>>>>
>>>>
>>>>
>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln]
>>>> libv8 C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>
>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>
>>>> Error C1083 Cannot open include file: 'lame/lame.h': No such file or
>>>> directory mod_shout C:\Git\freeswitch\src\mod\formats\mod_shout\mod_shout.c
>>>> 38
>>>>
>>>> Error LNK1181 cannot open input file 'icui18n.lib' mod_v8
>>>> C:\Git\freeswitch\src\mod\languages\mod_v8\LINK 1
>>>>
>>>>
>>>>
>>>> ?
>>>>
>>>>
>>>>
>>>> 2016-03-06 7:53 GMT+01:00 Peter Olsson <peter at olssononline.se>:
>>>>
>>>> One common mistake is that you allow Git to modify line endings. Make
>>>> sure autocrlf is turned off - then clone the repository again from scratch.
>>>>
>>>>
>>>>
>>>> Also, I'm not sure if it will work in VS2015, but give it a try.
>>>>
>>>>
>>>>
>>>> /Peter
>>>>
>>>>
>>>>
>>>> 2016-03-06 2:06 GMT+01:00 Gregor Nanger <gregor at infomedia.si>:
>>>>
>>>> Hi!
>>>>
>>>>
>>>>
>>>> I want to build Freeswitch on windows with visual studio 2015.
>>>>
>>>>
>>>>
>>>> Where should I start if I get 600 errors when try to Rebuild All. I
>>>> opened solution and start Rebuild All, but I get so many errors that I
>>>> belive that I am doing something wrong.
>>>>
>>>>
>>>>
>>>> Mainl yre errors regarding:
>>>>
>>>>
>>>>
>>>> Cannot open source file....
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Best regards, Gregor
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Gregor Nanger*
>>>>
>>>>
>>>>
>>>> *CTO*
>>>> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
>>>> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
>>>> ? www.infomedia.si
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>>
>>>> *Gregor Nanger*
>>>>
>>>>
>>>>
>>>> *CTO*
>>>> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
>>>> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
>>>> ? www.infomedia.si
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> Professional FreeSWITCH Consulting Services:
>>>>
>>>> consulting at freeswitch.org
>>>>
>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>> Official FreeSWITCH Sites
>>>>
>>>> http://www.freeswitch.org
>>>>
>>>> http://confluence.freeswitch.org
>>>>
>>>> http://www.cluecon.com
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Gregor Nanger
>>>
>>> *CTO*
>>> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
>>> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
>>> ? www.infomedia.si
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
> ? www.infomedia.si
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rutu.patel at inextrix.com  Wed Apr  6 11:35:23 2016
From: rutu.patel at inextrix.com (Rutu Patel)
Date: Wed, 6 Apr 2016 13:05:23 +0530
Subject: [Freeswitch-users] 480 Temporarily
	Unavailable[MANDATORY_IE_MISSING]
In-Reply-To: <CAPQet1ye8rw18g16DiLyNNSCuBbNd0X=yn5YuT7t2ZdJ-N65hw@mail.gmail.com>
References: <CAPQet1zRj3sogx3csggUCe9zkeCLN2X=99fr=z73WkLvD5=bPA@mail.gmail.com>
	<CAOrA2UYZTKucCxUC3XbB=yG_Z9ZOTOZpRH_MvrgLHhqoHnVc8g@mail.gmail.com>
	<CAKbxfG_BacH6FQqEPba7w6UTHbKhQsw+7UX9=pCgvr7mo1t=XA@mail.gmail.com>
	<CAPQet1zPPUVHSpNyUVJVvj_df7qfq+ty_yYatuBoTPboaQ0F+Q@mail.gmail.com>
	<CAKzWOxXUCNjRXGy=UPDNSQyZUx8i5eJ-iw9Xs4gYKdJR5RA3aA@mail.gmail.com>
	<CAPQet1wsOwEC3hWFqKns0-0gTOkMEs94B6D8mxp8HwyGsWqPHg@mail.gmail.com>
	<CAOrA2Ub=NFpdsQZxsDX2BhmoX-8DLtMFHBBQeJXQtrKSeq6m8A@mail.gmail.com>
	<CAPQet1z+=0rw=VO9Q-b95n4B_Bk-R6VEZvrM5VE5sMd-u4dTgQ@mail.gmail.com>
	<CAOrA2UZEYXX_bdThsjcqjdUXqtR9oHMjkr2k-FT0WAU-O9Y=9g@mail.gmail.com>
	<CAPQet1ye8rw18g16DiLyNNSCuBbNd0X=yn5YuT7t2ZdJ-N65hw@mail.gmail.com>
Message-ID: <CAPQet1zRi9yZQ7eYaLxGtSUXCHNo9R_KtmHq74OX-OhuXE_unw@mail.gmail.com>

Can anyone please help ?

--
Thanks,
Rutu Patel

<http://www.inextrix.com>

On Tue, Apr 5, 2016 at 10:46 AM, Rutu Patel <rutu.patel at inextrix.com> wrote:

> Hi,
>
> Thank you for your help to debug the issue.
>
> We are now trying without opensips but still facing authentication issue.
>
> We created two users of freeswitch in asterisk as peer.
>
> For example, First peer in sip.conf is 456789 and second is 123456.
>
> [123456]
> type=friend
> host=x.x.x.45
> username=123456
> secret=abcdef
> disallow=all
> allow=g729
> trustrpid=yes
> canreinvite=yes
> sendrpid=yes
> context=extensions-test
> qualify=yes
> directmedia=yes
>
> Now when we are trying to reach 123456, the request reach to asterisk but
> not authenticate proper user(123456). It authenticate first peer(456789)
> defined in sip.conf.
>
> If we define 123456 before 456789 then it works but we have multiple peers
> in asterisk so how can we manage that ?
>
> We also tried by sending user credential direct in bridge,but still it is
> authenticate the first peer defined in sip.conf .
>
>
> {sip_auth_username=123456,sip_auth_password=abcdef,sip_contact_user=123456}sofia/default/NUMBER at IP
>
> ?Can anyone please explain how to authenticate specific peer ?
>
> Please let me know if you want any other details.?
>
> --
> Thanks,
> Rutu Patel
> ??
>
> <http://www.inextrix.com>
>
> On Fri, Apr 1, 2016 at 3:56 PM, Jurijs Ivolga <jurijs.ivolga at gmail.com>
> wrote:
>
>> Hi,
>>
>> I'm not sure about your environment, but on this point I will recommend
>> you to isolate issue. So on this point you should understand if your sip
>> trunk between Asterisk and Freeswitch configured correctly, without
>> involving Opensips and etc. To test SIP trunk you can try to send call
>> directly from freeswitch to Asterisk and see if it works, you can send call
>> just directly from freeswitch console, something similar to:
>>
>> originate sofia/example/300 at foo.com 8600
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/mod_commands
>>
>> If you call will reach Asterisk without any issue, then this means that
>> sip trunk created correctly and issue is somewhere on Opensips side or any
>> other part of your set-up.
>>
>> Second option will be to try to configure sip trunk without password and
>> check if it helps
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
>>
>> With kind regards,
>>
>> Jurijs
>>
>> On Fri, Apr 1, 2016 at 12:54 PM, Rutu Patel <rutu.patel at inextrix.com>
>> wrote:
>>
>>> Hi Juris,
>>>
>>> Yes we added peer in asterisk sip.conf.
>>>
>>> Are we still missing any configurations?
>>>
>>>
>>> On Thursday, March 31, 2016, Jurijs Ivolga <jurijs.ivolga at gmail.com>
>>> wrote:
>>>
>>>> Hi Rutu,
>>>>
>>>> Just a shot in a dark. :)
>>>>
>>>> Did you added sip peer record in sip.conf, something similar to:
>>>>
>>>> [freeswitch]
>>>> type=peer
>>>> host=IP_ADDRESS_OF_FREESWITCH_SERVER
>>>> username=HOSTNAME.DOMAIN.COM
>>>> port=5080
>>>> fromdomain=IP_ADDRESS_OF_FREESWITCH_SERVER
>>>> secret=BOOTH_WAY_PASSWORD
>>>>
>>>> With kind regards,
>>>>
>>>>
>>>> Jurijs
>>>>
>>>> On Thu, Mar 31, 2016 at 5:19 PM, Rutu Patel <rutu.patel at inextrix.com>
>>>> wrote:
>>>>
>>>>> Hi Michael
>>>>>
>>>>> Thank you for your assistance.
>>>>>
>>>>> We are trying to make call between two freeswitch users.
>>>>>
>>>>> One user is registered in softphone and other is registered in
>>>>> Asterisk by register string but asterisk user cannot authenticate properly
>>>>> and freeswitch directly send 480 Temporary Unavailable.
>>>>>
>>>>> In sip profile we set challenge-realm= auto_to also tried with
>>>>> auto_from but still not getting success.
>>>>>
>>>>> I am attaching some more logs if anyone can guide in right direction.
>>>>>
>>>>> --
>>>>> Thanks,
>>>>> Rutu Patel
>>>>>
>>>>>
>>>>>
>>>>> On Mon, Mar 28, 2016 at 10:23 PM, Michael Collins <msc at freeswitch.org>
>>>>> wrote:
>>>>>
>>>>>> Try a gateway. Look in conf/sip_profiles/external/example.xml for a
>>>>>> heavily-commented sample. For more detailed discussion see this nice doc:
>>>>>>
>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration
>>>>>>
>>>>>> -MSC
>>>>>>
>>>>>> On Fri, Mar 25, 2016 at 11:34 PM, Rutu Patel <rutu.patel at inextrix.com
>>>>>> > wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> Thank you for reply.
>>>>>>>
>>>>>>> We have registered extension of Freeswitch in Asterisk as below
>>>>>>>
>>>>>>> register=>test-ext:secret at 192.168.1.64/test-ext
>>>>>>>
>>>>>>> ?We want to send call from Freeswitch to Asterisk?.
>>>>>>>
>>>>>>> What authentication parameters we need to set in Freeswitch ?
>>>>>>>
>>>>>>> --
>>>>>>> Thanks,
>>>>>>> Rutu Patel
>>>>>>> On Wed, Mar 23, 2016 at 8:46 PM, Anthony Minessale <
>>>>>>> anthony.minessale at gmail.com> wrote:
>>>>>>>
>>>>>>>> You need to supply auth credentials to Asterisk because its set to
>>>>>>>> authenticate calls.
>>>>>>>> The best thing to do would be to make a gateway pointing to
>>>>>>>> Asterisk with all the credentials and call it using that.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Wed, Mar 23, 2016 at 10:20 AM, Jurijs Ivolga <
>>>>>>>> jurijs.ivolga at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Hi Rutu,
>>>>>>>>>
>>>>>>>>> In logs I can see that user-agent Asterisk is sending INVITE to
>>>>>>>>> Freeswitch and Freeswitch replies with "480 Temporarily Unavailable". This
>>>>>>>>> a bit differs from what you explained in your mail...
>>>>>>>>>
>>>>>>>>> First think what pop up in my mind is that you need to configure
>>>>>>>>> Asterisk as a trunk on Freeswitch, not sure if this helps, but this is my
>>>>>>>>> first guess...
>>>>>>>>>
>>>>>>>>> https://freeswitch.org/confluence/display/FREESWITCH/Asterisk
>>>>>>>>>
>>>>>>>>> With kind regards,
>>>>>>>>>
>>>>>>>>> Jurijs
>>>>>>>>>
>>>>>>>>> On Wed, Mar 23, 2016 at 3:46 PM, Rutu Patel <
>>>>>>>>> rutu.patel at inextrix.com> wrote:
>>>>>>>>>
>>>>>>>>>> Hi,
>>>>>>>>>>
>>>>>>>>>> We are sending call to asterisk PBX but we are getting 480
>>>>>>>>>> Temporarily Unavailable immediately. from freeswitch server.
>>>>>>>>>>
>>>>>>>>>> Call flow is as below:
>>>>>>>>>>
>>>>>>>>>> User => opensips => FS => PBX,
>>>>>>>>>>
>>>>>>>>>> but call not sending out from FS and drop by FS.
>>>>>>>>>>
>>>>>>>>>> Here I am attaching both sip logs and freeswitch logs.
>>>>>>>>>>
>>>>>>>>>> Can anyone please help to sort out the issue.
>>>>>>>>>> --
>>>>>>>>>> Thanks,
>>>>>>>>>> Rutu Patel
>>>>>>>>>> <http://www.inextrix.com>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>>>>>>>
>>>>>>>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>>>>>>>> http://twitter.com/FreeSWITCH
>>>>>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>>>>>>>> <http://freeswitch.org/g+>*
>>>>>>>>
>>>>>>>> ClueCon Weekly Development Call
>>>>>>>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>>>>>>>
>>>>>>>> https://www.youtube.com/watch?v=9XXgW34t40s
>>>>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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,
>>> Rutu Patel
>>> iNextrix Technologies Pvt. Ltd.
>>> www.inextrix.com
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From steveayre at gmail.com  Wed Apr  6 13:44:20 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 6 Apr 2016 10:44:20 +0100
Subject: [Freeswitch-users] Is MySQL OK to use?
In-Reply-To: <CAKbxfG-yLgyzgq3D0RT+96RcfTSWBi7myX+t0ew37arRSLYkXQ@mail.gmail.com>
References: <528E45C5.8030509@digitalmail.com>
	<CAC-LwPME371eMDigDeZXTbf-WeOLx56La8iE9eyyS-f+xh6FGA@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C91C@IMBER.smb.curriegrad2004.ca>
	<CAFiqYu=F2w51wy4M15ygP_=iHSe5orqRqu6zdnYNND8D6=ix=g@mail.gmail.com>
	<CAKbxfG-yLgyzgq3D0RT+96RcfTSWBi7myX+t0ew37arRSLYkXQ@mail.gmail.com>
Message-ID: <CAFiqYumNo9wQHyZNx5ufuhohE_ivBQJTTrYm7suY4-8UbGzmNQ@mail.gmail.com>

I've been doing some debugging on this this week as we're currently seeing
regular crashes on our production servers. This happens with both
Threading=0 and Threading=2.

It appears it's a bug in MyODBC triggered because FreeSWITCH uses a new
ODBC environment handle every time it creates a connection.

When a environment handle is created myodbc_init is called and when it's
freed it calls myodbc_end. Those functions are not thread-safe. They're
protected by a reference counter but they aren't wrapped in a mutex so if
the server is busy and they're both called at the same time you can result
in some global variables being used after they're freed.

Rebuilding MyODBC with an empty myodbc_end appears to stop the crashes, but
I'm not saying that's a good solution. Still looking at it.

On 22 November 2013 at 20:31, Anthony Minessale <anthony.minessale at gmail.com
> wrote:

> I've always seen Threading=0 as the solution to stability probs.
> Basically if the driver already uses mutexing, its better to disable the
> arbitrary ones in the core of odbc.
> For postgres 0 is basically mandatory.
>
>
>
> On Fri, Nov 22, 2013 at 2:12 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> No such thing in the latest versions, they're all threadsafe now (_r is a
>> symlink to the other).
>>
>> I use it with success. I did find stability problems in libmyodbc when
>> upgrading from 5.0 to 5.5. My solution was to add these to odbc.ini
>>
>> Option      = 67108864
>> Threading   = 2
>>
>> The Threading setting is what stopped the crashes. There's a Jira on the
>> subject and someone else said manually upgrading to a newer version of
>> unixodbc+myodbc also fixed it.
>>
>> -Steve
>>
>>
>>
>> On 21 November 2013 18:23, Jeff Leung <jleung at v10networks.ca> wrote:
>>
>>> Try the non-thread safe MySQL library instead of the thread safe one.
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Vik Killa
>>> *Sent:* Thursday, November 21, 2013 10:22 AM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] Is MySQL OK to use?
>>>
>>>
>>>
>>> short answer no.
>>>
>>> use postgres
>>>
>>>
>>>
>>> On Thu, Nov 21, 2013 at 12:41 PM, Alex Lake <alex at digitalmail.com>
>>> wrote:
>>>
>>> I find these "Error in my_thread_global_end()" messages somewhat
>>> annoying in my fs1.2stable on Ubuntu12.04 box. Is there any advice
>>> (other than "don't use MySQL") for how to install it better? Might it be
>>> something to do with thread-safe libraries (I'm using libmyodbc_r.so at
>>> the moment)
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>> http://www.cudatel.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>> http://www.cudatel.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>> http://www.cudatel.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saurabhkv01 at gmail.com  Wed Apr  6 13:01:19 2016
From: saurabhkv01 at gmail.com (saurabh verrma)
Date: Wed, 6 Apr 2016 14:31:19 +0530
Subject: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
In-Reply-To: <CACot2ntmtZzcXs4bWqd3oW9i5URXMpqaouu8EXKWeN2thDJG4A@mail.gmail.com>
References: <CACot2ntTB+7OXvomm3sxMzxnNEnT6vHtQb=T=m8MTOFT0=zFZg@mail.gmail.com>
	<E93C0E00-0EEE-44E1-81F2-1E8CF374057B@jerris.com>
	<CACot2ntmtZzcXs4bWqd3oW9i5URXMpqaouu8EXKWeN2thDJG4A@mail.gmail.com>
Message-ID: <CACot2nvgSJOFj9HJ2JOc4Ofp1Ce_x-mZ0G95epFJA1AwMZ2X5g@mail.gmail.com>

Hi,

I?m working on an application where I?m trying to use FreeSWITCH as a
library. My intention is to use FreeSWITCH as a UAS endpoint. Basically it
needs to be supporting following:
1. WebRTC
2. Ability to act like UAS endpoint
3. Support for DTLS/SRTP
4. ICE support

I?m seeking community suggestion if that?s feasible to implement or not? If
yes, what are the possible starting directions we could explore above
points.

Any help would be greatly appreciated.


On Sat, Apr 2, 2016 at 12:24 PM, saurabh verrma <saurabhkv01 at gmail.com>
wrote:

> Thanks Michael,
>
> Basically we're writing a PJSIP based application & PJSIP doesn't have
> DTLS support. So we're thinking to use FreeSWITCH library for DTLS/SRTP.
>
> On Fri, Apr 1, 2016 at 7:29 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> we have full support for webrtc media profile which would include these
>> features  There are not a ton of people who use freeswitch as a library,
>> but it is built that way so that you can control it and host it in another
>> application instead of stand alone.  If you are trying to accomplish
>> something I'd try to handle it standalone first so you can learn all the
>> different ways you might control it before architecting a solution
>>
>> On Apr 1, 2016, at 9:48 AM, saurabh verrma <saurabhkv01 at gmail.com> wrote:
>>
>> Hi All,
>>
>> I want to use FreeSWITCH library for DTLS/SRTP support. I want to know in
>> freeswitch which library has the support of these features(SRTP/DTLS) ?
>> Is there any application available based on freeswitch library ?
>>
>> Any help would be appreciable.
>>
>> --
>> *With Warm Regards:*
>> *Saurabh Kumar Verma*
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 Warm Regards:*
> *Saurabh Kumar Verma*
>



-- 
*With Warm Regards:*
*Saurabh Kumar Verma*
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From mouli123 at gmail.com  Wed Apr  6 17:10:40 2016
From: mouli123 at gmail.com (Chandramouli P)
Date: Wed, 6 Apr 2016 18:40:40 +0530
Subject: [Freeswitch-users] NATIVE SQL ERR [cannot commit - no transaction
	is active]
Message-ID: <CALR0W68a6+iiimqtW6hyxVdKz8o8B1yg-P7tye0LjEmtajvOaw@mail.gmail.com>

Hi,

Please find my below deployed environment:

Environment: Microsoft Azure
OS: CentOS 7.0 (64 bit)
FreeSwitch Version: 1.6.6~64bit ( 64bit)

I installed FreeSwitch through "Yum" and works fine for couple of hours.
After that I am getting below errors:

[ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [attempt to write a readonly
database]
BEGIN EXCLUSIVE
2016-04-05 13:29:33.368355 [CRIT] switch_core_sqldb.c:1952 ERROR [attempt
to write a readonly database]
2016-04-05 13:29:33.368355 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
[cannot commit - no transaction is active]

Can anybody tell me what could be the issue and how to solve this?

Thanks in advance,
Chandra.
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From stszap at gmail.com  Wed Apr  6 17:16:26 2016
From: stszap at gmail.com (=?UTF-8?B?0KHRgtCw0L3QuNGB0LvQsNCyINCX0LDQv9C+0LvRjNGB0LrQuNC5?=)
Date: Wed, 6 Apr 2016 18:16:26 +0500
Subject: [Freeswitch-users] Codec renegotiation on Re-INVITE withot SDP
Message-ID: <CANtLK7P2JWntbupPU8m1HfC=X9Q8Vop7E=5BabE2oPUfz2SsQg@mail.gmail.com>

Hello. We are facing the following problem with our FS setup:

1. provider sends us INVITE with "a=sendonly" in SDP
2. FS picks up the call, playing ivr and replying with 200 and "a=recvonly"
in SDP
3. provider sends Re-INVITE without SDP
4. FS replying 200 with already negotiated SDP (single codec and
"a=recvonly")

Provider wants renegotiation of sdp on step 4, or a least "a=sendrecv"
parameter. Is there a way to achieve this without source modification? We
tried setting following profile parameters:
renegotiate-codec-on-hold=true
renegotiate-codec-on-reinvite=true
enable-3pcc=true
but no luck. The only solution we found was modifing sofia.c like this:
line 7374 - replace switch_core_media_gen_local_sdp(session,
SDP_TYPE_RESPONSE, NULL, 0, NULL, 0); with
switch_core_media_gen_local_sdp(session, SDP_TYPE_RESPONSE, NULL, 0,
"sendrecv", 0);
line 7376 - replace nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); with
nua_respond(tech_pvt->nh, SIP_200_OK,
SOATAG_USER_SDP_STR(tech_pvt->mparams.local_sdp_str), TAG_END());

Thanks.
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From aubalde at presenceco.com  Wed Apr  6 17:57:48 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Wed, 6 Apr 2016 15:57:48 +0200
Subject: [Freeswitch-users] Error Opening DB
Message-ID: <CAKThY04xWQ1eNQ00QW7PUz1=inB7pYibjUDcqD2QR8e3K2UzNw@mail.gmail.com>

Hello,

In the console Freeswitch I have observed the following error:

2016-04-06 11:35:08.864016 [ERR] switch_core_db.c:108 SQL ERR [unsupported
file format]
2016-04-06 11:35:08.864016 [ERR] switch_core_db.c:223 SQL ERR [unsupported
file format]
2016-04-06 11:35:08.864016 [CRIT] switch_core_sqldb.c:507 Failure to
connect to CORE_DB voicemail_default!
2016-04-06 11:35:08.864016 [ERR] mod_voicemail.c:297 Error Opening DB

Anyone know how to fix it? Is possible to disable the use of internal
databases?


Thanks and regards,
Agust?

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

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From luis.daniel.lucio at gmail.com  Wed Apr  6 18:20:35 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Wed, 6 Apr 2016 10:20:35 -0400
Subject: [Freeswitch-users] Error Opening DB
In-Reply-To: <CAKThY04xWQ1eNQ00QW7PUz1=inB7pYibjUDcqD2QR8e3K2UzNw@mail.gmail.com>
References: <CAKThY04xWQ1eNQ00QW7PUz1=inB7pYibjUDcqD2QR8e3K2UzNw@mail.gmail.com>
Message-ID: <CAFLo2Qym9Td+k_cosnDnZYE4tj6xGJSHOzgkLF_zDE=cN1OWUA@mail.gmail.com>

Have you done a major updates on your system? Otherwise sound like a sqlite
corruption. Just delete files and restart.

You always need a db, internal or using a db such as pgsql or MySQL by odbc
Le 6 avr. 2016 9:58 AM, "Agust? Ubalde" <aubalde at presenceco.com> a ?crit :

> Hello,
>
> In the console Freeswitch I have observed the following error:
>
> 2016-04-06 11:35:08.864016 [ERR] switch_core_db.c:108 SQL ERR [unsupported
> file format]
> 2016-04-06 11:35:08.864016 [ERR] switch_core_db.c:223 SQL ERR [unsupported
> file format]
> 2016-04-06 11:35:08.864016 [CRIT] switch_core_sqldb.c:507 Failure to
> connect to CORE_DB voicemail_default!
> 2016-04-06 11:35:08.864016 [ERR] mod_voicemail.c:297 Error Opening DB
>
> Anyone know how to fix it? Is possible to disable the use of internal
> databases?
>
>
> Thanks and regards,
> Agust?
>
> *PRESENCE TECHNOLOGY*
> *Agust? Ubalde Bellot*
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
>
> Ph: +34 93 10 10 300
> Fx: +34 93 10 10 333
>
> *www.presenceco.com* <http://www.presenceco.com/>
>
> *Follow us on:*
>
> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
> <http://www.youtube.com/PresenceTechnology>*[image: in]*
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> <http://www.slideshare.net/PresenceTech>*[image: fb]*
> <http://www.facebook.com/presencetech>
>
> For additional information, please visit our website *www.presenceco.com*
> <http://www.presenceco.com/>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
>
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From vincent.gire at gmail.com  Wed Apr  6 19:42:20 2016
From: vincent.gire at gmail.com (Vincent Gire)
Date: Wed, 6 Apr 2016 17:42:20 +0200
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <731C9BFA-0101-42F3-B368-17F8AEFD7580@jerris.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
	<CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
	<CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>
	<CAKsJLpa83aWDBMFuZaYeMFVR6XeN=2uQqwcuTXAtqTY3szckSg@mail.gmail.com>
	<E6948933-09AC-4A48-A0C9-70E206EED453@jerris.com>
	<CAKsJLpavsm2N7odd2xw6CP6zuBxhLR_QEy3acEPukujVAPU5HA@mail.gmail.com>
	<CAHtxdDfQGXWAUopsmOUO3hTV2A91i9t=9ss5P3+GtnxPtwwsfg@mail.gmail.com>
	<CAEwTsAid_21ysmiu3E+iUdvazMGpLpb0KOpVCOxK6WaurmnWEA@mail.gmail.com>
	<CAKsJLpYAR333mU6VuD2AjxepbPJ4quyzGKpW2ziOmMLds6rgsg@mail.gmail.com>
	<731C9BFA-0101-42F3-B368-17F8AEFD7580@jerris.com>
Message-ID: <CAKsJLpaCq3=ifM_FFav2-iE9X7f00FRHppiRjO25Yc-=rKeuew@mail.gmail.com>

We already handle the multiple providers on the HTTP server.
So you would recommend to implement a module against the speech interface
streaming to the HTTP server ?
Wouldn't it be easier (and less redundant) to add streaming capabilities to
mod_httpapi ?

On Wed, Apr 6, 2016 at 12:22 AM, Michael Jerris <mike at jerris.com> wrote:

> The best way to handle this is probably to write a custom module against
> the speech interface that sends the streams like you are describing, and
> supports multiple providers.  How exactly do you go about sending to
> multiple at the same time and combining the results, as this might actually
> be the trickiest part as youll need to integrate that into a module that
> uses the speech interface to be able to at all sanely handle that in
> freeswitch
>
> On Apr 5, 2016, at 1:24 PM, Vincent Gire <vincent.gire at gmail.com> wrote:
>
> We use multiple ones to compare confidence.
> They all support chunk transfer and provide a HTTP transaction similar to :
> http://developer.att.com/apis/speech/docs/v3
> Look at 3/ Make API Calls
>
> On Tue, Apr 5, 2016 at 7:14 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Unimrcp isn't going to provide the interface he's talking about.  Can I
>> ask which engine this is?
>>
>>
>> On Tuesday, April 5, 2016, Sergey Safarov <s.safarov at gmail.com> wrote:
>>
>>> Look at http://www.unimrcp.org/ and
>>> https://wiki.freeswitch.org/wiki/Mod_unimrcp
>>>
>>> On Tue, Apr 5, 2016, 19:38 Vincent Gire <vincent.gire at gmail.com> wrote:
>>>
>>>> We are building an IVR completely driven by ASR.
>>>> ASR is performed in a distant location by a HTTP service (supporting
>>>> chunked transfer) and adds an incompressible latency. We would like to
>>>> stream the record to the ASR service as soon as it starts to reduce the
>>>> overall latency before response.
>>>> Does it make sense ?
>>>>
>>>>
>>>> On Tue, Apr 5, 2016 at 5:39 PM, Michael Jerris <mike at jerris.com> wrote:
>>>>
>>>>> I would stay away from mod_vlc.  Its audio portions with recording
>>>>> have known issues.  We do use the rtmp streaming in mod_av heavily but
>>>>> thats obviously not wav.  Can you explain a bit more why you have this
>>>>> requirement?
>>>>>
>>>>> On Apr 5, 2016, at 5:36 AM, Vincent Gire <vincent.gire at gmail.com>
>>>>> wrote:
>>>>>
>>>>> Ok thanks.
>>>>> It looks promising !
>>>>> I'll dig into mod_vlc.
>>>>>
>>>>> Best
>>>>>
>>>>> Vincent
>>>>>
>>>>> On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <s.safarov at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Think is requred streaming feature of freeswitch.
>>>>>> Look at mod_esf
>>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_esf> and
>>>>>> mod_vlc <https://wiki.freeswitch.org/wiki/Mod_vlc>
>>>>>> Instruction to compile mod_vlc on provided link is to old but
>>>>>> helpfull to undestand how to stream media to http server.
>>>>>>
>>>>>> For compiling mod_vlc please use vlc repo
>>>>>> <https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos
>>>>>> instruction
>>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>
>>>>>> .
>>>>>> After you intall vlc, then you can enable mod_vlc module in
>>>>>> freeswitch sources(SPEC file) and compile freeswitch.
>>>>>>
>>>>>> Sergey
>>>>>>
>>>>>>
>>>>>> ??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com>:
>>>>>>
>>>>>>> webdav, mod_http_cache or mod_httapi all results in sending the
>>>>>>> recording only *after* it is complete.
>>>>>>> They all write the recording to a file, wait for the recording to
>>>>>>> complete and the file to close and then send it over HTTP.
>>>>>>>
>>>>>>> I would like to start sending the recording to the remove server as
>>>>>>> soon as it starts (max 1 sec latency).
>>>>>>> mod_http_cache or mod_httapi would be perfect if they were streaming
>>>>>>> the recording like mod_shout.
>>>>>>>
>>>>>>>
>>>>>>> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <s.safarov at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Input/output latency is not problem.  I use Kazoo on my servers and
>>>>>>>> call recording is stored to database during 5 seconds after hangup.
>>>>>>>> What is broken in your case if save file using webdav or http_cache?
>>>>>>>>
>>>>>>>> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> Hello Sergey,
>>>>>>>>>
>>>>>>>>> Thank you for your answer.
>>>>>>>>> I've looked into webdav mounted filesystem.
>>>>>>>>>
>>>>>>>>> Unfortunately, most WebDav clients (especially davfs2 on debian)
>>>>>>>>> do a lot of buffering, caching and even lock-null requests (lock a non
>>>>>>>>> existent resource before writing to it). I also suspect that they wait for
>>>>>>>>> the end of the write operation.
>>>>>>>>> The result is a latency of a few seconds witch is not much better
>>>>>>>>> than what I achieve with mod_shout if I transcode the MP3.
>>>>>>>>>
>>>>>>>>> Any other idea ?
>>>>>>>>>
>>>>>>>>> Thank you !
>>>>>>>>>
>>>>>>>>> Best regards
>>>>>>>>>
>>>>>>>>> Vincent
>>>>>>>>>
>>>>>>>>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <
>>>>>>>>> s.safarov at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Please look at webdav mounted filesystem.
>>>>>>>>>>
>>>>>>>>>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> Hi all,
>>>>>>>>>>>
>>>>>>>>>>> Thank you to all contributing to FreeSWITCH !
>>>>>>>>>>>
>>>>>>>>>>> I'm working on a IVR project where logic is implemented on a
>>>>>>>>>>> HTTP server.
>>>>>>>>>>> We are leaving Twilio because we now need to record and live
>>>>>>>>>>> stream the session to the HTTP server in WAV format (chunked transfer
>>>>>>>>>>> encoding).
>>>>>>>>>>>
>>>>>>>>>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's) but
>>>>>>>>>>> it seems that the records are first saved to disk before there are sent to
>>>>>>>>>>> the server as chunked data.
>>>>>>>>>>> We need the transfer to start as soon as the recording starts.
>>>>>>>>>>>
>>>>>>>>>>> *mod_shout* does start the request almost as the records starts
>>>>>>>>>>> but it does not support WAV file and shout:// is not exactly a HTTP request
>>>>>>>>>>> (SOURCE method instead of PUT).
>>>>>>>>>>>
>>>>>>>>>>> Is there a way to use these modules to achieve our goal ?
>>>>>>>>>>>
>>>>>>>>>>> If not, we are willing to author a specific module or rather
>>>>>>>>>>> contribute to the existing ones.
>>>>>>>>>>>
>>>>>>>>>>> We've identified two approaches:
>>>>>>>>>>>
>>>>>>>>>>>    1. From *mod_httapi* Modify mod_httapi to directly stream
>>>>>>>>>>>    the record instead of completely saving it to disk before the HTTP chunked
>>>>>>>>>>>    transfer starts.
>>>>>>>>>>>    This seems the most logical but with more than 3000 lines,
>>>>>>>>>>>    mod_httapi does not seem to be the easiest module to build upon for
>>>>>>>>>>>    newcomers!
>>>>>>>>>>>
>>>>>>>>>>>    2. From *mod_shout*
>>>>>>>>>>>       1. Modify libshoot to replace the custom SOURCE method
>>>>>>>>>>>       with standard HTTP PUT method
>>>>>>>>>>>       2. Modify mod_shout to support wav files
>>>>>>>>>>>       3. Implement our IVR in script (javascript/lua)
>>>>>>>>>>>
>>>>>>>>>>> What do you think ?
>>>>>>>>>>>
>>>>>>>>>>> Thank you for your help.
>>>>>>>>>>>
>>>>>>>>>>> Cheers,
>>>>>>>>>>>
>>>>>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Vincent Gire
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Vincent Gire
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Vincent Gire
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From mike at jerris.com  Wed Apr  6 19:58:37 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Apr 2016 11:58:37 -0400
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION ERROR. SDP
In-Reply-To: <c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
Message-ID: <CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>

yes it's supported

On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:

> Hi Itola;
>
>
>
> Sorry, same error.
>
>
>
> Does Freeswitch support media switching (srtp-dtls) between two
> chrome(sip.js as signal) browsers?
>
>
>
> Finding when FS runs in media mode:
>
> codec causes caller side ?488 not acceptable here| incompatible
> destination ?
>
>                 callee side: ?cancel |user not registered?
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *?talo Rossi
> *Sent:* Tuesday, April 05, 2016 8:58 PM
> *To:* FreeSWITCH Users Help
> *Cc:* freeswitch-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> Set media_mix_inbound_outbound_codecs=true
>
> ?talo Rossi
>
> italo at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','italo at freeswitch.org');>
>
> IRC chat.freenode.net #freeswitch #freeswitch-dev
>
> Bugs? https://freeswitch.org/jira
>
> Docs? https://freeswitch.org/jira
>
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6
>
>
>
>
>
> On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com
> <javascript:_e(%7B%7D,'cvml','quanhs at stee.stengg.com');>> wrote:
>
> Hi All;
>
>
>
> I want to use freeswitch to set up a WebRTC POC (Audio only,
> udp/tls/rtp/savp).  Setup is Anonymous (
> 192.168.199.216/chrome/sipjs:sip-0.7.0.min) ->freeswitch (192.168.199.22)
> ->1010 (192.168.199.216/chrome/sip-0.7.0.min),just following the
> information in book ?FreeSWITCH 1.6 Cookbook?.
>
> If using media bypass mode (inbound-bypass-media == true), all works fine,
> caller and called can hear each other.
>
> But if disabling media bypass mode, call is rejected by FS.
>
>
>
> FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?.
> Attachment is detailed FS log file.
>
> Chrome uses opus 111, FS uses opus 116.
>
> Mod_sofia.c::sofia_receive_message() -->
> sofia_media.c::sofia_media_negotiate_sdp() -->
> sofia_core_media.c::sofia_core_media_negotiate_sdp().
>
>
>
> Help is needed to troubleshoot this issue.
>
>
>
> Thanks advance.
>
> Smile.
>
>
>
>
> [This e-mail is confidential and may be privileged. If you are not the
> intended recipient, please kindly notify us immediately and delete the
> message
> from your system; please do not copy or use it for any purpose, nor
> disclose
> its contents to any other person. Thank you.]
> ---ST Electronics Group---
>
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From mike at jerris.com  Wed Apr  6 20:01:06 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Apr 2016 12:01:06 -0400
Subject: [Freeswitch-users] Windows build
In-Reply-To: <CAC7ttFFfGFD+mzKRZWvwNy3BYoKK7FXY1pBgPGUMbj79qYeMmA@mail.gmail.com>
References: <CAC7ttFFYf+dt3NfgRQy0R5PbNut0=-=mmxv6+gTvUYFB1RsKBA@mail.gmail.com>
	<CAPEdYThObr1ha7ffrFNijAj3H30Z6Zk41TETEgdoeZWxLu6igg@mail.gmail.com>
	<CAC7ttFE98byMZX+uNYUcLJ-A8+WnQwcwXBqqPO-g2fBu3XRfvA@mail.gmail.com>
	<CAPEdYTghm8J20E5COERvG1TidbYNUqcVsjD1TngTyH-zwWE52w@mail.gmail.com>
	<CAC7ttFGRJQAFn0vC3sCC1tguvWmZ==BpKR4i-=UWjDPRgGx-hA@mail.gmail.com>
	<9589f6e5d757412ba5d620cf8e2bf26c@imladris.sermotec.local>
	<56E10D5D.1000303@zg.t-com.hr>
	<38494249596647c6a08591eca41e3000@imladris.sermotec.local>
	<CAC7ttFFbv9YAxXNg7zjc1B9cxLS74v6f6HfBpSt09sULor6EoA@mail.gmail.com>
	<CAHtxdDdY2JRqVuw-W0b+UF0Z_ZUGbCj_iUFUOXi=nrxDDzKb9A@mail.gmail.com>
	<CAC7ttFErJO-gTLBQTHy_6zCcRWBxOYgsXtJXqMC0U8Tv+79AOg@mail.gmail.com>
	<3ED4D03E-7C8C-4DE6-9DC1-1C6E039C5355@jerris.com>
	<CAC7ttFFfGFD+mzKRZWvwNy3BYoKK7FXY1pBgPGUMbj79qYeMmA@mail.gmail.com>
Message-ID: <CAEwTsAgU8dn+BqO0K94PPRvqTOKDZFBHdgw51tmkfKSrtmzbXA@mail.gmail.com>

I'll work on getting new ones out there, but it's easy to build it yourself

On Wednesday, April 6, 2016, Gregor Nanger <gregor at infomedia.si> wrote:

> Thank you Michael!
>
> Maybe it is something with my configuration of VS...
>
> Can you copy your build also on
> http://files.freeswitch.org/windows/installer/x64/?
>
> 2016-04-06 1:08 GMT+02:00 Michael Jerris <mike at jerris.com
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>>:
>
>> I just built windows fine yesterday.  It may be unhappy with rebuild, but
>> I built fine. (this should cover master).... mod_avmd might have been
>> failing for a few commits, but I pushed the fix for that module yesterday,
>> all the others in default configuration built fine.
>>
>> On Apr 5, 2016, at 6:38 PM, Gregor Nanger <gregor at infomedia.si
>> <javascript:_e(%7B%7D,'cvml','gregor at infomedia.si');>> wrote:
>>
>> Just bumping this post if someone has any hints.
>>
>> I am trying to build Freeswitch with Visual studio 2015, but keep getting
>> error:
>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>> ? ?
>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>> ?"
>>
>>
>> Would realy like to successfuly build Freeswitch in windows. I tried with
>> 1.6 branch.
>>
>> Best regards, Gregor
>>
>> 2016-03-17 5:43 GMT+01:00 Sergey Safarov <s.safarov at gmail.com
>> <javascript:_e(%7B%7D,'cvml','s.safarov at gmail.com');>>:
>>
>>> One week ago I successfully compiled mod_V8 on CentOS 7.
>>> May be switch to Linux?
>>>
>>> On Thu, Mar 17, 2016 at 12:49 AM, Gregor Nanger <gregor at infomedia.si
>>> <javascript:_e(%7B%7D,'cvml','gregor at infomedia.si');>> wrote:
>>>
>>>> Thank you Harald.
>>>>
>>>> I tried with latest branch 1.7 and x64 and got errors regarding mod_V8.
>>>> I tried what Peter suggested, but I am more in c# and web projects and do
>>>> not have experience in building  C++ projects. There is already prebuilt
>>>> setup on freeswitch site, so someone successfully build it :-)) It is not
>>>> so important for us at this point to make own build, so will try again
>>>> later.
>>>>
>>>>
>>>> 2016-03-11 7:32 GMT+01:00 Harald Petrovitsch <
>>>> Harald.Petrovitsch at sermotec.at
>>>> <javascript:_e(%7B%7D,'cvml','Harald.Petrovitsch at sermotec.at');>>:
>>>>
>>>>> Hi Gregor,
>>>>>
>>>>>
>>>>>
>>>>> I only do a
>>>>>
>>>>> Git.exe clone ?bv1.6
>>>>> https://freeswitch.org/stash/scm/fs/freeswitch.git .
>>>>>
>>>>>
>>>>>
>>>>> Loaded the solution into vs2015, set configuration Win32 / Release and
>>>>> press f7 (need to do it two times)
>>>>>
>>>>>
>>>>>
>>>>> The build ended with
>>>>>
>>>>> ========== Build: 20 succeeded, 0 failed, 157 up-to-date, 15 skipped
>>>>> ==========
>>>>>
>>>>>
>>>>>
>>>>> I?ve attached a list of the generated mod folder
>>>>>
>>>>>
>>>>>
>>>>> Regards
>>>>>
>>>>> Harald
>>>>>
>>>>>
>>>>>
>>>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
>>>>> *Im Auftrag von *Shishko
>>>>> *Gesendet:* Donnerstag, 10. M?rz 2016 07:00
>>>>> *An:* freeswitch-users at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>
>>>>>
>>>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>>>
>>>>>
>>>>>
>>>>> Hi Harald,
>>>>>
>>>>> what did you do to build libv8 and mod_v8? I tried with VS2015 Update
>>>>> 1, branch 1.6, but to no avail.
>>>>>
>>>>> Thanks
>>>>>
>>>>> On 03/07/2016 08:20 AM, Harald Petrovitsch wrote:
>>>>>
>>>>> Hi Gregor !
>>>>>
>>>>>
>>>>>
>>>>> V8 libs and mod builds fine here (visual Studio 2015 Sp1, 1.6 branch,
>>>>> used tortoiseGit to download it))
>>>>>
>>>>>
>>>>>
>>>>> Regards
>>>>>
>>>>> Harald
>>>>>
>>>>>
>>>>>
>>>>> *Von:* freeswitch-users-bounces at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
>>>>> *Im Auftrag v**on *Gregor Nanger
>>>>> *Gesendet:* Montag, 07. M?rz 2016 00:19
>>>>> *An:* FreeSWITCH Users Help
>>>>> *Betreff:* Re: [Freeswitch-users] Windows build
>>>>>
>>>>>
>>>>>
>>>>> Thank you, H
>>>>>
>>>>> ?a?
>>>>>
>>>>> rald. This works: "AFAIK, for the ?'lame/lame.h'? you have to change
>>>>> the include line to  only ?lame.h?"
>>>>>
>>>>> B
>>>>>
>>>>> ?ut for v8 stil do  not have solution. I do not want to exclude
>>>>> mod_v8, since this module runs javascript. But, can you please confirm me
>>>>> that is not yet compatible, to stop trying to solve it.
>>>>>
>>>>>
>>>>> Any other suggestion what does this mean:
>>>>> "
>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>>>
>>>>> ? ?
>>>>>
>>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>
>>>>> ?"
>>>>>
>>>>>
>>>>>
>>>>> Best regards, Gregor?
>>>>>
>>>>>
>>>>>
>>>>> 2016-03-06 16:39 GMT+01:00 Peter Olsson <peter at olssononline.se
>>>>> <javascript:_e(%7B%7D,'cvml','peter at olssononline.se');>>:
>>>>>
>>>>> Remove mod_v8 from the build. I don't think it's compatible with
>>>>> VS2015 for now. However, all other modules should be ok.
>>>>>
>>>>>
>>>>>
>>>>> /Peter
>>>>>
>>>>>
>>>>>
>>>>> 2016-03-06 12:31 GMT+01:00 Gregor Nanger <gregor at infomedia.si
>>>>> <javascript:_e(%7B%7D,'cvml','gregor at infomedia.si');>>:
>>>>>
>>>>> ?This helped a lot, thank you. Now I have only few errors. Any hint?
>>>>>
>>>>>
>>>>>
>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln]
>>>>> libv8 C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>
>>>>> Error MSB4057 The target "v8:Rebuild" does not exist in the project.
>>>>> [C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln] libv8
>>>>> C:\Git\freeswitch\libs\v8-3.24.14\tools\gyp\v8.sln.metaproj 1
>>>>>
>>>>> Error C1083 Cannot open include file: 'lame/lame.h': No such file or
>>>>> directory mod_shout C:\Git\freeswitch\src\mod\formats\mod_shout\mod_shout.c
>>>>> 38
>>>>>
>>>>> Error LNK1181 cannot open input file 'icui18n.lib' mod_v8
>>>>> C:\Git\freeswitch\src\mod\languages\mod_v8\LINK 1
>>>>>
>>>>>
>>>>>
>>>>> ?
>>>>>
>>>>>
>>>>>
>>>>> 2016-03-06 7:53 GMT+01:00 Peter Olsson <peter at olssononline.se
>>>>> <javascript:_e(%7B%7D,'cvml','peter at olssononline.se');>>:
>>>>>
>>>>> One common mistake is that you allow Git to modify line endings. Make
>>>>> sure autocrlf is turned off - then clone the repository again from scratch.
>>>>>
>>>>>
>>>>>
>>>>> Also, I'm not sure if it will work in VS2015, but give it a try.
>>>>>
>>>>>
>>>>>
>>>>> /Peter
>>>>>
>>>>>
>>>>>
>>>>> 2016-03-06 2:06 GMT+01:00 Gregor Nanger <gregor at infomedia.si
>>>>> <javascript:_e(%7B%7D,'cvml','gregor at infomedia.si');>>:
>>>>>
>>>>> Hi!
>>>>>
>>>>>
>>>>>
>>>>> I want to build Freeswitch on windows with visual studio 2015.
>>>>>
>>>>>
>>>>>
>>>>> Where should I start if I get 600 errors when try to Rebuild All. I
>>>>> opened solution and start Rebuild All, but I get so many errors that I
>>>>> belive that I am doing something wrong.
>>>>>
>>>>>
>>>>>
>>>>> Mainl yre errors regarding:
>>>>>
>>>>>
>>>>>
>>>>> Cannot open source file....
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Best regards, Gregor
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> *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
>>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> *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 <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>>>
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>>
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>>
>>>>> http://www.freeswitch.org
>>>>>
>>>>> http://confluence.freeswitch.org
>>>>>
>>>>> http://www.cluecon.com
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>
>>>>> FreeSWITCH-users at lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>>>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> 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
>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> 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
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> 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 anthony.minessale at gmail.com  Wed Apr  6 20:10:39 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Apr 2016 11:10:39 -0500
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAKsJLpaCq3=ifM_FFav2-iE9X7f00FRHppiRjO25Yc-=rKeuew@mail.gmail.com>
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Firstly,

You cannot stream WAV.  Its not possible.  WAV is not a streaming format,
its a container.  Its basically a raw audio data stream with a header
explaining the characteristics of the stream.


Next,

You need to compare your specific needs to the goal of the software to be
flexible and usable to all.

The reason the speech interface was suggested is that the point of the
speech interface is precisely to tap into live audio streams and perform
asynchronous ASR operations while the original call does whatever it wants
like talk to another caller or listen to a file playing etc.

What you envision is more of a synchronous single instruction to record a
file in realtime to the remote server over httapi.
The considerations to implement this would be what format to send the audio
in. The source audio could be any combination of sample rate, channel count
and encoding format.
To record we would translate the encoded format into signed linear at the
original input sample rate and channels.  These variances must be
communicated before you start sending the data so it would be a matter of
figuring out an existing standard or implement a proprietary one.





On Wed, Apr 6, 2016 at 10:42 AM, Vincent Gire <vincent.gire at gmail.com>
wrote:

> We already handle the multiple providers on the HTTP server.
> So you would recommend to implement a module against the speech interface
> streaming to the HTTP server ?
> Wouldn't it be easier (and less redundant) to add streaming capabilities
> to mod_httpapi ?
>
> On Wed, Apr 6, 2016 at 12:22 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> The best way to handle this is probably to write a custom module against
>> the speech interface that sends the streams like you are describing, and
>> supports multiple providers.  How exactly do you go about sending to
>> multiple at the same time and combining the results, as this might actually
>> be the trickiest part as youll need to integrate that into a module that
>> uses the speech interface to be able to at all sanely handle that in
>> freeswitch
>>
>> On Apr 5, 2016, at 1:24 PM, Vincent Gire <vincent.gire at gmail.com> wrote:
>>
>> We use multiple ones to compare confidence.
>> They all support chunk transfer and provide a HTTP transaction similar to
>> :
>> http://developer.att.com/apis/speech/docs/v3
>> Look at 3/ Make API Calls
>>
>> On Tue, Apr 5, 2016 at 7:14 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Unimrcp isn't going to provide the interface he's talking about.  Can I
>>> ask which engine this is?
>>>
>>>
>>> On Tuesday, April 5, 2016, Sergey Safarov <s.safarov at gmail.com> wrote:
>>>
>>>> Look at http://www.unimrcp.org/ and
>>>> https://wiki.freeswitch.org/wiki/Mod_unimrcp
>>>>
>>>> On Tue, Apr 5, 2016, 19:38 Vincent Gire <vincent.gire at gmail.com> wrote:
>>>>
>>>>> We are building an IVR completely driven by ASR.
>>>>> ASR is performed in a distant location by a HTTP service (supporting
>>>>> chunked transfer) and adds an incompressible latency. We would like to
>>>>> stream the record to the ASR service as soon as it starts to reduce the
>>>>> overall latency before response.
>>>>> Does it make sense ?
>>>>>
>>>>>
>>>>> On Tue, Apr 5, 2016 at 5:39 PM, Michael Jerris <mike at jerris.com>
>>>>> wrote:
>>>>>
>>>>>> I would stay away from mod_vlc.  Its audio portions with recording
>>>>>> have known issues.  We do use the rtmp streaming in mod_av heavily but
>>>>>> thats obviously not wav.  Can you explain a bit more why you have this
>>>>>> requirement?
>>>>>>
>>>>>> On Apr 5, 2016, at 5:36 AM, Vincent Gire <vincent.gire at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>> Ok thanks.
>>>>>> It looks promising !
>>>>>> I'll dig into mod_vlc.
>>>>>>
>>>>>> Best
>>>>>>
>>>>>> Vincent
>>>>>>
>>>>>> On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <s.safarov at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Think is requred streaming feature of freeswitch.
>>>>>>> Look at mod_esf
>>>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_esf> and
>>>>>>> mod_vlc <https://wiki.freeswitch.org/wiki/Mod_vlc>
>>>>>>> Instruction to compile mod_vlc on provided link is to old but
>>>>>>> helpfull to undestand how to stream media to http server.
>>>>>>>
>>>>>>> For compiling mod_vlc please use vlc repo
>>>>>>> <https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos
>>>>>>> instruction
>>>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>
>>>>>>> .
>>>>>>> After you intall vlc, then you can enable mod_vlc module in
>>>>>>> freeswitch sources(SPEC file) and compile freeswitch.
>>>>>>>
>>>>>>> Sergey
>>>>>>>
>>>>>>>
>>>>>>> ??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com>:
>>>>>>>
>>>>>>>> webdav, mod_http_cache or mod_httapi all results in sending the
>>>>>>>> recording only *after* it is complete.
>>>>>>>> They all write the recording to a file, wait for the recording to
>>>>>>>> complete and the file to close and then send it over HTTP.
>>>>>>>>
>>>>>>>> I would like to start sending the recording to the remove server as
>>>>>>>> soon as it starts (max 1 sec latency).
>>>>>>>> mod_http_cache or mod_httapi would be perfect if they were
>>>>>>>> streaming the recording like mod_shout.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <s.safarov at gmail.com
>>>>>>>> > wrote:
>>>>>>>>
>>>>>>>>> Input/output latency is not problem.  I use Kazoo on my servers
>>>>>>>>> and call recording is stored to database during 5 seconds after hangup.
>>>>>>>>> What is broken in your case if save file using webdav or
>>>>>>>>> http_cache?
>>>>>>>>>
>>>>>>>>> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> Hello Sergey,
>>>>>>>>>>
>>>>>>>>>> Thank you for your answer.
>>>>>>>>>> I've looked into webdav mounted filesystem.
>>>>>>>>>>
>>>>>>>>>> Unfortunately, most WebDav clients (especially davfs2 on debian)
>>>>>>>>>> do a lot of buffering, caching and even lock-null requests (lock a non
>>>>>>>>>> existent resource before writing to it). I also suspect that they wait for
>>>>>>>>>> the end of the write operation.
>>>>>>>>>> The result is a latency of a few seconds witch is not much better
>>>>>>>>>> than what I achieve with mod_shout if I transcode the MP3.
>>>>>>>>>>
>>>>>>>>>> Any other idea ?
>>>>>>>>>>
>>>>>>>>>> Thank you !
>>>>>>>>>>
>>>>>>>>>> Best regards
>>>>>>>>>>
>>>>>>>>>> Vincent
>>>>>>>>>>
>>>>>>>>>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <
>>>>>>>>>> s.safarov at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Please look at webdav mounted filesystem.
>>>>>>>>>>>
>>>>>>>>>>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Hi all,
>>>>>>>>>>>>
>>>>>>>>>>>> Thank you to all contributing to FreeSWITCH !
>>>>>>>>>>>>
>>>>>>>>>>>> I'm working on a IVR project where logic is implemented on a
>>>>>>>>>>>> HTTP server.
>>>>>>>>>>>> We are leaving Twilio because we now need to record and live
>>>>>>>>>>>> stream the session to the HTTP server in WAV format (chunked transfer
>>>>>>>>>>>> encoding).
>>>>>>>>>>>>
>>>>>>>>>>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's)
>>>>>>>>>>>> but it seems that the records are first saved to disk before there are sent
>>>>>>>>>>>> to the server as chunked data.
>>>>>>>>>>>> We need the transfer to start as soon as the recording starts.
>>>>>>>>>>>>
>>>>>>>>>>>> *mod_shout* does start the request almost as the records
>>>>>>>>>>>> starts but it does not support WAV file and shout:// is not exactly a HTTP
>>>>>>>>>>>> request (SOURCE method instead of PUT).
>>>>>>>>>>>>
>>>>>>>>>>>> Is there a way to use these modules to achieve our goal ?
>>>>>>>>>>>>
>>>>>>>>>>>> If not, we are willing to author a specific module or rather
>>>>>>>>>>>> contribute to the existing ones.
>>>>>>>>>>>>
>>>>>>>>>>>> We've identified two approaches:
>>>>>>>>>>>>
>>>>>>>>>>>>    1. From *mod_httapi* Modify mod_httapi to directly stream
>>>>>>>>>>>>    the record instead of completely saving it to disk before the HTTP chunked
>>>>>>>>>>>>    transfer starts.
>>>>>>>>>>>>    This seems the most logical but with more than 3000 lines,
>>>>>>>>>>>>    mod_httapi does not seem to be the easiest module to build upon for
>>>>>>>>>>>>    newcomers!
>>>>>>>>>>>>
>>>>>>>>>>>>    2. From *mod_shout*
>>>>>>>>>>>>       1. Modify libshoot to replace the custom SOURCE method
>>>>>>>>>>>>       with standard HTTP PUT method
>>>>>>>>>>>>       2. Modify mod_shout to support wav files
>>>>>>>>>>>>       3. Implement our IVR in script (javascript/lua)
>>>>>>>>>>>>
>>>>>>>>>>>> What do you think ?
>>>>>>>>>>>>
>>>>>>>>>>>> Thank you for your help.
>>>>>>>>>>>>
>>>>>>>>>>>> Cheers,
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
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>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Vincent Gire
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
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>>>>> http://www.freeswitchsolutions.com
>>>>>
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>>>>
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>>
>>
>>
>> --
>> Vincent Gire
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
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>>
>>
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>>
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>
>
>
> --
> Vincent Gire
>
> _________________________________________________________________________
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-- 
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From mike at jerris.com  Wed Apr  6 20:20:37 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Apr 2016 12:20:37 -0400
Subject: [Freeswitch-users] Is MySQL OK to use?
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	<A70AC0C1B7E7FD478A6BF19705461AA0C91C@IMBER.smb.curriegrad2004.ca>
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This sounds like a bug in the mysql driver then right?  If those are not threadsafe then the driver should protect against use where it is not safe.

> On Apr 6, 2016, at 5:44 AM, Steven Ayre <steveayre at gmail.com> wrote:
> 
> I've been doing some debugging on this this week as we're currently seeing regular crashes on our production servers. This happens with both Threading=0 and Threading=2.
> 
> It appears it's a bug in MyODBC triggered because FreeSWITCH uses a new ODBC environment handle every time it creates a connection.
> 
> When a environment handle is created myodbc_init is called and when it's freed it calls myodbc_end. Those functions are not thread-safe. They're protected by a reference counter but they aren't wrapped in a mutex so if the server is busy and they're both called at the same time you can result in some global variables being used after they're freed.
> 
> Rebuilding MyODBC with an empty myodbc_end appears to stop the crashes, but I'm not saying that's a good solution. Still looking at it.
> 
> On 22 November 2013 at 20:31, Anthony Minessale <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> I've always seen Threading=0 as the solution to stability probs.
> Basically if the driver already uses mutexing, its better to disable the arbitrary ones in the core of odbc.
> For postgres 0 is basically mandatory.
> 
> 
> 
> On Fri, Nov 22, 2013 at 2:12 AM, Steven Ayre <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
> No such thing in the latest versions, they're all threadsafe now (_r is a symlink to the other).
> 
> I use it with success. I did find stability problems in libmyodbc when upgrading from 5.0 to 5.5. My solution was to add these to odbc.ini
> 
> Option      = 67108864
> Threading   = 2
> 
> The Threading setting is what stopped the crashes. There's a Jira on the subject and someone else said manually upgrading to a newer version of unixodbc+myodbc also fixed it.
> 
> -Steve
> 
> 
> 
> On 21 November 2013 18:23, Jeff Leung <jleung at v10networks.ca <mailto:jleung at v10networks.ca>> wrote:
> Try the non-thread safe MySQL library instead of the thread safe one.
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Vik Killa
> Sent: Thursday, November 21, 2013 10:22 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Is MySQL OK to use?
> 
>  
> 
> short answer no.
> 
> use postgres
> 
>  
> 
> On Thu, Nov 21, 2013 at 12:41 PM, Alex Lake <alex at digitalmail.com <mailto:alex at digitalmail.com>> wrote:
> 
> I find these "Error in my_thread_global_end()" messages somewhat
> annoying in my fs1.2stable on Ubuntu12.04 box. Is there any advice
> (other than "don't use MySQL") for how to install it better? Might it be
> something to do with thread-safe libraries (I'm using libmyodbc_r.so at
> the moment)
> 
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> Anthony Minessale II
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> 
> 
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> consulting at freeswitch.org
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From mike at jerris.com  Wed Apr  6 20:22:13 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Apr 2016 12:22:13 -0400
Subject: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
In-Reply-To: <CACot2nvgSJOFj9HJ2JOc4Ofp1Ce_x-mZ0G95epFJA1AwMZ2X5g@mail.gmail.com>
References: <CACot2ntTB+7OXvomm3sxMzxnNEnT6vHtQb=T=m8MTOFT0=zFZg@mail.gmail.com>
	<E93C0E00-0EEE-44E1-81F2-1E8CF374057B@jerris.com>
	<CACot2ntmtZzcXs4bWqd3oW9i5URXMpqaouu8EXKWeN2thDJG4A@mail.gmail.com>
	<CACot2nvgSJOFj9HJ2JOc4Ofp1Ce_x-mZ0G95epFJA1AwMZ2X5g@mail.gmail.com>
Message-ID: <A84CA188-7849-4678-98DC-33BA615A9A0C@jerris.com>

is this for a client or server application?  If it is for a client application, we may not be the right stack for you.  We make a number of assumptions in our ice that is based on it being a server in order to improve server efficiency.

> On Apr 6, 2016, at 5:01 AM, saurabh verrma <saurabhkv01 at gmail.com> wrote:
> 
> Hi,
> 
> I?m working on an application where I?m trying to use FreeSWITCH as a library. My intention is to use FreeSWITCH as a UAS endpoint. Basically it needs to be supporting following: 
> 1. WebRTC 
> 2. Ability to act like UAS endpoint 
> 3. Support for DTLS/SRTP 
> 4. ICE support
> 
> I?m seeking community suggestion if that?s feasible to implement or not? If yes, what are the possible starting directions we could explore above points. 
> 
> Any help would be greatly appreciated.
> 
> 
> On Sat, Apr 2, 2016 at 12:24 PM, saurabh verrma <saurabhkv01 at gmail.com <mailto:saurabhkv01 at gmail.com>> wrote:
> Thanks Michael, 
> 
> Basically we're writing a PJSIP based application & PJSIP doesn't have DTLS support. So we're thinking to use FreeSWITCH library for DTLS/SRTP.
> 
> On Fri, Apr 1, 2016 at 7:29 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> we have full support for webrtc media profile which would include these features  There are not a ton of people who use freeswitch as a library, but it is built that way so that you can control it and host it in another application instead of stand alone.  If you are trying to accomplish something I'd try to handle it standalone first so you can learn all the different ways you might control it before architecting a solution
> 
>> On Apr 1, 2016, at 9:48 AM, saurabh verrma <saurabhkv01 at gmail.com <mailto:saurabhkv01 at gmail.com>> wrote:
>> 
>> Hi All, 
>> 
>> I want to use FreeSWITCH library for DTLS/SRTP support. I want to know in freeswitch which library has the support of these features(SRTP/DTLS) ?
>> Is there any application available based on freeswitch library ?
>> 
>> Any help would be appreciable.

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From anthony.minessale at gmail.com  Wed Apr  6 20:32:00 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Apr 2016 11:32:00 -0500
Subject: [Freeswitch-users] Is MySQL OK to use?
In-Reply-To: <A8930F04-C0DF-4C38-9AAD-3E80189FBC2D@jerris.com>
References: <528E45C5.8030509@digitalmail.com>
	<CAC-LwPME371eMDigDeZXTbf-WeOLx56La8iE9eyyS-f+xh6FGA@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C91C@IMBER.smb.curriegrad2004.ca>
	<CAFiqYu=F2w51wy4M15ygP_=iHSe5orqRqu6zdnYNND8D6=ix=g@mail.gmail.com>
	<CAKbxfG-yLgyzgq3D0RT+96RcfTSWBi7myX+t0ew37arRSLYkXQ@mail.gmail.com>
	<CAFiqYumNo9wQHyZNx5ufuhohE_ivBQJTTrYm7suY4-8UbGzmNQ@mail.gmail.com>
	<A8930F04-C0DF-4C38-9AAD-3E80189FBC2D@jerris.com>
Message-ID: <CAKbxfG9HpySNbdGRc4K2HbC7WuHFwnx4mV=NWDEEhegp_K+mzA@mail.gmail.com>

There is no room for non-threadsafe code in the future of computers with
the number of cores only going up.
If they don't still have the "thread safe" version of mysql  (with the _r
at the end), then all versions should be probably considered ok in threaded
env and it would indeed be a bug unless there is some documentation to
suggest they gave up on being thread safe in which case, you may want to
use postgres rather than fight the hard battle with mysql.


On Wed, Apr 6, 2016 at 11:20 AM, Michael Jerris <mike at jerris.com> wrote:

> This sounds like a bug in the mysql driver then right?  If those are not
> threadsafe then the driver should protect against use where it is not safe.
>
> On Apr 6, 2016, at 5:44 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
> I've been doing some debugging on this this week as we're currently seeing
> regular crashes on our production servers. This happens with both
> Threading=0 and Threading=2.
>
> It appears it's a bug in MyODBC triggered because FreeSWITCH uses a new
> ODBC environment handle every time it creates a connection.
>
> When a environment handle is created myodbc_init is called and when it's
> freed it calls myodbc_end. Those functions are not thread-safe. They're
> protected by a reference counter but they aren't wrapped in a mutex so if
> the server is busy and they're both called at the same time you can result
> in some global variables being used after they're freed.
>
> Rebuilding MyODBC with an empty myodbc_end appears to stop the crashes,
> but I'm not saying that's a good solution. Still looking at it.
>
> On 22 November 2013 at 20:31, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I've always seen Threading=0 as the solution to stability probs.
>> Basically if the driver already uses mutexing, its better to disable the
>> arbitrary ones in the core of odbc.
>> For postgres 0 is basically mandatory.
>>
>>
>>
>> On Fri, Nov 22, 2013 at 2:12 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> No such thing in the latest versions, they're all threadsafe now (_r is
>>> a symlink to the other).
>>>
>>> I use it with success. I did find stability problems in libmyodbc when
>>> upgrading from 5.0 to 5.5. My solution was to add these to odbc.ini
>>>
>>> Option      = 67108864
>>> Threading   = 2
>>>
>>> The Threading setting is what stopped the crashes. There's a Jira on the
>>> subject and someone else said manually upgrading to a newer version of
>>> unixodbc+myodbc also fixed it.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 21 November 2013 18:23, Jeff Leung <jleung at v10networks.ca> wrote:
>>>
>>>> Try the non-thread safe MySQL library instead of the thread safe one.
>>>>
>>>>
>>>>
>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Vik Killa
>>>> *Sent:* Thursday, November 21, 2013 10:22 AM
>>>> *To:* FreeSWITCH Users Help
>>>> *Subject:* Re: [Freeswitch-users] Is MySQL OK to use?
>>>>
>>>>
>>>>
>>>> short answer no.
>>>>
>>>> use postgres
>>>>
>>>>
>>>>
>>>> On Thu, Nov 21, 2013 at 12:41 PM, Alex Lake <alex at digitalmail.com>
>>>> wrote:
>>>>
>>>> I find these "Error in my_thread_global_end()" messages somewhat
>>>> annoying in my fs1.2stable on Ubuntu12.04 box. Is there any advice
>>>> (other than "don't use MySQL") for how to install it better? Might it be
>>>> something to do with thread-safe libraries (I'm using libmyodbc_r.so at
>>>> the moment)
>>>>
>>>>
>>>> _________________________________________________________________________
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>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
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>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>> http://www.cudatel.com
>>>>
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>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>> http://www.cudatel.com
>>>
>>> Official FreeSWITCH Sites
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>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>> http://www.cudatel.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
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>>
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>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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>
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> http://www.freeswitchsolutions.com
>
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>



-- 
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
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From vincent.gire at gmail.com  Wed Apr  6 20:47:07 2016
From: vincent.gire at gmail.com (Vincent Gire)
Date: Wed, 6 Apr 2016 18:47:07 +0200
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAKbxfG-auQAvHZrztzgnF_4O363tK43VfkFC1xuwZgjmDKiaew@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
	<CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
	<CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>
	<CAKsJLpa83aWDBMFuZaYeMFVR6XeN=2uQqwcuTXAtqTY3szckSg@mail.gmail.com>
	<E6948933-09AC-4A48-A0C9-70E206EED453@jerris.com>
	<CAKsJLpavsm2N7odd2xw6CP6zuBxhLR_QEy3acEPukujVAPU5HA@mail.gmail.com>
	<CAHtxdDfQGXWAUopsmOUO3hTV2A91i9t=9ss5P3+GtnxPtwwsfg@mail.gmail.com>
	<CAEwTsAid_21ysmiu3E+iUdvazMGpLpb0KOpVCOxK6WaurmnWEA@mail.gmail.com>
	<CAKsJLpYAR333mU6VuD2AjxepbPJ4quyzGKpW2ziOmMLds6rgsg@mail.gmail.com>
	<731C9BFA-0101-42F3-B368-17F8AEFD7580@jerris.com>
	<CAKsJLpaCq3=ifM_FFav2-iE9X7f00FRHppiRjO25Yc-=rKeuew@mail.gmail.com>
	<CAKbxfG-auQAvHZrztzgnF_4O363tK43VfkFC1xuwZgjmDKiaew@mail.gmail.com>
Message-ID: <CAKsJLpY0jtfa6MBhVHHQTgOA_RC_nQv35b7DbUa99w6gCk635w@mail.gmail.com>

On Wed, Apr 6, 2016 at 6:10 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Firstly,
>
> You cannot stream WAV.  Its not possible.  WAV is not a streaming format,
> its a container.  Its basically a raw audio data stream with a header
> explaining the characteristics of the stream.
>

Yes. Sorry for the language abuse and thank you for clearing this out.


>
>
> Next,
>
> You need to compare your specific needs to the goal of the software to be
> flexible and usable to all.
>
> The reason the speech interface was suggested is that the point of the
> speech interface is precisely to tap into live audio streams and perform
> asynchronous ASR operations while the original call does whatever it wants
> like talk to another caller or listen to a file playing etc.
>
> What you envision is more of a synchronous single instruction to record a
> file in realtime to the remote server over httapi.
>

Exactly.


> The considerations to implement this would be what format to send the
> audio in. The source audio could be any combination of sample rate, channel
> count and encoding format.
> To record we would translate the encoded format into signed linear at the
> original input sample rate and channels.  These variances must be
> communicated before you start sending the data so it would be a matter of
> figuring out an existing standard or implement a proprietary one.
>

I am not sure I am following here.
I might again have misused the word streaming.
I was thinking that when we specify a http target for a record in
mod_httapi, it could start sending the corresponding file (WAV or other) to
the HTTP server as chunked data as soon as the record starts.
In the case of a WAV target, the WAV header would be sent first as part of
the first HTTP data chunks and the raw audio data stream would follow in
the next data chunks.
Does it make sense ?


>
>
>
>
> On Wed, Apr 6, 2016 at 10:42 AM, Vincent Gire <vincent.gire at gmail.com>
> wrote:
>
>> We already handle the multiple providers on the HTTP server.
>> So you would recommend to implement a module against the speech interface
>> streaming to the HTTP server ?
>> Wouldn't it be easier (and less redundant) to add streaming capabilities
>> to mod_httpapi ?
>>
>> On Wed, Apr 6, 2016 at 12:22 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> The best way to handle this is probably to write a custom module against
>>> the speech interface that sends the streams like you are describing, and
>>> supports multiple providers.  How exactly do you go about sending to
>>> multiple at the same time and combining the results, as this might actually
>>> be the trickiest part as youll need to integrate that into a module that
>>> uses the speech interface to be able to at all sanely handle that in
>>> freeswitch
>>>
>>> On Apr 5, 2016, at 1:24 PM, Vincent Gire <vincent.gire at gmail.com> wrote:
>>>
>>> We use multiple ones to compare confidence.
>>> They all support chunk transfer and provide a HTTP transaction similar
>>> to :
>>> http://developer.att.com/apis/speech/docs/v3
>>> Look at 3/ Make API Calls
>>>
>>> On Tue, Apr 5, 2016 at 7:14 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> Unimrcp isn't going to provide the interface he's talking about.  Can I
>>>> ask which engine this is?
>>>>
>>>>
>>>> On Tuesday, April 5, 2016, Sergey Safarov <s.safarov at gmail.com> wrote:
>>>>
>>>>> Look at http://www.unimrcp.org/ and
>>>>> https://wiki.freeswitch.org/wiki/Mod_unimrcp
>>>>>
>>>>> On Tue, Apr 5, 2016, 19:38 Vincent Gire <vincent.gire at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> We are building an IVR completely driven by ASR.
>>>>>> ASR is performed in a distant location by a HTTP service (supporting
>>>>>> chunked transfer) and adds an incompressible latency. We would like to
>>>>>> stream the record to the ASR service as soon as it starts to reduce the
>>>>>> overall latency before response.
>>>>>> Does it make sense ?
>>>>>>
>>>>>>
>>>>>> On Tue, Apr 5, 2016 at 5:39 PM, Michael Jerris <mike at jerris.com>
>>>>>> wrote:
>>>>>>
>>>>>>> I would stay away from mod_vlc.  Its audio portions with recording
>>>>>>> have known issues.  We do use the rtmp streaming in mod_av heavily but
>>>>>>> thats obviously not wav.  Can you explain a bit more why you have this
>>>>>>> requirement?
>>>>>>>
>>>>>>> On Apr 5, 2016, at 5:36 AM, Vincent Gire <vincent.gire at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>> Ok thanks.
>>>>>>> It looks promising !
>>>>>>> I'll dig into mod_vlc.
>>>>>>>
>>>>>>> Best
>>>>>>>
>>>>>>> Vincent
>>>>>>>
>>>>>>> On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <s.safarov at gmail.com
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> Think is requred streaming feature of freeswitch.
>>>>>>>> Look at mod_esf
>>>>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_esf> and
>>>>>>>> mod_vlc <https://wiki.freeswitch.org/wiki/Mod_vlc>
>>>>>>>> Instruction to compile mod_vlc on provided link is to old but
>>>>>>>> helpfull to undestand how to stream media to http server.
>>>>>>>>
>>>>>>>> For compiling mod_vlc please use vlc repo
>>>>>>>> <https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos
>>>>>>>> instruction
>>>>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>
>>>>>>>> .
>>>>>>>> After you intall vlc, then you can enable mod_vlc module in
>>>>>>>> freeswitch sources(SPEC file) and compile freeswitch.
>>>>>>>>
>>>>>>>> Sergey
>>>>>>>>
>>>>>>>>
>>>>>>>> ??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com>:
>>>>>>>>
>>>>>>>>> webdav, mod_http_cache or mod_httapi all results in sending the
>>>>>>>>> recording only *after* it is complete.
>>>>>>>>> They all write the recording to a file, wait for the recording to
>>>>>>>>> complete and the file to close and then send it over HTTP.
>>>>>>>>>
>>>>>>>>> I would like to start sending the recording to the remove server
>>>>>>>>> as soon as it starts (max 1 sec latency).
>>>>>>>>> mod_http_cache or mod_httapi would be perfect if they were
>>>>>>>>> streaming the recording like mod_shout.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <
>>>>>>>>> s.safarov at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Input/output latency is not problem.  I use Kazoo on my servers
>>>>>>>>>> and call recording is stored to database during 5 seconds after hangup.
>>>>>>>>>> What is broken in your case if save file using webdav or
>>>>>>>>>> http_cache?
>>>>>>>>>>
>>>>>>>>>> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> Hello Sergey,
>>>>>>>>>>>
>>>>>>>>>>> Thank you for your answer.
>>>>>>>>>>> I've looked into webdav mounted filesystem.
>>>>>>>>>>>
>>>>>>>>>>> Unfortunately, most WebDav clients (especially davfs2 on debian)
>>>>>>>>>>> do a lot of buffering, caching and even lock-null requests (lock a non
>>>>>>>>>>> existent resource before writing to it). I also suspect that they wait for
>>>>>>>>>>> the end of the write operation.
>>>>>>>>>>> The result is a latency of a few seconds witch is not much
>>>>>>>>>>> better than what I achieve with mod_shout if I transcode the MP3.
>>>>>>>>>>>
>>>>>>>>>>> Any other idea ?
>>>>>>>>>>>
>>>>>>>>>>> Thank you !
>>>>>>>>>>>
>>>>>>>>>>> Best regards
>>>>>>>>>>>
>>>>>>>>>>> Vincent
>>>>>>>>>>>
>>>>>>>>>>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <
>>>>>>>>>>> s.safarov at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Please look at webdav mounted filesystem.
>>>>>>>>>>>>
>>>>>>>>>>>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <vincent.gire at gmail.com>
>>>>>>>>>>>> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Hi all,
>>>>>>>>>>>>>
>>>>>>>>>>>>> Thank you to all contributing to FreeSWITCH !
>>>>>>>>>>>>>
>>>>>>>>>>>>> I'm working on a IVR project where logic is implemented on a
>>>>>>>>>>>>> HTTP server.
>>>>>>>>>>>>> We are leaving Twilio because we now need to record and live
>>>>>>>>>>>>> stream the session to the HTTP server in WAV format (chunked transfer
>>>>>>>>>>>>> encoding).
>>>>>>>>>>>>>
>>>>>>>>>>>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's)
>>>>>>>>>>>>> but it seems that the records are first saved to disk before there are sent
>>>>>>>>>>>>> to the server as chunked data.
>>>>>>>>>>>>> We need the transfer to start as soon as the recording starts.
>>>>>>>>>>>>>
>>>>>>>>>>>>> *mod_shout* does start the request almost as the records
>>>>>>>>>>>>> starts but it does not support WAV file and shout:// is not exactly a HTTP
>>>>>>>>>>>>> request (SOURCE method instead of PUT).
>>>>>>>>>>>>>
>>>>>>>>>>>>> Is there a way to use these modules to achieve our goal ?
>>>>>>>>>>>>>
>>>>>>>>>>>>> If not, we are willing to author a specific module or rather
>>>>>>>>>>>>> contribute to the existing ones.
>>>>>>>>>>>>>
>>>>>>>>>>>>> We've identified two approaches:
>>>>>>>>>>>>>
>>>>>>>>>>>>>    1. From *mod_httapi* Modify mod_httapi to directly stream
>>>>>>>>>>>>>    the record instead of completely saving it to disk before the HTTP chunked
>>>>>>>>>>>>>    transfer starts.
>>>>>>>>>>>>>    This seems the most logical but with more than 3000 lines,
>>>>>>>>>>>>>    mod_httapi does not seem to be the easiest module to build upon for
>>>>>>>>>>>>>    newcomers!
>>>>>>>>>>>>>
>>>>>>>>>>>>>    2. From *mod_shout*
>>>>>>>>>>>>>       1. Modify libshoot to replace the custom SOURCE method
>>>>>>>>>>>>>       with standard HTTP PUT method
>>>>>>>>>>>>>       2. Modify mod_shout to support wav files
>>>>>>>>>>>>>       3. Implement our IVR in script (javascript/lua)
>>>>>>>>>>>>>
>>>>>>>>>>>>> What do you think ?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Thank you for your help.
>>>>>>>>>>>>>
>>>>>>>>>>>>> Cheers,
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Vincent Gire
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Vincent Gire
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Vincent Gire
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Vincent Gire
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From omortimer at gmail.com  Wed Apr  6 20:48:50 2016
From: omortimer at gmail.com (Oz Mortimer)
Date: Wed, 6 Apr 2016 17:48:50 +0100
Subject: [Freeswitch-users] docker / NAT troubles..
Message-ID: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>

HI,

I?m trying to get FS running in Docker, which largely was pain free (i know, i know, VMs, etc), but I can?t get my head around what is going on with RTP. Ive set ext-rtp-ip and it seems to be taking affect:

freeswitch at 7ad22635059e> sofia status profile internal
=================================================================================================
Name             	internal
Domain Name      	N/A
Auto-NAT         	false
DBName           	sofia_reg_internal
Pres Hosts       	172.17.0.5,172.17.0.5
Dialplan         	XML
Context          	trusted
Challenge Realm  	auto_from
RTP-IP           	172.17.0.5
Ext-RTP-IP       	192.168.1.168
SIP-IP           	172.17.0.5
Ext-SIP-IP       	192.168.1.168
URL              	sip:mod_sofia at 192.168.1.168:5060
BIND-URL         	sip:mod_sofia at 192.168.1.168:5060;maddr=172.17.0.5;transport=udp,tcp
HOLD-MUSIC       	local_stream://moh
OUTBOUND-PROXY   	N/A
CODECS IN        	G729,PCMU,PCMA
CODECS OUT       	G729,PCMU,PCMA
TEL-EVENT        	101
DTMF-MODE        	none
CNG              	13
SESSION-TO       	0
MAX-DIALOG       	0
NOMEDIA          	false
LATE-NEG         	true
PROXY-MEDIA      	false
ZRTP-PASSTHRU    	true
AGGRESSIVENAT    	false
CALLS-IN         	2
FAILED-CALLS-IN  	2
CALLS-OUT        	0
FAILED-CALLS-OUT 	0
REGISTRATIONS    	0



but when a call is placed i seems to be incorrect in the SDP

2016-04-06 16:29:49.011107 [DEBUG] mod_sofia.c:2353 Ring SDP:
v=0
o=FreeSWITCH 1459942605 1459942606 IN IP4 172.17.0.5
s=FreeSWITCH
c=IN IP4 172.17.0.5
t=0 0
m=audio 17584 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Shouldn?t the SDP reflect the Ext-RTP-IP ?

Im sure i?ve missed some sort of config setting or have gone snow blind!.
fs version is FreeSWITCH (Version 1.6.7 -14-d38d065 64bit)

Any ideas will be greatly received.
Thanks
Oz.
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From omortimer at gmail.com  Wed Apr  6 20:48:50 2016
From: omortimer at gmail.com (Oz Mortimer)
Date: Wed, 6 Apr 2016 17:48:50 +0100
Subject: [Freeswitch-users] docker / NAT troubles..
Message-ID: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>

HI,

I?m trying to get FS running in Docker, which largely was pain free (i know, i know, VMs, etc), but I can?t get my head around what is going on with RTP. Ive set ext-rtp-ip and it seems to be taking affect:

freeswitch at 7ad22635059e> sofia status profile internal
=================================================================================================
Name             	internal
Domain Name      	N/A
Auto-NAT         	false
DBName           	sofia_reg_internal
Pres Hosts       	172.17.0.5,172.17.0.5
Dialplan         	XML
Context          	trusted
Challenge Realm  	auto_from
RTP-IP           	172.17.0.5
Ext-RTP-IP       	192.168.1.168
SIP-IP           	172.17.0.5
Ext-SIP-IP       	192.168.1.168
URL              	sip:mod_sofia at 192.168.1.168:5060
BIND-URL         	sip:mod_sofia at 192.168.1.168:5060;maddr=172.17.0.5;transport=udp,tcp
HOLD-MUSIC       	local_stream://moh
OUTBOUND-PROXY   	N/A
CODECS IN        	G729,PCMU,PCMA
CODECS OUT       	G729,PCMU,PCMA
TEL-EVENT        	101
DTMF-MODE        	none
CNG              	13
SESSION-TO       	0
MAX-DIALOG       	0
NOMEDIA          	false
LATE-NEG         	true
PROXY-MEDIA      	false
ZRTP-PASSTHRU    	true
AGGRESSIVENAT    	false
CALLS-IN         	2
FAILED-CALLS-IN  	2
CALLS-OUT        	0
FAILED-CALLS-OUT 	0
REGISTRATIONS    	0



but when a call is placed i seems to be incorrect in the SDP

2016-04-06 16:29:49.011107 [DEBUG] mod_sofia.c:2353 Ring SDP:
v=0
o=FreeSWITCH 1459942605 1459942606 IN IP4 172.17.0.5
s=FreeSWITCH
c=IN IP4 172.17.0.5
t=0 0
m=audio 17584 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Shouldn?t the SDP reflect the Ext-RTP-IP ?

Im sure i?ve missed some sort of config setting or have gone snow blind!.
fs version is FreeSWITCH (Version 1.6.7 -14-d38d065 64bit)

Any ideas will be greatly received.
Thanks
Oz.
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From mike at jerris.com  Wed Apr  6 20:58:01 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Apr 2016 12:58:01 -0400
Subject: [Freeswitch-users] docker / NAT troubles..
In-Reply-To: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>
References: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>
Message-ID: <AE01A0C6-615F-44F1-B74A-67916833B156@jerris.com>

you are natting from one rfc1918 address space to another?  If so, all the default nat acl's will be wrong, and you will have to make your own acl's that match your network environment.

> On Apr 6, 2016, at 12:48 PM, Oz Mortimer <omortimer at gmail.com> wrote:
> 
> HI,
> 
> I?m trying to get FS running in Docker, which largely was pain free (i know, i know, VMs, etc), but I can?t get my head around what is going on with RTP. Ive set ext-rtp-ip and it seems to be taking affect:
> 
> freeswitch at 7ad22635059e> sofia status profile internal
> =================================================================================================
> Name             	internal
> Domain Name      	N/A
> Auto-NAT         	false
> DBName           	sofia_reg_internal
> Pres Hosts       	172.17.0.5,172.17.0.5
> Dialplan         	XML
> Context          	trusted
> Challenge Realm  	auto_from
> RTP-IP           	172.17.0.5
> Ext-RTP-IP       	192.168.1.168
> SIP-IP           	172.17.0.5
> Ext-SIP-IP       	192.168.1.168
> URL              	sip:mod_sofia at 192.168.1.168:5060 <sip:mod_sofia at 192.168.1.168:5060>
> BIND-URL         	sip:mod_sofia at 192.168.1.168:5060;maddr= <sip:mod_sofia at 192.168.1.168:5060;maddr=>172.17.0.5;transport=udp,tcp
> HOLD-MUSIC       	local_stream://moh
> OUTBOUND-PROXY   	N/A
> CODECS IN        	G729,PCMU,PCMA
> CODECS OUT       	G729,PCMU,PCMA
> TEL-EVENT        	101
> DTMF-MODE        	none
> CNG              	13
> SESSION-TO       	0
> MAX-DIALOG       	0
> NOMEDIA          	false
> LATE-NEG         	true
> PROXY-MEDIA      	false
> ZRTP-PASSTHRU    	true
> AGGRESSIVENAT    	false
> CALLS-IN         	2
> FAILED-CALLS-IN  	2
> CALLS-OUT        	0
> FAILED-CALLS-OUT 	0
> REGISTRATIONS    	0
> 
> 
> 
> but when a call is placed i seems to be incorrect in the SDP
> 
> 2016-04-06 16:29:49.011107 [DEBUG] mod_sofia.c:2353 Ring SDP:
> v=0
> o=FreeSWITCH 1459942605 1459942606 IN IP4 172.17.0.5
> s=FreeSWITCH
> c=IN IP4 172.17.0.5
> t=0 0
> m=audio 17584 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
> 
> Shouldn?t the SDP reflect the Ext-RTP-IP ?
> 
> Im sure i?ve missed some sort of config setting or have gone snow blind!.
> fs version is FreeSWITCH (Version 1.6.7 -14-d38d065 64bit)
> 
> Any ideas will be greatly received.
> Thanks
> Oz.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Apr  6 20:59:53 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Apr 2016 11:59:53 -0500
Subject: [Freeswitch-users] Record and live stream WAV to HTTP server
In-Reply-To: <CAKsJLpY0jtfa6MBhVHHQTgOA_RC_nQv35b7DbUa99w6gCk635w@mail.gmail.com>
References: <CAKsJLpacO2KtiuNOOKm-Sck7PKKqZmr9pD2D=s0VrU2TnBxsMw@mail.gmail.com>
	<CAHtxdDcOvfJHk7CFL5edBkDC6DU05b-w=q57GMnyMyiM1GQqAg@mail.gmail.com>
	<CAKsJLpY0UGNoDC1E-Svnv6ULvaveLT1kH4jj9dmH2-Bn1yOJKQ@mail.gmail.com>
	<CAHtxdDc-UM0WhimHVRSaChCRyc+_rX4YGOLZ84bS5rUjk14c_Q@mail.gmail.com>
	<CAKsJLpbwDCK00MqGaGcFotr9LeJa9qUES9Q4ieujtSRc600VYw@mail.gmail.com>
	<CAHtxdDcGCOGtEZbCQbgna6h30KZ4_1ib5=1qY98f=X50+UHrog@mail.gmail.com>
	<CAKsJLpa83aWDBMFuZaYeMFVR6XeN=2uQqwcuTXAtqTY3szckSg@mail.gmail.com>
	<E6948933-09AC-4A48-A0C9-70E206EED453@jerris.com>
	<CAKsJLpavsm2N7odd2xw6CP6zuBxhLR_QEy3acEPukujVAPU5HA@mail.gmail.com>
	<CAHtxdDfQGXWAUopsmOUO3hTV2A91i9t=9ss5P3+GtnxPtwwsfg@mail.gmail.com>
	<CAEwTsAid_21ysmiu3E+iUdvazMGpLpb0KOpVCOxK6WaurmnWEA@mail.gmail.com>
	<CAKsJLpYAR333mU6VuD2AjxepbPJ4quyzGKpW2ziOmMLds6rgsg@mail.gmail.com>
	<731C9BFA-0101-42F3-B368-17F8AEFD7580@jerris.com>
	<CAKsJLpaCq3=ifM_FFav2-iE9X7f00FRHppiRjO25Yc-=rKeuew@mail.gmail.com>
	<CAKbxfG-auQAvHZrztzgnF_4O363tK43VfkFC1xuwZgjmDKiaew@mail.gmail.com>
	<CAKsJLpY0jtfa6MBhVHHQTgOA_RC_nQv35b7DbUa99w6gCk635w@mail.gmail.com>
Message-ID: <CAKbxfG8d49ghhs-7UWs9+ZkyTS55pWVX25ts2Aco=Xnu0Dsb9Q@mail.gmail.com>

On Wed, Apr 6, 2016 at 11:47 AM, Vincent Gire <vincent.gire at gmail.com>
wrote:

>
>
> On Wed, Apr 6, 2016 at 6:10 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Firstly,
>>
>> You cannot stream WAV.  Its not possible.  WAV is not a streaming format,
>> its a container.  Its basically a raw audio data stream with a header
>> explaining the characteristics of the stream.
>>
>
> Yes. Sorry for the language abuse and thank you for clearing this out.
>
>
>>
>>
>> Next,
>>
>> You need to compare your specific needs to the goal of the software to be
>> flexible and usable to all.
>>
>> The reason the speech interface was suggested is that the point of the
>> speech interface is precisely to tap into live audio streams and perform
>> asynchronous ASR operations while the original call does whatever it wants
>> like talk to another caller or listen to a file playing etc.
>>
>> What you envision is more of a synchronous single instruction to record a
>> file in realtime to the remote server over httapi.
>>
>
> Exactly.
>
>
>> The considerations to implement this would be what format to send the
>> audio in. The source audio could be any combination of sample rate, channel
>> count and encoding format.
>> To record we would translate the encoded format into signed linear at the
>> original input sample rate and channels.  These variances must be
>> communicated before you start sending the data so it would be a matter of
>> figuring out an existing standard or implement a proprietary one.
>>
>
> I am not sure I am following here.
> I might again have misused the word streaming.
> I was thinking that when we specify a http target for a record in
> mod_httapi, it could start sending the corresponding file (WAV or other) to
> the HTTP server as chunked data as soon as the record starts.
> In the case of a WAV target, the WAV header would be sent first as part of
> the first HTTP data chunks and the raw audio data stream would follow in
> the next data chunks.
> Does it make sense ?
>


I agree that you are not following as I gave a relatively lengthy and
specific response.  I know exactly what you are asking for so there is no
need to re-explain.
The way you ask for it is very specific to your own needs so it would not
make a very useful feature in FS.

Are you asking for someone to make this feature for you or are you asking
for advice on how you can implement it?
I think you are underestimating the complexity and potential scalability of
the overall project you are doing (not just the FS portion) and you should
review your entire architecture.




>
>
>>
>>
>>
>>
>> On Wed, Apr 6, 2016 at 10:42 AM, Vincent Gire <vincent.gire at gmail.com>
>> wrote:
>>
>>> We already handle the multiple providers on the HTTP server.
>>> So you would recommend to implement a module against the speech
>>> interface streaming to the HTTP server ?
>>> Wouldn't it be easier (and less redundant) to add streaming capabilities
>>> to mod_httpapi ?
>>>
>>> On Wed, Apr 6, 2016 at 12:22 AM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> The best way to handle this is probably to write a custom module
>>>> against the speech interface that sends the streams like you are
>>>> describing, and supports multiple providers.  How exactly do you go about
>>>> sending to multiple at the same time and combining the results, as this
>>>> might actually be the trickiest part as youll need to integrate that into a
>>>> module that uses the speech interface to be able to at all sanely handle
>>>> that in freeswitch
>>>>
>>>> On Apr 5, 2016, at 1:24 PM, Vincent Gire <vincent.gire at gmail.com>
>>>> wrote:
>>>>
>>>> We use multiple ones to compare confidence.
>>>> They all support chunk transfer and provide a HTTP transaction similar
>>>> to :
>>>> http://developer.att.com/apis/speech/docs/v3
>>>> Look at 3/ Make API Calls
>>>>
>>>> On Tue, Apr 5, 2016 at 7:14 PM, Michael Jerris <mike at jerris.com> wrote:
>>>>
>>>>> Unimrcp isn't going to provide the interface he's talking about.  Can
>>>>> I ask which engine this is?
>>>>>
>>>>>
>>>>> On Tuesday, April 5, 2016, Sergey Safarov <s.safarov at gmail.com> wrote:
>>>>>
>>>>>> Look at http://www.unimrcp.org/ and
>>>>>> https://wiki.freeswitch.org/wiki/Mod_unimrcp
>>>>>>
>>>>>> On Tue, Apr 5, 2016, 19:38 Vincent Gire <vincent.gire at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> We are building an IVR completely driven by ASR.
>>>>>>> ASR is performed in a distant location by a HTTP service (supporting
>>>>>>> chunked transfer) and adds an incompressible latency. We would like to
>>>>>>> stream the record to the ASR service as soon as it starts to reduce the
>>>>>>> overall latency before response.
>>>>>>> Does it make sense ?
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Apr 5, 2016 at 5:39 PM, Michael Jerris <mike at jerris.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> I would stay away from mod_vlc.  Its audio portions with recording
>>>>>>>> have known issues.  We do use the rtmp streaming in mod_av heavily but
>>>>>>>> thats obviously not wav.  Can you explain a bit more why you have this
>>>>>>>> requirement?
>>>>>>>>
>>>>>>>> On Apr 5, 2016, at 5:36 AM, Vincent Gire <vincent.gire at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>> Ok thanks.
>>>>>>>> It looks promising !
>>>>>>>> I'll dig into mod_vlc.
>>>>>>>>
>>>>>>>> Best
>>>>>>>>
>>>>>>>> Vincent
>>>>>>>>
>>>>>>>> On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <
>>>>>>>> s.safarov at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Think is requred streaming feature of freeswitch.
>>>>>>>>> Look at mod_esf
>>>>>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/mod_esf>
>>>>>>>>> and mod_vlc <https://wiki.freeswitch.org/wiki/Mod_vlc>
>>>>>>>>> Instruction to compile mod_vlc on provided link is to old but
>>>>>>>>> helpfull to undestand how to stream media to http server.
>>>>>>>>>
>>>>>>>>> For compiling mod_vlc please use vlc repo
>>>>>>>>> <https://freeswitch.org/stash/projects/SD/repos/vlc/browse> and centos
>>>>>>>>> instruction
>>>>>>>>> <https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary>
>>>>>>>>> .
>>>>>>>>> After you intall vlc, then you can enable mod_vlc module in
>>>>>>>>> freeswitch sources(SPEC file) and compile freeswitch.
>>>>>>>>>
>>>>>>>>> Sergey
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> ??, 5 ???. 2016 ?. ? 10:31, Vincent Gire <vincent.gire at gmail.com>:
>>>>>>>>>
>>>>>>>>>> webdav, mod_http_cache or mod_httapi all results in sending the
>>>>>>>>>> recording only *after* it is complete.
>>>>>>>>>> They all write the recording to a file, wait for the recording
>>>>>>>>>> to complete and the file to close and then send it over HTTP.
>>>>>>>>>>
>>>>>>>>>> I would like to start sending the recording to the remove server
>>>>>>>>>> as soon as it starts (max 1 sec latency).
>>>>>>>>>> mod_http_cache or mod_httapi would be perfect if they were
>>>>>>>>>> streaming the recording like mod_shout.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <
>>>>>>>>>> s.safarov at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Input/output latency is not problem.  I use Kazoo on my servers
>>>>>>>>>>> and call recording is stored to database during 5 seconds after hangup.
>>>>>>>>>>> What is broken in your case if save file using webdav or
>>>>>>>>>>> http_cache?
>>>>>>>>>>>
>>>>>>>>>>> On Mon, Apr 4, 2016, 21:10 Vincent Gire <vincent.gire at gmail.com>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Hello Sergey,
>>>>>>>>>>>>
>>>>>>>>>>>> Thank you for your answer.
>>>>>>>>>>>> I've looked into webdav mounted filesystem.
>>>>>>>>>>>>
>>>>>>>>>>>> Unfortunately, most WebDav clients (especially davfs2 on
>>>>>>>>>>>> debian) do a lot of buffering, caching and even lock-null requests (lock a
>>>>>>>>>>>> non existent resource before writing to it). I also suspect that they wait
>>>>>>>>>>>> for the end of the write operation.
>>>>>>>>>>>> The result is a latency of a few seconds witch is not much
>>>>>>>>>>>> better than what I achieve with mod_shout if I transcode the MP3.
>>>>>>>>>>>>
>>>>>>>>>>>> Any other idea ?
>>>>>>>>>>>>
>>>>>>>>>>>> Thank you !
>>>>>>>>>>>>
>>>>>>>>>>>> Best regards
>>>>>>>>>>>>
>>>>>>>>>>>> Vincent
>>>>>>>>>>>>
>>>>>>>>>>>> On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <
>>>>>>>>>>>> s.safarov at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Please look at webdav mounted filesystem.
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Sun, Apr 3, 2016, 19:17 Vincent Gire <
>>>>>>>>>>>>> vincent.gire at gmail.com> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Hi all,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Thank you to all contributing to FreeSWITCH !
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> I'm working on a IVR project where logic is implemented on a
>>>>>>>>>>>>>> HTTP server.
>>>>>>>>>>>>>> We are leaving Twilio because we now need to record and live
>>>>>>>>>>>>>> stream the session to the HTTP server in WAV format (chunked transfer
>>>>>>>>>>>>>> encoding).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> *mod_httapi* looks great (HT TAPI very similar to Twilio's)
>>>>>>>>>>>>>> but it seems that the records are first saved to disk before there are sent
>>>>>>>>>>>>>> to the server as chunked data.
>>>>>>>>>>>>>> We need the transfer to start as soon as the recording starts.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> *mod_shout* does start the request almost as the records
>>>>>>>>>>>>>> starts but it does not support WAV file and shout:// is not exactly a HTTP
>>>>>>>>>>>>>> request (SOURCE method instead of PUT).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Is there a way to use these modules to achieve our goal ?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> If not, we are willing to author a specific module or rather
>>>>>>>>>>>>>> contribute to the existing ones.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> We've identified two approaches:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>    1. From *mod_httapi* Modify mod_httapi to directly stream
>>>>>>>>>>>>>>    the record instead of completely saving it to disk before the HTTP chunked
>>>>>>>>>>>>>>    transfer starts.
>>>>>>>>>>>>>>    This seems the most logical but with more than 3000
>>>>>>>>>>>>>>    lines, mod_httapi does not seem to be the easiest module to build upon for
>>>>>>>>>>>>>>    newcomers!
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>    2. From *mod_shout*
>>>>>>>>>>>>>>       1. Modify libshoot to replace the custom SOURCE method
>>>>>>>>>>>>>>       with standard HTTP PUT method
>>>>>>>>>>>>>>       2. Modify mod_shout to support wav files
>>>>>>>>>>>>>>       3. Implement our IVR in script (javascript/lua)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> What do you think ?
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Thank you for your help.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Cheers,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Vincent Gire
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Vincent Gire
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Vincent Gire
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>>
>> ? http://freeswitch.org/  ? http://cluecon.com/  ?
>> http://twitter.com/FreeSWITCH
>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
>> <http://freeswitch.org/g+>*
>>
>> ClueCon Weekly Development Call
>> ? sip:888 at conference.freeswitch.org  ? +19193869900
>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Vincent Gire
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From omortimer at gmail.com  Wed Apr  6 21:31:45 2016
From: omortimer at gmail.com (Oz Mortimer)
Date: Wed, 6 Apr 2016 18:31:45 +0100
Subject: [Freeswitch-users] docker / NAT troubles..
In-Reply-To: <AE01A0C6-615F-44F1-B74A-67916833B156@jerris.com>
References: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>
	<AE01A0C6-615F-44F1-B74A-67916833B156@jerris.com>
Message-ID: <67B6621A-1705-4595-85F1-429F31819DC6@gmail.com>

Hi,
Thanks for the reply - I wish I understood it - but I don?t ;)
Yes, the natting is between one rfc1918 address space to another.

Based on your reply I tried

	<param name="apply-nat-acl" value="damnnat?/>

where acl.conf.xml has

 <list name="damnnat" default="allow">
      <node type="allow" cidr="192.168.1.0/24"/>
      <node type="allow" cidr="172.17.0.0/24"/>
    </list>

To no avail!. can you give me a pointer to what I need to change and where?

Thanks
Oz.

> On 6 Apr 2016, at 17:58, Michael Jerris <mike at jerris.com> wrote:
> 
> you are natting from one rfc1918 address space to another?  If so, all the default nat acl's will be wrong, and you will have to make your own acl's that match your network environment.
> 
>> On Apr 6, 2016, at 12:48 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>> 
>> HI,
>> 
>> I?m trying to get FS running in Docker, which largely was pain free (i know, i know, VMs, etc), but I can?t get my head around what is going on with RTP. Ive set ext-rtp-ip and it seems to be taking affect:
>> 
>> freeswitch at 7ad22635059e> sofia status profile internal
>> =================================================================================================
>> Name             	internal
>> Domain Name      	N/A
>> Auto-NAT         	false
>> DBName           	sofia_reg_internal
>> Pres Hosts       	172.17.0.5,172.17.0.5
>> Dialplan         	XML
>> Context          	trusted
>> Challenge Realm  	auto_from
>> RTP-IP           	172.17.0.5
>> Ext-RTP-IP       	192.168.1.168
>> SIP-IP           	172.17.0.5
>> Ext-SIP-IP       	192.168.1.168
>> URL              	sip:mod_sofia at 192.168.1.168:5060 <sip:mod_sofia at 192.168.1.168:5060>
>> BIND-URL         	sip:mod_sofia at 192.168.1.168:5060;maddr= <sip:mod_sofia at 192.168.1.168:5060;maddr=>172.17.0.5;transport=udp,tcp
>> HOLD-MUSIC       	local_stream://moh
>> OUTBOUND-PROXY   	N/A
>> CODECS IN        	G729,PCMU,PCMA
>> CODECS OUT       	G729,PCMU,PCMA
>> TEL-EVENT        	101
>> DTMF-MODE        	none
>> CNG              	13
>> SESSION-TO       	0
>> MAX-DIALOG       	0
>> NOMEDIA          	false
>> LATE-NEG         	true
>> PROXY-MEDIA      	false
>> ZRTP-PASSTHRU    	true
>> AGGRESSIVENAT    	false
>> CALLS-IN         	2
>> FAILED-CALLS-IN  	2
>> CALLS-OUT        	0
>> FAILED-CALLS-OUT 	0
>> REGISTRATIONS    	0
>> 
>> 
>> 
>> but when a call is placed i seems to be incorrect in the SDP
>> 
>> 2016-04-06 16:29:49.011107 [DEBUG] mod_sofia.c:2353 Ring SDP:
>> v=0
>> o=FreeSWITCH 1459942605 1459942606 IN IP4 172.17.0.5
>> s=FreeSWITCH
>> c=IN IP4 172.17.0.5
>> t=0 0
>> m=audio 17584 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>> 
>> Shouldn?t the SDP reflect the Ext-RTP-IP ?
>> 
>> Im sure i?ve missed some sort of config setting or have gone snow blind!.
>> fs version is FreeSWITCH (Version 1.6.7 -14-d38d065 64bit)
>> 
>> Any ideas will be greatly received.
>> Thanks
>> Oz.
>> _________________________________________________________________________
>> 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 mike at jerris.com  Wed Apr  6 21:39:03 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Apr 2016 13:39:03 -0400
Subject: [Freeswitch-users] docker / NAT troubles..
In-Reply-To: <67B6621A-1705-4595-85F1-429F31819DC6@gmail.com>
References: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>
	<AE01A0C6-615F-44F1-B74A-67916833B156@jerris.com>
	<67B6621A-1705-4595-85F1-429F31819DC6@gmail.com>
Message-ID: <45895411-F048-4DA9-B0F4-579E0A5E0E29@jerris.com>

The default acl's treat all rfc1918 addresses as internal.  you'll need to make one that treats your external addresses as external even tho they are rfc1918.  Why are you natting from one private address to another?  Its a very strange implementation


> On Apr 6, 2016, at 1:31 PM, Oz Mortimer <omortimer at gmail.com> wrote:
> 
> Hi,
> Thanks for the reply - I wish I understood it - but I don?t ;)
> Yes, the natting is between one rfc1918 address space to another.
> 
> Based on your reply I tried
> 
> 	<param name="apply-nat-acl" value="damnnat?/>
> 
> where acl.conf.xml has
> 
>  <list name="damnnat" default="allow">
>       <node type="allow" cidr="192.168.1.0/24"/>
>       <node type="allow" cidr="172.17.0.0/24"/>
>     </list>
> 
> To no avail!. can you give me a pointer to what I need to change and where?
> 
> Thanks
> Oz.
> 
>> On 6 Apr 2016, at 17:58, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> 
>> you are natting from one rfc1918 address space to another?  If so, all the default nat acl's will be wrong, and you will have to make your own acl's that match your network environment.
>> 
>>> On Apr 6, 2016, at 12:48 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>>> 
>>> HI,
>>> 
>>> I?m trying to get FS running in Docker, which largely was pain free (i know, i know, VMs, etc), but I can?t get my head around what is going on with RTP. Ive set ext-rtp-ip and it seems to be taking affect:
>>> 
>>> freeswitch at 7ad22635059e> sofia status profile internal
>>> =================================================================================================
>>> Name             	internal
>>> Domain Name      	N/A
>>> Auto-NAT         	false
>>> DBName           	sofia_reg_internal
>>> Pres Hosts       	172.17.0.5,172.17.0.5
>>> Dialplan         	XML
>>> Context          	trusted
>>> Challenge Realm  	auto_from
>>> RTP-IP           	172.17.0.5
>>> Ext-RTP-IP       	192.168.1.168
>>> SIP-IP           	172.17.0.5
>>> Ext-SIP-IP       	192.168.1.168
>>> URL              	sip:mod_sofia at 192.168.1.168:5060 <sip:mod_sofia at 192.168.1.168:5060>
>>> BIND-URL         	sip:mod_sofia at 192.168.1.168:5060;maddr= <sip:mod_sofia at 192.168.1.168:5060;maddr=>172.17.0.5;transport=udp,tcp
>>> HOLD-MUSIC       	local_stream://moh
>>> OUTBOUND-PROXY   	N/A
>>> CODECS IN        	G729,PCMU,PCMA
>>> CODECS OUT       	G729,PCMU,PCMA
>>> TEL-EVENT        	101
>>> DTMF-MODE        	none
>>> CNG              	13
>>> SESSION-TO       	0
>>> MAX-DIALOG       	0
>>> NOMEDIA          	false
>>> LATE-NEG         	true
>>> PROXY-MEDIA      	false
>>> ZRTP-PASSTHRU    	true
>>> AGGRESSIVENAT    	false
>>> CALLS-IN         	2
>>> FAILED-CALLS-IN  	2
>>> CALLS-OUT        	0
>>> FAILED-CALLS-OUT 	0
>>> REGISTRATIONS    	0
>>> 
>>> 
>>> 
>>> but when a call is placed i seems to be incorrect in the SDP
>>> 
>>> 2016-04-06 16:29:49.011107 [DEBUG] mod_sofia.c:2353 Ring SDP:
>>> v=0
>>> o=FreeSWITCH 1459942605 1459942606 IN IP4 172.17.0.5
>>> s=FreeSWITCH
>>> c=IN IP4 172.17.0.5
>>> t=0 0
>>> m=audio 17584 RTP/AVP 8 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> a=sendrecv
>>> 
>>> Shouldn?t the SDP reflect the Ext-RTP-IP ?
>>> 
>>> Im sure i?ve missed some sort of config setting or have gone snow blind!.
>>> fs version is FreeSWITCH (Version 1.6.7 -14-d38d065 64bit)
>>> 
>>> Any ideas will be greatly received.
>>> Thanks
>>> Oz.
>>> _________________________________________________________________________
>>> 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 omortimer at gmail.com  Wed Apr  6 21:49:24 2016
From: omortimer at gmail.com (Oz Mortimer)
Date: Wed, 6 Apr 2016 18:49:24 +0100
Subject: [Freeswitch-users] docker / NAT troubles..
In-Reply-To: <45895411-F048-4DA9-B0F4-579E0A5E0E29@jerris.com>
References: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>
	<AE01A0C6-615F-44F1-B74A-67916833B156@jerris.com>
	<67B6621A-1705-4595-85F1-429F31819DC6@gmail.com>
	<45895411-F048-4DA9-B0F4-579E0A5E0E29@jerris.com>
Message-ID: <17C2D436-F1A3-42B4-B742-524797B99C8B@gmail.com>

I know! I've never come across it! It's what I seem to have to do when using a docker container - unless someone knows different?
Docker uses a vm, which is on the network 192.168.. But the container has an IP of 172.17..
I'm no docker expert and in an ideal world the container should have a 192.168.. Address, but I can't find a way to make that happen.
Maybe I'm asking in the wrong mailing list - could be a question for docker. Either way, I'd like to figure out what I'm doing wrong! 
Nb. This is no way going to be a production setup - it's a development setup.

I "think" I understand what I need to do..
Thanks
Oz
> On 6 Apr 2016, at 18:39, Michael Jerris <mike at jerris.com> wrote:
> 
> The default acl's treat all rfc1918 addresses as internal.  you'll need to make one that treats your external addresses as external even tho they are rfc1918.  Why are you natting from one private address to another?  Its a very strange implementation
> 
> 
>> On Apr 6, 2016, at 1:31 PM, Oz Mortimer <omortimer at gmail.com> wrote:
>> 
>> Hi,
>> Thanks for the reply - I wish I understood it - but I don?t ;)
>> Yes, the natting is between one rfc1918 address space to another.
>> 
>> Based on your reply I tried
>> 
>> 	<param name="apply-nat-acl" value="damnnat?/>
>> 
>> where acl.conf.xml has
>> 
>>  <list name="damnnat" default="allow">
>>       <node type="allow" cidr="192.168.1.0/24"/>
>>       <node type="allow" cidr="172.17.0.0/24"/>
>>     </list>
>> 
>> To no avail!. can you give me a pointer to what I need to change and where?
>> 
>> Thanks
>> Oz.
>> 
>>> On 6 Apr 2016, at 17:58, Michael Jerris <mike at jerris.com> wrote:
>>> 
>>> you are natting from one rfc1918 address space to another?  If so, all the default nat acl's will be wrong, and you will have to make your own acl's that match your network environment.
>>> 
>>>> On Apr 6, 2016, at 12:48 PM, Oz Mortimer <omortimer at gmail.com> wrote:
>>>> 
>>>> HI,
>>>> 
>>>> I?m trying to get FS running in Docker, which largely was pain free (i know, i know, VMs, etc), but I can?t get my head around what is going on with RTP. Ive set ext-rtp-ip and it seems to be taking affect:
>>>> 
>>>> freeswitch at 7ad22635059e> sofia status profile internal
>>>> =================================================================================================
>>>> Name             	internal
>>>> Domain Name      	N/A
>>>> Auto-NAT         	false
>>>> DBName           	sofia_reg_internal
>>>> Pres Hosts       	172.17.0.5,172.17.0.5
>>>> Dialplan         	XML
>>>> Context          	trusted
>>>> Challenge Realm  	auto_from
>>>> RTP-IP           	172.17.0.5
>>>> Ext-RTP-IP       	192.168.1.168
>>>> SIP-IP           	172.17.0.5
>>>> Ext-SIP-IP       	192.168.1.168
>>>> URL              	sip:mod_sofia at 192.168.1.168:5060
>>>> BIND-URL         	sip:mod_sofia at 192.168.1.168:5060;maddr=172.17.0.5;transport=udp,tcp
>>>> HOLD-MUSIC       	local_stream://moh
>>>> OUTBOUND-PROXY   	N/A
>>>> CODECS IN        	G729,PCMU,PCMA
>>>> CODECS OUT       	G729,PCMU,PCMA
>>>> TEL-EVENT        	101
>>>> DTMF-MODE        	none
>>>> CNG              	13
>>>> SESSION-TO       	0
>>>> MAX-DIALOG       	0
>>>> NOMEDIA          	false
>>>> LATE-NEG         	true
>>>> PROXY-MEDIA      	false
>>>> ZRTP-PASSTHRU    	true
>>>> AGGRESSIVENAT    	false
>>>> CALLS-IN         	2
>>>> FAILED-CALLS-IN  	2
>>>> CALLS-OUT        	0
>>>> FAILED-CALLS-OUT 	0
>>>> REGISTRATIONS    	0
>>>> 
>>>> 
>>>> 
>>>> but when a call is placed i seems to be incorrect in the SDP
>>>> 
>>>> 2016-04-06 16:29:49.011107 [DEBUG] mod_sofia.c:2353 Ring SDP:
>>>> v=0
>>>> o=FreeSWITCH 1459942605 1459942606 IN IP4 172.17.0.5
>>>> s=FreeSWITCH
>>>> c=IN IP4 172.17.0.5
>>>> t=0 0
>>>> m=audio 17584 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>> a=sendrecv
>>>> 
>>>> Shouldn?t the SDP reflect the Ext-RTP-IP ?
>>>> 
>>>> Im sure i?ve missed some sort of config setting or have gone snow blind!.
>>>> fs version is FreeSWITCH (Version 1.6.7 -14-d38d065 64bit)
>>>> 
>>>> Any ideas will be greatly received.
>>>> Thanks
>>>> Oz.
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services: 
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
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From nneul at mst.edu  Wed Apr  6 21:49:42 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Wed, 6 Apr 2016 12:49:42 -0500
Subject: [Freeswitch-users] docker / NAT troubles..
In-Reply-To: <45895411-F048-4DA9-B0F4-579E0A5E0E29@jerris.com>
References: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>
	<AE01A0C6-615F-44F1-B74A-67916833B156@jerris.com>
	<67B6621A-1705-4595-85F1-429F31819DC6@gmail.com>
	<45895411-F048-4DA9-B0F4-579E0A5E0E29@jerris.com>
Message-ID: <57054C36.1000909@mst.edu>

Sounds like a running FS in a docker container (rfc1918 addrs) on a host which itself is assigned a rfc1918 addr.

-- Nathan

On 04/06/2016 12:39 PM, Michael Jerris wrote:
> The default acl's treat all rfc1918 addresses as internal.  you'll need to make one that treats your external addresses
> as external even tho they are rfc1918.  Why are you natting from one private address to another?  Its a very strange
> implementation
>
>
>> On Apr 6, 2016, at 1:31 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>>
>> Hi,
>> Thanks for the reply - I wish I understood it - but I don?t ;)
>> Yes, the natting is between one rfc1918 address space to another.
>>
>> Based on your reply I tried
>>
>> <param name="apply-nat-acl" value="damnnat?/>
>>
>> where acl.conf.xml has
>>
>>  <list name="damnnat" default="allow">
>>       <node type="allow" cidr="192.168.1.0/24"/>
>>       <node type="allow" cidr="172.17.0.0/24"/>
>>     </list>
>>
>> To no avail!. can you give me a pointer to what I need to change and where?
>>
>> Thanks
>> Oz.
>>
>>> On 6 Apr 2016, at 17:58, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>>
>>> you are natting from one rfc1918 address space to another?  If so, all the default nat acl's will be wrong, and you
>>> will have to make your own acl's that match your network environment.
>>>
>>>> On Apr 6, 2016, at 12:48 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>>>>
>>>> HI,
>>>>
>>>> I?m trying to get FS running in Docker, which largely was pain free (i know, i know, VMs, etc), but I can?t get my
>>>> head around what is going on with RTP. Ive set ext-rtp-ip and it seems to be taking affect:
>>>>
>>>> freeswitch at 7ad22635059e> sofia status profile internal
>>>> =================================================================================================
>>>> Name internal
>>>> Domain Name N/A
>>>> Auto-NAT false
>>>> DBName sofia_reg_internal
>>>> Pres Hosts 172.17.0.5,172.17.0.5
>>>> Dialplan XML
>>>> Context trusted
>>>> Challenge Realm auto_from
>>>> RTP-IP 172.17.0.5
>>>> Ext-RTP-IP 192.168.1.168
>>>> SIP-IP 172.17.0.5
>>>> Ext-SIP-IP 192.168.1.168
>>>> URL sip:mod_sofia at 192.168.1.168:5060
>>>> BIND-URL sip:mod_sofia at 192.168.1.168:5060;maddr=172.17.0.5;transport=udp,tcp
>>>> HOLD-MUSIC local_stream://moh
>>>> OUTBOUND-PROXY N/A
>>>> CODECS IN G729,PCMU,PCMA
>>>> CODECS OUT G729,PCMU,PCMA
>>>> TEL-EVENT 101
>>>> DTMF-MODE none
>>>> CNG 13
>>>> SESSION-TO 0
>>>> MAX-DIALOG 0
>>>> NOMEDIA false
>>>> LATE-NEG true
>>>> PROXY-MEDIA false
>>>> ZRTP-PASSTHRU true
>>>> AGGRESSIVENAT false
>>>> CALLS-IN 2
>>>> FAILED-CALLS-IN 2
>>>> CALLS-OUT 0
>>>> FAILED-CALLS-OUT 0
>>>> REGISTRATIONS 0
>>>>
>>>>
>>>>
>>>> but when a call is placed i seems to be incorrect in the SDP
>>>>
>>>> 2016-04-06 16:29:49.011107 [DEBUG] mod_sofia.c:2353 Ring SDP:
>>>> v=0
>>>> o=FreeSWITCH 1459942605 1459942606 IN IP4 172.17.0.5
>>>> s=FreeSWITCH
>>>> c=IN IP4 172.17.0.5
>>>> t=0 0
>>>> m=audio 17584 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>> a=sendrecv
>>>>
>>>> Shouldn?t the SDP reflect the Ext-RTP-IP ?
>>>>
>>>> Im sure i?ve missed some sort of config setting or have gone snow blind!.
>>>> fs version is FreeSWITCH (Version 1.6.7 -14-d38d065 64bit)
>>>>
>>>> Any ideas will be greatly received.
>>>> Thanks
>>>> Oz.
>>>> _________________________________________________________________________
>>>> 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 <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
>

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


From fvillarroel at yahoo.com  Wed Apr  6 21:50:11 2016
From: fvillarroel at yahoo.com (Fernando Villarroel)
Date: Wed, 06 Apr 2016 14:50:11 -0300
Subject: [Freeswitch-users] Verto firefox
Message-ID: <125801.43094.bm@smtp216.mail.bf1.yahoo.com>

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From mike at jerris.com  Wed Apr  6 22:00:27 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 6 Apr 2016 14:00:27 -0400
Subject: [Freeswitch-users] docker / NAT troubles..
In-Reply-To: <17C2D436-F1A3-42B4-B742-524797B99C8B@gmail.com>
References: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>
	<AE01A0C6-615F-44F1-B74A-67916833B156@jerris.com>
	<67B6621A-1705-4595-85F1-429F31819DC6@gmail.com>
	<45895411-F048-4DA9-B0F4-579E0A5E0E29@jerris.com>
	<17C2D436-F1A3-42B4-B742-524797B99C8B@gmail.com>
Message-ID: <828541E3-6097-4932-8C29-C5516EAB45CF@jerris.com>

And your clients are attaching from what network?  Also if your description is correct, then you have internal and external addresses backwards.  Does it actually nat those addresses or is it routed?


> On Apr 6, 2016, at 1:49 PM, Oz Mortimer <omortimer at gmail.com> wrote:
> 
> I know! I've never come across it! It's what I seem to have to do when using a docker container - unless someone knows different?
> Docker uses a vm, which is on the network 192.168.. But the container has an IP of 172.17..
> I'm no docker expert and in an ideal world the container should have a 192.168.. Address, but I can't find a way to make that happen.
> Maybe I'm asking in the wrong mailing list - could be a question for docker. Either way, I'd like to figure out what I'm doing wrong! 
> Nb. This is no way going to be a production setup - it's a development setup.
> 
> I "think" I understand what I need to do..
> Thanks
> Oz
> On 6 Apr 2016, at 18:39, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> 
>> The default acl's treat all rfc1918 addresses as internal.  you'll need to make one that treats your external addresses as external even tho they are rfc1918.  Why are you natting from one private address to another?  Its a very strange implementation
>> 
>> 
>>> On Apr 6, 2016, at 1:31 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>>> 
>>> Hi,
>>> Thanks for the reply - I wish I understood it - but I don?t ;)
>>> Yes, the natting is between one rfc1918 address space to another.
>>> 
>>> Based on your reply I tried
>>> 
>>> 	<param name="apply-nat-acl" value="damnnat?/>
>>> 
>>> where acl.conf.xml has
>>> 
>>>  <list name="damnnat" default="allow">
>>>       <node type="allow" cidr="192.168.1.0/24"/>
>>>       <node type="allow" cidr="172.17.0.0/24"/>
>>>     </list>
>>> 
>>> To no avail!. can you give me a pointer to what I need to change and where?
>>> 
>>> Thanks
>>> Oz.
>>> 
>>>> On 6 Apr 2016, at 17:58, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>>> 
>>>> you are natting from one rfc1918 address space to another?  If so, all the default nat acl's will be wrong, and you will have to make your own acl's that match your network environment.
>>>> 
>>>>> On Apr 6, 2016, at 12:48 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>>>>> 
>>>>> HI,
>>>>> 
>>>>> I?m trying to get FS running in Docker, which largely was pain free (i know, i know, VMs, etc), but I can?t get my head around what is going on with RTP. Ive set ext-rtp-ip and it seems to be taking affect:
>>>>> 
>>>>> freeswitch at 7ad22635059e> sofia status profile internal
>>>>> =================================================================================================
>>>>> Name             	internal
>>>>> Domain Name      	N/A
>>>>> Auto-NAT         	false
>>>>> DBName           	sofia_reg_internal
>>>>> Pres Hosts       	172.17.0.5,172.17.0.5
>>>>> Dialplan         	XML
>>>>> Context          	trusted
>>>>> Challenge Realm  	auto_from
>>>>> RTP-IP           	172.17.0.5
>>>>> Ext-RTP-IP       	192.168.1.168
>>>>> SIP-IP           	172.17.0.5
>>>>> Ext-SIP-IP       	192.168.1.168
>>>>> URL              	sip:mod_sofia at 192.168.1.168:5060 <sip:mod_sofia at 192.168.1.168:5060>
>>>>> BIND-URL         	sip:mod_sofia at 192.168.1.168:5060;maddr= <sip:mod_sofia at 192.168.1.168:5060;maddr=>172.17.0.5;transport=udp,tcp
>>>>> HOLD-MUSIC       	local_stream://moh
>>>>> OUTBOUND-PROXY   	N/A
>>>>> CODECS IN        	G729,PCMU,PCMA
>>>>> CODECS OUT       	G729,PCMU,PCMA
>>>>> TEL-EVENT        	101
>>>>> DTMF-MODE        	none
>>>>> CNG              	13
>>>>> SESSION-TO       	0
>>>>> MAX-DIALOG       	0
>>>>> NOMEDIA          	false
>>>>> LATE-NEG         	true
>>>>> PROXY-MEDIA      	false
>>>>> ZRTP-PASSTHRU    	true
>>>>> AGGRESSIVENAT    	false
>>>>> CALLS-IN         	2
>>>>> FAILED-CALLS-IN  	2
>>>>> CALLS-OUT        	0
>>>>> FAILED-CALLS-OUT 	0
>>>>> REGISTRATIONS    	0
>>>>> 
>>>>> 
>>>>> 
>>>>> but when a call is placed i seems to be incorrect in the SDP
>>>>> 
>>>>> 2016-04-06 16:29:49.011107 [DEBUG] mod_sofia.c:2353 Ring SDP:
>>>>> v=0
>>>>> o=FreeSWITCH 1459942605 1459942606 IN IP4 172.17.0.5
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 172.17.0.5
>>>>> t=0 0
>>>>> m=audio 17584 RTP/AVP 8 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=ptime:20
>>>>> a=sendrecv
>>>>> 
>>>>> Shouldn?t the SDP reflect the Ext-RTP-IP ?
>>>>> 
>>>>> Im sure i?ve missed some sort of config setting or have gone snow blind!.
>>>>> fs version is FreeSWITCH (Version 1.6.7 -14-d38d065 64bit)
>>>>> 
>>>>> Any ideas will be greatly received.
>>>>> Thanks
>>>>> Oz.
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services: 
>>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services: 
>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>> http://www.cluecon.com <http://www.cluecon.com/>
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From omortimer at gmail.com  Wed Apr  6 22:13:58 2016
From: omortimer at gmail.com (Oz Mortimer)
Date: Wed, 6 Apr 2016 19:13:58 +0100
Subject: [Freeswitch-users] docker / NAT troubles..
In-Reply-To: <828541E3-6097-4932-8C29-C5516EAB45CF@jerris.com>
References: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>
	<AE01A0C6-615F-44F1-B74A-67916833B156@jerris.com>
	<67B6621A-1705-4595-85F1-429F31819DC6@gmail.com>
	<45895411-F048-4DA9-B0F4-579E0A5E0E29@jerris.com>
	<17C2D436-F1A3-42B4-B742-524797B99C8B@gmail.com>
	<828541E3-6097-4932-8C29-C5516EAB45CF@jerris.com>
Message-ID: <E57993A6-A251-45FE-BC60-3593DC39837C@gmail.com>

The clients are on the 192. Network.
The network internal to freeswitch is 17. 
Docker assigns the 17. Ips to the container (which fs is running on).

The signalling is fine, but the sdp needs to show the 192.168.1.168 address as its currently showing the internal network (even though ext-rtp-Ip is set and confirmed). 

192.168.1.168 is a bridged network interface. 

192.168.1.something -> 192.168.1.168 (docker vm)->172.17.0.5(docker container)

The docker container has the rtp ports "exposed" (in the same way as it does for the signalling port).

I actually didn't like docker when I first looked at it, but it actually quite nice for development (bar this issue)
Thanks
Oz

> On 6 Apr 2016, at 19:00, Michael Jerris <mike at jerris.com> wrote:
> 
> And your clients are attaching from what network?  Also if your description is correct, then you have internal and external addresses backwards.  Does it actually nat those addresses or is it routed?
> 
> 
>> On Apr 6, 2016, at 1:49 PM, Oz Mortimer <omortimer at gmail.com> wrote:
>> 
>> I know! I've never come across it! It's what I seem to have to do when using a docker container - unless someone knows different?
>> Docker uses a vm, which is on the network 192.168.. But the container has an IP of 172.17..
>> I'm no docker expert and in an ideal world the container should have a 192.168.. Address, but I can't find a way to make that happen.
>> Maybe I'm asking in the wrong mailing list - could be a question for docker. Either way, I'd like to figure out what I'm doing wrong! 
>> Nb. This is no way going to be a production setup - it's a development setup.
>> 
>> I "think" I understand what I need to do..
>> Thanks
>> Oz
>>> On 6 Apr 2016, at 18:39, Michael Jerris <mike at jerris.com> wrote:
>>> 
>>> The default acl's treat all rfc1918 addresses as internal.  you'll need to make one that treats your external addresses as external even tho they are rfc1918.  Why are you natting from one private address to another?  Its a very strange implementation
>>> 
>>> 
>>>> On Apr 6, 2016, at 1:31 PM, Oz Mortimer <omortimer at gmail.com> wrote:
>>>> 
>>>> Hi,
>>>> Thanks for the reply - I wish I understood it - but I don?t ;)
>>>> Yes, the natting is between one rfc1918 address space to another.
>>>> 
>>>> Based on your reply I tried
>>>> 
>>>> 	<param name="apply-nat-acl" value="damnnat?/>
>>>> 
>>>> where acl.conf.xml has
>>>> 
>>>>  <list name="damnnat" default="allow">
>>>>       <node type="allow" cidr="192.168.1.0/24"/>
>>>>       <node type="allow" cidr="172.17.0.0/24"/>
>>>>     </list>
>>>> 
>>>> To no avail!. can you give me a pointer to what I need to change and where?
>>>> 
>>>> Thanks
>>>> Oz.
>>>> 
>>>>> On 6 Apr 2016, at 17:58, Michael Jerris <mike at jerris.com> wrote:
>>>>> 
>>>>> you are natting from one rfc1918 address space to another?  If so, all the default nat acl's will be wrong, and you will have to make your own acl's that match your network environment.
>>>>> 
>>>>>> On Apr 6, 2016, at 12:48 PM, Oz Mortimer <omortimer at gmail.com> wrote:
>>>>>> 
>>>>>> HI,
>>>>>> 
>>>>>> I?m trying to get FS running in Docker, which largely was pain free (i know, i know, VMs, etc), but I can?t get my head around what is going on with RTP. Ive set ext-rtp-ip and it seems to be taking affect:
>>>>>> 
>>>>>> freeswitch at 7ad22635059e> sofia status profile internal
>>>>>> =================================================================================================
>>>>>> Name             	internal
>>>>>> Domain Name      	N/A
>>>>>> Auto-NAT         	false
>>>>>> DBName           	sofia_reg_internal
>>>>>> Pres Hosts       	172.17.0.5,172.17.0.5
>>>>>> Dialplan         	XML
>>>>>> Context          	trusted
>>>>>> Challenge Realm  	auto_from
>>>>>> RTP-IP           	172.17.0.5
>>>>>> Ext-RTP-IP       	192.168.1.168
>>>>>> SIP-IP           	172.17.0.5
>>>>>> Ext-SIP-IP       	192.168.1.168
>>>>>> URL              	sip:mod_sofia at 192.168.1.168:5060
>>>>>> BIND-URL         	sip:mod_sofia at 192.168.1.168:5060;maddr=172.17.0.5;transport=udp,tcp
>>>>>> HOLD-MUSIC       	local_stream://moh
>>>>>> OUTBOUND-PROXY   	N/A
>>>>>> CODECS IN        	G729,PCMU,PCMA
>>>>>> CODECS OUT       	G729,PCMU,PCMA
>>>>>> TEL-EVENT        	101
>>>>>> DTMF-MODE        	none
>>>>>> CNG              	13
>>>>>> SESSION-TO       	0
>>>>>> MAX-DIALOG       	0
>>>>>> NOMEDIA          	false
>>>>>> LATE-NEG         	true
>>>>>> PROXY-MEDIA      	false
>>>>>> ZRTP-PASSTHRU    	true
>>>>>> AGGRESSIVENAT    	false
>>>>>> CALLS-IN         	2
>>>>>> FAILED-CALLS-IN  	2
>>>>>> CALLS-OUT        	0
>>>>>> FAILED-CALLS-OUT 	0
>>>>>> REGISTRATIONS    	0
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> but when a call is placed i seems to be incorrect in the SDP
>>>>>> 
>>>>>> 2016-04-06 16:29:49.011107 [DEBUG] mod_sofia.c:2353 Ring SDP:
>>>>>> v=0
>>>>>> o=FreeSWITCH 1459942605 1459942606 IN IP4 172.17.0.5
>>>>>> s=FreeSWITCH
>>>>>> c=IN IP4 172.17.0.5
>>>>>> t=0 0
>>>>>> m=audio 17584 RTP/AVP 8 101
>>>>>> a=rtpmap:8 PCMA/8000
>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>> a=fmtp:101 0-16
>>>>>> a=ptime:20
>>>>>> a=sendrecv
>>>>>> 
>>>>>> Shouldn?t the SDP reflect the Ext-RTP-IP ?
>>>>>> 
>>>>>> Im sure i?ve missed some sort of config setting or have gone snow blind!.
>>>>>> fs version is FreeSWITCH (Version 1.6.7 -14-d38d065 64bit)
>>>>>> 
>>>>>> Any ideas will be greatly received.
>>>>>> Thanks
>>>>>> Oz.
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services: 
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services: 
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services: 
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: 
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Wed Apr  6 22:22:16 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Wed, 6 Apr 2016 13:22:16 -0500
Subject: [Freeswitch-users] docker / NAT troubles..
In-Reply-To: <E57993A6-A251-45FE-BC60-3593DC39837C@gmail.com>
References: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>
	<AE01A0C6-615F-44F1-B74A-67916833B156@jerris.com>
	<67B6621A-1705-4595-85F1-429F31819DC6@gmail.com>
	<45895411-F048-4DA9-B0F4-579E0A5E0E29@jerris.com>
	<17C2D436-F1A3-42B4-B742-524797B99C8B@gmail.com>
	<828541E3-6097-4932-8C29-C5516EAB45CF@jerris.com>
	<E57993A6-A251-45FE-BC60-3593DC39837C@gmail.com>
Message-ID: <570553D8.3000902@mst.edu>

As long as you don't have yet another layer of external access required - the original suggestion I think is probably 
right - you need to make a new set of ACLs for the freeswitch nat acl that do not include the 192.x.x.x network. Then 
that net will be interpreted as external.

-- Nathan

On 04/06/2016 01:13 PM, Oz Mortimer wrote:
> The clients are on the 192. Network.
> The network internal to freeswitch is 17.
> Docker assigns the 17. Ips to the container (which fs is running on).
>
> The signalling is fine, but the sdp needs to show the 192.168.1.168 address as its currently showing the internal
> network (even though ext-rtp-Ip is set and confirmed).
>
> 192.168.1.168 is a bridged network interface.
>
> 192.168.1.something -> 192.168.1.168 (docker vm)->172.17.0.5(docker container)
>
> The docker container has the rtp ports "exposed" (in the same way as it does for the signalling port).
>
> I actually didn't like docker when I first looked at it, but it actually quite nice for development (bar this issue)
> Thanks
> Oz
>
> On 6 Apr 2016, at 19:00, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>
>> And your clients are attaching from what network?  Also if your description is correct, then you have internal and
>> external addresses backwards.  Does it actually nat those addresses or is it routed?
>>
>>
>>> On Apr 6, 2016, at 1:49 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>>>
>>> I know! I've never come across it! It's what I seem to have to do when using a docker container - unless someone
>>> knows different?
>>> Docker uses a vm, which is on the network 192.168.. But the container has an IP of 172.17..
>>> I'm no docker expert and in an ideal world the container should have a 192.168.. Address, but I can't find a way to
>>> make that happen.
>>> Maybe I'm asking in the wrong mailing list - could be a question for docker. Either way, I'd like to figure out what
>>> I'm doing wrong!
>>> Nb. This is no way going to be a production setup - it's a development setup.
>>>
>>> I "think" I understand what I need to do..
>>> Thanks
>>> Oz
>>> On 6 Apr 2016, at 18:39, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>>
>>>> The default acl's treat all rfc1918 addresses as internal.  you'll need to make one that treats your external
>>>> addresses as external even tho they are rfc1918.  Why are you natting from one private address to another?  Its a
>>>> very strange implementation
>>>>
>>>>
>>>>> On Apr 6, 2016, at 1:31 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>>>>>
>>>>> Hi,
>>>>> Thanks for the reply - I wish I understood it - but I don?t ;)
>>>>> Yes, the natting is between one rfc1918 address space to another.
>>>>>
>>>>> Based on your reply I tried
>>>>>
>>>>> <param name="apply-nat-acl" value="damnnat?/>
>>>>>
>>>>> where acl.conf.xml has
>>>>>
>>>>>  <list name="damnnat" default="allow">
>>>>>       <node type="allow" cidr="192.168.1.0/24"/>
>>>>>       <node type="allow" cidr="172.17.0.0/24"/>
>>>>>     </list>
>>>>>
>>>>> To no avail!. can you give me a pointer to what I need to change and where?
>>>>>
>>>>> Thanks
>>>>> Oz.
>>>>>
>>>>>> On 6 Apr 2016, at 17:58, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>>>>>
>>>>>> you are natting from one rfc1918 address space to another?  If so, all the default nat acl's will be wrong, and
>>>>>> you will have to make your own acl's that match your network environment.
>>>>>>
>>>>>>> On Apr 6, 2016, at 12:48 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>>>>>>>
>>>>>>> HI,
>>>>>>>
>>>>>>> I?m trying to get FS running in Docker, which largely was pain free (i know, i know, VMs, etc), but I can?t get
>>>>>>> my head around what is going on with RTP. Ive set ext-rtp-ip and it seems to be taking affect:
>>>>>>>
>>>>>>> freeswitch at 7ad22635059e> sofia status profile internal
>>>>>>> =================================================================================================
>>>>>>> Name internal
>>>>>>> Domain Name N/A
>>>>>>> Auto-NAT false
>>>>>>> DBName sofia_reg_internal
>>>>>>> Pres Hosts 172.17.0.5,172.17.0.5
>>>>>>> Dialplan XML
>>>>>>> Context trusted
>>>>>>> Challenge Realm auto_from
>>>>>>> RTP-IP 172.17.0.5
>>>>>>> Ext-RTP-IP 192.168.1.168
>>>>>>> SIP-IP 172.17.0.5
>>>>>>> Ext-SIP-IP 192.168.1.168
>>>>>>> URL sip:mod_sofia at 192.168.1.168:5060
>>>>>>> BIND-URL sip:mod_sofia at 192.168.1.168:5060;maddr=172.17.0.5;transport=udp,tcp
>>>>>>> HOLD-MUSIC local_stream://moh
>>>>>>> OUTBOUND-PROXY N/A
>>>>>>> CODECS IN G729,PCMU,PCMA
>>>>>>> CODECS OUT G729,PCMU,PCMA
>>>>>>> TEL-EVENT 101
>>>>>>> DTMF-MODE none
>>>>>>> CNG 13
>>>>>>> SESSION-TO 0
>>>>>>> MAX-DIALOG 0
>>>>>>> NOMEDIA false
>>>>>>> LATE-NEG true
>>>>>>> PROXY-MEDIA false
>>>>>>> ZRTP-PASSTHRU true
>>>>>>> AGGRESSIVENAT false
>>>>>>> CALLS-IN 2
>>>>>>> FAILED-CALLS-IN 2
>>>>>>> CALLS-OUT 0
>>>>>>> FAILED-CALLS-OUT 0
>>>>>>> REGISTRATIONS 0
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> but when a call is placed i seems to be incorrect in the SDP
>>>>>>>
>>>>>>> 2016-04-06 16:29:49.011107 [DEBUG] mod_sofia.c:2353 Ring SDP:
>>>>>>> v=0
>>>>>>> o=FreeSWITCH 1459942605 1459942606 IN IP4 172.17.0.5
>>>>>>> s=FreeSWITCH
>>>>>>> c=IN IP4 172.17.0.5
>>>>>>> t=0 0
>>>>>>> m=audio 17584 RTP/AVP 8 101
>>>>>>> a=rtpmap:8 PCMA/8000
>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>> a=fmtp:101 0-16
>>>>>>> a=ptime:20
>>>>>>> a=sendrecv
>>>>>>>
>>>>>>> Shouldn?t the SDP reflect the Ext-RTP-IP ?
>>>>>>>
>>>>>>> Im sure i?ve missed some sort of config setting or have gone snow blind!.
>>>>>>> fs version is FreeSWITCH (Version 1.6.7 -14-d38d065 64bit)
>>>>>>>
>>>>>>> Any ideas will be greatly received.
>>>>>>> Thanks
>>>>>>> Oz.
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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


From omortimer at gmail.com  Wed Apr  6 22:29:14 2016
From: omortimer at gmail.com (Oz Mortimer)
Date: Wed, 6 Apr 2016 19:29:14 +0100
Subject: [Freeswitch-users] docker / NAT troubles..
In-Reply-To: <570553D8.3000902@mst.edu>
References: <F00E19D7-70F2-4FBB-B3B5-3166F78747E0@gmail.com>
	<AE01A0C6-615F-44F1-B74A-67916833B156@jerris.com>
	<67B6621A-1705-4595-85F1-429F31819DC6@gmail.com>
	<45895411-F048-4DA9-B0F4-579E0A5E0E29@jerris.com>
	<17C2D436-F1A3-42B4-B742-524797B99C8B@gmail.com>
	<828541E3-6097-4932-8C29-C5516EAB45CF@jerris.com>
	<E57993A6-A251-45FE-BC60-3593DC39837C@gmail.com>
	<570553D8.3000902@mst.edu>
Message-ID: <2E4B566F-D8BD-45E6-8788-9DE8C05EB0C5@gmail.com>

Gocha! And absolutely not (though I might be tempted to try it for the sake of external demos ;))
Thanks again
Oz 

> On 6 Apr 2016, at 19:22, Nathan Neulinger <nneul at mst.edu> wrote:
> 
> As long as you don't have yet another layer of external access required - the original suggestion I think is probably 
> right - you need to make a new set of ACLs for the freeswitch nat acl that do not include the 192.x.x.x network. Then 
> that net will be interpreted as external.
> 
> -- Nathan
> 
>> On 04/06/2016 01:13 PM, Oz Mortimer wrote:
>> The clients are on the 192. Network.
>> The network internal to freeswitch is 17.
>> Docker assigns the 17. Ips to the container (which fs is running on).
>> 
>> The signalling is fine, but the sdp needs to show the 192.168.1.168 address as its currently showing the internal
>> network (even though ext-rtp-Ip is set and confirmed).
>> 
>> 192.168.1.168 is a bridged network interface.
>> 
>> 192.168.1.something -> 192.168.1.168 (docker vm)->172.17.0.5(docker container)
>> 
>> The docker container has the rtp ports "exposed" (in the same way as it does for the signalling port).
>> 
>> I actually didn't like docker when I first looked at it, but it actually quite nice for development (bar this issue)
>> Thanks
>> Oz
>> 
>>> On 6 Apr 2016, at 19:00, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>> 
>>> And your clients are attaching from what network?  Also if your description is correct, then you have internal and
>>> external addresses backwards.  Does it actually nat those addresses or is it routed?
>>> 
>>> 
>>>> On Apr 6, 2016, at 1:49 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>>>> 
>>>> I know! I've never come across it! It's what I seem to have to do when using a docker container - unless someone
>>>> knows different?
>>>> Docker uses a vm, which is on the network 192.168.. But the container has an IP of 172.17..
>>>> I'm no docker expert and in an ideal world the container should have a 192.168.. Address, but I can't find a way to
>>>> make that happen.
>>>> Maybe I'm asking in the wrong mailing list - could be a question for docker. Either way, I'd like to figure out what
>>>> I'm doing wrong!
>>>> Nb. This is no way going to be a production setup - it's a development setup.
>>>> 
>>>> I "think" I understand what I need to do..
>>>> Thanks
>>>> Oz
>>>>> On 6 Apr 2016, at 18:39, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>>>> 
>>>>> The default acl's treat all rfc1918 addresses as internal.  you'll need to make one that treats your external
>>>>> addresses as external even tho they are rfc1918.  Why are you natting from one private address to another?  Its a
>>>>> very strange implementation
>>>>> 
>>>>> 
>>>>>> On Apr 6, 2016, at 1:31 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>>>>>> 
>>>>>> Hi,
>>>>>> Thanks for the reply - I wish I understood it - but I don?t ;)
>>>>>> Yes, the natting is between one rfc1918 address space to another.
>>>>>> 
>>>>>> Based on your reply I tried
>>>>>> 
>>>>>> <param name="apply-nat-acl" value="damnnat?/>
>>>>>> 
>>>>>> where acl.conf.xml has
>>>>>> 
>>>>>> <list name="damnnat" default="allow">
>>>>>>      <node type="allow" cidr="192.168.1.0/24"/>
>>>>>>      <node type="allow" cidr="172.17.0.0/24"/>
>>>>>>    </list>
>>>>>> 
>>>>>> To no avail!. can you give me a pointer to what I need to change and where?
>>>>>> 
>>>>>> Thanks
>>>>>> Oz.
>>>>>> 
>>>>>>> On 6 Apr 2016, at 17:58, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>>>>>>> 
>>>>>>> you are natting from one rfc1918 address space to another?  If so, all the default nat acl's will be wrong, and
>>>>>>> you will have to make your own acl's that match your network environment.
>>>>>>> 
>>>>>>>> On Apr 6, 2016, at 12:48 PM, Oz Mortimer <omortimer at gmail.com <mailto:omortimer at gmail.com>> wrote:
>>>>>>>> 
>>>>>>>> HI,
>>>>>>>> 
>>>>>>>> I?m trying to get FS running in Docker, which largely was pain free (i know, i know, VMs, etc), but I can?t get
>>>>>>>> my head around what is going on with RTP. Ive set ext-rtp-ip and it seems to be taking affect:
>>>>>>>> 
>>>>>>>> freeswitch at 7ad22635059e> sofia status profile internal
>>>>>>>> =================================================================================================
>>>>>>>> Name internal
>>>>>>>> Domain Name N/A
>>>>>>>> Auto-NAT false
>>>>>>>> DBName sofia_reg_internal
>>>>>>>> Pres Hosts 172.17.0.5,172.17.0.5
>>>>>>>> Dialplan XML
>>>>>>>> Context trusted
>>>>>>>> Challenge Realm auto_from
>>>>>>>> RTP-IP 172.17.0.5
>>>>>>>> Ext-RTP-IP 192.168.1.168
>>>>>>>> SIP-IP 172.17.0.5
>>>>>>>> Ext-SIP-IP 192.168.1.168
>>>>>>>> URL sip:mod_sofia at 192.168.1.168:5060
>>>>>>>> BIND-URL sip:mod_sofia at 192.168.1.168:5060;maddr=172.17.0.5;transport=udp,tcp
>>>>>>>> HOLD-MUSIC local_stream://moh
>>>>>>>> OUTBOUND-PROXY N/A
>>>>>>>> CODECS IN G729,PCMU,PCMA
>>>>>>>> CODECS OUT G729,PCMU,PCMA
>>>>>>>> TEL-EVENT 101
>>>>>>>> DTMF-MODE none
>>>>>>>> CNG 13
>>>>>>>> SESSION-TO 0
>>>>>>>> MAX-DIALOG 0
>>>>>>>> NOMEDIA false
>>>>>>>> LATE-NEG true
>>>>>>>> PROXY-MEDIA false
>>>>>>>> ZRTP-PASSTHRU true
>>>>>>>> AGGRESSIVENAT false
>>>>>>>> CALLS-IN 2
>>>>>>>> FAILED-CALLS-IN 2
>>>>>>>> CALLS-OUT 0
>>>>>>>> FAILED-CALLS-OUT 0
>>>>>>>> REGISTRATIONS 0
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> but when a call is placed i seems to be incorrect in the SDP
>>>>>>>> 
>>>>>>>> 2016-04-06 16:29:49.011107 [DEBUG] mod_sofia.c:2353 Ring SDP:
>>>>>>>> v=0
>>>>>>>> o=FreeSWITCH 1459942605 1459942606 IN IP4 172.17.0.5
>>>>>>>> s=FreeSWITCH
>>>>>>>> c=IN IP4 172.17.0.5
>>>>>>>> t=0 0
>>>>>>>> m=audio 17584 RTP/AVP 8 101
>>>>>>>> a=rtpmap:8 PCMA/8000
>>>>>>>> a=rtpmap:101 telephone-event/8000
>>>>>>>> a=fmtp:101 0-16
>>>>>>>> a=ptime:20
>>>>>>>> a=sendrecv
>>>>>>>> 
>>>>>>>> Shouldn?t the SDP reflect the Ext-RTP-IP ?
>>>>>>>> 
>>>>>>>> Im sure i?ve missed some sort of config setting or have gone snow blind!.
>>>>>>>> fs version is FreeSWITCH (Version 1.6.7 -14-d38d065 64bit)
>>>>>>>> 
>>>>>>>> Any ideas will be greatly received.
>>>>>>>> Thanks
>>>>>>>> Oz.
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>>>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>>>>>> 
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>> 
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>>>>> 
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>>>> 
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>>>>> http://www.cluecon.com <http://www.cluecon.com/>
>>>>> 
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>>> http://www.freeswitch.org <http://www.freeswitch.org/>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>>> http://www.freeswitch.org
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://www.freeswitch.org
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> -- 
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 mandra at gmail.com  Thu Apr  7 02:06:28 2016
From: mandra at gmail.com (Chris Mandra)
Date: Wed, 6 Apr 2016 18:06:28 -0400
Subject: [Freeswitch-users] Module initialization failure status
In-Reply-To: <16D92427-15DA-49F7-9D7D-B4CF86CC43B2@jerris.com>
References: <CAJdRzN4XOaj=-F6=CX0tOLv9FDMFDkiVi8ETgAhPh-RHEzONsg@mail.gmail.com>
	<CAJdRzN4Ni6dg6_cx5VsPWpe1+EY8n3559qcqDvgAumpF4Pw79g@mail.gmail.com>
	<16D92427-15DA-49F7-9D7D-B4CF86CC43B2@jerris.com>
Message-ID: <CAJdRzN4uF5jsWmz3=Z8BWKZX-7Lab7dHADmkdo2+P6+A=EFe2g@mail.gmail.com>

Thanks Michael

On Mon, Apr 4, 2016 at 10:44 AM, Michael Jerris <mike at jerris.com> wrote:

> anything but
>
> SWITCH_STATUS_SUCCESS
> SWITCH_STATUS_NOUNLOAD
>
> is considered an error.
>
>
> On Apr 3, 2016, at 6:16 PM, Chris Mandra <mandra at gmail.com> wrote:
>
> Anything?
>
> On Saturday, April 2, 2016, Chris Mandra <mandra at gmail.com> wrote:
>
>> Hi Guys - Happy Saturday.
>> Quick question: what is  the proper status to return if a module fails on
>> initialization?
>> That is: should it always return SWITCH_STATUS_SUCCESS ?
>> What if something goes wrong?  What status should it return?
>> Thanks!chris
>>
>>
>>
>
> --
> mandra
> c:410.258.5281
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
mandra
c:410.258.5281
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From mandra at gmail.com  Thu Apr  7 02:07:14 2016
From: mandra at gmail.com (Chris Mandra)
Date: Wed, 6 Apr 2016 18:07:14 -0400
Subject: [Freeswitch-users] Unit tests for modules?
Message-ID: <CAJdRzN4YSuNp48syEfAezt=iXmiHJD+2Ea8UrhbvP1g-wmYvCQ@mail.gmail.com>

Is there a best practice way to do unit testing of new freeswitch modules?
Any recommendations?
thanks, chris
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From jelena at misticnabica.hr  Thu Apr  7 02:19:04 2016
From: jelena at misticnabica.hr (jelena at misticnabica.hr)
Date: Wed, 6 Apr 2016 22:19:04 GMT
Subject: [Freeswitch-users] Unit tests for modules?
Message-ID: <61A7DCD8430442889C55F829D134C675.MAI@server2.totohost.hr>



From steveayre at gmail.com  Thu Apr  7 02:34:46 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 6 Apr 2016 23:34:46 +0100
Subject: [Freeswitch-users] Is MySQL OK to use?
In-Reply-To: <A8930F04-C0DF-4C38-9AAD-3E80189FBC2D@jerris.com>
References: <528E45C5.8030509@digitalmail.com>
	<CAC-LwPME371eMDigDeZXTbf-WeOLx56La8iE9eyyS-f+xh6FGA@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C91C@IMBER.smb.curriegrad2004.ca>
	<CAFiqYu=F2w51wy4M15ygP_=iHSe5orqRqu6zdnYNND8D6=ix=g@mail.gmail.com>
	<CAKbxfG-yLgyzgq3D0RT+96RcfTSWBi7myX+t0ew37arRSLYkXQ@mail.gmail.com>
	<CAFiqYumNo9wQHyZNx5ufuhohE_ivBQJTTrYm7suY4-8UbGzmNQ@mail.gmail.com>
	<A8930F04-C0DF-4C38-9AAD-3E80189FBC2D@jerris.com>
Message-ID: <CAFiqYu=WyEu29MeCwnei7Xct91V1m2Par=261Kg0bKz0W0Jngw@mail.gmail.com>

It's within the MyODBC driver and not within the libmysqlclient library
which means the driver is not thread-safe even if you're using
libmysqlclient_r. Incidentally in recent versions libmysqlclient is
thread-safe and libmysqlclient_r is just a symlink to libmysqlclient.

On 6 April 2016 at 17:20, Michael Jerris <mike at jerris.com> wrote:

> This sounds like a bug in the mysql driver then right?  If those are not
> threadsafe then the driver should protect against use where it is not safe.
>
> On Apr 6, 2016, at 5:44 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
> I've been doing some debugging on this this week as we're currently seeing
> regular crashes on our production servers. This happens with both
> Threading=0 and Threading=2.
>
> It appears it's a bug in MyODBC triggered because FreeSWITCH uses a new
> ODBC environment handle every time it creates a connection.
>
> When a environment handle is created myodbc_init is called and when it's
> freed it calls myodbc_end. Those functions are not thread-safe. They're
> protected by a reference counter but they aren't wrapped in a mutex so if
> the server is busy and they're both called at the same time you can result
> in some global variables being used after they're freed.
>
> Rebuilding MyODBC with an empty myodbc_end appears to stop the crashes,
> but I'm not saying that's a good solution. Still looking at it.
>
> On 22 November 2013 at 20:31, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I've always seen Threading=0 as the solution to stability probs.
>> Basically if the driver already uses mutexing, its better to disable the
>> arbitrary ones in the core of odbc.
>> For postgres 0 is basically mandatory.
>>
>>
>>
>> On Fri, Nov 22, 2013 at 2:12 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> No such thing in the latest versions, they're all threadsafe now (_r is
>>> a symlink to the other).
>>>
>>> I use it with success. I did find stability problems in libmyodbc when
>>> upgrading from 5.0 to 5.5. My solution was to add these to odbc.ini
>>>
>>> Option      = 67108864
>>> Threading   = 2
>>>
>>> The Threading setting is what stopped the crashes. There's a Jira on the
>>> subject and someone else said manually upgrading to a newer version of
>>> unixodbc+myodbc also fixed it.
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 21 November 2013 18:23, Jeff Leung <jleung at v10networks.ca> wrote:
>>>
>>>> Try the non-thread safe MySQL library instead of the thread safe one.
>>>>
>>>>
>>>>
>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Vik Killa
>>>> *Sent:* Thursday, November 21, 2013 10:22 AM
>>>> *To:* FreeSWITCH Users Help
>>>> *Subject:* Re: [Freeswitch-users] Is MySQL OK to use?
>>>>
>>>>
>>>>
>>>> short answer no.
>>>>
>>>> use postgres
>>>>
>>>>
>>>>
>>>> On Thu, Nov 21, 2013 at 12:41 PM, Alex Lake <alex at digitalmail.com>
>>>> wrote:
>>>>
>>>> I find these "Error in my_thread_global_end()" messages somewhat
>>>> annoying in my fs1.2stable on Ubuntu12.04 box. Is there any advice
>>>> (other than "don't use MySQL") for how to install it better? Might it be
>>>> something to do with thread-safe libraries (I'm using libmyodbc_r.so at
>>>> the moment)
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>> http://www.cudatel.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>> http://www.cudatel.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>> http://www.cudatel.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>> http://www.cudatel.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mandra at gmail.com  Thu Apr  7 02:48:47 2016
From: mandra at gmail.com (Chris Mandra)
Date: Wed, 6 Apr 2016 18:48:47 -0400
Subject: [Freeswitch-users] Unit tests for modules?
In-Reply-To: <61A7DCD8430442889C55F829D134C675.MAI@server2.totohost.hr>
References: <61A7DCD8430442889C55F829D134C675.MAI@server2.totohost.hr>
Message-ID: <CAJdRzN7jyg9JvgLBmcZM-fQYFA3sg=ZFkcDeAayqd2iM7mDdkw@mail.gmail.com>

Why do you send messages like this?

On Wednesday, April 6, 2016, <jelena at misticnabica.hr> wrote:

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


-- 
mandra
c:410.258.5281
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From luis.daniel.lucio at gmail.com  Thu Apr  7 02:55:10 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Wed, 6 Apr 2016 18:55:10 -0400
Subject: [Freeswitch-users] Is MySQL OK to use?
In-Reply-To: <CAFiqYu=WyEu29MeCwnei7Xct91V1m2Par=261Kg0bKz0W0Jngw@mail.gmail.com>
References: <528E45C5.8030509@digitalmail.com>
	<CAC-LwPME371eMDigDeZXTbf-WeOLx56La8iE9eyyS-f+xh6FGA@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C91C@IMBER.smb.curriegrad2004.ca>
	<CAFiqYu=F2w51wy4M15ygP_=iHSe5orqRqu6zdnYNND8D6=ix=g@mail.gmail.com>
	<CAKbxfG-yLgyzgq3D0RT+96RcfTSWBi7myX+t0ew37arRSLYkXQ@mail.gmail.com>
	<CAFiqYumNo9wQHyZNx5ufuhohE_ivBQJTTrYm7suY4-8UbGzmNQ@mail.gmail.com>
	<A8930F04-C0DF-4C38-9AAD-3E80189FBC2D@jerris.com>
	<CAFiqYu=WyEu29MeCwnei7Xct91V1m2Par=261Kg0bKz0W0Jngw@mail.gmail.com>
Message-ID: <CAFLo2QyQjeRBBtUkAffCAMeEPyXeTiZj2JHGWzhM7xcnSyv2UA@mail.gmail.com>

Let me get home and I will post releases I have. Checkout I hold about 10
FreeSWITCH using MySQL as a backbend without issues
Le 6 avr. 2016 6:35 PM, "Steven Ayre" <steveayre at gmail.com> a ?crit :

> It's within the MyODBC driver and not within the libmysqlclient library
> which means the driver is not thread-safe even if you're using
> libmysqlclient_r. Incidentally in recent versions libmysqlclient is
> thread-safe and libmysqlclient_r is just a symlink to libmysqlclient.
>
> On 6 April 2016 at 17:20, Michael Jerris <mike at jerris.com> wrote:
>
>> This sounds like a bug in the mysql driver then right?  If those are not
>> threadsafe then the driver should protect against use where it is not safe.
>>
>> On Apr 6, 2016, at 5:44 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> I've been doing some debugging on this this week as we're currently
>> seeing regular crashes on our production servers. This happens with both
>> Threading=0 and Threading=2.
>>
>> It appears it's a bug in MyODBC triggered because FreeSWITCH uses a new
>> ODBC environment handle every time it creates a connection.
>>
>> When a environment handle is created myodbc_init is called and when it's
>> freed it calls myodbc_end. Those functions are not thread-safe. They're
>> protected by a reference counter but they aren't wrapped in a mutex so if
>> the server is busy and they're both called at the same time you can result
>> in some global variables being used after they're freed.
>>
>> Rebuilding MyODBC with an empty myodbc_end appears to stop the crashes,
>> but I'm not saying that's a good solution. Still looking at it.
>>
>> On 22 November 2013 at 20:31, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> I've always seen Threading=0 as the solution to stability probs.
>>> Basically if the driver already uses mutexing, its better to disable the
>>> arbitrary ones in the core of odbc.
>>> For postgres 0 is basically mandatory.
>>>
>>>
>>>
>>> On Fri, Nov 22, 2013 at 2:12 AM, Steven Ayre <steveayre at gmail.com>
>>> wrote:
>>>
>>>> No such thing in the latest versions, they're all threadsafe now (_r is
>>>> a symlink to the other).
>>>>
>>>> I use it with success. I did find stability problems in libmyodbc when
>>>> upgrading from 5.0 to 5.5. My solution was to add these to odbc.ini
>>>>
>>>> Option      = 67108864
>>>> Threading   = 2
>>>>
>>>> The Threading setting is what stopped the crashes. There's a Jira on
>>>> the subject and someone else said manually upgrading to a newer version of
>>>> unixodbc+myodbc also fixed it.
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>> On 21 November 2013 18:23, Jeff Leung <jleung at v10networks.ca> wrote:
>>>>
>>>>> Try the non-thread safe MySQL library instead of the thread safe one.
>>>>>
>>>>>
>>>>>
>>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Vik
>>>>> Killa
>>>>> *Sent:* Thursday, November 21, 2013 10:22 AM
>>>>> *To:* FreeSWITCH Users Help
>>>>> *Subject:* Re: [Freeswitch-users] Is MySQL OK to use?
>>>>>
>>>>>
>>>>>
>>>>> short answer no.
>>>>>
>>>>> use postgres
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Nov 21, 2013 at 12:41 PM, Alex Lake <alex at digitalmail.com>
>>>>> wrote:
>>>>>
>>>>> I find these "Error in my_thread_global_end()" messages somewhat
>>>>> annoying in my fs1.2stable on Ubuntu12.04 box. Is there any advice
>>>>> (other than "don't use MySQL") for how to install it better? Might it
>>>>> be
>>>>> something to do with thread-safe libraries (I'm using libmyodbc_r.so at
>>>>> the moment)
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>> http://www.cudatel.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>> http://www.cudatel.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>> http://www.cudatel.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>> http://www.cudatel.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 linuxpenguins.xyz  Thu Apr  7 05:02:49 2016
From: brian at linuxpenguins.xyz (Brian May)
Date: Thu, 07 Apr 2016 11:02:49 +1000
Subject: [Freeswitch-users] Message is less than minimum record length
Message-ID: <87wpoai62u.fsf@prune.linuxpenguins.xyz>

Some reason my freeswitch system has suddenly stopped listening for
messages. Instead it immediately cuts the talker off and says the
message is too short. Any ideas? How do I debug this?

i am going to conduct an audio test later on to make sure this is
working, last I checked phone calls were working fine however.


2016-04-07 10:09:31.172293 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16 at 8000hz 1 channels 20ms
2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:1910 done playing file /var/lib/freeswitch/storage/voicemail/default/microcomaustralia.com.au/2000/greeting_1.wav
2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en]
2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-record_message.wav] (en:en)
2016-04-07 10:09:40.232300 [DEBUG] mod_loopback.c:601 loopback/voicemail-b CHANNEL KILL
2016-04-07 10:09:40.232300 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16 at 8000hz 1 channels 20ms
2016-04-07 10:09:44.792345 [DEBUG] switch_ivr_play_say.c:1910 done playing file /usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.wav
2016-04-07 10:09:45.912334 [DEBUG] switch_ivr_play_say.c:559 Raw Codec Activated, ready to waste resources!
2016-04-07 10:09:45.912334 [DEBUG] switch_ivr_play_say.c:673 Raw Codec Activated
2016-04-07 10:09:45.912334 [DEBUG] switch_core_codec.c:221 loopback/voicemail-b Push codec L16:100
2016-04-07 10:09:47.912303 [DEBUG] switch_core_codec.c:246 loopback/voicemail-b Restore previous codec PCMA:8.
2016-04-07 10:09:47.932384 [DEBUG] mod_voicemail.c:1245 Message is less than minimum record length: 3, discarding it.
2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en]
2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-too-small.wav] (en:en)
2016-04-07 10:09:47.932384 [DEBUG] mod_loopback.c:601 loopback/voicemail-b CHANNEL KILL
2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16 at 8000hz 1 channels 20ms
2016-04-07 10:09:51.492336 [DEBUG] sofia.c:6858 Channel sofia/external/61419503953 at 203.31.79.18 entering state [terminated][487]
2016-04-07 10:09:51.492336 [NOTICE] sofia.c:7879 Hangup sofia/external/61419503953 at 203.31.79.18 [CS_EXECUTE] [ORIGINATOR_CANCEL]
2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:707 sofia/external/61419503953 at 203.31.79.18 ending bridge by request from read function
2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:780 BRIDGE THREAD DONE [sofia/external/61419503953 at 203.31.79.18]
2016-04-07 10:09:51.492336 [DEBUG] mod_loopback.c:601 loopback/voicemail-a CHANNEL KILL
2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:701 sofia/external/61419503953 at 203.31.79.18 ending bridge by request from write function

-- 
Brian May <brian at linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/


From quanhs at stee.stengg.com  Thu Apr  7 06:40:50 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Thu, 7 Apr 2016 02:40:50 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION ERROR. SDP
In-Reply-To: <CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
Message-ID: <efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>

OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.

Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.

As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.

In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).

Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Wednesday, April 06, 2016 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

yes it's supported

On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Itola;

Sorry, same error.

Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?

Finding when FS runs in media mode:
codec causes caller side ?488 not acceptable here| incompatible destination ?
                callee side: ?cancel |user not registered?

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');> [mailto:freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>] On Behalf Of ?talo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Set media_mix_inbound_outbound_codecs=true

?talo Rossi
italo at freeswitch.org<javascript:_e(%7B%7D,'cvml','italo at freeswitch.org');>
IRC chat.freenode.net<http://chat.freenode.net> #freeswitch #freeswitch-dev
Bugs? https://freeswitch.org/jira
Docs? https://freeswitch.org/jira
Chat? https://hipchat.freeswitch.org/gUdAgy0m6


On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<javascript:_e(%7B%7D,'cvml','quanhs at stee.stengg.com');>> wrote:

Hi All;



I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min<http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min<http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.

If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.

But if disabling media bypass mode, call is rejected by FS.



FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.

Chrome uses opus 111, FS uses opus 116.

Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().



Help is needed to troubleshoot this issue.



Thanks advance.

Smile.



[This e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
from your system; please do not copy or use it for any purpose, nor disclose
its contents to any other person. Thank you.]
---ST Electronics Group---
-------------- next part --------------
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From quanhs at stee.stengg.com  Thu Apr  7 06:58:25 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Thu, 7 Apr 2016 02:58:25 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION ERROR. SDP
In-Reply-To: <efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
Message-ID: <ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>

Hi;

Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Quan Huo Sheng
Sent: Thursday, April 07, 2016 10:41 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.

Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.

As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.

In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).

Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Wednesday, April 06, 2016 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

yes it's supported

On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Itola;

Sorry, same error.

Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?

Finding when FS runs in media mode:
codec causes caller side ?488 not acceptable here| incompatible destination ?
                callee side: ?cancel |user not registered?

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');> [mailto:freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>] On Behalf Of ?talo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Set media_mix_inbound_outbound_codecs=true

?talo Rossi
italo at freeswitch.org<javascript:_e(%7B%7D,'cvml','italo at freeswitch.org');>
IRC chat.freenode.net<http://chat.freenode.net> #freeswitch #freeswitch-dev
Bugs? https://freeswitch.org/jira
Docs? https://freeswitch.org/jira
Chat? https://hipchat.freeswitch.org/gUdAgy0m6


On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<javascript:_e(%7B%7D,'cvml','quanhs at stee.stengg.com');>> wrote:

Hi All;



I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min<http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min<http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.

If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.

But if disabling media bypass mode, call is rejected by FS.



FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.

Chrome uses opus 111, FS uses opus 116.

Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().



Help is needed to troubleshoot this issue.



Thanks advance.

Smile.



[This e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
from your system; please do not copy or use it for any purpose, nor disclose
its contents to any other person. Thank you.]
---ST Electronics Group---
-------------- next part --------------
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From mike at jerris.com  Thu Apr  7 09:52:33 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 7 Apr 2016 01:52:33 -0400
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION ERROR. SDP
In-Reply-To: <ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
Message-ID: <CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>

you keep saying

(UDP/TLS/RTP/SAVPF)

are you thinking that is supposed to be in the sip message somehow?  sounds
like you are using jssip, which is known to have issues.

On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:

> Hi;
>
>
>
> Forgot another information. Does CODEC VP8 must be included in codec_prefs
> in WebRTC regardless there is not at all video involved?
>
>
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *Quan Huo Sheng
> *Sent:* Thursday, April 07, 2016 10:41 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I
> am familiar with SIP.
>
>
>
> Well, can help to provide workable configuration. Or troubleshooting the
> issue. Thanks a lot.
>
>
>
> As understanding, CODEC NEGOTIATION DEAY is set, how does FS include
> SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome
> endpoint.
>
>
>
> In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered
> via WSS), there is no such information but normal RTP (RTP/SAVP).
>
>
>
> Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.
>
>
>
>
>
> Regards
>
> Smile.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *Michael Jerris
> *Sent:* Wednesday, April 06, 2016 11:59 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> yes it's supported
>
> On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com
> <javascript:_e(%7B%7D,'cvml','quanhs at stee.stengg.com');>> wrote:
>
> Hi Itola;
>
>
>
> Sorry, same error.
>
>
>
> Does Freeswitch support media switching (srtp-dtls) between two
> chrome(sip.js as signal) browsers?
>
>
>
> Finding when FS runs in media mode:
>
> codec causes caller side ?488 not acceptable here| incompatible
> destination ?
>
>                 callee side: ?cancel |user not registered?
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *?talo Rossi
> *Sent:* Tuesday, April 05, 2016 8:58 PM
> *To:* FreeSWITCH Users Help
> *Cc:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> Set media_mix_inbound_outbound_codecs=true
>
> ?talo Rossi
>
> italo at freeswitch.org
>
> IRC chat.freenode.net #freeswitch #freeswitch-dev
>
> Bugs? https://freeswitch.org/jira
>
> Docs? https://freeswitch.org/jira
>
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6
>
>
>
>
>
> On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com>
> wrote:
>
> Hi All;
>
>
>
> I want to use freeswitch to set up a WebRTC POC (Audio only,
> udp/tls/rtp/savp).  Setup is Anonymous (
> 192.168.199.216/chrome/sipjs:sip-0.7.0.min) ->freeswitch (192.168.199.22)
> ->1010 (192.168.199.216/chrome/sip-0.7.0.min),just following the
> information in book ?FreeSWITCH 1.6 Cookbook?.
>
> If using media bypass mode (inbound-bypass-media == true), all works fine,
> caller and called can hear each other.
>
> But if disabling media bypass mode, call is rejected by FS.
>
>
>
> FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?.
> Attachment is detailed FS log file.
>
> Chrome uses opus 111, FS uses opus 116.
>
> Mod_sofia.c::sofia_receive_message() -->
> sofia_media.c::sofia_media_negotiate_sdp() -->
> sofia_core_media.c::sofia_core_media_negotiate_sdp().
>
>
>
> Help is needed to troubleshoot this issue.
>
>
>
> Thanks advance.
>
> Smile.
>
>
>
>
> [This e-mail is confidential and may be privileged. If you are not the
> intended recipient, please kindly notify us immediately and delete the
> message
> from your system; please do not copy or use it for any purpose, nor
> disclose
> its contents to any other person. Thank you.]
> ---ST Electronics Group---
>
-------------- next part --------------
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From quanhs at stee.stengg.com  Thu Apr  7 10:01:57 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Thu, 7 Apr 2016 06:01:57 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION ERROR. SDP
In-Reply-To: <CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
Message-ID: <46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>

I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 1:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

you keep saying

(UDP/TLS/RTP/SAVPF)

are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.

On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi;

Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');> [mailto:freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>] On Behalf Of Quan Huo Sheng
Sent: Thursday, April 07, 2016 10:41 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.

Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.

As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.

In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).

Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');> [mailto:freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>] On Behalf Of Michael Jerris
Sent: Wednesday, April 06, 2016 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

yes it's supported

On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<javascript:_e(%7B%7D,'cvml','quanhs at stee.stengg.com');>> wrote:
Hi Itola;

Sorry, same error.

Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?

Finding when FS runs in media mode:
codec causes caller side ?488 not acceptable here| incompatible destination ?
                callee side: ?cancel |user not registered?

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Set media_mix_inbound_outbound_codecs=true

?talo Rossi
italo at freeswitch.org<mailto:italo at freeswitch.org>
IRC chat.freenode.net<http://chat.freenode.net> #freeswitch #freeswitch-dev
Bugs? https://freeswitch.org/jira
Docs? https://freeswitch.org/jira
Chat? https://hipchat.freeswitch.org/gUdAgy0m6


On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

Hi All;



I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min<http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min<http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.

If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.

But if disabling media bypass mode, call is rejected by FS.



FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.

Chrome uses opus 111, FS uses opus 116.

Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().



Help is needed to troubleshoot this issue.



Thanks advance.

Smile.



[This e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
from your system; please do not copy or use it for any purpose, nor disclose
its contents to any other person. Thank you.]
---ST Electronics Group---
-------------- next part --------------
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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160407/f3e748c0/attachment-0001.html 

From mike at jerris.com  Thu Apr  7 10:46:49 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 7 Apr 2016 02:46:49 -0400
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
	NEGOTIATION ERROR. SDP
In-Reply-To: <46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
Message-ID: <94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>

freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?

> On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
> 
> I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 1:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> you keep saying 
> 
> (UDP/TLS/RTP/SAVPF)
>  
> are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.
> 
> On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi;
>  
> Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved? 
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <> [mailto:freeswitch-users-bounces at lists.freeswitch.org <>] On Behalf Of Quan Huo Sheng
> Sent: Thursday, April 07, 2016 10:41 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.
>  
> Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.
>  
> As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.
>  
> In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).
>  
> Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.
>  
>  
> Regards
> Smile.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <> [mailto:freeswitch-users-bounces at lists.freeswitch.org <>] On Behalf Of Michael Jerris
> Sent: Wednesday, April 06, 2016 11:59 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> yes it's supported
> 
> On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com <>> wrote:
> Hi Itola;
>  
> Sorry, same error.
>  
> Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?
>  
> Finding when FS runs in media mode:
> codec causes caller side ?488 not acceptable here| incompatible destination ?
>                 callee side: ?cancel |user not registered?
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of ?talo Rossi
> Sent: Tuesday, April 05, 2016 8:58 PM
> To: FreeSWITCH Users Help
> Cc: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> Set media_mix_inbound_outbound_codecs=true
> 
> ?talo Rossi
> italo at freeswitch.org <mailto:italo at freeswitch.org>
> IRC chat.freenode.net <http://chat.freenode.net/> #freeswitch #freeswitch-dev
> Bugs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Docs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6 <https://hipchat.freeswitch.org/gUdAgy0m6>
>  
>  
> 
> On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi All;
>  
> I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min <http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min <http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.
> If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
> But if disabling media bypass mode, call is rejected by FS.
>  
> FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.
> Chrome uses opus 111, FS uses opus 116.
> Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().
>  
> Help is needed to troubleshoot this issue.
>  
> Thanks advance.
> Smile.
>  
> 
> [This e-mail is confidential and may be privileged. If you are not the
> intended recipient, please kindly notify us immediately and delete the message
> from your system; please do not copy or use it for any purpose, nor disclose
> its contents to any other person. Thank you.]
> ---ST Electronics Group---
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From quanhs at stee.stengg.com  Thu Apr  7 11:05:35 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Thu, 7 Apr 2016 07:05:35 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299
	CODEC	NEGOTIATION ERROR. SDP
In-Reply-To: <94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
Message-ID: <f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>

Hi Michael;

Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as WebRTC signaling.

You can see SDP from chrome (+sipjs) for this in previous attachments.

If set inbound-bypass-media=true, then chrome caller can talk with chrome callee using opus without any issue.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 2:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?

On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 1:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

you keep saying

(UDP/TLS/RTP/SAVPF)

are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.

On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi;

Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Quan Huo Sheng
Sent: Thursday, April 07, 2016 10:41 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.

Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.

As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.

In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).

Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Wednesday, April 06, 2016 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

yes it's supported

On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Itola;

Sorry, same error.

Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?

Finding when FS runs in media mode:
codec causes caller side ?488 not acceptable here| incompatible destination ?
                callee side: ?cancel |user not registered?

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Set media_mix_inbound_outbound_codecs=true

?talo Rossi
italo at freeswitch.org<mailto:italo at freeswitch.org>
IRC chat.freenode.net<http://chat.freenode.net/> #freeswitch #freeswitch-dev
Bugs? https://freeswitch.org/jira
Docs? https://freeswitch.org/jira
Chat? https://hipchat.freeswitch.org/gUdAgy0m6


On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi All;

I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min<http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min<http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.
If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
But if disabling media bypass mode, call is rejected by FS.

FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.
Chrome uses opus 111, FS uses opus 116.
Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().

Help is needed to troubleshoot this issue.

Thanks advance.
Smile.


[This e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
from your system; please do not copy or use it for any purpose, nor disclose
its contents to any other person. Thank you.]
---ST Electronics Group---
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>

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

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

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From babak.freeswitch at gmail.com  Thu Apr  7 11:56:55 2016
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Thu, 7 Apr 2016 12:26:55 +0430
Subject: [Freeswitch-users] detecting tone
Message-ID: <CAEVfcaaBPgJujRJSer8rbmg8HRHp-TGiePm_D97YfG=uhAb+Jw@mail.gmail.com>

Hi
I want to detect the tone that is attached to this email with spandsp. what
descriptor should I use. I'm trying 49 but it is not working.
thanks
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From kalyaneekulkarni at gmail.com  Thu Apr  7 14:11:37 2016
From: kalyaneekulkarni at gmail.com (Kalyani Kulkarni)
Date: Thu, 7 Apr 2016 15:41:37 +0530
Subject: [Freeswitch-users] detecting tone
In-Reply-To: <CAEVfcaaBPgJujRJSer8rbmg8HRHp-TGiePm_D97YfG=uhAb+Jw@mail.gmail.com>
References: <CAEVfcaaBPgJujRJSer8rbmg8HRHp-TGiePm_D97YfG=uhAb+Jw@mail.gmail.com>
Message-ID: <CAHrOMBkXyk78uWaFtuTAVkggmBbb3HqrPchRG89wW5R4s3s54w@mail.gmail.com>

Hi Babak,

You can set the debug level high to check which tone and duration is being
detected. From Audacity the frequency is 425 Hz so 49 descriptor reorder
should be detected. You might need to fine tune the duration looking at the
debug messages.

Hope this helps.
Kalyani

On Thu, Apr 7, 2016 at 1:26 PM, Babak Yakhchali <babak.freeswitch at gmail.com>
wrote:

> Hi
> I want to detect the tone that is attached to this email with spandsp.
> what descriptor should I use. I'm trying 49 but it is not working.
> 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 piotrek.gregor at gmail.com  Thu Apr  7 16:04:46 2016
From: piotrek.gregor at gmail.com (Piotr Gregor)
Date: Thu, 7 Apr 2016 13:04:46 +0100
Subject: [Freeswitch-users] detecting tone
In-Reply-To: <CAHrOMBkXyk78uWaFtuTAVkggmBbb3HqrPchRG89wW5R4s3s54w@mail.gmail.com>
References: <CAEVfcaaBPgJujRJSer8rbmg8HRHp-TGiePm_D97YfG=uhAb+Jw@mail.gmail.com>
	<CAHrOMBkXyk78uWaFtuTAVkggmBbb3HqrPchRG89wW5R4s3s54w@mail.gmail.com>
Message-ID: <CAO3obxbxS5Xcaj1rK7_QeyCmPBgwx_vNmOTUo41JFqgMZtmw5g@mail.gmail.com>

Hi Babak,

you can also try avmd module for a detection of single frequency sound.
I have tested avmd on your audio and it detected 425.69 Hz frequency
when it's variance threshold was set to 0.0006, 446.23 Hz with default
threshold of 0.00025.


2016-04-07 12:32:05.820909 [NOTICE] switch_channel.c:1104 New Channel
sofia/internal/1000 at 192.168.1.27 [4e41e5be-51a0-42fa-87b5-8fde76744530]
2016-04-07 12:32:06.060908 [INFO] mod_dialplan_xml.c:637 Processing
beautiful display name <1000>->1705 in context default

    api avmd 4e41e5be-51a0-42fa-87b5-8fde76744530 start

    Content-Type: api/response
    Content-Length: 89

    +OK
     [4e41e5be-51a0-42fa-87b5-8fde76744530] [sofia/internal/
1000 at 192.168.1.27] started!

2016-04-07 12:32:12.760908 [INFO] switch_core_media_bug.c:701 Sending early
media
2016-04-07 12:32:12.760908 [NOTICE] sofia_media.c:92 Pre-Answer
sofia/internal/1000 at 192.168.1.27!
2016-04-07 12:32:12.760908 [INFO] mod_avmd.c:279 Avmd session initialized,
[8000] samples/s
2016-04-07 12:32:12.760908 [INFO] mod_avmd.c:739 Avmd on channel
[sofia/internal/1000 at 192.168.1.27] started!
2016-04-07 12:32:17.480907 [NOTICE] mod_avmd.c:919 <<< AVMD - Beep
Detected: f = [425.688780], variance = [0.000533] >>>


Avmd works for single frequency tones. Be sure to checkout latest master
(this module
is yet not of a production quality at this moment).

cheers,
Piotr
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From stasan89 at gmail.com  Thu Apr  7 16:06:17 2016
From: stasan89 at gmail.com (=?UTF-8?B?0KHRgtCw0YEg0KLQtdC70YzQvdC+0LI=?=)
Date: Thu, 7 Apr 2016 15:06:17 +0300
Subject: [Freeswitch-users] Call from SIP server to external SIP provider:
	488 INCOMPATIBLE_DESTINATION
Message-ID: <CAC89h=K-syXf7KiEdo=ThoMd=0y5WUqGFP9XLT6FuTAq3SKdZw@mail.gmail.com>

Hello.
I`am doing external calls via sip provider. And I have internal SIP server
(opensips) with users.
My case: two my users connecting via my sip server. One of him starting
conference and calling to external number (carrier number or mobile phone).

But when I doing call to external number, I got error 488
INCOMPATIBLE_DESTINATION on call.

For setup I done next steps:
1) create file /etc/freeswitch/sip_profiles/external/freelycall.com.xml
with provider-access params

2) set dialplan extension with my mobile number (hardcoded expression for
testing)
    <extension name="outbound_calls_to_freelycall">
      <condition field="destination_number" expression="^(7906*******)$"
break="on-true">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="bridge"
data="sofia/gateway/freelycall.com/$1"/>

      </condition>
    </extension>

3) set default provider params and external hosts in vars.xml
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:freelycall.com"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:freelycall.com"/>
  ...
  <X-PRE-PROCESS cmd="set" data="default_provider=freelycall.com"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_username=MY_ID"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_password=MY_PASS"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=freelycall.com
"/>
  <!-- true or false -->
  <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>
  <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>

I starting debug:
freeswitch -c

command "sofia status" say:
                     Name       Type
                                      Data    State
=================================================================================================
            172.31.22.124      alias
internal    ALIASED
                 external    profile
sip:mod_sofia at 172.31.22.124:5060    RUNNING (0)
                 external    profile
sip:mod_sofia at 172.31.22.124:5061    RUNNING (0) (TLS)
 external::freelycall.com    gateway
sip:MY_ID at freelycall.com    REGED
                 internal    profile
sip:mod_sofia at 172.31.22.124:5080    RUNNING (0)
                 internal    profile
sip:mod_sofia at 172.31.22.124:5081    RUNNING (0) (TLS)
=================================================================================================
2 profiles 1 alias

Thats good I think.

command "sofia status profile external" say:
=================================================================================================
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               172.31.22.124
SIP-IP               172.31.22.124
URL                  sip:mod_sofia at 172.31.22.124:5060
BIND-URL             sip:mod_sofia at 172.31.22.124:5060;transport=udp,tcp
TLS-URL              sip:mod_sofia at 172.31.22.124:5061
TLS-BIND-URL         sips:mod_sofia at 172.31.22.124:5061;transport=tls
HOLD-MUSIC           local_stream://moh
OUTBOUND-PROXY       N/A
CODECS IN            GSM
CODECS OUT           GSM
TEL-EVENT            101
DTMF-MODE            rfc2833
CNG                  13
SESSION-TO           0
MAX-DIALOG           0
NOMEDIA              false
LATE-NEG             true
PROXY-MEDIA          false
ZRTP-PASSTHRU        true
AGGRESSIVENAT        false
CALLS-IN             0
FAILED-CALLS-IN      0
CALLS-OUT            0
FAILED-CALLS-OUT     0
REGISTRATIONS        0

I`am not sure, that all correct, but I don`t know how should look like the
right output of this command


When I starting call, I see next row in log:
2016-04-07 07:52:09.195582 [NOTICE] switch_ivr.c:2092 Transfer
sofia/external/8 at MY_OPENSIPS_DOMAIN to XML[7906*******@default]

I think, that @default is incorrect, I think that domain should be look
like as @freelycall.com

What I doing incorrect? Any idea?

Full log on call:
2016-04-07 07:52:09.195582 [NOTICE] switch_channel.c:1101 New Channel
sofia/external/8 at MY_OPENSIPS_DOMAIN [286b6a34-fcb7-11e5-a94f-99d758825eaf]
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_NEW
2016-04-07 07:52:09.195582 [DEBUG] sofia.c:9248
sofia/external/8 at MY_OPENSIPS_DOMAIN receiving invite from 172.31.0.169:5060
version: 1.6.6  64bit
2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6760 Channel
sofia/external/8 at MY_OPENSIPS_DOMAIN entering state [received][100]
2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6770 Remote SDP:
v=0
o=- 1460029927 1 IN IP4 MY_CALLER_IP
s=portsip.com
c=IN IP4 MY_OPENSIPS_IP
t=0 0
m=audio 42386 RTP/AVP 18 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=nortpproxy:yes

2016-04-07 07:52:09.195582 [DEBUG] sofia.c:7125
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_NEW -> CS_INIT
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:492
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State NEW
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_INIT
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State INIT
2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:88
sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA INIT
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:40
sofia/external/8 at MY_OPENSIPS_DOMAIN Standard INIT
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:48
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_INIT -> CS_ROUTING
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State INIT going to sleep
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_ROUTING
2016-04-07 07:52:09.195582 [DEBUG] switch_channel.c:2247
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Callstate Change DOWN -> RINGING
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING
2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141
sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA ROUTING
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:166
sofia/external/8 at MY_OPENSIPS_DOMAIN Standard ROUTING
2016-04-07 07:52:09.195582 [INFO] mod_dialplan_xml.c:637 Processing 8
<8>->7906******* in context public
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing
[public->from_opensips] continue=false
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS) [from_opensips]
network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/ break=on-false
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action
transfer(${destination_number} XML default)
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:216
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_ROUTING -> CS_EXECUTE
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING going to sleep
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_EXECUTE
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE
2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:196
sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA EXECUTE
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:258
sofia/external/8 at MY_OPENSIPS_DOMAIN Standard EXECUTE
EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN transfer(7906******* XML
default)
2016-04-07 07:52:09.195582 [DEBUG] switch_ivr.c:2085
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_EXECUTE -> CS_ROUTING
2016-04-07 07:52:09.195582 [NOTICE] switch_ivr.c:2092 Transfer
sofia/external/8 at MY_OPENSIPS_DOMAIN to XML[7906*******@default]
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE going to sleep
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_ROUTING
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING
2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141
sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA ROUTING
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:166
sofia/external/8 at MY_OPENSIPS_DOMAIN Standard ROUTING
2016-04-07 07:52:09.195582 [INFO] mod_dialplan_xml.c:637 Processing 8
<8>->7906******* in context default
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing [default->unloop]
continue=false
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing
[default->tod_example] continue=true
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Date/TimeMatch (FAIL)
[tod_example] break=on-false
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing
[default->outbound_calls_to_freelycall] continue=false
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS)
[outbound_calls_to_freelycall] destination_number(7906*******) =~
/^(7906*******)$/ break=on-true
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action
set(hangup_after_bridge=true)
Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action bridge(sofia/gateway/
freelycall.com/7906*******)
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:216
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_ROUTING -> CS_EXECUTE
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING going to sleep
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_EXECUTE
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE
2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:196
sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA EXECUTE
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:258
sofia/external/8 at MY_OPENSIPS_DOMAIN Standard EXECUTE
EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN set(hangup_after_bridge=true)
2016-04-07 07:52:09.195582 [DEBUG] mod_dptools.c:1498 SET
sofia/external/8 at MY_OPENSIPS_DOMAIN [hangup_after_bridge]=[true]
EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN bridge(sofia/gateway/
freelycall.com/7906*******)
2016-04-07 07:52:09.195582 [DEBUG] switch_ivr_originate.c:2128 Parsing
global variables
2016-04-07 07:52:09.195582 [NOTICE] switch_channel.c:1101 New Channel
sofia/external/7906******* [286bb5b6-fcb7-11e5-a957-99d758825eaf]
2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:4776
(sofia/external/7906*******) State Change CS_NEW -> CS_INIT
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/7906*******) Running State Change CS_INIT
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
(sofia/external/7906*******) State INIT
2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:88
sofia/external/7906******* SOFIA INIT
2016-04-07 07:52:09.195582 [DEBUG] sofia_glue.c:1257
sofia/external/7906******* sending invite version: 1.6.6  64bit
Local SDP:
v=0
o=FreeSWITCH 1460003999 1460004000 IN IP4 172.31.22.124
s=FreeSWITCH
c=IN IP4 172.31.22.124
t=0 0
m=audio 25930 RTP/AVP 3 101 13
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv

2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:40
sofia/external/7906******* Standard INIT
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:48
(sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
(sofia/external/7906*******) State INIT going to sleep
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/7906*******) Running State Change CS_ROUTING
2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6760 Channel
sofia/external/7906******* entering state [calling][0]
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
(sofia/external/7906*******) State ROUTING
2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141
sofia/external/7906******* SOFIA ROUTING
2016-04-07 07:52:09.195582 [DEBUG] switch_ivr_originate.c:67
(sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
(sofia/external/7906*******) State ROUTING going to sleep
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:551
(sofia/external/7906*******) State CONSUME_MEDIA
2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:551
(sofia/external/7906*******) State CONSUME_MEDIA going to sleep
2016-04-07 07:52:09.215582 [DEBUG] sofia.c:6760 Channel
sofia/external/7906******* entering state [terminated][488]
2016-04-07 07:52:09.215582 [NOTICE] sofia.c:7779 Hangup
sofia/external/7906******* [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/7906*******) Running State Change CS_HANGUP
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:739
(sofia/external/7906*******) Callstate Change DOWN -> HANGUP
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
(sofia/external/7906*******) State HANGUP
2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:431 Channel
sofia/external/7906******* hanging up, cause: INCOMPATIBLE_DESTINATION
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:60
sofia/external/7906******* Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
(sofia/external/7906*******) State HANGUP going to sleep
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:508
(sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/7906*******) Running State Change CS_REPORTING
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
(sofia/external/7906*******) State REPORTING
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:104
sofia/external/7906******* Standard REPORTING, cause:
INCOMPATIBLE_DESTINATION
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
(sofia/external/7906*******) State REPORTING going to sleep
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:499
(sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY
2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:1646 Session 2
(sofia/external/7906*******) Locked, Waiting on external entities
2016-04-07 07:52:09.215582 [DEBUG] switch_ivr_originate.c:3751 Originate
Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
2016-04-07 07:52:09.215582 [INFO] mod_dptools.c:3379 Originate Failed.
Cause: INCOMPATIBLE_DESTINATION
2016-04-07 07:52:09.215582 [NOTICE] switch_channel.c:4804 Hangup
sofia/external/8 at MY_OPENSIPS_DOMAIN [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:2796
sofia/external/8 at MY_OPENSIPS_DOMAIN skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:539
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE going to sleep
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_HANGUP
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:739
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Callstate Change RINGING -> HANGUP
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State HANGUP
2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:425
sofia/external/8 at MY_OPENSIPS_DOMAIN Overriding SIP cause 488 with 488 from
the other leg
2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:431 Channel
sofia/external/8 at MY_OPENSIPS_DOMAIN hanging up, cause:
INCOMPATIBLE_DESTINATION
2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:568 Responding to INVITE
with: 488
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:60
sofia/external/8 at MY_OPENSIPS_DOMAIN Standard HANGUP, cause:
INCOMPATIBLE_DESTINATION
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State HANGUP going to sleep
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:508
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_HANGUP -> CS_REPORTING
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_REPORTING
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State REPORTING
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:104
sofia/external/8 at MY_OPENSIPS_DOMAIN Standard REPORTING, cause:
INCOMPATIBLE_DESTINATION
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State REPORTING going to sleep
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:499
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_REPORTING ->
CS_DESTROY
2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:1646 Session 1
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Locked, Waiting on external entities
2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1664 Session 1
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Ended
2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1668 Close
Channel sofia/external/8 at MY_OPENSIPS_DOMAIN [CS_DESTROY]
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:630
(sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_DESTROY
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State DESTROY
2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:341
sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA DESTROY
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:111
sofia/external/8 at MY_OPENSIPS_DOMAIN Standard DESTROY
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
(sofia/external/8 at MY_OPENSIPS_DOMAIN) State DESTROY going to sleep
2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1664 Session 2
(sofia/external/7906*******) Ended
2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1668 Close
Channel sofia/external/7906******* [CS_DESTROY]
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:630
(sofia/external/7906*******) Running State Change CS_DESTROY
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
(sofia/external/7906*******) State DESTROY
2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:341
sofia/external/7906******* SOFIA DESTROY
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:111
sofia/external/7906******* Standard DESTROY
2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
(sofia/external/7906*******) State DESTROY going to sleep
-------------- next part --------------
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From adahary at gmail.com  Thu Apr  7 16:07:11 2016
From: adahary at gmail.com (Assaf Dahary)
Date: Thu, 07 Apr 2016 15:07:11 +0300
Subject: [Freeswitch-users] extended FS as ZRTP client
In-Reply-To: <F486B3C0-9B54-44C8-AA46-2E8C8513ECFC@jerris.com>
References: <001401d188eb$6dbba080$4932e180$@gmail.com>
	<F486B3C0-9B54-44C8-AA46-2E8C8513ECFC@jerris.com>
Message-ID: <982a4c67-f1e7-4ebb-ab0d-48d2f6a9f701@typeapp.com>

I've reinstalled both FS's, client and main, with the latest version 1.6.7.

None ZRTP SIP phone -> FS client -> NAT Internet -> FS main -> CSipSimple ZRTP enabled).

All set to PCMU codec (no transcoding).
All connect on same network (no NAT).

The FS client is setup with media-proxy=false (dialplan and profiles).

And

The FS main is setup with media-proxy=true.

I still get the same problem, I get zrtp CRC warning on the FS client - and no end-to-end encryption.

As before (with version 1.4) the FS client is NOT sending any zrtp hash, as I suppose it should do.

When FS main is set to media-proxy=false then there are no waning as it now acting as MITM.

Any tip/advise on the right direction will be very much appreciated.

Regards

Assaf



On Mar 28, 2016, 18:13, at 18:13, Michael Jerris <mike at jerris.com> wrote:
>did you try and see if this is the case with a current release instead
>of the old one?
>
>> On Mar 28, 2016, at 8:14 AM, Assaf Dahary <adahary at gmail.com> wrote:
>> 
>> Hi,
>>  
>> I would like to use FS (multiple) as a ZRTP client register on a the
>main FS.
>>  
>> I have already managed to setup a Gateway with user/pass on the FS
>client and register it on the main FS for regular incoming/outgoing
>calls (without ZRTP).
>>  
>> To enable ZRTP calls I setup the FS client/main as follow:
>> None ZRTP SIP phone -> FS client -> NAT Internet -> FS main ->
>CSipSimple ZRTP enabled).
>>  
>> FS Client (ver 1.4) :
>> ZRTP enabled globally in VARS and in dialplan.
>> Media proxy disabled on both internal and external profiles and in
>dialplan.
>>  
>> FS Main (ver 1.4):
>> Media proxy enabled ? including late negotiation.
>>  
>> I forced the FS client and the CSipSimple to use only PCMU codec to
>avoid transcoding.
>>  
>> The problem is that on a call from the SIP phone via the FS client
>there are always CRC errors on the ZRTP log.
>> Only if the FS main is set to disable media proxy then there are no
>CRC errors ? but then it becomes a MITM with incompatible SASs.
>>  
>> From reading other posts about FS and ZRTP CRC errors I assume that
>it happens because the FS client is not creating a zrtp hash in the
>invite SDP.
>>  
>> So my question is how to make the FS client to generate the zrtp hash
>in the invite SDP to act as real ZRTP enabled client? 
>> I've already tried to set the FS client internal/external/dialplan
>with several zrtp configs with no success.
>>  
>> I would appreciate any tip to resolve this issue.
>>  
>> Regards
>>  
>> Assaf  
>>  
>>  
>> The client FS 
>>  
>>
>_________________________________________________________________________
>> 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
>
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>FreeSWITCH-users at lists.freeswitch.org
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From ahabiba at gmail.com  Thu Apr  7 16:17:31 2016
From: ahabiba at gmail.com (Ahmed habiba)
Date: Thu, 7 Apr 2016 15:17:31 +0300
Subject: [Freeswitch-users] Call from SIP server to external SIP
	provider: 488 INCOMPATIBLE_DESTINATION
In-Reply-To: <mailman.266.1460030809.8223.freeswitch-users@lists.freeswitch.org>
References: <mailman.266.1460030809.8223.freeswitch-users@lists.freeswitch.org>
Message-ID: <313CE2EA-5E22-4F62-99E6-A454C467E3A3@gmail.com>

It looks that the profile you are using use only GSM, however the phone is trying to use G729 only 

look at the colored text.

> On Apr 7, 2016, at 3:06 PM, freeswitch-users-request at lists.freeswitch.org wrote:
> 
> From: ???? ??????? <stasan89 at gmail.com <mailto:stasan89 at gmail.com>>
> Subject: [Freeswitch-users] Call from SIP server to external SIP provider: 488 INCOMPATIBLE_DESTINATION
> Date: April 7, 2016 at 3:06:17 PM GMT+3
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> 
> 
> Hello.
> I`am doing external calls via sip provider. And I have internal SIP server (opensips) with users.
> My case: two my users connecting via my sip server. One of him starting conference and calling to external number (carrier number or mobile phone).
> 
> But when I doing call to external number, I got error 488 INCOMPATIBLE_DESTINATION on call.
> 
> For setup I done next steps:
> 1) create file /etc/freeswitch/sip_profiles/external/freelycall.com.xml with provider-access params
> 
> 2) set dialplan extension with my mobile number (hardcoded expression for testing)
>     <extension name="outbound_calls_to_freelycall">       
>       <condition field="destination_number" expression="^(7906*******)$" break="on-true">                 
>         <action application="set" data="hangup_after_bridge=true"/>         
>         <action application="bridge" data="sofia/gateway/freelycall.com/$1 <http://freelycall.com/$1>"/>    
>       </condition>     
>     </extension>
>     
> 3) set default provider params and external hosts in vars.xml
>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:freelycall.com <http://freelycall.com/>"/>
>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:freelycall.com <http://freelycall.com/>"/>
>   ...
>   <X-PRE-PROCESS cmd="set" data="default_provider=freelycall.com <http://freelycall.com/>"/>
>   <X-PRE-PROCESS cmd="set" data="default_provider_username=MY_ID"/>
>   <X-PRE-PROCESS cmd="set" data="default_provider_password=MY_PASS"/>
>   <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=freelycall.com <http://freelycall.com/>"/>
>   <!-- true or false -->
>   <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>
>   <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>
>   
> I starting debug:
> freeswitch -c
> 
> command "sofia status" say:
>                      Name       Type                                          Data    State
> =================================================================================================
>             172.31.22.124      alias                                      internal    ALIASED
>                  external    profile              sip:mod_sofia at 172.31.22.124:5060 <http://sip:mod_sofia at 172.31.22.124:5060/>    RUNNING (0)
>                  external    profile              sip:mod_sofia at 172.31.22.124:5061 <http://sip:mod_sofia at 172.31.22.124:5061/>    RUNNING (0) (TLS)
>  external::freelycall.com <http://freelycall.com/>    gateway                      sip:MY_ID at freelycall.com <mailto:sip%3AMY_ID at freelycall.com>    REGED
>                  internal    profile              sip:mod_sofia at 172.31.22.124:5080 <http://sip:mod_sofia at 172.31.22.124:5080/>    RUNNING (0)
>                  internal    profile              sip:mod_sofia at 172.31.22.124:5081 <http://sip:mod_sofia at 172.31.22.124:5081/>    RUNNING (0) (TLS)
> =================================================================================================
> 2 profiles 1 alias
> 
> Thats good I think.
> 
> command "sofia status profile external" say:
> =================================================================================================
> 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               172.31.22.124
> SIP-IP               172.31.22.124
> URL                  sip:mod_sofia at 172.31.22.124:5060 <http://sip:mod_sofia at 172.31.22.124:5060/>
> BIND-URL             sip:mod_sofia at 172.31.22.124:5060;transport=udp,tcp <sip:mod_sofia at 172.31.22.124:5060;transport=udp,tcp>
> TLS-URL              sip:mod_sofia at 172.31.22.124:5061 <http://sip:mod_sofia at 172.31.22.124:5061/>
> TLS-BIND-URL         sips:mod_sofia at 172.31.22.124:5061;transport=tls
> HOLD-MUSIC           local_stream://moh
> OUTBOUND-PROXY       N/A
> CODECS IN            GSM
> CODECS OUT           GSM
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             true
> PROXY-MEDIA          false
> ZRTP-PASSTHRU        true
> AGGRESSIVENAT        false
> CALLS-IN             0
> FAILED-CALLS-IN      0
> CALLS-OUT            0
> FAILED-CALLS-OUT     0
> REGISTRATIONS        0
> 
> I`am not sure, that all correct, but I don`t know how should look like the right output of this command
> 
> 
> When I starting call, I see next row in log:
> 2016-04-07 07:52:09.195582 [NOTICE] switch_ivr.c:2092 Transfer sofia/external/8 at MY_OPENSIPS_DOMAIN to XML[7906*******@default]
> 
> I think, that @default is incorrect, I think that domain should be look like as @freelycall.com <http://freelycall.com/>
> 
> What I doing incorrect? Any idea?
> 
> Full log on call:
> 2016-04-07 07:52:09.195582 [NOTICE] switch_channel.c:1101 New Channel sofia/external/8 at MY_OPENSIPS_DOMAIN [286b6a34-fcb7-11e5-a94f-99d758825eaf]
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_NEW
> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:9248 sofia/external/8 at MY_OPENSIPS_DOMAIN receiving invite from 172.31.0.169:5060 <http://172.31.0.169:5060/> version: 1.6.6  64bit
> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6760 Channel sofia/external/8 at MY_OPENSIPS_DOMAIN entering state [received][100]
> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6770 Remote SDP:
> v=0
> o=- 1460029927 1 IN IP4 MY_CALLER_IP
> s=portsip.com <http://portsip.com/>
> c=IN IP4 MY_OPENSIPS_IP
> t=0 0
> m=audio 42386 RTP/AVP 18 3 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=nortpproxy:yes
> 
> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:7125 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_NEW -> CS_INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:492 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State NEW
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State INIT
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:88 sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:40 sofia/external/8 at MY_OPENSIPS_DOMAIN Standard INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:48 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_INIT -> CS_ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State INIT going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_channel.c:2247 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Callstate Change DOWN -> RINGING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141 sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:166 sofia/external/8 at MY_OPENSIPS_DOMAIN Standard ROUTING
> 2016-04-07 07:52:09.195582 [INFO] mod_dialplan_xml.c:637 Processing 8 <8>->7906******* in context public
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing [public->from_opensips] continue=false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS) [from_opensips] network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/ break=on-false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action transfer(${destination_number} XML default) 
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:216 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_ROUTING -> CS_EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:196 sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:258 sofia/external/8 at MY_OPENSIPS_DOMAIN Standard EXECUTE
> EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN transfer(7906******* XML default)
> 2016-04-07 07:52:09.195582 [DEBUG] switch_ivr.c:2085 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_EXECUTE -> CS_ROUTING
> 2016-04-07 07:52:09.195582 [NOTICE] switch_ivr.c:2092 Transfer sofia/external/8 at MY_OPENSIPS_DOMAIN to XML[7906*******@default]
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141 sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:166 sofia/external/8 at MY_OPENSIPS_DOMAIN Standard ROUTING
> 2016-04-07 07:52:09.195582 [INFO] mod_dialplan_xml.c:637 Processing 8 <8>->7906******* in context default
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing [default->unloop] continue=false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing [default->tod_example] continue=true
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Date/TimeMatch (FAIL) [tod_example] break=on-false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing [default->outbound_calls_to_freelycall] continue=false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS) [outbound_calls_to_freelycall] destination_number(7906*******) =~ /^(7906*******)$/ break=on-true
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action set(hangup_after_bridge=true) 
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action bridge(sofia/gateway/freelycall.com/7906******* <http://freelycall.com/7906*******>) 
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:216 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_ROUTING -> CS_EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:196 sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:258 sofia/external/8 at MY_OPENSIPS_DOMAIN Standard EXECUTE
> EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN set(hangup_after_bridge=true)
> 2016-04-07 07:52:09.195582 [DEBUG] mod_dptools.c:1498 SET sofia/external/8 at MY_OPENSIPS_DOMAIN [hangup_after_bridge]=[true]
> EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN bridge(sofia/gateway/freelycall.com/7906******* <http://freelycall.com/7906*******>)
> 2016-04-07 07:52:09.195582 [DEBUG] switch_ivr_originate.c:2128 Parsing global variables
> 2016-04-07 07:52:09.195582 [NOTICE] switch_channel.c:1101 New Channel sofia/external/7906******* [286bb5b6-fcb7-11e5-a957-99d758825eaf]
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:4776 (sofia/external/7906*******) State Change CS_NEW -> CS_INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/7906*******) Running State Change CS_INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516 (sofia/external/7906*******) State INIT
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:88 sofia/external/7906******* SOFIA INIT
> 2016-04-07 07:52:09.195582 [DEBUG] sofia_glue.c:1257 sofia/external/7906******* sending invite version: 1.6.6  64bit
> Local SDP:
> v=0
> o=FreeSWITCH 1460003999 1460004000 IN IP4 172.31.22.124
> s=FreeSWITCH
> c=IN IP4 172.31.22.124
> t=0 0
> m=audio 25930 RTP/AVP 3 101 13
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
> a=sendrecv
> 
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:40 sofia/external/7906******* Standard INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:48 (sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516 (sofia/external/7906*******) State INIT going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/7906*******) Running State Change CS_ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6760 Channel sofia/external/7906******* entering state [calling][0]
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532 (sofia/external/7906*******) State ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141 sofia/external/7906******* SOFIA ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_ivr_originate.c:67 (sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532 (sofia/external/7906*******) State ROUTING going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:551 (sofia/external/7906*******) State CONSUME_MEDIA
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:551 (sofia/external/7906*******) State CONSUME_MEDIA going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] sofia.c:6760 Channel sofia/external/7906******* entering state [terminated][488]
> 2016-04-07 07:52:09.215582 [NOTICE] sofia.c:7779 Hangup sofia/external/7906******* [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/7906*******) Running State Change CS_HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:739 (sofia/external/7906*******) Callstate Change DOWN -> HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741 (sofia/external/7906*******) State HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:431 Channel sofia/external/7906******* hanging up, cause: INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:60 sofia/external/7906******* Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741 (sofia/external/7906*******) State HANGUP going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:508 (sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/7906*******) Running State Change CS_REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827 (sofia/external/7906*******) State REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:104 sofia/external/7906******* Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827 (sofia/external/7906*******) State REPORTING going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:499 (sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:1646 Session 2 (sofia/external/7906*******) Locked, Waiting on external entities
> 2016-04-07 07:52:09.215582 [DEBUG] switch_ivr_originate.c:3751 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> 2016-04-07 07:52:09.215582 [INFO] mod_dptools.c:3379 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [NOTICE] switch_channel.c:4804 Hangup sofia/external/8 at MY_OPENSIPS_DOMAIN [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:2796 sofia/external/8 at MY_OPENSIPS_DOMAIN skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:539 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:739 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Callstate Change RINGING -> HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:425 sofia/external/8 at MY_OPENSIPS_DOMAIN Overriding SIP cause 488 with 488 from the other leg
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:431 Channel sofia/external/8 at MY_OPENSIPS_DOMAIN hanging up, cause: INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:568 Responding to INVITE with: 488
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:60 sofia/external/8 at MY_OPENSIPS_DOMAIN Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State HANGUP going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:508 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_HANGUP -> CS_REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:104 sofia/external/8 at MY_OPENSIPS_DOMAIN Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State REPORTING going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:499 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_REPORTING -> CS_DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:1646 Session 1 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Locked, Waiting on external entities
> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1664 Session 1 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Ended
> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1668 Close Channel sofia/external/8 at MY_OPENSIPS_DOMAIN [CS_DESTROY]
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:630 (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:341 sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:111 sofia/external/8 at MY_OPENSIPS_DOMAIN Standard DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640 (sofia/external/8 at MY_OPENSIPS_DOMAIN) State DESTROY going to sleep
> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1664 Session 2 (sofia/external/7906*******) Ended
> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1668 Close Channel sofia/external/7906******* [CS_DESTROY]
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:630 (sofia/external/7906*******) Running State Change CS_DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640 (sofia/external/7906*******) State DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:341 sofia/external/7906******* SOFIA DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:111 sofia/external/7906******* Standard DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640 (sofia/external/7906*******) State DESTROY going to sleep

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From stasan89 at gmail.com  Thu Apr  7 17:39:02 2016
From: stasan89 at gmail.com (=?UTF-8?B?0KHRgtCw0YEg0KLQtdC70YzQvdC+0LI=?=)
Date: Thu, 7 Apr 2016 16:39:02 +0300
Subject: [Freeswitch-users] Call from SIP server to external SIP
 provider: 488 INCOMPATIBLE_DESTINATION
In-Reply-To: <313CE2EA-5E22-4F62-99E6-A454C467E3A3@gmail.com>
References: <mailman.266.1460030809.8223.freeswitch-users@lists.freeswitch.org>
	<313CE2EA-5E22-4F62-99E6-A454C467E3A3@gmail.com>
Message-ID: <CAC89h=KaEvzXyETijxPC_tEVn9c6h9yyzDG7gqFFVuCwdCcK7Q@mail.gmail.com>

Ahmed, thank you.

I did not know that the gsm codec is not suitable for VOIP providers.

2016-04-07 15:17 GMT+03:00 Ahmed habiba <ahabiba at gmail.com>:

> It looks that the profile you are using use only GSM, however the phone is
> trying to use G729 only
>
> look at the colored text.
>
> On Apr 7, 2016, at 3:06 PM, freeswitch-users-request at lists.freeswitch.org
> wrote:
>
> *From: *???? ??????? <stasan89 at gmail.com>
> *Subject: **[Freeswitch-users] Call from SIP server to external SIP
> provider: 488 INCOMPATIBLE_DESTINATION*
> *Date: *April 7, 2016 at 3:06:17 PM GMT+3
> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> Hello.
> I`am doing external calls via sip provider. And I have internal SIP server
> (opensips) with users.
> My case: two my users connecting via my sip server. One of him starting
> conference and calling to external number (carrier number or mobile phone).
>
> But when I doing call to external number, I got error 488
> INCOMPATIBLE_DESTINATION on call.
>
> For setup I done next steps:
> 1) create file /etc/freeswitch/sip_profiles/external/freelycall.com.xml
> with provider-access params
>
> 2) set dialplan extension with my mobile number (hardcoded expression for
> testing)
>     <extension name="outbound_calls_to_freelycall">
>       <condition field="destination_number" expression="^(7906*******)$"
> break="on-true">
>         <action application="set" data="hangup_after_bridge=true"/>
>
>         <action application="bridge" data="sofia/gateway/freelycall.com/$1
> "/>
>       </condition>
>     </extension>
>
> 3) set default provider params and external hosts in vars.xml
>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:freelycall.com"/>
>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:freelycall.com"/>
>   ...
>   <X-PRE-PROCESS cmd="set" data="default_provider=freelycall.com"/>
>   <X-PRE-PROCESS cmd="set" data="default_provider_username=MY_ID"/>
>   <X-PRE-PROCESS cmd="set" data="default_provider_password=MY_PASS"/>
>   <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=
> freelycall.com"/>
>   <!-- true or false -->
>   <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>
>   <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>
>
> I starting debug:
> freeswitch -c
>
> command "sofia status" say:
>                      Name       Type
>                                       Data    State
>
> =================================================================================================
>             172.31.22.124      alias
> internal    ALIASED
>                  external    profile
> sip:mod_sofia at 172.31.22.124:5060    RUNNING (0)
>                  external    profile
> sip:mod_sofia at 172.31.22.124:5061    RUNNING (0) (TLS)
>  external::freelycall.com    gateway
> sip:MY_ID at freelycall.com    REGED
>                  internal    profile
> sip:mod_sofia at 172.31.22.124:5080    RUNNING (0)
>                  internal    profile
> sip:mod_sofia at 172.31.22.124:5081    RUNNING (0) (TLS)
>
> =================================================================================================
> 2 profiles 1 alias
>
> Thats good I think.
>
> command "sofia status profile external" say:
>
> =================================================================================================
> 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               172.31.22.124
> SIP-IP               172.31.22.124
> URL                  sip:mod_sofia at 172.31.22.124:5060
> BIND-URL             sip:mod_sofia at 172.31.22.124:5060;transport=udp,tcp
> TLS-URL              sip:mod_sofia at 172.31.22.124:5061
> TLS-BIND-URL         sips:mod_sofia at 172.31.22.124:5061;transport=tls
> HOLD-MUSIC           local_stream://moh
> OUTBOUND-PROXY       N/A
> CODECS IN            GSM
> CODECS OUT           GSM
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             true
> PROXY-MEDIA          false
> ZRTP-PASSTHRU        true
> AGGRESSIVENAT        false
> CALLS-IN             0
> FAILED-CALLS-IN      0
> CALLS-OUT            0
> FAILED-CALLS-OUT     0
> REGISTRATIONS        0
>
> I`am not sure, that all correct, but I don`t know how should look like the
> right output of this command
>
>
> When I starting call, I see next row in log:
> 2016-04-07 07:52:09.195582 [NOTICE] switch_ivr.c:2092 Transfer
> sofia/external/8 at MY_OPENSIPS_DOMAIN to XML[7906*******@default]
>
> I think, that @default is incorrect, I think that domain should be look
> like as @freelycall.com
>
> What I doing incorrect? Any idea?
>
> Full log on call:
> 2016-04-07 07:52:09.195582 [NOTICE] switch_channel.c:1101 New Channel
> sofia/external/8 at MY_OPENSIPS_DOMAIN [286b6a34-fcb7-11e5-a94f-99d758825eaf]
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_NEW
> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:9248
> sofia/external/8 at MY_OPENSIPS_DOMAIN receiving invite from
> 172.31.0.169:5060 version: 1.6.6  64bit
> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6760 Channel
> sofia/external/8 at MY_OPENSIPS_DOMAIN entering state [received][100]
> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6770 Remote SDP:
> v=0
> o=- 1460029927 1 IN IP4 MY_CALLER_IP
> s=portsip.com
> c=IN IP4 MY_OPENSIPS_IP
> t=0 0
> m=audio 42386 RTP/AVP 18 3 101
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=nortpproxy:yes
>
> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:7125
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_NEW -> CS_INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:492
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State NEW
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State INIT
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:88
> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:40
> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:48
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_INIT -> CS_ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State INIT going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_channel.c:2247
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Callstate Change DOWN -> RINGING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141
> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:166
> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard ROUTING
> 2016-04-07 07:52:09.195582 [INFO] mod_dialplan_xml.c:637 Processing 8
> <8>->7906******* in context public
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing
> [public->from_opensips] continue=false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS)
> [from_opensips] network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/
> break=on-false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action
> transfer(${destination_number} XML default)
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:216
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_ROUTING ->
> CS_EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:196
> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:258
> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard EXECUTE
> EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN transfer(7906******* XML
> default)
> 2016-04-07 07:52:09.195582 [DEBUG] switch_ivr.c:2085
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_EXECUTE ->
> CS_ROUTING
> 2016-04-07 07:52:09.195582 [NOTICE] switch_ivr.c:2092 Transfer
> sofia/external/8 at MY_OPENSIPS_DOMAIN to XML[7906*******@default]
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141
> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:166
> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard ROUTING
> 2016-04-07 07:52:09.195582 [INFO] mod_dialplan_xml.c:637 Processing 8
> <8>->7906******* in context default
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing [default->unloop]
> continue=false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing
> [default->tod_example] continue=true
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Date/TimeMatch (FAIL)
> [tod_example] break=on-false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing
> [default->outbound_calls_to_freelycall] continue=false
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS)
> [outbound_calls_to_freelycall] destination_number(7906*******) =~
> /^(7906*******)$/ break=on-true
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action
> set(hangup_after_bridge=true)
> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action bridge(sofia/gateway/
> freelycall.com/7906*******)
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:216
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_ROUTING ->
> CS_EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:196
> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA EXECUTE
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:258
> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard EXECUTE
> EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN set(hangup_after_bridge=true)
> 2016-04-07 07:52:09.195582 [DEBUG] mod_dptools.c:1498 SET
> sofia/external/8 at MY_OPENSIPS_DOMAIN [hangup_after_bridge]=[true]
> EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN bridge(sofia/gateway/
> freelycall.com/7906*******)
> 2016-04-07 07:52:09.195582 [DEBUG] switch_ivr_originate.c:2128 Parsing
> global variables
> 2016-04-07 07:52:09.195582 [NOTICE] switch_channel.c:1101 New Channel
> sofia/external/7906******* [286bb5b6-fcb7-11e5-a957-99d758825eaf]
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:4776
> (sofia/external/7906*******) State Change CS_NEW -> CS_INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/7906*******) Running State Change CS_INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
> (sofia/external/7906*******) State INIT
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:88
> sofia/external/7906******* SOFIA INIT
> 2016-04-07 07:52:09.195582 [DEBUG] sofia_glue.c:1257
> sofia/external/7906******* sending invite version: 1.6.6  64bit
> Local SDP:
> v=0
> o=FreeSWITCH 1460003999 1460004000 IN IP4 172.31.22.124
> s=FreeSWITCH
> c=IN IP4 172.31.22.124
> t=0 0
> m=audio 25930 RTP/AVP 3 101 13
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
> a=sendrecv
>
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:40
> sofia/external/7906******* Standard INIT
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:48
> (sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
> (sofia/external/7906*******) State INIT going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/7906*******) Running State Change CS_ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6760 Channel
> sofia/external/7906******* entering state [calling][0]
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/7906*******) State ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141
> sofia/external/7906******* SOFIA ROUTING
> 2016-04-07 07:52:09.195582 [DEBUG] switch_ivr_originate.c:67
> (sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/7906*******) State ROUTING going to sleep
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:551
> (sofia/external/7906*******) State CONSUME_MEDIA
> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:551
> (sofia/external/7906*******) State CONSUME_MEDIA going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] sofia.c:6760 Channel
> sofia/external/7906******* entering state [terminated][488]
> 2016-04-07 07:52:09.215582 [NOTICE] sofia.c:7779 Hangup
> sofia/external/7906******* [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/7906*******) Running State Change CS_HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:739
> (sofia/external/7906*******) Callstate Change DOWN -> HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
> (sofia/external/7906*******) State HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:431 Channel
> sofia/external/7906******* hanging up, cause: INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:60
> sofia/external/7906******* Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
> (sofia/external/7906*******) State HANGUP going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:508
> (sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/7906*******) Running State Change CS_REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
> (sofia/external/7906*******) State REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:104
> sofia/external/7906******* Standard REPORTING, cause:
> INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
> (sofia/external/7906*******) State REPORTING going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:1646 Session 2
> (sofia/external/7906*******) Locked, Waiting on external entities
> 2016-04-07 07:52:09.215582 [DEBUG] switch_ivr_originate.c:3751 Originate
> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> 2016-04-07 07:52:09.215582 [INFO] mod_dptools.c:3379 Originate Failed.
> Cause: INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [NOTICE] switch_channel.c:4804 Hangup
> sofia/external/8 at MY_OPENSIPS_DOMAIN [CS_EXECUTE]
> [INCOMPATIBLE_DESTINATION]
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:2796
> sofia/external/8 at MY_OPENSIPS_DOMAIN skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:539
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:739
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Callstate Change RINGING -> HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State HANGUP
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:425
> sofia/external/8 at MY_OPENSIPS_DOMAIN Overriding SIP cause 488 with 488
> from the other leg
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:431 Channel
> sofia/external/8 at MY_OPENSIPS_DOMAIN hanging up, cause:
> INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:568 Responding to INVITE
> with: 488
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:60
> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard HANGUP, cause:
> INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State HANGUP going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:508
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_HANGUP ->
> CS_REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State REPORTING
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:104
> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard REPORTING, cause:
> INCOMPATIBLE_DESTINATION
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State REPORTING going to sleep
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_REPORTING ->
> CS_DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:1646 Session 1
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Locked, Waiting on external entities
> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1664 Session 1
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Ended
> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1668 Close
> Channel sofia/external/8 at MY_OPENSIPS_DOMAIN [CS_DESTROY]
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:630
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:341
> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:111
> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State DESTROY going to sleep
> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1664 Session 2
> (sofia/external/7906*******) Ended
> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1668 Close
> Channel sofia/external/7906******* [CS_DESTROY]
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:630
> (sofia/external/7906*******) Running State Change CS_DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
> (sofia/external/7906*******) State DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:341
> sofia/external/7906******* SOFIA DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:111
> sofia/external/7906******* Standard DESTROY
> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
> (sofia/external/7906*******) State DESTROY going to sleep
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From saurabhkv01 at gmail.com  Wed Apr  6 22:11:24 2016
From: saurabhkv01 at gmail.com (saurabh verrma)
Date: Wed, 6 Apr 2016 23:41:24 +0530
Subject: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
In-Reply-To: <A84CA188-7849-4678-98DC-33BA615A9A0C@jerris.com>
References: <CACot2ntTB+7OXvomm3sxMzxnNEnT6vHtQb=T=m8MTOFT0=zFZg@mail.gmail.com>
	<E93C0E00-0EEE-44E1-81F2-1E8CF374057B@jerris.com>
	<CACot2ntmtZzcXs4bWqd3oW9i5URXMpqaouu8EXKWeN2thDJG4A@mail.gmail.com>
	<CACot2nvgSJOFj9HJ2JOc4Ofp1Ce_x-mZ0G95epFJA1AwMZ2X5g@mail.gmail.com>
	<A84CA188-7849-4678-98DC-33BA615A9A0C@jerris.com>
Message-ID: <CACot2ntmk5aUM9kyLXwD3C+1phvib-TLYaszW3cvTm7Texg9VQ@mail.gmail.com>

Thanks Michael,

What do you exactly mean by client & server application.?
Is it some type of SIP server/stack(server application) & SIP
endpoint(client application) that you're referring ?

Also could you please explain a bit what's the limitation in ICE in case of
client application.

On Wed, Apr 6, 2016 at 9:52 PM, Michael Jerris <mike at jerris.com> wrote:

> is this for a client or server application?  If it is for a client
> application, we may not be the right stack for you.  We make a number of
> assumptions in our ice that is based on it being a server in order to
> improve server efficiency.
>
> On Apr 6, 2016, at 5:01 AM, saurabh verrma <saurabhkv01 at gmail.com> wrote:
>
> Hi,
>
> I?m working on an application where I?m trying to use FreeSWITCH as a
> library. My intention is to use FreeSWITCH as a UAS endpoint. Basically it
> needs to be supporting following:
> 1. WebRTC
> 2. Ability to act like UAS endpoint
> 3. Support for DTLS/SRTP
> 4. ICE support
>
> I?m seeking community suggestion if that?s feasible to implement or not?
> If yes, what are the possible starting directions we could explore above
> points.
>
> Any help would be greatly appreciated.
>
>
> On Sat, Apr 2, 2016 at 12:24 PM, saurabh verrma <saurabhkv01 at gmail.com>
> wrote:
>
>> Thanks Michael,
>>
>> Basically we're writing a PJSIP based application & PJSIP doesn't have
>> DTLS support. So we're thinking to use FreeSWITCH library for DTLS/SRTP.
>>
>> On Fri, Apr 1, 2016 at 7:29 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> we have full support for webrtc media profile which would include these
>>> features  There are not a ton of people who use freeswitch as a library,
>>> but it is built that way so that you can control it and host it in another
>>> application instead of stand alone.  If you are trying to accomplish
>>> something I'd try to handle it standalone first so you can learn all the
>>> different ways you might control it before architecting a solution
>>>
>>> On Apr 1, 2016, at 9:48 AM, saurabh verrma <saurabhkv01 at gmail.com>
>>> wrote:
>>>
>>> Hi All,
>>>
>>> I want to use FreeSWITCH library for DTLS/SRTP support. I want to know
>>> in freeswitch which library has the support of these features(SRTP/DTLS) ?
>>> Is there any application available based on freeswitch library ?
>>>
>>> Any help would be appreciable.
>>>
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Warm Regards:*
*Saurabh Kumar Verma*
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From saurabhkv01 at gmail.com  Thu Apr  7 11:17:56 2016
From: saurabhkv01 at gmail.com (saurabh verrma)
Date: Thu, 7 Apr 2016 12:47:56 +0530
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION ERROR. SDP
In-Reply-To: <f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
Message-ID: <CACot2nsE48=5pKZs25g3=_syjj91sTN=giYqAyX4Qu-gqhLHHA@mail.gmail.com>

Hi Michael,

I?ve been working on having FreeSWITCH interact with JsSIP for WebRTC
calling. You?ve mentioned here that JsSIP is known to be having issues.

Could you please point me to some link which gives details about the issue
or do the favour of telling me what are the known issues in FreeSWITCH +
JsSIP.

It would be appreciable.

On Thu, Apr 7, 2016 at 12:35 PM, Quan Huo Sheng <quanhs at stee.stengg.com>
wrote:

> Hi Michael;
>
>
>
> Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only
> supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as
> WebRTC signaling.
>
>
>
> You can see SDP from chrome (+sipjs) for this in previous attachments.
>
>
>
> If set inbound-bypass-media=true, then chrome caller can talk with chrome
> callee using opus without any issue.
>
>
>
>
>
> Regards
>
> Smile.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Thursday, April 07, 2016 2:47 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an
> sdp.  Can you please explain what exactly you think is missing from the sdp
> in our offer?
>
>
>
> On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
>
>
> I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6
> cookbook |author: Anthony Minessale ?), not jssip.
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Michael
> Jerris
> *Sent:* Thursday, April 07, 2016 1:53 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> you keep saying
>
> (UDP/TLS/RTP/SAVPF)
>
>
>
> are you thinking that is supposed to be in the sip message somehow?
>  sounds like you are using jssip, which is known to have issues.
>
>
> On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com>
> wrote:
>
> Hi;
>
>
>
> Forgot another information. Does CODEC VP8 must be included in codec_prefs
> in WebRTC regardless there is not at all video involved?
>
>
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Quan Huo
> Sheng
> *Sent:* Thursday, April 07, 2016 10:41 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I
> am familiar with SIP.
>
>
>
> Well, can help to provide workable configuration. Or troubleshooting the
> issue. Thanks a lot.
>
>
>
> As understanding, CODEC NEGOTIATION DEAY is set, how does FS include
> SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome
> endpoint.
>
>
>
> In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered
> via WSS), there is no such information but normal RTP (RTP/SAVP).
>
>
>
> Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.
>
>
>
>
>
> Regards
>
> Smile.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Michael
> Jerris
> *Sent:* Wednesday, April 06, 2016 11:59 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> yes it's supported
>
> On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
> Hi Itola;
>
>
>
> Sorry, same error.
>
>
>
> Does Freeswitch support media switching (srtp-dtls) between two
> chrome(sip.js as signal) browsers?
>
>
>
> Finding when FS runs in media mode:
>
> codec causes caller side ?488 not acceptable here| incompatible
> destination ?
>
>                 callee side: ?cancel |user not registered?
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *?talo
> Rossi
> *Sent:* Tuesday, April 05, 2016 8:58 PM
> *To:* FreeSWITCH Users Help
> *Cc:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> Set media_mix_inbound_outbound_codecs=true
>
> ?talo Rossi
>
> italo at freeswitch.org
>
> IRC chat.freenode.net #freeswitch #freeswitch-dev
>
> Bugs? https://freeswitch.org/jira
>
> Docs? https://freeswitch.org/jira
>
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6
>
>
>
>
>
> On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
> Hi All;
>
>
>
> I want to use freeswitch to set up a WebRTC POC (Audio only,
> udp/tls/rtp/savp).  Setup is Anonymous (
> 192.168.199.216/chrome/sipjs:sip-0.7.0.min) ->freeswitch (192.168.199.22)
> ->1010 (192.168.199.216/chrome/sip-0.7.0.min),just following the
> information in book ?FreeSWITCH 1.6 Cookbook?.
>
> If using media bypass mode (inbound-bypass-media == true), all works fine,
> caller and called can hear each other.
>
> But if disabling media bypass mode, call is rejected by FS.
>
>
>
> FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?.
> Attachment is detailed FS log file.
>
> Chrome uses opus 111, FS uses opus 116.
>
> Mod_sofia.c::sofia_receive_message() -->
> sofia_media.c::sofia_media_negotiate_sdp() -->
> sofia_core_media.c::sofia_core_media_negotiate_sdp().
>
>
>
> Help is needed to troubleshoot this issue.
>
>
>
> Thanks advance.
>
> Smile.
>
>
>
>
> [This e-mail is confidential and may be privileged. If you are not the
> intended recipient, please kindly notify us immediately and delete the
> message
> from your system; please do not copy or use it for any purpose, nor
> disclose
> its contents to any other person. Thank you.]
> ---ST Electronics Group---
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Warm Regards:*
*Saurabh Kumar Verma*
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From mike at jerris.com  Thu Apr  7 18:17:23 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 7 Apr 2016 10:17:23 -0400
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
	NEGOTIATION ERROR. SDP
In-Reply-To: <f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
Message-ID: <38419347-D033-43AB-92DA-91D7D4C90465@jerris.com>

media_webrtc=true


> On Apr 7, 2016, at 3:05 AM, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
> 
> Hi Michael;
>  
> Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as WebRTC signaling.
>  
> You can see SDP from chrome (+sipjs) for this in previous attachments.
>  
> If set inbound-bypass-media=true, then chrome caller can talk with chrome callee using opus without any issue.
>  
>  
> Regards
> Smile.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 2:47 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?
>  
> On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
>  
> I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 1:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> you keep saying 
> 
> (UDP/TLS/RTP/SAVPF)
>  
> are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.
> 
> On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi;
>  
> Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved? 
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Quan Huo Sheng
> Sent: Thursday, April 07, 2016 10:41 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.
>  
> Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.
>  
> As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.
>  
> In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).
>  
> Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.
>  
>  
> Regards
> Smile.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Wednesday, April 06, 2016 11:59 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> yes it's supported
> 
> On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi Itola;
>  
> Sorry, same error.
>  
> Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?
>  
> Finding when FS runs in media mode:
> codec causes caller side ?488 not acceptable here| incompatible destination ?
>                 callee side: ?cancel |user not registered?
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of ?talo Rossi
> Sent: Tuesday, April 05, 2016 8:58 PM
> To: FreeSWITCH Users Help
> Cc: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> Set media_mix_inbound_outbound_codecs=true
> 
> ?talo Rossi
> italo at freeswitch.org <mailto:italo at freeswitch.org>
> IRC chat.freenode.net <http://chat.freenode.net/> #freeswitch #freeswitch-dev
> Bugs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Docs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6 <https://hipchat.freeswitch.org/gUdAgy0m6>
>  
>  
> 
> On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi All;
>  
> I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min <http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min <http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.
> If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
> But if disabling media bypass mode, call is rejected by FS.
>  
> FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.
> Chrome uses opus 111, FS uses opus 116.
> Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().
>  
> Help is needed to troubleshoot this issue.
>  
> Thanks advance.
> Smile.
>  
> 
> [This e-mail is confidential and may be privileged. If you are not the
> intended recipient, please kindly notify us immediately and delete the message
> from your system; please do not copy or use it for any purpose, nor disclose
> its contents to any other person. Thank you.]
> ---ST Electronics Group---
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From mike at jerris.com  Thu Apr  7 18:40:28 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 7 Apr 2016 10:40:28 -0400
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
	NEGOTIATION ERROR. SDP
In-Reply-To: <CACot2nsE48=5pKZs25g3=_syjj91sTN=giYqAyX4Qu-gqhLHHA@mail.gmail.com>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CACot2nsE48=5pKZs25g3=_syjj91sTN=giYqAyX4Qu-gqhLHHA@mail.gmail.com>
Message-ID: <AA1DBD40-0136-48F1-B295-04FBB76811F6@jerris.com>

you can search jira and the mailing list.. there were many issues.  I guess its possible they came back and fixed them, but last i looked they went many months with no response to the bugs.


> On Apr 7, 2016, at 3:17 AM, saurabh verrma <saurabhkv01 at gmail.com> wrote:
> 
> Hi Michael, 
> 
> I?ve been working on having FreeSWITCH interact with JsSIP for WebRTC calling. You?ve mentioned here that JsSIP is known to be having issues. 
> 
> Could you please point me to some link which gives details about the issue or do the favour of telling me what are the known issues in FreeSWITCH + JsSIP. 
> 
> It would be appreciable.
> 
> On Thu, Apr 7, 2016 at 12:35 PM, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi Michael;
> 
>  
> 
> Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as WebRTC signaling.
> 
>  
> 
> You can see SDP from chrome (+sipjs) for this in previous attachments.
> 
>  
> 
> If set inbound-bypass-media=true, then chrome caller can talk with chrome callee using opus without any issue.
> 
>  
> 
>  
> 
> Regards
> 
> Smile.
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 2:47 PM
> 
> 
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
> 
>  
> 
> freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?
> 
>  
> 
> On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> 
>  
> 
> I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.
> 
>  
> 
> Regards
> 
> Smile
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 1:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
> 
>  
> 
> you keep saying 
> 
> (UDP/TLS/RTP/SAVPF)
> 
>  
> 
> are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.
> 
> 
> On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> 
> Hi;
> 
>  
> 
> Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved? 
> 
>  
> 
>  
> 
> Regards
> 
> Smile
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Quan Huo Sheng
> Sent: Thursday, April 07, 2016 10:41 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
> 
>  
> 
> OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.
> 
>  
> 
> Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.
> 
>  
> 
> As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.
> 
>  
> 
> In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).
> 
>  
> 
> Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.
> 
>  
> 
>  
> 
> Regards
> 
> Smile.
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Wednesday, April 06, 2016 11:59 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
> 
>  
> 
> yes it's supported
> 
> On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> 
> Hi Itola;
> 
>  
> 
> Sorry, same error.
> 
>  
> 
> Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?
> 
>  
> 
> Finding when FS runs in media mode:
> 
> codec causes caller side ?488 not acceptable here| incompatible destination ?
> 
>                 callee side: ?cancel |user not registered?
> 
>  
> 
> Regards
> 
> Smile
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of ?talo Rossi
> Sent: Tuesday, April 05, 2016 8:58 PM
> To: FreeSWITCH Users Help
> Cc: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
> 
>  
> 
> Set media_mix_inbound_outbound_codecs=true
> 
> ?talo Rossi
> 
> italo at freeswitch.org <mailto:italo at freeswitch.org>
> IRC chat.freenode.net <http://chat.freenode.net/> #freeswitch #freeswitch-dev
> 
> Bugs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Docs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6 <https://hipchat.freeswitch.org/gUdAgy0m6>
>  
> 
>  
> 
> On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> 
> Hi All;
> 
>  
> 
> I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min <http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min <http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.
> 
> If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
> 
> But if disabling media bypass mode, call is rejected by FS.
> 
>  
> 
> FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.
> 
> Chrome uses opus 111, FS uses opus 116.
> 
> Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().
> 
>  
> 
> Help is needed to troubleshoot this issue.
> 
>  
> 
> Thanks advance.
> 
> Smile.
> 
>  
> 
> 
> [This e-mail is confidential and may be privileged. If you are not the
> intended recipient, please kindly notify us immediately and delete the message
> from your system; please do not copy or use it for any purpose, nor disclose
> its contents to any other person. Thank you.]
> ---ST Electronics Group---
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> With Warm Regards:
> Saurabh Kumar Verma
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org

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From amani.mansour2 at gmail.com  Thu Apr  7 19:01:38 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Thu, 07 Apr 2016 15:01:38 +0000
Subject: [Freeswitch-users] double 401
Message-ID: <CAFETSWhqYnsX9VBpD3LOmBh93asL28AwvXe6N11CF6uaKcDQZw@mail.gmail.com>

Hi ,
i want to oblige an extension example 400 to not be registered  .
so
soft --------------------register--------------------------------> server

soft <-----------------unauthorized (first 401)----------------server

soft ---------------------register (with authorization header computed with
the nonce of the
401)----------------------------------------------------->server

soft <--------------------unauthorized (with stale ==False) ---->server

i did this in dialplan/default.xml:
<extension name="fail register">
    <condition field="caller_id_number" expression="^400$">
        <action application="respond" data="401"/>
        <action application="hangup"/>
    </condition>
</extension>
i have only one unauthorized then i have ok ,but how can i do to have this
result .
thank you
with best regards ,
amani
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From aubalde at presenceco.com  Thu Apr  7 20:25:15 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Thu, 7 Apr 2016 18:25:15 +0200
Subject: [Freeswitch-users] OPUS and CPU load
Message-ID: <CAKThY064St3fVKNEfjyYnWaCKxOW4uGQ7-zKVniM4z4FQFMUYA@mail.gmail.com>

Hi all,,

I have found that the CPU usage with users WebRTC + OPUS is high. With
2vCPUs, 10 users use 25% of CPU. Anyone know if this load is normal?


Thank you,

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

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From mike at jerris.com  Thu Apr  7 20:39:49 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 7 Apr 2016 12:39:49 -0400
Subject: [Freeswitch-users] OPUS and CPU load
In-Reply-To: <CAKThY064St3fVKNEfjyYnWaCKxOW4uGQ7-zKVniM4z4FQFMUYA@mail.gmail.com>
References: <CAKThY064St3fVKNEfjyYnWaCKxOW4uGQ7-zKVniM4z4FQFMUYA@mail.gmail.com>
Message-ID: <F944D37F-985B-47E1-A3DA-6569B9638DAE@jerris.com>

I've seen estimates that opus is up to twice as heavy as g729.  webrtc encryption adds some (although not huge) load on top of that.  20 opus sessions per cpu seems low, are these particularly slow processors or virtual processors?


> On Apr 7, 2016, at 12:25 PM, Agust? Ubalde <aubalde at presenceco.com> wrote:
> 
> Hi all,,
> 
> I have found that the CPU usage with users WebRTC + OPUS is high. With 2vCPUs, 10 users use 25% of CPU. Anyone know if this load is normal?



From olegstolyar at gmail.com  Thu Apr  7 20:49:45 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Thu, 7 Apr 2016 09:49:45 -0700
Subject: [Freeswitch-users] OPUS and CPU load
In-Reply-To: <F944D37F-985B-47E1-A3DA-6569B9638DAE@jerris.com>
References: <CAKThY064St3fVKNEfjyYnWaCKxOW4uGQ7-zKVniM4z4FQFMUYA@mail.gmail.com>
	<F944D37F-985B-47E1-A3DA-6569B9638DAE@jerris.com>
Message-ID: <CAC8JMOJR_g751uKz_eduy1m2kGzHMx3UbOqwcOhydE+Mg_n11Q@mail.gmail.com>

In my tests OPUS transcoding seems to consume 5-6 times more CPU then G.711
or G.722.

On Thu, Apr 7, 2016 at 9:39 AM, Michael Jerris <mike at jerris.com> wrote:

> I've seen estimates that opus is up to twice as heavy as g729.  webrtc
> encryption adds some (although not huge) load on top of that.  20 opus
> sessions per cpu seems low, are these particularly slow processors or
> virtual processors?
>
>
> > On Apr 7, 2016, at 12:25 PM, Agust? Ubalde <aubalde at presenceco.com>
> wrote:
> >
> > Hi all,,
> >
> > I have found that the CPU usage with users WebRTC + OPUS is high. With
> 2vCPUs, 10 users use 25% of CPU. Anyone know if this load is normal?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From quanhs at stee.stengg.com  Fri Apr  8 07:49:23 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Fri, 8 Apr 2016 03:49:23 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299
	CODEC	NEGOTIATION ERROR. SDP
In-Reply-To: <38419347-D033-43AB-92DA-91D7D4C90465@jerris.com>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
	<38419347-D033-43AB-92DA-91D7D4C90465@jerris.com>
Message-ID: <5c023f40f79d430c85dc5d902b33ebee@RESDRCMBX001.Resources.STELECT.LOCAL>

Hi Michael;

Same complaint at mod_sofia.c 2299.
Mod_sofia.c: 2299 (sofia_media_tech_media(tech_pvt, r_sdp) != SWITCH_STATUS_SUCCESS).
Switch_core_media.c :3573: switch_core_media_negotiation_sdp().

Eval ${webrtc_enable_dtls}, Eval ${media_webrtc}, Eval ${rtp_use_dtls}, Eval ${rtp_secure_media}, all return true.

Does anyone successfully set up this WebRTC demo (excluding video) using media mode as described by cookbook.


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 10:17 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

media_webrtc=true


On Apr 7, 2016, at 3:05 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

Hi Michael;

Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as WebRTC signaling.

You can see SDP from chrome (+sipjs) for this in previous attachments.

If set inbound-bypass-media=true, then chrome caller can talk with chrome callee using opus without any issue.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 2:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?

On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 1:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

you keep saying

(UDP/TLS/RTP/SAVPF)

are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.

On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi;

Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Quan Huo Sheng
Sent: Thursday, April 07, 2016 10:41 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.

Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.

As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.

In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).

Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Wednesday, April 06, 2016 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

yes it's supported

On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Itola;

Sorry, same error.

Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?

Finding when FS runs in media mode:
codec causes caller side ?488 not acceptable here| incompatible destination ?
                callee side: ?cancel |user not registered?

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Set media_mix_inbound_outbound_codecs=true

?talo Rossi
italo at freeswitch.org<mailto:italo at freeswitch.org>
IRC chat.freenode.net<http://chat.freenode.net/> #freeswitch #freeswitch-dev
Bugs? https://freeswitch.org/jira
Docs? https://freeswitch.org/jira
Chat? https://hipchat.freeswitch.org/gUdAgy0m6


On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi All;

I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min<http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min<http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.
If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
But if disabling media bypass mode, call is rejected by FS.

FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.
Chrome uses opus 111, FS uses opus 116.
Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().

Help is needed to troubleshoot this issue.

Thanks advance.
Smile.


[This e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
from your system; please do not copy or use it for any purpose, nor disclose
its contents to any other person. Thank you.]
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http://www.freeswitch.org<http://www.freeswitch.org/>

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

Official FreeSWITCH Sites
http://www.freeswitch.org<http://www.freeswitch.org/>
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From s.safarov at gmail.com  Fri Apr  8 08:00:11 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 08 Apr 2016 04:00:11 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION ERROR. SDP
In-Reply-To: <5c023f40f79d430c85dc5d902b33ebee@RESDRCMBX001.Resources.STELECT.LOCAL>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
	<38419347-D033-43AB-92DA-91D7D4C90465@jerris.com>
	<5c023f40f79d430c85dc5d902b33ebee@RESDRCMBX001.Resources.STELECT.LOCAL>
Message-ID: <CAHtxdDf7KAQHEi+ennJs3Wnfd-pnuVjpzJ9Gn2Bert2AY-wzfA@mail.gmail.com>

One week ago I has configured master with sipML5.
You can try reproduce.

On Fri, Apr 8, 2016, 06:52 Quan Huo Sheng <quanhs at stee.stengg.com> wrote:

> Hi Michael;
>
>
>
> Same complaint at mod_sofia.c 2299.
>
> Mod_sofia.c: 2299 (sofia_media_tech_media(tech_pvt, r_sdp) !=
> SWITCH_STATUS_SUCCESS).
>
> Switch_core_media.c :3573: switch_core_media_negotiation_sdp().
>
>
>
> Eval ${webrtc_enable_dtls}, Eval ${media_webrtc}, Eval ${rtp_use_dtls},
> Eval ${rtp_secure_media}, all return true.
>
>
>
> Does anyone successfully set up this WebRTC demo (excluding video) using
> media mode as described by cookbook.
>
>
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Thursday, April 07, 2016 10:17 PM
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> media_webrtc=true
>
>
>
>
>
> On Apr 7, 2016, at 3:05 AM, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
>
>
> Hi Michael;
>
>
>
> Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only
> supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as
> WebRTC signaling.
>
>
>
> You can see SDP from chrome (+sipjs) for this in previous attachments.
>
>
>
> If set inbound-bypass-media=true, then chrome caller can talk with chrome
> callee using opus without any issue.
>
>
>
>
>
> Regards
>
> Smile.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Michael
> Jerris
> *Sent:* Thursday, April 07, 2016 2:47 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an
> sdp.  Can you please explain what exactly you think is missing from the sdp
> in our offer?
>
>
>
> On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
>
>
> I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6
> cookbook |author: Anthony Minessale ?), not jssip.
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Michael
> Jerris
> *Sent:* Thursday, April 07, 2016 1:53 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> you keep saying
>
> (UDP/TLS/RTP/SAVPF)
>
>
>
> are you thinking that is supposed to be in the sip message somehow?
>  sounds like you are using jssip, which is known to have issues.
>
>
> On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com>
> wrote:
>
> Hi;
>
>
>
> Forgot another information. Does CODEC VP8 must be included in codec_prefs
> in WebRTC regardless there is not at all video involved?
>
>
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Quan Huo
> Sheng
> *Sent:* Thursday, April 07, 2016 10:41 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I
> am familiar with SIP.
>
>
>
> Well, can help to provide workable configuration. Or troubleshooting the
> issue. Thanks a lot.
>
>
>
> As understanding, CODEC NEGOTIATION DEAY is set, how does FS include
> SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome
> endpoint.
>
>
>
> In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered
> via WSS), there is no such information but normal RTP (RTP/SAVP).
>
>
>
> Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.
>
>
>
>
>
> Regards
>
> Smile.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Michael
> Jerris
> *Sent:* Wednesday, April 06, 2016 11:59 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> yes it's supported
>
> On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
> Hi Itola;
>
>
>
> Sorry, same error.
>
>
>
> Does Freeswitch support media switching (srtp-dtls) between two
> chrome(sip.js as signal) browsers?
>
>
>
> Finding when FS runs in media mode:
>
> codec causes caller side ?488 not acceptable here| incompatible
> destination ?
>
>                 callee side: ?cancel |user not registered?
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *?talo
> Rossi
> *Sent:* Tuesday, April 05, 2016 8:58 PM
> *To:* FreeSWITCH Users Help
> *Cc:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> Set media_mix_inbound_outbound_codecs=true
>
> ?talo Rossi
>
> italo at freeswitch.org
>
> IRC chat.freenode.net #freeswitch #freeswitch-dev
>
> Bugs? https://freeswitch.org/jira
>
> Docs? https://freeswitch.org/jira
>
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6
>
>
>
>
>
> On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
> Hi All;
>
>
>
> I want to use freeswitch to set up a WebRTC POC (Audio only,
> udp/tls/rtp/savp).  Setup is Anonymous (
> 192.168.199.216/chrome/sipjs:sip-0.7.0.min) ->freeswitch (192.168.199.22)
> ->1010 (192.168.199.216/chrome/sip-0.7.0.min),just following the
> information in book ?FreeSWITCH 1.6 Cookbook?.
>
> If using media bypass mode (inbound-bypass-media == true), all works fine,
> caller and called can hear each other.
>
> But if disabling media bypass mode, call is rejected by FS.
>
>
>
> FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?.
> Attachment is detailed FS log file.
>
> Chrome uses opus 111, FS uses opus 116.
>
> Mod_sofia.c::sofia_receive_message() -->
> sofia_media.c::sofia_media_negotiate_sdp() -->
> sofia_core_media.c::sofia_core_media_negotiate_sdp().
>
>
>
> Help is needed to troubleshoot this issue.
>
>
>
> Thanks advance.
>
> Smile.
>
>
>
>
> [This e-mail is confidential and may be privileged. If you are not the
> intended recipient, please kindly notify us immediately and delete the
> message
> from your system; please do not copy or use it for any purpose, nor
> disclose
> its contents to any other person. Thank you.]
> ---ST Electronics Group---
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jelena at misticnabica.hr  Fri Apr  8 08:02:02 2016
From: jelena at misticnabica.hr (jelena at misticnabica.hr)
Date: Fri, 8 Apr 2016 04:02:02 GMT
Subject: [Freeswitch-users] webrtc: [ERR]
	mod_sofia.c:2299	CODEC	NEGOTIATION ERROR. SDP
Message-ID: <6B35DEED784543549521C9F80CCE9A0D.MAI@server2.totohost.hr>



From quanhs at stee.stengg.com  Fri Apr  8 08:01:26 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Fri, 8 Apr 2016 04:01:26 +0000
Subject: [Freeswitch-users] OPUS and CPU load
In-Reply-To: <CAC8JMOJR_g751uKz_eduy1m2kGzHMx3UbOqwcOhydE+Mg_n11Q@mail.gmail.com>
References: <CAKThY064St3fVKNEfjyYnWaCKxOW4uGQ7-zKVniM4z4FQFMUYA@mail.gmail.com>
	<F944D37F-985B-47E1-A3DA-6569B9638DAE@jerris.com>
	<CAC8JMOJR_g751uKz_eduy1m2kGzHMx3UbOqwcOhydE+Mg_n11Q@mail.gmail.com>
Message-ID: <48bb9084a61846c8b691be96d5437815@RESDRCMBX001.Resources.STELECT.LOCAL>

Hi;

I want to set up Webrtc Demo using freeswitch with media switching or media transcode. Setup is chrome (sip.js) ->fs->chrome (sip.js), but end up with fail.
Can you share your successful case using Webrtc + opus. (SIP preferred to Verto as signal).


Thanks
Smile

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Oleg Stolyar
Sent: Friday, April 08, 2016 12:50 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] OPUS and CPU load

In my tests OPUS transcoding seems to consume 5-6 times more CPU then G.711 or G.722.

On Thu, Apr 7, 2016 at 9:39 AM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
I've seen estimates that opus is up to twice as heavy as g729.  webrtc encryption adds some (although not huge) load on top of that.  20 opus sessions per cpu seems low, are these particularly slow processors or virtual processors?


> On Apr 7, 2016, at 12:25 PM, Agust? Ubalde <aubalde at presenceco.com<mailto:aubalde at presenceco.com>> wrote:
>
> Hi all,,
>
> I have found that the CPU usage with users WebRTC + OPUS is high. With 2vCPUs, 10 users use 25% of CPU. Anyone know if this load is normal?


_________________________________________________________________________
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 e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
from your system; please do not copy or use it for any purpose, nor disclose
its contents to any other person. Thank you.]
---ST Electronics Group---
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From steveu at coppice.org  Fri Apr  8 08:04:11 2016
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 8 Apr 2016 12:04:11 +0800
Subject: [Freeswitch-users] OPUS and CPU load
In-Reply-To: <CAC8JMOJR_g751uKz_eduy1m2kGzHMx3UbOqwcOhydE+Mg_n11Q@mail.gmail.com>
References: <CAKThY064St3fVKNEfjyYnWaCKxOW4uGQ7-zKVniM4z4FQFMUYA@mail.gmail.com>
	<F944D37F-985B-47E1-A3DA-6569B9638DAE@jerris.com>
	<CAC8JMOJR_g751uKz_eduy1m2kGzHMx3UbOqwcOhydE+Mg_n11Q@mail.gmail.com>
Message-ID: <57072DBB.9050004@coppice.org>

You should be seeing a significant different between G.711 and G.722, so 
bundling them together doesn't make sense. G.711 takes only a minute 
amount of CPU effort. G.722 takes a significant amount.

Regards,
Steve

On 04/08/2016 12:49 AM, Oleg Stolyar wrote:
> In my tests OPUS transcoding seems to consume 5-6 times more CPU then 
> G.711 or G.722.
>
> On Thu, Apr 7, 2016 at 9:39 AM, Michael Jerris <mike at jerris.com 
> <mailto:mike at jerris.com>> wrote:
>
>     I've seen estimates that opus is up to twice as heavy as g729. 
>     webrtc encryption adds some (although not huge) load on top of
>     that.  20 opus sessions per cpu seems low, are these particularly
>     slow processors or virtual processors?
>
>
>     > On Apr 7, 2016, at 12:25 PM, Agust? Ubalde
>     <aubalde at presenceco.com <mailto:aubalde at presenceco.com>> wrote:
>     >
>     > Hi all,,
>     >
>     > I have found that the CPU usage with users WebRTC + OPUS is
>     high. With 2vCPUs, 10 users use 25% of CPU. Anyone know if this
>     load is normal?
>



From quanhs at stee.stengg.com  Fri Apr  8 10:31:41 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Fri, 8 Apr 2016 06:31:41 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION ERROR. SDP
In-Reply-To: <CAHtxdDf7KAQHEi+ennJs3Wnfd-pnuVjpzJ9Gn2Bert2AY-wzfA@mail.gmail.com>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
	<38419347-D033-43AB-92DA-91D7D4C90465@jerris.com>
	<5c023f40f79d430c85dc5d902b33ebee@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAHtxdDf7KAQHEi+ennJs3Wnfd-pnuVjpzJ9Gn2Bert2AY-wzfA@mail.gmail.com>
Message-ID: <302e815fb7c642d094445f4465256b92@RESDRCMBX002.Resources.STELECT.LOCAL>

what is setting of inbound-bypass-media and inbound-proxy-media in your case?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
Sent: Friday, April 08, 2016 12:00 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP


One week ago I has configured master with sipML5.
You can try reproduce.

On Fri, Apr 8, 2016, 06:52 Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Michael;

Same complaint at mod_sofia.c 2299.
Mod_sofia.c: 2299 (sofia_media_tech_media(tech_pvt, r_sdp) != SWITCH_STATUS_SUCCESS).
Switch_core_media.c :3573: switch_core_media_negotiation_sdp().

Eval ${webrtc_enable_dtls}, Eval ${media_webrtc}, Eval ${rtp_use_dtls}, Eval ${rtp_secure_media}, all return true.

Does anyone successfully set up this WebRTC demo (excluding video) using media mode as described by cookbook.


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 10:17 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

media_webrtc=true


On Apr 7, 2016, at 3:05 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

Hi Michael;

Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as WebRTC signaling.

You can see SDP from chrome (+sipjs) for this in previous attachments.

If set inbound-bypass-media=true, then chrome caller can talk with chrome callee using opus without any issue.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 2:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?

On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 1:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

you keep saying

(UDP/TLS/RTP/SAVPF)

are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.

On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi;

Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Quan Huo Sheng
Sent: Thursday, April 07, 2016 10:41 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.

Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.

As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.

In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).

Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Wednesday, April 06, 2016 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

yes it's supported

On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Itola;

Sorry, same error.

Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?

Finding when FS runs in media mode:
codec causes caller side ?488 not acceptable here| incompatible destination ?
                callee side: ?cancel |user not registered?

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Set media_mix_inbound_outbound_codecs=true

?talo Rossi
italo at freeswitch.org<mailto:italo at freeswitch.org>
IRC chat.freenode.net<http://chat.freenode.net/> #freeswitch #freeswitch-dev
Bugs? https://freeswitch.org/jira
Docs? https://freeswitch.org/jira
Chat? https://hipchat.freeswitch.org/gUdAgy0m6


On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi All;

I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min<http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min<http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.
If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
But if disabling media bypass mode, call is rejected by FS.

FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.
Chrome uses opus 111, FS uses opus 116.
Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().

Help is needed to troubleshoot this issue.

Thanks advance.
Smile.


[This e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
from your system; please do not copy or use it for any purpose, nor disclose
its contents to any other person. Thank you.]
---ST Electronics Group---
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>

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

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

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

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

FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
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From mike at jerris.com  Fri Apr  8 11:13:06 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 Apr 2016 03:13:06 -0400
Subject: [Freeswitch-users] OPUS and CPU load
In-Reply-To: <48bb9084a61846c8b691be96d5437815@RESDRCMBX001.Resources.STELECT.LOCAL>
References: <CAKThY064St3fVKNEfjyYnWaCKxOW4uGQ7-zKVniM4z4FQFMUYA@mail.gmail.com>
	<F944D37F-985B-47E1-A3DA-6569B9638DAE@jerris.com>
	<CAC8JMOJR_g751uKz_eduy1m2kGzHMx3UbOqwcOhydE+Mg_n11Q@mail.gmail.com>
	<48bb9084a61846c8b691be96d5437815@RESDRCMBX001.Resources.STELECT.LOCAL>
Message-ID: <CAEwTsAgdr-4sgiQtzLqAZBw4kwD17Mkuh1QbxGzgcHkFPKWCpA@mail.gmail.com>

Can someone explain why sip would be preferred in this scenario?  Is there
any real reason?

On Friday, April 8, 2016, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:

> Hi;
>
>
>
> I want to set up Webrtc Demo using freeswitch with media switching or
> media transcode. Setup is chrome (sip.js) ->fs->chrome (sip.js), but end up
> with fail.
>
> Can you share your successful case using Webrtc + opus. (SIP preferred to
> Verto as signal).
>
>
>
>
>
> Thanks
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *Oleg Stolyar
> *Sent:* Friday, April 08, 2016 12:50 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] OPUS and CPU load
>
>
>
> In my tests OPUS transcoding seems to consume 5-6 times more CPU then
> G.711 or G.722.
>
>
>
> On Thu, Apr 7, 2016 at 9:39 AM, Michael Jerris <mike at jerris.com
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>
> I've seen estimates that opus is up to twice as heavy as g729.  webrtc
> encryption adds some (although not huge) load on top of that.  20 opus
> sessions per cpu seems low, are these particularly slow processors or
> virtual processors?
>
>
> > On Apr 7, 2016, at 12:25 PM, Agust? Ubalde <aubalde at presenceco.com
> <javascript:_e(%7B%7D,'cvml','aubalde at presenceco.com');>> wrote:
> >
> > Hi all,,
> >
> > I have found that the CPU usage with users WebRTC + OPUS is high. With
> 2vCPUs, 10 users use 25% of CPU. Anyone know if this load is normal?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> [This e-mail is confidential and may be privileged. If you are not the
> intended recipient, please kindly notify us immediately and delete the
> message
> from your system; please do not copy or use it for any purpose, nor
> disclose
> its contents to any other person. Thank you.]
> ---ST Electronics Group---
>
>
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From s.safarov at gmail.com  Fri Apr  8 11:17:13 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 08 Apr 2016 07:17:13 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION ERROR. SDP
In-Reply-To: <302e815fb7c642d094445f4465256b92@RESDRCMBX002.Resources.STELECT.LOCAL>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
	<38419347-D033-43AB-92DA-91D7D4C90465@jerris.com>
	<5c023f40f79d430c85dc5d902b33ebee@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAHtxdDf7KAQHEi+ennJs3Wnfd-pnuVjpzJ9Gn2Bert2AY-wzfA@mail.gmail.com>
	<302e815fb7c642d094445f4465256b92@RESDRCMBX002.Resources.STELECT.LOCAL>
Message-ID: <CAHtxdDfHgBza0eiWqqxGCi_PH-Y8cGxGdGJFDRLhrnxC3rbWAA@mail.gmail.com>

All call with media transcoding enabled. In WebRTC case OPUS <-> G711a

On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs at stee.stengg.com> wrote:

> what is setting of inbound-bypass-media and inbound-proxy-media in your
> case?
>
>
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sergey
> Safarov
> *Sent:* Friday, April 08, 2016 12:00 PM
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> One week ago I has configured master with sipML5.
> You can try reproduce.
>
>
>
> On Fri, Apr 8, 2016, 06:52 Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
> Hi Michael;
>
>
>
> Same complaint at mod_sofia.c 2299.
>
> Mod_sofia.c: 2299 (sofia_media_tech_media(tech_pvt, r_sdp) !=
> SWITCH_STATUS_SUCCESS).
>
> Switch_core_media.c :3573: switch_core_media_negotiation_sdp().
>
>
>
> Eval ${webrtc_enable_dtls}, Eval ${media_webrtc}, Eval ${rtp_use_dtls},
> Eval ${rtp_secure_media}, all return true.
>
>
>
> Does anyone successfully set up this WebRTC demo (excluding video) using
> media mode as described by cookbook.
>
>
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Thursday, April 07, 2016 10:17 PM
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> media_webrtc=true
>
>
>
>
>
> On Apr 7, 2016, at 3:05 AM, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
>
>
> Hi Michael;
>
>
>
> Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only
> supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as
> WebRTC signaling.
>
>
>
> You can see SDP from chrome (+sipjs) for this in previous attachments.
>
>
>
> If set inbound-bypass-media=true, then chrome caller can talk with chrome
> callee using opus without any issue.
>
>
>
>
>
> Regards
>
> Smile.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Michael
> Jerris
> *Sent:* Thursday, April 07, 2016 2:47 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an
> sdp.  Can you please explain what exactly you think is missing from the sdp
> in our offer?
>
>
>
> On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
>
>
> I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6
> cookbook |author: Anthony Minessale ?), not jssip.
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Michael
> Jerris
> *Sent:* Thursday, April 07, 2016 1:53 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> you keep saying
>
> (UDP/TLS/RTP/SAVPF)
>
>
>
> are you thinking that is supposed to be in the sip message somehow?
>  sounds like you are using jssip, which is known to have issues.
>
>
> On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com>
> wrote:
>
> Hi;
>
>
>
> Forgot another information. Does CODEC VP8 must be included in codec_prefs
> in WebRTC regardless there is not at all video involved?
>
>
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Quan Huo
> Sheng
> *Sent:* Thursday, April 07, 2016 10:41 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I
> am familiar with SIP.
>
>
>
> Well, can help to provide workable configuration. Or troubleshooting the
> issue. Thanks a lot.
>
>
>
> As understanding, CODEC NEGOTIATION DEAY is set, how does FS include
> SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome
> endpoint.
>
>
>
> In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered
> via WSS), there is no such information but normal RTP (RTP/SAVP).
>
>
>
> Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.
>
>
>
>
>
> Regards
>
> Smile.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Michael
> Jerris
> *Sent:* Wednesday, April 06, 2016 11:59 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> yes it's supported
>
> On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
> Hi Itola;
>
>
>
> Sorry, same error.
>
>
>
> Does Freeswitch support media switching (srtp-dtls) between two
> chrome(sip.js as signal) browsers?
>
>
>
> Finding when FS runs in media mode:
>
> codec causes caller side ?488 not acceptable here| incompatible
> destination ?
>
>                 callee side: ?cancel |user not registered?
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *?talo
> Rossi
> *Sent:* Tuesday, April 05, 2016 8:58 PM
> *To:* FreeSWITCH Users Help
> *Cc:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> Set media_mix_inbound_outbound_codecs=true
>
> ?talo Rossi
>
> italo at freeswitch.org
>
> IRC chat.freenode.net #freeswitch #freeswitch-dev
>
> Bugs? https://freeswitch.org/jira
>
> Docs? https://freeswitch.org/jira
>
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6
>
>
>
>
>
> On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
> Hi All;
>
>
>
> I want to use freeswitch to set up a WebRTC POC (Audio only,
> udp/tls/rtp/savp).  Setup is Anonymous (
> 192.168.199.216/chrome/sipjs:sip-0.7.0.min) ->freeswitch (192.168.199.22)
> ->1010 (192.168.199.216/chrome/sip-0.7.0.min),just following the
> information in book ?FreeSWITCH 1.6 Cookbook?.
>
> If using media bypass mode (inbound-bypass-media == true), all works fine,
> caller and called can hear each other.
>
> But if disabling media bypass mode, call is rejected by FS.
>
>
>
> FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?.
> Attachment is detailed FS log file.
>
> Chrome uses opus 111, FS uses opus 116.
>
> Mod_sofia.c::sofia_receive_message() -->
> sofia_media.c::sofia_media_negotiate_sdp() -->
> sofia_core_media.c::sofia_core_media_negotiate_sdp().
>
>
>
> Help is needed to troubleshoot this issue.
>
>
>
> Thanks advance.
>
> Smile.
>
>
>
>
> [This e-mail is confidential and may be privileged. If you are not the
> intended recipient, please kindly notify us immediately and delete the
> message
> from your system; please do not copy or use it for any purpose, nor
> disclose
> its contents to any other person. Thank you.]
> ---ST Electronics Group---
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From quanhs at stee.stengg.com  Fri Apr  8 12:04:50 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Fri, 8 Apr 2016 08:04:50 +0000
Subject: [Freeswitch-users] OPUS and CPU load
In-Reply-To: <CAEwTsAgdr-4sgiQtzLqAZBw4kwD17Mkuh1QbxGzgcHkFPKWCpA@mail.gmail.com>
References: <CAKThY064St3fVKNEfjyYnWaCKxOW4uGQ7-zKVniM4z4FQFMUYA@mail.gmail.com>
	<F944D37F-985B-47E1-A3DA-6569B9638DAE@jerris.com>
	<CAC8JMOJR_g751uKz_eduy1m2kGzHMx3UbOqwcOhydE+Mg_n11Q@mail.gmail.com>
	<48bb9084a61846c8b691be96d5437815@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAgdr-4sgiQtzLqAZBw4kwD17Mkuh1QbxGzgcHkFPKWCpA@mail.gmail.com>
Message-ID: <736951f4272746c4b150a16110f5f330@RESDRCMBX002.Resources.STELECT.LOCAL>

I just want to study a successful case for the Webrtc demo (using media mode) in the chapter 6 of book ?FreeSwitch 1.6 Cookbook? which uses sip.js (sip-0.7.0.js) in Chrome.

So far I fail until now. Please refer to the  thread ?Webrtc [err] mod_sofia.c 2299 codec negotiation error. Sdp?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Friday, April 08, 2016 3:13 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] OPUS and CPU load

Can someone explain why sip would be preferred in this scenario?  Is there any real reason?

On Friday, April 8, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi;

I want to set up Webrtc Demo using freeswitch with media switching or media transcode. Setup is chrome (sip.js) ->fs->chrome (sip.js), but end up with fail.
Can you share your successful case using Webrtc + opus. (SIP preferred to Verto as signal).


Thanks
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');> [mailto:freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>] On Behalf Of Oleg Stolyar
Sent: Friday, April 08, 2016 12:50 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] OPUS and CPU load

In my tests OPUS transcoding seems to consume 5-6 times more CPU then G.711 or G.722.

On Thu, Apr 7, 2016 at 9:39 AM, Michael Jerris <mike at jerris.com<javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
I've seen estimates that opus is up to twice as heavy as g729.  webrtc encryption adds some (although not huge) load on top of that.  20 opus sessions per cpu seems low, are these particularly slow processors or virtual processors?


> On Apr 7, 2016, at 12:25 PM, Agust? Ubalde <aubalde at presenceco.com<javascript:_e(%7B%7D,'cvml','aubalde at presenceco.com');>> wrote:
>
> Hi all,,
>
> I have found that the CPU usage with users WebRTC + OPUS is high. With 2vCPUs, 10 users use 25% of CPU. Anyone know if this load is normal?


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
http://www.freeswitchsolutions.com

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From quanhs at stee.stengg.com  Fri Apr  8 12:16:36 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Fri, 8 Apr 2016 08:16:36 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
 NEGOTIATION ERROR. SDP
In-Reply-To: <CAHtxdDfHgBza0eiWqqxGCi_PH-Y8cGxGdGJFDRLhrnxC3rbWAA@mail.gmail.com>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
	<38419347-D033-43AB-92DA-91D7D4C90465@jerris.com>
	<5c023f40f79d430c85dc5d902b33ebee@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAHtxdDf7KAQHEi+ennJs3Wnfd-pnuVjpzJ9Gn2Bert2AY-wzfA@mail.gmail.com>
	<302e815fb7c642d094445f4465256b92@RESDRCMBX002.Resources.STELECT.LOCAL>
	<CAHtxdDfHgBza0eiWqqxGCi_PH-Y8cGxGdGJFDRLhrnxC3rbWAA@mail.gmail.com>
Message-ID: <a0e8cc45ffae4f42bb7be231bb03a163@RESDRCMBX002.Resources.STELECT.LOCAL>

Good. Can you share your scenario ?

Chrome (sipML5) ->FS (1.6.5-64bit  Media mode) ->Chrome (sipML5).


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
Sent: Friday, April 08, 2016 3:17 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP


All call with media transcoding enabled. In WebRTC case OPUS <-> G711a

On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
what is setting of inbound-bypass-media and inbound-proxy-media in your case?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Sergey Safarov
Sent: Friday, April 08, 2016 12:00 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP


One week ago I has configured master with sipML5.
You can try reproduce.

On Fri, Apr 8, 2016, 06:52 Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Michael;

Same complaint at mod_sofia.c 2299.
Mod_sofia.c: 2299 (sofia_media_tech_media(tech_pvt, r_sdp) != SWITCH_STATUS_SUCCESS).
Switch_core_media.c :3573: switch_core_media_negotiation_sdp().

Eval ${webrtc_enable_dtls}, Eval ${media_webrtc}, Eval ${rtp_use_dtls}, Eval ${rtp_secure_media}, all return true.

Does anyone successfully set up this WebRTC demo (excluding video) using media mode as described by cookbook.


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 10:17 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

media_webrtc=true


On Apr 7, 2016, at 3:05 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

Hi Michael;

Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as WebRTC signaling.

You can see SDP from chrome (+sipjs) for this in previous attachments.

If set inbound-bypass-media=true, then chrome caller can talk with chrome callee using opus without any issue.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 2:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?

On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 1:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

you keep saying

(UDP/TLS/RTP/SAVPF)

are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.

On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi;

Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Quan Huo Sheng
Sent: Thursday, April 07, 2016 10:41 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.

Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.

As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.

In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).

Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Wednesday, April 06, 2016 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

yes it's supported

On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Itola;

Sorry, same error.

Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?

Finding when FS runs in media mode:
codec causes caller side ?488 not acceptable here| incompatible destination ?
                callee side: ?cancel |user not registered?

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Set media_mix_inbound_outbound_codecs=true

?talo Rossi
italo at freeswitch.org<mailto:italo at freeswitch.org>
IRC chat.freenode.net<http://chat.freenode.net/> #freeswitch #freeswitch-dev
Bugs? https://freeswitch.org/jira
Docs? https://freeswitch.org/jira
Chat? https://hipchat.freeswitch.org/gUdAgy0m6


On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi All;

I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min<http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min<http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.
If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
But if disabling media bypass mode, call is rejected by FS.

FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.
Chrome uses opus 111, FS uses opus 116.
Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().

Help is needed to troubleshoot this issue.

Thanks advance.
Smile.


[This e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
from your system; please do not copy or use it for any purpose, nor disclose
its contents to any other person. Thank you.]
---ST Electronics Group---
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From amani.mansour2 at gmail.com  Fri Apr  8 13:21:46 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Fri, 08 Apr 2016 09:21:46 +0000
Subject: [Freeswitch-users] 401 Unauthorized register
Message-ID: <CAFETSWhQzZmQpvvQOWEPFNkWTNgzqKqGr5z=cUr6enhHc0RGVw@mail.gmail.com>

Good morning  ,

Can you help me please ?, I need to configure a number X that  when i dial
it from my soft phone it must return to me 401 Unauthorized then hi send a
new regester with authorization (with nonce of the 401 ) and then he
receive a second 401 with stale = false .


Please i need your help ,Thanks a lot .

regards
amani
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From steveayre at gmail.com  Fri Apr  8 14:22:09 2016
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 8 Apr 2016 11:22:09 +0100
Subject: [Freeswitch-users] Call from SIP server to external SIP
 provider: 488 INCOMPATIBLE_DESTINATION
In-Reply-To: <CAC89h=KaEvzXyETijxPC_tEVn9c6h9yyzDG7gqFFVuCwdCcK7Q@mail.gmail.com>
References: <mailman.266.1460030809.8223.freeswitch-users@lists.freeswitch.org>
	<313CE2EA-5E22-4F62-99E6-A454C467E3A3@gmail.com>
	<CAC89h=KaEvzXyETijxPC_tEVn9c6h9yyzDG7gqFFVuCwdCcK7Q@mail.gmail.com>
Message-ID: <CAFiqYumm8A8titL5uR7oFb6tynBsOVUUPbmZLEouSurtD8HAsQ@mail.gmail.com>

It can be suitable, but every SIP server is configured with a list of the
codecs it accepts/offers so that list can be different depending on which
server you're connecting to.

In this case the phone is offering only G729 and your server is only trying
to use GSM so they don't have any codecs in common hence the 488 error.
Some servers would accept the call and you could too if you reconfigure
your codec options.

G729 is a licensed codec. Unless you're bridging it directly to another
server/phone that accepts G729 or you can reconfigure the phone to use
another codec then you'll need the mod_com_g729 commercial module to
transcode it to another codec.

GSM will be supported by some servers, but most probably won't support it
(there are other better codecs now). You probably want to set your server
to use several codecs so it can figure out which one is best for whichever
server/phone you're connecting to.

On 7 April 2016 at 14:39, ???? ??????? <stasan89 at gmail.com> wrote:

> Ahmed, thank you.
>
> I did not know that the gsm codec is not suitable for VOIP providers.
>
> 2016-04-07 15:17 GMT+03:00 Ahmed habiba <ahabiba at gmail.com>:
>
>> It looks that the profile you are using use only GSM, however the phone
>> is trying to use G729 only
>>
>> look at the colored text.
>>
>> On Apr 7, 2016, at 3:06 PM, freeswitch-users-request at lists.freeswitch.org
>> wrote:
>>
>> *From: *???? ??????? <stasan89 at gmail.com>
>> *Subject: **[Freeswitch-users] Call from SIP server to external SIP
>> provider: 488 INCOMPATIBLE_DESTINATION*
>> *Date: *April 7, 2016 at 3:06:17 PM GMT+3
>> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>
>>
>> Hello.
>> I`am doing external calls via sip provider. And I have internal SIP
>> server (opensips) with users.
>> My case: two my users connecting via my sip server. One of him starting
>> conference and calling to external number (carrier number or mobile phone).
>>
>> But when I doing call to external number, I got error 488
>> INCOMPATIBLE_DESTINATION on call.
>>
>> For setup I done next steps:
>> 1) create file /etc/freeswitch/sip_profiles/external/freelycall.com.xml
>> with provider-access params
>>
>> 2) set dialplan extension with my mobile number (hardcoded expression for
>> testing)
>>     <extension name="outbound_calls_to_freelycall">
>>       <condition field="destination_number" expression="^(7906*******)$"
>> break="on-true">
>>         <action application="set"
>> data="hangup_after_bridge=true"/>
>>         <action application="bridge" data="sofia/gateway/
>> freelycall.com/$1"/>
>>       </condition>
>>     </extension>
>>
>> 3) set default provider params and external hosts in vars.xml
>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:freelycall.com"/>
>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:freelycall.com"/>
>>   ...
>>   <X-PRE-PROCESS cmd="set" data="default_provider=freelycall.com"/>
>>   <X-PRE-PROCESS cmd="set" data="default_provider_username=MY_ID"/>
>>   <X-PRE-PROCESS cmd="set" data="default_provider_password=MY_PASS"/>
>>   <X-PRE-PROCESS cmd="set" data="default_provider_from_domain=
>> freelycall.com"/>
>>   <!-- true or false -->
>>   <X-PRE-PROCESS cmd="set" data="default_provider_register=true"/>
>>   <X-PRE-PROCESS cmd="set" data="default_provider_contact=5000"/>
>>
>> I starting debug:
>> freeswitch -c
>>
>> command "sofia status" say:
>>                      Name       Type
>>                                       Data    State
>>
>> =================================================================================================
>>             172.31.22.124      alias
>> internal    ALIASED
>>                  external    profile
>> sip:mod_sofia at 172.31.22.124:5060    RUNNING (0)
>>                  external    profile
>> sip:mod_sofia at 172.31.22.124:5061    RUNNING (0) (TLS)
>>  external::freelycall.com    gateway
>> sip:MY_ID at freelycall.com    REGED
>>                  internal    profile
>> sip:mod_sofia at 172.31.22.124:5080    RUNNING (0)
>>                  internal    profile
>> sip:mod_sofia at 172.31.22.124:5081    RUNNING (0) (TLS)
>>
>> =================================================================================================
>> 2 profiles 1 alias
>>
>> Thats good I think.
>>
>> command "sofia status profile external" say:
>>
>> =================================================================================================
>> 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               172.31.22.124
>> SIP-IP               172.31.22.124
>> URL                  sip:mod_sofia at 172.31.22.124:5060
>> BIND-URL             sip:mod_sofia at 172.31.22.124:5060;transport=udp,tcp
>> TLS-URL              sip:mod_sofia at 172.31.22.124:5061
>> TLS-BIND-URL         sips:mod_sofia at 172.31.22.124:5061;transport=tls
>> HOLD-MUSIC           local_stream://moh
>> OUTBOUND-PROXY       N/A
>> CODECS IN            GSM
>> CODECS OUT           GSM
>> TEL-EVENT            101
>> DTMF-MODE            rfc2833
>> CNG                  13
>> SESSION-TO           0
>> MAX-DIALOG           0
>> NOMEDIA              false
>> LATE-NEG             true
>> PROXY-MEDIA          false
>> ZRTP-PASSTHRU        true
>> AGGRESSIVENAT        false
>> CALLS-IN             0
>> FAILED-CALLS-IN      0
>> CALLS-OUT            0
>> FAILED-CALLS-OUT     0
>> REGISTRATIONS        0
>>
>> I`am not sure, that all correct, but I don`t know how should look like
>> the right output of this command
>>
>>
>> When I starting call, I see next row in log:
>> 2016-04-07 07:52:09.195582 [NOTICE] switch_ivr.c:2092 Transfer
>> sofia/external/8 at MY_OPENSIPS_DOMAIN to XML[7906*******@default]
>>
>> I think, that @default is incorrect, I think that domain should be look
>> like as @freelycall.com
>>
>> What I doing incorrect? Any idea?
>>
>> Full log on call:
>> 2016-04-07 07:52:09.195582 [NOTICE] switch_channel.c:1101 New Channel
>> sofia/external/8 at MY_OPENSIPS_DOMAIN
>> [286b6a34-fcb7-11e5-a94f-99d758825eaf]
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_NEW
>> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:9248
>> sofia/external/8 at MY_OPENSIPS_DOMAIN receiving invite from
>> 172.31.0.169:5060 version: 1.6.6  64bit
>> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6760 Channel
>> sofia/external/8 at MY_OPENSIPS_DOMAIN entering state [received][100]
>> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6770 Remote SDP:
>> v=0
>> o=- 1460029927 1 IN IP4 MY_CALLER_IP
>> s=portsip.com
>> c=IN IP4 MY_OPENSIPS_IP
>> t=0 0
>> m=audio 42386 RTP/AVP 18 3 101
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=nortpproxy:yes
>>
>> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:7125
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_NEW -> CS_INIT
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:492
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State NEW
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_INIT
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State INIT
>> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:88
>> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA INIT
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:40
>> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard INIT
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:48
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_INIT -> CS_ROUTING
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State INIT going to sleep
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_ROUTING
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_channel.c:2247
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Callstate Change DOWN -> RINGING
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING
>> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141
>> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA ROUTING
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:166
>> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard ROUTING
>> 2016-04-07 07:52:09.195582 [INFO] mod_dialplan_xml.c:637 Processing 8
>> <8>->7906******* in context public
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing
>> [public->from_opensips] continue=false
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS)
>> [from_opensips] network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/
>> break=on-false
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action
>> transfer(${destination_number} XML default)
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:216
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_ROUTING ->
>> CS_EXECUTE
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING going to sleep
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_EXECUTE
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE
>> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:196
>> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA EXECUTE
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:258
>> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard EXECUTE
>> EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN transfer(7906******* XML
>> default)
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_ivr.c:2085
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_EXECUTE ->
>> CS_ROUTING
>> 2016-04-07 07:52:09.195582 [NOTICE] switch_ivr.c:2092 Transfer
>> sofia/external/8 at MY_OPENSIPS_DOMAIN to XML[7906*******@default]
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE going to sleep
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_ROUTING
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING
>> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141
>> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA ROUTING
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:166
>> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard ROUTING
>> 2016-04-07 07:52:09.195582 [INFO] mod_dialplan_xml.c:637 Processing 8
>> <8>->7906******* in context default
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing [default->unloop]
>> continue=false
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (FAIL) [unloop]
>> ${sip_looped_call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing
>> [default->tod_example] continue=true
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Date/TimeMatch (FAIL)
>> [tod_example] break=on-false
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN parsing
>> [default->outbound_calls_to_freelycall] continue=false
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Regex (PASS)
>> [outbound_calls_to_freelycall] destination_number(7906*******) =~
>> /^(7906*******)$/ break=on-true
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action
>> set(hangup_after_bridge=true)
>> Dialplan: sofia/external/8 at MY_OPENSIPS_DOMAIN Action
>> bridge(sofia/gateway/freelycall.com/7906*******)
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:216
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_ROUTING ->
>> CS_EXECUTE
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State ROUTING going to sleep
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_EXECUTE
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:539
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE
>> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:196
>> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA EXECUTE
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:258
>> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard EXECUTE
>> EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN set(hangup_after_bridge=true)
>> 2016-04-07 07:52:09.195582 [DEBUG] mod_dptools.c:1498 SET
>> sofia/external/8 at MY_OPENSIPS_DOMAIN [hangup_after_bridge]=[true]
>> EXECUTE sofia/external/8 at MY_OPENSIPS_DOMAIN bridge(sofia/gateway/
>> freelycall.com/7906*******)
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_ivr_originate.c:2128 Parsing
>> global variables
>> 2016-04-07 07:52:09.195582 [NOTICE] switch_channel.c:1101 New Channel
>> sofia/external/7906******* [286bb5b6-fcb7-11e5-a957-99d758825eaf]
>> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:4776
>> (sofia/external/7906*******) State Change CS_NEW -> CS_INIT
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/7906*******) Running State Change CS_INIT
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
>> (sofia/external/7906*******) State INIT
>> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:88
>> sofia/external/7906******* SOFIA INIT
>> 2016-04-07 07:52:09.195582 [DEBUG] sofia_glue.c:1257
>> sofia/external/7906******* sending invite version: 1.6.6  64bit
>> Local SDP:
>> v=0
>> o=FreeSWITCH 1460003999 1460004000 IN IP4 172.31.22.124
>> s=FreeSWITCH
>> c=IN IP4 172.31.22.124
>> t=0 0
>> m=audio 25930 RTP/AVP 3 101 13
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>> a=sendrecv
>>
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:40
>> sofia/external/7906******* Standard INIT
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:48
>> (sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:516
>> (sofia/external/7906*******) State INIT going to sleep
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/7906*******) Running State Change CS_ROUTING
>> 2016-04-07 07:52:09.195582 [DEBUG] sofia.c:6760 Channel
>> sofia/external/7906******* entering state [calling][0]
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/7906*******) State ROUTING
>> 2016-04-07 07:52:09.195582 [DEBUG] mod_sofia.c:141
>> sofia/external/7906******* SOFIA ROUTING
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_ivr_originate.c:67
>> (sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/7906*******) State ROUTING going to sleep
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:551
>> (sofia/external/7906*******) State CONSUME_MEDIA
>> 2016-04-07 07:52:09.195582 [DEBUG] switch_core_state_machine.c:551
>> (sofia/external/7906*******) State CONSUME_MEDIA going to sleep
>> 2016-04-07 07:52:09.215582 [DEBUG] sofia.c:6760 Channel
>> sofia/external/7906******* entering state [terminated][488]
>> 2016-04-07 07:52:09.215582 [NOTICE] sofia.c:7779 Hangup
>> sofia/external/7906******* [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/7906*******) Running State Change CS_HANGUP
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:739
>> (sofia/external/7906*******) Callstate Change DOWN -> HANGUP
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
>> (sofia/external/7906*******) State HANGUP
>> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:431 Channel
>> sofia/external/7906******* hanging up, cause: INCOMPATIBLE_DESTINATION
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/7906******* Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
>> (sofia/external/7906*******) State HANGUP going to sleep
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:508
>> (sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/7906*******) Running State Change CS_REPORTING
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
>> (sofia/external/7906*******) State REPORTING
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:104
>> sofia/external/7906******* Standard REPORTING, cause:
>> INCOMPATIBLE_DESTINATION
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
>> (sofia/external/7906*******) State REPORTING going to sleep
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:499
>> (sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:1646 Session 2
>> (sofia/external/7906*******) Locked, Waiting on external entities
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_ivr_originate.c:3751 Originate
>> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
>> 2016-04-07 07:52:09.215582 [INFO] mod_dptools.c:3379 Originate Failed.
>> Cause: INCOMPATIBLE_DESTINATION
>> 2016-04-07 07:52:09.215582 [NOTICE] switch_channel.c:4804 Hangup
>> sofia/external/8 at MY_OPENSIPS_DOMAIN [CS_EXECUTE]
>> [INCOMPATIBLE_DESTINATION]
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:2796
>> sofia/external/8 at MY_OPENSIPS_DOMAIN skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:539
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State EXECUTE going to sleep
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_HANGUP
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:739
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Callstate Change RINGING -> HANGUP
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State HANGUP
>> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:425
>> sofia/external/8 at MY_OPENSIPS_DOMAIN Overriding SIP cause 488 with 488
>> from the other leg
>> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:431 Channel
>> sofia/external/8 at MY_OPENSIPS_DOMAIN hanging up, cause:
>> INCOMPATIBLE_DESTINATION
>> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:568 Responding to INVITE
>> with: 488
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard HANGUP, cause:
>> INCOMPATIBLE_DESTINATION
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:741
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State HANGUP going to sleep
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:508
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_REPORTING
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State REPORTING
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:104
>> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard REPORTING, cause:
>> INCOMPATIBLE_DESTINATION
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:827
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State REPORTING going to sleep
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:499
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_session.c:1646 Session 1
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Locked, Waiting on external
>> entities
>> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1664 Session 1
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Ended
>> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1668 Close
>> Channel sofia/external/8 at MY_OPENSIPS_DOMAIN [CS_DESTROY]
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:630
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) Running State Change CS_DESTROY
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State DESTROY
>> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:341
>> sofia/external/8 at MY_OPENSIPS_DOMAIN SOFIA DESTROY
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:111
>> sofia/external/8 at MY_OPENSIPS_DOMAIN Standard DESTROY
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
>> (sofia/external/8 at MY_OPENSIPS_DOMAIN) State DESTROY going to sleep
>> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1664 Session 2
>> (sofia/external/7906*******) Ended
>> 2016-04-07 07:52:09.215582 [NOTICE] switch_core_session.c:1668 Close
>> Channel sofia/external/7906******* [CS_DESTROY]
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:630
>> (sofia/external/7906*******) Running State Change CS_DESTROY
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
>> (sofia/external/7906*******) State DESTROY
>> 2016-04-07 07:52:09.215582 [DEBUG] mod_sofia.c:341
>> sofia/external/7906******* SOFIA DESTROY
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:111
>> sofia/external/7906******* Standard DESTROY
>> 2016-04-07 07:52:09.215582 [DEBUG] switch_core_state_machine.c:640
>> (sofia/external/7906*******) State DESTROY going to sleep
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
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From olegstolyar at gmail.com  Fri Apr  8 17:49:09 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Fri, 8 Apr 2016 06:49:09 -0700
Subject: [Freeswitch-users] OPUS and CPU load
In-Reply-To: <57072DBB.9050004@coppice.org>
References: <CAKThY064St3fVKNEfjyYnWaCKxOW4uGQ7-zKVniM4z4FQFMUYA@mail.gmail.com>
	<F944D37F-985B-47E1-A3DA-6569B9638DAE@jerris.com>
	<CAC8JMOJR_g751uKz_eduy1m2kGzHMx3UbOqwcOhydE+Mg_n11Q@mail.gmail.com>
	<57072DBB.9050004@coppice.org>
Message-ID: <CAC8JMOLQTvPPTVqLUdB=Z2cSwo+YQUvgAYMqVwQfS0-5-6KUgA@mail.gmail.com>

Steve, you are correct, I was comparing Opus to my current production mix
of G.711 and  G.722.  Here are more precise results on AWS M3.2xl instance
with 8 CPUs for a 10 person conference with all clients connecting from
Chrome via WebRTC.
FS CPU usage was jumping more or less within these ranges throughout the
conference:

PCMU: 4.3% - 5.0%
722: 7.6% - 8.3%
OPUS: 25% - 50%

On Thu, Apr 7, 2016 at 9:04 PM, Steve Underwood <steveu at coppice.org> wrote:

> You should be seeing a significant different between G.711 and G.722, so
> bundling them together doesn't make sense. G.711 takes only a minute
> amount of CPU effort. G.722 takes a significant amount.
>
> Regards,
> Steve
>
> On 04/08/2016 12:49 AM, Oleg Stolyar wrote:
> > In my tests OPUS transcoding seems to consume 5-6 times more CPU then
> > G.711 or G.722.
> >
> > On Thu, Apr 7, 2016 at 9:39 AM, Michael Jerris <mike at jerris.com
> > <mailto:mike at jerris.com>> wrote:
> >
> >     I've seen estimates that opus is up to twice as heavy as g729.
> >     webrtc encryption adds some (although not huge) load on top of
> >     that.  20 opus sessions per cpu seems low, are these particularly
> >     slow processors or virtual processors?
> >
> >
> >     > On Apr 7, 2016, at 12:25 PM, Agust? Ubalde
> >     <aubalde at presenceco.com <mailto:aubalde at presenceco.com>> wrote:
> >     >
> >     > Hi all,,
> >     >
> >     > I have found that the CPU usage with users WebRTC + OPUS is
> >     high. With 2vCPUs, 10 users use 25% of CPU. Anyone know if this
> >     load is normal?
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From stasan89 at gmail.com  Fri Apr  8 17:59:54 2016
From: stasan89 at gmail.com (=?UTF-8?B?0KHRgtCw0YEg0KLQtdC70YzQvdC+0LI=?=)
Date: Fri, 8 Apr 2016 16:59:54 +0300
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
	configuring NAT on Amazon EC2
Message-ID: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>

Hello.

When using a call or conference through sip ? freeswitch with external
provider there is a problem ? the call is interrupted in 30 seconds. Though
the sound goes all right.
I think that it caused by the NAT settings for freeswitch, but I don't
understand how to adjust it correctly.
At start of freeswitch I see the following mistakes in the tracking data:
2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP 1/5
2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP 2/5
2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP 3/5
2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP 4/5
2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP 5/5
2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for PMP
[general error]
2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP
2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT
devices detected!

Despite of this mistake, conference communication between two internal
users works normally. The problem arises at a call through external
provider.

We have the following architecture:
In a cloud of Amazon EC2 there are 2 servers ? opensips and freeswitch,
both for NAT for external clients, but have an opportunity to work with
each other directly.
opensips has the internal address 172.31.0.169 and external 52. *.*.177
freeswitch has the internal address 172.31.22.124 and external 52. *.*.198

In fact, freeswitch acts only for conferences, and is ready for use of a
remote DB on opensips.
The auto-nat settings by default didn't work. The problem is in the
external profile settings as far as I understand.

I have filled and created the following configuration:
vars.xml
  <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto?/>
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198?/> <!? public
freeswitch ip ?>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198?/> <!? public
freeswitch ip ?>
  <!-- External SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
  <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
  <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/tls"/>

sip_profile/external.xml
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>

    <param name="ext-rtp-ip" value=?52.*.*.198?/> <!? public freeswitch ip
?>
    <param name="ext-sip-ip" value=?52.*.*.198?/> <!? public freeswitch ip
?>
In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
(that would be logical), but in that case conferences didn't work at all
and errors below appeared:
[ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
Also I tried to put such configuration:
    <param name="rtp-ip" value="auto"/>
    <param name="sip-ip" value="52.*.*.198?/>
but it also hasn't helped to solve the problem.

autoload_configs/switch.conf.xml
    <param name="rtp-start-port" value="16384"/>
    <param name="rtp-end-port" value="32768"/>

"sofia status" looks as follows:
                     Name       Type
                                      Data    State
=================================================================================================
            172.31.22.124      alias
internal    ALIASED
                 external    profile
sip:mod_sofia at 52.*.*.198:5060
RUNNING (0)
                 external    profile
sip:mod_sofia at 52.*.*.198:5061
RUNNING (0) (TLS)
 external::*********.com    gateway
sip:USER@*********.com
REGED
                 internal    profile
sip:mod_sofia at 52.*.*.198:5080
RUNNING (0)
                 internal    profile
sip:mod_sofia at 52.*.*.198:5081
RUNNING (0) (TLS)
=================================================================================================
2 profiles 1 alias

"sofia status profile external" looks as follows:
=================================================================================================
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               172.31.22.124
Ext-RTP-IP           52.*.*.198
SIP-IP               172.31.22.124
Ext-SIP-IP           52.*.*.198
URL                  sip:mod_sofia at 52.*.*.198:5060
BIND-URL             sip:mod_sofia at 52.
*.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
TLS-URL              sip:mod_sofia at 52.*.*.198:5061
TLS-BIND-URL         sips:mod_sofia at 52.
*.*.198:5061;maddr=172.31.22.124;transport=tls
HOLD-MUSIC           local_stream://moh
OUTBOUND-PROXY       N/A
CODECS IN            PCMA
CODECS OUT           PCMA
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



What do I adjust wrong? Whether there is some opportunity, to tell
freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
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From olegstolyar at gmail.com  Fri Apr  8 18:19:09 2016
From: olegstolyar at gmail.com (Oleg Stolyar)
Date: Fri, 8 Apr 2016 07:19:09 -0700
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
 configuring NAT on Amazon EC2
In-Reply-To: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>
References: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>
Message-ID: <CAC8JMOKX7wkSG_140qWMedyz56E5aHyFxG0jb5+87G9iG9qVzQ@mail.gmail.com>

Try disabling session timers in the sip profile.  I think that line is
commented out by default, so uncomment it.

<param name="enable-timer" value="false"/>

On Fri, Apr 8, 2016 at 6:59 AM, ???? ??????? <stasan89 at gmail.com> wrote:

> Hello.
>
> When using a call or conference through sip ? freeswitch with external
> provider there is a problem ? the call is interrupted in 30 seconds. Though
> the sound goes all right.
> I think that it caused by the NAT settings for freeswitch, but I don't
> understand how to adjust it correctly.
> At start of freeswitch I see the following mistakes in the tracking data:
> 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
> 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP 1/5
> 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP 2/5
> 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP 3/5
> 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP 4/5
> 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP 5/5
> 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for PMP
> [general error]
> 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP
> 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT
> devices detected!
>
> Despite of this mistake, conference communication between two internal
> users works normally. The problem arises at a call through external
> provider.
>
> We have the following architecture:
> In a cloud of Amazon EC2 there are 2 servers ? opensips and freeswitch,
> both for NAT for external clients, but have an opportunity to work with
> each other directly.
> opensips has the internal address 172.31.0.169 and external 52. *.*.177
> freeswitch has the internal address 172.31.22.124 and external 52. *.*.198
>
> In fact, freeswitch acts only for conferences, and is ready for use of a
> remote DB on opensips.
> The auto-nat settings by default didn't work. The problem is in the
> external profile settings as far as I understand.
>
> I have filled and created the following configuration:
> vars.xml
>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto?/>
>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198?/> <!? public
> freeswitch ip ?>
>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198?/> <!? public
> freeswitch ip ?>
>   <!-- External SIP Profile -->
>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>   <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/tls"/>
>
> sip_profile/external.xml
>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>     <param name="sip-ip" value="$${local_ip_v4}"/>
>
>     <param name="ext-rtp-ip" value=?52.*.*.198?/> <!? public freeswitch ip
> ?>
>     <param name="ext-sip-ip" value=?52.*.*.198?/> <!? public freeswitch ip
> ?>
> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
> ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
> (that would be logical), but in that case conferences didn't work at all
> and errors below appeared:
> [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
> Also I tried to put such configuration:
>     <param name="rtp-ip" value="auto"/>
>     <param name="sip-ip" value="52.*.*.198?/>
> but it also hasn't helped to solve the problem.
>
> autoload_configs/switch.conf.xml
>     <param name="rtp-start-port" value="16384"/>
>     <param name="rtp-end-port" value="32768"/>
>
> "sofia status" looks as follows:
>                      Name       Type
>                                       Data    State
>
> =================================================================================================
>             172.31.22.124      alias
> internal    ALIASED
>                  external    profile               sip:mod_sofia at 52.*.*.198:5060
> RUNNING (0)
>                  external    profile               sip:mod_sofia at 52.*.*.198:5061
> RUNNING (0) (TLS)
>  external::*********.com    gateway                      sip:USER@*********.com
> REGED
>                  internal    profile               sip:mod_sofia at 52.*.*.198:5080
> RUNNING (0)
>                  internal    profile               sip:mod_sofia at 52.*.*.198:5081
> RUNNING (0) (TLS)
>
> =================================================================================================
> 2 profiles 1 alias
>
> "sofia status profile external" looks as follows:
>
> =================================================================================================
> 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               172.31.22.124
> Ext-RTP-IP           52.*.*.198
> SIP-IP               172.31.22.124
> Ext-SIP-IP           52.*.*.198
> URL                  sip:mod_sofia at 52.*.*.198:5060
> BIND-URL             sip:mod_sofia at 52.
> *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
> TLS-URL              sip:mod_sofia at 52.*.*.198:5061
> TLS-BIND-URL         sips:mod_sofia at 52.
> *.*.198:5061;maddr=172.31.22.124;transport=tls
> HOLD-MUSIC           local_stream://moh
> OUTBOUND-PROXY       N/A
> CODECS IN            PCMA
> CODECS OUT           PCMA
> 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
>
>
>
> What do I adjust wrong? Whether there is some opportunity, to tell
> freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From stasan89 at gmail.com  Fri Apr  8 18:34:21 2016
From: stasan89 at gmail.com (=?UTF-8?B?0KHRgtCw0YEg0KLQtdC70YzQvdC+0LI=?=)
Date: Fri, 8 Apr 2016 17:34:21 +0300
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
 configuring NAT on Amazon EC2
In-Reply-To: <CAC8JMOKX7wkSG_140qWMedyz56E5aHyFxG0jb5+87G9iG9qVzQ@mail.gmail.com>
References: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>
	<CAC8JMOKX7wkSG_140qWMedyz56E5aHyFxG0jb5+87G9iG9qVzQ@mail.gmail.com>
Message-ID: <CAC89h=L3a3ZzOvYYpq4Jma8JsN6+WmvUGkEmF+7mO4WvyNoGTg@mail.gmail.com>

I already tried disabling timers, does not work.

2016-04-08 17:19 GMT+03:00 Oleg Stolyar <olegstolyar at gmail.com>:

> Try disabling session timers in the sip profile.  I think that line is
> commented out by default, so uncomment it.
>
> <param name="enable-timer" value="false"/>
>
> On Fri, Apr 8, 2016 at 6:59 AM, ???? ??????? <stasan89 at gmail.com> wrote:
>
>> Hello.
>>
>> When using a call or conference through sip ? freeswitch with external
>> provider there is a problem ? the call is interrupted in 30 seconds. Though
>> the sound goes all right.
>> I think that it caused by the NAT settings for freeswitch, but I don't
>> understand how to adjust it correctly.
>> At start of freeswitch I see the following mistakes in the tracking data:
>> 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
>> 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP 1/5
>> 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP 2/5
>> 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP 3/5
>> 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP 4/5
>> 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP 5/5
>> 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for PMP
>> [general error]
>> 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP
>> 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT
>> devices detected!
>>
>> Despite of this mistake, conference communication between two internal
>> users works normally. The problem arises at a call through external
>> provider.
>>
>> We have the following architecture:
>> In a cloud of Amazon EC2 there are 2 servers ? opensips and freeswitch,
>> both for NAT for external clients, but have an opportunity to work with
>> each other directly.
>> opensips has the internal address 172.31.0.169 and external 52. *.*.177
>> freeswitch has the internal address 172.31.22.124 and external 52. *.*.198
>>
>> In fact, freeswitch acts only for conferences, and is ready for use of a
>> remote DB on opensips.
>> The auto-nat settings by default didn't work. The problem is in the
>> external profile settings as far as I understand.
>>
>> I have filled and created the following configuration:
>> vars.xml
>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto?/>
>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198?/> <!? public
>> freeswitch ip ?>
>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198?/> <!? public
>> freeswitch ip ?>
>>   <!-- External SIP Profile -->
>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>   <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/tls"/>
>>
>> sip_profile/external.xml
>>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>>     <param name="sip-ip" value="$${local_ip_v4}"/>
>>
>>     <param name="ext-rtp-ip" value=?52.*.*.198?/> <!? public freeswitch
>> ip ?>
>>     <param name="ext-sip-ip" value=?52.*.*.198?/> <!? public freeswitch
>> ip ?>
>> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
>> ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
>> (that would be logical), but in that case conferences didn't work at all
>> and errors below appeared:
>> [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
>> Also I tried to put such configuration:
>>     <param name="rtp-ip" value="auto"/>
>>     <param name="sip-ip" value="52.*.*.198?/>
>> but it also hasn't helped to solve the problem.
>>
>> autoload_configs/switch.conf.xml
>>     <param name="rtp-start-port" value="16384"/>
>>     <param name="rtp-end-port" value="32768"/>
>>
>> "sofia status" looks as follows:
>>                      Name       Type
>>                                       Data    State
>>
>> =================================================================================================
>>             172.31.22.124      alias
>> internal    ALIASED
>>                  external    profile               sip:mod_sofia at 52.*.*.198:5060
>> RUNNING (0)
>>                  external    profile               sip:mod_sofia at 52.*.*.198:5061
>> RUNNING (0) (TLS)
>>  external::*********.com    gateway                      sip:USER@*********.com
>> REGED
>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5080
>> RUNNING (0)
>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5081
>> RUNNING (0) (TLS)
>>
>> =================================================================================================
>> 2 profiles 1 alias
>>
>> "sofia status profile external" looks as follows:
>>
>> =================================================================================================
>> 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               172.31.22.124
>> Ext-RTP-IP           52.*.*.198
>> SIP-IP               172.31.22.124
>> Ext-SIP-IP           52.*.*.198
>> URL                  sip:mod_sofia at 52.*.*.198:5060
>> BIND-URL             sip:mod_sofia at 52.
>> *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
>> TLS-URL              sip:mod_sofia at 52.*.*.198:5061
>> TLS-BIND-URL         sips:mod_sofia at 52.
>> *.*.198:5061;maddr=172.31.22.124;transport=tls
>> HOLD-MUSIC           local_stream://moh
>> OUTBOUND-PROXY       N/A
>> CODECS IN            PCMA
>> CODECS OUT           PCMA
>> 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
>>
>>
>>
>> What do I adjust wrong? Whether there is some opportunity, to tell
>> freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jurijs.ivolga at gmail.com  Fri Apr  8 18:37:44 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Fri, 8 Apr 2016 17:37:44 +0300
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
 configuring NAT on Amazon EC2
In-Reply-To: <CAC89h=L3a3ZzOvYYpq4Jma8JsN6+WmvUGkEmF+7mO4WvyNoGTg@mail.gmail.com>
References: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>
	<CAC8JMOKX7wkSG_140qWMedyz56E5aHyFxG0jb5+87G9iG9qVzQ@mail.gmail.com>
	<CAC89h=L3a3ZzOvYYpq4Jma8JsN6+WmvUGkEmF+7mO4WvyNoGTg@mail.gmail.com>
Message-ID: <CAOrA2UbnqXs49O18a3eeb-8YRvvxmOS7fPEORtyHEd57O_Q94w@mail.gmail.com>

Hi,

I would recommend you to capture SIP packets  during call  on Freeswitch
server and send it here, I will take a look on it.

With kind regards,

Jurijs

On Fri, Apr 8, 2016 at 5:34 PM, ???? ??????? <stasan89 at gmail.com> wrote:

> I already tried disabling timers, does not work.
>
> 2016-04-08 17:19 GMT+03:00 Oleg Stolyar <olegstolyar at gmail.com>:
>
>> Try disabling session timers in the sip profile.  I think that line is
>> commented out by default, so uncomment it.
>>
>> <param name="enable-timer" value="false"/>
>>
>> On Fri, Apr 8, 2016 at 6:59 AM, ???? ??????? <stasan89 at gmail.com> wrote:
>>
>>> Hello.
>>>
>>> When using a call or conference through sip ? freeswitch with external
>>> provider there is a problem ? the call is interrupted in 30 seconds. Though
>>> the sound goes all right.
>>> I think that it caused by the NAT settings for freeswitch, but I don't
>>> understand how to adjust it correctly.
>>> At start of freeswitch I see the following mistakes in the tracking data:
>>> 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
>>> 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP 1/5
>>> 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP 2/5
>>> 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP 3/5
>>> 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP 4/5
>>> 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP 5/5
>>> 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for PMP
>>> [general error]
>>> 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP
>>> 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT
>>> devices detected!
>>>
>>> Despite of this mistake, conference communication between two internal
>>> users works normally. The problem arises at a call through external
>>> provider.
>>>
>>> We have the following architecture:
>>> In a cloud of Amazon EC2 there are 2 servers ? opensips and freeswitch,
>>> both for NAT for external clients, but have an opportunity to work with
>>> each other directly.
>>> opensips has the internal address 172.31.0.169 and external 52. *.*.177
>>> freeswitch has the internal address 172.31.22.124 and external 52.
>>> *.*.198
>>>
>>> In fact, freeswitch acts only for conferences, and is ready for use of a
>>> remote DB on opensips.
>>> The auto-nat settings by default didn't work. The problem is in the
>>> external profile settings as far as I understand.
>>>
>>> I have filled and created the following configuration:
>>> vars.xml
>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto?/>
>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198?/> <!?
>>> public freeswitch ip ?>
>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198?/> <!?
>>> public freeswitch ip ?>
>>>   <!-- External SIP Profile -->
>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>   <X-PRE-PROCESS cmd="set"
>>> data="external_ssl_dir=$${base_dir}/conf/tls"/>
>>>
>>> sip_profile/external.xml
>>>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>>>     <param name="sip-ip" value="$${local_ip_v4}"/>
>>>
>>>     <param name="ext-rtp-ip" value=?52.*.*.198?/> <!? public freeswitch
>>> ip ?>
>>>     <param name="ext-sip-ip" value=?52.*.*.198?/> <!? public freeswitch
>>> ip ?>
>>> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
>>> ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
>>> (that would be logical), but in that case conferences didn't work at all
>>> and errors below appeared:
>>> [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
>>> Also I tried to put such configuration:
>>>     <param name="rtp-ip" value="auto"/>
>>>     <param name="sip-ip" value="52.*.*.198?/>
>>> but it also hasn't helped to solve the problem.
>>>
>>> autoload_configs/switch.conf.xml
>>>     <param name="rtp-start-port" value="16384"/>
>>>     <param name="rtp-end-port" value="32768"/>
>>>
>>> "sofia status" looks as follows:
>>>                      Name       Type
>>>                                       Data    State
>>>
>>> =================================================================================================
>>>             172.31.22.124      alias
>>>                                   internal    ALIASED
>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5060
>>> RUNNING (0)
>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5061
>>> RUNNING (0) (TLS)
>>>  external::*********.com    gateway                      sip:USER@*********.com
>>> REGED
>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5080
>>> RUNNING (0)
>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5081
>>> RUNNING (0) (TLS)
>>>
>>> =================================================================================================
>>> 2 profiles 1 alias
>>>
>>> "sofia status profile external" looks as follows:
>>>
>>> =================================================================================================
>>> 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               172.31.22.124
>>> Ext-RTP-IP           52.*.*.198
>>> SIP-IP               172.31.22.124
>>> Ext-SIP-IP           52.*.*.198
>>> URL                  sip:mod_sofia at 52.*.*.198:5060
>>> BIND-URL             sip:mod_sofia at 52.
>>> *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
>>> TLS-URL              sip:mod_sofia at 52.*.*.198:5061
>>> TLS-BIND-URL         sips:mod_sofia at 52.
>>> *.*.198:5061;maddr=172.31.22.124;transport=tls
>>> HOLD-MUSIC           local_stream://moh
>>> OUTBOUND-PROXY       N/A
>>> CODECS IN            PCMA
>>> CODECS OUT           PCMA
>>> 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
>>>
>>>
>>>
>>> What do I adjust wrong? Whether there is some opportunity, to tell
>>> freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From stasan89 at gmail.com  Fri Apr  8 19:02:09 2016
From: stasan89 at gmail.com (=?UTF-8?B?0KHRgtCw0YEg0KLQtdC70YzQvdC+0LI=?=)
Date: Fri, 8 Apr 2016 18:02:09 +0300
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
 configuring NAT on Amazon EC2
In-Reply-To: <CAOrA2UbnqXs49O18a3eeb-8YRvvxmOS7fPEORtyHEd57O_Q94w@mail.gmail.com>
References: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>
	<CAC8JMOKX7wkSG_140qWMedyz56E5aHyFxG0jb5+87G9iG9qVzQ@mail.gmail.com>
	<CAC89h=L3a3ZzOvYYpq4Jma8JsN6+WmvUGkEmF+7mO4WvyNoGTg@mail.gmail.com>
	<CAOrA2UbnqXs49O18a3eeb-8YRvvxmOS7fPEORtyHEd57O_Q94w@mail.gmail.com>
Message-ID: <CAC89h=KCcOx5ftG4etYsrV1unEp=BuB-hMBcHQrOzJ2OJvRBog@mail.gmail.com>

Yes, of cause. I hide some ip and real phone numbers.
178.*.*.12 - ip of provider.

*On start call:*
2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
sofia/external/8 at sip0.MY_DOMAIN.com [c618eafe-fd98-11e5-a353-831849fc41a3]
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_NEW
2016-04-08 10:47:10.503262 [DEBUG] sofia.c:9248 sofia/external/
8 at sip0.MY_DOMAIN.com receiving invite from 172.31.0.169:5060 version:
1.6.6  64bit
2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel sofia/external/
8 at sip0.MY_DOMAIN.com entering state [received][100]
2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6770 Remote SDP:
v=0
o=- 1460126829 1 IN IP4 85.*.*.4
s=portsip.com
c=IN IP4 52.*.*.177
t=0 0
m=audio 40082 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=nortpproxy:yes

2016-04-08 10:47:10.503262 [DEBUG] sofia.c:7125 (sofia/external/
8 at sip0.MY_DOMAIN.com) State Change CS_NEW -> CS_INIT
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:492
(sofia/external/8 at sip0.MY_DOMAIN.com) State NEW
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_INIT
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
(sofia/external/8 at sip0.MY_DOMAIN.com) State INIT
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88 sofia/external/
8 at sip0.MY_DOMAIN.com SOFIA INIT
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
sofia/external/8 at sip0.MY_DOMAIN.com Standard INIT
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
(sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_INIT -> CS_ROUTING
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
(sofia/external/8 at sip0.MY_DOMAIN.com) State INIT going to sleep
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
2016-04-08 10:47:10.503262 [DEBUG] switch_channel.c:2247 (sofia/external/
8 at sip0.MY_DOMAIN.com) Callstate Change DOWN -> RINGING
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
(sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
8 at sip0.MY_DOMAIN.com SOFIA ROUTING
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
<8>->7906******* in context public
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
[public->from_opensips] continue=false
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS) [from_opensips]
network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/ break=on-false
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
transfer(${destination_number} XML default)
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
(sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING -> CS_EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
(sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
(sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com transfer(7906******* XML
default)
2016-04-08 10:47:10.503262 [DEBUG] switch_ivr.c:2085 (sofia/external/
8 at sip0.MY_DOMAIN.com) State Change CS_EXECUTE -> CS_ROUTING
2016-04-08 10:47:10.503262 [NOTICE] switch_ivr.c:2092 Transfer
sofia/external/8 at sip0.MY_DOMAIN.com to XML[7906*******@default]
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
(sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
(sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
8 at sip0.MY_DOMAIN.com SOFIA ROUTING
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
<8>->7906******* in context default
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing [default->unloop]
continue=false
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
[default->tod_example] continue=true
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Date/Time Match (PASS)
[tod_example] break=on-false
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action set(open=true)
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
[default->outbound_calls_to_freelycall] continue=false
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
[outbound_calls_to_freelycall] destination_number(7906*******) =~ /^(.+)/
break=on-true
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
set(hangup_after_bridge=true)
Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action bridge(sofia/gateway/
freelycall.com/7906*******)
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
(sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING -> CS_EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
(sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
(sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com set(open=true)
2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET sofia/external/
8 at sip0.MY_DOMAIN.com [open]=[true]
EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com set(hangup_after_bridge=true)
2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET sofia/external/
8 at sip0.MY_DOMAIN.com [hangup_after_bridge]=[true]
EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com bridge(sofia/gateway/
freelycall.com/7906*******)
2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:2128 Parsing
global variables
2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
sofia/external/7906******* [c619366c-fd98-11e5-a35c-831849fc41a3]
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:4776
(sofia/external/7906*******) State Change CS_NEW -> CS_INIT
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
(sofia/external/7906*******) Running State Change CS_INIT
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
(sofia/external/7906*******) State INIT
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88
sofia/external/7906******* SOFIA INIT
2016-04-08 10:47:10.503262 [DEBUG] sofia_glue.c:1257
sofia/external/7906******* sending invite version: 1.6.6  64bit
Local SDP:
v=0
o=FreeSWITCH 1460100428 1460100429 IN IP4 52.*.*.198
s=FreeSWITCH
c=IN IP4 52.*.*.198
t=0 0
m=audio 26402 RTP/AVP 8 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
a=sendrecv

2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
sofia/external/7906******* Standard INIT
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
(sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
(sofia/external/7906*******) State INIT going to sleep
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
(sofia/external/7906*******) Running State Change CS_ROUTING
2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel
sofia/external/7906******* entering state [calling][0]
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
(sofia/external/7906*******) State ROUTING
2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141
sofia/external/7906******* SOFIA ROUTING
2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:67
(sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
(sofia/external/7906*******) State ROUTING going to sleep
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
(sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
(sofia/external/7906*******) State CONSUME_MEDIA
2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
(sofia/external/7906*******) State CONSUME_MEDIA going to sleep
2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel
sofia/external/7906******* entering state [proceeding][183]
2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6770 Remote SDP:
v=0
o=root 153112258 153112258 IN IP4 178.*.*.12
s=Asterisk PBX 11.11.0
c=IN IP4 178.*.*.12
t=0 0
m=audio 17362 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
telephone-event payload to 101 at 8000
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
sofia/external/7906******* PCMA/8000 20 ms 160 samples 64000 bits 1 channels
2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111
sofia/external/7906******* Original read codec set to PCMA:8
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
telephone-event payload to 101 at 8000
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
sofia/external/7906******* Set 2833 dtmf send payload to 101 recv payload
to 101
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
[sofia/external/7906*******] 172.31.22.124 port 26402 -> 178.*.*.12 port
17362 codec: 8 ms: 20
2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer [soft]
160 bytes per 20ms
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
sofia/external/7906******* Set 2833 dtmf send payload to 101
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
sofia/external/7906******* Set 2833 dtmf receive payload to 101
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
sofia/external/7906******* Set rtp dtmf delay to 40
2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
sofia/external/7906*******!
2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468
(sofia/external/7906*******) Callstate Change DOWN -> EARLY
2016-04-08 10:47:17.443282 [INFO] switch_ivr_originate.c:3557 Sending early
media
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
telephone-event payload to 101 at 8000
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
sofia/external/8 at sip0.MY_DOMAIN.com PCMA/8000 20 ms 160 samples 64000 bits
1 channels
2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111 sofia/external/
8 at sip0.MY_DOMAIN.com Original read codec set to PCMA:8
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
telephone-event payload to 101 at 8000
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485 sofia/external/
8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101 recv payload to 101
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
[sofia/external/8 at sip0.MY_DOMAIN.com] 172.31.22.124 port 30630 ->
52.*.*.177 port 40082 codec: 8 ms: 20
2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer [soft]
160 bytes per 20ms
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332 sofia/external/
8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339 sofia/external/
8 at sip0.MY_DOMAIN.com Set 2833 dtmf receive payload to 101
2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362 sofia/external/
8 at sip0.MY_DOMAIN.com Set rtp dtmf delay to 40
2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
sofia/external/8 at sip0.MY_DOMAIN.com!
2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468 (sofia/external/
8 at sip0.MY_DOMAIN.com) Callstate Change RINGING -> EARLY
2016-04-08 10:47:17.443282 [DEBUG] mod_sofia.c:2330 Ring SDP:
v=0
o=FreeSWITCH 1460096207 1460096208 IN IP4 52.*.*.198
s=FreeSWITCH
c=IN IP4 52.*.*.198
t=0 0
m=audio 30630 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel sofia/external/
8 at sip0.MY_DOMAIN.com entering state [early][183]
2016-04-08 10:47:17.443282 [DEBUG] switch_ivr_originate.c:3608 Originate
Resulted in Success: [sofia/external/7906*******]
2016-04-08 10:47:17.463254 [DEBUG] switch_ivr_bridge.c:1591
(sofia/external/7906*******) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:473
(sofia/external/7906*******) Running State Change CS_EXCHANGE_MEDIA
2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:542
(sofia/external/7906*******) State EXCHANGE_MEDIA
2016-04-08 10:47:17.463254 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
2016-04-08 10:47:17.503264 [DEBUG] switch_rtp.c:6654 Correct audio ip/port
confirmed.
2016-04-08 10:47:17.663261 [DEBUG] switch_rtp.c:6654 Correct audio ip/port
confirmed.
2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
sofia/external/7906******* entering state [completing][200]
2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6767 Duplicate SDP
v=0
o=root 153112258 153112258 IN IP4 178.*.*.12
s=Asterisk PBX 11.11.0
c=IN IP4 178.*.*.12
t=0 0
m=audio 17362 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
sofia/external/7906******* entering state [ready][200]
2016-04-08 10:47:21.323279 [NOTICE] sofia.c:7665 Channel
[sofia/external/7906*******] has been answered
2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767
(sofia/external/7906*******) Callstate Change EARLY -> ACTIVE
2016-04-08 10:47:21.323279 [DEBUG] mod_sofia.c:799 Local SDP sofia/external/
8 at sip0.MY_DOMAIN.com:
v=0
o=FreeSWITCH 1460096207 1460096209 IN IP4 52.*.*.198
s=FreeSWITCH
c=IN IP4 52.*.*.198
t=0 0
m=audio 30630 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel sofia/external/
8 at sip0.MY_DOMAIN.com entering state [completed][200]
2016-04-08 10:47:21.323279 [NOTICE] switch_ivr_bridge.c:616 Channel
[sofia/external/8 at sip0.MY_DOMAIN.com] has been answered
2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767 (sofia/external/
8 at sip0.MY_DOMAIN.com) Callstate Change EARLY -> ACTIVE
2016-04-08 10:47:21.383279 [DEBUG] switch_rtp.c:6654 Correct audio ip/port
confirmed.
2016-04-08 10:47:21.423259 [DEBUG] switch_rtp.c:6654 Correct audio ip/port
confirmed.


*And after 30 seconds:*
2016-04-08 10:47:53.343283 [DEBUG] sofia.c:6760 Channel sofia/external/
8 at sip0.MY_DOMAIN.com entering state [terminating][0]
2016-04-08 10:47:53.343283 [NOTICE] sofia.c:7779 Hangup sofia/external/
8 at sip0.MY_DOMAIN.com [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:699 sofia/external/
8 at sip0.MY_DOMAIN.com ending bridge by request from write function
2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD
DONE [sofia/external/7906*******]
2016-04-08 10:47:53.343283 [NOTICE] switch_ivr_bridge.c:881 Hangup
sofia/external/7906******* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:542
(sofia/external/7906*******) State EXCHANGE_MEDIA going to sleep
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
(sofia/external/7906*******) Running State Change CS_HANGUP
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
(sofia/external/7906*******) Callstate Change ACTIVE -> HANGUP
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
(sofia/external/7906*******) State HANGUP
2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel
sofia/external/7906******* hanging up, cause: NORMAL_CLEARING
2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:484 Sending BYE to
sofia/external/7906*******
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
sofia/external/7906******* Standard HANGUP, cause: NORMAL_CLEARING
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
(sofia/external/7906*******) State HANGUP going to sleep
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
(sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
(sofia/external/7906*******) Running State Change CS_REPORTING
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
(sofia/external/7906*******) State REPORTING
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
sofia/external/7906******* Standard REPORTING, cause: NORMAL_CLEARING
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
(sofia/external/7906*******) State REPORTING going to sleep
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
(sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY
2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 2
(sofia/external/7906*******) Locked, Waiting on external entities
2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:705 sofia/external/
8 at sip0.MY_DOMAIN.com ending bridge by request from read function
2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD
DONE [sofia/external/8 at sip0.MY_DOMAIN.com]
2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:1692 sofia/external/
8 at sip0.MY_DOMAIN.com skip receive message [UNBRIDGE] (channel is hungup
already)
2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:2796
sofia/external/8 at sip0.MY_DOMAIN.com skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:539
(sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_HANGUP
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
(sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change ACTIVE -> HANGUP
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
(sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP
2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel sofia/external/
8 at sip0.MY_DOMAIN.com hanging up, cause: NORMAL_UNSPECIFIED
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
sofia/external/8 at sip0.MY_DOMAIN.com Standard HANGUP, cause:
NORMAL_UNSPECIFIED
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
(sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP going to sleep
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
(sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_HANGUP -> CS_REPORTING
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
(sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_REPORTING
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
(sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
sofia/external/8 at sip0.MY_DOMAIN.com Standard REPORTING, cause:
NORMAL_UNSPECIFIED
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
(sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING going to sleep
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
(sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_REPORTING ->
CS_DESTROY
2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 1
(sofia/external/8 at sip0.MY_DOMAIN.com) Locked, Waiting on external entities
2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session 1
(sofia/external/8 at sip0.MY_DOMAIN.com) Ended
2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
Channel sofia/external/8 at sip0.MY_DOMAIN.com [CS_DESTROY]
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
(sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_DESTROY
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
(sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY
2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341 sofia/external/
8 at sip0.MY_DOMAIN.com SOFIA DESTROY
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
sofia/external/8 at sip0.MY_DOMAIN.com Standard DESTROY
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
(sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY going to sleep
2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session 2
(sofia/external/7906*******) Ended
2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
Channel sofia/external/7906******* [CS_DESTROY]
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
(sofia/external/7906*******) Running State Change CS_DESTROY
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
(sofia/external/7906*******) State DESTROY
2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341
sofia/external/7906******* SOFIA DESTROY
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
sofia/external/7906******* Standard DESTROY
2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
(sofia/external/7906*******) State DESTROY going to sleep

2016-04-08 17:37 GMT+03:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:

> Hi,
>
> I would recommend you to capture SIP packets  during call  on Freeswitch
> server and send it here, I will take a look on it.
>
> With kind regards,
>
> Jurijs
>
> On Fri, Apr 8, 2016 at 5:34 PM, ???? ??????? <stasan89 at gmail.com> wrote:
>
>> I already tried disabling timers, does not work.
>>
>> 2016-04-08 17:19 GMT+03:00 Oleg Stolyar <olegstolyar at gmail.com>:
>>
>>> Try disabling session timers in the sip profile.  I think that line is
>>> commented out by default, so uncomment it.
>>>
>>> <param name="enable-timer" value="false"/>
>>>
>>> On Fri, Apr 8, 2016 at 6:59 AM, ???? ??????? <stasan89 at gmail.com> wrote:
>>>
>>>> Hello.
>>>>
>>>> When using a call or conference through sip ? freeswitch with external
>>>> provider there is a problem ? the call is interrupted in 30 seconds. Though
>>>> the sound goes all right.
>>>> I think that it caused by the NAT settings for freeswitch, but I don't
>>>> understand how to adjust it correctly.
>>>> At start of freeswitch I see the following mistakes in the tracking
>>>> data:
>>>> 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
>>>> 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP 1/5
>>>> 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP 2/5
>>>> 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP 3/5
>>>> 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP 4/5
>>>> 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP 5/5
>>>> 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for
>>>> PMP [general error]
>>>> 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP
>>>> 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT
>>>> devices detected!
>>>>
>>>> Despite of this mistake, conference communication between two internal
>>>> users works normally. The problem arises at a call through external
>>>> provider.
>>>>
>>>> We have the following architecture:
>>>> In a cloud of Amazon EC2 there are 2 servers ? opensips and freeswitch,
>>>> both for NAT for external clients, but have an opportunity to work with
>>>> each other directly.
>>>> opensips has the internal address 172.31.0.169 and external 52. *.*.177
>>>> freeswitch has the internal address 172.31.22.124 and external 52.
>>>> *.*.198
>>>>
>>>> In fact, freeswitch acts only for conferences, and is ready for use of
>>>> a remote DB on opensips.
>>>> The auto-nat settings by default didn't work. The problem is in the
>>>> external profile settings as far as I understand.
>>>>
>>>> I have filled and created the following configuration:
>>>> vars.xml
>>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto?/>
>>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198?/> <!?
>>>> public freeswitch ip ?>
>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198?/> <!?
>>>> public freeswitch ip ?>
>>>>   <!-- External SIP Profile -->
>>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
>>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
>>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>>   <X-PRE-PROCESS cmd="set"
>>>> data="external_ssl_dir=$${base_dir}/conf/tls"/>
>>>>
>>>> sip_profile/external.xml
>>>>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>>>>     <param name="sip-ip" value="$${local_ip_v4}"/>
>>>>
>>>>     <param name="ext-rtp-ip" value=?52.*.*.198?/> <!? public freeswitch
>>>> ip ?>
>>>>     <param name="ext-sip-ip" value=?52.*.*.198?/> <!? public freeswitch
>>>> ip ?>
>>>> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
>>>> ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
>>>> (that would be logical), but in that case conferences didn't work at all
>>>> and errors below appeared:
>>>> [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
>>>> Also I tried to put such configuration:
>>>>     <param name="rtp-ip" value="auto"/>
>>>>     <param name="sip-ip" value="52.*.*.198?/>
>>>> but it also hasn't helped to solve the problem.
>>>>
>>>> autoload_configs/switch.conf.xml
>>>>     <param name="rtp-start-port" value="16384"/>
>>>>     <param name="rtp-end-port" value="32768"/>
>>>>
>>>> "sofia status" looks as follows:
>>>>                      Name       Type
>>>>                                       Data    State
>>>>
>>>> =================================================================================================
>>>>             172.31.22.124      alias
>>>>                                   internal    ALIASED
>>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5060
>>>> RUNNING (0)
>>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5061
>>>> RUNNING (0) (TLS)
>>>>  external::*********.com    gateway                      sip:USER@*********.com
>>>> REGED
>>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5080
>>>> RUNNING (0)
>>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5081
>>>> RUNNING (0) (TLS)
>>>>
>>>> =================================================================================================
>>>> 2 profiles 1 alias
>>>>
>>>> "sofia status profile external" looks as follows:
>>>>
>>>> =================================================================================================
>>>> 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               172.31.22.124
>>>> Ext-RTP-IP           52.*.*.198
>>>> SIP-IP               172.31.22.124
>>>> Ext-SIP-IP           52.*.*.198
>>>> URL                  sip:mod_sofia at 52.*.*.198:5060
>>>> BIND-URL             sip:mod_sofia at 52.
>>>> *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
>>>> TLS-URL              sip:mod_sofia at 52.*.*.198:5061
>>>> TLS-BIND-URL         sips:mod_sofia at 52.
>>>> *.*.198:5061;maddr=172.31.22.124;transport=tls
>>>> HOLD-MUSIC           local_stream://moh
>>>> OUTBOUND-PROXY       N/A
>>>> CODECS IN            PCMA
>>>> CODECS OUT           PCMA
>>>> 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
>>>>
>>>>
>>>>
>>>> What do I adjust wrong? Whether there is some opportunity, to tell
>>>> freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
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From jurijs.ivolga at gmail.com  Fri Apr  8 19:06:43 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Fri, 8 Apr 2016 18:06:43 +0300
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
 configuring NAT on Amazon EC2
In-Reply-To: <CAC89h=KCcOx5ftG4etYsrV1unEp=BuB-hMBcHQrOzJ2OJvRBog@mail.gmail.com>
References: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>
	<CAC8JMOKX7wkSG_140qWMedyz56E5aHyFxG0jb5+87G9iG9qVzQ@mail.gmail.com>
	<CAC89h=L3a3ZzOvYYpq4Jma8JsN6+WmvUGkEmF+7mO4WvyNoGTg@mail.gmail.com>
	<CAOrA2UbnqXs49O18a3eeb-8YRvvxmOS7fPEORtyHEd57O_Q94w@mail.gmail.com>
	<CAC89h=KCcOx5ftG4etYsrV1unEp=BuB-hMBcHQrOzJ2OJvRBog@mail.gmail.com>
Message-ID: <CAOrA2UbxHQ+9TJeX0gCErFMD1uRCmH9bDu76JQo7Rt_mBfqOkw@mail.gmail.com>

Hi,

This is not what I need, please use ngrep:

http://jonathanmanning.com/2009/11/17/how-to-sip-capture-using-ngrep-debug-sip-packets/

With kind regards,

Jurijs

On Fri, Apr 8, 2016 at 6:02 PM, ???? ??????? <stasan89 at gmail.com> wrote:

> Yes, of cause. I hide some ip and real phone numbers.
> 178.*.*.12 - ip of provider.
>
> *On start call:*
> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
> sofia/external/8 at sip0.MY_DOMAIN.com [c618eafe-fd98-11e5-a353-831849fc41a3]
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_NEW
> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:9248 sofia/external/
> 8 at sip0.MY_DOMAIN.com receiving invite from 172.31.0.169:5060 version:
> 1.6.6  64bit
> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel sofia/external/
> 8 at sip0.MY_DOMAIN.com entering state [received][100]
> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6770 Remote SDP:
> v=0
> o=- 1460126829 1 IN IP4 85.*.*.4
> s=portsip.com
> c=IN IP4 52.*.*.177
> t=0 0
> m=audio 40082 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=nortpproxy:yes
>
> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:7125 (sofia/external/
> 8 at sip0.MY_DOMAIN.com) State Change CS_NEW -> CS_INIT
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:492
> (sofia/external/8 at sip0.MY_DOMAIN.com) State NEW
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_INIT
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT
> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88 sofia/external/
> 8 at sip0.MY_DOMAIN.com SOFIA INIT
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
> sofia/external/8 at sip0.MY_DOMAIN.com Standard INIT
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_INIT -> CS_ROUTING
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT going to sleep
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
> 2016-04-08 10:47:10.503262 [DEBUG] switch_channel.c:2247 (sofia/external/
> 8 at sip0.MY_DOMAIN.com) Callstate Change DOWN -> RINGING
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
> <8>->7906******* in context public
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
> [public->from_opensips] continue=false
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
> [from_opensips] network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/
> break=on-false
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
> transfer(${destination_number} XML default)
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
> CS_EXECUTE
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com transfer(7906******* XML
> default)
> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr.c:2085 (sofia/external/
> 8 at sip0.MY_DOMAIN.com) State Change CS_EXECUTE -> CS_ROUTING
> 2016-04-08 10:47:10.503262 [NOTICE] switch_ivr.c:2092 Transfer
> sofia/external/8 at sip0.MY_DOMAIN.com to XML[7906*******@default]
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
> <8>->7906******* in context default
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing [default->unloop]
> continue=false
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
> [default->tod_example] continue=true
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Date/Time Match (PASS)
> [tod_example] break=on-false
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action set(open=true)
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
> [default->outbound_calls_to_freelycall] continue=false
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
> [outbound_calls_to_freelycall] destination_number(7906*******) =~ /^(.+)/
> break=on-true
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
> set(hangup_after_bridge=true)
> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action bridge(sofia/gateway/
> freelycall.com/7906*******)
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
> CS_EXECUTE
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com set(open=true)
> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET sofia/external/
> 8 at sip0.MY_DOMAIN.com [open]=[true]
> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com set(hangup_after_bridge=true)
> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET sofia/external/
> 8 at sip0.MY_DOMAIN.com [hangup_after_bridge]=[true]
> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com bridge(sofia/gateway/
> freelycall.com/7906*******)
> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:2128 Parsing
> global variables
> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
> sofia/external/7906******* [c619366c-fd98-11e5-a35c-831849fc41a3]
> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:4776
> (sofia/external/7906*******) State Change CS_NEW -> CS_INIT
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/7906*******) Running State Change CS_INIT
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
> (sofia/external/7906*******) State INIT
> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88
> sofia/external/7906******* SOFIA INIT
> 2016-04-08 10:47:10.503262 [DEBUG] sofia_glue.c:1257
> sofia/external/7906******* sending invite version: 1.6.6  64bit
> Local SDP:
> v=0
> o=FreeSWITCH 1460100428 1460100429 IN IP4 52.*.*.198
> s=FreeSWITCH
> c=IN IP4 52.*.*.198
> t=0 0
> m=audio 26402 RTP/AVP 8 101 13
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
> a=sendrecv
>
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
> sofia/external/7906******* Standard INIT
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
> (sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
> (sofia/external/7906*******) State INIT going to sleep
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/7906*******) Running State Change CS_ROUTING
> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel
> sofia/external/7906******* entering state [calling][0]
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/7906*******) State ROUTING
> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141
> sofia/external/7906******* SOFIA ROUTING
> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:67
> (sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
> (sofia/external/7906*******) State ROUTING going to sleep
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
> (sofia/external/7906*******) State CONSUME_MEDIA
> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
> (sofia/external/7906*******) State CONSUME_MEDIA going to sleep
> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel
> sofia/external/7906******* entering state [proceeding][183]
> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6770 Remote SDP:
> v=0
> o=root 153112258 153112258 IN IP4 178.*.*.12
> s=Asterisk PBX 11.11.0
> c=IN IP4 178.*.*.12
> t=0 0
> m=audio 17362 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
> telephone-event payload to 101 at 8000
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
> sofia/external/7906******* PCMA/8000 20 ms 160 samples 64000 bits 1 channels
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111
> sofia/external/7906******* Original read codec set to PCMA:8
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
> telephone-event payload to 101 at 8000
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
> sofia/external/7906******* Set 2833 dtmf send payload to 101 recv payload
> to 101
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
> [sofia/external/7906*******] 172.31.22.124 port 26402 -> 178.*.*.12 port
> 17362 codec: 8 ms: 20
> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer [soft]
> 160 bytes per 20ms
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
> sofia/external/7906******* Set 2833 dtmf send payload to 101
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
> sofia/external/7906******* Set 2833 dtmf receive payload to 101
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
> sofia/external/7906******* Set rtp dtmf delay to 40
> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
> sofia/external/7906*******!
> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468
> (sofia/external/7906*******) Callstate Change DOWN -> EARLY
> 2016-04-08 10:47:17.443282 [INFO] switch_ivr_originate.c:3557 Sending
> early media
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
> telephone-event payload to 101 at 8000
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
> sofia/external/8 at sip0.MY_DOMAIN.com PCMA/8000 20 ms 160 samples 64000
> bits 1 channels
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111 sofia/external/
> 8 at sip0.MY_DOMAIN.com Original read codec set to PCMA:8
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
> telephone-event payload to 101 at 8000
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485 sofia/external/
> 8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101 recv payload to 101
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
> [sofia/external/8 at sip0.MY_DOMAIN.com] 172.31.22.124 port 30630 ->
> 52.*.*.177 port 40082 codec: 8 ms: 20
> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer [soft]
> 160 bytes per 20ms
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332 sofia/external/
> 8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339 sofia/external/
> 8 at sip0.MY_DOMAIN.com Set 2833 dtmf receive payload to 101
> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362 sofia/external/
> 8 at sip0.MY_DOMAIN.com Set rtp dtmf delay to 40
> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
> sofia/external/8 at sip0.MY_DOMAIN.com!
> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468 (sofia/external/
> 8 at sip0.MY_DOMAIN.com) Callstate Change RINGING -> EARLY
> 2016-04-08 10:47:17.443282 [DEBUG] mod_sofia.c:2330 Ring SDP:
> v=0
> o=FreeSWITCH 1460096207 1460096208 IN IP4 52.*.*.198
> s=FreeSWITCH
> c=IN IP4 52.*.*.198
> t=0 0
> m=audio 30630 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel sofia/external/
> 8 at sip0.MY_DOMAIN.com entering state [early][183]
> 2016-04-08 10:47:17.443282 [DEBUG] switch_ivr_originate.c:3608 Originate
> Resulted in Success: [sofia/external/7906*******]
> 2016-04-08 10:47:17.463254 [DEBUG] switch_ivr_bridge.c:1591
> (sofia/external/7906*******) State Change CS_CONSUME_MEDIA ->
> CS_EXCHANGE_MEDIA
> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/7906*******) Running State Change CS_EXCHANGE_MEDIA
> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:542
> (sofia/external/7906*******) State EXCHANGE_MEDIA
> 2016-04-08 10:47:17.463254 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
> 2016-04-08 10:47:17.503264 [DEBUG] switch_rtp.c:6654 Correct audio ip/port
> confirmed.
> 2016-04-08 10:47:17.663261 [DEBUG] switch_rtp.c:6654 Correct audio ip/port
> confirmed.
> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
> sofia/external/7906******* entering state [completing][200]
> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6767 Duplicate SDP
> v=0
> o=root 153112258 153112258 IN IP4 178.*.*.12
> s=Asterisk PBX 11.11.0
> c=IN IP4 178.*.*.12
> t=0 0
> m=audio 17362 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
> sofia/external/7906******* entering state [ready][200]
> 2016-04-08 10:47:21.323279 [NOTICE] sofia.c:7665 Channel
> [sofia/external/7906*******] has been answered
> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767
> (sofia/external/7906*******) Callstate Change EARLY -> ACTIVE
> 2016-04-08 10:47:21.323279 [DEBUG] mod_sofia.c:799 Local SDP
> sofia/external/8 at sip0.MY_DOMAIN.com:
> v=0
> o=FreeSWITCH 1460096207 1460096209 IN IP4 52.*.*.198
> s=FreeSWITCH
> c=IN IP4 52.*.*.198
> t=0 0
> m=audio 30630 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel sofia/external/
> 8 at sip0.MY_DOMAIN.com entering state [completed][200]
> 2016-04-08 10:47:21.323279 [NOTICE] switch_ivr_bridge.c:616 Channel
> [sofia/external/8 at sip0.MY_DOMAIN.com] has been answered
> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767 (sofia/external/
> 8 at sip0.MY_DOMAIN.com) Callstate Change EARLY -> ACTIVE
> 2016-04-08 10:47:21.383279 [DEBUG] switch_rtp.c:6654 Correct audio ip/port
> confirmed.
> 2016-04-08 10:47:21.423259 [DEBUG] switch_rtp.c:6654 Correct audio ip/port
> confirmed.
>
>
> *And after 30 seconds:*
> 2016-04-08 10:47:53.343283 [DEBUG] sofia.c:6760 Channel sofia/external/
> 8 at sip0.MY_DOMAIN.com entering state [terminating][0]
> 2016-04-08 10:47:53.343283 [NOTICE] sofia.c:7779 Hangup sofia/external/
> 8 at sip0.MY_DOMAIN.com [CS_EXECUTE] [NORMAL_UNSPECIFIED]
> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:699 sofia/external/
> 8 at sip0.MY_DOMAIN.com ending bridge by request from write function
> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD
> DONE [sofia/external/7906*******]
> 2016-04-08 10:47:53.343283 [NOTICE] switch_ivr_bridge.c:881 Hangup
> sofia/external/7906******* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:542
> (sofia/external/7906*******) State EXCHANGE_MEDIA going to sleep
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/7906*******) Running State Change CS_HANGUP
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
> (sofia/external/7906*******) Callstate Change ACTIVE -> HANGUP
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
> (sofia/external/7906*******) State HANGUP
> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel
> sofia/external/7906******* hanging up, cause: NORMAL_CLEARING
> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:484 Sending BYE to
> sofia/external/7906*******
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
> sofia/external/7906******* Standard HANGUP, cause: NORMAL_CLEARING
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
> (sofia/external/7906*******) State HANGUP going to sleep
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
> (sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/7906*******) Running State Change CS_REPORTING
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
> (sofia/external/7906*******) State REPORTING
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
> sofia/external/7906******* Standard REPORTING, cause: NORMAL_CLEARING
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
> (sofia/external/7906*******) State REPORTING going to sleep
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 2
> (sofia/external/7906*******) Locked, Waiting on external entities
> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:705 sofia/external/
> 8 at sip0.MY_DOMAIN.com ending bridge by request from read function
> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD
> DONE [sofia/external/8 at sip0.MY_DOMAIN.com]
> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:1692 sofia/external/
> 8 at sip0.MY_DOMAIN.com skip receive message [UNBRIDGE] (channel is hungup
> already)
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:2796
> sofia/external/8 at sip0.MY_DOMAIN.com skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:539
> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_HANGUP
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change ACTIVE -> HANGUP
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP
> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel sofia/external/
> 8 at sip0.MY_DOMAIN.com hanging up, cause: NORMAL_UNSPECIFIED
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
> sofia/external/8 at sip0.MY_DOMAIN.com Standard HANGUP, cause:
> NORMAL_UNSPECIFIED
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP going to sleep
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_HANGUP ->
> CS_REPORTING
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_REPORTING
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
> sofia/external/8 at sip0.MY_DOMAIN.com Standard REPORTING, cause:
> NORMAL_UNSPECIFIED
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING going to sleep
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_REPORTING ->
> CS_DESTROY
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 1
> (sofia/external/8 at sip0.MY_DOMAIN.com) Locked, Waiting on external entities
> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session 1
> (sofia/external/8 at sip0.MY_DOMAIN.com) Ended
> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
> Channel sofia/external/8 at sip0.MY_DOMAIN.com [CS_DESTROY]
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_DESTROY
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY
> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341 sofia/external/
> 8 at sip0.MY_DOMAIN.com SOFIA DESTROY
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
> sofia/external/8 at sip0.MY_DOMAIN.com Standard DESTROY
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY going to sleep
> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session 2
> (sofia/external/7906*******) Ended
> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
> Channel sofia/external/7906******* [CS_DESTROY]
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
> (sofia/external/7906*******) Running State Change CS_DESTROY
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
> (sofia/external/7906*******) State DESTROY
> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341
> sofia/external/7906******* SOFIA DESTROY
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
> sofia/external/7906******* Standard DESTROY
> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
> (sofia/external/7906*******) State DESTROY going to sleep
>
>
> 2016-04-08 17:37 GMT+03:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:
>
>> Hi,
>>
>> I would recommend you to capture SIP packets  during call  on Freeswitch
>> server and send it here, I will take a look on it.
>>
>> With kind regards,
>>
>> Jurijs
>>
>> On Fri, Apr 8, 2016 at 5:34 PM, ???? ??????? <stasan89 at gmail.com> wrote:
>>
>>> I already tried disabling timers, does not work.
>>>
>>> 2016-04-08 17:19 GMT+03:00 Oleg Stolyar <olegstolyar at gmail.com>:
>>>
>>>> Try disabling session timers in the sip profile.  I think that line is
>>>> commented out by default, so uncomment it.
>>>>
>>>> <param name="enable-timer" value="false"/>
>>>>
>>>> On Fri, Apr 8, 2016 at 6:59 AM, ???? ??????? <stasan89 at gmail.com>
>>>> wrote:
>>>>
>>>>> Hello.
>>>>>
>>>>> When using a call or conference through sip ? freeswitch with external
>>>>> provider there is a problem ? the call is interrupted in 30 seconds. Though
>>>>> the sound goes all right.
>>>>> I think that it caused by the NAT settings for freeswitch, but I don't
>>>>> understand how to adjust it correctly.
>>>>> At start of freeswitch I see the following mistakes in the tracking
>>>>> data:
>>>>> 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
>>>>> 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>> 1/5
>>>>> 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>> 2/5
>>>>> 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>> 3/5
>>>>> 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>> 4/5
>>>>> 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>> 5/5
>>>>> 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for
>>>>> PMP [general error]
>>>>> 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP
>>>>> 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT
>>>>> devices detected!
>>>>>
>>>>> Despite of this mistake, conference communication between two internal
>>>>> users works normally. The problem arises at a call through external
>>>>> provider.
>>>>>
>>>>> We have the following architecture:
>>>>> In a cloud of Amazon EC2 there are 2 servers ? opensips and
>>>>> freeswitch, both for NAT for external clients, but have an opportunity to
>>>>> work with each other directly.
>>>>> opensips has the internal address 172.31.0.169 and external 52. *.*.177
>>>>> freeswitch has the internal address 172.31.22.124 and external 52.
>>>>> *.*.198
>>>>>
>>>>> In fact, freeswitch acts only for conferences, and is ready for use of
>>>>> a remote DB on opensips.
>>>>> The auto-nat settings by default didn't work. The problem is in the
>>>>> external profile settings as far as I understand.
>>>>>
>>>>> I have filled and created the following configuration:
>>>>> vars.xml
>>>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto?/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198?/> <!?
>>>>> public freeswitch ip ?>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198?/> <!?
>>>>> public freeswitch ip ?>
>>>>>   <!-- External SIP Profile -->
>>>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
>>>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>>>   <X-PRE-PROCESS cmd="set"
>>>>> data="external_ssl_dir=$${base_dir}/conf/tls"/>
>>>>>
>>>>> sip_profile/external.xml
>>>>>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>>>>>     <param name="sip-ip" value="$${local_ip_v4}"/>
>>>>>
>>>>>     <param name="ext-rtp-ip" value=?52.*.*.198?/> <!? public
>>>>> freeswitch ip ?>
>>>>>     <param name="ext-sip-ip" value=?52.*.*.198?/> <!? public
>>>>> freeswitch ip ?>
>>>>> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
>>>>> ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
>>>>> (that would be logical), but in that case conferences didn't work at all
>>>>> and errors below appeared:
>>>>> [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
>>>>> Also I tried to put such configuration:
>>>>>     <param name="rtp-ip" value="auto"/>
>>>>>     <param name="sip-ip" value="52.*.*.198?/>
>>>>> but it also hasn't helped to solve the problem.
>>>>>
>>>>> autoload_configs/switch.conf.xml
>>>>>     <param name="rtp-start-port" value="16384"/>
>>>>>     <param name="rtp-end-port" value="32768"/>
>>>>>
>>>>> "sofia status" looks as follows:
>>>>>                      Name       Type
>>>>>                                       Data    State
>>>>>
>>>>> =================================================================================================
>>>>>             172.31.22.124      alias
>>>>>                                   internal    ALIASED
>>>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5060
>>>>> RUNNING (0)
>>>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5061
>>>>> RUNNING (0) (TLS)
>>>>>  external::*********.com    gateway                      sip:USER@*********.com
>>>>> REGED
>>>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5080
>>>>> RUNNING (0)
>>>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5081
>>>>> RUNNING (0) (TLS)
>>>>>
>>>>> =================================================================================================
>>>>> 2 profiles 1 alias
>>>>>
>>>>> "sofia status profile external" looks as follows:
>>>>>
>>>>> =================================================================================================
>>>>> 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               172.31.22.124
>>>>> Ext-RTP-IP           52.*.*.198
>>>>> SIP-IP               172.31.22.124
>>>>> Ext-SIP-IP           52.*.*.198
>>>>> URL                  sip:mod_sofia at 52.*.*.198:5060
>>>>> BIND-URL             sip:mod_sofia at 52.
>>>>> *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
>>>>> TLS-URL              sip:mod_sofia at 52.*.*.198:5061
>>>>> TLS-BIND-URL         sips:mod_sofia at 52.
>>>>> *.*.198:5061;maddr=172.31.22.124;transport=tls
>>>>> HOLD-MUSIC           local_stream://moh
>>>>> OUTBOUND-PROXY       N/A
>>>>> CODECS IN            PCMA
>>>>> CODECS OUT           PCMA
>>>>> 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
>>>>>
>>>>>
>>>>>
>>>>> What do I adjust wrong? Whether there is some opportunity, to tell
>>>>> freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr  8 19:29:05 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 Apr 2016 11:29:05 -0400
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
	NEGOTIATION ERROR. SDP
In-Reply-To: <a0e8cc45ffae4f42bb7be231bb03a163@RESDRCMBX002.Resources.STELECT.LOCAL>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
	<38419347-D033-43AB-92DA-91D7D4C90465@jerris.com>
	<5c023f40f79d430c85dc5d902b33ebee@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAHtxdDf7KAQHEi+ennJs3Wnfd-pnuVjpzJ9Gn2Bert2AY-wzfA@mail.gmail.com>
	<302e815fb7c642d094445f4465256b92@RESDRCMBX002.Resources.STELECT.LOCAL>
	<CAHtxdDfHgBza0eiWqqxGCi_PH-Y8cGxGdGJFDRLhrnxC3rbWAA@mail.gmail.com>
	<a0e8cc45ffae4f42bb7be231bb03a163@RESDRCMBX002.Resources.STELECT.LOCAL>
Message-ID: <C4A0FD03-EF2C-476F-B011-2950CA655689@jerris.com>

Seriously, use verto... it will be much cleaner... also, the issue you are likely having has to do with codec negotiation settings, but we can't say for sure without seeing a debug log.

> On Apr 8, 2016, at 4:16 AM, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
> 
> Good. Can you share your scenario ?
>  
> Chrome (sipML5) ->FS (1.6.5-64bit  Media mode) ->Chrome (sipML5).
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
> Sent: Friday, April 08, 2016 3:17 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> All call with media transcoding enabled. In WebRTC case OPUS <-> G711a
> 
>  
> On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> what is setting of inbound-bypass-media and inbound-proxy-media in your case?
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Sergey Safarov
> Sent: Friday, April 08, 2016 12:00 PM
> 
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> One week ago I has configured master with sipML5.
> You can try reproduce.
> 
>  
> On Fri, Apr 8, 2016, 06:52 Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi Michael;
>  
> Same complaint at mod_sofia.c 2299.
> Mod_sofia.c: 2299 (sofia_media_tech_media(tech_pvt, r_sdp) != SWITCH_STATUS_SUCCESS).
> Switch_core_media.c :3573: switch_core_media_negotiation_sdp().
>  
> Eval ${webrtc_enable_dtls}, Eval ${media_webrtc}, Eval ${rtp_use_dtls}, Eval ${rtp_secure_media}, all return true.
>  
> Does anyone successfully set up this WebRTC demo (excluding video) using media mode as described by cookbook.  
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 10:17 PM
> 
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> media_webrtc=true
>  
>  
> On Apr 7, 2016, at 3:05 AM, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
>  
> Hi Michael;
>  
> Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as WebRTC signaling.
>  
> You can see SDP from chrome (+sipjs) for this in previous attachments.
>  
> If set inbound-bypass-media=true, then chrome caller can talk with chrome callee using opus without any issue.
>  
>  
> Regards
> Smile.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 2:47 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?
>  
> On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
>  
> I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 1:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> you keep saying 
> 
> (UDP/TLS/RTP/SAVPF)
>  
> are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.
> 
> On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi;
>  
> Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved? 
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Quan Huo Sheng
> Sent: Thursday, April 07, 2016 10:41 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.
>  
> Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.
>  
> As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.
>  
> In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).
>  
> Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.
>  
>  
> Regards
> Smile.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Wednesday, April 06, 2016 11:59 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> yes it's supported
> 
> On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi Itola;
>  
> Sorry, same error.
>  
> Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?
>  
> Finding when FS runs in media mode:
> codec causes caller side ?488 not acceptable here| incompatible destination ?
>                 callee side: ?cancel |user not registered?
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of ?talo Rossi
> Sent: Tuesday, April 05, 2016 8:58 PM
> To: FreeSWITCH Users Help
> Cc: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> Set media_mix_inbound_outbound_codecs=true
> 
> ?talo Rossi
> italo at freeswitch.org <mailto:italo at freeswitch.org>
> IRC chat.freenode.net <http://chat.freenode.net/> #freeswitch #freeswitch-dev
> Bugs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Docs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6 <https://hipchat.freeswitch.org/gUdAgy0m6>
>  
>  
> 
> On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi All;
>  
> I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min <http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min <http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.
> If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
> But if disabling media bypass mode, call is rejected by FS.
>  
> FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.
> Chrome uses opus 111, FS uses opus 116.
> Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().
>  
> Help is needed to troubleshoot this issue.
>  
> Thanks advance.
> Smile.
>  
> 
> [This e-mail is confidential and may be privileged. If you are not the
> intended recipient, please kindly notify us immediately and delete the message
> from your system; please do not copy or use it for any purpose, nor disclose
> its contents to any other person. Thank you.]
> ---ST Electronics Group---
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
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>  
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
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>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
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> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
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> 
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> 
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From regis.freeswitch.org at tornad.net  Fri Apr  8 20:13:01 2016
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Fri, 8 Apr 2016 18:13:01 +0200
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
 configuring NAT on Amazon EC2
In-Reply-To: <CAOrA2UbxHQ+9TJeX0gCErFMD1uRCmH9bDu76JQo7Rt_mBfqOkw@mail.gmail.com>
References: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>
	<CAC8JMOKX7wkSG_140qWMedyz56E5aHyFxG0jb5+87G9iG9qVzQ@mail.gmail.com>
	<CAC89h=L3a3ZzOvYYpq4Jma8JsN6+WmvUGkEmF+7mO4WvyNoGTg@mail.gmail.com>
	<CAOrA2UbnqXs49O18a3eeb-8YRvvxmOS7fPEORtyHEd57O_Q94w@mail.gmail.com>
	<CAC89h=KCcOx5ftG4etYsrV1unEp=BuB-hMBcHQrOzJ2OJvRBog@mail.gmail.com>
	<CAOrA2UbxHQ+9TJeX0gCErFMD1uRCmH9bDu76JQo7Rt_mBfqOkw@mail.gmail.com>
Message-ID: <CAHWtr1TdBN2J61eFGjnPw6i633qwLRYd7Ti7mVs4uTWHAz4NxQ@mail.gmail.com>

99% of the time 30 seconds hangup (or 32seconds) means a NAT problem... or
a response not come back to source...

2016-04-08 17:06 GMT+02:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:

> Hi,
>
> This is not what I need, please use ngrep:
>
>
> http://jonathanmanning.com/2009/11/17/how-to-sip-capture-using-ngrep-debug-sip-packets/
>
> With kind regards,
>
> Jurijs
>
> On Fri, Apr 8, 2016 at 6:02 PM, ???? ??????? <stasan89 at gmail.com> wrote:
>
>> Yes, of cause. I hide some ip and real phone numbers.
>> 178.*.*.12 - ip of provider.
>>
>> *On start call:*
>> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
>> sofia/external/8 at sip0.MY_DOMAIN.com
>> [c618eafe-fd98-11e5-a353-831849fc41a3]
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_NEW
>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:9248 sofia/external/
>> 8 at sip0.MY_DOMAIN.com receiving invite from 172.31.0.169:5060 version:
>> 1.6.6  64bit
>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel sofia/external/
>> 8 at sip0.MY_DOMAIN.com entering state [received][100]
>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6770 Remote SDP:
>> v=0
>> o=- 1460126829 1 IN IP4 85.*.*.4
>> s=portsip.com
>> c=IN IP4 52.*.*.177
>> t=0 0
>> m=audio 40082 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=nortpproxy:yes
>>
>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:7125 (sofia/external/
>> 8 at sip0.MY_DOMAIN.com) State Change CS_NEW -> CS_INIT
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:492
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State NEW
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_INIT
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT
>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88 sofia/external/
>> 8 at sip0.MY_DOMAIN.com SOFIA INIT
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
>> sofia/external/8 at sip0.MY_DOMAIN.com Standard INIT
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_INIT -> CS_ROUTING
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT going to sleep
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_channel.c:2247 (sofia/external/
>> 8 at sip0.MY_DOMAIN.com) Callstate Change DOWN -> RINGING
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
>> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
>> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
>> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
>> <8>->7906******* in context public
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>> [public->from_opensips] continue=false
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
>> [from_opensips] network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/
>> break=on-false
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>> transfer(${destination_number} XML default)
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
>> CS_EXECUTE
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
>> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
>> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com transfer(7906******* XML
>> default)
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr.c:2085 (sofia/external/
>> 8 at sip0.MY_DOMAIN.com) State Change CS_EXECUTE -> CS_ROUTING
>> 2016-04-08 10:47:10.503262 [NOTICE] switch_ivr.c:2092 Transfer
>> sofia/external/8 at sip0.MY_DOMAIN.com to XML[7906*******@default]
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
>> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
>> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
>> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
>> <8>->7906******* in context default
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing [default->unloop]
>> continue=false
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS) [unloop]
>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (FAIL) [unloop]
>> ${sip_looped_call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>> [default->tod_example] continue=true
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Date/Time Match (PASS)
>> [tod_example] break=on-false
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action set(open=true)
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>> [default->outbound_calls_to_freelycall] continue=false
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
>> [outbound_calls_to_freelycall] destination_number(7906*******) =~ /^(.+)/
>> break=on-true
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>> set(hangup_after_bridge=true)
>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>> bridge(sofia/gateway/freelycall.com/7906*******)
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
>> CS_EXECUTE
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
>> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
>> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com set(open=true)
>> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET sofia/external/
>> 8 at sip0.MY_DOMAIN.com [open]=[true]
>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com set(hangup_after_bridge=true)
>> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET sofia/external/
>> 8 at sip0.MY_DOMAIN.com [hangup_after_bridge]=[true]
>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com bridge(sofia/gateway/
>> freelycall.com/7906*******)
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:2128 Parsing
>> global variables
>> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
>> sofia/external/7906******* [c619366c-fd98-11e5-a35c-831849fc41a3]
>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:4776
>> (sofia/external/7906*******) State Change CS_NEW -> CS_INIT
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/7906*******) Running State Change CS_INIT
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>> (sofia/external/7906*******) State INIT
>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88
>> sofia/external/7906******* SOFIA INIT
>> 2016-04-08 10:47:10.503262 [DEBUG] sofia_glue.c:1257
>> sofia/external/7906******* sending invite version: 1.6.6  64bit
>> Local SDP:
>> v=0
>> o=FreeSWITCH 1460100428 1460100429 IN IP4 52.*.*.198
>> s=FreeSWITCH
>> c=IN IP4 52.*.*.198
>> t=0 0
>> m=audio 26402 RTP/AVP 8 101 13
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=rtpmap:13 CN/8000
>> a=ptime:20
>> a=sendrecv
>>
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
>> sofia/external/7906******* Standard INIT
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
>> (sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>> (sofia/external/7906*******) State INIT going to sleep
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/7906*******) Running State Change CS_ROUTING
>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel
>> sofia/external/7906******* entering state [calling][0]
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/7906*******) State ROUTING
>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141
>> sofia/external/7906******* SOFIA ROUTING
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:67
>> (sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>> (sofia/external/7906*******) State ROUTING going to sleep
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
>> (sofia/external/7906*******) State CONSUME_MEDIA
>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
>> (sofia/external/7906*******) State CONSUME_MEDIA going to sleep
>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel
>> sofia/external/7906******* entering state [proceeding][183]
>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6770 Remote SDP:
>> v=0
>> o=root 153112258 153112258 IN IP4 178.*.*.12
>> s=Asterisk PBX 11.11.0
>> c=IN IP4 178.*.*.12
>> t=0 0
>> m=audio 17362 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
>> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
>> telephone-event payload to 101 at 8000
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
>> sofia/external/7906******* PCMA/8000 20 ms 160 samples 64000 bits 1 channels
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111
>> sofia/external/7906******* Original read codec set to PCMA:8
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
>> telephone-event payload to 101 at 8000
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
>> sofia/external/7906******* Set 2833 dtmf send payload to 101 recv payload
>> to 101
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
>> [sofia/external/7906*******] 172.31.22.124 port 26402 -> 178.*.*.12 port
>> 17362 codec: 8 ms: 20
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer
>> [soft] 160 bytes per 20ms
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
>> sofia/external/7906******* Set 2833 dtmf send payload to 101
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
>> sofia/external/7906******* Set 2833 dtmf receive payload to 101
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
>> sofia/external/7906******* Set rtp dtmf delay to 40
>> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
>> sofia/external/7906*******!
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468
>> (sofia/external/7906*******) Callstate Change DOWN -> EARLY
>> 2016-04-08 10:47:17.443282 [INFO] switch_ivr_originate.c:3557 Sending
>> early media
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
>> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
>> telephone-event payload to 101 at 8000
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
>> sofia/external/8 at sip0.MY_DOMAIN.com PCMA/8000 20 ms 160 samples 64000
>> bits 1 channels
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111 sofia/external/
>> 8 at sip0.MY_DOMAIN.com Original read codec set to PCMA:8
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
>> telephone-event payload to 101 at 8000
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
>> recv payload to 101
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
>> [sofia/external/8 at sip0.MY_DOMAIN.com] 172.31.22.124 port 30630 ->
>> 52.*.*.177 port 40082 codec: 8 ms: 20
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer
>> [soft] 160 bytes per 20ms
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf receive payload to 101
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
>> sofia/external/8 at sip0.MY_DOMAIN.com Set rtp dtmf delay to 40
>> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
>> sofia/external/8 at sip0.MY_DOMAIN.com!
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468 (sofia/external/
>> 8 at sip0.MY_DOMAIN.com) Callstate Change RINGING -> EARLY
>> 2016-04-08 10:47:17.443282 [DEBUG] mod_sofia.c:2330 Ring SDP:
>> v=0
>> o=FreeSWITCH 1460096207 1460096208 IN IP4 52.*.*.198
>> s=FreeSWITCH
>> c=IN IP4 52.*.*.198
>> t=0 0
>> m=audio 30630 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>>
>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel sofia/external/
>> 8 at sip0.MY_DOMAIN.com entering state [early][183]
>> 2016-04-08 10:47:17.443282 [DEBUG] switch_ivr_originate.c:3608 Originate
>> Resulted in Success: [sofia/external/7906*******]
>> 2016-04-08 10:47:17.463254 [DEBUG] switch_ivr_bridge.c:1591
>> (sofia/external/7906*******) State Change CS_CONSUME_MEDIA ->
>> CS_EXCHANGE_MEDIA
>> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/7906*******) Running State Change CS_EXCHANGE_MEDIA
>> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:542
>> (sofia/external/7906*******) State EXCHANGE_MEDIA
>> 2016-04-08 10:47:17.463254 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
>> 2016-04-08 10:47:17.503264 [DEBUG] switch_rtp.c:6654 Correct audio
>> ip/port confirmed.
>> 2016-04-08 10:47:17.663261 [DEBUG] switch_rtp.c:6654 Correct audio
>> ip/port confirmed.
>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
>> sofia/external/7906******* entering state [completing][200]
>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6767 Duplicate SDP
>> v=0
>> o=root 153112258 153112258 IN IP4 178.*.*.12
>> s=Asterisk PBX 11.11.0
>> c=IN IP4 178.*.*.12
>> t=0 0
>> m=audio 17362 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
>> sofia/external/7906******* entering state [ready][200]
>> 2016-04-08 10:47:21.323279 [NOTICE] sofia.c:7665 Channel
>> [sofia/external/7906*******] has been answered
>> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767
>> (sofia/external/7906*******) Callstate Change EARLY -> ACTIVE
>> 2016-04-08 10:47:21.323279 [DEBUG] mod_sofia.c:799 Local SDP
>> sofia/external/8 at sip0.MY_DOMAIN.com:
>> v=0
>> o=FreeSWITCH 1460096207 1460096209 IN IP4 52.*.*.198
>> s=FreeSWITCH
>> c=IN IP4 52.*.*.198
>> t=0 0
>> m=audio 30630 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>>
>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel sofia/external/
>> 8 at sip0.MY_DOMAIN.com entering state [completed][200]
>> 2016-04-08 10:47:21.323279 [NOTICE] switch_ivr_bridge.c:616 Channel
>> [sofia/external/8 at sip0.MY_DOMAIN.com] has been answered
>> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767 (sofia/external/
>> 8 at sip0.MY_DOMAIN.com) Callstate Change EARLY -> ACTIVE
>> 2016-04-08 10:47:21.383279 [DEBUG] switch_rtp.c:6654 Correct audio
>> ip/port confirmed.
>> 2016-04-08 10:47:21.423259 [DEBUG] switch_rtp.c:6654 Correct audio
>> ip/port confirmed.
>>
>>
>> *And after 30 seconds:*
>> 2016-04-08 10:47:53.343283 [DEBUG] sofia.c:6760 Channel sofia/external/
>> 8 at sip0.MY_DOMAIN.com entering state [terminating][0]
>> 2016-04-08 10:47:53.343283 [NOTICE] sofia.c:7779 Hangup sofia/external/
>> 8 at sip0.MY_DOMAIN.com [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:699 sofia/external/
>> 8 at sip0.MY_DOMAIN.com ending bridge by request from write function
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD
>> DONE [sofia/external/7906*******]
>> 2016-04-08 10:47:53.343283 [NOTICE] switch_ivr_bridge.c:881 Hangup
>> sofia/external/7906******* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:542
>> (sofia/external/7906*******) State EXCHANGE_MEDIA going to sleep
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/7906*******) Running State Change CS_HANGUP
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
>> (sofia/external/7906*******) Callstate Change ACTIVE -> HANGUP
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>> (sofia/external/7906*******) State HANGUP
>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel
>> sofia/external/7906******* hanging up, cause: NORMAL_CLEARING
>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:484 Sending BYE to
>> sofia/external/7906*******
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/7906******* Standard HANGUP, cause: NORMAL_CLEARING
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>> (sofia/external/7906*******) State HANGUP going to sleep
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
>> (sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/7906*******) Running State Change CS_REPORTING
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>> (sofia/external/7906*******) State REPORTING
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
>> sofia/external/7906******* Standard REPORTING, cause: NORMAL_CLEARING
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>> (sofia/external/7906*******) State REPORTING going to sleep
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
>> (sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 2
>> (sofia/external/7906*******) Locked, Waiting on external entities
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:705 sofia/external/
>> 8 at sip0.MY_DOMAIN.com ending bridge by request from read function
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD
>> DONE [sofia/external/8 at sip0.MY_DOMAIN.com]
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:1692
>> sofia/external/8 at sip0.MY_DOMAIN.com skip receive message [UNBRIDGE]
>> (channel is hungup already)
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:2796
>> sofia/external/8 at sip0.MY_DOMAIN.com skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:539
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_HANGUP
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change ACTIVE -> HANGUP
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP
>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel sofia/external/
>> 8 at sip0.MY_DOMAIN.com hanging up, cause: NORMAL_UNSPECIFIED
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
>> sofia/external/8 at sip0.MY_DOMAIN.com Standard HANGUP, cause:
>> NORMAL_UNSPECIFIED
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP going to sleep
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_REPORTING
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
>> sofia/external/8 at sip0.MY_DOMAIN.com Standard REPORTING, cause:
>> NORMAL_UNSPECIFIED
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING going to sleep
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 1
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Locked, Waiting on external
>> entities
>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session 1
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Ended
>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
>> Channel sofia/external/8 at sip0.MY_DOMAIN.com [CS_DESTROY]
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_DESTROY
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY
>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341 sofia/external/
>> 8 at sip0.MY_DOMAIN.com SOFIA DESTROY
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
>> sofia/external/8 at sip0.MY_DOMAIN.com Standard DESTROY
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY going to sleep
>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session 2
>> (sofia/external/7906*******) Ended
>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
>> Channel sofia/external/7906******* [CS_DESTROY]
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
>> (sofia/external/7906*******) Running State Change CS_DESTROY
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>> (sofia/external/7906*******) State DESTROY
>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341
>> sofia/external/7906******* SOFIA DESTROY
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
>> sofia/external/7906******* Standard DESTROY
>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>> (sofia/external/7906*******) State DESTROY going to sleep
>>
>>
>> 2016-04-08 17:37 GMT+03:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:
>>
>>> Hi,
>>>
>>> I would recommend you to capture SIP packets  during call  on Freeswitch
>>> server and send it here, I will take a look on it.
>>>
>>> With kind regards,
>>>
>>> Jurijs
>>>
>>> On Fri, Apr 8, 2016 at 5:34 PM, ???? ??????? <stasan89 at gmail.com> wrote:
>>>
>>>> I already tried disabling timers, does not work.
>>>>
>>>> 2016-04-08 17:19 GMT+03:00 Oleg Stolyar <olegstolyar at gmail.com>:
>>>>
>>>>> Try disabling session timers in the sip profile.  I think that line is
>>>>> commented out by default, so uncomment it.
>>>>>
>>>>> <param name="enable-timer" value="false"/>
>>>>>
>>>>> On Fri, Apr 8, 2016 at 6:59 AM, ???? ??????? <stasan89 at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hello.
>>>>>>
>>>>>> When using a call or conference through sip ? freeswitch with
>>>>>> external provider there is a problem ? the call is interrupted in 30
>>>>>> seconds. Though the sound goes all right.
>>>>>> I think that it caused by the NAT settings for freeswitch, but I
>>>>>> don't understand how to adjust it correctly.
>>>>>> At start of freeswitch I see the following mistakes in the tracking
>>>>>> data:
>>>>>> 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
>>>>>> 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>>> 1/5
>>>>>> 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>>> 2/5
>>>>>> 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>>> 3/5
>>>>>> 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>>> 4/5
>>>>>> 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>>> 5/5
>>>>>> 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for
>>>>>> PMP [general error]
>>>>>> 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP
>>>>>> 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT
>>>>>> devices detected!
>>>>>>
>>>>>> Despite of this mistake, conference communication between two
>>>>>> internal users works normally. The problem arises at a call through
>>>>>> external provider.
>>>>>>
>>>>>> We have the following architecture:
>>>>>> In a cloud of Amazon EC2 there are 2 servers ? opensips and
>>>>>> freeswitch, both for NAT for external clients, but have an opportunity to
>>>>>> work with each other directly.
>>>>>> opensips has the internal address 172.31.0.169 and external 52.
>>>>>> *.*.177
>>>>>> freeswitch has the internal address 172.31.22.124 and external 52.
>>>>>> *.*.198
>>>>>>
>>>>>> In fact, freeswitch acts only for conferences, and is ready for use
>>>>>> of a remote DB on opensips.
>>>>>> The auto-nat settings by default didn't work. The problem is in the
>>>>>> external profile settings as far as I understand.
>>>>>>
>>>>>> I have filled and created the following configuration:
>>>>>> vars.xml
>>>>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto?/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198?/> <!?
>>>>>> public freeswitch ip ?>
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198?/> <!?
>>>>>> public freeswitch ip ?>
>>>>>>   <!-- External SIP Profile -->
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
>>>>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>>>>   <X-PRE-PROCESS cmd="set"
>>>>>> data="external_ssl_dir=$${base_dir}/conf/tls"/>
>>>>>>
>>>>>> sip_profile/external.xml
>>>>>>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>>>>>>     <param name="sip-ip" value="$${local_ip_v4}"/>
>>>>>>
>>>>>>     <param name="ext-rtp-ip" value=?52.*.*.198?/> <!? public
>>>>>> freeswitch ip ?>
>>>>>>     <param name="ext-sip-ip" value=?52.*.*.198?/> <!? public
>>>>>> freeswitch ip ?>
>>>>>> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
>>>>>> ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
>>>>>> (that would be logical), but in that case conferences didn't work at all
>>>>>> and errors below appeared:
>>>>>> [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
>>>>>> Also I tried to put such configuration:
>>>>>>     <param name="rtp-ip" value="auto"/>
>>>>>>     <param name="sip-ip" value="52.*.*.198?/>
>>>>>> but it also hasn't helped to solve the problem.
>>>>>>
>>>>>> autoload_configs/switch.conf.xml
>>>>>>     <param name="rtp-start-port" value="16384"/>
>>>>>>     <param name="rtp-end-port" value="32768"/>
>>>>>>
>>>>>> "sofia status" looks as follows:
>>>>>>                      Name       Type
>>>>>>                                       Data    State
>>>>>>
>>>>>> =================================================================================================
>>>>>>             172.31.22.124      alias
>>>>>>                                   internal    ALIASED
>>>>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5060
>>>>>> RUNNING (0)
>>>>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5061
>>>>>> RUNNING (0) (TLS)
>>>>>>  external::*********.com    gateway                      sip:USER@*********.com
>>>>>> REGED
>>>>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5080
>>>>>> RUNNING (0)
>>>>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5081
>>>>>> RUNNING (0) (TLS)
>>>>>>
>>>>>> =================================================================================================
>>>>>> 2 profiles 1 alias
>>>>>>
>>>>>> "sofia status profile external" looks as follows:
>>>>>>
>>>>>> =================================================================================================
>>>>>> 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               172.31.22.124
>>>>>> Ext-RTP-IP           52.*.*.198
>>>>>> SIP-IP               172.31.22.124
>>>>>> Ext-SIP-IP           52.*.*.198
>>>>>> URL                  sip:mod_sofia at 52.*.*.198:5060
>>>>>> BIND-URL             sip:mod_sofia at 52.
>>>>>> *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
>>>>>> TLS-URL              sip:mod_sofia at 52.*.*.198:5061
>>>>>> TLS-BIND-URL         sips:mod_sofia at 52.
>>>>>> *.*.198:5061;maddr=172.31.22.124;transport=tls
>>>>>> HOLD-MUSIC           local_stream://moh
>>>>>> OUTBOUND-PROXY       N/A
>>>>>> CODECS IN            PCMA
>>>>>> CODECS OUT           PCMA
>>>>>> 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
>>>>>>
>>>>>>
>>>>>>
>>>>>> What do I adjust wrong? Whether there is some opportunity, to tell
>>>>>> freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Apr  8 21:01:13 2016
From: findmeinwland at gmail.com (Artur Mega)
Date: Fri, 8 Apr 2016 22:01:13 +0500
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
 configuring NAT on Amazon EC2
In-Reply-To: <CAHWtr1TdBN2J61eFGjnPw6i633qwLRYd7Ti7mVs4uTWHAz4NxQ@mail.gmail.com>
References: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>
	<CAC8JMOKX7wkSG_140qWMedyz56E5aHyFxG0jb5+87G9iG9qVzQ@mail.gmail.com>
	<CAC89h=L3a3ZzOvYYpq4Jma8JsN6+WmvUGkEmF+7mO4WvyNoGTg@mail.gmail.com>
	<CAOrA2UbnqXs49O18a3eeb-8YRvvxmOS7fPEORtyHEd57O_Q94w@mail.gmail.com>
	<CAC89h=KCcOx5ftG4etYsrV1unEp=BuB-hMBcHQrOzJ2OJvRBog@mail.gmail.com>
	<CAOrA2UbxHQ+9TJeX0gCErFMD1uRCmH9bDu76JQo7Rt_mBfqOkw@mail.gmail.com>
	<CAHWtr1TdBN2J61eFGjnPw6i633qwLRYd7Ti7mVs4uTWHAz4NxQ@mail.gmail.com>
Message-ID: <CAOLEQ7fZ6FJH9v5wiKv87MFKihc+gCXSbc2-CUnQhw5xDXogJg@mail.gmail.com>

?Do you use sip-proxy? If you use sip proxy, maybe you forget to add
"Record-Route" header? to sip-query

2016-04-08 21:13 GMT+05:00 Regis M <regis.freeswitch.org at tornad.net>:

> 99% of the time 30 seconds hangup (or 32seconds) means a NAT problem... or
> a response not come back to source...
>
> 2016-04-08 17:06 GMT+02:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:
>
>> Hi,
>>
>> This is not what I need, please use ngrep:
>>
>>
>> http://jonathanmanning.com/2009/11/17/how-to-sip-capture-using-ngrep-debug-sip-packets/
>>
>> With kind regards,
>>
>> Jurijs
>>
>> On Fri, Apr 8, 2016 at 6:02 PM, ???? ??????? <stasan89 at gmail.com> wrote:
>>
>>> Yes, of cause. I hide some ip and real phone numbers.
>>> 178.*.*.12 - ip of provider.
>>>
>>> *On start call:*
>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
>>> sofia/external/8 at sip0.MY_DOMAIN.com
>>> [c618eafe-fd98-11e5-a353-831849fc41a3]
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_NEW
>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:9248 sofia/external/
>>> 8 at sip0.MY_DOMAIN.com receiving invite from 172.31.0.169:5060 version:
>>> 1.6.6  64bit
>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel sofia/external/
>>> 8 at sip0.MY_DOMAIN.com entering state [received][100]
>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6770 Remote SDP:
>>> v=0
>>> o=- 1460126829 1 IN IP4 85.*.*.4
>>> s=portsip.com
>>> c=IN IP4 52.*.*.177
>>> t=0 0
>>> m=audio 40082 RTP/AVP 8 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=nortpproxy:yes
>>>
>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:7125 (sofia/external/
>>> 8 at sip0.MY_DOMAIN.com) State Change CS_NEW -> CS_INIT
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:492
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State NEW
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_INIT
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT
>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88 sofia/external/
>>> 8 at sip0.MY_DOMAIN.com SOFIA INIT
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard INIT
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_INIT -> CS_ROUTING
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT going to sleep
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_channel.c:2247 (sofia/external/
>>> 8 at sip0.MY_DOMAIN.com) Callstate Change DOWN -> RINGING
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
>>> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
>>> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
>>> <8>->7906******* in context public
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>> [public->from_opensips] continue=false
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
>>> [from_opensips] network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/
>>> break=on-false
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>> transfer(${destination_number} XML default)
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
>>> CS_EXECUTE
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
>>> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com transfer(7906******* XML
>>> default)
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr.c:2085 (sofia/external/
>>> 8 at sip0.MY_DOMAIN.com) State Change CS_EXECUTE -> CS_ROUTING
>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_ivr.c:2092 Transfer
>>> sofia/external/8 at sip0.MY_DOMAIN.com to XML[7906*******@default]
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
>>> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
>>> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
>>> <8>->7906******* in context default
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing [default->unloop]
>>> continue=false
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS) [unloop]
>>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (FAIL) [unloop]
>>> ${sip_looped_call}() =~ /^true$/ break=on-false
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>> [default->tod_example] continue=true
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Date/Time Match (PASS)
>>> [tod_example] break=on-false
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action set(open=true)
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>> [default->outbound_calls_to_freelycall] continue=false
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
>>> [outbound_calls_to_freelycall] destination_number(7906*******) =~ /^(.+)/
>>> break=on-true
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>> set(hangup_after_bridge=true)
>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>> bridge(sofia/gateway/freelycall.com/7906*******)
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
>>> CS_EXECUTE
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
>>> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com set(open=true)
>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET sofia/external/
>>> 8 at sip0.MY_DOMAIN.com [open]=[true]
>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com
>>> set(hangup_after_bridge=true)
>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET sofia/external/
>>> 8 at sip0.MY_DOMAIN.com [hangup_after_bridge]=[true]
>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com bridge(sofia/gateway/
>>> freelycall.com/7906*******)
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:2128 Parsing
>>> global variables
>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
>>> sofia/external/7906******* [c619366c-fd98-11e5-a35c-831849fc41a3]
>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:4776
>>> (sofia/external/7906*******) State Change CS_NEW -> CS_INIT
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/7906*******) Running State Change CS_INIT
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>> (sofia/external/7906*******) State INIT
>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88
>>> sofia/external/7906******* SOFIA INIT
>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia_glue.c:1257
>>> sofia/external/7906******* sending invite version: 1.6.6  64bit
>>> Local SDP:
>>> v=0
>>> o=FreeSWITCH 1460100428 1460100429 IN IP4 52.*.*.198
>>> s=FreeSWITCH
>>> c=IN IP4 52.*.*.198
>>> t=0 0
>>> m=audio 26402 RTP/AVP 8 101 13
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=rtpmap:13 CN/8000
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
>>> sofia/external/7906******* Standard INIT
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
>>> (sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>> (sofia/external/7906*******) State INIT going to sleep
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/7906*******) Running State Change CS_ROUTING
>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel
>>> sofia/external/7906******* entering state [calling][0]
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>> (sofia/external/7906*******) State ROUTING
>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141
>>> sofia/external/7906******* SOFIA ROUTING
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:67
>>> (sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>> (sofia/external/7906*******) State ROUTING going to sleep
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
>>> (sofia/external/7906*******) State CONSUME_MEDIA
>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
>>> (sofia/external/7906*******) State CONSUME_MEDIA going to sleep
>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel
>>> sofia/external/7906******* entering state [proceeding][183]
>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6770 Remote SDP:
>>> v=0
>>> o=root 153112258 153112258 IN IP4 178.*.*.12
>>> s=Asterisk PBX 11.11.0
>>> c=IN IP4 178.*.*.12
>>> t=0 0
>>> m=audio 17362 RTP/AVP 8 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
>>> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
>>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
>>> telephone-event payload to 101 at 8000
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
>>> sofia/external/7906******* PCMA/8000 20 ms 160 samples 64000 bits 1 channels
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111
>>> sofia/external/7906******* Original read codec set to PCMA:8
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
>>> telephone-event payload to 101 at 8000
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
>>> sofia/external/7906******* Set 2833 dtmf send payload to 101 recv payload
>>> to 101
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
>>> [sofia/external/7906*******] 172.31.22.124 port 26402 -> 178.*.*.12 port
>>> 17362 codec: 8 ms: 20
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer
>>> [soft] 160 bytes per 20ms
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
>>> sofia/external/7906******* Set 2833 dtmf send payload to 101
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
>>> sofia/external/7906******* Set 2833 dtmf receive payload to 101
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
>>> sofia/external/7906******* Set rtp dtmf delay to 40
>>> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
>>> sofia/external/7906*******!
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468
>>> (sofia/external/7906*******) Callstate Change DOWN -> EARLY
>>> 2016-04-08 10:47:17.443282 [INFO] switch_ivr_originate.c:3557 Sending
>>> early media
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
>>> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
>>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
>>> telephone-event payload to 101 at 8000
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
>>> sofia/external/8 at sip0.MY_DOMAIN.com PCMA/8000 20 ms 160 samples 64000
>>> bits 1 channels
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111
>>> sofia/external/8 at sip0.MY_DOMAIN.com Original read codec set to PCMA:8
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
>>> telephone-event payload to 101 at 8000
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
>>> recv payload to 101
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
>>> [sofia/external/8 at sip0.MY_DOMAIN.com] 172.31.22.124 port 30630 ->
>>> 52.*.*.177 port 40082 codec: 8 ms: 20
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer
>>> [soft] 160 bytes per 20ms
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf receive payload to 101
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
>>> sofia/external/8 at sip0.MY_DOMAIN.com Set rtp dtmf delay to 40
>>> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
>>> sofia/external/8 at sip0.MY_DOMAIN.com!
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468 (sofia/external/
>>> 8 at sip0.MY_DOMAIN.com) Callstate Change RINGING -> EARLY
>>> 2016-04-08 10:47:17.443282 [DEBUG] mod_sofia.c:2330 Ring SDP:
>>> v=0
>>> o=FreeSWITCH 1460096207 1460096208 IN IP4 52.*.*.198
>>> s=FreeSWITCH
>>> c=IN IP4 52.*.*.198
>>> t=0 0
>>> m=audio 30630 RTP/AVP 8 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel sofia/external/
>>> 8 at sip0.MY_DOMAIN.com entering state [early][183]
>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_ivr_originate.c:3608 Originate
>>> Resulted in Success: [sofia/external/7906*******]
>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_ivr_bridge.c:1591
>>> (sofia/external/7906*******) State Change CS_CONSUME_MEDIA ->
>>> CS_EXCHANGE_MEDIA
>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/7906*******) Running State Change CS_EXCHANGE_MEDIA
>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:542
>>> (sofia/external/7906*******) State EXCHANGE_MEDIA
>>> 2016-04-08 10:47:17.463254 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
>>> 2016-04-08 10:47:17.503264 [DEBUG] switch_rtp.c:6654 Correct audio
>>> ip/port confirmed.
>>> 2016-04-08 10:47:17.663261 [DEBUG] switch_rtp.c:6654 Correct audio
>>> ip/port confirmed.
>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
>>> sofia/external/7906******* entering state [completing][200]
>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6767 Duplicate SDP
>>> v=0
>>> o=root 153112258 153112258 IN IP4 178.*.*.12
>>> s=Asterisk PBX 11.11.0
>>> c=IN IP4 178.*.*.12
>>> t=0 0
>>> m=audio 17362 RTP/AVP 8 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
>>> sofia/external/7906******* entering state [ready][200]
>>> 2016-04-08 10:47:21.323279 [NOTICE] sofia.c:7665 Channel
>>> [sofia/external/7906*******] has been answered
>>> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767
>>> (sofia/external/7906*******) Callstate Change EARLY -> ACTIVE
>>> 2016-04-08 10:47:21.323279 [DEBUG] mod_sofia.c:799 Local SDP
>>> sofia/external/8 at sip0.MY_DOMAIN.com:
>>> v=0
>>> o=FreeSWITCH 1460096207 1460096209 IN IP4 52.*.*.198
>>> s=FreeSWITCH
>>> c=IN IP4 52.*.*.198
>>> t=0 0
>>> m=audio 30630 RTP/AVP 8 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel sofia/external/
>>> 8 at sip0.MY_DOMAIN.com entering state [completed][200]
>>> 2016-04-08 10:47:21.323279 [NOTICE] switch_ivr_bridge.c:616 Channel
>>> [sofia/external/8 at sip0.MY_DOMAIN.com] has been answered
>>> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767 (sofia/external/
>>> 8 at sip0.MY_DOMAIN.com) Callstate Change EARLY -> ACTIVE
>>> 2016-04-08 10:47:21.383279 [DEBUG] switch_rtp.c:6654 Correct audio
>>> ip/port confirmed.
>>> 2016-04-08 10:47:21.423259 [DEBUG] switch_rtp.c:6654 Correct audio
>>> ip/port confirmed.
>>>
>>>
>>> *And after 30 seconds:*
>>> 2016-04-08 10:47:53.343283 [DEBUG] sofia.c:6760 Channel sofia/external/
>>> 8 at sip0.MY_DOMAIN.com entering state [terminating][0]
>>> 2016-04-08 10:47:53.343283 [NOTICE] sofia.c:7779 Hangup sofia/external/
>>> 8 at sip0.MY_DOMAIN.com [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:699
>>> sofia/external/8 at sip0.MY_DOMAIN.com ending bridge by request from write
>>> function
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD
>>> DONE [sofia/external/7906*******]
>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_ivr_bridge.c:881 Hangup
>>> sofia/external/7906******* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:542
>>> (sofia/external/7906*******) State EXCHANGE_MEDIA going to sleep
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/7906*******) Running State Change CS_HANGUP
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
>>> (sofia/external/7906*******) Callstate Change ACTIVE -> HANGUP
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>> (sofia/external/7906*******) State HANGUP
>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel
>>> sofia/external/7906******* hanging up, cause: NORMAL_CLEARING
>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:484 Sending BYE to
>>> sofia/external/7906*******
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
>>> sofia/external/7906******* Standard HANGUP, cause: NORMAL_CLEARING
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>> (sofia/external/7906*******) State HANGUP going to sleep
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
>>> (sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/7906*******) Running State Change CS_REPORTING
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>> (sofia/external/7906*******) State REPORTING
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
>>> sofia/external/7906******* Standard REPORTING, cause: NORMAL_CLEARING
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>> (sofia/external/7906*******) State REPORTING going to sleep
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
>>> (sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 2
>>> (sofia/external/7906*******) Locked, Waiting on external entities
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:705
>>> sofia/external/8 at sip0.MY_DOMAIN.com ending bridge by request from read
>>> function
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE THREAD
>>> DONE [sofia/external/8 at sip0.MY_DOMAIN.com]
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:1692
>>> sofia/external/8 at sip0.MY_DOMAIN.com skip receive message [UNBRIDGE]
>>> (channel is hungup already)
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:2796
>>> sofia/external/8 at sip0.MY_DOMAIN.com skip receive message
>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:539
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_HANGUP
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change ACTIVE -> HANGUP
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP
>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel
>>> sofia/external/8 at sip0.MY_DOMAIN.com hanging up, cause:
>>> NORMAL_UNSPECIFIED
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard HANGUP, cause:
>>> NORMAL_UNSPECIFIED
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP going to sleep
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_HANGUP ->
>>> CS_REPORTING
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_REPORTING
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard REPORTING, cause:
>>> NORMAL_UNSPECIFIED
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING going to sleep
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_REPORTING ->
>>> CS_DESTROY
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 1
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Locked, Waiting on external
>>> entities
>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session 1
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Ended
>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
>>> Channel sofia/external/8 at sip0.MY_DOMAIN.com [CS_DESTROY]
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_DESTROY
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY
>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341 sofia/external/
>>> 8 at sip0.MY_DOMAIN.com SOFIA DESTROY
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard DESTROY
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY going to sleep
>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session 2
>>> (sofia/external/7906*******) Ended
>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
>>> Channel sofia/external/7906******* [CS_DESTROY]
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
>>> (sofia/external/7906*******) Running State Change CS_DESTROY
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>> (sofia/external/7906*******) State DESTROY
>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341
>>> sofia/external/7906******* SOFIA DESTROY
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
>>> sofia/external/7906******* Standard DESTROY
>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>> (sofia/external/7906*******) State DESTROY going to sleep
>>>
>>>
>>> 2016-04-08 17:37 GMT+03:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:
>>>
>>>> Hi,
>>>>
>>>> I would recommend you to capture SIP packets  during call  on
>>>> Freeswitch server and send it here, I will take a look on it.
>>>>
>>>> With kind regards,
>>>>
>>>> Jurijs
>>>>
>>>> On Fri, Apr 8, 2016 at 5:34 PM, ???? ??????? <stasan89 at gmail.com>
>>>> wrote:
>>>>
>>>>> I already tried disabling timers, does not work.
>>>>>
>>>>> 2016-04-08 17:19 GMT+03:00 Oleg Stolyar <olegstolyar at gmail.com>:
>>>>>
>>>>>> Try disabling session timers in the sip profile.  I think that line
>>>>>> is commented out by default, so uncomment it.
>>>>>>
>>>>>> <param name="enable-timer" value="false"/>
>>>>>>
>>>>>> On Fri, Apr 8, 2016 at 6:59 AM, ???? ??????? <stasan89 at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Hello.
>>>>>>>
>>>>>>> When using a call or conference through sip ? freeswitch with
>>>>>>> external provider there is a problem ? the call is interrupted in 30
>>>>>>> seconds. Though the sound goes all right.
>>>>>>> I think that it caused by the NAT settings for freeswitch, but I
>>>>>>> don't understand how to adjust it correctly.
>>>>>>> At start of freeswitch I see the following mistakes in the tracking
>>>>>>> data:
>>>>>>> 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
>>>>>>> 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>>>> 1/5
>>>>>>> 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>>>> 2/5
>>>>>>> 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>>>> 3/5
>>>>>>> 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>>>> 4/5
>>>>>>> 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP
>>>>>>> 5/5
>>>>>>> 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for
>>>>>>> PMP [general error]
>>>>>>> 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP
>>>>>>> 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP
>>>>>>> NAT devices detected!
>>>>>>>
>>>>>>> Despite of this mistake, conference communication between two
>>>>>>> internal users works normally. The problem arises at a call through
>>>>>>> external provider.
>>>>>>>
>>>>>>> We have the following architecture:
>>>>>>> In a cloud of Amazon EC2 there are 2 servers ? opensips and
>>>>>>> freeswitch, both for NAT for external clients, but have an opportunity to
>>>>>>> work with each other directly.
>>>>>>> opensips has the internal address 172.31.0.169 and external 52.
>>>>>>> *.*.177
>>>>>>> freeswitch has the internal address 172.31.22.124 and external 52.
>>>>>>> *.*.198
>>>>>>>
>>>>>>> In fact, freeswitch acts only for conferences, and is ready for use
>>>>>>> of a remote DB on opensips.
>>>>>>> The auto-nat settings by default didn't work. The problem is in the
>>>>>>> external profile settings as far as I understand.
>>>>>>>
>>>>>>> I have filled and created the following configuration:
>>>>>>> vars.xml
>>>>>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto?/>
>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198?/> <!?
>>>>>>> public freeswitch ip ?>
>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198?/> <!?
>>>>>>> public freeswitch ip ?>
>>>>>>>   <!-- External SIP Profile -->
>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>>>>>   <X-PRE-PROCESS cmd="set"
>>>>>>> data="external_ssl_dir=$${base_dir}/conf/tls"/>
>>>>>>>
>>>>>>> sip_profile/external.xml
>>>>>>>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>>>>>>>     <param name="sip-ip" value="$${local_ip_v4}"/>
>>>>>>>
>>>>>>>     <param name="ext-rtp-ip" value=?52.*.*.198?/> <!? public
>>>>>>> freeswitch ip ?>
>>>>>>>     <param name="ext-sip-ip" value=?52.*.*.198?/> <!? public
>>>>>>> freeswitch ip ?>
>>>>>>> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
>>>>>>> ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
>>>>>>> (that would be logical), but in that case conferences didn't work at all
>>>>>>> and errors below appeared:
>>>>>>> [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
>>>>>>> Also I tried to put such configuration:
>>>>>>>     <param name="rtp-ip" value="auto"/>
>>>>>>>     <param name="sip-ip" value="52.*.*.198?/>
>>>>>>> but it also hasn't helped to solve the problem.
>>>>>>>
>>>>>>> autoload_configs/switch.conf.xml
>>>>>>>     <param name="rtp-start-port" value="16384"/>
>>>>>>>     <param name="rtp-end-port" value="32768"/>
>>>>>>>
>>>>>>> "sofia status" looks as follows:
>>>>>>>                      Name       Type
>>>>>>>                                       Data    State
>>>>>>>
>>>>>>> =================================================================================================
>>>>>>>             172.31.22.124      alias
>>>>>>>                                   internal    ALIASED
>>>>>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5060
>>>>>>> RUNNING (0)
>>>>>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5061
>>>>>>> RUNNING (0) (TLS)
>>>>>>>  external::*********.com    gateway                      sip:USER@*********.com
>>>>>>> REGED
>>>>>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5080
>>>>>>> RUNNING (0)
>>>>>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5081
>>>>>>> RUNNING (0) (TLS)
>>>>>>>
>>>>>>> =================================================================================================
>>>>>>> 2 profiles 1 alias
>>>>>>>
>>>>>>> "sofia status profile external" looks as follows:
>>>>>>>
>>>>>>> =================================================================================================
>>>>>>> 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               172.31.22.124
>>>>>>> Ext-RTP-IP           52.*.*.198
>>>>>>> SIP-IP               172.31.22.124
>>>>>>> Ext-SIP-IP           52.*.*.198
>>>>>>> URL                  sip:mod_sofia at 52.*.*.198:5060
>>>>>>> BIND-URL             sip:mod_sofia at 52.
>>>>>>> *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
>>>>>>> TLS-URL              sip:mod_sofia at 52.*.*.198:5061
>>>>>>> TLS-BIND-URL         sips:mod_sofia at 52.
>>>>>>> *.*.198:5061;maddr=172.31.22.124;transport=tls
>>>>>>> HOLD-MUSIC           local_stream://moh
>>>>>>> OUTBOUND-PROXY       N/A
>>>>>>> CODECS IN            PCMA
>>>>>>> CODECS OUT           PCMA
>>>>>>> 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> What do I adjust wrong? Whether there is some opportunity, to tell
>>>>>>> freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

Arthur
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From sausageside at yahoo.com  Fri Apr  8 21:49:41 2016
From: sausageside at yahoo.com (Sausage Side)
Date: Fri, 8 Apr 2016 17:49:41 +0000 (UTC)
Subject: [Freeswitch-users] Lua 5.2 Install Issue
References: <634007670.1840370.1460137781421.JavaMail.yahoo.ref@mail.yahoo.com>
Message-ID: <634007670.1840370.1460137781421.JavaMail.yahoo@mail.yahoo.com>

I have installed FS 1.6.7 master onto a new Linode Centos 7.1 VM. This comes with Lua 5.1 preinstalled, and I cannot uninstall it, it would seem, as it reports it would break things (like yum). My problem is that when I make, it fails with this :
make[3]: Entering directory `/usr/local/src/freeswitch-1.6/src/mod/languages/mod_lua'
? CXX????? mod_lua_la-mod_lua.lo
mod_lua.cpp:37:17: fatal error: lua.h: No such file or directory
?#include "lua.h"

Not surprising considering configure reports this :
"checking for lua5.2... checking for lua-5.2... checking for lua5.1... checking for lua-5.1... checking for lua... checking for alMidiGainSOFT in -lopenal... no"

If I do "yum install lua-devel" it all compiles but with version 5.1, not 5.2, obviously as Centos is installing the 5.1 headers as that is its current repo version. 

So, my question is how do I make FS pick up the bundled 5.2 during the "configure" step, that I believe is has in "/usr/local/src/freeswitch/src/mod/languages/mod_lua/lua" and not fail unless the -devel package for the 5.1 version is installed via yum ?
FreeSWITCH Version 1.6.7+git~20160401T134007Z~f0c3870be3~64bit (git f0c3870 2016-04-01 13:40:07Z 64bit)
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From mike at jerris.com  Fri Apr  8 22:21:45 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 8 Apr 2016 14:21:45 -0400
Subject: [Freeswitch-users] Lua 5.2 Install Issue
In-Reply-To: <634007670.1840370.1460137781421.JavaMail.yahoo@mail.yahoo.com>
References: <634007670.1840370.1460137781421.JavaMail.yahoo.ref@mail.yahoo.com>
	<634007670.1840370.1460137781421.JavaMail.yahoo@mail.yahoo.com>
Message-ID: <F91223B9-D415-4514-8E46-C38323F05487@jerris.com>

It does not use the one in src/mod/languages/mod_lua/lua.  Its only using system lua now as of 1.6 release.

It's using pkg-config to find it.. searching in order for the following pkg-config package names:

lua5.2 lua-5.2 lua5.1 lua-5.1 lua

Whichever one it finds first of those via pkg-config is what it tries to use.  If you manually or otherwise install lua5.2, just make sure that the .pc file is somewhere in PKG_CONFIG_PATH, and has a name in that list before the system one it is finding, it will then insert the CFLAGS and LDFLAGS in order to find the headers and libs in the paths from that pc file before the system one.

Also, if libdir for that lua libraries dir is not already in ld.so.conf you may need to add there and ldconfig.


> On Apr 8, 2016, at 1:49 PM, Sausage Side <sausageside at yahoo.com> wrote:
> 
> I have installed FS 1.6.7 master onto a new Linode Centos 7.1 VM. This comes with Lua 5.1 preinstalled, and I cannot uninstall it, it would seem, as it reports it would break things (like yum). My problem is that when I make, it fails with this :
> 
> make[3]: Entering directory `/usr/local/src/freeswitch-1.6/src/mod/languages/mod_lua'
>   CXX      mod_lua_la-mod_lua.lo
> mod_lua.cpp:37:17: fatal error: lua.h: No such file or directory
>  #include "lua.h"
> 
> Not surprising considering configure reports this :
> 
> "checking for lua5.2... checking for lua-5.2... checking for lua5.1... checking for lua-5.1... checking for lua... checking for alMidiGainSOFT in -lopenal... no"
> 
> 
> If I do "yum install lua-devel" it all compiles but with version 5.1, not 5.2, obviously as Centos is installing the 5.1 headers as that is its current repo version. 
> 
> So, my question is how do I make FS pick up the bundled 5.2 during the "configure" step, that I believe is has in "/usr/local/src/freeswitch/src/mod/languages/mod_lua/lua" and not fail unless the -devel package for the 5.1 version is installed via yum ?
> 
> FreeSWITCH Version 1.6.7+git~20160401T134007Z~f0c3870be3~64bit (git f0c3870 2016-04-01 13:40:07Z 64bit)
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From fabiomargarido at gmail.com  Fri Apr  8 22:26:16 2016
From: fabiomargarido at gmail.com (Fabio Margarido)
Date: Fri, 08 Apr 2016 18:26:16 +0000
Subject: [Freeswitch-users] mod-vpx 1.6.7 Debian package missing
Message-ID: <CAGib9DRUDD96At26HuoZXQUJienEP7AnqW2e-8C__rrXZUfdzg@mail.gmail.com>

I've noticed there is no freeswitch-mod-vpx Debian package for version
1.6.7. When I upgraded my box the package was uninstalled. Does this mean
the module's functionality has been moved to the core?

Thanks.
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From gmaruzz at gmail.com  Fri Apr  8 22:31:07 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 8 Apr 2016 20:31:07 +0200
Subject: [Freeswitch-users] mod-vpx 1.6.7 Debian package missing
In-Reply-To: <CAGib9DRUDD96At26HuoZXQUJienEP7AnqW2e-8C__rrXZUfdzg@mail.gmail.com>
References: <CAGib9DRUDD96At26HuoZXQUJienEP7AnqW2e-8C__rrXZUfdzg@mail.gmail.com>
Message-ID: <CALXCt0qgqSeuj+P0WwWDmUt1Mwq8SVgphFMJq9bq-KbcUfkWgQ@mail.gmail.com>

vpx is now a core codec, is embedded into FreeSWITCH itself, no more need
for a separate module.

On Fri, Apr 8, 2016 at 8:26 PM, Fabio Margarido <fabiomargarido at gmail.com>
wrote:

> I've noticed there is no freeswitch-mod-vpx Debian package for version
> 1.6.7. When I upgraded my box the package was uninstalled. Does this mean
> the module's functionality has been moved to the core?
>
> 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From danny.gershman at gmail.com  Fri Apr  8 22:36:06 2016
From: danny.gershman at gmail.com (Danny Gershman)
Date: Fri, 08 Apr 2016 18:36:06 +0000
Subject: [Freeswitch-users] Controlling `presence_id`
Message-ID: <CAEorZymMLMhu=CK5+Kk2u4udr8a6MgWxVcoWDS9BZbiLZUv1MQ@mail.gmail.com>

Is there a way to control the domain on the presence_id?  When I run `verto
status` it's showing up as the internal alias.  I want to be able to
specify this as a different value (the external_sip_ip for example).  We
use this value by querying the database presence_id value in the channels
table.

                     Name	   Type	
 Data	State
=================================================================================================
            external-ipv6	profile	
sip:mod_sofia@[::1]:5080	RUNNING (0)
            external-ipv6	profile	
sip:mod_sofia@[::1]:5081	RUNNING (0) (TLS)
              10.200.82.5	  alias	
internal	ALIASED
                 external	profile	
sip:mod_sofia at 10.0.82.91:5080	RUNNING (0)
                 external	profile	
sip:mod_sofia at 10.0.82.91:5081	RUNNING (0) (TLS)
            internal-ipv6	profile	
sip:mod_sofia@[::1]:5060	RUNNING (0)
            internal-ipv6	profile	
sip:mod_sofia@[::1]:5061	RUNNING (0) (TLS)
                 internal	profile	
sip:mod_sofia at 10.0.82.91:5060	RUNNING (0)
                 internal	profile	
sip:mod_sofia at 10.0.82.91:5061	RUNNING (0) (TLS)
=================================================================================================
4 profiles 1 alias


Here is the output of `verto status`

                     Name	   Type	
 Data	State
=================================================================================================
               default-v6	profile	
ws:[::1]:8081	RUNNING
               default-v6	profile	
wss:[::1]:8082	RUNNING
               default-v4	profile	
ws:0.0.0.0:8081	RUNNING
               default-v4	profile	
wss:0.0.0.0:8082	RUNNING
default-v4::1rVWHg2fbA5Ydx9hAceSdKGREwY=@10.200.82.5	client	
          192.168.202.156:50064	CONN_REG (WSS)
=================================================================================================
2 profiles , 1 client

Thanks,

Danny
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From brian.beebe at vivint.com  Fri Apr  8 23:39:39 2016
From: brian.beebe at vivint.com (Brian Beebe)
Date: Fri, 8 Apr 2016 19:39:39 +0000
Subject: [Freeswitch-users] Freeswitch malloc errors
Message-ID: <72cc256d96fd4de3a31710c9c0891eca@MAILDB04.APEX.Local>

For the past day or so, freeswitch has been crashing with a malloc error.  We have been running with this configuration for over a month without errors.  Now we are seeing these errors. We have 4 servers and all of them are experiencing this.   Has anyone seen this before?

We are running 1.7.0.599 on Centos 7.2.1511.

Apr  8 13:21:35 freeswitch-pl1a freeswitch: *** Error in `/usr/bin/freeswitch': malloc(): smallbin double linked list corrupted: 0x00007f90d01818c0 ***
Apr  8 12:43:30 freeswitch-pl2b freeswitch: *** Error in `/usr/bin/freeswitch': corrupted double-linked list: 0x00007f4eb80d84a0 ***
Apr  8 12:39:23 freeswitch-pl2a freeswitch: *** Error in `/usr/bin/freeswitch': malloc(): smallbin double linked list corrupted: 0x0000000004363fa0 ***
Apr  8 11:38:38 freeswitch-pl1a freeswitch: *** Error in `/usr/bin/freeswitch': malloc(): smallbin double linked list corrupted: 0x00007fab65103130 ***

Thanks,

Brian
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From babak.freeswitch at gmail.com  Sat Apr  9 08:46:52 2016
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Sat, 9 Apr 2016 09:16:52 +0430
Subject: [Freeswitch-users] detecting tone
In-Reply-To: <CAO3obxbxS5Xcaj1rK7_QeyCmPBgwx_vNmOTUo41JFqgMZtmw5g@mail.gmail.com>
References: <CAEVfcaaBPgJujRJSer8rbmg8HRHp-TGiePm_D97YfG=uhAb+Jw@mail.gmail.com>
	<CAHrOMBkXyk78uWaFtuTAVkggmBbb3HqrPchRG89wW5R4s3s54w@mail.gmail.com>
	<CAO3obxbxS5Xcaj1rK7_QeyCmPBgwx_vNmOTUo41JFqgMZtmw5g@mail.gmail.com>
Message-ID: <CAEVfcaZofTcOGeD-dChGeuvOQs5zSkRnC8XQhpWfuvjyzzMusA@mail.gmail.com>

thanks
I think the my test scenario is not correct. I'm playing the sound file to
a test call and it seems spandsp is not monitoring the played file rtp.
when I play the file using laptop mic it detects it with descriptor 49. the
original problem is false detections during normal calls. I will test your
suggestions when I can record a conversation in which false detection
happened.

On Thu, Apr 7, 2016 at 4:34 PM, Piotr Gregor <piotrek.gregor at gmail.com>
wrote:

> Hi Babak,
>
> you can also try avmd module for a detection of single frequency sound.
> I have tested avmd on your audio and it detected 425.69 Hz frequency
> when it's variance threshold was set to 0.0006, 446.23 Hz with default
> threshold of 0.00025.
>
>
> 2016-04-07 12:32:05.820909 [NOTICE] switch_channel.c:1104 New Channel
> sofia/internal/1000 at 192.168.1.27 [4e41e5be-51a0-42fa-87b5-8fde76744530]
> 2016-04-07 12:32:06.060908 [INFO] mod_dialplan_xml.c:637 Processing
> beautiful display name <1000>->1705 in context default
>
>     api avmd 4e41e5be-51a0-42fa-87b5-8fde76744530 start
>
>     Content-Type: api/response
>     Content-Length: 89
>
>     +OK
>      [4e41e5be-51a0-42fa-87b5-8fde76744530] [sofia/internal/
> 1000 at 192.168.1.27] started!
>
> 2016-04-07 12:32:12.760908 [INFO] switch_core_media_bug.c:701 Sending
> early media
> 2016-04-07 12:32:12.760908 [NOTICE] sofia_media.c:92 Pre-Answer
> sofia/internal/1000 at 192.168.1.27!
> 2016-04-07 12:32:12.760908 [INFO] mod_avmd.c:279 Avmd session initialized,
> [8000] samples/s
> 2016-04-07 12:32:12.760908 [INFO] mod_avmd.c:739 Avmd on channel
> [sofia/internal/1000 at 192.168.1.27] started!
> 2016-04-07 12:32:17.480907 [NOTICE] mod_avmd.c:919 <<< AVMD - Beep
> Detected: f = [425.688780], variance = [0.000533] >>>
>
>
> Avmd works for single frequency tones. Be sure to checkout latest master
> (this module
> is yet not of a production quality at this moment).
>
> cheers,
> Piotr
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jelena at misticnabica.hr  Sat Apr  9 08:57:15 2016
From: jelena at misticnabica.hr (jelena at misticnabica.hr)
Date: Sat, 9 Apr 2016 04:57:15 GMT
Subject: [Freeswitch-users] detecting tone
Message-ID: <6941A25E420D495AA1EE512083BDE9A1.MAI@server2.totohost.hr>



From jungleboogie0 at gmail.com  Sat Apr  9 09:28:32 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Fri, 8 Apr 2016 22:28:32 -0700
Subject: [Freeswitch-users] Mastering FreeSWITCH Book
In-Reply-To: <CAKbxfG_E1Pz=UcKhKJSQxBWDNXhhUhBek_YSahyCY95MH0zPRg@mail.gmail.com>
References: <CAKbxfG_E1Pz=UcKhKJSQxBWDNXhhUhBek_YSahyCY95MH0zPRg@mail.gmail.com>
Message-ID: <CAKE2PDuJRTu7qLP2OSX6ojxPuEkWO90wB_mS3Y7d_GO53mpKCw@mail.gmail.com>

Hi All,
On 16 September 2014 at 12:07, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
>
> Hi,
>
> We've been asked by packt to produce a 3rd book called "Mastering FreeSWITCH"
> Would anyone be interested in working on the authoring of that book and get their name in print?
>
> We need help with creating the content as well as formatting the text into the word templates they require etc.
>
> Please contact consulting at freeswitch.org if you are interested.

Just curious how the progress on the book is coming along. ;)

William's outline looks great:
https://docs.google.com/document/d/1D39Tm9eTrmBMkR1kDSO89jNpyOcCbRuXARd53B4A90M/edit



-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From gmaruzz at gmail.com  Sat Apr  9 12:16:23 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 9 Apr 2016 10:16:23 +0200
Subject: [Freeswitch-users] [Freeswitch-docs] Mastering FreeSWITCH Book
In-Reply-To: <CAKE2PDuJRTu7qLP2OSX6ojxPuEkWO90wB_mS3Y7d_GO53mpKCw@mail.gmail.com>
References: <CAKbxfG_E1Pz=UcKhKJSQxBWDNXhhUhBek_YSahyCY95MH0zPRg@mail.gmail.com>
	<CAKE2PDuJRTu7qLP2OSX6ojxPuEkWO90wB_mS3Y7d_GO53mpKCw@mail.gmail.com>
Message-ID: <CALXCt0rTcHDN5KpXNoxps8sbCFwb3Exp_+f6OobYojvg=r94mw@mail.gmail.com>

Progressing well, almost there !  :)

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT
Il 09/Apr/2016 07:29, "jungle Boogie" <jungleboogie0 at gmail.com> ha scritto:

> Hi All,
> On 16 September 2014 at 12:07, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> >
> > Hi,
> >
> > We've been asked by packt to produce a 3rd book called "Mastering
> FreeSWITCH"
> > Would anyone be interested in working on the authoring of that book and
> get their name in print?
> >
> > We need help with creating the content as well as formatting the text
> into the word templates they require etc.
> >
> > Please contact consulting at freeswitch.org if you are interested.
>
> Just curious how the progress on the book is coming along. ;)
>
> William's outline looks great:
>
> https://docs.google.com/document/d/1D39Tm9eTrmBMkR1kDSO89jNpyOcCbRuXARd53B4A90M/edit
>
>
>
> --
> -------
> inum: 883510009027723
> sip: jungleboogie at sip2sip.info
> xmpp: jungle-boogie at jit.si
>
> _______________________________________________
> Freeswitch-docs mailing list
> Freeswitch-docs at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs
>
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From amani.mansour2 at gmail.com  Sat Apr  9 16:38:27 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Sat, 9 Apr 2016 13:38:27 +0100
Subject: [Freeswitch-users] Scenario : registering phase
Message-ID: <CAFETSWifMR_Y1LVsRmRDihHFi_--CSMa9bGygTnHWs1Tmbp96g@mail.gmail.com>

Good morning  ,

Can you help me please ?, I need to configure a number X in my soft phone .

I want to do a scenario in the regestering phase which  must be like this :
soft --------------------register--------------------------------> server

soft <-----------------unauthorized (first 401)----------------server

soft ---------------------register (with authorization header computed with
the nonce of the
401)----------------------------------------------------->server

soft <--------------------unauthorized 401 (with stale ==False) ---->server

i did this in dialplan/default.xml:
<extension name="fail register">
    <condition field="caller_id_number" expression="^400$">
        <action application="respond" data="401"/>
        <action application="hangup"/>
    </condition>
</extension>
i have only one unauthorized then i have ok ,but how can i do to have this
result .
thank you
with best regards ,
amani
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From mouli123 at gmail.com  Sat Apr  9 17:57:31 2016
From: mouli123 at gmail.com (Chandramouli P)
Date: Sat, 9 Apr 2016 19:27:31 +0530
Subject: [Freeswitch-users] NATIVE SQL ERR [cannot commit - no
	transaction is active]
In-Reply-To: <CALR0W68a6+iiimqtW6hyxVdKz8o8B1yg-P7tye0LjEmtajvOaw@mail.gmail.com>
References: <CALR0W68a6+iiimqtW6hyxVdKz8o8B1yg-P7tye0LjEmtajvOaw@mail.gmail.com>
Message-ID: <CALR0W6_r=z0en7hYNeZYuqdZCvxN1kbmXZs51CKAx9WFTx6mqQ@mail.gmail.com>

Hi,

Any help would be appreciated.

Thank you,
Chandra.


On Wed, Apr 6, 2016 at 6:40 PM, Chandramouli P <mouli123 at gmail.com> wrote:

> Hi,
>
> Please find my below deployed environment:
>
> Environment: Microsoft Azure
> OS: CentOS 7.0 (64 bit)
> FreeSwitch Version: 1.6.6~64bit ( 64bit)
>
> I installed FreeSwitch through "Yum" and works fine for couple of hours.
> After that I am getting below errors:
>
> [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [attempt to write a readonly
> database]
> BEGIN EXCLUSIVE
> 2016-04-05 13:29:33.368355 [CRIT] switch_core_sqldb.c:1952 ERROR [attempt
> to write a readonly database]
> 2016-04-05 13:29:33.368355 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
> [cannot commit - no transaction is active]
>
> Can anybody tell me what could be the issue and how to solve this?
>
> Thanks in advance,
> Chandra.
>
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From luis.daniel.lucio at gmail.com  Sat Apr  9 18:29:04 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sat, 9 Apr 2016 10:29:04 -0400
Subject: [Freeswitch-users] [Freeswitch-docs] Mastering FreeSWITCH Book
In-Reply-To: <CALXCt0rTcHDN5KpXNoxps8sbCFwb3Exp_+f6OobYojvg=r94mw@mail.gmail.com>
References: <CAKbxfG_E1Pz=UcKhKJSQxBWDNXhhUhBek_YSahyCY95MH0zPRg@mail.gmail.com>
	<CAKE2PDuJRTu7qLP2OSX6ojxPuEkWO90wB_mS3Y7d_GO53mpKCw@mail.gmail.com>
	<CALXCt0rTcHDN5KpXNoxps8sbCFwb3Exp_+f6OobYojvg=r94mw@mail.gmail.com>
Message-ID: <CAFLo2QzypUzQPsH+QS==AiSQCdSELbZ+wmad+9uj_nBgOL+SoA@mail.gmail.com>

Count me in. I would like to write about security. Not only on the involved
protocols, but operating a VoIP system as well
Le 9 avr. 2016 4:17 AM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> a ?crit :

> Progressing well, almost there !  :)
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
> Il 09/Apr/2016 07:29, "jungle Boogie" <jungleboogie0 at gmail.com> ha
> scritto:
>
>> Hi All,
>> On 16 September 2014 at 12:07, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> >
>> > Hi,
>> >
>> > We've been asked by packt to produce a 3rd book called "Mastering
>> FreeSWITCH"
>> > Would anyone be interested in working on the authoring of that book and
>> get their name in print?
>> >
>> > We need help with creating the content as well as formatting the text
>> into the word templates they require etc.
>> >
>> > Please contact consulting at freeswitch.org if you are interested.
>>
>> Just curious how the progress on the book is coming along. ;)
>>
>> William's outline looks great:
>>
>> https://docs.google.com/document/d/1D39Tm9eTrmBMkR1kDSO89jNpyOcCbRuXARd53B4A90M/edit
>>
>>
>>
>> --
>> -------
>> inum: 883510009027723
>> sip: jungleboogie at sip2sip.info
>> xmpp: jungle-boogie at jit.si
>>
>> _______________________________________________
>> Freeswitch-docs mailing list
>> Freeswitch-docs at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Apr  9 18:34:27 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 9 Apr 2016 16:34:27 +0200
Subject: [Freeswitch-users] [Freeswitch-docs] Mastering FreeSWITCH Book
In-Reply-To: <CAFLo2QzypUzQPsH+QS==AiSQCdSELbZ+wmad+9uj_nBgOL+SoA@mail.gmail.com>
References: <CAKbxfG_E1Pz=UcKhKJSQxBWDNXhhUhBek_YSahyCY95MH0zPRg@mail.gmail.com>
	<CAKE2PDuJRTu7qLP2OSX6ojxPuEkWO90wB_mS3Y7d_GO53mpKCw@mail.gmail.com>
	<CALXCt0rTcHDN5KpXNoxps8sbCFwb3Exp_+f6OobYojvg=r94mw@mail.gmail.com>
	<CAFLo2QzypUzQPsH+QS==AiSQCdSELbZ+wmad+9uj_nBgOL+SoA@mail.gmail.com>
Message-ID: <CALXCt0oDo=vechMMT9oS+R2TEjqT55h=U-eThY4KxpFWtLb=cw@mail.gmail.com>

Book is practically closed at this time, too late.

Anyway, if you have something already written, please send it to me, I'll
see if I can merge it.

If not, wait for next announcement, other books are coming this way :))



On Sat, Apr 9, 2016 at 4:29 PM, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> Count me in. I would like to write about security. Not only on the
> involved protocols, but operating a VoIP system as well
> Le 9 avr. 2016 4:17 AM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> a
> ?crit :
>
>> Progressing well, almost there !  :)
>>
>> sent from mobile
>> cell: +39 347 266 56 18
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> Il 09/Apr/2016 07:29, "jungle Boogie" <jungleboogie0 at gmail.com> ha
>> scritto:
>>
>>> Hi All,
>>> On 16 September 2014 at 12:07, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>> >
>>> > Hi,
>>> >
>>> > We've been asked by packt to produce a 3rd book called "Mastering
>>> FreeSWITCH"
>>> > Would anyone be interested in working on the authoring of that book
>>> and get their name in print?
>>> >
>>> > We need help with creating the content as well as formatting the text
>>> into the word templates they require etc.
>>> >
>>> > Please contact consulting at freeswitch.org if you are interested.
>>>
>>> Just curious how the progress on the book is coming along. ;)
>>>
>>> William's outline looks great:
>>>
>>> https://docs.google.com/document/d/1D39Tm9eTrmBMkR1kDSO89jNpyOcCbRuXARd53B4A90M/edit
>>>
>>>
>>>
>>> --
>>> -------
>>> inum: 883510009027723
>>> sip: jungleboogie at sip2sip.info
>>> xmpp: jungle-boogie at jit.si
>>>
>>> _______________________________________________
>>> Freeswitch-docs mailing list
>>> Freeswitch-docs at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From mandra at gmail.com  Sat Apr  9 20:40:50 2016
From: mandra at gmail.com (Chris Mandra)
Date: Sat, 9 Apr 2016 12:40:50 -0400
Subject: [Freeswitch-users] Mastering FreeSWITCH Book
In-Reply-To: <CALXCt0oDo=vechMMT9oS+R2TEjqT55h=U-eThY4KxpFWtLb=cw@mail.gmail.com>
References: <CAKbxfG_E1Pz=UcKhKJSQxBWDNXhhUhBek_YSahyCY95MH0zPRg@mail.gmail.com>
	<CAKE2PDuJRTu7qLP2OSX6ojxPuEkWO90wB_mS3Y7d_GO53mpKCw@mail.gmail.com>
	<CALXCt0rTcHDN5KpXNoxps8sbCFwb3Exp_+f6OobYojvg=r94mw@mail.gmail.com>
	<CAFLo2QzypUzQPsH+QS==AiSQCdSELbZ+wmad+9uj_nBgOL+SoA@mail.gmail.com>
	<CALXCt0oDo=vechMMT9oS+R2TEjqT55h=U-eThY4KxpFWtLb=cw@mail.gmail.com>
Message-ID: <CAJdRzN5mGVSE9JKzXXL2EpBPmJC5RFXszZSWKd5aXnmaiRJdkA@mail.gmail.com>

If you keep a list or anything, I'd like to help out, give back, so keep me
in mind
Chris

On Saturday, April 9, 2016, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> Book is practically closed at this time, too late.
>
> Anyway, if you have something already written, please send it to me, I'll
> see if I can merge it.
>
> If not, wait for next announcement, other books are coming this way :))
>
>
>
> On Sat, Apr 9, 2016 at 4:29 PM, Luis Daniel Lucio Quiroz <
> luis.daniel.lucio at gmail.com
> <javascript:_e(%7B%7D,'cvml','luis.daniel.lucio at gmail.com');>> wrote:
>
>> Count me in. I would like to write about security. Not only on the
>> involved protocols, but operating a VoIP system as well
>> Le 9 avr. 2016 4:17 AM, "Giovanni Maruzzelli" <gmaruzz at gmail.com
>> <javascript:_e(%7B%7D,'cvml','gmaruzz at gmail.com');>> a ?crit :
>>
>>> Progressing well, almost there !  :)
>>>
>>> sent from mobile
>>> cell: +39 347 266 56 18
>>> Giovanni Maruzzelli
>>> OpenTelecom.IT
>>> Il 09/Apr/2016 07:29, "jungle Boogie" <jungleboogie0 at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','jungleboogie0 at gmail.com');>> ha scritto:
>>>
>>>> Hi All,
>>>> On 16 September 2014 at 12:07, Anthony Minessale
>>>> <anthony.minessale at gmail.com
>>>> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>> wrote:
>>>> >
>>>> > Hi,
>>>> >
>>>> > We've been asked by packt to produce a 3rd book called "Mastering
>>>> FreeSWITCH"
>>>> > Would anyone be interested in working on the authoring of that book
>>>> and get their name in print?
>>>> >
>>>> > We need help with creating the content as well as formatting the text
>>>> into the word templates they require etc.
>>>> >
>>>> > Please contact consulting at freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');> if you are
>>>> interested.
>>>>
>>>> Just curious how the progress on the book is coming along. ;)
>>>>
>>>> William's outline looks great:
>>>>
>>>> https://docs.google.com/document/d/1D39Tm9eTrmBMkR1kDSO89jNpyOcCbRuXARd53B4A90M/edit
>>>>
>>>>
>>>>
>>>> --
>>>> -------
>>>> inum: 883510009027723
>>>> sip: jungleboogie at sip2sip.info
>>>> <javascript:_e(%7B%7D,'cvml','jungleboogie at sip2sip.info');>
>>>> xmpp: jungle-boogie at jit.si
>>>> <javascript:_e(%7B%7D,'cvml','jungle-boogie at jit.si');>
>>>>
>>>> _______________________________________________
>>>> Freeswitch-docs mailing list
>>>> Freeswitch-docs at lists.freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','Freeswitch-docs at lists.freeswitch.org');>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>


-- 
mandra
c:410.258.5281
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From gmaruzz at gmail.com  Sat Apr  9 20:46:45 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 9 Apr 2016 18:46:45 +0200
Subject: [Freeswitch-users] Mastering FreeSWITCH Book
In-Reply-To: <CAJdRzN5mGVSE9JKzXXL2EpBPmJC5RFXszZSWKd5aXnmaiRJdkA@mail.gmail.com>
References: <CAKbxfG_E1Pz=UcKhKJSQxBWDNXhhUhBek_YSahyCY95MH0zPRg@mail.gmail.com>
	<CAKE2PDuJRTu7qLP2OSX6ojxPuEkWO90wB_mS3Y7d_GO53mpKCw@mail.gmail.com>
	<CALXCt0rTcHDN5KpXNoxps8sbCFwb3Exp_+f6OobYojvg=r94mw@mail.gmail.com>
	<CAFLo2QzypUzQPsH+QS==AiSQCdSELbZ+wmad+9uj_nBgOL+SoA@mail.gmail.com>
	<CALXCt0oDo=vechMMT9oS+R2TEjqT55h=U-eThY4KxpFWtLb=cw@mail.gmail.com>
	<CAJdRzN5mGVSE9JKzXXL2EpBPmJC5RFXszZSWKd5aXnmaiRJdkA@mail.gmail.com>
Message-ID: <CALXCt0qVBrnJz0+B7Ww1qSKQwN_V8ToS46ita1=GW0zYFrD8jQ@mail.gmail.com>

Definitely!

Thanks a lot, and help would also be precious in Confluence documentation
writing and updating.

Please, if you feel to help and give back, write to the FreeSWITCH
Documentation Mailing Lists signaling your availability, we'll be *very*
glad.

http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs



On Sat, Apr 9, 2016 at 6:40 PM, Chris Mandra <mandra at gmail.com> wrote:

> If you keep a list or anything, I'd like to help out, give back, so keep
> me in mind
> Chris
>
>
> On Saturday, April 9, 2016, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>
>> Book is practically closed at this time, too late.
>>
>> Anyway, if you have something already written, please send it to me, I'll
>> see if I can merge it.
>>
>> If not, wait for next announcement, other books are coming this way :))
>>
>>
>>
>> On Sat, Apr 9, 2016 at 4:29 PM, Luis Daniel Lucio Quiroz <
>> luis.daniel.lucio at gmail.com> wrote:
>>
>>> Count me in. I would like to write about security. Not only on the
>>> involved protocols, but operating a VoIP system as well
>>> Le 9 avr. 2016 4:17 AM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> a
>>> ?crit :
>>>
>>>> Progressing well, almost there !  :)
>>>>
>>>> sent from mobile
>>>> cell: +39 347 266 56 18
>>>> Giovanni Maruzzelli
>>>> OpenTelecom.IT
>>>> Il 09/Apr/2016 07:29, "jungle Boogie" <jungleboogie0 at gmail.com> ha
>>>> scritto:
>>>>
>>>>> Hi All,
>>>>> On 16 September 2014 at 12:07, Anthony Minessale
>>>>> <anthony.minessale at gmail.com> wrote:
>>>>> >
>>>>> > Hi,
>>>>> >
>>>>> > We've been asked by packt to produce a 3rd book called "Mastering
>>>>> FreeSWITCH"
>>>>> > Would anyone be interested in working on the authoring of that book
>>>>> and get their name in print?
>>>>> >
>>>>> > We need help with creating the content as well as formatting the
>>>>> text into the word templates they require etc.
>>>>> >
>>>>> > Please contact consulting at freeswitch.org if you are interested.
>>>>>
>>>>> Just curious how the progress on the book is coming along. ;)
>>>>>
>>>>> William's outline looks great:
>>>>>
>>>>> https://docs.google.com/document/d/1D39Tm9eTrmBMkR1kDSO89jNpyOcCbRuXARd53B4A90M/edit
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> -------
>>>>> inum: 883510009027723
>>>>> sip: jungleboogie at sip2sip.info
>>>>> xmpp: jungle-boogie at jit.si
>>>>>
>>>>> _______________________________________________
>>>>> Freeswitch-docs mailing list
>>>>> Freeswitch-docs at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>
>
> --
> mandra
> c:410.258.5281
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From mandra at gmail.com  Sat Apr  9 21:12:10 2016
From: mandra at gmail.com (Chris Mandra)
Date: Sat, 9 Apr 2016 13:12:10 -0400
Subject: [Freeswitch-users] Mastering FreeSWITCH Book
In-Reply-To: <CALXCt0qVBrnJz0+B7Ww1qSKQwN_V8ToS46ita1=GW0zYFrD8jQ@mail.gmail.com>
References: <CAKbxfG_E1Pz=UcKhKJSQxBWDNXhhUhBek_YSahyCY95MH0zPRg@mail.gmail.com>
	<CAKE2PDuJRTu7qLP2OSX6ojxPuEkWO90wB_mS3Y7d_GO53mpKCw@mail.gmail.com>
	<CALXCt0rTcHDN5KpXNoxps8sbCFwb3Exp_+f6OobYojvg=r94mw@mail.gmail.com>
	<CAFLo2QzypUzQPsH+QS==AiSQCdSELbZ+wmad+9uj_nBgOL+SoA@mail.gmail.com>
	<CALXCt0oDo=vechMMT9oS+R2TEjqT55h=U-eThY4KxpFWtLb=cw@mail.gmail.com>
	<CAJdRzN5mGVSE9JKzXXL2EpBPmJC5RFXszZSWKd5aXnmaiRJdkA@mail.gmail.com>
	<CALXCt0qVBrnJz0+B7Ww1qSKQwN_V8ToS46ita1=GW0zYFrD8jQ@mail.gmail.com>
Message-ID: <CAJdRzN6kbWne3-K=8sETfD1bvyZJ0zfnhTF9Xyaxedfc6kKJ7A@mail.gmail.com>

I've joined that list. :)

On Saturday, April 9, 2016, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> Definitely!
>
> Thanks a lot, and help would also be precious in Confluence documentation
> writing and updating.
>
> Please, if you feel to help and give back, write to the FreeSWITCH
> Documentation Mailing Lists signaling your availability, we'll be *very*
> glad.
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs
>
>
>
> On Sat, Apr 9, 2016 at 6:40 PM, Chris Mandra <mandra at gmail.com
> <javascript:_e(%7B%7D,'cvml','mandra at gmail.com');>> wrote:
>
>> If you keep a list or anything, I'd like to help out, give back, so keep
>> me in mind
>> Chris
>>
>>
>> On Saturday, April 9, 2016, Giovanni Maruzzelli <gmaruzz at gmail.com
>> <javascript:_e(%7B%7D,'cvml','gmaruzz at gmail.com');>> wrote:
>>
>>> Book is practically closed at this time, too late.
>>>
>>> Anyway, if you have something already written, please send it to me,
>>> I'll see if I can merge it.
>>>
>>> If not, wait for next announcement, other books are coming this way :))
>>>
>>>
>>>
>>> On Sat, Apr 9, 2016 at 4:29 PM, Luis Daniel Lucio Quiroz <
>>> luis.daniel.lucio at gmail.com> wrote:
>>>
>>>> Count me in. I would like to write about security. Not only on the
>>>> involved protocols, but operating a VoIP system as well
>>>> Le 9 avr. 2016 4:17 AM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> a
>>>> ?crit :
>>>>
>>>>> Progressing well, almost there !  :)
>>>>>
>>>>> sent from mobile
>>>>> cell: +39 347 266 56 18
>>>>> Giovanni Maruzzelli
>>>>> OpenTelecom.IT
>>>>> Il 09/Apr/2016 07:29, "jungle Boogie" <jungleboogie0 at gmail.com> ha
>>>>> scritto:
>>>>>
>>>>>> Hi All,
>>>>>> On 16 September 2014 at 12:07, Anthony Minessale
>>>>>> <anthony.minessale at gmail.com> wrote:
>>>>>> >
>>>>>> > Hi,
>>>>>> >
>>>>>> > We've been asked by packt to produce a 3rd book called "Mastering
>>>>>> FreeSWITCH"
>>>>>> > Would anyone be interested in working on the authoring of that book
>>>>>> and get their name in print?
>>>>>> >
>>>>>> > We need help with creating the content as well as formatting the
>>>>>> text into the word templates they require etc.
>>>>>> >
>>>>>> > Please contact consulting at freeswitch.org if you are interested.
>>>>>>
>>>>>> Just curious how the progress on the book is coming along. ;)
>>>>>>
>>>>>> William's outline looks great:
>>>>>>
>>>>>> https://docs.google.com/document/d/1D39Tm9eTrmBMkR1kDSO89jNpyOcCbRuXARd53B4A90M/edit
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> -------
>>>>>> inum: 883510009027723
>>>>>> sip: jungleboogie at sip2sip.info
>>>>>> xmpp: jungle-boogie at jit.si
>>>>>>
>>>>>> _______________________________________________
>>>>>> Freeswitch-docs mailing list
>>>>>> Freeswitch-docs at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>
>>
>> --
>> mandra
>> c:410.258.5281
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>


-- 
mandra
c:410.258.5281
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From brian at freeswitch.org  Sun Apr 10 03:58:31 2016
From: brian at freeswitch.org (Brian West)
Date: Sat, 9 Apr 2016 18:58:31 -0500
Subject: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
In-Reply-To: <CACot2ntmk5aUM9kyLXwD3C+1phvib-TLYaszW3cvTm7Texg9VQ@mail.gmail.com>
References: <CACot2ntTB+7OXvomm3sxMzxnNEnT6vHtQb=T=m8MTOFT0=zFZg@mail.gmail.com>
	<E93C0E00-0EEE-44E1-81F2-1E8CF374057B@jerris.com>
	<CACot2ntmtZzcXs4bWqd3oW9i5URXMpqaouu8EXKWeN2thDJG4A@mail.gmail.com>
	<CACot2nvgSJOFj9HJ2JOc4Ofp1Ce_x-mZ0G95epFJA1AwMZ2X5g@mail.gmail.com>
	<A84CA188-7849-4678-98DC-33BA615A9A0C@jerris.com>
	<CACot2ntmk5aUM9kyLXwD3C+1phvib-TLYaszW3cvTm7Texg9VQ@mail.gmail.com>
Message-ID: <CAEJMVkBHuXYqMJ5E=jFv_Onhs=UosXzZ4DjuLzuPcCBC-gapwQ@mail.gmail.com>

I think he's asking what you're attempting to accomplish.

On Wed, Apr 6, 2016 at 1:11 PM, saurabh verrma <saurabhkv01 at gmail.com>
wrote:

> Thanks Michael,
>
> What do you exactly mean by client & server application.?
> Is it some type of SIP server/stack(server application) & SIP
> endpoint(client application) that you're referring ?
>
> Also could you please explain a bit what's the limitation in ICE in case
> of client application.
>
> On Wed, Apr 6, 2016 at 9:52 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> is this for a client or server application?  If it is for a client
>> application, we may not be the right stack for you.  We make a number of
>> assumptions in our ice that is based on it being a server in order to
>> improve server efficiency.
>>
>> On Apr 6, 2016, at 5:01 AM, saurabh verrma <saurabhkv01 at gmail.com> wrote:
>>
>> Hi,
>>
>> I?m working on an application where I?m trying to use FreeSWITCH as a
>> library. My intention is to use FreeSWITCH as a UAS endpoint. Basically it
>> needs to be supporting following:
>> 1. WebRTC
>> 2. Ability to act like UAS endpoint
>> 3. Support for DTLS/SRTP
>> 4. ICE support
>>
>> I?m seeking community suggestion if that?s feasible to implement or not?
>> If yes, what are the possible starting directions we could explore above
>> points.
>>
>> Any help would be greatly appreciated.
>>
>>
>> On Sat, Apr 2, 2016 at 12:24 PM, saurabh verrma <saurabhkv01 at gmail.com>
>> wrote:
>>
>>> Thanks Michael,
>>>
>>> Basically we're writing a PJSIP based application & PJSIP doesn't have
>>> DTLS support. So we're thinking to use FreeSWITCH library for DTLS/SRTP.
>>>
>>> On Fri, Apr 1, 2016 at 7:29 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> we have full support for webrtc media profile which would include these
>>>> features  There are not a ton of people who use freeswitch as a library,
>>>> but it is built that way so that you can control it and host it in another
>>>> application instead of stand alone.  If you are trying to accomplish
>>>> something I'd try to handle it standalone first so you can learn all the
>>>> different ways you might control it before architecting a solution
>>>>
>>>> On Apr 1, 2016, at 9:48 AM, saurabh verrma <saurabhkv01 at gmail.com>
>>>> wrote:
>>>>
>>>> Hi All,
>>>>
>>>> I want to use FreeSWITCH library for DTLS/SRTP support. I want to know
>>>> in freeswitch which library has the support of these features(SRTP/DTLS) ?
>>>> Is there any application available based on freeswitch library ?
>>>>
>>>> Any help would be appreciable.
>>>>
>>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 Warm Regards:*
> *Saurabh Kumar Verma*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
https://www.gofundme.com/freeswitch_ubuntu

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From amani.mansour2 at gmail.com  Sun Apr 10 04:04:44 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Sun, 10 Apr 2016 01:04:44 +0100
Subject: [Freeswitch-users] Configure SDP
In-Reply-To: <3BD6E704-4227-4FB7-A12E-02AB326679F7@jerris.com>
References: <CAFETSWiskTaDA3fFw4qbBo7h7CsXKjiFcY7EXF5CYLk917YQ0Q@mail.gmail.com>
	<3BD6E704-4227-4FB7-A12E-02AB326679F7@jerris.com>
Message-ID: <CAFETSWj7_NHZi7g4SDxAJu+hJcmvY0VkO0a-ABQtG5E2WQQF5A@mail.gmail.com>

Hi mr ,
thank you sir , after enabling 3pcc what i must do , because the INVITE
message is with SDP .
regards
Amani

2016-04-04 15:44 GMT+01:00 Michael Jerris <mike at jerris.com>:

> you are looking for 3pcc
>
> > On Apr 4, 2016, at 4:22 AM, amani mansour <amani.mansour2 at gmail.com>
> wrote:
> >
> > Good morning ,
> >
> > I need to configure a number to send invite without SDP , can anyone
> help me please ?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jelena at misticnabica.hr  Sun Apr 10 04:07:57 2016
From: jelena at misticnabica.hr (jelena at misticnabica.hr)
Date: Sun, 10 Apr 2016 00:07:57 GMT
Subject: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
Message-ID: <4249F84C38D04AFDA4D0D91AFAF5A017.MAI@server2.totohost.hr>



From mandra at gmail.com  Sun Apr 10 06:19:26 2016
From: mandra at gmail.com (Chris Mandra)
Date: Sat, 9 Apr 2016 22:19:26 -0400
Subject: [Freeswitch-users] curl connect timeout question
Message-ID: <CAJdRzN5MsyG8gcHEsbR3udnt4vGV0QWciypgJbAkwzNypsMJ7A@mail.gmail.com>

Hey guys, if I'm using curl in a module and the connection times out do you
have suggestion for how this should be handled?

For example, if I disconnect freeswitch from the internet and have the
following settings:


curl_easy_setopt(curl_handle, CURLOPT_CONNECTTIMEOUT, 5L);
curl_easy_setopt(curl_handle, CURLOPT_TIMEOUT, 5L);
switch_curl_easy_setopt(curl_handle, CURLOPT_NOSIGNAL, 1L);

it never times out and then crashes freeswitch. Is this expected behavior?
Is there something I should be doing instead?
Thanks, chris

--
mandra
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From daveh at beachdognet.com  Sun Apr 10 11:23:08 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Sun, 10 Apr 2016 03:23:08 -0400
Subject: [Freeswitch-users] question on handling of nonce count (nc)
Message-ID: <850AC07B-9D14-4C1A-94BF-19E96A2550E8@beachdognet.com>

In investigating some REGISTER storms on one of my networks, I am seeing some client devices interacting with Freeswitch in a manner that can lead to excessive registration traffic.  
It looks to me to be more of an endpoint problem than a freeswitch problem, but I would like confirmation of that as well as any ideas on how to handle this (i.e., throttle back this traffic).

The basic problem flow is this:

- Client sends a REGISTER with a large nc value and nonce value A
- Freeswitch replies 401 with stale=true (nonce is stale) and nonce value B
- Client sends another REGISTER with nc value incremented by 1 and nonce value A again
- Freeswitch replies 401 with stale=true (nonce is stale) and nonce value C
- Client sends another REGISTER with nc value incremented again and nonce value A again
?.etc.

This seems particularly problematic with some Yealink, Communicator, and Polycomm IP Soundlink endpoint

Here is a specific example (some information redacted)

recv 804 bytes from udp/[]:5060 at 23:54:11.906859:
   ------------------------------------------------------------------------
   REGISTER sip:x.x.x.x:5060 SIP/2.0
   Authorization: Digest username="123371",realm="sip.foo.com",nonce="41adc443-57c8-4325-831e-ffd006a922d4",uri=?sip:x.x.x.x:6060",response="3a4b5f05ec1897a58865b4ba0cdb0b4d",cnonce="b5d06adf6a4c7c0592f5fc1d7766a605",nc=0000008a,qop=auth,algorithm=MD5
   
send 641 bytes to udp/[10.128.77.170]:5060 at 23:54:11.909722:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   WWW-Authenticate: Digest realm=sip.foo.com", nonce="888c8919-b28f-4be4-be12-753430aafa88", stale=true, algorithm=MD5, qop=?auth?


recv 804 bytes from udp/[]:5060 at 23:54:12.007622:
   ------------------------------------------------------------------------
   REGISTER sip:x.x.x.x:5060 SIP/2.0
   Authorization: Digest username="123371",realm="sip.foo.com",nonce="41adc443-57c8-4325-831e-ffd006a922d4",uri=?sip:x.x.x.x:6060",response="556498e38d27c944f10e3a0c11a5ea41",cnonce="5585e516afcf2f95bfbc4bef11a075ee",nc=0000008b,qop=auth,algorithm=MD5

send 641 bytes to udp/[10.128.77.170]:5060 at 23:54:12.010376:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   WWW-Authenticate: Digest realm=?sip.foo.com", nonce="1ff3b9a3-4cbb-4569-b6c7-7bee203547ac", stale=true, algorithm=MD5, qop="auth"

recv 804 bytes from udp/[10.128.77.170]:5060 at 23:54:12.108742:
   ------------------------------------------------------------------------
   REGISTER sip:x.x.x.x:5060 SIP/2.0
   Authorization: Digest username="123371",realm="sip.foo.com",nonce="41adc443-57c8-4325-831e-ffd006a922d4",uri=?sip:x.x.x.x:6060",response="9cf2360ef5f28684e667ac878362d0c0",cnonce="9833d8d3889d3ae8875e0f6f00c4d3f3",nc=0000008c,qop=auth,algorithm=MD5

   

   




From d.mordovin at dwide.com  Sun Apr 10 11:51:28 2016
From: d.mordovin at dwide.com (Dmitry Mordovin)
Date: Sun, 10 Apr 2016 11:51:28 +0400
Subject: [Freeswitch-users] FreeSWITCH in cluster
Message-ID: <570A0600.9060204@dwide.com>

Hi,

I want configure two FreeSWITCH servers as cluster.

As I understand, I need:
  - Configure one core DB (PostgreSQL) for each FreeSWITCH instances.
  - Add two A records in DNS to resolve domain name and load balancing.

*Scheme, 2 FS servers and one DB.*

FS-1 server  192.168.0.1
        Core-DB server  192.168.0.100
FS-2 server  192.168.0.2

*Add two A records in DNS*

mydomain.com.  43200 IN A       192.168.0.1
mydomain.com.  43200 IN A       192.168.0.2


I expect, FreeSWITCH instances will mirror each other in this configuration.

So, if Bob register on 192.168.0.1 server, and Alice register on 
192.168.0.2, they can call each other like local users.
Also, in this case, SIP messages from Bob during the call session will 
sending to any of nodes, depend on DNS resolve.

Am I right?
Is this correct way to build cluster?

Please direct me in right way.

BR, Dmitry
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From mouli123 at gmail.com  Sun Apr 10 12:04:32 2016
From: mouli123 at gmail.com (Chandramouli P)
Date: Sun, 10 Apr 2016 13:34:32 +0530
Subject: [Freeswitch-users] FreeSWITCH in cluster
In-Reply-To: <570A0600.9060204@dwide.com>
References: <570A0600.9060204@dwide.com>
Message-ID: <CALR0W68x8tY=OOP5Xrra2rR-3i+3qdJNi08TY-hsGixtDxjodA@mail.gmail.com>

Hello Dmitry,

I guess the below link with help you:

http://saevolgo.blogspot.in/2012/07/freeswitch-with-sip-users-in-mysql-mod.html

Thanks,
Chandra.




On Sun, Apr 10, 2016 at 1:21 PM, Dmitry Mordovin <d.mordovin at dwide.com>
wrote:

> Hi,
>
> I want configure two FreeSWITCH servers as cluster.
>
> As I understand, I need:
>  - Configure one core DB (PostgreSQL) for each FreeSWITCH instances.
>  - Add two A records in DNS to resolve domain name and load balancing.
>
> *Scheme, 2 FS servers and one DB.*
>
> FS-1 server  192.168.0.1
>        Core-DB server  192.168.0.100
> FS-2 server  192.168.0.2
>
> *Add two A records in DNS*
>
> mydomain.com.  43200 IN A       192.168.0.1
> mydomain.com.  43200 IN A       192.168.0.2
>
>
> I expect, FreeSWITCH instances will mirror each other in this
> configuration.
>
> So, if Bob register on 192.168.0.1 server, and Alice register on
> 192.168.0.2, they can call each other like local users.
> Also, in this case, SIP messages from Bob during the call session will
> sending to any of nodes, depend on DNS resolve.
>
> Am I right?
> Is this correct way to build cluster?
>
> Please direct me in right way.
>
> BR, Dmitry
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From d.mordovin at dwide.com  Sun Apr 10 12:43:42 2016
From: d.mordovin at dwide.com (Dmitry Mordovin)
Date: Sun, 10 Apr 2016 12:43:42 +0400
Subject: [Freeswitch-users] FreeSWITCH in cluster
In-Reply-To: <CALR0W68x8tY=OOP5Xrra2rR-3i+3qdJNi08TY-hsGixtDxjodA@mail.gmail.com>
References: <570A0600.9060204@dwide.com>
	<CALR0W68x8tY=OOP5Xrra2rR-3i+3qdJNi08TY-hsGixtDxjodA@mail.gmail.com>
Message-ID: <570A123E.4000304@dwide.com>

Thank you Chandra!

Yes, in my case I need use external DB for xml_curl.

Main different, I don't want use SIP Proxy as load balancer, I want 
balance calls with DNS.

And I worrying about FreeSWITHes behavior.

Imagine one steps, Bob register and make a call:
  1. Bob send REGISTER to @mydomain.com. DNS will resolve @mydomain.com 
to 192.168.0.1 (FS-1 server). Ok, registered!
  2. Bob send INVITE to @mydomain.com. DNS will/may resolve 
@mydomain.com next time to second node 192.168.0.2 (FS-2 server).
  3. Bob send BYE to @mydomain.com. DNS will/may resolve @mydomain.com 
next time to first node again 192.168.0.1 (FS-1 server).


I don't know FS behavior on 2 and 3 steps

Step 2. FS on FS-2 server must to know Bob, which registered on FS-1 server
Step 3. FS on FS-1 server must inform FS-2 to finish the Bob's call

BR, Dmitry



On 04/10/2016 12:04 PM, Chandramouli P wrote:
> Hello Dmitry,
>
> I guess the below link with help you:
>
> http://saevolgo.blogspot.in/2012/07/freeswitch-with-sip-users-in-mysql-mod.html
>
> Thanks,
> Chandra.
>
>
>
>
> On Sun, Apr 10, 2016 at 1:21 PM, Dmitry Mordovin <d.mordovin at dwide.com 
> <mailto:d.mordovin at dwide.com>> wrote:
>
>     Hi,
>
>     I want configure two FreeSWITCH servers as cluster.
>
>     As I understand, I need:
>      - Configure one core DB (PostgreSQL) for each FreeSWITCH instances.
>      - Add two A records in DNS to resolve domain name and load balancing.
>
>     *Scheme, 2 FS servers and one DB.*
>
>     FS-1 server  192.168.0.1
>            Core-DB server  192.168.0.100
>     FS-2 server  192.168.0.2
>
>     *Add two A records in DNS*
>
>     mydomain.com <http://mydomain.com>.  43200 IN A 192.168.0.1
>     mydomain.com <http://mydomain.com>.  43200 IN A 192.168.0.2
>
>
>     I expect, FreeSWITCH instances will mirror each other in this
>     configuration.
>
>     So, if Bob register on 192.168.0.1 server, and Alice register on
>     192.168.0.2, they can call each other like local users.
>     Also, in this case, SIP messages from Bob during the call session
>     will sending to any of nodes, depend on DNS resolve.
>
>     Am I right?
>     Is this correct way to build cluster?
>
>     Please direct me in right way.
>
>     BR, Dmitry
>
>     _________________________________________________________________________
>     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 phenix at vfemail.net  Sun Apr 10 15:21:12 2016
From: phenix at vfemail.net (Tanguy)
Date: Sun, 10 Apr 2016 13:21:12 +0200
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
Message-ID: <570A3728.6020900@vfemail.net>

Hello
I would like to use nibblebilling for fraud prevention on international 
and premium numbers ( national or emergency calls should never be 
blocked and should still working even if the billing database is 
unavailable  )

Unfortunately that never hangup my calls using standard dial plan

*nibblebill.conf.xml*

<configuration name="nibblebill.conf" description="Nibble Billing">
   <settings>
     <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help 
with these options -->

     <!-- Information for connecting to your database -->
     <param name="db_dsn" value="pgsql://hostaddr=127.0.0.1 dbname=xxxx 
user=xxxx password=xxxx"/>

     <!-- The database table where your CASH column is located -->
     <param name="db_table" value="accounts"/>

     <!-- The column name where we store the value of the account -->
     <param name="db_column_cash" value="cash"/>

     <!-- The column name for the unique ID identifying the account -->
     <param name="db_column_account" value="id"/>

     <!-- Default heartbeat interval. Set to 'off' for no heartbeat 
(i.e. bill only at end of call) -->
     <param name="global_heartbeat" value="20"/>

     <!-- By default, warn a caller when their balance is at $5.00. You 
can set this to a negative number. -->
     <param name="lowbal_amt" value="-100"/><!-- i don't need this -->
     <param name="lowbal_action" value="play ding"/>

     <!-- By default, terminate a caller when their balance hits $0.00. 
You can set this to a negative number. -->
     <param name="nobal_amt" value="0"/>
     <param name="nobal_action" value="hangup"/>

     <!-- If a call goes beyond a certain dollar amount, flag or 
terminate it -->
     <param name="percall_max_amt" value="10"/>
     <param name="percall_action" value="hangup"/>

   </settings>
</configuration>

*outbound route*

<extension name="international" >
    <condition field="destination_number" expression="^00(\d{6,20})$" >
        <action application="nibblebill" data="flush"/>
         <action application="set" data="nibble_account=${accountcode}"/>
         <action application="set" data="nibble_rate=0.1"/>
        <action application="bridge" data="sofia/gateway/gw_idt/$1" />
    </condition>
</extension>


*My balance*
My balance is already negative

  select * from accounts ;
                 id                |   cash
----------------------------------+-----------
  company.voip.mydomain.com | -0.789471

*A sample call*
My balance already negative, the call will be billed but never blocked

Dialplan: sofia/internal/5003 at company.voip.domain.com Action 
nibblebill(flush)
Dialplan: sofia/internal/5003 at company.voip.domain.com Action 
set(nibble_account=${accountcode})
Dialplan: sofia/internal/5003 at company.voip.domain.com Action 
set(nibble_rate=0.1)
EXECUTE sofia/internal/5003 at company.voip.domain.com nibblebill(flush)
EXECUTE sofia/internal/5003 at company.voip.domain.com 
set(nibble_account=company.voip.domain.com)
2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477 
sofia/internal/5003 at company.voip.domain.com SET 
[nibble_account]=[company.voip.domain.com]
EXECUTE sofia/internal/5003 at company.voip.domain.com set(nibble_rate=0.1)
2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477 
sofia/internal/5003 at company.voip.domain.com SET [nibble_rate]=[0.1]
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:488 Attempting to 
bill at $0.1 per minute to account company.voip.domain.com
2016-04-10 12:10:52.132458 [INFO] mod_nibblebill.c:540 Beginning new 
billing on 644488fc-ff04-11e5-9a27-fd2791153af9
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:546 42 seconds 
passed since last bill time of 2016-04-10 12:10:09
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:563 Billing 
$0.071033 to company.voip.domain.com (Call: / 0.000000 so far)
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:393 Doing update query
2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:420 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE 
id='company.voip.domain.com']
2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:428 Retrieved 
current balance for account company.voip.domain.com (balance = -0.860504)


*Using b-leg only*
Alternatively i tried an alternative dialplan ( Even if i don't relay 
understand what is the meaning of b-leg billing  )

<action application="bridge" 
data="{enable_heartbeat_events=5,nibble_rate=0.1,nibble_account=${accountcode},originate_timeout=90}sofia/gateway/gw_idt/33$1" 
/>

The result is better because the pending call is hanged up when the 
balance reach 0.

2016-04-10 12:39:59.012462 [CRIT] mod_nibblebill.c:607 Balance of 
-0.003237 fell below allowed amount of 0.000000! (Account 
company.voip.domain.com)

But if i make a new call ( when my balance is negative )  , the caller 
party is immediately hanged but this did not cancel immediately the 
bridge:  The called party ring and can stay bridged for 61 seconds after 
answer.





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From jungleboogie0 at gmail.com  Sun Apr 10 20:22:42 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Sun, 10 Apr 2016 09:22:42 -0700
Subject: [Freeswitch-users] freeswitch with postgres setup process?
Message-ID: <CAKE2PDs-cYJJYE_KuYYX_mJq0A5_gMy9tZEvf95x32rtKyKyUQ@mail.gmail.com>

Hello All,

I have freeswitch installed from source on Debian 8.3 64 bit and
Postgresql 9.4 installed from the Postgres packages[1].

What's the best way to have Freeswitch use a database that I'll setup
and create? Do I need to recompile freeswitch with certain modules
enabled? Do I need to enable to config flags?

I've completed these things:
install unixodbc-dev
install odbc-postgresql
update /etc/odbc.ini:

[freeswitch]
; WARNING: The old psql odbc driver psqlodbc.so is now renamed psqlodbcw.so
; in version 08.x. Note that the library can also be installed under an other
; path than /usr/local/lib/ following your installation.
Driver = /usr/lib/x86_64-linux-gnu/odbc/psqlodbcw.so
Description=Connection to LDAP/POSTGRESQL
Servername=localhost
Port=5432
Protocol=6.4
FetchBufferSize=99
Username=freeswitch
Password=xxx
Database=freeswitch
ReadOnly=no
Debug=1
CommLog=1


view etc/odbcinst.ini
[PostgreSQL ANSI]
Description=PostgreSQL ODBC driver (ANSI version)
Driver=psqlodbca.so
Setup=libodbcpsqlS.so
Debug=0
CommLog=1
UsageCount=1

[PostgreSQL Unicode]
Description=PostgreSQL ODBC driver (Unicode version)
Driver=psqlodbcw.so
Setup=libodbcpsqlS.so
Debug=0
CommLog=1
UsageCount=1



And finally, test the connection:
# isql -v freeswitch
+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+



[1] http://www.postgresql.org/download/linux/debian/

-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From krice at freeswitch.org  Sun Apr 10 21:19:52 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 10 Apr 2016 12:19:52 -0500
Subject: [Freeswitch-users] freeswitch with postgres setup process?
In-Reply-To: <CAKE2PDs-cYJJYE_KuYYX_mJq0A5_gMy9tZEvf95x32rtKyKyUQ@mail.gmail.com>
References: <CAKE2PDs-cYJJYE_KuYYX_mJq0A5_gMy9tZEvf95x32rtKyKyUQ@mail.gmail.com>
Message-ID: <8f7701d1934d$3287f1f0$9797d5d0$@freeswitch.org>

FreeSWITCH if configured at build time to use postgresql can use it directly
or via odbc just by setting the DSN settings appropriately in the configs.
Check confluence for the various module configs where you might want to sue
a DB backend

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of jungle
Boogie
Sent: Sunday, April 10, 2016 11:23 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] freeswitch with postgres setup process?

Hello All,

I have freeswitch installed from source on Debian 8.3 64 bit and Postgresql
9.4 installed from the Postgres packages[1].

What's the best way to have Freeswitch use a database that I'll setup and
create? Do I need to recompile freeswitch with certain modules enabled? Do I
need to enable to config flags?

I've completed these things:
install unixodbc-dev
install odbc-postgresql
update /etc/odbc.ini:

[freeswitch]
; WARNING: The old psql odbc driver psqlodbc.so is now renamed psqlodbcw.so
; in version 08.x. Note that the library can also be installed under an
other ; path than /usr/local/lib/ following your installation.
Driver = /usr/lib/x86_64-linux-gnu/odbc/psqlodbcw.so
Description=Connection to LDAP/POSTGRESQL Servername=localhost
Port=5432
Protocol=6.4
FetchBufferSize=99
Username=freeswitch
Password=xxx
Database=freeswitch
ReadOnly=no
Debug=1
CommLog=1


view etc/odbcinst.ini
[PostgreSQL ANSI]
Description=PostgreSQL ODBC driver (ANSI version) Driver=psqlodbca.so
Setup=libodbcpsqlS.so
Debug=0
CommLog=1
UsageCount=1

[PostgreSQL Unicode]
Description=PostgreSQL ODBC driver (Unicode version) Driver=psqlodbcw.so
Setup=libodbcpsqlS.so
Debug=0
CommLog=1
UsageCount=1



And finally, test the connection:
# isql -v freeswitch
+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+



[1] http://www.postgresql.org/download/linux/debian/

--
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si

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

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

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



From s.safarov at gmail.com  Sun Apr 10 21:20:19 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 10 Apr 2016 17:20:19 +0000
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
In-Reply-To: <570A3728.6020900@vfemail.net>
References: <570A3728.6020900@vfemail.net>
Message-ID: <CAHtxdDdz3XZ1WKAXUU_aZ7RT=BWz6_H9PWZDQbbdN+jMa-hr+A@mail.gmail.com>

I use following dialplan extension. All work fine

    <extension name="setNibbleBillRate" continue="true">
        <condition field="${destination_number}"
expression="^\+7(495|499)\d+$" break="on-true">
            <action application="set" data="nibble_rate=1"/>
        </condition>
        <condition field="${destination_number}" expression="^\+7800\d+$"
break="on-true">
            <action application="set" data="nibble_rate=0"/>
        </condition>
        <condition field="${destination_number}" expression="^\+7\d+$"
break="on-true">
            <action application="set" data="nibble_rate=3"/>
        </condition>
        <condition field="${destination_number}" expression="^\+\d+$"
break="on-true">
            <action application="set" data="nibble_rate=20"/>
        </condition>
        <condition field="${destination_number}"
expression="^[a-zA-Z0-9]+$" break="on-true">
            <action application="set" data="nibble_rate=0"/>
            <anti-action application="set" data="nibble_rate=200000000"/>
        </condition>
    </extension>

    <extension name="froudContol" continue="true">
      <condition>
        <action application="nibblebill" data="flush"/>
      </condition>
    </extension>


??, 10 ???. 2016 ?. ? 14:22, Tanguy <phenix at vfemail.net>:

> Hello
> I would like to use nibblebilling for fraud prevention on international
> and premium numbers ( national or emergency calls should never be blocked
> and should still working even if the billing database is unavailable  )
>
> Unfortunately that never hangup my calls using standard dial plan
>
> *nibblebill.conf.xml*
>
> <configuration name="nibblebill.conf" description="Nibble Billing">
>   <settings>
>     <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with
> these options -->
>
>     <!-- Information for connecting to your database -->
>     <param name="db_dsn" value="pgsql://hostaddr=127.0.0.1 dbname=xxxx
> user=xxxx password=xxxx"/>
>
>     <!-- The database table where your CASH column is located -->
>     <param name="db_table" value="accounts"/>
>
>     <!-- The column name where we store the value of the account -->
>     <param name="db_column_cash" value="cash"/>
>
>     <!-- The column name for the unique ID identifying the account -->
>     <param name="db_column_account" value="id"/>
>
>     <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
> bill only at end of call) -->
>     <param name="global_heartbeat" value="20"/>
>
>     <!-- By default, warn a caller when their balance is at $5.00. You can
> set this to a negative number. -->
>     <param name="lowbal_amt" value="-100"/> <!-- i don't need this -->
>     <param name="lowbal_action" value="play ding"/>
>
>     <!-- By default, terminate a caller when their balance hits $0.00. You
> can set this to a negative number. -->
>     <param name="nobal_amt" value="0"/>
>     <param name="nobal_action" value="hangup"/>
>
>     <!-- If a call goes beyond a certain dollar amount, flag or terminate
> it -->
>     <param name="percall_max_amt" value="10"/>
>     <param name="percall_action" value="hangup"/>
>
>   </settings>
> </configuration>
>
> *outbound route*
>
> <extension name="international" >
>    <condition field="destination_number" expression="^00(\d{6,20})$" >
>        <action application="nibblebill" data="flush"/>
>         <action application="set" data="nibble_account=${accountcode}"/>
>         <action application="set" data="nibble_rate=0.1"/>
>        <action application="bridge" data="sofia/gateway/gw_idt/$1" />
>    </condition>
> </extension>
>
>
> *My balance*
> My balance is already negative
>
>  select * from accounts ;
>                 id                |   cash
> ----------------------------------+-----------
>  company.voip.mydomain.com | -0.789471
>
> *A sample call*
> My balance already negative, the call will be billed but never blocked
>
> Dialplan: sofia/internal/5003 at company.voip.domain.com Action
> nibblebill(flush)
> Dialplan: sofia/internal/5003 at company.voip.domain.com Action
> set(nibble_account=${accountcode})
> Dialplan: sofia/internal/5003 at company.voip.domain.com Action
> set(nibble_rate=0.1)
> EXECUTE sofia/internal/5003 at company.voip.domain.com nibblebill(flush)
> EXECUTE sofia/internal/5003 at company.voip.domain.com set(nibble_account=
> company.voip.domain.com)
> 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477
> sofia/internal/5003 at company.voip.domain.com SET [nibble_account]=[
> company.voip.domain.com]
> EXECUTE sofia/internal/5003 at company.voip.domain.com set(nibble_rate=0.1)
> 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477
> sofia/internal/5003 at company.voip.domain.com SET [nibble_rate]=[0.1]
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:488 Attempting to bill
> at $0.1 per minute to account company.voip.domain.com
> 2016-04-10 12:10:52.132458 [INFO] mod_nibblebill.c:540 Beginning new
> billing on 644488fc-ff04-11e5-9a27-fd2791153af9
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:546 42 seconds passed
> since last bill time of 2016-04-10 12:10:09
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:563 Billing $0.071033
> to company.voip.domain.com (Call: / 0.000000 so far)
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:393 Doing update query
> 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:420 Doing lookup query
> [SELECT cash AS nibble_balance FROM accounts WHERE id='
> company.voip.domain.com']
> 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:428 Retrieved current
> balance for account company.voip.domain.com (balance = -0.860504)
>
>
> *Using b-leg only*
> Alternatively i tried an alternative dialplan ( Even if i don't relay
> understand what is the meaning of b-leg billing  )
>
> <action application="bridge"
> data="{enable_heartbeat_events=5,nibble_rate=0.1,nibble_account=${accountcode},originate_timeout=90}sofia/gateway/gw_idt/33$1"
> />
>
> The result is better because the pending call is hanged up when the
> balance reach 0.
>
> 2016-04-10 12:39:59.012462 [CRIT] mod_nibblebill.c:607 Balance of
> -0.003237 fell below allowed amount of 0.000000! (Account
> company.voip.domain.com)
>
> But if i make a new call ( when my balance is negative )  , the caller
> party is immediately hanged but this did not cancel immediately the
> bridge:  The called party ring and can stay bridged for 61 seconds after
> answer.
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 10 21:30:19 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 10 Apr 2016 17:30:19 +0000
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
In-Reply-To: <CAHtxdDdz3XZ1WKAXUU_aZ7RT=BWz6_H9PWZDQbbdN+jMa-hr+A@mail.gmail.com>
References: <570A3728.6020900@vfemail.net>
	<CAHtxdDdz3XZ1WKAXUU_aZ7RT=BWz6_H9PWZDQbbdN+jMa-hr+A@mail.gmail.com>
Message-ID: <CAHtxdDc-Sj4UZCBoFQTYkknd6f56qHoxWApGc-MRbuXJ2Rjapg@mail.gmail.com>

Second extension

    <extension name="froudContol" continue="true">
      <condition>
        <action application="set" data="nibble_account=${accountcode}"/>
        <action application="nibblebill" data="flush"/>
      </condition>
    </extension>



??, 10 ???. 2016 ?. ? 20:21, Sergey Safarov <s.safarov at gmail.com>:

> I use following dialplan extension. All work fine
>
>     <extension name="setNibbleBillRate" continue="true">
>         <condition field="${destination_number}"
> expression="^\+7(495|499)\d+$" break="on-true">
>             <action application="set" data="nibble_rate=1"/>
>         </condition>
>         <condition field="${destination_number}" expression="^\+7800\d+$"
> break="on-true">
>             <action application="set" data="nibble_rate=0"/>
>         </condition>
>         <condition field="${destination_number}" expression="^\+7\d+$"
> break="on-true">
>             <action application="set" data="nibble_rate=3"/>
>         </condition>
>         <condition field="${destination_number}" expression="^\+\d+$"
> break="on-true">
>             <action application="set" data="nibble_rate=20"/>
>         </condition>
>         <condition field="${destination_number}"
> expression="^[a-zA-Z0-9]+$" break="on-true">
>             <action application="set" data="nibble_rate=0"/>
>             <anti-action application="set" data="nibble_rate=200000000"/>
>         </condition>
>     </extension>
>
>     <extension name="froudContol" continue="true">
>       <condition>
>
>
>         <action application="nibblebill" data="flush"/>
>
>       </condition>
>     </extension>
>
>
> ??, 10 ???. 2016 ?. ? 14:22, Tanguy <phenix at vfemail.net>:
>
>> Hello
>> I would like to use nibblebilling for fraud prevention on international
>> and premium numbers ( national or emergency calls should never be blocked
>> and should still working even if the billing database is unavailable  )
>>
>> Unfortunately that never hangup my calls using standard dial plan
>>
>> *nibblebill.conf.xml*
>>
>> <configuration name="nibblebill.conf" description="Nibble Billing">
>>   <settings>
>>     <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help
>> with these options -->
>>
>>     <!-- Information for connecting to your database -->
>>     <param name="db_dsn" value="pgsql://hostaddr=127.0.0.1 dbname=xxxx
>> user=xxxx password=xxxx"/>
>>
>>     <!-- The database table where your CASH column is located -->
>>     <param name="db_table" value="accounts"/>
>>
>>     <!-- The column name where we store the value of the account -->
>>     <param name="db_column_cash" value="cash"/>
>>
>>     <!-- The column name for the unique ID identifying the account -->
>>     <param name="db_column_account" value="id"/>
>>
>>     <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
>> bill only at end of call) -->
>>     <param name="global_heartbeat" value="20"/>
>>
>>     <!-- By default, warn a caller when their balance is at $5.00. You
>> can set this to a negative number. -->
>>     <param name="lowbal_amt" value="-100"/> <!-- i don't need this -->
>>     <param name="lowbal_action" value="play ding"/>
>>
>>     <!-- By default, terminate a caller when their balance hits $0.00.
>> You can set this to a negative number. -->
>>     <param name="nobal_amt" value="0"/>
>>     <param name="nobal_action" value="hangup"/>
>>
>>     <!-- If a call goes beyond a certain dollar amount, flag or terminate
>> it -->
>>     <param name="percall_max_amt" value="10"/>
>>     <param name="percall_action" value="hangup"/>
>>
>>   </settings>
>> </configuration>
>>
>> *outbound route*
>>
>> <extension name="international" >
>>    <condition field="destination_number" expression="^00(\d{6,20})$" >
>>        <action application="nibblebill" data="flush"/>
>>         <action application="set" data="nibble_account=${accountcode}"/>
>>         <action application="set" data="nibble_rate=0.1"/>
>>        <action application="bridge" data="sofia/gateway/gw_idt/$1" />
>>    </condition>
>> </extension>
>>
>>
>> *My balance*
>> My balance is already negative
>>
>>  select * from accounts ;
>>                 id                |   cash
>> ----------------------------------+-----------
>>  company.voip.mydomain.com | -0.789471
>>
>> *A sample call*
>> My balance already negative, the call will be billed but never blocked
>>
>> Dialplan: sofia/internal/5003 at company.voip.domain.com Action
>> nibblebill(flush)
>> Dialplan: sofia/internal/5003 at company.voip.domain.com Action
>> set(nibble_account=${accountcode})
>> Dialplan: sofia/internal/5003 at company.voip.domain.com Action
>> set(nibble_rate=0.1)
>> EXECUTE sofia/internal/5003 at company.voip.domain.com nibblebill(flush)
>> EXECUTE sofia/internal/5003 at company.voip.domain.com set(nibble_account=
>> company.voip.domain.com)
>> 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477
>> sofia/internal/5003 at company.voip.domain.com SET [nibble_account]=[
>> company.voip.domain.com]
>> EXECUTE sofia/internal/5003 at company.voip.domain.com set(nibble_rate=0.1)
>> 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477
>> sofia/internal/5003 at company.voip.domain.com SET [nibble_rate]=[0.1]
>> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:488 Attempting to
>> bill at $0.1 per minute to account company.voip.domain.com
>> 2016-04-10 12:10:52.132458 [INFO] mod_nibblebill.c:540 Beginning new
>> billing on 644488fc-ff04-11e5-9a27-fd2791153af9
>> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:546 42 seconds passed
>> since last bill time of 2016-04-10 12:10:09
>> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:563 Billing $0.071033
>> to company.voip.domain.com (Call: / 0.000000 so far)
>> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:393 Doing update query
>> 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:420 Doing lookup query
>> [SELECT cash AS nibble_balance FROM accounts WHERE id='
>> company.voip.domain.com']
>> 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:428 Retrieved current
>> balance for account company.voip.domain.com (balance = -0.860504)
>>
>>
>> *Using b-leg only*
>> Alternatively i tried an alternative dialplan ( Even if i don't relay
>> understand what is the meaning of b-leg billing  )
>>
>> <action application="bridge"
>> data="{enable_heartbeat_events=5,nibble_rate=0.1,nibble_account=${accountcode},originate_timeout=90}sofia/gateway/gw_idt/33$1"
>> />
>>
>> The result is better because the pending call is hanged up when the
>> balance reach 0.
>>
>> 2016-04-10 12:39:59.012462 [CRIT] mod_nibblebill.c:607 Balance of
>> -0.003237 fell below allowed amount of 0.000000! (Account
>> company.voip.domain.com)
>>
>> But if i make a new call ( when my balance is negative )  , the caller
>> party is immediately hanged but this did not cancel immediately the
>> bridge:  The called party ring and can stay bridged for 61 seconds after
>> answer.
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From luis.daniel.lucio at gmail.com  Sun Apr 10 21:40:10 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sun, 10 Apr 2016 13:40:10 -0400
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
In-Reply-To: <570A3728.6020900@vfemail.net>
References: <570A3728.6020900@vfemail.net>
Message-ID: <CAFLo2Qzi5oSHzXMyFEiDExdzHsmew_2a+Wne_f1Z7H6PUNTe4A@mail.gmail.com>

There is a known bug and a known patch to fix it. I can't remember right
now which one it is.
Hello
I would like to use nibblebilling for fraud prevention on international and
premium numbers ( national or emergency calls should never be blocked and
should still working even if the billing database is unavailable  )

Unfortunately that never hangup my calls using standard dial plan

*nibblebill.conf.xml*

<configuration name="nibblebill.conf" description="Nibble Billing">
  <settings>
    <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with
these options -->

    <!-- Information for connecting to your database -->
    <param name="db_dsn" value="pgsql://hostaddr=127.0.0.1 dbname=xxxx
user=xxxx password=xxxx"/>

    <!-- The database table where your CASH column is located -->
    <param name="db_table" value="accounts"/>

    <!-- The column name where we store the value of the account -->
    <param name="db_column_cash" value="cash"/>

    <!-- The column name for the unique ID identifying the account -->
    <param name="db_column_account" value="id"/>

    <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
bill only at end of call) -->
    <param name="global_heartbeat" value="20"/>

    <!-- By default, warn a caller when their balance is at $5.00. You can
set this to a negative number. -->
    <param name="lowbal_amt" value="-100"/> <!-- i don't need this -->
    <param name="lowbal_action" value="play ding"/>

    <!-- By default, terminate a caller when their balance hits $0.00. You
can set this to a negative number. -->
    <param name="nobal_amt" value="0"/>
    <param name="nobal_action" value="hangup"/>

    <!-- If a call goes beyond a certain dollar amount, flag or terminate
it -->
    <param name="percall_max_amt" value="10"/>
    <param name="percall_action" value="hangup"/>

  </settings>
</configuration>

*outbound route*

<extension name="international" >
   <condition field="destination_number" expression="^00(\d{6,20})$" >
       <action application="nibblebill" data="flush"/>
        <action application="set" data="nibble_account=${accountcode}"/>
        <action application="set" data="nibble_rate=0.1"/>
       <action application="bridge" data="sofia/gateway/gw_idt/$1" />
   </condition>
</extension>


*My balance*
My balance is already negative

 select * from accounts ;
                id                |   cash
----------------------------------+-----------
 company.voip.mydomain.com | -0.789471

*A sample call*
My balance already negative, the call will be billed but never blocked

Dialplan: sofia/internal/5003 at company.voip.domain.com Action
nibblebill(flush)
Dialplan: sofia/internal/5003 at company.voip.domain.com Action
set(nibble_account=${accountcode})
Dialplan: sofia/internal/5003 at company.voip.domain.com Action
set(nibble_rate=0.1)
EXECUTE sofia/internal/5003 at company.voip.domain.com nibblebill(flush)
EXECUTE sofia/internal/5003 at company.voip.domain.com set(nibble_account=
company.voip.domain.com)
2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477
sofia/internal/5003 at company.voip.domain.com SET [nibble_account]=[
company.voip.domain.com]
EXECUTE sofia/internal/5003 at company.voip.domain.com set(nibble_rate=0.1)
2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477
sofia/internal/5003 at company.voip.domain.com SET [nibble_rate]=[0.1]
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:488 Attempting to bill
at $0.1 per minute to account company.voip.domain.com
2016-04-10 12:10:52.132458 [INFO] mod_nibblebill.c:540 Beginning new
billing on 644488fc-ff04-11e5-9a27-fd2791153af9
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:546 42 seconds passed
since last bill time of 2016-04-10 12:10:09
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:563 Billing $0.071033
to company.voip.domain.com (Call: / 0.000000 so far)
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:393 Doing update query
2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:420 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='
company.voip.domain.com']
2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:428 Retrieved current
balance for account company.voip.domain.com (balance = -0.860504)


*Using b-leg only*
Alternatively i tried an alternative dialplan ( Even if i don't relay
understand what is the meaning of b-leg billing  )

<action application="bridge"
data="{enable_heartbeat_events=5,nibble_rate=0.1,nibble_account=${accountcode},originate_timeout=90}sofia/gateway/gw_idt/33$1"
/>

The result is better because the pending call is hanged up when the balance
reach 0.

2016-04-10 12:39:59.012462 [CRIT] mod_nibblebill.c:607 Balance of -0.003237
fell below allowed amount of 0.000000! (Account company.voip.domain.com)

But if i make a new call ( when my balance is negative )  , the caller
party is immediately hanged but this did not cancel immediately the
bridge:  The called party ring and can stay bridged for 61 seconds after
answer.






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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From krice at freeswitch.org  Sun Apr 10 21:43:53 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 10 Apr 2016 12:43:53 -0500
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
In-Reply-To: <CAFLo2Qzi5oSHzXMyFEiDExdzHsmew_2a+Wne_f1Z7H6PUNTe4A@mail.gmail.com>
References: <570A3728.6020900@vfemail.net>
	<CAFLo2Qzi5oSHzXMyFEiDExdzHsmew_2a+Wne_f1Z7H6PUNTe4A@mail.gmail.com>
Message-ID: <B700FF78-C2BA-41C3-8E9C-AD49B658DD60@freeswitch.org>

If there is a known bug and a patch for this, there should be a jira and a pull request for the patch

Sent from my iPhone

> On Apr 10, 2016, at 12:40 PM, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com> wrote:
> 
> There is a known bug and a known patch to fix it. I can't remember right now which one it is.
> 
> Hello
> I would like to use nibblebilling for fraud prevention on international and premium numbers ( national or emergency calls should never be blocked and should still working even if the billing database is unavailable  )
> 
> Unfortunately that never hangup my calls using standard dial plan
> 
> nibblebill.conf.xml
> 
> <configuration name="nibblebill.conf" description="Nibble Billing">
>   <settings>
>     <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with these options -->
> 
>     <!-- Information for connecting to your database -->
>     <param name="db_dsn" value="pgsql://hostaddr=127.0.0.1 dbname=xxxx user=xxxx password=xxxx"/>
> 
>     <!-- The database table where your CASH column is located -->
>     <param name="db_table" value="accounts"/>
> 
>     <!-- The column name where we store the value of the account -->
>     <param name="db_column_cash" value="cash"/>
> 
>     <!-- The column name for the unique ID identifying the account -->
>     <param name="db_column_account" value="id"/>
> 
>     <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e. bill only at end of call) -->
>     <param name="global_heartbeat" value="20"/>
> 
>     <!-- By default, warn a caller when their balance is at $5.00. You can set this to a negative number. -->
>     <param name="lowbal_amt" value="-100"/> <!-- i don't need this -->
>     <param name="lowbal_action" value="play ding"/>
> 
>     <!-- By default, terminate a caller when their balance hits $0.00. You can set this to a negative number. -->
>     <param name="nobal_amt" value="0"/>
>     <param name="nobal_action" value="hangup"/>
> 
>     <!-- If a call goes beyond a certain dollar amount, flag or terminate it -->
>     <param name="percall_max_amt" value="10"/>
>     <param name="percall_action" value="hangup"/>
> 
>   </settings>
> </configuration>
> 
> outbound route
> 
> <extension name="international" >
>    <condition field="destination_number" expression="^00(\d{6,20})$" >
>        <action application="nibblebill" data="flush"/>
>         <action application="set" data="nibble_account=${accountcode}"/>
>         <action application="set" data="nibble_rate=0.1"/>
>        <action application="bridge"       data="sofia/gateway/gw_idt/$1" />
>    </condition>
> </extension>
> 
> 
> My balance
> My balance is already negative
> 
>  select * from accounts ;
>                 id                |   cash    
> ----------------------------------+-----------
>  company.voip.mydomain.com | -0.789471
> 
> A sample call
> My balance already negative, the call will be billed but never blocked
> 
> Dialplan: sofia/internal/5003 at company.voip.domain.com Action nibblebill(flush)
> Dialplan: sofia/internal/5003 at company.voip.domain.com Action set(nibble_account=${accountcode})
> Dialplan: sofia/internal/5003 at company.voip.domain.com Action set(nibble_rate=0.1)
> EXECUTE sofia/internal/5003 at company.voip.domain.com nibblebill(flush)
> EXECUTE sofia/internal/5003 at company.voip.domain.com set(nibble_account=company.voip.domain.com)
> 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477 sofia/internal/5003 at company.voip.domain.com SET [nibble_account]=[company.voip.domain.com]
> EXECUTE sofia/internal/5003 at company.voip.domain.com set(nibble_rate=0.1)
> 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477 sofia/internal/5003 at company.voip.domain.com SET [nibble_rate]=[0.1]
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:488 Attempting to bill at $0.1 per minute to account company.voip.domain.com
> 2016-04-10 12:10:52.132458 [INFO] mod_nibblebill.c:540 Beginning new billing on 644488fc-ff04-11e5-9a27-fd2791153af9
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:546 42 seconds passed since last bill time of 2016-04-10 12:10:09
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:563 Billing $0.071033 to company.voip.domain.com (Call: / 0.000000 so far)
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:393 Doing update query
> 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:420 Doing lookup query
> [SELECT cash AS nibble_balance FROM accounts WHERE id='company.voip.domain.com']
> 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:428 Retrieved current balance for account company.voip.domain.com (balance = -0.860504)
> 
> 
> Using b-leg only
> Alternatively i tried an alternative dialplan ( Even if i don't relay understand what is the meaning of b-leg billing  )
> 
> <action application="bridge" data="{enable_heartbeat_events=5,nibble_rate=0.1,nibble_account=${accountcode},originate_timeout=90}sofia/gateway/gw_idt/33$1" /> 
> 
> The result is better because the pending call is hanged up when the balance reach 0.
> 
> 2016-04-10 12:39:59.012462 [CRIT] mod_nibblebill.c:607 Balance of -0.003237 fell below allowed amount of 0.000000! (Account company.voip.domain.com)
> 
> But if i make a new call ( when my balance is negative )  , the caller party is immediately hanged but this did not cancel immediately the bridge:  The called party ring and can stay bridged for 61 seconds after answer.
> 
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From luis.daniel.lucio at gmail.com  Sun Apr 10 22:09:00 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sun, 10 Apr 2016 14:09:00 -0400
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
In-Reply-To: <B700FF78-C2BA-41C3-8E9C-AD49B658DD60@freeswitch.org>
References: <570A3728.6020900@vfemail.net>
	<CAFLo2Qzi5oSHzXMyFEiDExdzHsmew_2a+Wne_f1Z7H6PUNTe4A@mail.gmail.com>
	<B700FF78-C2BA-41C3-8E9C-AD49B658DD60@freeswitch.org>
Message-ID: <CAFLo2QwD6eM6=B1TUa0ObWa9-3GLraVJxeu0UfaV6QS4zkeLmA@mail.gmail.com>

There is. It is not from myself. I just don't remember it
Le 10 avr. 2016 1:44 PM, "Ken Rice" <krice at freeswitch.org> a ?crit :

> If there is a known bug and a patch for this, there should be a jira and a
> pull request for the patch
>
> Sent from my iPhone
>
> On Apr 10, 2016, at 12:40 PM, Luis Daniel Lucio Quiroz <
> luis.daniel.lucio at gmail.com> wrote:
>
> There is a known bug and a known patch to fix it. I can't remember right
> now which one it is.
> Hello
> I would like to use nibblebilling for fraud prevention on international
> and premium numbers ( national or emergency calls should never be blocked
> and should still working even if the billing database is unavailable  )
>
> Unfortunately that never hangup my calls using standard dial plan
>
> *nibblebill.conf.xml*
>
> <configuration name="nibblebill.conf" description="Nibble Billing">
>   <settings>
>     <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with
> these options -->
>
>     <!-- Information for connecting to your database -->
>     <param name="db_dsn" value="pgsql://hostaddr=127.0.0.1 dbname=xxxx
> user=xxxx password=xxxx"/>
>
>     <!-- The database table where your CASH column is located -->
>     <param name="db_table" value="accounts"/>
>
>     <!-- The column name where we store the value of the account -->
>     <param name="db_column_cash" value="cash"/>
>
>     <!-- The column name for the unique ID identifying the account -->
>     <param name="db_column_account" value="id"/>
>
>     <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
> bill only at end of call) -->
>     <param name="global_heartbeat" value="20"/>
>
>     <!-- By default, warn a caller when their balance is at $5.00. You can
> set this to a negative number. -->
>     <param name="lowbal_amt" value="-100"/> <!-- i don't need this -->
>     <param name="lowbal_action" value="play ding"/>
>
>     <!-- By default, terminate a caller when their balance hits $0.00. You
> can set this to a negative number. -->
>     <param name="nobal_amt" value="0"/>
>     <param name="nobal_action" value="hangup"/>
>
>     <!-- If a call goes beyond a certain dollar amount, flag or terminate
> it -->
>     <param name="percall_max_amt" value="10"/>
>     <param name="percall_action" value="hangup"/>
>
>   </settings>
> </configuration>
>
> *outbound route*
>
> <extension name="international" >
>    <condition field="destination_number" expression="^00(\d{6,20})$" >
>        <action application="nibblebill" data="flush"/>
>         <action application="set" data="nibble_account=${accountcode}"/>
>         <action application="set" data="nibble_rate=0.1"/>
>        <action application="bridge" data="sofia/gateway/gw_idt/$1" />
>    </condition>
> </extension>
>
>
> *My balance*
> My balance is already negative
>
>  select * from accounts ;
>                 id                |   cash
> ----------------------------------+-----------
>  company.voip.mydomain.com | -0.789471
>
> *A sample call*
> My balance already negative, the call will be billed but never blocked
>
> Dialplan: sofia/internal/5003 at company.voip.domain.com Action
> nibblebill(flush)
> Dialplan: sofia/internal/5003 at company.voip.domain.com Action
> set(nibble_account=${accountcode})
> Dialplan: sofia/internal/5003 at company.voip.domain.com Action
> set(nibble_rate=0.1)
> EXECUTE sofia/internal/5003 at company.voip.domain.com nibblebill(flush)
> EXECUTE sofia/internal/5003 at company.voip.domain.com set(nibble_account=
> company.voip.domain.com)
> 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477
> sofia/internal/5003 at company.voip.domain.com SET [nibble_account]=[
> company.voip.domain.com]
> EXECUTE sofia/internal/5003 at company.voip.domain.com set(nibble_rate=0.1)
> 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477
> sofia/internal/5003 at company.voip.domain.com SET [nibble_rate]=[0.1]
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:488 Attempting to bill
> at $0.1 per minute to account company.voip.domain.com
> 2016-04-10 12:10:52.132458 [INFO] mod_nibblebill.c:540 Beginning new
> billing on 644488fc-ff04-11e5-9a27-fd2791153af9
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:546 42 seconds passed
> since last bill time of 2016-04-10 12:10:09
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:563 Billing $0.071033
> to company.voip.domain.com (Call: / 0.000000 so far)
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:393 Doing update query
> 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:420 Doing lookup query
> [SELECT cash AS nibble_balance FROM accounts WHERE id='
> company.voip.domain.com']
> 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:428 Retrieved current
> balance for account company.voip.domain.com (balance = -0.860504)
>
>
> *Using b-leg only*
> Alternatively i tried an alternative dialplan ( Even if i don't relay
> understand what is the meaning of b-leg billing  )
>
> <action application="bridge"
> data="{enable_heartbeat_events=5,nibble_rate=0.1,nibble_account=${accountcode},originate_timeout=90}sofia/gateway/gw_idt/33$1"
> />
>
> The result is better because the pending call is hanged up when the
> balance reach 0.
>
> 2016-04-10 12:39:59.012462 [CRIT] mod_nibblebill.c:607 Balance of
> -0.003237 fell below allowed amount of 0.000000! (Account
> company.voip.domain.com)
>
> But if i make a new call ( when my balance is negative )  , the caller
> party is immediately hanged but this did not cancel immediately the
> bridge:  The called party ring and can stay bridged for 61 seconds after
> answer.
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From saurabhkv01 at gmail.com  Sun Apr 10 22:40:20 2016
From: saurabhkv01 at gmail.com (saurabh verrma)
Date: Mon, 11 Apr 2016 00:10:20 +0530
Subject: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
In-Reply-To: <4249F84C38D04AFDA4D0D91AFAF5A017.MAI@server2.totohost.hr>
References: <4249F84C38D04AFDA4D0D91AFAF5A017.MAI@server2.totohost.hr>
Message-ID: <CACot2nvS_4h2ZxjsKtGadGcaCzVPuS7D6HHtNDJf=tmauUme0Q@mail.gmail.com>

Hi Brian/Michael,

Let me try to re-phrase what I?m trying to do if that?s creating
confusion.

I?m planning to use FreeSWITCH library in my application to create a WebRTC
endpoint (more like a SIP user agent; be able to act like UAC as well as
UAS as desired). To be able to act like that; FreeSWITCH library shall be
able support following:
-> SIP over Websockets
-> DTLS-SRTP
-> ICE

So I was just seeking guidance whether this seems to be feasible to
implement using FreeSWITCH or not? And if there are limitation; what are
they?

Help would be appreciated.

On Sun, Apr 10, 2016 at 5:37 AM, <jelena at misticnabica.hr> wrote:

>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Warm Regards:*
*Saurabh Kumar Verma*
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From jungleboogie0 at gmail.com  Mon Apr 11 00:37:28 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Sun, 10 Apr 2016 13:37:28 -0700
Subject: [Freeswitch-users] freeswitch with postgres setup process?
In-Reply-To: <8f7701d1934d$3287f1f0$9797d5d0$@freeswitch.org>
References: <CAKE2PDs-cYJJYE_KuYYX_mJq0A5_gMy9tZEvf95x32rtKyKyUQ@mail.gmail.com>
	<8f7701d1934d$3287f1f0$9797d5d0$@freeswitch.org>
Message-ID: <CAKE2PDsxVZLWbKtKZ4VHNZ+xRYViuzeQ4rWsH=vGdWmtC+nypA@mail.gmail.com>

HI Ken,
On 10 April 2016 at 10:19, Ken Rice <krice at freeswitch.org> wrote:
> FreeSWITCH if configured at build time to use postgresql can use it directly
> or via odbc just by setting the DSN settings appropriately in the configs.
> Check confluence for the various module configs where you might want to sue
> a DB backend
>
>

Okay, in the conf/autoload_configs/switch.conf.xml, I made this change:

!-- <param name="core-db-dsn" value="dsn:username:password" /> -->
<param name="core-db-dsn" value="dsn:freeswitch:xxx" />


Stopped and started FS. I still don't see any tables on the db.

Is this the config flag to use posgres:
./configure --enable-core-pgsql-support

Thanks!


From max at nysolutions.com  Mon Apr 11 00:54:15 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Sun, 10 Apr 2016 20:54:15 +0000
Subject: [Freeswitch-users] freeswitch with postgres setup process?
In-Reply-To: <CAKE2PDsxVZLWbKtKZ4VHNZ+xRYViuzeQ4rWsH=vGdWmtC+nypA@mail.gmail.com>
References: <CAKE2PDs-cYJJYE_KuYYX_mJq0A5_gMy9tZEvf95x32rtKyKyUQ@mail.gmail.com>
	<8f7701d1934d$3287f1f0$9797d5d0$@freeswitch.org>
	<CAKE2PDsxVZLWbKtKZ4VHNZ+xRYViuzeQ4rWsH=vGdWmtC+nypA@mail.gmail.com>
Message-ID: <2dd85db760054fc7880cf257510d0b32@nysolutions.com>

See https://freeswitch.org/confluence/display/FREESWITCH/PostgreSQL+in+the+core 

Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com 

Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of jungle Boogie
Sent: Sunday, April 10, 2016 4:37 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] freeswitch with postgres setup process?

HI Ken,
On 10 April 2016 at 10:19, Ken Rice <krice at freeswitch.org> wrote:
> FreeSWITCH if configured at build time to use postgresql can use it 
> directly or via odbc just by setting the DSN settings appropriately in the configs.
> Check confluence for the various module configs where you might want 
> to sue a DB backend
>
>

Okay, in the conf/autoload_configs/switch.conf.xml, I made this change:

!-- <param name="core-db-dsn" value="dsn:username:password" /> --> <param name="core-db-dsn" value="dsn:freeswitch:xxx" />


Stopped and started FS. I still don't see any tables on the db.

Is this the config flag to use posgres:
./configure --enable-core-pgsql-support

Thanks!

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

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

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


From jungleboogie0 at gmail.com  Mon Apr 11 00:59:01 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Sun, 10 Apr 2016 13:59:01 -0700
Subject: [Freeswitch-users] freeswitch with postgres setup process?
In-Reply-To: <2dd85db760054fc7880cf257510d0b32@nysolutions.com>
References: <CAKE2PDs-cYJJYE_KuYYX_mJq0A5_gMy9tZEvf95x32rtKyKyUQ@mail.gmail.com>
	<8f7701d1934d$3287f1f0$9797d5d0$@freeswitch.org>
	<CAKE2PDsxVZLWbKtKZ4VHNZ+xRYViuzeQ4rWsH=vGdWmtC+nypA@mail.gmail.com>
	<2dd85db760054fc7880cf257510d0b32@nysolutions.com>
Message-ID: <CAKE2PDvGeT3iMrqygcmAmXctu+j=c4QdYLBqCk14QPE2hiOTFA@mail.gmail.com>

On 10 April 2016 at 13:54, Moishe Grunstein <max at nysolutions.com> wrote:
> See https://freeswitch.org/confluence/display/FREESWITCH/PostgreSQL+in+the+core

I was told odbc is preferred but I'll that page out and see what happens.

Thanks!


-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From krice at freeswitch.org  Mon Apr 11 01:44:40 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 10 Apr 2016 16:44:40 -0500
Subject: [Freeswitch-users] freeswitch with postgres setup process?
In-Reply-To: <CAKE2PDsxVZLWbKtKZ4VHNZ+xRYViuzeQ4rWsH=vGdWmtC+nypA@mail.gmail.com>
References: <CAKE2PDs-cYJJYE_KuYYX_mJq0A5_gMy9tZEvf95x32rtKyKyUQ@mail.gmail.com>
	<8f7701d1934d$3287f1f0$9797d5d0$@freeswitch.org>
	<CAKE2PDsxVZLWbKtKZ4VHNZ+xRYViuzeQ4rWsH=vGdWmtC+nypA@mail.gmail.com>
Message-ID: <EB891440-F42E-4998-8474-3C55C9832E1B@freeswitch.org>

Thats because the db strings for native pgsql are libpq connect strings dsn:user:pass is odbc

Sent from my iPhone

> On Apr 10, 2016, at 3:37 PM, jungle Boogie <jungleboogie0 at gmail.com> wrote:
> 
> HI Ken,
>> On 10 April 2016 at 10:19, Ken Rice <krice at freeswitch.org> wrote:
>> FreeSWITCH if configured at build time to use postgresql can use it directly
>> or via odbc just by setting the DSN settings appropriately in the configs.
>> Check confluence for the various module configs where you might want to sue
>> a DB backend
> 
> Okay, in the conf/autoload_configs/switch.conf.xml, I made this change:
> 
> !-- <param name="core-db-dsn" value="dsn:username:password" /> -->
> <param name="core-db-dsn" value="dsn:freeswitch:xxx" />
> 
> 
> Stopped and started FS. I still don't see any tables on the db.
> 
> Is this the config flag to use posgres:
> ./configure --enable-core-pgsql-support
> 
> Thanks!
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From krice at freeswitch.org  Mon Apr 11 01:45:31 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 10 Apr 2016 16:45:31 -0500
Subject: [Freeswitch-users] freeswitch with postgres setup process?
In-Reply-To: <CAKE2PDvGeT3iMrqygcmAmXctu+j=c4QdYLBqCk14QPE2hiOTFA@mail.gmail.com>
References: <CAKE2PDs-cYJJYE_KuYYX_mJq0A5_gMy9tZEvf95x32rtKyKyUQ@mail.gmail.com>
	<8f7701d1934d$3287f1f0$9797d5d0$@freeswitch.org>
	<CAKE2PDsxVZLWbKtKZ4VHNZ+xRYViuzeQ4rWsH=vGdWmtC+nypA@mail.gmail.com>
	<2dd85db760054fc7880cf257510d0b32@nysolutions.com>
	<CAKE2PDvGeT3iMrqygcmAmXctu+j=c4QdYLBqCk14QPE2hiOTFA@mail.gmail.com>
Message-ID: <FC1D0F78-0E42-4D9B-A152-1BB5A7B09B45@freeswitch.org>

Whi said odbc is preferred? I prefer native pgsql over odbc every day

Sent from my iPhone

> On Apr 10, 2016, at 3:59 PM, jungle Boogie <jungleboogie0 at gmail.com> wrote:
> 
>> On 10 April 2016 at 13:54, Moishe Grunstein <max at nysolutions.com> wrote:
>> See https://freeswitch.org/confluence/display/FREESWITCH/PostgreSQL+in+the+core
> 
> I was told odbc is preferred but I'll that page out and see what happens.
> 
> Thanks!
> 
> 
> -- 
> -------
> inum: 883510009027723
> sip: jungleboogie at sip2sip.info
> xmpp: jungle-boogie at jit.si
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jungleboogie0 at gmail.com  Mon Apr 11 02:06:25 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Sun, 10 Apr 2016 15:06:25 -0700
Subject: [Freeswitch-users] freeswitch with postgres setup process?
In-Reply-To: <EB891440-F42E-4998-8474-3C55C9832E1B@freeswitch.org>
References: <CAKE2PDs-cYJJYE_KuYYX_mJq0A5_gMy9tZEvf95x32rtKyKyUQ@mail.gmail.com>
	<8f7701d1934d$3287f1f0$9797d5d0$@freeswitch.org>
	<CAKE2PDsxVZLWbKtKZ4VHNZ+xRYViuzeQ4rWsH=vGdWmtC+nypA@mail.gmail.com>
	<EB891440-F42E-4998-8474-3C55C9832E1B@freeswitch.org>
Message-ID: <CAKE2PDsiert0L8GYBc_vdNsZ5oV=r6whE_ROTnAdrBvGxmrASQ@mail.gmail.com>

On 10 April 2016 at 14:44, Ken Rice <krice at freeswitch.org> wrote:
> Thats because the db strings for native pgsql are libpq connect strings dsn:user:pass is odbc

Which one is correct if I do ./configure --enable-core-pgsql-support

1. <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1
dbname=freeswitch user=freeswitch password='xxx' options='-c
client_min_messages=NOTICE'" />

2.  <param name="core-db-dsn" value="dsn:freeswitch:xxx" />


Also, I see event_handlers/mod_cdr_pg_csv in modules.conf Does that
need to be uncommented?


Thanks,
jb


-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From jelena at misticnabica.hr  Mon Apr 11 02:13:45 2016
From: jelena at misticnabica.hr (jelena at misticnabica.hr)
Date: Sun, 10 Apr 2016 22:13:45 GMT
Subject: [Freeswitch-users] freeswitch with postgres setup process?
Message-ID: <CC4859A745754E66B9B8A5259A98452F.MAI@server2.totohost.hr>



From max at nysolutions.com  Mon Apr 11 02:45:02 2016
From: max at nysolutions.com (Moishe Grunstein)
Date: Sun, 10 Apr 2016 22:45:02 +0000
Subject: [Freeswitch-users] freeswitch with postgres setup process?
In-Reply-To: <CAKE2PDsiert0L8GYBc_vdNsZ5oV=r6whE_ROTnAdrBvGxmrASQ@mail.gmail.com>
References: <CAKE2PDs-cYJJYE_KuYYX_mJq0A5_gMy9tZEvf95x32rtKyKyUQ@mail.gmail.com>
	<8f7701d1934d$3287f1f0$9797d5d0$@freeswitch.org>
	<CAKE2PDsxVZLWbKtKZ4VHNZ+xRYViuzeQ4rWsH=vGdWmtC+nypA@mail.gmail.com>
	<EB891440-F42E-4998-8474-3C55C9832E1B@freeswitch.org>
	<CAKE2PDsiert0L8GYBc_vdNsZ5oV=r6whE_ROTnAdrBvGxmrASQ@mail.gmail.com>
Message-ID: <1581118448714ea7be3e70ff9a9f0dea@nysolutions.com>

If you use postgres in the core 
1. <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='xxx' options='-c client_min_messages=NOTICE'" />
https://freeswitch.org/confluence/display/FREESWITCH/PostgreSQL+in+the+core

If you use odbc 
2.  <param name="core-db-dsn" value="dsn:freeswitch:xxx" />
https://freeswitch.org/confluence/display/FREESWITCH/ODBC+DSN




Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com 

Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of jungle Boogie
Sent: Sunday, April 10, 2016 6:06 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] freeswitch with postgres setup process?

On 10 April 2016 at 14:44, Ken Rice <krice at freeswitch.org> wrote:
> Thats because the db strings for native pgsql are libpq connect 
> strings dsn:user:pass is odbc

Which one is correct if I do ./configure --enable-core-pgsql-support

1. <param name="core-db-dsn" value="pgsql://hostaddr=127.0.0.1 dbname=freeswitch user=freeswitch password='xxx' options='-c client_min_messages=NOTICE'" />

2.  <param name="core-db-dsn" value="dsn:freeswitch:xxx" />


Also, I see event_handlers/mod_cdr_pg_csv in modules.conf Does that need to be uncommented?


Thanks,
jb


--
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si

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

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

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


From mike at jerris.com  Mon Apr 11 03:47:56 2016
From: mike at jerris.com (Michael Jerris)
Date: Sun, 10 Apr 2016 19:47:56 -0400
Subject: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
In-Reply-To: <CACot2nvS_4h2ZxjsKtGadGcaCzVPuS7D6HHtNDJf=tmauUme0Q@mail.gmail.com>
References: <4249F84C38D04AFDA4D0D91AFAF5A017.MAI@server2.totohost.hr>
	<CACot2nvS_4h2ZxjsKtGadGcaCzVPuS7D6HHtNDJf=tmauUme0Q@mail.gmail.com>
Message-ID: <CAEwTsAgZKq4OSeFFCx+1QHSRyrpGa_qXLopLwwqNGdR=dXObxg@mail.gmail.com>

I already answered with the limitation that probably makes it a bad match.

On Sunday, April 10, 2016, saurabh verrma <saurabhkv01 at gmail.com> wrote:

> Hi Brian/Michael,
>
> Let me try to re-phrase what I?m trying to do if that?s creating
> confusion.
>
> I?m planning to use FreeSWITCH library in my application to create a
> WebRTC endpoint (more like a SIP user agent; be able to act like UAC as
> well as UAS as desired). To be able to act like that; FreeSWITCH library
> shall be able support following:
> -> SIP over Websockets
> -> DTLS-SRTP
> -> ICE
>
> So I was just seeking guidance whether this seems to be feasible to
> implement using FreeSWITCH or not? And if there are limitation; what are
> they?
>
> Help would be appreciated.
>
> On Sun, Apr 10, 2016 at 5:37 AM, <jelena at misticnabica.hr
> <javascript:_e(%7B%7D,'cvml','jelena at misticnabica.hr');>> wrote:
>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> *With Warm Regards:*
> *Saurabh Kumar Verma*
>
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From quanhs at stee.stengg.com  Mon Apr 11 07:11:06 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Mon, 11 Apr 2016 03:11:06 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299
	CODEC	NEGOTIATION ERROR. SDP
In-Reply-To: <C4A0FD03-EF2C-476F-B011-2950CA655689@jerris.com>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
	<38419347-D033-43AB-92DA-91D7D4C90465@jerris.com>
	<5c023f40f79d430c85dc5d902b33ebee@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAHtxdDf7KAQHEi+ennJs3Wnfd-pnuVjpzJ9Gn2Bert2AY-wzfA@mail.gmail.com>
	<302e815fb7c642d094445f4465256b92@RESDRCMBX002.Resources.STELECT.LOCAL>
	<CAHtxdDfHgBza0eiWqqxGCi_PH-Y8cGxGdGJFDRLhrnxC3rbWAA@mail.gmail.com>
	<a0e8cc45ffae4f42bb7be231bb03a163@RESDRCMBX002.Resources.STELECT.LOCAL>
	<C4A0FD03-EF2C-476F-B011-2950CA655689@jerris.com>
Message-ID: <9b88e0f313a841d8af36b79c396b87a1@RESAMKMBX002.Resources.STELECT.LOCAL>

Because existing SIP endpoint doesn?t support SRTP-DTLS, the transcoding (SRTP-DTLS and SRTP-SDES)is required to bridge RTP stream between WebRTC browser and SIP endpoint.


1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR.  SDP:
1709c7c4-fa17-11e5-9184-9d31ebce310e v=0
1709c7c4-fa17-11e5-9184-9d31ebce310e o=- 2269318439683772682 2 IN IP4 127.0.0.1
1709c7c4-fa17-11e5-9184-9d31ebce310e s=-
1709c7c4-fa17-11e5-9184-9d31ebce310e t=0 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=group:BUNDLE audio
1709c7c4-fa17-11e5-9184-9d31ebce310e a=msid-semantic: WMS rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o
1709c7c4-fa17-11e5-9184-9d31ebce310e m=audio 1107 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
1709c7c4-fa17-11e5-9184-9d31ebce310e c=IN IP4 192.168.199.216
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:111 opus/48000/2
1709c7c4-fa17-11e5-9184-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:103 ISAC/16000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:104 ISAC/32000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:9 G722/8000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:0 PCMU/8000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:8 PCMA/8000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:106 CN/32000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:105 CN/16000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:13 CN/8000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:126 telephone-event/8000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp:1110 IN IP4 192.168.199.216
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 1 udp 2122260223 192.168.199.216 1107 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 1 udp 2122194687 10.20.102.216 1108 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 1 udp 2122129151 10.10.13.216 1109 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 2 udp 2122260222 192.168.199.216 1110 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 2 udp 2122194686 10.20.102.216 1111 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 2 udp 2122129150 10.10.13.216 1112 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 1 tcp 1518280447 192.168.199.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 1 tcp 1518214911 10.20.102.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 1 tcp 1518149375 10.10.13.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 2 tcp 1518280446 192.168.199.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 2 tcp 1518214910 10.20.102.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 2 tcp 1518149374 10.10.13.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-ufrag:4PWOC8zOrgkF0JcH
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-pwd:KoIliPYZgGkk+b6K7bJDPVLK
1709c7c4-fa17-11e5-9184-9d31ebce310e a=fingerprint:sha-256 18:9D:FC:F1:74:38:63:AE:F1:F8:F0:26:F5:83:A8:41:97:53:67:F5:35:FB:7E:F3:06:BF:D5:71:FD:A2:F6:38
1709c7c4-fa17-11e5-9184-9d31ebce310e a=setup:actpass
1709c7c4-fa17-11e5-9184-9d31ebce310e a=mid:audio
1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp-mux
1709c7c4-fa17-11e5-9184-9d31ebce310e a=maxptime:60
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 cname:mrCkzHA0WH9Dd+8n
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 msid:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 62a659a4-ef68-469a-871c-0e61ef3dd211
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 mslabel:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 label:62a659a4-ef68-469a-871c-0e61ef3dd211

1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [NOTICE] switch_channel.c:3501 Hangup sofia/internal/anonymous at 192.168.199.22 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_ivr_originate.c:1217 sofia/internal/anonymous at 192.168.199.22 Media Establishment Failed.

###############################################
B leg:

1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.471064 [DEBUG] mod_sofia.c:88 sofia/internal/5shm6jpu at tibvb7p7p6od.invalid SOFIA INIT
1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1228 sip:5shm6jpu at 192.168.199.216:1106;transport=wss Setting proxy route to sofia/internal/5shm6jpu at tibvb7p7p6od.invalid
1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1257 sofia/internal/5shm6jpu at tibvb7p7p6od.invalid sending invite version: 1.6.5  64bit
1d260ca8-fa17-11e5-91a9-9d31ebce310e Local SDP:
1d260ca8-fa17-11e5-91a9-9d31ebce310e v=0
1d260ca8-fa17-11e5-91a9-9d31ebce310e o=FreeSWITCH 1459715932 1459715933 IN IP4 192.168.199.22
1d260ca8-fa17-11e5-91a9-9d31ebce310e s=FreeSWITCH
1d260ca8-fa17-11e5-91a9-9d31ebce310e c=IN IP4 192.168.199.22
1d260ca8-fa17-11e5-91a9-9d31ebce310e t=0 0
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=msid-semantic: WMS V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp
1d260ca8-fa17-11e5-91a9-9d31ebce310e m=audio 25356 RTP/SAVPF 111 8 102 101
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:111 opus/48000/2
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:8 PCMA/8000
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:102 telephone-event/48000
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:101 telephone-event/8000
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fingerprint:sha-256 E7:50:3F:56:DE:21:E5:17:DA:F7:3A:3F:28:F8:BC:2A:C6:E7:E0:67:5C:6C:61:39:00:6D:99:DE:A4:A2:8A:80
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=setup:actpass
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp-mux
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp:25356 IN IP4 192.168.199.22
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 cname:OFXIzFBV4sOBkyJG
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 msid:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp a0
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 mslabel:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 label:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzpa0
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-ufrag:UUr7xgAHzqT3ejiT
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-pwd:O6reBS0vbEXl3xRNh8KFfbcE
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 1 udp 659136 192.168.199.22 25356 typ host generation 0
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 2 udp 659136 192.168.199.22 25356 typ host generation 0
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ptime:20
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=sendrecv


@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@
Profile: internal

2016-04-04 11:17:29.618618 [DEBUG] sofia.c:4236 debug [9]
2016-04-04 11:17:29.618624 [DEBUG] sofia.c:4236 sip-trace [no]
2016-04-04 11:17:29.618627 [DEBUG] sofia.c:4236 sip-capture [no]
2016-04-04 11:17:29.618630 [DEBUG] sofia.c:4236 watchdog-enabled [no]
2016-04-04 11:17:29.618634 [DEBUG] sofia.c:4236 watchdog-step-timeout [30000]
2016-04-04 11:17:29.618638 [DEBUG] sofia.c:4236 watchdog-event-timeout [30000]
2016-04-04 11:17:29.618642 [DEBUG] sofia.c:4236 log-auth-failures [true]
2016-04-04 11:17:29.618645 [DEBUG] sofia.c:4236 forward-unsolicited-mwi-notify [false]
2016-04-04 11:17:29.618648 [DEBUG] sofia.c:4236 context [public]
2016-04-04 11:17:29.618654 [DEBUG] sofia.c:4236 rfc2833-pt [101]
2016-04-04 11:17:29.618657 [DEBUG] sofia.c:4236 sip-port [5062]
2016-04-04 11:17:29.618661 [DEBUG] sofia.c:4236 dialplan [XML]
2016-04-04 11:17:29.618667 [DEBUG] sofia.c:4236 dtmf-duration [2000]
2016-04-04 11:17:29.618677 [DEBUG] sofia.c:4236 inbound-codec-prefs [OPUS]
2016-04-04 11:17:29.618684 [DEBUG] sofia.c:4236 outbound-codec-prefs [OPUS]
2016-04-04 11:17:29.618690 [DEBUG] sofia.c:4236 rtp-timer-name [soft]
2016-04-04 11:17:29.618694 [DEBUG] sofia.c:4236 rtp-ip [192.168.199.22]
2016-04-04 11:17:29.618698 [DEBUG] sofia.c:4236 sip-ip [192.168.199.22]
2016-04-04 11:17:29.618702 [DEBUG] sofia.c:4236 hold-music [local_stream://moh]
2016-04-04 11:17:29.618720 [DEBUG] sofia.c:4236 apply-nat-acl [nat.auto]
2016-04-04 11:17:29.618734 [ERR] sofia.c:5185 Not adding acl nat.auto because it's the local network
2016-04-04 11:17:29.618739 [DEBUG] sofia.c:4236 apply-inbound-acl [domains]
2016-04-04 11:17:29.618745 [DEBUG] sofia.c:4236 local-network-acl [localnet.auto]
2016-04-04 11:17:29.618749 [DEBUG] sofia.c:4236 record-path [/usr/local/freeswitch/recordings]
2016-04-04 11:17:29.618754 [DEBUG] sofia.c:4236 record-template [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
2016-04-04 11:17:29.618763 [DEBUG] sofia.c:4236 manage-presence [true]
2016-04-04 11:17:29.618768 [DEBUG] sofia.c:4236 presence-hosts [192.168.199.22,192.168.199.22]
2016-04-04 11:17:29.618774 [DEBUG] sofia.c:4236 presence-privacy [false]
2016-04-04 11:17:29.618778 [DEBUG] sofia.c:4236 inbound-codec-negotiation [generous]
2016-04-04 11:17:29.618783 [DEBUG] sofia.c:4236 tls [true]
2016-04-04 11:17:29.618791 [DEBUG] sofia.c:4236 tls-only [false]
2016-04-04 11:17:29.618798 [DEBUG] sofia.c:4236 tls-bind-params [transport=tls]
2016-04-04 11:17:29.618804 [DEBUG] sofia.c:4236 tls-sip-port [5063]
2016-04-04 11:17:29.618810 [DEBUG] sofia.c:4236 tls-passphrase []
2016-04-04 11:17:29.618816 [DEBUG] sofia.c:4236 tls-verify-date [true]
2016-04-04 11:17:29.618822 [DEBUG] sofia.c:4236 tls-verify-policy [none]
2016-04-04 11:17:29.618829 [ERR] sofia_glue.c:329 Invalid tls-verify-policy value: none
2016-04-04 11:17:29.618835 [DEBUG] sofia.c:4236 tls-verify-depth [2]
2016-04-04 11:17:29.618844 [DEBUG] sofia.c:4236 tls-verify-in-subjects []
2016-04-04 11:17:29.618850 [DEBUG] sofia.c:4236 tls-version [tlsv1,tlsv1.1,tlsv1.2]
2016-04-04 11:17:29.618860 [DEBUG] sofia.c:4236 tls-ciphers [ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH]
2016-04-04 11:17:29.618868 [DEBUG] sofia.c:4236 inbound-late-negotiation [true]
2016-04-04 11:17:29.618872 [DEBUG] sofia.c:4236 inbound-zrtp-passthru [true]
2016-04-04 11:17:29.618876 [DEBUG] sofia.c:4236 nonce-ttl [60]
2016-04-04 11:17:29.618881 [DEBUG] sofia.c:4236 auth-calls [true]
2016-04-04 11:17:29.618887 [DEBUG] sofia.c:4236 inbound-reg-force-matching-username [true]
2016-04-04 11:17:29.618893 [DEBUG] sofia.c:4236 auth-all-packets [false]
2016-04-04 11:17:29.618898 [DEBUG] sofia.c:4236 ext-rtp-ip [auto-nat]
2016-04-04 11:17:29.618902 [DEBUG] sofia.c:4236 ext-sip-ip [auto-nat]
2016-04-04 11:17:29.618908 [DEBUG] sofia.c:4236 rtp-timeout-sec [300]
2016-04-04 11:17:29.618913 [DEBUG] sofia.c:4236 rtp-hold-timeout-sec [1800]
2016-04-04 11:17:29.618917 [DEBUG] sofia.c:4236 force-register-domain [192.168.199.22]
2016-04-04 11:17:29.618922 [DEBUG] sofia.c:4236 force-subscription-domain [192.168.199.22]
2016-04-04 11:17:29.618927 [DEBUG] sofia.c:4236 force-register-db-domain [192.168.199.22]
2016-04-04 11:17:29.618933 [DEBUG] sofia.c:4236 ws-binding [:5066]
2016-04-04 11:17:29.618939 [DEBUG] sofia.c:4236 wss-binding [:7443]
2016-04-04 11:17:29.618945 [DEBUG] sofia.c:4236 challenge-realm [auto_from]
2016-04-04 11:17:29.618953 [INFO] sofia.c:5513 Setting MAX Auth Validity to 0 Attempts
2016-04-04 11:17:29.619091 [NOTICE] sofia.c:5680 Started Profile internal [sofia_reg_internal]

2016-04-04 11:24:00.891443 [DEBUG] switch_loadable_module.c:735 Chat Thread Started
2016-04-04 11:24:00.891453 [INFO] switch_core.c:2418
FreeSWITCH Version 1.6.5~64bit ( 64bit)

The detailed log is attached in previous emails.


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Friday, April 08, 2016 11:29 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Seriously, use verto... it will be much cleaner... also, the issue you are likely having has to do with codec negotiation settings, but we can't say for sure without seeing a debug log.

On Apr 8, 2016, at 4:16 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

Good. Can you share your scenario ?

Chrome (sipML5) ->FS (1.6.5-64bit  Media mode) ->Chrome (sipML5).


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
Sent: Friday, April 08, 2016 3:17 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

All call with media transcoding enabled. In WebRTC case OPUS <-> G711a

On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
what is setting of inbound-bypass-media and inbound-proxy-media in your case?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Sergey Safarov
Sent: Friday, April 08, 2016 12:00 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

One week ago I has configured master with sipML5.
You can try reproduce.

On Fri, Apr 8, 2016, 06:52 Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Michael;

Same complaint at mod_sofia.c 2299.
Mod_sofia.c: 2299 (sofia_media_tech_media(tech_pvt, r_sdp) != SWITCH_STATUS_SUCCESS).
Switch_core_media.c :3573: switch_core_media_negotiation_sdp().

Eval ${webrtc_enable_dtls}, Eval ${media_webrtc}, Eval ${rtp_use_dtls}, Eval ${rtp_secure_media}, all return true.

Does anyone successfully set up this WebRTC demo (excluding video) using media mode as described by cookbook.


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 10:17 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

media_webrtc=true


On Apr 7, 2016, at 3:05 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

Hi Michael;

Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as WebRTC signaling.

You can see SDP from chrome (+sipjs) for this in previous attachments.

If set inbound-bypass-media=true, then chrome caller can talk with chrome callee using opus without any issue.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 2:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?

On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 1:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

you keep saying

(UDP/TLS/RTP/SAVPF)

are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.

On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi;

Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Quan Huo Sheng
Sent: Thursday, April 07, 2016 10:41 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.

Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.

As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.

In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).

Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Wednesday, April 06, 2016 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

yes it's supported

On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Itola;

Sorry, same error.

Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?

Finding when FS runs in media mode:
codec causes caller side ?488 not acceptable here| incompatible destination ?
                callee side: ?cancel |user not registered?

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Set media_mix_inbound_outbound_codecs=true

?talo Rossi
italo at freeswitch.org<mailto:italo at freeswitch.org>
IRC chat.freenode.net<http://chat.freenode.net/> #freeswitch #freeswitch-dev
Bugs? https://freeswitch.org/jira
Docs? https://freeswitch.org/jira
Chat? https://hipchat.freeswitch.org/gUdAgy0m6


On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi All;

I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min<http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min<http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.
If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
But if disabling media bypass mode, call is rejected by FS.

FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.
Chrome uses opus 111, FS uses opus 116.
Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().

Help is needed to troubleshoot this issue.

Thanks advance.
Smile.


[This e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
from your system; please do not copy or use it for any purpose, nor disclose
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From mandra at gmail.com  Mon Apr 11 07:47:54 2016
From: mandra at gmail.com (Chris Mandra)
Date: Sun, 10 Apr 2016 23:47:54 -0400
Subject: [Freeswitch-users] curl connect timeout question
In-Reply-To: <CAJdRzN5MsyG8gcHEsbR3udnt4vGV0QWciypgJbAkwzNypsMJ7A@mail.gmail.com>
References: <CAJdRzN5MsyG8gcHEsbR3udnt4vGV0QWciypgJbAkwzNypsMJ7A@mail.gmail.com>
Message-ID: <CAJdRzN4rDu=ZOsgSX-C6qGhwg2iTGdNDj7WrZKPwF+se=8LGBw@mail.gmail.com>

Any ideas?

On Saturday, April 9, 2016, Chris Mandra <mandra at gmail.com> wrote:

> Hey guys, if I'm using curl in a module and the connection times out do
> you have suggestion for how this should be handled?
>
> For example, if I disconnect freeswitch from the internet and have the
> following settings:
>
>
> curl_easy_setopt(curl_handle, CURLOPT_CONNECTTIMEOUT, 5L);
> curl_easy_setopt(curl_handle, CURLOPT_TIMEOUT, 5L);
> switch_curl_easy_setopt(curl_handle, CURLOPT_NOSIGNAL, 1L);
>
> it never times out and then crashes freeswitch. Is this expected behavior?
> Is there something I should be doing instead?
> Thanks, chris
>
> --
> mandra
>
>

-- 
mandra
c:410.258.5281
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From d at d-man.org  Mon Apr 11 08:27:25 2016
From: d at d-man.org (Darren)
Date: Mon, 11 Apr 2016 04:27:25 +0000
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
In-Reply-To: <CAFLo2QwD6eM6=B1TUa0ObWa9-3GLraVJxeu0UfaV6QS4zkeLmA@mail.gmail.com>
References: <570A3728.6020900@vfemail.net>
	<CAFLo2Qzi5oSHzXMyFEiDExdzHsmew_2a+Wne_f1Z7H6PUNTe4A@mail.gmail.com>
	<B700FF78-C2BA-41C3-8E9C-AD49B658DD60@freeswitch.org>
	<CAFLo2QwD6eM6=B1TUa0ObWa9-3GLraVJxeu0UfaV6QS4zkeLmA@mail.gmail.com>
Message-ID: <B6118ED6-AAA7-462D-B4CB-06D3713948A8@d-man.org>

If you can point me to the pull request, I can take a look and see if I can integrate it in.

From: <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com<mailto:luis.daniel.lucio at gmail.com>>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Date: Sunday, April 10, 2016 at 11:09 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] nibblebill do not terminate calls properly


There is. It is not from myself. I just don't remember it

Le 10 avr. 2016 1:44 PM, "Ken Rice" <krice at freeswitch.org<mailto:krice at freeswitch.org>> a ?crit :
If there is a known bug and a patch for this, there should be a jira and a pull request for the patch

Sent from my iPhone

On Apr 10, 2016, at 12:40 PM, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com<mailto:luis.daniel.lucio at gmail.com>> wrote:


There is a known bug and a known patch to fix it. I can't remember right now which one it is.

Hello
I would like to use nibblebilling for fraud prevention on international and premium numbers ( national or emergency calls should never be blocked and should still working even if the billing database is unavailable  )

Unfortunately that never hangup my calls using standard dial plan

nibblebill.conf.xml

<configuration name="nibblebill.conf" description="Nibble Billing">
  <settings>
    <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with these options -->

    <!-- Information for connecting to your database -->
    <param name="db_dsn" value="pgsql://hostaddr=127.0.0.1 dbname=xxxx user=xxxx password=xxxx"/>

    <!-- The database table where your CASH column is located -->
    <param name="db_table" value="accounts"/>

    <!-- The column name where we store the value of the account -->
    <param name="db_column_cash" value="cash"/>

    <!-- The column name for the unique ID identifying the account -->
    <param name="db_column_account" value="id"/>

    <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e. bill only at end of call) -->
    <param name="global_heartbeat" value="20"/>

    <!-- By default, warn a caller when their balance is at $5.00. You can set this to a negative number. -->
    <param name="lowbal_amt" value="-100"/> <!-- i don't need this -->
    <param name="lowbal_action" value="play ding"/>

    <!-- By default, terminate a caller when their balance hits $0.00. You can set this to a negative number. -->
    <param name="nobal_amt" value="0"/>
    <param name="nobal_action" value="hangup"/>

    <!-- If a call goes beyond a certain dollar amount, flag or terminate it -->
    <param name="percall_max_amt" value="10"/>
    <param name="percall_action" value="hangup"/>

  </settings>
</configuration>

outbound route

<extension name="international" >
   <condition field="destination_number" expression="^00(\d{6,20})$" >
       <action application="nibblebill" data="flush"/>
        <action application="set" data="nibble_account=${accountcode}"/>
        <action application="set" data="nibble_rate=0.1"/>
       <action application="bridge" data="sofia/gateway/gw_idt/$1" />
   </condition>
</extension>


My balance
My balance is already negative

 select * from accounts ;
                id                |   cash
----------------------------------+-----------
 company.voip.mydomain.com<http://company.voip.mydomain.com> | -0.789471

A sample call
My balance already negative, the call will be billed but never blocked

Dialplan: sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> Action nibblebill(flush)
Dialplan: sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> Action set(nibble_account=${accountcode})
Dialplan: sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> Action set(nibble_rate=0.1)
EXECUTE sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> nibblebill(flush)
EXECUTE sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> set(nibble_account=company.voip.domain.com<http://company.voip.domain.com>)
2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477 sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> SET [nibble_account]=[company.voip.domain.com<http://company.voip.domain.com>]
EXECUTE sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> set(nibble_rate=0.1)
2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477 sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> SET [nibble_rate]=[0.1]
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:488 Attempting to bill at $0.1 per minute to account company.voip.domain.com<http://company.voip.domain.com>
2016-04-10 12:10:52.132458 [INFO] mod_nibblebill.c:540 Beginning new billing on 644488fc-ff04-11e5-9a27-fd2791153af9
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:546 42 seconds passed since last bill time of 2016-04-10 12:10:09
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:563 Billing $0.071033 to company.voip.domain.com<http://company.voip.domain.com> (Call: / 0.000000 so far)
2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:393 Doing update query
2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:420 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='company.voip.domain.com<http://company.voip.domain.com>']
2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:428 Retrieved current balance for account company.voip.domain.com<http://company.voip.domain.com> (balance = -0.860504)


Using b-leg only
Alternatively i tried an alternative dialplan ( Even if i don't relay understand what is the meaning of b-leg billing  )

<action application="bridge" data="{enable_heartbeat_events=5,nibble_rate=0.1,nibble_account=${accountcode},originate_timeout=90}sofia/gateway/gw_idt/33$1" />

The result is better because the pending call is hanged up when the balance reach 0.

2016-04-10 12:39:59.012462 [CRIT] mod_nibblebill.c:607 Balance of -0.003237 fell below allowed amount of 0.000000! (Account company.voip.domain.com<http://company.voip.domain.com>)

But if i make a new call ( when my balance is negative )  , the caller party is immediately hanged but this did not cancel immediately the bridge:  The called party ring and can stay bridged for 61 seconds after answer.






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From stasan89 at gmail.com  Mon Apr 11 13:59:05 2016
From: stasan89 at gmail.com (=?UTF-8?B?0KHRgtCw0YEg0KLQtdC70YzQvdC+0LI=?=)
Date: Mon, 11 Apr 2016 12:59:05 +0300
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
 configuring NAT on Amazon EC2
In-Reply-To: <CAOLEQ7fZ6FJH9v5wiKv87MFKihc+gCXSbc2-CUnQhw5xDXogJg@mail.gmail.com>
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Message-ID: <CAC89h=KjuTnY5yoiHkUT6bvbFbDzt9imTCCZBsMHBmJ3bJi4pA@mail.gmail.com>

>>Artur Mega: Do you use sip-proxy? If you use sip proxy, maybe you forget
to add "Record-Route" header? to sip-query
"Record-Route" header ?already exists.

>>Regis M: 99% of the time 30 seconds hangup (or 32seconds) means a NAT
problem... or a response not come back to source...
I understand, but I don`t know how fix it.

Jurijs Ivolga,
I analize packets in ngrep.
First BYE packet include reason: NORMAL_CLEARING. BYE packet for caller
include reason: ACK Timeout.
About NORMAL_CLEARING in freeswitch documentation (
https://wiki.freeswitch.org/wiki/Hangup_Causes):
This cause indicates that the call is being cleared because one of the
users involved in the call has requested that the call be cleared. Under
normal situations, the source of this cause is not the network.

But phone clients do not send this command.

I suppose that this is because the server has not received confirmation from
one of the customers that the call took place.

All packages on call:
#
U 172.31.0.169:5060 -> 172.31.22.124:5060
INVITE sip:7906*******@sip0.MY_SIP_DOMAIN.com SIP/2.0.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Max-Forwards: 69.
Contact: <sip:8 at 85.
*.*.4:53712;ob;transport=tls>;+sip.instance="<urn:uuid:E458598F-FBED-B020-670D-167CE2ADB38A>".
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>.
From: "8"<sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER,
SUBSCRIBE, INFO.
Content-Type: application/sdp.
Supported: replaces, outbound, path.
User-Agent: PortSIP SDK for IOS.
Content-Length: 219.
.
v=0.
o=- 1460361915 1 IN IP4 85.*.*.4.
s=portsip.com.
c=IN IP4 52.*.*.177.
t=0 0.
m=audio 40772 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=sendrecv.
a=nortpproxy:yes.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Content-Length: 0.
.

#
U 172.31.22.124:5060 -> 178.*.*.12:5060
INVITE sip:7906*******@freelycall.com SIP/2.0.
Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK5vNr86BpDFNSN.
Max-Forwards: 67.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844894 INVITE.
Contact: <sip:gw+freelycall.com at 52.*.*.198:5060;transport=udp;gw=
freelycall.com>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 244.
X-FS-Support: update_display,send_info.
Remote-Party-ID: "8" <sip:8 at freelycall.com
>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1460338046 1460338047 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 23870 RTP/AVP 8 101 13.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
U 178.*.*.12:5060 -> 172.31.22.124:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP
52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844894 INVITE.
Server: Asterisk PBX 11.11.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces.
Contact: <sip:7906*******@178.*.*.12:5060>.
Content-Length: 0.
.

#
U 178.*.*.12:5060 -> 172.31.22.124:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP
52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844894 INVITE.
Server: Asterisk PBX 11.11.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces.
Contact: <sip:7906*******@178.*.*.12:5060>.
Content-Type: application/sdp.
Content-Length: 240.
.
v=0.
o=root 1395806664 1395806664 IN IP4 178.*.*.12.
s=Asterisk PBX 11.11.0.
c=IN IP4 178.*.*.12.
t=0 0.
m=audio 11574 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 183 Session Progress.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 178.*.*.12:5060 -> 172.31.22.124:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844894 INVITE.
Server: Asterisk PBX 11.11.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces.
Contact: <sip:7906*******@178.*.*.12:5060>.
Content-Type: application/sdp.
Content-Length: 240.
.
v=0.
o=root 1395806664 1395806664 IN IP4 178.*.*.12.
s=Asterisk PBX 11.11.0.
c=IN IP4 178.*.*.12.
t=0 0.
m=audio 11574 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.

#
U 172.31.22.124:5060 -> 178.*.*.12:5060
ACK sip:7906*******@178.*.*.12:5060 SIP/2.0.
Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK65eHa2vSarBcH.
Max-Forwards: 70.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844894 ACK.
Contact: <sip:gw+freelycall.com at 52.*.*.198:5060;transport=udp;gw=
freelycall.com>.
Content-Length: 0.
.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 172.31.22.124:5060 -> 172.31.0.169:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
Via: SIP/2.0/TLS 192.168.0.110:10977
;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 2 INVITE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 220.
Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
>;party=calling;privacy=off;screen=no.
.
v=0.
o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
s=FreeSWITCH.
c=IN IP4 52.*.*.198.
t=0 0.
m=audio 27728 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.

#
U 172.31.22.124:5060 -> 178.*.*.12:5060
BYE sip:7906*******@178.*.*.12:5060 SIP/2.0.
Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK8Q12Dry049QHr.
Max-Forwards: 70.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844895 BYE.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.

#
U 178.*.*.12:5060 -> 172.31.22.124:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
52.*.*.198;branch=z9hG4bK8Q12Dry049QHr;received=52.*.*.198;rport=5060.
From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
CSeq: 89844895 BYE.
Server: Asterisk PBX 11.11.0.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE.
Supported: replaces.
Content-Length: 0.
.

#
U 172.31.22.124:5060 -> 52.*.*.177:5060
BYE sip:8 at 85.*.*.4:53712;ob;transport=tls SIP/2.0.
Via: SIP/2.0/UDP 172.31.22.124;rport;branch=z9hG4bK7e89BXDX701yc.
Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
Max-Forwards: 70.
From: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
To: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 89844917 BYE.
Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Reason: SIP;cause=408;text="ACK Timeout".
Content-Length: 0.
.

#
U 52.*.*.177:5060 -> 172.31.22.124:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP
172.31.22.124;received=52.*.*.198;rport=5060;branch=z9hG4bK7e89BXDX701yc.
Contact: <sip:8 at 192.168.0.110:10977
;ob;transport=tls>;+sip.instance="<urn:uuid:E458598F-FBED-B020-670D-167CE2ADB38A>".
To: "8"<sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
From: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
CSeq: 89844917 BYE.
User-Agent: PortSIP SDK for IOS.
Content-Length: 0.
.




2016-04-08 20:01 GMT+03:00 Artur Mega <findmeinwland at gmail.com>:

> ?Do you use sip-proxy? If you use sip proxy, maybe you forget to add
> "Record-Route" header? to sip-query
>
> 2016-04-08 21:13 GMT+05:00 Regis M <regis.freeswitch.org at tornad.net>:
>
>> 99% of the time 30 seconds hangup (or 32seconds) means a NAT problem...
>> or a response not come back to source...
>>
>> 2016-04-08 17:06 GMT+02:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:
>>
>>> Hi,
>>>
>>> This is not what I need, please use ngrep:
>>>
>>>
>>> http://jonathanmanning.com/2009/11/17/how-to-sip-capture-using-ngrep-debug-sip-packets/
>>>
>>> With kind regards,
>>>
>>> Jurijs
>>>
>>> On Fri, Apr 8, 2016 at 6:02 PM, ???? ??????? <stasan89 at gmail.com> wrote:
>>>
>>>> Yes, of cause. I hide some ip and real phone numbers.
>>>> 178.*.*.12 - ip of provider.
>>>>
>>>> *On start call:*
>>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
>>>> sofia/external/8 at sip0.MY_DOMAIN.com
>>>> [c618eafe-fd98-11e5-a353-831849fc41a3]
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_NEW
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:9248 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com receiving invite from 172.31.0.169:5060 version:
>>>> 1.6.6  64bit
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com entering state [received][100]
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6770 Remote SDP:
>>>> v=0
>>>> o=- 1460126829 1 IN IP4 85.*.*.4
>>>> s=portsip.com
>>>> c=IN IP4 52.*.*.177
>>>> t=0 0
>>>> m=audio 40082 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=nortpproxy:yes
>>>>
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:7125 (sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com) State Change CS_NEW -> CS_INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:492
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State NEW
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_INIT ->
>>>> CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_channel.c:2247
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change DOWN -> RINGING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
>>>> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
>>>> <8>->7906******* in context public
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>> [public->from_opensips] continue=false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
>>>> [from_opensips] network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/
>>>> break=on-false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>>> transfer(${destination_number} XML default)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
>>>> CS_EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com transfer(7906******* XML
>>>> default)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr.c:2085 (sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com) State Change CS_EXECUTE -> CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_ivr.c:2092 Transfer
>>>> sofia/external/8 at sip0.MY_DOMAIN.com to XML[7906*******@default]
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
>>>> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
>>>> <8>->7906******* in context default
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>> [default->unloop] continue=false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS) [unloop]
>>>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (FAIL) [unloop]
>>>> ${sip_looped_call}() =~ /^true$/ break=on-false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>> [default->tod_example] continue=true
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Date/Time Match (PASS)
>>>> [tod_example] break=on-false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action set(open=true)
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>> [default->outbound_calls_to_freelycall] continue=false
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
>>>> [outbound_calls_to_freelycall] destination_number(7906*******) =~ /^(.+)/
>>>> break=on-true
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>>> set(hangup_after_bridge=true)
>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>>> bridge(sofia/gateway/freelycall.com/7906*******)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
>>>> CS_EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com set(open=true)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET
>>>> sofia/external/8 at sip0.MY_DOMAIN.com [open]=[true]
>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com
>>>> set(hangup_after_bridge=true)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET
>>>> sofia/external/8 at sip0.MY_DOMAIN.com [hangup_after_bridge]=[true]
>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com bridge(sofia/gateway/
>>>> freelycall.com/7906*******)
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:2128 Parsing
>>>> global variables
>>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
>>>> sofia/external/7906******* [c619366c-fd98-11e5-a35c-831849fc41a3]
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:4776
>>>> (sofia/external/7906*******) State Change CS_NEW -> CS_INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>> (sofia/external/7906*******) State INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88
>>>> sofia/external/7906******* SOFIA INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia_glue.c:1257
>>>> sofia/external/7906******* sending invite version: 1.6.6  64bit
>>>> Local SDP:
>>>> v=0
>>>> o=FreeSWITCH 1460100428 1460100429 IN IP4 52.*.*.198
>>>> s=FreeSWITCH
>>>> c=IN IP4 52.*.*.198
>>>> t=0 0
>>>> m=audio 26402 RTP/AVP 8 101 13
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=rtpmap:13 CN/8000
>>>> a=ptime:20
>>>> a=sendrecv
>>>>
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
>>>> sofia/external/7906******* Standard INIT
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
>>>> (sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>> (sofia/external/7906*******) State INIT going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel
>>>> sofia/external/7906******* entering state [calling][0]
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/7906*******) State ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141
>>>> sofia/external/7906******* SOFIA ROUTING
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:67
>>>> (sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>> (sofia/external/7906*******) State ROUTING going to sleep
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
>>>> (sofia/external/7906*******) State CONSUME_MEDIA
>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
>>>> (sofia/external/7906*******) State CONSUME_MEDIA going to sleep
>>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel
>>>> sofia/external/7906******* entering state [proceeding][183]
>>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6770 Remote SDP:
>>>> v=0
>>>> o=root 153112258 153112258 IN IP4 178.*.*.12
>>>> s=Asterisk PBX 11.11.0
>>>> c=IN IP4 178.*.*.12
>>>> t=0 0
>>>> m=audio 17362 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>>
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
>>>> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
>>>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
>>>> telephone-event payload to 101 at 8000
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
>>>> sofia/external/7906******* PCMA/8000 20 ms 160 samples 64000 bits 1 channels
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111
>>>> sofia/external/7906******* Original read codec set to PCMA:8
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
>>>> telephone-event payload to 101 at 8000
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
>>>> sofia/external/7906******* Set 2833 dtmf send payload to 101 recv payload
>>>> to 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
>>>> [sofia/external/7906*******] 172.31.22.124 port 26402 -> 178.*.*.12 port
>>>> 17362 codec: 8 ms: 20
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer
>>>> [soft] 160 bytes per 20ms
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
>>>> sofia/external/7906******* Set 2833 dtmf send payload to 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
>>>> sofia/external/7906******* Set 2833 dtmf receive payload to 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
>>>> sofia/external/7906******* Set rtp dtmf delay to 40
>>>> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
>>>> sofia/external/7906*******!
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468
>>>> (sofia/external/7906*******) Callstate Change DOWN -> EARLY
>>>> 2016-04-08 10:47:17.443282 [INFO] switch_ivr_originate.c:3557 Sending
>>>> early media
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio Codec
>>>> Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio Codec
>>>> Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
>>>> telephone-event payload to 101 at 8000
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
>>>> sofia/external/8 at sip0.MY_DOMAIN.com PCMA/8000 20 ms 160 samples 64000
>>>> bits 1 channels
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Original read codec set to PCMA:8
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
>>>> telephone-event payload to 101 at 8000
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
>>>> recv payload to 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
>>>> [sofia/external/8 at sip0.MY_DOMAIN.com] 172.31.22.124 port 30630 ->
>>>> 52.*.*.177 port 40082 codec: 8 ms: 20
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer
>>>> [soft] 160 bytes per 20ms
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf receive payload to
>>>> 101
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set rtp dtmf delay to 40
>>>> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
>>>> sofia/external/8 at sip0.MY_DOMAIN.com!
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change RINGING -> EARLY
>>>> 2016-04-08 10:47:17.443282 [DEBUG] mod_sofia.c:2330 Ring SDP:
>>>> v=0
>>>> o=FreeSWITCH 1460096207 1460096208 IN IP4 52.*.*.198
>>>> s=FreeSWITCH
>>>> c=IN IP4 52.*.*.198
>>>> t=0 0
>>>> m=audio 30630 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>> a=sendrecv
>>>>
>>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com entering state [early][183]
>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_ivr_originate.c:3608
>>>> Originate Resulted in Success: [sofia/external/7906*******]
>>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_ivr_bridge.c:1591
>>>> (sofia/external/7906*******) State Change CS_CONSUME_MEDIA ->
>>>> CS_EXCHANGE_MEDIA
>>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_EXCHANGE_MEDIA
>>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:542
>>>> (sofia/external/7906*******) State EXCHANGE_MEDIA
>>>> 2016-04-08 10:47:17.463254 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
>>>> 2016-04-08 10:47:17.503264 [DEBUG] switch_rtp.c:6654 Correct audio
>>>> ip/port confirmed.
>>>> 2016-04-08 10:47:17.663261 [DEBUG] switch_rtp.c:6654 Correct audio
>>>> ip/port confirmed.
>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
>>>> sofia/external/7906******* entering state [completing][200]
>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6767 Duplicate SDP
>>>> v=0
>>>> o=root 153112258 153112258 IN IP4 178.*.*.12
>>>> s=Asterisk PBX 11.11.0
>>>> c=IN IP4 178.*.*.12
>>>> t=0 0
>>>> m=audio 17362 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>>
>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
>>>> sofia/external/7906******* entering state [ready][200]
>>>> 2016-04-08 10:47:21.323279 [NOTICE] sofia.c:7665 Channel
>>>> [sofia/external/7906*******] has been answered
>>>> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767
>>>> (sofia/external/7906*******) Callstate Change EARLY -> ACTIVE
>>>> 2016-04-08 10:47:21.323279 [DEBUG] mod_sofia.c:799 Local SDP
>>>> sofia/external/8 at sip0.MY_DOMAIN.com:
>>>> v=0
>>>> o=FreeSWITCH 1460096207 1460096209 IN IP4 52.*.*.198
>>>> s=FreeSWITCH
>>>> c=IN IP4 52.*.*.198
>>>> t=0 0
>>>> m=audio 30630 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>> a=sendrecv
>>>>
>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com entering state [completed][200]
>>>> 2016-04-08 10:47:21.323279 [NOTICE] switch_ivr_bridge.c:616 Channel
>>>> [sofia/external/8 at sip0.MY_DOMAIN.com] has been answered
>>>> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change EARLY -> ACTIVE
>>>> 2016-04-08 10:47:21.383279 [DEBUG] switch_rtp.c:6654 Correct audio
>>>> ip/port confirmed.
>>>> 2016-04-08 10:47:21.423259 [DEBUG] switch_rtp.c:6654 Correct audio
>>>> ip/port confirmed.
>>>>
>>>>
>>>> *And after 30 seconds:*
>>>> 2016-04-08 10:47:53.343283 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com entering state [terminating][0]
>>>> 2016-04-08 10:47:53.343283 [NOTICE] sofia.c:7779 Hangup sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:699
>>>> sofia/external/8 at sip0.MY_DOMAIN.com ending bridge by request from
>>>> write function
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE
>>>> THREAD DONE [sofia/external/7906*******]
>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_ivr_bridge.c:881 Hangup
>>>> sofia/external/7906******* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:542
>>>> (sofia/external/7906*******) State EXCHANGE_MEDIA going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
>>>> (sofia/external/7906*******) Callstate Change ACTIVE -> HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>> (sofia/external/7906*******) State HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel
>>>> sofia/external/7906******* hanging up, cause: NORMAL_CLEARING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:484 Sending BYE to
>>>> sofia/external/7906*******
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/7906******* Standard HANGUP, cause: NORMAL_CLEARING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>> (sofia/external/7906*******) State HANGUP going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
>>>> (sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/7906*******) Running State Change CS_REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>> (sofia/external/7906*******) State REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
>>>> sofia/external/7906******* Standard REPORTING, cause: NORMAL_CLEARING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>> (sofia/external/7906*******) State REPORTING going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
>>>> (sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 2
>>>> (sofia/external/7906*******) Locked, Waiting on external entities
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:705
>>>> sofia/external/8 at sip0.MY_DOMAIN.com ending bridge by request from read
>>>> function
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE
>>>> THREAD DONE [sofia/external/8 at sip0.MY_DOMAIN.com]
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:1692
>>>> sofia/external/8 at sip0.MY_DOMAIN.com skip receive message [UNBRIDGE]
>>>> (channel is hungup already)
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:2796
>>>> sofia/external/8 at sip0.MY_DOMAIN.com skip receive message
>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:539
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change ACTIVE -> HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP
>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel
>>>> sofia/external/8 at sip0.MY_DOMAIN.com hanging up, cause:
>>>> NORMAL_UNSPECIFIED
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard HANGUP, cause:
>>>> NORMAL_UNSPECIFIED
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_HANGUP ->
>>>> CS_REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard REPORTING, cause:
>>>> NORMAL_UNSPECIFIED
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING going to sleep
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_REPORTING ->
>>>> CS_DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session 1
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Locked, Waiting on external
>>>> entities
>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session
>>>> 1 (sofia/external/8 at sip0.MY_DOMAIN.com) Ended
>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
>>>> Channel sofia/external/8 at sip0.MY_DOMAIN.com [CS_DESTROY]
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341 sofia/external/
>>>> 8 at sip0.MY_DOMAIN.com SOFIA DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY going to sleep
>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session
>>>> 2 (sofia/external/7906*******) Ended
>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
>>>> Channel sofia/external/7906******* [CS_DESTROY]
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
>>>> (sofia/external/7906*******) Running State Change CS_DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>> (sofia/external/7906*******) State DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341
>>>> sofia/external/7906******* SOFIA DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
>>>> sofia/external/7906******* Standard DESTROY
>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>> (sofia/external/7906*******) State DESTROY going to sleep
>>>>
>>>>
>>>> 2016-04-08 17:37 GMT+03:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:
>>>>
>>>>> Hi,
>>>>>
>>>>> I would recommend you to capture SIP packets  during call  on
>>>>> Freeswitch server and send it here, I will take a look on it.
>>>>>
>>>>> With kind regards,
>>>>>
>>>>> Jurijs
>>>>>
>>>>> On Fri, Apr 8, 2016 at 5:34 PM, ???? ??????? <stasan89 at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> I already tried disabling timers, does not work.
>>>>>>
>>>>>> 2016-04-08 17:19 GMT+03:00 Oleg Stolyar <olegstolyar at gmail.com>:
>>>>>>
>>>>>>> Try disabling session timers in the sip profile.  I think that line
>>>>>>> is commented out by default, so uncomment it.
>>>>>>>
>>>>>>> <param name="enable-timer" value="false"/>
>>>>>>>
>>>>>>> On Fri, Apr 8, 2016 at 6:59 AM, ???? ??????? <stasan89 at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Hello.
>>>>>>>>
>>>>>>>> When using a call or conference through sip ? freeswitch with
>>>>>>>> external provider there is a problem ? the call is interrupted in 30
>>>>>>>> seconds. Though the sound goes all right.
>>>>>>>> I think that it caused by the NAT settings for freeswitch, but I
>>>>>>>> don't understand how to adjust it correctly.
>>>>>>>> At start of freeswitch I see the following mistakes in the tracking
>>>>>>>> data:
>>>>>>>> 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
>>>>>>>> 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>> PMP 1/5
>>>>>>>> 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>> PMP 2/5
>>>>>>>> 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>> PMP 3/5
>>>>>>>> 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>> PMP 4/5
>>>>>>>> 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>> PMP 5/5
>>>>>>>> 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking
>>>>>>>> for PMP [general error]
>>>>>>>> 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for
>>>>>>>> UPnP
>>>>>>>> 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP
>>>>>>>> NAT devices detected!
>>>>>>>>
>>>>>>>> Despite of this mistake, conference communication between two
>>>>>>>> internal users works normally. The problem arises at a call through
>>>>>>>> external provider.
>>>>>>>>
>>>>>>>> We have the following architecture:
>>>>>>>> In a cloud of Amazon EC2 there are 2 servers ? opensips and
>>>>>>>> freeswitch, both for NAT for external clients, but have an opportunity to
>>>>>>>> work with each other directly.
>>>>>>>> opensips has the internal address 172.31.0.169 and external 52.
>>>>>>>> *.*.177
>>>>>>>> freeswitch has the internal address 172.31.22.124 and external 52.
>>>>>>>> *.*.198
>>>>>>>>
>>>>>>>> In fact, freeswitch acts only for conferences, and is ready for use
>>>>>>>> of a remote DB on opensips.
>>>>>>>> The auto-nat settings by default didn't work. The problem is in the
>>>>>>>> external profile settings as far as I understand.
>>>>>>>>
>>>>>>>> I have filled and created the following configuration:
>>>>>>>> vars.xml
>>>>>>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto?/>
>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198?/> <!?
>>>>>>>> public freeswitch ip ?>
>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198?/> <!?
>>>>>>>> public freeswitch ip ?>
>>>>>>>>   <!-- External SIP Profile -->
>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>>>>>>   <X-PRE-PROCESS cmd="set"
>>>>>>>> data="external_ssl_dir=$${base_dir}/conf/tls"/>
>>>>>>>>
>>>>>>>> sip_profile/external.xml
>>>>>>>>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>>>>>>>>     <param name="sip-ip" value="$${local_ip_v4}"/>
>>>>>>>>
>>>>>>>>     <param name="ext-rtp-ip" value=?52.*.*.198?/> <!? public
>>>>>>>> freeswitch ip ?>
>>>>>>>>     <param name="ext-sip-ip" value=?52.*.*.198?/> <!? public
>>>>>>>> freeswitch ip ?>
>>>>>>>> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
>>>>>>>> ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
>>>>>>>> (that would be logical), but in that case conferences didn't work at all
>>>>>>>> and errors below appeared:
>>>>>>>> [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
>>>>>>>> Also I tried to put such configuration:
>>>>>>>>     <param name="rtp-ip" value="auto"/>
>>>>>>>>     <param name="sip-ip" value="52.*.*.198?/>
>>>>>>>> but it also hasn't helped to solve the problem.
>>>>>>>>
>>>>>>>> autoload_configs/switch.conf.xml
>>>>>>>>     <param name="rtp-start-port" value="16384"/>
>>>>>>>>     <param name="rtp-end-port" value="32768"/>
>>>>>>>>
>>>>>>>> "sofia status" looks as follows:
>>>>>>>>                      Name       Type
>>>>>>>>                                       Data    State
>>>>>>>>
>>>>>>>> =================================================================================================
>>>>>>>>             172.31.22.124      alias
>>>>>>>>                                   internal    ALIASED
>>>>>>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5060
>>>>>>>> RUNNING (0)
>>>>>>>>                  external    profile               sip:mod_sofia at 52.*.*.198:5061
>>>>>>>> RUNNING (0) (TLS)
>>>>>>>>  external::*********.com    gateway                      sip:USER@*********.com
>>>>>>>> REGED
>>>>>>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5080
>>>>>>>> RUNNING (0)
>>>>>>>>                  internal    profile               sip:mod_sofia at 52.*.*.198:5081
>>>>>>>> RUNNING (0) (TLS)
>>>>>>>>
>>>>>>>> =================================================================================================
>>>>>>>> 2 profiles 1 alias
>>>>>>>>
>>>>>>>> "sofia status profile external" looks as follows:
>>>>>>>>
>>>>>>>> =================================================================================================
>>>>>>>> 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               172.31.22.124
>>>>>>>> Ext-RTP-IP           52.*.*.198
>>>>>>>> SIP-IP               172.31.22.124
>>>>>>>> Ext-SIP-IP           52.*.*.198
>>>>>>>> URL                  sip:mod_sofia at 52.*.*.198:5060
>>>>>>>> BIND-URL             sip:mod_sofia at 52.
>>>>>>>> *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
>>>>>>>> TLS-URL              sip:mod_sofia at 52.*.*.198:5061
>>>>>>>> TLS-BIND-URL         sips:mod_sofia at 52.
>>>>>>>> *.*.198:5061;maddr=172.31.22.124;transport=tls
>>>>>>>> HOLD-MUSIC           local_stream://moh
>>>>>>>> OUTBOUND-PROXY       N/A
>>>>>>>> CODECS IN            PCMA
>>>>>>>> CODECS OUT           PCMA
>>>>>>>> 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
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> What do I adjust wrong? Whether there is some opportunity, to tell
>>>>>>>> freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
>
> 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 jurijs.ivolga at gmail.com  Mon Apr 11 16:04:44 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Mon, 11 Apr 2016 15:04:44 +0300
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
 configuring NAT on Amazon EC2
In-Reply-To: <CAC89h=KjuTnY5yoiHkUT6bvbFbDzt9imTCCZBsMHBmJ3bJi4pA@mail.gmail.com>
References: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>
	<CAC8JMOKX7wkSG_140qWMedyz56E5aHyFxG0jb5+87G9iG9qVzQ@mail.gmail.com>
	<CAC89h=L3a3ZzOvYYpq4Jma8JsN6+WmvUGkEmF+7mO4WvyNoGTg@mail.gmail.com>
	<CAOrA2UbnqXs49O18a3eeb-8YRvvxmOS7fPEORtyHEd57O_Q94w@mail.gmail.com>
	<CAC89h=KCcOx5ftG4etYsrV1unEp=BuB-hMBcHQrOzJ2OJvRBog@mail.gmail.com>
	<CAOrA2UbxHQ+9TJeX0gCErFMD1uRCmH9bDu76JQo7Rt_mBfqOkw@mail.gmail.com>
	<CAHWtr1TdBN2J61eFGjnPw6i633qwLRYd7Ti7mVs4uTWHAz4NxQ@mail.gmail.com>
	<CAOLEQ7fZ6FJH9v5wiKv87MFKihc+gCXSbc2-CUnQhw5xDXogJg@mail.gmail.com>
	<CAC89h=KjuTnY5yoiHkUT6bvbFbDzt9imTCCZBsMHBmJ3bJi4pA@mail.gmail.com>
Message-ID: <CAOrA2UZ=sE-RDZCg-CC-n-HBgErBKq4aWWFCR3z8N3uRA_GiZw@mail.gmail.com>

Hi Stas,

I assume:

UA => OpenSIPS => Freeswitch => VoIP provider

172.31.0.169:5060 - OpenSIPS
172.31.22.124:5060 - Freeswitch
178.*.*.12:5060 -  VoIP provider


>From log what you provided it looks like OpenSIPS is not configured
properly...

If you check RFC:

https://tools.ietf.org/html/rfc3665#section-3.1

Successful call looks like this:

 Alice                     Bob
     |                        |
     |       INVITE F1        |
     |----------------------->|
     |    180 Ringing F2      |
     |<-----------------------|
     |                        |
     |       200 OK F3        |
     |<-----------------------|
     |         ACK F4         |
     |----------------------->|
     |   Both Way RTP Media   |
     |<======================>|
     |                        |
     |         BYE F5         |
     |<-----------------------|
     |       200 OK F6        |
     |----------------------->|
     |                        |


In your case somehow Freeswitch never receive ACK from OpenSIPS.

As you can see following packet was sent several times by Freeswitch:

U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
And on this OK Freeswitch should receive ACK, but never receive it
from Opensips and because of this FreeSWITCH hang-up call:

U 172.31.22.124:5060 -> 52.*.*.177:5060
> BYE sip:8 at 85.*.*.4:53712;ob;transport=tls SIP/2.0.
> Via: SIP/2.0/UDP 172.31.22.124;rport;branch=z9hG4bK7e89BXDX701yc.
> Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> Max-Forwards: 70.
> From: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> To: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 89844917 BYE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Reason: SIP;cause=408;text="ACK Timeout".
> Content-Length: 0.
>
As you can see BYE is strange too... Bye should be sent back to
OpenSIPS, but not to 52.*.*.177:5060.

On this point I think you have incorrect Record-Route records:

Good example you can find below, please check 16.12.1.1.

https://www.ietf.org/rfc/rfc3261.txt

Somehow you have following record-route record:
Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr> but your
opensips never use 5061 port, it is using 5060, so something is
incorrect here...

On this point I think you have miss-configured OpenSIPS.

With kind regards,


Jurijs

On Mon, Apr 11, 2016 at 12:59 PM, ???? ??????? <stasan89 at gmail.com> wrote:

> >>Artur Mega: Do you use sip-proxy? If you use sip proxy, maybe you forget
> to add "Record-Route" header? to sip-query
> "Record-Route" header ?already exists.
>
> >>Regis M: 99% of the time 30 seconds hangup (or 32seconds) means a NAT
> problem... or a response not come back to source...
> I understand, but I don`t know how fix it.
>
> Jurijs Ivolga,
> I analize packets in ngrep.
> First BYE packet include reason: NORMAL_CLEARING. BYE packet for caller
> include reason: ACK Timeout.
> About NORMAL_CLEARING in freeswitch documentation (
> https://wiki.freeswitch.org/wiki/Hangup_Causes):
> This cause indicates that the call is being cleared because one of the
> users involved in the call has requested that the call be cleared. Under
> normal situations, the source of this cause is not the network.
>
> But phone clients do not send this command.
>
> I suppose that this is because the server has not received confirmation from
> one of the customers that the call took place.
>
> All packages on call:
> #
> U 172.31.0.169:5060 -> 172.31.22.124:5060
> INVITE sip:7906*******@sip0.MY_SIP_DOMAIN.com SIP/2.0.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Max-Forwards: 69.
> Contact: <sip:8 at 85.
> *.*.4:53712;ob;transport=tls>;+sip.instance="<urn:uuid:E458598F-FBED-B020-670D-167CE2ADB38A>".
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>.
> From: "8"<sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> REGISTER, SUBSCRIBE, INFO.
> Content-Type: application/sdp.
> Supported: replaces, outbound, path.
> User-Agent: PortSIP SDK for IOS.
> Content-Length: 219.
> .
> v=0.
> o=- 1460361915 1 IN IP4 85.*.*.4.
> s=portsip.com.
> c=IN IP4 52.*.*.177.
> t=0 0.
> m=audio 40772 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=sendrecv.
> a=nortpproxy:yes.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Content-Length: 0.
> .
>
> #
> U 172.31.22.124:5060 -> 178.*.*.12:5060
> INVITE sip:7906*******@freelycall.com SIP/2.0.
> Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK5vNr86BpDFNSN.
> Max-Forwards: 67.
> From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
> To: <sip:7906*******@freelycall.com>.
> Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
> CSeq: 89844894 INVITE.
> Contact: <sip:gw+freelycall.com at 52.*.*.198:5060;transport=udp;gw=
> freelycall.com>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 244.
> X-FS-Support: update_display,send_info.
> Remote-Party-ID: "8" <sip:8 at freelycall.com
> >;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1460338046 1460338047 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 23870 RTP/AVP 8 101 13.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
> #
> U 178.*.*.12:5060 -> 172.31.22.124:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP
> 52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.
> From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
> To: <sip:7906*******@freelycall.com>.
> Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
> CSeq: 89844894 INVITE.
> Server: Asterisk PBX 11.11.0.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE.
> Supported: replaces.
> Contact: <sip:7906*******@178.*.*.12:5060>.
> Content-Length: 0.
> .
>
> #
> U 178.*.*.12:5060 -> 172.31.22.124:5060
> SIP/2.0 183 Session Progress.
> Via: SIP/2.0/UDP
> 52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.
> From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
> To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
> Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
> CSeq: 89844894 INVITE.
> Server: Asterisk PBX 11.11.0.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE.
> Supported: replaces.
> Contact: <sip:7906*******@178.*.*.12:5060>.
> Content-Type: application/sdp.
> Content-Length: 240.
> .
> v=0.
> o=root 1395806664 1395806664 IN IP4 178.*.*.12.
> s=Asterisk PBX 11.11.0.
> c=IN IP4 178.*.*.12.
> t=0 0.
> m=audio 11574 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 183 Session Progress.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 178.*.*.12:5060 -> 172.31.22.124:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 52.*.*.198;branch=z9hG4bK5vNr86BpDFNSN;received=52.*.*.198;rport=5060.
> From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
> To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
> Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
> CSeq: 89844894 INVITE.
> Server: Asterisk PBX 11.11.0.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE.
> Supported: replaces.
> Contact: <sip:7906*******@178.*.*.12:5060>.
> Content-Type: application/sdp.
> Content-Length: 240.
> .
> v=0.
> o=root 1395806664 1395806664 IN IP4 178.*.*.12.
> s=Asterisk PBX 11.11.0.
> c=IN IP4 178.*.*.12.
> t=0 0.
> m=audio 11574 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
> a=sendrecv.
>
> #
> U 172.31.22.124:5060 -> 178.*.*.12:5060
> ACK sip:7906*******@178.*.*.12:5060 SIP/2.0.
> Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK65eHa2vSarBcH.
> Max-Forwards: 70.
> From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
> To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
> Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
> CSeq: 89844894 ACK.
> Contact: <sip:gw+freelycall.com at 52.*.*.198:5060;transport=udp;gw=
> freelycall.com>.
> Content-Length: 0.
> .
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 172.31.22.124:5060 -> 172.31.0.169:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP sip0.MY_SIP_DOMAIN.com:5060
> ;branch=z9hG4bK2d16.a6dca045.0;i=191;received=172.31.0.169.
> Via: SIP/2.0/TLS 192.168.0.110:10977
> ;received=85.*.*.4;branch=z9hG4bK-524287-1---a27bae4af7c48619;rport=53712.
> Record-Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Record-Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> From: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> To: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 2 INVITE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 220.
> Remote-Party-ID: "Outbound Call" <sip:7906*******@sip0.MY_SIP_DOMAIN.com
> >;party=calling;privacy=off;screen=no.
> .
> v=0.
> o=FreeSWITCH 1460334195 1460334196 IN IP4 52.*.*.198.
> s=FreeSWITCH.
> c=IN IP4 52.*.*.198.
> t=0 0.
> m=audio 27728 RTP/AVP 8 101.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=ptime:20.
>
> #
> U 172.31.22.124:5060 -> 178.*.*.12:5060
> BYE sip:7906*******@178.*.*.12:5060 SIP/2.0.
> Via: SIP/2.0/UDP 52.*.*.198;rport;branch=z9hG4bK8Q12Dry049QHr.
> Max-Forwards: 70.
> From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
> To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
> Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
> CSeq: 89844895 BYE.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> .
>
> #
> U 178.*.*.12:5060 -> 172.31.22.124:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 52.*.*.198;branch=z9hG4bK8Q12Dry049QHr;received=52.*.*.198;rport=5060.
> From: "8" <sip:21***@freelycall.com>;tag=52eDp9a81B1mg.
> To: <sip:7906*******@freelycall.com>;tag=as75cc44fb.
> Call-ID: f7a88b46-7a5e-1234-a88c-06bbc71f1ffb.
> CSeq: 89844895 BYE.
> Server: Asterisk PBX 11.11.0.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH, MESSAGE.
> Supported: replaces.
> Content-Length: 0.
> .
>
> #
> U 172.31.22.124:5060 -> 52.*.*.177:5060
> BYE sip:8 at 85.*.*.4:53712;ob;transport=tls SIP/2.0.
> Via: SIP/2.0/UDP 172.31.22.124;rport;branch=z9hG4bK7e89BXDX701yc.
> Route: <sip:sip0.MY_SIP_DOMAIN.com;r2=on;lr>.
> Route: <sip:172.31.0.169:5061;transport=tls;r2=on;lr>.
> Max-Forwards: 70.
> From: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> To: "8" <sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 89844917 BYE.
> Contact: <sip:*7906*******@52.*.*.198:5060;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.6.6~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Reason: SIP;cause=408;text="ACK Timeout".
> Content-Length: 0.
> .
>
> #
> U 52.*.*.177:5060 -> 172.31.22.124:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP
> 172.31.22.124;received=52.*.*.198;rport=5060;branch=z9hG4bK7e89BXDX701yc.
> Contact: <sip:8 at 192.168.0.110:10977
> ;ob;transport=tls>;+sip.instance="<urn:uuid:E458598F-FBED-B020-670D-167CE2ADB38A>".
> To: "8"<sip:8 at sip0.MY_SIP_DOMAIN.com>;tag=4c913b30.
> From: <sip:*7906*******@sip0.MY_SIP_DOMAIN.com>;tag=4SNmmet442a2m.
> Call-ID: tE-jiL0Ft5GdKtpQbcLLoA...
> CSeq: 89844917 BYE.
> User-Agent: PortSIP SDK for IOS.
> Content-Length: 0.
> .
>
>
>
>
> 2016-04-08 20:01 GMT+03:00 Artur Mega <findmeinwland at gmail.com>:
>
>> ?Do you use sip-proxy? If you use sip proxy, maybe you forget to add
>> "Record-Route" header? to sip-query
>>
>> 2016-04-08 21:13 GMT+05:00 Regis M <regis.freeswitch.org at tornad.net>:
>>
>>> 99% of the time 30 seconds hangup (or 32seconds) means a NAT problem...
>>> or a response not come back to source...
>>>
>>> 2016-04-08 17:06 GMT+02:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:
>>>
>>>> Hi,
>>>>
>>>> This is not what I need, please use ngrep:
>>>>
>>>>
>>>> http://jonathanmanning.com/2009/11/17/how-to-sip-capture-using-ngrep-debug-sip-packets/
>>>>
>>>> With kind regards,
>>>>
>>>> Jurijs
>>>>
>>>> On Fri, Apr 8, 2016 at 6:02 PM, ???? ??????? <stasan89 at gmail.com>
>>>> wrote:
>>>>
>>>>> Yes, of cause. I hide some ip and real phone numbers.
>>>>> 178.*.*.12 - ip of provider.
>>>>>
>>>>> *On start call:*
>>>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com
>>>>> [c618eafe-fd98-11e5-a353-831849fc41a3]
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_NEW
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:9248 sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com receiving invite from 172.31.0.169:5060 version:
>>>>> 1.6.6  64bit
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com entering state [received][100]
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6770 Remote SDP:
>>>>> v=0
>>>>> o=- 1460126829 1 IN IP4 85.*.*.4
>>>>> s=portsip.com
>>>>> c=IN IP4 52.*.*.177
>>>>> t=0 0
>>>>> m=audio 40082 RTP/AVP 8 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=nortpproxy:yes
>>>>>
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:7125 (sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com) State Change CS_NEW -> CS_INIT
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:492
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State NEW
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_INIT
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88 sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com SOFIA INIT
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard INIT
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_INIT ->
>>>>> CS_ROUTING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State INIT going to sleep
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_channel.c:2247
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change DOWN -> RINGING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
>>>>> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
>>>>> <8>->7906******* in context public
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>>> [public->from_opensips] continue=false
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
>>>>> [from_opensips] network_addr(172.31.0.169) =~ /^172\.31\.0\.169$/
>>>>> break=on-false
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>>>> transfer(${destination_number} XML default)
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
>>>>> CS_EXECUTE
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
>>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com transfer(7906******* XML
>>>>> default)
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr.c:2085 (sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com) State Change CS_EXECUTE -> CS_ROUTING
>>>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_ivr.c:2092 Transfer
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com to XML[7906*******@default]
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_ROUTING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141 sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com SOFIA ROUTING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:166
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard ROUTING
>>>>> 2016-04-08 10:47:10.503262 [INFO] mod_dialplan_xml.c:637 Processing 8
>>>>> <8>->7906******* in context default
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>>> [default->unloop] continue=false
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS) [unloop]
>>>>> ${unroll_loops}(true) =~ /^true$/ break=on-false
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (FAIL) [unloop]
>>>>> ${sip_looped_call}() =~ /^true$/ break=on-false
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>>> [default->tod_example] continue=true
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Date/Time Match (PASS)
>>>>> [tod_example] break=on-false
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action set(open=true)
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com parsing
>>>>> [default->outbound_calls_to_freelycall] continue=false
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Regex (PASS)
>>>>> [outbound_calls_to_freelycall] destination_number(7906*******) =~ /^(.+)/
>>>>> break=on-true
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>>>> set(hangup_after_bridge=true)
>>>>> Dialplan: sofia/external/8 at sip0.MY_DOMAIN.com Action
>>>>> bridge(sofia/gateway/freelycall.com/7906*******)
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:216
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_ROUTING ->
>>>>> CS_EXECUTE
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State ROUTING going to sleep
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_EXECUTE
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:539
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:196 sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com SOFIA EXECUTE
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:258
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard EXECUTE
>>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com set(open=true)
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com [open]=[true]
>>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com
>>>>> set(hangup_after_bridge=true)
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_dptools.c:1498 SET
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com [hangup_after_bridge]=[true]
>>>>> EXECUTE sofia/external/8 at sip0.MY_DOMAIN.com bridge(sofia/gateway/
>>>>> freelycall.com/7906*******)
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:2128 Parsing
>>>>> global variables
>>>>> 2016-04-08 10:47:10.503262 [NOTICE] switch_channel.c:1101 New Channel
>>>>> sofia/external/7906******* [c619366c-fd98-11e5-a35c-831849fc41a3]
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:4776
>>>>> (sofia/external/7906*******) State Change CS_NEW -> CS_INIT
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/7906*******) Running State Change CS_INIT
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>>> (sofia/external/7906*******) State INIT
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:88
>>>>> sofia/external/7906******* SOFIA INIT
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia_glue.c:1257
>>>>> sofia/external/7906******* sending invite version: 1.6.6  64bit
>>>>> Local SDP:
>>>>> v=0
>>>>> o=FreeSWITCH 1460100428 1460100429 IN IP4 52.*.*.198
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 52.*.*.198
>>>>> t=0 0
>>>>> m=audio 26402 RTP/AVP 8 101 13
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=rtpmap:13 CN/8000
>>>>> a=ptime:20
>>>>> a=sendrecv
>>>>>
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:40
>>>>> sofia/external/7906******* Standard INIT
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:48
>>>>> (sofia/external/7906*******) State Change CS_INIT -> CS_ROUTING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:516
>>>>> (sofia/external/7906*******) State INIT going to sleep
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/7906*******) Running State Change CS_ROUTING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] sofia.c:6760 Channel
>>>>> sofia/external/7906******* entering state [calling][0]
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>>> (sofia/external/7906*******) State ROUTING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] mod_sofia.c:141
>>>>> sofia/external/7906******* SOFIA ROUTING
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_ivr_originate.c:67
>>>>> (sofia/external/7906*******) State Change CS_ROUTING -> CS_CONSUME_MEDIA
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:532
>>>>> (sofia/external/7906*******) State ROUTING going to sleep
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/7906*******) Running State Change CS_CONSUME_MEDIA
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
>>>>> (sofia/external/7906*******) State CONSUME_MEDIA
>>>>> 2016-04-08 10:47:10.503262 [DEBUG] switch_core_state_machine.c:551
>>>>> (sofia/external/7906*******) State CONSUME_MEDIA going to sleep
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel
>>>>> sofia/external/7906******* entering state [proceeding][183]
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6770 Remote SDP:
>>>>> v=0
>>>>> o=root 153112258 153112258 IN IP4 178.*.*.12
>>>>> s=Asterisk PBX 11.11.0
>>>>> c=IN IP4 178.*.*.12
>>>>> t=0 0
>>>>> m=audio 17362 RTP/AVP 8 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=ptime:20
>>>>>
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio
>>>>> Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio
>>>>> Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
>>>>> telephone-event payload to 101 at 8000
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
>>>>> sofia/external/7906******* PCMA/8000 20 ms 160 samples 64000 bits 1 channels
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111
>>>>> sofia/external/7906******* Original read codec set to PCMA:8
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
>>>>> telephone-event payload to 101 at 8000
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
>>>>> sofia/external/7906******* Set 2833 dtmf send payload to 101 recv payload
>>>>> to 101
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
>>>>> [sofia/external/7906*******] 172.31.22.124 port 26402 -> 178.*.*.12 port
>>>>> 17362 codec: 8 ms: 20
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer
>>>>> [soft] 160 bytes per 20ms
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
>>>>> sofia/external/7906******* Set 2833 dtmf send payload to 101
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
>>>>> sofia/external/7906******* Set 2833 dtmf receive payload to 101
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
>>>>> sofia/external/7906******* Set rtp dtmf delay to 40
>>>>> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
>>>>> sofia/external/7906*******!
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468
>>>>> (sofia/external/7906*******) Callstate Change DOWN -> EARLY
>>>>> 2016-04-08 10:47:17.443282 [INFO] switch_ivr_originate.c:3557 Sending
>>>>> early media
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4161 Audio
>>>>> Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4216 Audio
>>>>> Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4077 Set
>>>>> telephone-event payload to 101 at 8000
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:2906 Set Codec
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com PCMA/8000 20 ms 160 samples 64000
>>>>> bits 1 channels
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_codec.c:111
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Original read codec set to PCMA:8
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4429 Set
>>>>> telephone-event payload to 101 at 8000
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:4485
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
>>>>> recv payload to 101
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6033 AUDIO RTP
>>>>> [sofia/external/8 at sip0.MY_DOMAIN.com] 172.31.22.124 port 30630 ->
>>>>> 52.*.*.177 port 40082 codec: 8 ms: 20
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_rtp.c:3802 Starting timer
>>>>> [soft] 160 bytes per 20ms
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6332
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf send payload to 101
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6339
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set 2833 dtmf receive payload to
>>>>> 101
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_core_media.c:6362
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Set rtp dtmf delay to 40
>>>>> 2016-04-08 10:47:17.443282 [NOTICE] sofia_media.c:92 Pre-Answer
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com!
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_channel.c:3468
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change RINGING ->
>>>>> EARLY
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] mod_sofia.c:2330 Ring SDP:
>>>>> v=0
>>>>> o=FreeSWITCH 1460096207 1460096208 IN IP4 52.*.*.198
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 52.*.*.198
>>>>> t=0 0
>>>>> m=audio 30630 RTP/AVP 8 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=ptime:20
>>>>> a=sendrecv
>>>>>
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com entering state [early][183]
>>>>> 2016-04-08 10:47:17.443282 [DEBUG] switch_ivr_originate.c:3608
>>>>> Originate Resulted in Success: [sofia/external/7906*******]
>>>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_ivr_bridge.c:1591
>>>>> (sofia/external/7906*******) State Change CS_CONSUME_MEDIA ->
>>>>> CS_EXCHANGE_MEDIA
>>>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/7906*******) Running State Change CS_EXCHANGE_MEDIA
>>>>> 2016-04-08 10:47:17.463254 [DEBUG] switch_core_state_machine.c:542
>>>>> (sofia/external/7906*******) State EXCHANGE_MEDIA
>>>>> 2016-04-08 10:47:17.463254 [DEBUG] mod_sofia.c:613 SOFIA EXCHANGE_MEDIA
>>>>> 2016-04-08 10:47:17.503264 [DEBUG] switch_rtp.c:6654 Correct audio
>>>>> ip/port confirmed.
>>>>> 2016-04-08 10:47:17.663261 [DEBUG] switch_rtp.c:6654 Correct audio
>>>>> ip/port confirmed.
>>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
>>>>> sofia/external/7906******* entering state [completing][200]
>>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6767 Duplicate SDP
>>>>> v=0
>>>>> o=root 153112258 153112258 IN IP4 178.*.*.12
>>>>> s=Asterisk PBX 11.11.0
>>>>> c=IN IP4 178.*.*.12
>>>>> t=0 0
>>>>> m=audio 17362 RTP/AVP 8 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=ptime:20
>>>>>
>>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel
>>>>> sofia/external/7906******* entering state [ready][200]
>>>>> 2016-04-08 10:47:21.323279 [NOTICE] sofia.c:7665 Channel
>>>>> [sofia/external/7906*******] has been answered
>>>>> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767
>>>>> (sofia/external/7906*******) Callstate Change EARLY -> ACTIVE
>>>>> 2016-04-08 10:47:21.323279 [DEBUG] mod_sofia.c:799 Local SDP
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com:
>>>>> v=0
>>>>> o=FreeSWITCH 1460096207 1460096209 IN IP4 52.*.*.198
>>>>> s=FreeSWITCH
>>>>> c=IN IP4 52.*.*.198
>>>>> t=0 0
>>>>> m=audio 30630 RTP/AVP 8 101
>>>>> a=rtpmap:8 PCMA/8000
>>>>> a=rtpmap:101 telephone-event/8000
>>>>> a=fmtp:101 0-16
>>>>> a=ptime:20
>>>>> a=sendrecv
>>>>>
>>>>> 2016-04-08 10:47:21.323279 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com entering state [completed][200]
>>>>> 2016-04-08 10:47:21.323279 [NOTICE] switch_ivr_bridge.c:616 Channel
>>>>> [sofia/external/8 at sip0.MY_DOMAIN.com] has been answered
>>>>> 2016-04-08 10:47:21.323279 [DEBUG] switch_channel.c:3767
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change EARLY -> ACTIVE
>>>>> 2016-04-08 10:47:21.383279 [DEBUG] switch_rtp.c:6654 Correct audio
>>>>> ip/port confirmed.
>>>>> 2016-04-08 10:47:21.423259 [DEBUG] switch_rtp.c:6654 Correct audio
>>>>> ip/port confirmed.
>>>>>
>>>>>
>>>>> *And after 30 seconds:*
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] sofia.c:6760 Channel sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com entering state [terminating][0]
>>>>> 2016-04-08 10:47:53.343283 [NOTICE] sofia.c:7779 Hangup sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com [CS_EXECUTE] [NORMAL_UNSPECIFIED]
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:699
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com ending bridge by request from
>>>>> write function
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE
>>>>> THREAD DONE [sofia/external/7906*******]
>>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_ivr_bridge.c:881 Hangup
>>>>> sofia/external/7906******* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:542
>>>>> (sofia/external/7906*******) State EXCHANGE_MEDIA going to sleep
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/7906*******) Running State Change CS_HANGUP
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
>>>>> (sofia/external/7906*******) Callstate Change ACTIVE -> HANGUP
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>>> (sofia/external/7906*******) State HANGUP
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel
>>>>> sofia/external/7906******* hanging up, cause: NORMAL_CLEARING
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:484 Sending BYE to
>>>>> sofia/external/7906*******
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
>>>>> sofia/external/7906******* Standard HANGUP, cause: NORMAL_CLEARING
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>>> (sofia/external/7906*******) State HANGUP going to sleep
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
>>>>> (sofia/external/7906*******) State Change CS_HANGUP -> CS_REPORTING
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/7906*******) Running State Change CS_REPORTING
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>>> (sofia/external/7906*******) State REPORTING
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
>>>>> sofia/external/7906******* Standard REPORTING, cause: NORMAL_CLEARING
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>>> (sofia/external/7906*******) State REPORTING going to sleep
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
>>>>> (sofia/external/7906*******) State Change CS_REPORTING -> CS_DESTROY
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session
>>>>> 2 (sofia/external/7906*******) Locked, Waiting on external entities
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:705
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com ending bridge by request from
>>>>> read function
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:778 BRIDGE
>>>>> THREAD DONE [sofia/external/8 at sip0.MY_DOMAIN.com]
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_ivr_bridge.c:1692
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com skip receive message [UNBRIDGE]
>>>>> (channel is hungup already)
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:2796
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com skip receive message
>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:539
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State EXECUTE going to sleep
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_HANGUP
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:739
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Callstate Change ACTIVE ->
>>>>> HANGUP
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:431 Channel
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com hanging up, cause:
>>>>> NORMAL_UNSPECIFIED
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:60
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard HANGUP, cause:
>>>>> NORMAL_UNSPECIFIED
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:741
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State HANGUP going to sleep
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:508
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_HANGUP ->
>>>>> CS_REPORTING
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:473
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change
>>>>> CS_REPORTING
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:104
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard REPORTING, cause:
>>>>> NORMAL_UNSPECIFIED
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:827
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State REPORTING going to sleep
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:499
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State Change CS_REPORTING ->
>>>>> CS_DESTROY
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_session.c:1646 Session
>>>>> 1 (sofia/external/8 at sip0.MY_DOMAIN.com) Locked, Waiting on external
>>>>> entities
>>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session
>>>>> 1 (sofia/external/8 at sip0.MY_DOMAIN.com) Ended
>>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
>>>>> Channel sofia/external/8 at sip0.MY_DOMAIN.com [CS_DESTROY]
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) Running State Change CS_DESTROY
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341 sofia/external/
>>>>> 8 at sip0.MY_DOMAIN.com SOFIA DESTROY
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
>>>>> sofia/external/8 at sip0.MY_DOMAIN.com Standard DESTROY
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>>> (sofia/external/8 at sip0.MY_DOMAIN.com) State DESTROY going to sleep
>>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1664 Session
>>>>> 2 (sofia/external/7906*******) Ended
>>>>> 2016-04-08 10:47:53.343283 [NOTICE] switch_core_session.c:1668 Close
>>>>> Channel sofia/external/7906******* [CS_DESTROY]
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:630
>>>>> (sofia/external/7906*******) Running State Change CS_DESTROY
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>>> (sofia/external/7906*******) State DESTROY
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] mod_sofia.c:341
>>>>> sofia/external/7906******* SOFIA DESTROY
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:111
>>>>> sofia/external/7906******* Standard DESTROY
>>>>> 2016-04-08 10:47:53.343283 [DEBUG] switch_core_state_machine.c:640
>>>>> (sofia/external/7906*******) State DESTROY going to sleep
>>>>>
>>>>>
>>>>> 2016-04-08 17:37 GMT+03:00 Jurijs Ivolga <jurijs.ivolga at gmail.com>:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> I would recommend you to capture SIP packets  during call  on
>>>>>> Freeswitch server and send it here, I will take a look on it.
>>>>>>
>>>>>> With kind regards,
>>>>>>
>>>>>> Jurijs
>>>>>>
>>>>>> On Fri, Apr 8, 2016 at 5:34 PM, ???? ??????? <stasan89 at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> I already tried disabling timers, does not work.
>>>>>>>
>>>>>>> 2016-04-08 17:19 GMT+03:00 Oleg Stolyar <olegstolyar at gmail.com>:
>>>>>>>
>>>>>>>> Try disabling session timers in the sip profile.  I think that line
>>>>>>>> is commented out by default, so uncomment it.
>>>>>>>>
>>>>>>>> <param name="enable-timer" value="false"/>
>>>>>>>>
>>>>>>>> On Fri, Apr 8, 2016 at 6:59 AM, ???? ??????? <stasan89 at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> Hello.
>>>>>>>>>
>>>>>>>>> When using a call or conference through sip ? freeswitch with
>>>>>>>>> external provider there is a problem ? the call is interrupted in 30
>>>>>>>>> seconds. Though the sound goes all right.
>>>>>>>>> I think that it caused by the NAT settings for freeswitch, but I
>>>>>>>>> don't understand how to adjust it correctly.
>>>>>>>>> At start of freeswitch I see the following mistakes in the
>>>>>>>>> tracking data:
>>>>>>>>> 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT
>>>>>>>>> 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>>> PMP 1/5
>>>>>>>>> 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>>> PMP 2/5
>>>>>>>>> 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>>> PMP 3/5
>>>>>>>>> 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>>> PMP 4/5
>>>>>>>>> 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for
>>>>>>>>> PMP 5/5
>>>>>>>>> 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking
>>>>>>>>> for PMP [general error]
>>>>>>>>> 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for
>>>>>>>>> UPnP
>>>>>>>>> 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP
>>>>>>>>> NAT devices detected!
>>>>>>>>>
>>>>>>>>> Despite of this mistake, conference communication between two
>>>>>>>>> internal users works normally. The problem arises at a call through
>>>>>>>>> external provider.
>>>>>>>>>
>>>>>>>>> We have the following architecture:
>>>>>>>>> In a cloud of Amazon EC2 there are 2 servers ? opensips and
>>>>>>>>> freeswitch, both for NAT for external clients, but have an opportunity to
>>>>>>>>> work with each other directly.
>>>>>>>>> opensips has the internal address 172.31.0.169 and external 52.
>>>>>>>>> *.*.177
>>>>>>>>> freeswitch has the internal address 172.31.22.124 and external 52.
>>>>>>>>> *.*.198
>>>>>>>>>
>>>>>>>>> In fact, freeswitch acts only for conferences, and is ready for
>>>>>>>>> use of a remote DB on opensips.
>>>>>>>>> The auto-nat settings by default didn't work. The problem is in
>>>>>>>>> the external profile settings as far as I understand.
>>>>>>>>>
>>>>>>>>> I have filled and created the following configuration:
>>>>>>>>> vars.xml
>>>>>>>>>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto?/>
>>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198?/> <!?
>>>>>>>>> public freeswitch ip ?>
>>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198?/> <!?
>>>>>>>>> public freeswitch ip ?>
>>>>>>>>>   <!-- External SIP Profile -->
>>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/>
>>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
>>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/>
>>>>>>>>>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>>>>>>>>>   <X-PRE-PROCESS cmd="set"
>>>>>>>>> data="external_ssl_dir=$${base_dir}/conf/tls"/>
>>>>>>>>>
>>>>>>>>> sip_profile/external.xml
>>>>>>>>>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>>>>>>>>>     <param name="sip-ip" value="$${local_ip_v4}"/>
>>>>>>>>>
>>>>>>>>>     <param name="ext-rtp-ip" value=?52.*.*.198?/> <!? public
>>>>>>>>> freeswitch ip ?>
>>>>>>>>>     <param name="ext-sip-ip" value=?52.*.*.198?/> <!? public
>>>>>>>>> freeswitch ip ?>
>>>>>>>>> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and
>>>>>>>>> ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server
>>>>>>>>> (that would be logical), but in that case conferences didn't work at all
>>>>>>>>> and errors below appeared:
>>>>>>>>> [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ...
>>>>>>>>> Also I tried to put such configuration:
>>>>>>>>>     <param name="rtp-ip" value="auto"/>
>>>>>>>>>     <param name="sip-ip" value="52.*.*.198?/>
>>>>>>>>> but it also hasn't helped to solve the problem.
>>>>>>>>>
>>>>>>>>> autoload_configs/switch.conf.xml
>>>>>>>>>     <param name="rtp-start-port" value="16384"/>
>>>>>>>>>     <param name="rtp-end-port" value="32768"/>
>>>>>>>>>
>>>>>>>>> "sofia status" looks as follows:
>>>>>>>>>                      Name       Type
>>>>>>>>>                                       Data    State
>>>>>>>>>
>>>>>>>>> =================================================================================================
>>>>>>>>>             172.31.22.124      alias
>>>>>>>>>                                   internal    ALIASED
>>>>>>>>>                  external    profile
>>>>>>>>> sip:mod_sofia at 52.*.*.198:5060    RUNNING (0)
>>>>>>>>>                  external    profile
>>>>>>>>> sip:mod_sofia at 52.*.*.198:5061    RUNNING (0) (TLS)
>>>>>>>>>  external::*********.com    gateway                      sip:USER@*********.com
>>>>>>>>> REGED
>>>>>>>>>                  internal    profile
>>>>>>>>> sip:mod_sofia at 52.*.*.198:5080    RUNNING (0)
>>>>>>>>>                  internal    profile
>>>>>>>>> sip:mod_sofia at 52.*.*.198:5081    RUNNING (0) (TLS)
>>>>>>>>>
>>>>>>>>> =================================================================================================
>>>>>>>>> 2 profiles 1 alias
>>>>>>>>>
>>>>>>>>> "sofia status profile external" looks as follows:
>>>>>>>>>
>>>>>>>>> =================================================================================================
>>>>>>>>> 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               172.31.22.124
>>>>>>>>> Ext-RTP-IP           52.*.*.198
>>>>>>>>> SIP-IP               172.31.22.124
>>>>>>>>> Ext-SIP-IP           52.*.*.198
>>>>>>>>> URL                  sip:mod_sofia at 52.*.*.198:5060
>>>>>>>>> BIND-URL             sip:mod_sofia at 52.
>>>>>>>>> *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp
>>>>>>>>> TLS-URL              sip:mod_sofia at 52.*.*.198:5061
>>>>>>>>> TLS-BIND-URL         sips:mod_sofia at 52.
>>>>>>>>> *.*.198:5061;maddr=172.31.22.124;transport=tls
>>>>>>>>> HOLD-MUSIC           local_stream://moh
>>>>>>>>> OUTBOUND-PROXY       N/A
>>>>>>>>> CODECS IN            PCMA
>>>>>>>>> CODECS OUT           PCMA
>>>>>>>>> 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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> What do I adjust wrong? Whether there is some opportunity, to tell
>>>>>>>>> freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>>
>> 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
>
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From 35633 at heb.be  Mon Apr 11 16:25:58 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Mon, 11 Apr 2016 14:25:58 +0200
Subject: [Freeswitch-users] Freeswitch and opensips
Message-ID: <CAFSzUtb8hyWgY+sugSsUDd5BNRDVsVTCX5NUkqzq505e9+0rig@mail.gmail.com>

I'm using Opensips as load balancer in front of FreeSwitch. To tell
Freeswitch that calls come from Opensips proxy, do i have to create a new
external profile in sip_profiles directory or add an extension in
dialplan/public.xml or both of two?

Second question, in this file
:/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml , i read that i
have to specify the CIDR, is the ip address  of Opensips?

Thank you



NDUWAYEZU Joselyne

-- 
Haute ?cole de Bruxelles
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From dimitry.nagorny at robot5.de  Mon Apr 11 18:44:30 2016
From: dimitry.nagorny at robot5.de (Nagorny, Dimitry)
Date: Mon, 11 Apr 2016 14:44:30 +0000
Subject: [Freeswitch-users] FreeSWITCH malforming SIP Package
Message-ID: <6b1322f737984885b795b820a6387306@r5prod-exchange.robot5.de>

Good afternoon Ladies and Gentlemen,

I'm using FreeSWITCH (1.7.0 git cae2553 2015-12-11 00:23:48Z 64bit) as B2BUA between Kamailio and our own Server. Our Server is attached to FreeSWICH via ESL.
>From Kamailio into our own Server it is connecting w/o problems. But when I initiate a call from our Server to Kamailio via FreeSWITCH I've a weird reaction:

Network-Overview:
PSTN(192.168.1.30) --> Kamailio(192.168.1.150/10.250.5.74) --> FreeSWITCH(10.250.5.72) --> Our Server(10.250.5.67)

1st Initial Invite to FS:
INVITE sip:7 at 10.250.5.72 SIP/2.0
   Via: SIP/2.0/UDP 10.250.5.17:5090;branch=z9hG4bK.GSRDySx-y;rport
   From: <sip:1001 at 10.250.5.72>;tag=nDaJDf3lz
   To: sip:7 at 10.250.5.72
   Route: <sip:10.250.5.72:5080;lr>
   Content-Type: application/sdp

7 is mapped to go to our Server and our Server then is trying to connect to a UA (2031) at PSTN via FreeSWITCH gateway(Kamailio):
INVITE sip:2031 at 10.250.5.74:5070 SIP/2.0
   Via: SIP/2.0/UDP 10.250.5.72:5070;rport;branch=z9hG4bKmgrc4De2Q0ySN
   From: "" <sip:1001@[10.250.5.72:5070]>;tag=U5jj715ZB6maK
   To: <sip:2031 at 10.250.5.74:5070>

This Invite was built by FreeSWITCH and I could confirm that the "SIP-Invite" sent via ESL to FreeSWITCH from our Server was correct but as you can see here FreeSWITCH "malformed" the From-Header with the square brackets and due to that I'm getting error from PSTN later:

SIP/2.0 400 Bad Request
   Warning: 399 192.168.1.30 "Malformed headers : From "
   To: <sip:2031 at 10.250.5.74:5070>;tag=ae0f6cb7ca85c1fb958bd8bed1f9b7b7

Here some config data that is set in FreeSWITCH:
Vars.xml:
  <X-PRE-PROCESS cmd="set" data="domain=10.250.5.72"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5070"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5080"/>

Sip-profiles/external/testpbx.xml:
<include>
  <gateway name="testpbx">
  <param name="realm" value="10.250.5.74:5070"/>
  <param name="from-domain" value="10.250.5.72:5070"/>
  <param name="proxy" value="10.250.5.74:5070"/>
  ...
  </gateway>
</include>

Sip-profiles/external.xml and internal.xml:
    <param name="rtp-ip" value="10.250.5.72"/>
    <param name="sip-ip" value="10.250.5.72"/>
    <param name="ext-rtp-ip" value="10.250.5.72"/>
    <param name="ext-sip-ip" value="10.250.5.72"/>

Can anyone help why FreeSWITCH is doing this or can point to a class where the the From Header is built? Maybe someone knows a solution?
Here you can find full SIP-Trace Logs and Full FreeSWITCH Debugging Log: https://we.tl/FtPuknRrSw


Very Respectfully
Dimitry Nagorny
Trainee

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From brian at freeswitch.org  Mon Apr 11 19:04:19 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Apr 2016 10:04:19 -0500
Subject: [Freeswitch-users] FreeSWITCH malforming SIP Package
In-Reply-To: <6b1322f737984885b795b820a6387306@r5prod-exchange.robot5.de>
References: <6b1322f737984885b795b820a6387306@r5prod-exchange.robot5.de>
Message-ID: <CAEJMVkAYtxeKV3+x6JUQb+cCBUjavjjpJnpyPDd919yyQL+OWQ@mail.gmail.com>

You don't a port in those settings, it's triggering IPv6 code.

On Monday, April 11, 2016, Nagorny, Dimitry <dimitry.nagorny at robot5.de>
wrote:

> Good afternoon Ladies and Gentlemen,
>
>
>
> I?m using FreeSWITCH (1.7.0 git cae2553 2015-12-11 00:23:48Z 64bit) as
> B2BUA between Kamailio and our own Server. Our Server is attached to
> FreeSWICH via ESL.
>
> From Kamailio into our own Server it is connecting w/o problems. But when
> I initiate a call from our Server to Kamailio via FreeSWITCH I?ve a weird
> reaction:
>
>
>
> Network-Overview:
>
> PSTN(192.168.1.30) ? Kamailio(192.168.1.150/10.250.5.74) ?
> FreeSWITCH(10.250.5.72) ? Our Server(10.250.5.67)
>
>
>
> 1st Initial Invite to FS:
>
> INVITE sip:7 at 10.250.5.72
> <javascript:_e(%7B%7D,'cvml','sip:7 at 10.250.5.72');> SIP/2.0
>
>    Via: SIP/2.0/UDP 10.250.5.17:5090;branch=z9hG4bK.GSRDySx-y;rport
>
>    From: <sip:1001 at 10.250.5.72
> <javascript:_e(%7B%7D,'cvml','sip:1001 at 10.250.5.72');>>;tag=nDaJDf3lz
>
>    To: sip:7 at 10.250.5.72
> <javascript:_e(%7B%7D,'cvml','sip:7 at 10.250.5.72');>
>
>    Route: <sip:10.250.5.72:5080;lr>
>
>    Content-Type: application/sdp
>
>
>
> 7 is mapped to go to our Server and our Server then is trying to connect
> to a UA (2031) at PSTN via FreeSWITCH gateway(Kamailio):
>
> INVITE sip:2031 at 10.250.5.74:5070 SIP/2.0
>
>    Via: SIP/2.0/UDP 10.250.5.72:5070;rport;branch=z9hG4bKmgrc4De2Q0ySN
>
>    From: "" <sip:1001@*[10.250.5.72:5070 <http://10.250.5.72:5070>]*
> >;tag=U5jj715ZB6maK
>
>    To: <sip:2031 at 10.250.5.74:5070>
>
>
>
> This Invite was built by FreeSWITCH and I could confirm that the
> ?SIP-Invite? sent via ESL to FreeSWITCH from our Server was correct but as
> you can see here FreeSWITCH ?malformed? the From-Header with the *square
> brackets* and due to that I?m getting error from PSTN later:
>
>
>
> SIP/2.0 400 Bad Request
>
>    Warning: 399 192.168.1.30 "Malformed headers : From "
>
>    To: <sip:2031 at 10.250.5.74:5070>;tag=ae0f6cb7ca85c1fb958bd8bed1f9b7b7
>
>
>
> Here some config data that is set in FreeSWITCH:
>
> Vars.xml:
>
>   <X-PRE-PROCESS cmd="set" data="domain=10.250.5.72"/>
>
>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5070"/>
>
>   <X-PRE-PROCESS cmd="set" data="internal_sip_port=5080"/>
>
>
>
> Sip-profiles/external/testpbx.xml:
>
> <include>
>
>   <gateway name="testpbx">
>
>   <param name="realm" value="10.250.5.74:5070"/>
>
>   <param name="from-domain" value="10.250.5.72:5070"/>
>
>   <param name="proxy" value="10.250.5.74:5070"/>
>
>   ?
>
>   </gateway>
>
> </include>
>
>
>
> Sip-profiles/external.xml and internal.xml:
>
>     <param name="rtp-ip" value="10.250.5.72"/>
>
>     <param name="sip-ip" value="10.250.5.72"/>
>
>     <param name="ext-rtp-ip" value="10.250.5.72"/>
>
>     <param name="ext-sip-ip" value="10.250.5.72"/>
>
>
>
> Can anyone help why FreeSWITCH is doing this or can point to a class where
> the the From Header is built? Maybe someone knows a solution?
>
> Here you can find full SIP-Trace Logs and Full FreeSWITCH Debugging Log:
> https://we.tl/FtPuknRrSw
>
>
>
>
>
> Very Respectfully
>
> *Dimitry Nagorny*
>
> Trainee
>
>
>


-- 

*Brian West*
brian at freeswitch.org


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

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From stasan89 at gmail.com  Mon Apr 11 19:13:30 2016
From: stasan89 at gmail.com (=?UTF-8?B?0KHRgtCw0YEg0KLQtdC70YzQvdC+0LI=?=)
Date: Mon, 11 Apr 2016 18:13:30 +0300
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
 configuring NAT on Amazon EC2
In-Reply-To: <CAOrA2UZ=sE-RDZCg-CC-n-HBgErBKq4aWWFCR3z8N3uRA_GiZw@mail.gmail.com>
References: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>
	<CAC8JMOKX7wkSG_140qWMedyz56E5aHyFxG0jb5+87G9iG9qVzQ@mail.gmail.com>
	<CAC89h=L3a3ZzOvYYpq4Jma8JsN6+WmvUGkEmF+7mO4WvyNoGTg@mail.gmail.com>
	<CAOrA2UbnqXs49O18a3eeb-8YRvvxmOS7fPEORtyHEd57O_Q94w@mail.gmail.com>
	<CAC89h=KCcOx5ftG4etYsrV1unEp=BuB-hMBcHQrOzJ2OJvRBog@mail.gmail.com>
	<CAOrA2UbxHQ+9TJeX0gCErFMD1uRCmH9bDu76JQo7Rt_mBfqOkw@mail.gmail.com>
	<CAHWtr1TdBN2J61eFGjnPw6i633qwLRYd7Ti7mVs4uTWHAz4NxQ@mail.gmail.com>
	<CAOLEQ7fZ6FJH9v5wiKv87MFKihc+gCXSbc2-CUnQhw5xDXogJg@mail.gmail.com>
	<CAC89h=KjuTnY5yoiHkUT6bvbFbDzt9imTCCZBsMHBmJ3bJi4pA@mail.gmail.com>
	<CAOrA2UZ=sE-RDZCg-CC-n-HBgErBKq4aWWFCR3z8N3uRA_GiZw@mail.gmail.com>
Message-ID: <CAC89h=J3Nw_h9B2xe0ZHydNm2r4ivALHue1SX1gWrDNaBhXPxg@mail.gmail.com>

Hi Jurijs,

> I assume:
> UA => OpenSIPS => Freeswitch => VoIP provider
>
> 172.31.0.169:5060 - OpenSIPS
> 172.31.22.124:5060 - Freeswitch
> 178.*.*.12:5060 -  VoIP provider
>

Yes, it is correct.

As you can see BYE is strange too... Bye should be sent back to
OpenSIPS, but not to 52.*.*.177:5060.
>
> It is external sip IP. 172.31.0.16 <http://172.31.0.169:5060> - it is IP
in amazon local network; 52.*.*.177==sip0.MY_SIP_DOMAIN.com - it is public
IP and domain of opensips.

Trouble was in opensips ports configures. On clients available only 5061
(tls) port and disablied 5060 port. In opensips server enabled 5060 and
5061 ports and opensips send request to freeswitch by 5060 port.
Aftrer opensips send 200 OK command and ACK to client by 5060 port, but on
clients available only 5061 port and packets dont delivered.

Thanks for help, a programmer who does not have experience in servers
administration and VOIP-telephony is not easy to quickly deal with the fact
that there is. But with your help, I managed to make it work.
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From pstarzyk at general-devices.com  Mon Apr 11 19:19:39 2016
From: pstarzyk at general-devices.com (Piotr Starzyk)
Date: Mon, 11 Apr 2016 11:19:39 -0400
Subject: [Freeswitch-users] Best way to record each leg of a call separately?
Message-ID: <438ebfce0662b9cc0af3e4b29c769677@mail.gmail.com>

I?m trying to record phone calls, and phone conferences, so that later on
their audio can be reviewed.  The requirement is for each leg of the call
to be recorded separately.  After doing some googling, one way of doing it,
is by using record_session and setting RECORD_STEREO to true.  That will
result in caller and receiver audio streams being placed in separate
channels.  At that point you could split the channels to get the individual
streams.



Is that the recommended approach, or is there a better way (or module) to
record legs separately?



Also, am I correct assuming the recording is done on FreeSWITCH server?
Does that mean that each SIP client will need to send two audio streams out
(one to the other client and one to freeswitch for recording?).



Thanks.
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From jurijs.ivolga at gmail.com  Mon Apr 11 19:19:46 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Mon, 11 Apr 2016 18:19:46 +0300
Subject: [Freeswitch-users] Call is interrupted in 30 seconds,
 configuring NAT on Amazon EC2
In-Reply-To: <CAC89h=J3Nw_h9B2xe0ZHydNm2r4ivALHue1SX1gWrDNaBhXPxg@mail.gmail.com>
References: <CAC89h=L1EmskzrmbFT+TNaXMdM84vY2Yz-zt1KpUxtfpFmz9SA@mail.gmail.com>
	<CAC8JMOKX7wkSG_140qWMedyz56E5aHyFxG0jb5+87G9iG9qVzQ@mail.gmail.com>
	<CAC89h=L3a3ZzOvYYpq4Jma8JsN6+WmvUGkEmF+7mO4WvyNoGTg@mail.gmail.com>
	<CAOrA2UbnqXs49O18a3eeb-8YRvvxmOS7fPEORtyHEd57O_Q94w@mail.gmail.com>
	<CAC89h=KCcOx5ftG4etYsrV1unEp=BuB-hMBcHQrOzJ2OJvRBog@mail.gmail.com>
	<CAOrA2UbxHQ+9TJeX0gCErFMD1uRCmH9bDu76JQo7Rt_mBfqOkw@mail.gmail.com>
	<CAHWtr1TdBN2J61eFGjnPw6i633qwLRYd7Ti7mVs4uTWHAz4NxQ@mail.gmail.com>
	<CAOLEQ7fZ6FJH9v5wiKv87MFKihc+gCXSbc2-CUnQhw5xDXogJg@mail.gmail.com>
	<CAC89h=KjuTnY5yoiHkUT6bvbFbDzt9imTCCZBsMHBmJ3bJi4pA@mail.gmail.com>
	<CAOrA2UZ=sE-RDZCg-CC-n-HBgErBKq4aWWFCR3z8N3uRA_GiZw@mail.gmail.com>
	<CAC89h=J3Nw_h9B2xe0ZHydNm2r4ivALHue1SX1gWrDNaBhXPxg@mail.gmail.com>
Message-ID: <CAOrA2UYKYbV8N7REYSDdNWZHbUwyTs0Xe+j10ebGWAZRrxbMyg@mail.gmail.com>

Hi Stas,

You are welcome.

Keep in mind, that in your set-up freeswitch should send back BYE to
internal ip(172.31.0.16), it may even work with external IP(52.*.*.177),
but it will be incorrect set-up...

With kind regards,

Jurijs

On Mon, Apr 11, 2016 at 6:13 PM, ???? ??????? <stasan89 at gmail.com> wrote:

> Hi Jurijs,
>
>> I assume:
>> UA => OpenSIPS => Freeswitch => VoIP provider
>>
>> 172.31.0.169:5060 - OpenSIPS
>> 172.31.22.124:5060 - Freeswitch
>> 178.*.*.12:5060 -  VoIP provider
>>
>
> Yes, it is correct.
>
> As you can see BYE is strange too... Bye should be sent back to OpenSIPS, but not to 52.*.*.177:5060.
>>
>> It is external sip IP. 172.31.0.16 <http://172.31.0.169:5060> - it is IP
> in amazon local network; 52.*.*.177==sip0.MY_SIP_DOMAIN.com - it is
> public IP and domain of opensips.
>
> Trouble was in opensips ports configures. On clients available only 5061
> (tls) port and disablied 5060 port. In opensips server enabled 5060 and
> 5061 ports and opensips send request to freeswitch by 5060 port.
> Aftrer opensips send 200 OK command and ACK to client by 5060 port, but on
> clients available only 5061 port and packets dont delivered.
>
> Thanks for help, a programmer who does not have experience in servers
> administration and VOIP-telephony is not easy to quickly deal with the
> fact that there is. But with your help, I managed to make it work.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 11 21:27:13 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Apr 2016 12:27:13 -0500
Subject: [Freeswitch-users] Best way to record each leg of a call
	separately?
In-Reply-To: <438ebfce0662b9cc0af3e4b29c769677@mail.gmail.com>
References: <438ebfce0662b9cc0af3e4b29c769677@mail.gmail.com>
Message-ID: <CAEJMVkALdwGPbUHOxrTrHRwFib4UhZ-HDMEJEgJnrVTr=Qn50g@mail.gmail.com>

I would use stereo in this manner.

/b


On Mon, Apr 11, 2016 at 10:19 AM, Piotr Starzyk <
pstarzyk at general-devices.com> wrote:

> I?m trying to record phone calls, and phone conferences, so that later on
> their audio can be reviewed.  The requirement is for each leg of the call
> to be recorded separately.  After doing some googling, one way of doing it,
> is by using record_session and setting RECORD_STEREO to true.  That will
> result in caller and receiver audio streams being placed in separate
> channels.  At that point you could split the channels to get the individual
> streams.
>
>
>
> Is that the recommended approach, or is there a better way (or module) to
> record legs separately?
>
>
>
> Also, am I correct assuming the recording is done on FreeSWITCH server?
> Does that mean that each SIP client will need to send two audio streams out
> (one to the other client and one to freeswitch for recording?).
>
>
>
> Thanks.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From italo at freeswitch.org  Mon Apr 11 21:38:37 2016
From: italo at freeswitch.org (=?utf-8?q?=C3=8Dtalo_Rossi?=)
Date: Mon, 11 Apr 2016 10:38:37 -0700 (PDT)
Subject: [Freeswitch-users] Best way to record each leg of a
 call	separately?
In-Reply-To: <CAEJMVkALdwGPbUHOxrTrHRwFib4UhZ-HDMEJEgJnrVTr=Qn50g@mail.gmail.com>
References: <438ebfce0662b9cc0af3e4b29c769677@mail.gmail.com>
	<CAEJMVkALdwGPbUHOxrTrHRwFib4UhZ-HDMEJEgJnrVTr=Qn50g@mail.gmail.com>
Message-ID: <56x3rmy6k9rsh1vkcvgx76qet-0@mailer.nylas.com>

RECORD_STEREO or RECORD_READ_ONLY/RECORD_WRITE_ONLY  
  
?talo Rossi

italo at freeswitch.org

IRC chat.freenode.net #freeswitch #freeswitch-dev

Bugs? https://freeswitch.org/jira

Docs? https://freeswitch.org/jira

Chat? https://hipchat.freeswitch.org/gUdAgy0m6

  

  

On Apr 11 2016, at 2:30 pm, Brian West &lt;brian at freeswitch.org&gt; wrote:  

> I would use stereo in this manner.

>

>  

>

> /b

>

>  

>

>  

>

> On Mon, Apr 11, 2016 at 10:19 AM, Piotr Starzyk &lt;[pstarzyk at general-
devices.com](mailto:pstarzyk at general-devices.com)&gt; wrote:  

>

>> I?m trying to record phone calls, and phone conferences, so that later on
their audio can be reviewed.  The requirement is for each leg of the call to
be recorded separately.  After doing some googling, one way of doing it, is by
using record_session and setting RECORD_STEREO to true.  That will result in
caller and receiver audio streams being placed in separate channels.  At that
point you could split the channels to get the individual streams.

>>

>>  
>>

>> Is that the recommended approach, or is there a better way (or module) to
record legs separately?

>>

>>  
>>

>> Also, am I correct assuming the recording is done on FreeSWITCH server?
Does that mean that each SIP client will need to send two audio streams out
(one to the other client and one to freeswitch for recording?).

>>

>>  
>>

>> Thanks.

>>

>>  
_________________________________________________________________________  
Professional FreeSWITCH Consulting Services:  
[consulting at freeswitch.org](mailto:consulting at freeswitch.org)  
<http://www.freeswitchsolutions.com>  
  
Official FreeSWITCH Sites  
<http://www.freeswitch.org>  
<http://confluence.freeswitch.org>  
<http://www.cluecon.com>  
  
FreeSWITCH-users mailing list  
[FreeSWITCH-users at lists.freeswitch.org](mailto:FreeSWITCH-
users at lists.freeswitch.org)  
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>  
UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>  
<http://www.freeswitch.org>  

>

>  
  

>

>  

>

> \--  

>

> **_Brian West_**  
[brian at freeswitch.org](mailto:brian at freeswitch.org)

>

> ![](http://billing.freeswitch.org/templates/default/img/whmcslogo.png)  

>

> **_Twitter: @FreeSWITCH , @briankwest_**  
<http://www.freeswitchbook.com>  
<http://www.freeswitchcookbook.com>  
<https://www.gofundme.com/freeswitch_ubuntu>

>

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

>

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

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From dimitry.nagorny at robot5.de  Mon Apr 11 19:29:14 2016
From: dimitry.nagorny at robot5.de (Nagorny, Dimitry)
Date: Mon, 11 Apr 2016 15:29:14 +0000
Subject: [Freeswitch-users] FreeSWITCH malforming SIP Package
In-Reply-To: <CAEJMVkAYtxeKV3+x6JUQb+cCBUjavjjpJnpyPDd919yyQL+OWQ@mail.gmail.com>
References: <6b1322f737984885b795b820a6387306@r5prod-exchange.robot5.de>
	<CAEJMVkAYtxeKV3+x6JUQb+cCBUjavjjpJnpyPDd919yyQL+OWQ@mail.gmail.com>
Message-ID: <e9c47d2775aa4422b271839edeb5bd68@r5prod-exchange.robot5.de>

Thank you very much Brian, that helped getting rid of the error. Now I?ve to find out how to tell FreeSWITCH to send gateway stuff on port 5070, because right now it keeps trying to send on 5060. Thanks again!


Best Regards
Dimitry Nagorny
Trainee

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian West
Gesendet: Montag, 11. April 2016 17:04
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] FreeSWITCH malforming SIP Package

You don't a port in those settings, it's triggering IPv6 code.

On Monday, April 11, 2016, Nagorny, Dimitry <dimitry.nagorny at robot5.de<mailto:dimitry.nagorny at robot5.de>> wrote:
Sip-profiles/external/testpbx.xml:
<include>
  <gateway name="testpbx">
  <param name="realm" value="10.250.5.74:5070<http://10.250.5.74:5070>"/>
  <param name="from-domain" value="10.250.5.72:5070<http://10.250.5.72:5070>"/>
  <param name="proxy" value="10.250.5.74:5070<http://10.250.5.74:5070>"/>
  ?
  </gateway>
</include>



--

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

[Das Bild wurde vom Absender entfernt.]

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

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

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

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From aqsyounas at gmail.com  Mon Apr 11 23:12:15 2016
From: aqsyounas at gmail.com (Aqs Younas)
Date: Tue, 12 Apr 2016 00:12:15 +0500
Subject: [Freeswitch-users] Freeswitch and opensips
In-Reply-To: <CAFSzUtb8hyWgY+sugSsUDd5BNRDVsVTCX5NUkqzq505e9+0rig@mail.gmail.com>
References: <CAFSzUtb8hyWgY+sugSsUDd5BNRDVsVTCX5NUkqzq505e9+0rig@mail.gmail.com>
Message-ID: <CAGVOjVXRaF5nDv6SdtbsM0vabXi1u+9ToDq_qv=evdt_U0dVqA@mail.gmail.com>

you need to do both things.

On 11 April 2016 at 17:25, Nduwayezu, Joselyne <35633 at heb.be> wrote:

> I'm using Opensips as load balancer in front of FreeSwitch. To tell
> Freeswitch that calls come from Opensips proxy, do i have to create a new
> external profile in sip_profiles directory or add an extension in
> dialplan/public.xml or both of two?
>
> Second question, in this file
> :/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml , i read that i
> have to specify the CIDR, is the ip address  of Opensips?
>
> Thank you
>
>
>
> NDUWAYEZU Joselyne
>
> Haute ?cole de Bruxelles
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 11 23:22:22 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 11 Apr 2016 14:22:22 -0500
Subject: [Freeswitch-users] Configure SDP
In-Reply-To: <CAFETSWj7_NHZi7g4SDxAJu+hJcmvY0VkO0a-ABQtG5E2WQQF5A@mail.gmail.com>
References: <CAFETSWiskTaDA3fFw4qbBo7h7CsXKjiFcY7EXF5CYLk917YQ0Q@mail.gmail.com>
	<3BD6E704-4227-4FB7-A12E-02AB326679F7@jerris.com>
	<CAFETSWj7_NHZi7g4SDxAJu+hJcmvY0VkO0a-ABQtG5E2WQQF5A@mail.gmail.com>
Message-ID: <E2AD75FA-C1C7-4E3B-A7A4-A0B4F2D6E428@jerris.com>

We have no way to trigger that for a stand alone outbound call.  We can do it when we get one inbound and are bridging outbound... thats proxy for 3pcc.  Do you need it for an outbound call thats not the result of an inbound 3pcc call?  If you want to add the feature to do it for a new outbound call you can contact consulting at freeswitch.org and we could quote a cost for that work.

Mike



> On Apr 9, 2016, at 7:04 PM, amani mansour <amani.mansour2 at gmail.com> wrote:
> 
> Hi mr ,
> thank you sir , after enabling 3pcc what i must do , because the INVITE message is with SDP .
> regards 
> Amani
> 
> 2016-04-04 15:44 GMT+01:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> you are looking for 3pcc
> 
> > On Apr 4, 2016, at 4:22 AM, amani mansour <amani.mansour2 at gmail.com <mailto:amani.mansour2 at gmail.com>> wrote:
> >
> > Good morning ,
> >
> > I need to configure a number to send invite without SDP , can anyone help me please ?

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From mike at jerris.com  Mon Apr 11 23:46:33 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 11 Apr 2016 14:46:33 -0500
Subject: [Freeswitch-users] question on handling of nonce count (nc)
In-Reply-To: <850AC07B-9D14-4C1A-94BF-19E96A2550E8@beachdognet.com>
References: <850AC07B-9D14-4C1A-94BF-19E96A2550E8@beachdognet.com>
Message-ID: <AE35A6CA-CC33-443C-A8A0-D03D2C12AEA8@jerris.com>

sounds like devices that are broken to me.  

> On Apr 10, 2016, at 2:23 AM, Dave Horton <daveh at beachdognet.com> wrote:
> 
> In investigating some REGISTER storms on one of my networks, I am seeing some client devices interacting with Freeswitch in a manner that can lead to excessive registration traffic.  
> It looks to me to be more of an endpoint problem than a freeswitch problem, but I would like confirmation of that as well as any ideas on how to handle this (i.e., throttle back this traffic).
> 
> The basic problem flow is this:
> 
> - Client sends a REGISTER with a large nc value and nonce value A
> - Freeswitch replies 401 with stale=true (nonce is stale) and nonce value B
> - Client sends another REGISTER with nc value incremented by 1 and nonce value A again
> - Freeswitch replies 401 with stale=true (nonce is stale) and nonce value C
> - Client sends another REGISTER with nc value incremented again and nonce value A again
> ?.etc.
> 
> This seems particularly problematic with some Yealink, Communicator, and Polycomm IP Soundlink endpoint
> 
> Here is a specific example (some information redacted)
> 
> recv 804 bytes from udp/[]:5060 at 23:54:11.906859:
>   ------------------------------------------------------------------------
>   REGISTER sip:x.x.x.x:5060 SIP/2.0
>   Authorization: Digest username="123371",realm="sip.foo.com",nonce="41adc443-57c8-4325-831e-ffd006a922d4",uri=?sip:x.x.x.x:6060",response="3a4b5f05ec1897a58865b4ba0cdb0b4d",cnonce="b5d06adf6a4c7c0592f5fc1d7766a605",nc=0000008a,qop=auth,algorithm=MD5
> 
> send 641 bytes to udp/[10.128.77.170]:5060 at 23:54:11.909722:
>   ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   WWW-Authenticate: Digest realm=sip.foo.com", nonce="888c8919-b28f-4be4-be12-753430aafa88", stale=true, algorithm=MD5, qop=?auth?
> 
> 
> recv 804 bytes from udp/[]:5060 at 23:54:12.007622:
>   ------------------------------------------------------------------------
>   REGISTER sip:x.x.x.x:5060 SIP/2.0
>   Authorization: Digest username="123371",realm="sip.foo.com",nonce="41adc443-57c8-4325-831e-ffd006a922d4",uri=?sip:x.x.x.x:6060",response="556498e38d27c944f10e3a0c11a5ea41",cnonce="5585e516afcf2f95bfbc4bef11a075ee",nc=0000008b,qop=auth,algorithm=MD5
> 
> send 641 bytes to udp/[10.128.77.170]:5060 at 23:54:12.010376:
>   ------------------------------------------------------------------------
>   SIP/2.0 401 Unauthorized
>   WWW-Authenticate: Digest realm=?sip.foo.com", nonce="1ff3b9a3-4cbb-4569-b6c7-7bee203547ac", stale=true, algorithm=MD5, qop="auth"
> 
> recv 804 bytes from udp/[10.128.77.170]:5060 at 23:54:12.108742:
>   ------------------------------------------------------------------------
>   REGISTER sip:x.x.x.x:5060 SIP/2.0
>   Authorization: Digest username="123371",realm="sip.foo.com",nonce="41adc443-57c8-4325-831e-ffd006a922d4",uri=?sip:x.x.x.x:6060",response="9cf2360ef5f28684e667ac878362d0c0",cnonce="9833d8d3889d3ae8875e0f6f00c4d3f3",nc=0000008c,qop=auth,algorithm=MD5
> 
> 
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 phenix at vfemail.net  Mon Apr 11 15:42:40 2016
From: phenix at vfemail.net (tanguy)
Date: Mon, 11 Apr 2016 06:42:40 -0500
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
In-Reply-To: <B6118ED6-AAA7-462D-B4CB-06D3713948A8@d-man.org>
References: <570A3728.6020900@vfemail.net>
	<CAFLo2Qzi5oSHzXMyFEiDExdzHsmew_2a+Wne_f1Z7H6PUNTe4A@mail.gmail.com>
	<B700FF78-C2BA-41C3-8E9C-AD49B658DD60@freeswitch.org>
	<CAFLo2QwD6eM6=B1TUa0ObWa9-3GLraVJxeu0UfaV6QS4zkeLmA@mail.gmail.com>
	<B6118ED6-AAA7-462D-B4CB-06D3713948A8@d-man.org>
Message-ID: <20160411064240.Horde.6M0V77qRE_MGQRkFnMD4-w1@www.vfemail.net>

> Sergey
Thank you for your piece of dialplan but i don't think it's the  
solution. You offer a different solution to set the nibble_rate  
variable, but in my sample case this variable is properly set because  
the call is billed.

> Luis
A bug, maybe. I don't remember my freeswitch version ( i can't connect  
to my servers yet ) but i think it's 1.4 series.

I can try the last 1.6 version, i will also try a standard freeswitch  
sample configuration ( without config files provided by fusionpbx )


Quoting Darren <d at d-man.org>:

> If you can point me to the pull request, I can take a look and see  
> if I can integrate it in.
>
> From:  
> <freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Luis Daniel Lucio Quiroz  
> <luis.daniel.lucio at gmail.com<mailto:luis.daniel.lucio at gmail.com>>
> Reply-To: FreeSWITCH Users Help  
> <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
> Date: Sunday, April 10, 2016 at 11:09 AM
> To: FreeSWITCH Users Help  
> <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] nibblebill do not terminate calls properly
>
>
> There is. It is not from myself. I just don't remember it
>
> Le 10 avr. 2016 1:44 PM, "Ken Rice"  
> <krice at freeswitch.org<mailto:krice at freeswitch.org>> a ?crit :
> If there is a known bug and a patch for this, there should be a jira  
> and a pull request for the patch
>
> Sent from my iPhone
>
> On Apr 10, 2016, at 12:40 PM, Luis Daniel Lucio Quiroz  
> <luis.daniel.lucio at gmail.com<mailto:luis.daniel.lucio at gmail.com>>  
> wrote:
>
>
> There is a known bug and a known patch to fix it. I can't remember  
> right now which one it is.
>
> Hello
> I would like to use nibblebilling for fraud prevention on  
> international and premium numbers ( national or emergency calls  
> should never be blocked and should still working even if the billing  
> database is unavailable  )
>
> Unfortunately that never hangup my calls using standard dial plan
>
> nibblebill.conf.xml
>
> <configuration name="nibblebill.conf" description="Nibble Billing">
>   <settings>
>     <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help  
> with these options -->
>
>     <!-- Information for connecting to your database -->
>     <param name="db_dsn" value="pgsql://hostaddr=127.0.0.1  
> dbname=xxxx user=xxxx password=xxxx"/>
>
>     <!-- The database table where your CASH column is located -->
>     <param name="db_table" value="accounts"/>
>
>     <!-- The column name where we store the value of the account -->
>     <param name="db_column_cash" value="cash"/>
>
>     <!-- The column name for the unique ID identifying the account -->
>     <param name="db_column_account" value="id"/>
>
>     <!-- Default heartbeat interval. Set to 'off' for no heartbeat  
> (i.e. bill only at end of call) -->
>     <param name="global_heartbeat" value="20"/>
>
>     <!-- By default, warn a caller when their balance is at $5.00.  
> You can set this to a negative number. -->
>     <param name="lowbal_amt" value="-100"/> <!-- i don't need this -->
>     <param name="lowbal_action" value="play ding"/>
>
>     <!-- By default, terminate a caller when their balance hits  
> $0.00. You can set this to a negative number. -->
>     <param name="nobal_amt" value="0"/>
>     <param name="nobal_action" value="hangup"/>
>
>     <!-- If a call goes beyond a certain dollar amount, flag or  
> terminate it -->
>     <param name="percall_max_amt" value="10"/>
>     <param name="percall_action" value="hangup"/>
>
>   </settings>
> </configuration>
>
> outbound route
>
> <extension name="international" >
>    <condition field="destination_number" expression="^00(\d{6,20})$" >
>        <action application="nibblebill" data="flush"/>
>         <action application="set" data="nibble_account=${accountcode}"/>
>         <action application="set" data="nibble_rate=0.1"/>
>        <action application="bridge" data="sofia/gateway/gw_idt/$1" />
>    </condition>
> </extension>
>
>
> My balance
> My balance is already negative
>
>  select * from accounts ;
>                 id                |   cash
> ----------------------------------+-----------
>  company.voip.mydomain.com<http://company.voip.mydomain.com> | -0.789471
>
> A sample call
> My balance already negative, the call will be billed but never blocked
>
> Dialplan:  
> sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> Action  
> nibblebill(flush)
> Dialplan:  
> sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> Action  
> set(nibble_account=${accountcode})
> Dialplan:  
> sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> Action  
> set(nibble_rate=0.1)
> EXECUTE  
> sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com>  
> nibblebill(flush)
> EXECUTE  
> sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com>  
> set(nibble_account=company.voip.domain.com<http://company.voip.domain.com>)
> 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477  
> sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> SET  
> [nibble_account]=[company.voip.domain.com<http://company.voip.domain.com>]
> EXECUTE  
> sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com>  
> set(nibble_rate=0.1)
> 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477  
> sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/5003 at company.voip.domain.com> SET  
> [nibble_rate]=[0.1]
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:488 Attempting  
> to bill at $0.1 per minute to account  
> company.voip.domain.com<http://company.voip.domain.com>
> 2016-04-10 12:10:52.132458 [INFO] mod_nibblebill.c:540 Beginning new  
> billing on 644488fc-ff04-11e5-9a27-fd2791153af9
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:546 42 seconds  
> passed since last bill time of 2016-04-10 12:10:09
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:563 Billing  
> $0.071033 to company.voip.domain.com<http://company.voip.domain.com>  
> (Call: / 0.000000 so far)
> 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:393 Doing update query
> 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:420 Doing lookup query
> [SELECT cash AS nibble_balance FROM accounts WHERE  
> id='company.voip.domain.com<http://company.voip.domain.com>']
> 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:428 Retrieved  
> current balance for account  
> company.voip.domain.com<http://company.voip.domain.com> (balance =  
> -0.860504)
>
>
> Using b-leg only
> Alternatively i tried an alternative dialplan ( Even if i don't  
> relay understand what is the meaning of b-leg billing  )
>
> <action application="bridge"  
> data="{enable_heartbeat_events=5,nibble_rate=0.1,nibble_account=${accountcode},originate_timeout=90}sofia/gateway/gw_idt/33$1"  
> />
>
> The result is better because the pending call is hanged up when the  
> balance reach 0.
>
> 2016-04-10 12:39:59.012462 [CRIT] mod_nibblebill.c:607 Balance of  
> -0.003237 fell below allowed amount of 0.000000! (Account  
> company.voip.domain.com<http://company.voip.domain.com>)
>
> But if i make a new call ( when my balance is negative )  , the  
> caller party is immediately hanged but this did not cancel  
> immediately the bridge:  The called party ring and can stay bridged  
> for 61 seconds after answer.
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




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From mario_fs at mgtech.com  Mon Apr 11 23:32:14 2016
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 11 Apr 2016 12:32:14 -0700
Subject: [Freeswitch-users] FreeSWITCH malforming SIP Package
In-Reply-To: <e9c47d2775aa4422b271839edeb5bd68@r5prod-exchange.robot5.de>
References: <6b1322f737984885b795b820a6387306@r5prod-exchange.robot5.de>
	<CAEJMVkAYtxeKV3+x6JUQb+cCBUjavjjpJnpyPDd919yyQL+OWQ@mail.gmail.com>
	<e9c47d2775aa4422b271839edeb5bd68@r5prod-exchange.robot5.de>
Message-ID: <08741874-5BE6-45D9-BB48-BD7F709F31EF@mgtech.com>

You just need to add (or replace the old line) in the settings section at the bottom for the testpbx profile:
<param name="sip-port" value="5070"/>
   

> On Apr 11, 2016, at 8:29 AM, Nagorny, Dimitry <dimitry.nagorny at robot5.de> wrote:
> 
> Thank you very much Brian, that helped getting rid of the error. Now I?ve to find out how to tell FreeSWITCH to send gateway stuff on port 5070, because right now it keeps trying to send on 5060. Thanks again!
>  
>  
> Best Regards
> Dimitry Nagorny
> Trainee
>  
> Von: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>[mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Brian West
> Gesendet: Montag, 11. April 2016 17:04
> An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Betreff: Re: [Freeswitch-users] FreeSWITCH malforming SIP Package
>  
> You don't a port in those settings, it's triggering IPv6 code.
> 
> On Monday, April 11, 2016, Nagorny, Dimitry <dimitry.nagorny at robot5.de <mailto:dimitry.nagorny at robot5.de>> wrote:
> Sip-profiles/external/testpbx.xml:
> <include>
>   <gateway name="testpbx">
>   <param name="realm" value="10.250.5.74:5070 <http://10.250.5.74:5070/>"/>
>   <param name="from-domain" value="10.250.5.72:5070 <http://10.250.5.72:5070/>"/>
>   <param name="proxy" value="10.250.5.74:5070 <http://10.250.5.74:5070/>"/>
>   ?
>   </gateway>
> </include>
>  
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> <~WRD000.jpg>
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
> https://www.gofundme.com/freeswitch_ubuntu <https://www.gofundme.com/freeswitch_ubuntu>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype: <skype:briankwest>briankwest <skype:briankwest>
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From pstarzyk at general-devices.com  Mon Apr 11 23:54:26 2016
From: pstarzyk at general-devices.com (Piotr Starzyk)
Date: Mon, 11 Apr 2016 15:54:26 -0400
Subject: [Freeswitch-users] Cannot get automatic call recording working.
Message-ID: <42b77762b2cff94364249083ee8113dd@mail.gmail.com>

I?m running FreeSWITCH in console mode on a Windows 7 machine, using
unmodified demo setup.  I?m having a hard time getting automatic call
recording to work.  I?m using the following instructions to record a call
(by modifying  Local_Extension dialplan):



https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+record_session

*Record Calls To Extensions*

To record all phone calls between extensions do the following. Make a
directory under freeswitch/recordings/archive/. Then edit Local_Extension
in dialplan in conf/dialplan/default.xml

<extension name="Local_Extension">

    <condition field="destination_number" expression="^(10[01][0-8])$">

Then add the following actions.

<action application="set" data="RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/>

<action application="set" data="RECORD_COPYRIGHT=(c) 2011"/>

<action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>

<action application="set" data="RECORD_ARTIST=FreeSWITCH"/>

<action application="set" data="RECORD_COMMENT=FreeSWITCH"/>

<action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>

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

<action application="record_session"
data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>





But when I place a call, no files are getting created in the recordings or
archive folder.  I tried calling both ways, still nothing.  I am running in
admin mode, and have firewalls disabled.



When looking through the logs, the only two lines relevant to the recording
are the following:



892ef279-18c7-4d8d-8795-fa1c366b68c7 EXECUTE sofia/internal/1000 at 10.10.10.77
 bind_meta_app(2 b s record_session::C:/Program
Files/FreeSWITCH/recordings/1000.2016-04-11-15-16-58.wav)

892ef279-18c7-4d8d-8795-fa1c366b68c7 2016-04-11 15:16:58.521344 [INFO]
switch_ivr_async.c:4152 Bound B-Leg: *2 record_session::C:/Program
Files/FreeSWITCH/recordings/1000.2016-04-11-15-16-58.wav



>From FreeSWITCH cookbook, I found out they allow manual recording by
pressing *2 on receiving end.  When I tested that, it did work, and the wav
file was created in the recordings directory, so the recording module seems
to be working.  But for some reason, I still can?t get the automatic
session recording to work.  What am I doing wrong?





Thanks.
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From mike at jerris.com  Mon Apr 11 23:55:17 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 11 Apr 2016 14:55:17 -0500
Subject: [Freeswitch-users] NATIVE SQL ERR [cannot commit - no
	transaction is active]
In-Reply-To: <CALR0W6_r=z0en7hYNeZYuqdZCvxN1kbmXZs51CKAx9WFTx6mqQ@mail.gmail.com>
References: <CALR0W68a6+iiimqtW6hyxVdKz8o8B1yg-P7tye0LjEmtajvOaw@mail.gmail.com>
	<CALR0W6_r=z0en7hYNeZYuqdZCvxN1kbmXZs51CKAx9WFTx6mqQ@mail.gmail.com>
Message-ID: <BF3451FF-D102-4F48-89F0-59FBE35553C9@jerris.com>

I have seen a number of similar reports lately about something similar, but all of them appear to be on Centos 7.  I'm guessing that sqlite library has a bug on Centos 7.

> On Apr 9, 2016, at 8:57 AM, Chandramouli P <mouli123 at gmail.com> wrote:
> 
> Hi,
> 
> Any help would be appreciated.
> 
> Thank you,
> Chandra.
> 
> 
> On Wed, Apr 6, 2016 at 6:40 PM, Chandramouli P <mouli123 at gmail.com <mailto:mouli123 at gmail.com>> wrote:
> Hi,
> 
> Please find my below deployed environment:
> 
> Environment: Microsoft Azure
> OS: CentOS 7.0 (64 bit)
> FreeSwitch Version: 1.6.6~64bit ( 64bit)
> 
> I installed FreeSwitch through "Yum" and works fine for couple of hours. After that I am getting below errors:
> 
> [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [attempt to write a readonly database]
> BEGIN EXCLUSIVE
> 2016-04-05 13:29:33.368355 [CRIT] switch_core_sqldb.c:1952 ERROR [attempt to write a readonly database]
> 2016-04-05 13:29:33.368355 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [cannot commit - no transaction is active]
> 
> Can anybody tell me what could be the issue and how to solve this?
> 
> Thanks in advance,
> Chandra.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From s.safarov at gmail.com  Mon Apr 11 23:59:18 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 11 Apr 2016 19:59:18 +0000
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
In-Reply-To: <20160411064240.Horde.6M0V77qRE_MGQRkFnMD4-w1@www.vfemail.net>
References: <570A3728.6020900@vfemail.net>
	<CAFLo2Qzi5oSHzXMyFEiDExdzHsmew_2a+Wne_f1Z7H6PUNTe4A@mail.gmail.com>
	<B700FF78-C2BA-41C3-8E9C-AD49B658DD60@freeswitch.org>
	<CAFLo2QwD6eM6=B1TUa0ObWa9-3GLraVJxeu0UfaV6QS4zkeLmA@mail.gmail.com>
	<B6118ED6-AAA7-462D-B4CB-06D3713948A8@d-man.org>
	<20160411064240.Horde.6M0V77qRE_MGQRkFnMD4-w1@www.vfemail.net>
Message-ID: <CAHtxdDeTKfpCrN1snWDTnuCTvvezfrzaKPWR6pf2V8M-NMRdwA@mail.gmail.com>

I think problem in order of flush command. To take effect nibble rate must
be set before execution flush.

On Mon, Apr 11, 2016, 22:50 tanguy <phenix at vfemail.net> wrote:

> > Sergey
> Thank you for your piece of dialplan but i don't think it's the
> solution. You offer a different solution to set the nibble_rate
> variable, but in my sample case this variable is properly set because
> the call is billed.
>
> > Luis
> A bug, maybe. I don't remember my freeswitch version ( i can't connect
> to my servers yet ) but i think it's 1.4 series.
>
> I can try the last 1.6 version, i will also try a standard freeswitch
> sample configuration ( without config files provided by fusionpbx )
>
>
> Quoting Darren <d at d-man.org>:
>
> > If you can point me to the pull request, I can take a look and see
> > if I can integrate it in.
> >
> > From:
> > <freeswitch-users-bounces at lists.freeswitch.org<mailto:
> freeswitch-users-bounces at lists.freeswitch.org>> on behalf of Luis Daniel
> Lucio Quiroz
> > <luis.daniel.lucio at gmail.com<mailto:luis.daniel.lucio at gmail.com>>
> > Reply-To: FreeSWITCH Users Help
> > <freeswitch-users at lists.freeswitch.org<mailto:
> freeswitch-users at lists.freeswitch.org>>
> > Date: Sunday, April 10, 2016 at 11:09 AM
> > To: FreeSWITCH Users Help
> > <freeswitch-users at lists.freeswitch.org<mailto:
> freeswitch-users at lists.freeswitch.org>>
> > Subject: Re: [Freeswitch-users] nibblebill do not terminate calls
> properly
> >
> >
> > There is. It is not from myself. I just don't remember it
> >
> > Le 10 avr. 2016 1:44 PM, "Ken Rice"
> > <krice at freeswitch.org<mailto:krice at freeswitch.org>> a ?crit :
> > If there is a known bug and a patch for this, there should be a jira
> > and a pull request for the patch
> >
> > Sent from my iPhone
> >
> > On Apr 10, 2016, at 12:40 PM, Luis Daniel Lucio Quiroz
> > <luis.daniel.lucio at gmail.com<mailto:luis.daniel.lucio at gmail.com>>
> > wrote:
> >
> >
> > There is a known bug and a known patch to fix it. I can't remember
> > right now which one it is.
> >
> > Hello
> > I would like to use nibblebilling for fraud prevention on
> > international and premium numbers ( national or emergency calls
> > should never be blocked and should still working even if the billing
> > database is unavailable  )
> >
> > Unfortunately that never hangup my calls using standard dial plan
> >
> > nibblebill.conf.xml
> >
> > <configuration name="nibblebill.conf" description="Nibble Billing">
> >   <settings>
> >     <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help
> > with these options -->
> >
> >     <!-- Information for connecting to your database -->
> >     <param name="db_dsn" value="pgsql://hostaddr=127.0.0.1
> > dbname=xxxx user=xxxx password=xxxx"/>
> >
> >     <!-- The database table where your CASH column is located -->
> >     <param name="db_table" value="accounts"/>
> >
> >     <!-- The column name where we store the value of the account -->
> >     <param name="db_column_cash" value="cash"/>
> >
> >     <!-- The column name for the unique ID identifying the account -->
> >     <param name="db_column_account" value="id"/>
> >
> >     <!-- Default heartbeat interval. Set to 'off' for no heartbeat
> > (i.e. bill only at end of call) -->
> >     <param name="global_heartbeat" value="20"/>
> >
> >     <!-- By default, warn a caller when their balance is at $5.00.
> > You can set this to a negative number. -->
> >     <param name="lowbal_amt" value="-100"/> <!-- i don't need this -->
> >     <param name="lowbal_action" value="play ding"/>
> >
> >     <!-- By default, terminate a caller when their balance hits
> > $0.00. You can set this to a negative number. -->
> >     <param name="nobal_amt" value="0"/>
> >     <param name="nobal_action" value="hangup"/>
> >
> >     <!-- If a call goes beyond a certain dollar amount, flag or
> > terminate it -->
> >     <param name="percall_max_amt" value="10"/>
> >     <param name="percall_action" value="hangup"/>
> >
> >   </settings>
> > </configuration>
> >
> > outbound route
> >
> > <extension name="international" >
> >    <condition field="destination_number" expression="^00(\d{6,20})$" >
> >        <action application="nibblebill" data="flush"/>
> >         <action application="set" data="nibble_account=${accountcode}"/>
> >         <action application="set" data="nibble_rate=0.1"/>
> >        <action application="bridge" data="sofia/gateway/gw_idt/$1" />
> >    </condition>
> > </extension>
> >
> >
> > My balance
> > My balance is already negative
> >
> >  select * from accounts ;
> >                 id                |   cash
> > ----------------------------------+-----------
> >  company.voip.mydomain.com<http://company.voip.mydomain.com> | -0.789471
> >
> > A sample call
> > My balance already negative, the call will be billed but never blocked
> >
> > Dialplan:
> > sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/
> 5003 at company.voip.domain.com> Action
> > nibblebill(flush)
> > Dialplan:
> > sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/
> 5003 at company.voip.domain.com> Action
> > set(nibble_account=${accountcode})
> > Dialplan:
> > sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/
> 5003 at company.voip.domain.com> Action
> > set(nibble_rate=0.1)
> > EXECUTE
> > sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/
> 5003 at company.voip.domain.com>
> > nibblebill(flush)
> > EXECUTE
> > sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/
> 5003 at company.voip.domain.com>
> > set(nibble_account=company.voip.domain.com<
> http://company.voip.domain.com>)
> > 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477
> > sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/
> 5003 at company.voip.domain.com> SET
> > [nibble_account]=[company.voip.domain.com<http://company.voip.domain.com
> >]
> > EXECUTE
> > sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/
> 5003 at company.voip.domain.com>
> > set(nibble_rate=0.1)
> > 2016-04-10 12:10:04.049418 [DEBUG] mod_dptools.c:1477
> > sofia/internal/5003 at company.voip.domain.com<mailto:sofia/internal/
> 5003 at company.voip.domain.com> SET
> > [nibble_rate]=[0.1]
> > 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:488 Attempting
> > to bill at $0.1 per minute to account
> > company.voip.domain.com<http://company.voip.domain.com>
> > 2016-04-10 12:10:52.132458 [INFO] mod_nibblebill.c:540 Beginning new
> > billing on 644488fc-ff04-11e5-9a27-fd2791153af9
> > 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:546 42 seconds
> > passed since last bill time of 2016-04-10 12:10:09
> > 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:563 Billing
> > $0.071033 to company.voip.domain.com<http://company.voip.domain.com>
> > (Call: / 0.000000 so far)
> > 2016-04-10 12:10:52.132458 [DEBUG] mod_nibblebill.c:393 Doing update
> query
> > 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:420 Doing lookup
> query
> > [SELECT cash AS nibble_balance FROM accounts WHERE
> > id='company.voip.domain.com<http://company.voip.domain.com>']
> > 2016-04-10 12:10:52.172440 [DEBUG] mod_nibblebill.c:428 Retrieved
> > current balance for account
> > company.voip.domain.com<http://company.voip.domain.com> (balance =
> > -0.860504)
> >
> >
> > Using b-leg only
> > Alternatively i tried an alternative dialplan ( Even if i don't
> > relay understand what is the meaning of b-leg billing  )
> >
> > <action application="bridge"
> >
> data="{enable_heartbeat_events=5,nibble_rate=0.1,nibble_account=${accountcode},originate_timeout=90}sofia/gateway/gw_idt/33$1"
> > />
> >
> > The result is better because the pending call is hanged up when the
> > balance reach 0.
> >
> > 2016-04-10 12:39:59.012462 [CRIT] mod_nibblebill.c:607 Balance of
> > -0.003237 fell below allowed amount of 0.000000! (Account
> > company.voip.domain.com<http://company.voip.domain.com>)
> >
> > But if i make a new call ( when my balance is negative )  , the
> > caller party is immediately hanged but this did not cancel
> > immediately the bridge:  The called party ring and can stay bridged
> > for 61 seconds after answer.
> >
> >
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org<mailto:consulting at freeswitch.org>
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> > http://www.freeswitch.org
> >
> > _________________________________________________________________________
> > 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
>
>
>
>
> -------------------------------------------------
>
> ONLY AT VFEmail! - Use our Metadata Mitigator to keep your email out of
> the NSA's hands!
> $24.95 ONETIME Lifetime accounts with Privacy Features!
> 15GB disk! No bandwidth quotas!
> Commercial and Bulk Mail Options!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 11 23:59:45 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 11 Apr 2016 14:59:45 -0500
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
	NEGOTIATION ERROR. SDP
In-Reply-To: <9b88e0f313a841d8af36b79c396b87a1@RESAMKMBX002.Resources.STELECT.LOCAL>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
	<38419347-D033-43AB-92DA-91D7D4C90465@jerris.com>
	<5c023f40f79d430c85dc5d902b33ebee@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAHtxdDf7KAQHEi+ennJs3Wnfd-pnuVjpzJ9Gn2Bert2AY-wzfA@mail.gmail.com>
	<302e815fb7c642d094445f4465256b92@RESDRCMBX002.Resources.STELECT.LOCAL>
	<CAHtxdDfHgBza0eiWqqxGCi_PH-Y8cGxGdGJFDRLhrnxC3rbWAA@mail.gmail.com>
	<a0e8cc45ffae4f42bb7be231bb03a163@RESDRCMBX002.Resources.STELECT.LOCAL>
	<C4A0FD03-EF2C-476F-B011-2950CA655689@jerris.com>
	<9b88e0f313a841d8af36b79c396b87a1@RESAMKMBX002.Resources.STELECT.LOCAL>
Message-ID: <EBDBB225-9755-4222-8B20-55A3B00EC658@jerris.com>

This is never attempting to call the second endpoint:

1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_ivr_originate.c:3750 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]

It then tries to negotiate the ice session with freeswitch, as it is trying to answer the call locally and send it to voicemail.  then we have:

1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_core_media.c:3380 sofia/internal/anonymous at 192.168.199.22 no suitable candidates found.
1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_core_media.c:4473 sofia/internal/anonymous at 192.168.199.22 Set 2833 dtmf send payload to 126 recv payload to 126

the issue is your box is on a rfc1918 ip, and the default candidate acl's do not consider those valid.  You will need to adjust the acl if you plan on having freeswitch as a server answering webrtc calls from private address space.



> On Apr 10, 2016, at 10:11 PM, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
> 
> Because existing SIP endpoint doesn?t support SRTP-DTLS, the transcoding (SRTP-DTLS and SRTP-SDES)is required to bridge RTP stream between WebRTC browser and SIP endpoint.
>  
>  
> 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR.  SDP:
> 1709c7c4-fa17-11e5-9184-9d31ebce310e v=0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e o=- 2269318439683772682 2 IN IP4 127.0.0.1
> 1709c7c4-fa17-11e5-9184-9d31ebce310e s=-
> 1709c7c4-fa17-11e5-9184-9d31ebce310e t=0 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=group:BUNDLE audio
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=msid-semantic: WMS rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o
> 1709c7c4-fa17-11e5-9184-9d31ebce310e m=audio 1107 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
> 1709c7c4-fa17-11e5-9184-9d31ebce310e c=IN IP4 192.168.199.216
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:111 opus/48000/2
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:103 ISAC/16000
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:104 ISAC/32000
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:9 G722/8000
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:0 PCMU/8000
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:8 PCMA/8000
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:106 CN/32000
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:105 CN/16000
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:13 CN/8000
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:126 telephone-event/8000
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp:1110 IN IP4 192.168.199.216
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 1 udp 2122260223 192.168.199.216 1107 typ host generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 1 udp 2122194687 10.20.102.216 1108 typ host generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 1 udp 2122129151 10.10.13.216 1109 typ host generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 2 udp 2122260222 192.168.199.216 1110 typ host generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 2 udp 2122194686 10.20.102.216 1111 typ host generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 2 udp 2122129150 10.10.13.216 1112 typ host generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 1 tcp 1518280447 192.168.199.216 0 typ host tcptype active generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 1 tcp 1518214911 10.20.102.216 0 typ host tcptype active generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 1 tcp 1518149375 10.10.13.216 0 typ host tcptype active generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 2 tcp 1518280446 192.168.199.216 0 typ host tcptype active generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 2 tcp 1518214910 10.20.102.216 0 typ host tcptype active generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 2 tcp 1518149374 10.10.13.216 0 typ host tcptype active generation 0
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-ufrag:4PWOC8zOrgkF0JcH
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-pwd:KoIliPYZgGkk+b6K7bJDPVLK
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=fingerprint:sha-256 18:9D:FC:F1:74:38:63:AE:F1:F8:F0:26:F5:83:A8:41:97:53:67:F5:35:FB:7E:F3:06:BF:D5:71:FD:A2:F6:38
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=setup:actpass
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=mid:audio
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time>
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp-mux
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=maxptime:60
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 cname:mrCkzHA0WH9Dd+8n
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 msid:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 62a659a4-ef68-469a-871c-0e61ef3dd211
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 mslabel:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o
> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 label:62a659a4-ef68-469a-871c-0e61ef3dd211
>  
> 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [NOTICE] switch_channel.c:3501 Hangup sofia/internal/anonymous at 192.168.199.22 <mailto:sofia/internal/anonymous at 192.168.199.22> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
> 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_ivr_originate.c:1217 sofia/internal/anonymous at 192.168.199.22 <mailto:sofia/internal/anonymous at 192.168.199.22> Media Establishment Failed.
>  
> ###############################################
> B leg:
>  
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.471064 [DEBUG] mod_sofia.c:88 sofia/internal/5shm6jpu at tibvb7p7p6od.invalid <mailto:sofia/internal/5shm6jpu at tibvb7p7p6od.invalid> SOFIA INIT
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1228 sip:5shm6jpu at 192.168.199.216:1106;transport=wss <sip:5shm6jpu at 192.168.199.216:1106;transport=wss> Setting proxy route to sofia/internal/5shm6jpu at tibvb7p7p6od.invalid <mailto:sofia/internal/5shm6jpu at tibvb7p7p6od.invalid>
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1257 sofia/internal/5shm6jpu at tibvb7p7p6od.invalid <mailto:sofia/internal/5shm6jpu at tibvb7p7p6od.invalid> sending invite version: 1.6.5  64bit
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e Local SDP:
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e v=0
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e o=FreeSWITCH 1459715932 1459715933 IN IP4 192.168.199.22
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e s=FreeSWITCH
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e c=IN IP4 192.168.199.22
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e t=0 0
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=msid-semantic: WMS V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e m=audio 25356 RTP/SAVPF 111 8 102 101
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:111 opus/48000/2
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:8 PCMA/8000
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:102 telephone-event/48000
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:101 telephone-event/8000
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fingerprint:sha-256 E7:50:3F:56:DE:21:E5:17:DA:F7:3A:3F:28:F8:BC:2A:C6:E7:E0:67:5C:6C:61:39:00:6D:99:DE:A4:A2:8A:80
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=setup:actpass
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp-mux
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp:25356 IN IP4 192.168.199.22
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 cname:OFXIzFBV4sOBkyJG
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 msid:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp a0
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 mslabel:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 label:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzpa0
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-ufrag:UUr7xgAHzqT3ejiT
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-pwd:O6reBS0vbEXl3xRNh8KFfbcE
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 1 udp 659136 192.168.199.22 25356 typ host generation 0
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 2 udp 659136 192.168.199.22 25356 typ host generation 0
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ptime:20
> 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=sendrecv
>  
>  
> @@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@
> Profile: internal
>  
> 2016-04-04 11:17:29.618618 [DEBUG] sofia.c:4236 debug [9]
> 2016-04-04 11:17:29.618624 [DEBUG] sofia.c:4236 sip-trace [no]
> 2016-04-04 11:17:29.618627 [DEBUG] sofia.c:4236 sip-capture [no]
> 2016-04-04 11:17:29.618630 [DEBUG] sofia.c:4236 watchdog-enabled [no]
> 2016-04-04 11:17:29.618634 [DEBUG] sofia.c:4236 watchdog-step-timeout [30000]
> 2016-04-04 11:17:29.618638 [DEBUG] sofia.c:4236 watchdog-event-timeout [30000]
> 2016-04-04 11:17:29.618642 [DEBUG] sofia.c:4236 log-auth-failures [true]
> 2016-04-04 11:17:29.618645 [DEBUG] sofia.c:4236 forward-unsolicited-mwi-notify [false]
> 2016-04-04 11:17:29.618648 [DEBUG] sofia.c:4236 context [public]
> 2016-04-04 11:17:29.618654 [DEBUG] sofia.c:4236 rfc2833-pt [101]
> 2016-04-04 11:17:29.618657 [DEBUG] sofia.c:4236 sip-port [5062]
> 2016-04-04 11:17:29.618661 [DEBUG] sofia.c:4236 dialplan [XML]
> 2016-04-04 11:17:29.618667 [DEBUG] sofia.c:4236 dtmf-duration [2000]
> 2016-04-04 11:17:29.618677 [DEBUG] sofia.c:4236 inbound-codec-prefs [OPUS]
> 2016-04-04 11:17:29.618684 [DEBUG] sofia.c:4236 outbound-codec-prefs [OPUS]
> 2016-04-04 11:17:29.618690 [DEBUG] sofia.c:4236 rtp-timer-name [soft]
> 2016-04-04 11:17:29.618694 [DEBUG] sofia.c:4236 rtp-ip [192.168.199.22]
> 2016-04-04 11:17:29.618698 [DEBUG] sofia.c:4236 sip-ip [192.168.199.22]
> 2016-04-04 11:17:29.618702 [DEBUG] sofia.c:4236 hold-music [local_stream://moh]
> 2016-04-04 11:17:29.618720 [DEBUG] sofia.c:4236 apply-nat-acl [nat.auto]
> 2016-04-04 11:17:29.618734 [ERR] sofia.c:5185 Not adding acl nat.auto because it's the local network
> 2016-04-04 11:17:29.618739 [DEBUG] sofia.c:4236 apply-inbound-acl [domains]
> 2016-04-04 11:17:29.618745 [DEBUG] sofia.c:4236 local-network-acl [localnet.auto]
> 2016-04-04 11:17:29.618749 [DEBUG] sofia.c:4236 record-path [/usr/local/freeswitch/recordings]
> 2016-04-04 11:17:29.618754 [DEBUG] sofia.c:4236 record-template [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
> 2016-04-04 11:17:29.618763 [DEBUG] sofia.c:4236 manage-presence [true]
> 2016-04-04 11:17:29.618768 [DEBUG] sofia.c:4236 presence-hosts [192.168.199.22,192.168.199.22]
> 2016-04-04 11:17:29.618774 [DEBUG] sofia.c:4236 presence-privacy [false]
> 2016-04-04 11:17:29.618778 [DEBUG] sofia.c:4236 inbound-codec-negotiation [generous]
> 2016-04-04 11:17:29.618783 [DEBUG] sofia.c:4236 tls [true]
> 2016-04-04 11:17:29.618791 [DEBUG] sofia.c:4236 tls-only [false]
> 2016-04-04 11:17:29.618798 [DEBUG] sofia.c:4236 tls-bind-params [transport=tls]
> 2016-04-04 11:17:29.618804 [DEBUG] sofia.c:4236 tls-sip-port [5063]
> 2016-04-04 11:17:29.618810 [DEBUG] sofia.c:4236 tls-passphrase []
> 2016-04-04 11:17:29.618816 [DEBUG] sofia.c:4236 tls-verify-date [true]
> 2016-04-04 11:17:29.618822 [DEBUG] sofia.c:4236 tls-verify-policy [none]
> 2016-04-04 11:17:29.618829 [ERR] sofia_glue.c:329 Invalid tls-verify-policy value: none
> 2016-04-04 11:17:29.618835 [DEBUG] sofia.c:4236 tls-verify-depth [2]
> 2016-04-04 11:17:29.618844 [DEBUG] sofia.c:4236 tls-verify-in-subjects []
> 2016-04-04 11:17:29.618850 [DEBUG] sofia.c:4236 tls-version [tlsv1,tlsv1.1,tlsv1.2]
> 2016-04-04 11:17:29.618860 [DEBUG] sofia.c:4236 tls-ciphers [ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH]
> 2016-04-04 11:17:29.618868 [DEBUG] sofia.c:4236 inbound-late-negotiation [true]
> 2016-04-04 11:17:29.618872 [DEBUG] sofia.c:4236 inbound-zrtp-passthru [true]
> 2016-04-04 11:17:29.618876 [DEBUG] sofia.c:4236 nonce-ttl [60]
> 2016-04-04 11:17:29.618881 [DEBUG] sofia.c:4236 auth-calls [true]
> 2016-04-04 11:17:29.618887 [DEBUG] sofia.c:4236 inbound-reg-force-matching-username [true]
> 2016-04-04 11:17:29.618893 [DEBUG] sofia.c:4236 auth-all-packets [false]
> 2016-04-04 11:17:29.618898 [DEBUG] sofia.c:4236 ext-rtp-ip [auto-nat]
> 2016-04-04 11:17:29.618902 [DEBUG] sofia.c:4236 ext-sip-ip [auto-nat]
> 2016-04-04 11:17:29.618908 [DEBUG] sofia.c:4236 rtp-timeout-sec [300]
> 2016-04-04 11:17:29.618913 [DEBUG] sofia.c:4236 rtp-hold-timeout-sec [1800]
> 2016-04-04 11:17:29.618917 [DEBUG] sofia.c:4236 force-register-domain [192.168.199.22]
> 2016-04-04 11:17:29.618922 [DEBUG] sofia.c:4236 force-subscription-domain [192.168.199.22]
> 2016-04-04 11:17:29.618927 [DEBUG] sofia.c:4236 force-register-db-domain [192.168.199.22]
> 2016-04-04 11:17:29.618933 [DEBUG] sofia.c:4236 ws-binding [:5066]
> 2016-04-04 11:17:29.618939 [DEBUG] sofia.c:4236 wss-binding [:7443]
> 2016-04-04 11:17:29.618945 [DEBUG] sofia.c:4236 challenge-realm [auto_from]
> 2016-04-04 11:17:29.618953 [INFO] sofia.c:5513 Setting MAX Auth Validity to 0 Attempts
> 2016-04-04 11:17:29.619091 [NOTICE] sofia.c:5680 Started Profile internal [sofia_reg_internal]
>  
> 2016-04-04 11:24:00.891443 [DEBUG] switch_loadable_module.c:735 Chat Thread Started
> 2016-04-04 11:24:00.891453 [INFO] switch_core.c:2418
> FreeSWITCH Version 1.6.5~64bit ( 64bit)
>  
> The detailed log is attached in previous emails.
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Friday, April 08, 2016 11:29 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> Seriously, use verto... it will be much cleaner... also, the issue you are likely having has to do with codec negotiation settings, but we can't say for sure without seeing a debug log.
>  
> On Apr 8, 2016, at 4:16 AM, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
>  
> Good. Can you share your scenario ?
>  
> Chrome (sipML5) ->FS (1.6.5-64bit  Media mode) ->Chrome (sipML5).
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Sergey Safarov
> Sent: Friday, April 08, 2016 3:17 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> All call with media transcoding enabled. In WebRTC case OPUS <-> G711a
>  
> On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> what is setting of inbound-bypass-media and inbound-proxy-media in your case?
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Sergey Safarov
> Sent: Friday, April 08, 2016 12:00 PM
> 
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> One week ago I has configured master with sipML5.
> You can try reproduce.
>  
> On Fri, Apr 8, 2016, 06:52 Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi Michael;
>  
> Same complaint at mod_sofia.c 2299.
> Mod_sofia.c: 2299 (sofia_media_tech_media(tech_pvt, r_sdp) != SWITCH_STATUS_SUCCESS).
> Switch_core_media.c :3573: switch_core_media_negotiation_sdp().
>  
> Eval ${webrtc_enable_dtls}, Eval ${media_webrtc}, Eval ${rtp_use_dtls}, Eval ${rtp_secure_media}, all return true.
>  
> Does anyone successfully set up this WebRTC demo (excluding video) using media mode as described by cookbook.  
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 10:17 PM
> 
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> media_webrtc=true
>  
>  
> On Apr 7, 2016, at 3:05 AM, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
>  
> Hi Michael;
>  
> Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as WebRTC signaling.
>  
> You can see SDP from chrome (+sipjs) for this in previous attachments.
>  
> If set inbound-bypass-media=true, then chrome caller can talk with chrome callee using opus without any issue.
>  
>  
> Regards
> Smile.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 2:47 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?
>  
> On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
>  
> I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 1:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> you keep saying 
> 
> (UDP/TLS/RTP/SAVPF)
>  
> are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.
> 
> On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi;
>  
> Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved? 
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Quan Huo Sheng
> Sent: Thursday, April 07, 2016 10:41 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.
>  
> Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.
>  
> As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.
>  
> In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).
>  
> Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.
>  
>  
> Regards
> Smile.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Wednesday, April 06, 2016 11:59 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> yes it's supported
> 
> On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi Itola;
>  
> Sorry, same error.
>  
> Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?
>  
> Finding when FS runs in media mode:
> codec causes caller side ?488 not acceptable here| incompatible destination ?
>                 callee side: ?cancel |user not registered?
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of ?talo Rossi
> Sent: Tuesday, April 05, 2016 8:58 PM
> To: FreeSWITCH Users Help
> Cc: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> Set media_mix_inbound_outbound_codecs=true
> 
> ?talo Rossi
> italo at freeswitch.org <mailto:italo at freeswitch.org>
> IRC chat.freenode.net <http://chat.freenode.net/> #freeswitch #freeswitch-dev
> Bugs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Docs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6 <https://hipchat.freeswitch.org/gUdAgy0m6>
>  
>  
> 
> On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi All;
>  
> I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min <http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min <http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.
> If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
> But if disabling media bypass mode, call is rejected by FS.
>  
> FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.
> Chrome uses opus 111, FS uses opus 116.
> Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().
>  
> Help is needed to troubleshoot this issue.
>  
> Thanks advance.
> Smile.
>  
> 
> [This e-mail is confidential and may be privileged. If you are not the
> intended recipient, please kindly notify us immediately and delete the message
> from your system; please do not copy or use it for any purpose, nor disclose
> its contents to any other person. Thank you.]
> ---ST Electronics Group---
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
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From mike at jerris.com  Tue Apr 12 00:01:29 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 11 Apr 2016 15:01:29 -0500
Subject: [Freeswitch-users] curl connect timeout question
In-Reply-To: <CAJdRzN4rDu=ZOsgSX-C6qGhwg2iTGdNDj7WrZKPwF+se=8LGBw@mail.gmail.com>
References: <CAJdRzN5MsyG8gcHEsbR3udnt4vGV0QWciypgJbAkwzNypsMJ7A@mail.gmail.com>
	<CAJdRzN4rDu=ZOsgSX-C6qGhwg2iTGdNDj7WrZKPwF+se=8LGBw@mail.gmail.com>
Message-ID: <F2A75B3F-6627-4809-9AA2-B296B4FBAC7E@jerris.com>

So you are adding code to it and then its crashing?

> On Apr 10, 2016, at 10:47 PM, Chris Mandra <mandra at gmail.com> wrote:
> 
> Any ideas?
> 
> On Saturday, April 9, 2016, Chris Mandra <mandra at gmail.com <mailto:mandra at gmail.com>> wrote:
> Hey guys, if I'm using curl in a module and the connection times out do you have suggestion for how this should be handled?
> 
> For example, if I disconnect freeswitch from the internet and have the following settings:
> 
> 
> curl_easy_setopt(curl_handle, CURLOPT_CONNECTTIMEOUT, 5L);
> curl_easy_setopt(curl_handle, CURLOPT_TIMEOUT, 5L);
> switch_curl_easy_setopt(curl_handle, CURLOPT_NOSIGNAL, 1L);
> 
> it never times out and then crashes freeswitch. Is this expected behavior?
> Is there something I should be doing instead?
> Thanks, chris

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From s.safarov at gmail.com  Tue Apr 12 00:08:17 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 11 Apr 2016 20:08:17 +0000
Subject: [Freeswitch-users] NATIVE SQL ERR [cannot commit - no
 transaction is active]
In-Reply-To: <BF3451FF-D102-4F48-89F0-59FBE35553C9@jerris.com>
References: <CALR0W68a6+iiimqtW6hyxVdKz8o8B1yg-P7tye0LjEmtajvOaw@mail.gmail.com>
	<CALR0W6_r=z0en7hYNeZYuqdZCvxN1kbmXZs51CKAx9WFTx6mqQ@mail.gmail.com>
	<BF3451FF-D102-4F48-89F0-59FBE35553C9@jerris.com>
Message-ID: <CAHtxdDehb65Ofi6d2LbsY41m44gQD60eN1AquWED-6PfP9Jsog@mail.gmail.com>

As version.
I has troubles with XFS on CentOS. I switched to EXT4 filesystem and this
resolves issue's related to filesystem.

On Mon, Apr 11, 2016, 22:55 Michael Jerris <mike at jerris.com> wrote:

> I have seen a number of similar reports lately about something similar,
> but all of them appear to be on Centos 7.  I'm guessing that sqlite library
> has a bug on Centos 7.
>
> On Apr 9, 2016, at 8:57 AM, Chandramouli P <mouli123 at gmail.com> wrote:
>
> Hi,
>
> Any help would be appreciated.
>
> Thank you,
> Chandra.
>
>
> On Wed, Apr 6, 2016 at 6:40 PM, Chandramouli P <mouli123 at gmail.com> wrote:
>
>> Hi,
>>
>> Please find my below deployed environment:
>>
>> Environment: Microsoft Azure
>> OS: CentOS 7.0 (64 bit)
>> FreeSwitch Version: 1.6.6~64bit ( 64bit)
>>
>> I installed FreeSwitch through "Yum" and works fine for couple of hours.
>> After that I am getting below errors:
>>
>> [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [attempt to write a readonly
>> database]
>> BEGIN EXCLUSIVE
>> 2016-04-05 13:29:33.368355 [CRIT] switch_core_sqldb.c:1952 ERROR [attempt
>> to write a readonly database]
>> 2016-04-05 13:29:33.368355 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR
>> [cannot commit - no transaction is active]
>>
>> Can anybody tell me what could be the issue and how to solve this?
>>
>> Thanks in advance,
>> Chandra.
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
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From italo at freeswitch.org  Tue Apr 12 01:15:55 2016
From: italo at freeswitch.org (=?utf-8?q?=C3=8Dtalo_Rossi?=)
Date: Mon, 11 Apr 2016 14:15:55 -0700 (PDT)
Subject: [Freeswitch-users] Thursday FreeSWITCH Bug Hunt
Message-ID: <6bjlu2bd4iupvvqncu1sf4ff3-0@mailer.nylas.com>

FreeSWITCHers,

  

Join us thursday 2PM CST for the Thursday FreeSWITCH Bug Hunt!

  

Where? [conference.freeswitch.org/vc/#/?autocall=888](https://conference.frees
witch.org/vc/#/?autocall=888
"https://conference.freeswitch.org/vc/#/?autocall=888" )

  

Chat? <https://hipchat.freeswitch.org/gYzPMg3Oc>

  

What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support!

  

Help us help you, Join the Bug Hunt!

?talo Rossi

italo at freeswitch.org

IRC chat.freenode.net #freeswitch #freeswitch-dev

Bugs? https://freeswitch.org/jira

Docs? https://freeswitch.org/jira

Chat? https://hipchat.freeswitch.org/gUdAgy0m6

  

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From phenix at vfemail.net  Tue Apr 12 01:23:33 2016
From: phenix at vfemail.net (Tanguy)
Date: Mon, 11 Apr 2016 23:23:33 +0200
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
In-Reply-To: <CAHtxdDeTKfpCrN1snWDTnuCTvvezfrzaKPWR6pf2V8M-NMRdwA@mail.gmail.com>
References: <570A3728.6020900@vfemail.net>
	<CAFLo2Qzi5oSHzXMyFEiDExdzHsmew_2a+Wne_f1Z7H6PUNTe4A@mail.gmail.com>
	<B700FF78-C2BA-41C3-8E9C-AD49B658DD60@freeswitch.org>
	<CAFLo2QwD6eM6=B1TUa0ObWa9-3GLraVJxeu0UfaV6QS4zkeLmA@mail.gmail.com>
	<B6118ED6-AAA7-462D-B4CB-06D3713948A8@d-man.org>
	<20160411064240.Horde.6M0V77qRE_MGQRkFnMD4-w1@www.vfemail.net>
	<CAHtxdDeTKfpCrN1snWDTnuCTvvezfrzaKPWR6pf2V8M-NMRdwA@mail.gmail.com>
Message-ID: <570C15D5.9010000@vfemail.net>

Hello Sergey

Thanks for your advice, now i can't place a new call with a negative balance

[DEBUG] mod_nibblebill.c:488 Attempting to bill at $0.1 per minute to 
account company.voip.domain.com
[DEBUG] mod_nibblebill.c:500 Not billing account company.voip.domain.com 
- call is not in answered state
[DEBUG] mod_nibblebill.c:420 Doing lookup query
[DEBUG] mod_nibblebill.c:428 Retrieved current balance for account 
account company.voip.domain.com (balance = -0.028433)
[DEBUG] mod_nibblebill.c:504 Comparing -0.028433 to hangup balance of 
0.000000

But currents calls are not hanged up when my balance reach 0 and the 
database is never updated during the call ( but only after hangup ), i 
probably have a issue with the "global_heartbeat" parameter.






On 11/04/2016 21:59, Sergey Safarov wrote:
>
> I think problem in order of flush command. To take effect nibble rate 
> must be set before execution flush.
>
>
> On Mon, Apr 11, 2016, 22:50 tanguy <phenix at vfemail.net 
> <mailto:phenix at vfemail.net>> wrote:
>
>     > Sergey
>     Thank you for your piece of dialplan but i don't think it's the
>     solution. You offer a different solution to set the nibble_rate
>     variable, but in my sample case this variable is properly set because
>     the call is billed.
>
>     > Luis
>     A bug, maybe. I don't remember my freeswitch version ( i can't connect
>     to my servers yet ) but i think it's 1.4 series.
>
>     I can try the last 1.6 version, i will also try a standard freeswitch
>     sample configuration ( without config files provided by fusionpbx )
>

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From s.safarov at gmail.com  Tue Apr 12 01:55:06 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 11 Apr 2016 21:55:06 +0000
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
In-Reply-To: <570C15D5.9010000@vfemail.net>
References: <570A3728.6020900@vfemail.net>
	<CAFLo2Qzi5oSHzXMyFEiDExdzHsmew_2a+Wne_f1Z7H6PUNTe4A@mail.gmail.com>
	<B700FF78-C2BA-41C3-8E9C-AD49B658DD60@freeswitch.org>
	<CAFLo2QwD6eM6=B1TUa0ObWa9-3GLraVJxeu0UfaV6QS4zkeLmA@mail.gmail.com>
	<B6118ED6-AAA7-462D-B4CB-06D3713948A8@d-man.org>
	<20160411064240.Horde.6M0V77qRE_MGQRkFnMD4-w1@www.vfemail.net>
	<CAHtxdDeTKfpCrN1snWDTnuCTvvezfrzaKPWR6pf2V8M-NMRdwA@mail.gmail.com>
	<570C15D5.9010000@vfemail.net>
Message-ID: <CAHtxdDfrtThmDh0GpETSiHDBwRFX4vy+-WhBXW6mb-pH3Q-nfw@mail.gmail.com>

Probable. Yes.
On my servers, all work as documented in confluence.

On Tue, Apr 12, 2016, 00:24 Tanguy <phenix at vfemail.net> wrote:

> Hello Sergey
>
> Thanks for your advice, now i can't place a new call with a negative
> balance
>
>
> [DEBUG] mod_nibblebill.c:488 Attempting to bill at $0.1 per minute to
> account company.voip.domain.com
> [DEBUG] mod_nibblebill.c:500 Not billing account company.voip.domain.com
> - call is not in answered state
>
> [DEBUG] mod_nibblebill.c:420 Doing lookup query
> [DEBUG] mod_nibblebill.c:428 Retrieved current balance for account account
> company.voip.domain.com (balance = -0.028433)
> [DEBUG] mod_nibblebill.c:504 Comparing -0.028433 to hangup balance of
> 0.000000
>
> But currents calls are not hanged up when my balance reach 0 and the
> database is never updated during the call ( but only after hangup ), i
> probably have a issue with the "global_heartbeat" parameter.
>
>
>
>
>
>
>
> On 11/04/2016 21:59, Sergey Safarov wrote:
>
> I think problem in order of flush command. To take effect nibble rate must
> be set before execution flush.
>
> On Mon, Apr 11, 2016, 22:50 tanguy <phenix at vfemail.net> wrote:
>
>> > Sergey
>> Thank you for your piece of dialplan but i don't think it's the
>> solution. You offer a different solution to set the nibble_rate
>> variable, but in my sample case this variable is properly set because
>> the call is billed.
>>
>> > Luis
>> A bug, maybe. I don't remember my freeswitch version ( i can't connect
>> to my servers yet ) but i think it's 1.4 series.
>>
>> I can try the last 1.6 version, i will also try a standard freeswitch
>> sample configuration ( without config files provided by fusionpbx )
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jelena at misticnabica.hr  Tue Apr 12 02:05:33 2016
From: jelena at misticnabica.hr (jelena at misticnabica.hr)
Date: Mon, 11 Apr 2016 22:05:33 GMT
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
Message-ID: <EA7B29E9302E4251871DE2063CFA7023.MAI@server2.totohost.hr>



From abaci64 at gmail.com  Tue Apr 12 02:30:35 2016
From: abaci64 at gmail.com (Abaci B)
Date: Mon, 11 Apr 2016 18:30:35 -0400
Subject: [Freeswitch-users] Bounty
Message-ID: <CAFmZFiP2=uyWTKfyb44SdawJzSXSEOcmok2pbFwWJZXcs6QAaA@mail.gmail.com>

Looking for developer that can help with a feature development, I currently
have a $200 bounty on it, if anyone is interested please let me know.
https://freeswitch.org/jira/browse/FS-8700
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From abalashov at evaristesys.com  Tue Apr 12 03:23:35 2016
From: abalashov at evaristesys.com (Alex Balashov)
Date: Mon, 11 Apr 2016 19:23:35 -0400
Subject: [Freeswitch-users] Bounty
In-Reply-To: <CAFmZFiP2=uyWTKfyb44SdawJzSXSEOcmok2pbFwWJZXcs6QAaA@mail.gmail.com>
References: <CAFmZFiP2=uyWTKfyb44SdawJzSXSEOcmok2pbFwWJZXcs6QAaA@mail.gmail.com>
Message-ID: <20160411232335.5419088.7719.228864@evaristesys.com>

?If you want this done in the so-called developed world, at least, you're going to have to up the price considerably.
?
--
Alex?Balashov?|?Principal?|?Evariste?Systems?LLC
1447?Peachtree?Street?NE,?Suite?700
Atlanta,?GA?30309
United?States

Tel:?+1-800-250-5920?(toll-free)?/?+1-678-954-0671 (direct)
Web:?http://www.evaristesys.com/, http://www.csrpswitch.com/

Sent?from?my?BlackBerry.
? Original Message ?
From: Abaci B
Sent: Monday, April 11, 2016 18:32
To: FreeSWITCH Users Help
Reply To: FreeSWITCH Users Help
Cc: freeswitch-biz at lists.freeswitch.org
Subject: [Freeswitch-users] Bounty



From jude.mukundane at gmail.com  Tue Apr 12 03:30:59 2016
From: jude.mukundane at gmail.com (Jude Mukundane)
Date: Tue, 12 Apr 2016 00:30:59 +0100
Subject: [Freeswitch-users] Bounty
In-Reply-To: <20160411232335.5419088.7719.228864@evaristesys.com>
References: <CAFmZFiP2=uyWTKfyb44SdawJzSXSEOcmok2pbFwWJZXcs6QAaA@mail.gmail.com>
	<20160411232335.5419088.7719.228864@evaristesys.com>
Message-ID: <CAP=KgfaPFhAdYGsy-wiD=5sqGob9xDE-n3BL3D=bpAWYQCCcWA@mail.gmail.com>

what's the feature anyway?

If it is a few lines, maybe.

Jude

On Tue, Apr 12, 2016 at 12:23 AM, Alex Balashov <abalashov at evaristesys.com>
wrote:

> ?If you want this done in the so-called developed world, at least, you're
> going to have to up the price considerably.
> ?
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 1447 Peachtree Street NE, Suite 700
> Atlanta, GA 30309
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> Sent from my BlackBerry.
>   Original Message
> From: Abaci B
> Sent: Monday, April 11, 2016 18:32
> To: FreeSWITCH Users Help
> Reply To: FreeSWITCH Users Help
> Cc: freeswitch-biz at lists.freeswitch.org
> Subject: [Freeswitch-users] Bounty
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From 35633 at heb.be  Tue Apr 12 11:49:27 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Tue, 12 Apr 2016 09:49:27 +0200
Subject: [Freeswitch-users] Freeswitch and opensips
In-Reply-To: <CAGVOjVXRaF5nDv6SdtbsM0vabXi1u+9ToDq_qv=evdt_U0dVqA@mail.gmail.com>
References: <CAFSzUtb8hyWgY+sugSsUDd5BNRDVsVTCX5NUkqzq505e9+0rig@mail.gmail.com>
	<CAGVOjVXRaF5nDv6SdtbsM0vabXi1u+9ToDq_qv=evdt_U0dVqA@mail.gmail.com>
Message-ID: <CAFSzUtbCFei06ro7O_H69kU=bpSAksqDcE4f0CbLZMpsaK4mZA@mail.gmail.com>

Thaank you, i'll try it.

NDUWAYEZU Joselyne

2016-04-11 21:12 GMT+02:00 Aqs Younas <aqsyounas at gmail.com>:

> you need to do both things.
>
> On 11 April 2016 at 17:25, Nduwayezu, Joselyne <35633 at heb.be> wrote:
>
>> I'm using Opensips as load balancer in front of FreeSwitch. To tell
>> Freeswitch that calls come from Opensips proxy, do i have to create a new
>> external profile in sip_profiles directory or add an extension in
>> dialplan/public.xml or both of two?
>>
>> Second question, in this file
>> :/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml , i read that i
>> have to specify the CIDR, is the ip address  of Opensips?
>>
>> Thank you
>>
>>
>>
>> NDUWAYEZU Joselyne
>>
>> Haute ?cole de Bruxelles
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Haute ?cole de Bruxelles
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From quanhs at stee.stengg.com  Tue Apr 12 13:50:32 2016
From: quanhs at stee.stengg.com (Quan Huo Sheng)
Date: Tue, 12 Apr 2016 09:50:32 +0000
Subject: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299
	CODEC	NEGOTIATION ERROR. SDP
In-Reply-To: <EBDBB225-9755-4222-8B20-55A3B00EC658@jerris.com>
References: <9e08d9f45fe245acac2abe6842174566@RESDRCMBX001.Resources.STELECT.LOCAL>
	<cwpfgawvv5ij6y8hbsi2xcdqc-0@mailer.nylas.com>
	<c1d78e0220d94d5d8420310c2132626b@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhcbGzQ0biocsLVx6hjkBrCh+LxCf0g3=96tVO5LH2crA@mail.gmail.com>
	<efb29975fbea4e20b6985d88a852826e@RESDRCMBX001.Resources.STELECT.LOCAL>
	<ad54f64492684d99bc6a2235067a5d0a@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAEwTsAhCZsrshgiVsP+C09HLJDY_Bdrdt1b-FgDT7dxOQiRzmw@mail.gmail.com>
	<46e6e378b7e7434aa35047ff33122af4@RESDRCMBX001.Resources.STELECT.LOCAL>
	<94396DE8-9B58-4CEC-82A9-B8EA84479DCD@jerris.com>
	<f7f4d8a3fbba473cb42a006601938a10@RESDRCMBX001.Resources.STELECT.LOCAL>
	<38419347-D033-43AB-92DA-91D7D4C90465@jerris.com>
	<5c023f40f79d430c85dc5d902b33ebee@RESDRCMBX001.Resources.STELECT.LOCAL>
	<CAHtxdDf7KAQHEi+ennJs3Wnfd-pnuVjpzJ9Gn2Bert2AY-wzfA@mail.gmail.com>
	<302e815fb7c642d094445f4465256b92@RESDRCMBX002.Resources.STELECT.LOCAL>
	<CAHtxdDfHgBza0eiWqqxGCi_PH-Y8cGxGdGJFDRLhrnxC3rbWAA@mail.gmail.com>
	<a0e8cc45ffae4f42bb7be231bb03a163@RESDRCMBX002.Resources.STELECT.LOCAL>
	<C4A0FD03-EF2C-476F-B011-2950CA655689@jerris.com>
	<9b88e0f313a841d8af36b79c396b87a1@RESAMKMBX002.Resources.STELECT.LOCAL>
	<EBDBB225-9755-4222-8B20-55A3B00EC658@jerris.com>
Message-ID: <298c6228dbf549e7aefd65083e77ea6d@RESAMKMBX002.Resources.STELECT.LOCAL>

Hi Michael;

It finally functions after adjusting ACL. Great, I almost give up and change to try Verto that FreeSwitch prefers to. Thanks a lot.
Next step is to integrate with existing SIP domain or sip phone.


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, April 12, 2016 4:00 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

This is never attempting to call the second endpoint:

1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_ivr_originate.c:3750 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED]

It then tries to negotiate the ice session with freeswitch, as it is trying to answer the call locally and send it to voicemail.  then we have:

1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_core_media.c:3380 sofia/internal/anonymous at 192.168.199.22<mailto:sofia/internal/anonymous at 192.168.199.22> no suitable candidates found.
1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_core_media.c:4473 sofia/internal/anonymous at 192.168.199.22<mailto:sofia/internal/anonymous at 192.168.199.22> Set 2833 dtmf send payload to 126 recv payload to 126

the issue is your box is on a rfc1918 ip, and the default candidate acl's do not consider those valid.  You will need to adjust the acl if you plan on having freeswitch as a server answering webrtc calls from private address space.



On Apr 10, 2016, at 10:11 PM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

Because existing SIP endpoint doesn?t support SRTP-DTLS, the transcoding (SRTP-DTLS and SRTP-SDES)is required to bridge RTP stream between WebRTC browser and SIP endpoint.


1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR.  SDP:
1709c7c4-fa17-11e5-9184-9d31ebce310e v=0
1709c7c4-fa17-11e5-9184-9d31ebce310e o=- 2269318439683772682 2 IN IP4 127.0.0.1
1709c7c4-fa17-11e5-9184-9d31ebce310e s=-
1709c7c4-fa17-11e5-9184-9d31ebce310e t=0 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=group:BUNDLE audio
1709c7c4-fa17-11e5-9184-9d31ebce310e a=msid-semantic: WMS rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o
1709c7c4-fa17-11e5-9184-9d31ebce310e m=audio 1107 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
1709c7c4-fa17-11e5-9184-9d31ebce310e c=IN IP4 192.168.199.216
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:111 opus/48000/2
1709c7c4-fa17-11e5-9184-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:103 ISAC/16000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:104 ISAC/32000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:9 G722/8000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:0 PCMU/8000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:8 PCMA/8000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:106 CN/32000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:105 CN/16000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:13 CN/8000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:126 telephone-event/8000
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp:1110 IN IP4 192.168.199.216
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 1 udp 2122260223 192.168.199.216 1107 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 1 udp 2122194687 10.20.102.216 1108 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 1 udp 2122129151 10.10.13.216 1109 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 2 udp 2122260222 192.168.199.216 1110 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 2 udp 2122194686 10.20.102.216 1111 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 2 udp 2122129150 10.10.13.216 1112 typ host generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 1 tcp 1518280447 192.168.199.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 1 tcp 1518214911 10.20.102.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 1 tcp 1518149375 10.10.13.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 2 tcp 1518280446 192.168.199.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 2 tcp 1518214910 10.20.102.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 2 tcp 1518149374 10.10.13.216 0 typ host tcptype active generation 0
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-ufrag:4PWOC8zOrgkF0JcH
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-pwd:KoIliPYZgGkk+b6K7bJDPVLK
1709c7c4-fa17-11e5-9184-9d31ebce310e a=fingerprint:sha-256 18:9D:FC:F1:74:38:63:AE:F1:F8:F0:26:F5:83:A8:41:97:53:67:F5:35:FB:7E:F3:06:BF:D5:71:FD:A2:F6:38
1709c7c4-fa17-11e5-9184-9d31ebce310e a=setup:actpass
1709c7c4-fa17-11e5-9184-9d31ebce310e a=mid:audio
1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp-mux
1709c7c4-fa17-11e5-9184-9d31ebce310e a=maxptime:60
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 cname:mrCkzHA0WH9Dd+8n
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 msid:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 62a659a4-ef68-469a-871c-0e61ef3dd211
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 mslabel:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o
1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 label:62a659a4-ef68-469a-871c-0e61ef3dd211

1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [NOTICE] switch_channel.c:3501 Hangup sofia/internal/anonymous at 192.168.199.22<mailto:sofia/internal/anonymous at 192.168.199.22> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_ivr_originate.c:1217 sofia/internal/anonymous at 192.168.199.22<mailto:sofia/internal/anonymous at 192.168.199.22> Media Establishment Failed.

###############################################
B leg:

1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.471064 [DEBUG] mod_sofia.c:88 sofia/internal/5shm6jpu at tibvb7p7p6od.invalid<mailto:sofia/internal/5shm6jpu at tibvb7p7p6od.invalid> SOFIA INIT
1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1228 sip:5shm6jpu at 192.168.199.216:1106;transport=wss Setting proxy route to sofia/internal/5shm6jpu at tibvb7p7p6od.invalid<mailto:sofia/internal/5shm6jpu at tibvb7p7p6od.invalid>
1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1257 sofia/internal/5shm6jpu at tibvb7p7p6od.invalid<mailto:sofia/internal/5shm6jpu at tibvb7p7p6od.invalid> sending invite version: 1.6.5  64bit
1d260ca8-fa17-11e5-91a9-9d31ebce310e Local SDP:
1d260ca8-fa17-11e5-91a9-9d31ebce310e v=0
1d260ca8-fa17-11e5-91a9-9d31ebce310e o=FreeSWITCH 1459715932 1459715933 IN IP4 192.168.199.22
1d260ca8-fa17-11e5-91a9-9d31ebce310e s=FreeSWITCH
1d260ca8-fa17-11e5-91a9-9d31ebce310e c=IN IP4 192.168.199.22
1d260ca8-fa17-11e5-91a9-9d31ebce310e t=0 0
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=msid-semantic: WMS V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp
1d260ca8-fa17-11e5-91a9-9d31ebce310e m=audio 25356 RTP/SAVPF 111 8 102 101
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:111 opus/48000/2
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:8 PCMA/8000
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:102 telephone-event/48000
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:101 telephone-event/8000
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fingerprint:sha-256 E7:50:3F:56:DE:21:E5:17:DA:F7:3A:3F:28:F8:BC:2A:C6:E7:E0:67:5C:6C:61:39:00:6D:99:DE:A4:A2:8A:80
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=setup:actpass
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp-mux
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp:25356 IN IP4 192.168.199.22
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 cname:OFXIzFBV4sOBkyJG
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 msid:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp a0
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 mslabel:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 label:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzpa0
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-ufrag:UUr7xgAHzqT3ejiT
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-pwd:O6reBS0vbEXl3xRNh8KFfbcE
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 1 udp 659136 192.168.199.22 25356 typ host generation 0
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 2 udp 659136 192.168.199.22 25356 typ host generation 0
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ptime:20
1d260ca8-fa17-11e5-91a9-9d31ebce310e a=sendrecv


@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@
Profile: internal

2016-04-04 11:17:29.618618 [DEBUG] sofia.c:4236 debug [9]
2016-04-04 11:17:29.618624 [DEBUG] sofia.c:4236 sip-trace [no]
2016-04-04 11:17:29.618627 [DEBUG] sofia.c:4236 sip-capture [no]
2016-04-04 11:17:29.618630 [DEBUG] sofia.c:4236 watchdog-enabled [no]
2016-04-04 11:17:29.618634 [DEBUG] sofia.c:4236 watchdog-step-timeout [30000]
2016-04-04 11:17:29.618638 [DEBUG] sofia.c:4236 watchdog-event-timeout [30000]
2016-04-04 11:17:29.618642 [DEBUG] sofia.c:4236 log-auth-failures [true]
2016-04-04 11:17:29.618645 [DEBUG] sofia.c:4236 forward-unsolicited-mwi-notify [false]
2016-04-04 11:17:29.618648 [DEBUG] sofia.c:4236 context [public]
2016-04-04 11:17:29.618654 [DEBUG] sofia.c:4236 rfc2833-pt [101]
2016-04-04 11:17:29.618657 [DEBUG] sofia.c:4236 sip-port [5062]
2016-04-04 11:17:29.618661 [DEBUG] sofia.c:4236 dialplan [XML]
2016-04-04 11:17:29.618667 [DEBUG] sofia.c:4236 dtmf-duration [2000]
2016-04-04 11:17:29.618677 [DEBUG] sofia.c:4236 inbound-codec-prefs [OPUS]
2016-04-04 11:17:29.618684 [DEBUG] sofia.c:4236 outbound-codec-prefs [OPUS]
2016-04-04 11:17:29.618690 [DEBUG] sofia.c:4236 rtp-timer-name [soft]
2016-04-04 11:17:29.618694 [DEBUG] sofia.c:4236 rtp-ip [192.168.199.22]
2016-04-04 11:17:29.618698 [DEBUG] sofia.c:4236 sip-ip [192.168.199.22]
2016-04-04 11:17:29.618702 [DEBUG] sofia.c:4236 hold-music [local_stream://moh]
2016-04-04 11:17:29.618720 [DEBUG] sofia.c:4236 apply-nat-acl [nat.auto]
2016-04-04 11:17:29.618734 [ERR] sofia.c:5185 Not adding acl nat.auto because it's the local network
2016-04-04 11:17:29.618739 [DEBUG] sofia.c:4236 apply-inbound-acl [domains]
2016-04-04 11:17:29.618745 [DEBUG] sofia.c:4236 local-network-acl [localnet.auto]
2016-04-04 11:17:29.618749 [DEBUG] sofia.c:4236 record-path [/usr/local/freeswitch/recordings]
2016-04-04 11:17:29.618754 [DEBUG] sofia.c:4236 record-template [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
2016-04-04 11:17:29.618763 [DEBUG] sofia.c:4236 manage-presence [true]
2016-04-04 11:17:29.618768 [DEBUG] sofia.c:4236 presence-hosts [192.168.199.22,192.168.199.22]
2016-04-04 11:17:29.618774 [DEBUG] sofia.c:4236 presence-privacy [false]
2016-04-04 11:17:29.618778 [DEBUG] sofia.c:4236 inbound-codec-negotiation [generous]
2016-04-04 11:17:29.618783 [DEBUG] sofia.c:4236 tls [true]
2016-04-04 11:17:29.618791 [DEBUG] sofia.c:4236 tls-only [false]
2016-04-04 11:17:29.618798 [DEBUG] sofia.c:4236 tls-bind-params [transport=tls]
2016-04-04 11:17:29.618804 [DEBUG] sofia.c:4236 tls-sip-port [5063]
2016-04-04 11:17:29.618810 [DEBUG] sofia.c:4236 tls-passphrase []
2016-04-04 11:17:29.618816 [DEBUG] sofia.c:4236 tls-verify-date [true]
2016-04-04 11:17:29.618822 [DEBUG] sofia.c:4236 tls-verify-policy [none]
2016-04-04 11:17:29.618829 [ERR] sofia_glue.c:329 Invalid tls-verify-policy value: none
2016-04-04 11:17:29.618835 [DEBUG] sofia.c:4236 tls-verify-depth [2]
2016-04-04 11:17:29.618844 [DEBUG] sofia.c:4236 tls-verify-in-subjects []
2016-04-04 11:17:29.618850 [DEBUG] sofia.c:4236 tls-version [tlsv1,tlsv1.1,tlsv1.2]
2016-04-04 11:17:29.618860 [DEBUG] sofia.c:4236 tls-ciphers [ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH]
2016-04-04 11:17:29.618868 [DEBUG] sofia.c:4236 inbound-late-negotiation [true]
2016-04-04 11:17:29.618872 [DEBUG] sofia.c:4236 inbound-zrtp-passthru [true]
2016-04-04 11:17:29.618876 [DEBUG] sofia.c:4236 nonce-ttl [60]
2016-04-04 11:17:29.618881 [DEBUG] sofia.c:4236 auth-calls [true]
2016-04-04 11:17:29.618887 [DEBUG] sofia.c:4236 inbound-reg-force-matching-username [true]
2016-04-04 11:17:29.618893 [DEBUG] sofia.c:4236 auth-all-packets [false]
2016-04-04 11:17:29.618898 [DEBUG] sofia.c:4236 ext-rtp-ip [auto-nat]
2016-04-04 11:17:29.618902 [DEBUG] sofia.c:4236 ext-sip-ip [auto-nat]
2016-04-04 11:17:29.618908 [DEBUG] sofia.c:4236 rtp-timeout-sec [300]
2016-04-04 11:17:29.618913 [DEBUG] sofia.c:4236 rtp-hold-timeout-sec [1800]
2016-04-04 11:17:29.618917 [DEBUG] sofia.c:4236 force-register-domain [192.168.199.22]
2016-04-04 11:17:29.618922 [DEBUG] sofia.c:4236 force-subscription-domain [192.168.199.22]
2016-04-04 11:17:29.618927 [DEBUG] sofia.c:4236 force-register-db-domain [192.168.199.22]
2016-04-04 11:17:29.618933 [DEBUG] sofia.c:4236 ws-binding [:5066]
2016-04-04 11:17:29.618939 [DEBUG] sofia.c:4236 wss-binding [:7443]
2016-04-04 11:17:29.618945 [DEBUG] sofia.c:4236 challenge-realm [auto_from]
2016-04-04 11:17:29.618953 [INFO] sofia.c:5513 Setting MAX Auth Validity to 0 Attempts
2016-04-04 11:17:29.619091 [NOTICE] sofia.c:5680 Started Profile internal [sofia_reg_internal]

2016-04-04 11:24:00.891443 [DEBUG] switch_loadable_module.c:735 Chat Thread Started
2016-04-04 11:24:00.891453 [INFO] switch_core.c:2418
FreeSWITCH Version 1.6.5~64bit ( 64bit)

The detailed log is attached in previous emails.


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Friday, April 08, 2016 11:29 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Seriously, use verto... it will be much cleaner... also, the issue you are likely having has to do with codec negotiation settings, but we can't say for sure without seeing a debug log.

On Apr 8, 2016, at 4:16 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

Good. Can you share your scenario ?

Chrome (sipML5) ->FS (1.6.5-64bit  Media mode) ->Chrome (sipML5).


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
Sent: Friday, April 08, 2016 3:17 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

All call with media transcoding enabled. In WebRTC case OPUS <-> G711a

On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
what is setting of inbound-bypass-media and inbound-proxy-media in your case?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Sergey Safarov
Sent: Friday, April 08, 2016 12:00 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

One week ago I has configured master with sipML5.
You can try reproduce.

On Fri, Apr 8, 2016, 06:52 Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Michael;

Same complaint at mod_sofia.c 2299.
Mod_sofia.c: 2299 (sofia_media_tech_media(tech_pvt, r_sdp) != SWITCH_STATUS_SUCCESS).
Switch_core_media.c :3573: switch_core_media_negotiation_sdp().

Eval ${webrtc_enable_dtls}, Eval ${media_webrtc}, Eval ${rtp_use_dtls}, Eval ${rtp_secure_media}, all return true.

Does anyone successfully set up this WebRTC demo (excluding video) using media mode as described by cookbook.


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 10:17 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

media_webrtc=true


On Apr 7, 2016, at 3:05 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

Hi Michael;

Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as WebRTC signaling.

You can see SDP from chrome (+sipjs) for this in previous attachments.

If set inbound-bypass-media=true, then chrome caller can talk with chrome callee using opus without any issue.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 2:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?

On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:

I am using sip.js (sip-0.7.0.min as mentioned at book ?FreeSWITCH 1.6 cookbook |author: Anthony Minessale ?), not jssip.

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Thursday, April 07, 2016 1:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

you keep saying

(UDP/TLS/RTP/SAVPF)

are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.

On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi;

Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved?


Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Quan Huo Sheng
Sent: Thursday, April 07, 2016 10:41 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.

Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.

As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.

In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).

Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Wednesday, April 06, 2016 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

yes it's supported

On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Itola;

Sorry, same error.

Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?

Finding when FS runs in media mode:
codec causes caller side ?488 not acceptable here| incompatible destination ?
                callee side: ?cancel |user not registered?

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Set media_mix_inbound_outbound_codecs=true

?talo Rossi
italo at freeswitch.org<mailto:italo at freeswitch.org>
IRC chat.freenode.net<http://chat.freenode.net/> #freeswitch #freeswitch-dev
Bugs? https://freeswitch.org/jira
Docs? https://freeswitch.org/jira
Chat? https://hipchat.freeswitch.org/gUdAgy0m6


On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi All;

I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min<http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min<http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book ?FreeSWITCH 1.6 Cookbook?.
If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.
But if disabling media bypass mode, call is rejected by FS.

FS log shows ?[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP?. Attachment is detailed FS log file.
Chrome uses opus 111, FS uses opus 116.
Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().

Help is needed to troubleshoot this issue.

Thanks advance.
Smile.


[This e-mail is confidential and may be privileged. If you are not the
intended recipient, please kindly notify us immediately and delete the message
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>

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

FreeSWITCH-users mailing list
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<http://www.freeswitch.org/>

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From dutangp at gmail.com  Tue Apr 12 13:56:15 2016
From: dutangp at gmail.com (Pierre-Emmanuel Dutang)
Date: Tue, 12 Apr 2016 11:56:15 +0200
Subject: [Freeswitch-users] Recording
Message-ID: <CAMvMm-AdXXjk7nRWpYrAA1rdTFcO-KqQUD=tMw64L_T1h7A2xw@mail.gmail.com>

Hi all,

I try to record conversation with my FS and I have got a strange issue
during my recording test.

When I call my number I can do record and everything works fine, I hear my
voice perfectly.
When I simulate 1000 calls from FS1 to FS2, the calls works but I got only
files of 4Ko with nothing inside.

The dialplan that I call is simple:
        <action application="ring_ready"/>
        <action application="answer"/>
        <action application="sched_hangup" data="+120"/>
        <action application="playback"
data="/usr/share/freeswitch/sounds/machine.wav"/>

When I try to call this FS with my phone I perfectly hear the wav file.

I guess it's a media problem but I totally don't know what's wrong.

Best,
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From joel at gogii.net  Tue Apr 12 07:44:23 2016
From: joel at gogii.net (Joel Serrano)
Date: Mon, 11 Apr 2016 20:44:23 -0700
Subject: [Freeswitch-users] Billsec in dialplan
Message-ID: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>

Hi,

Is it possible to access ${billsec} in the dialplan after the bridge
application (having hangup_after_bridge=false)??

I want to log the billsec but currently it is always empty.

I have a workaround in a test server using api_hangup_hook to call a lua
script with session_in_hangup_hook=true, and inside the lua script I can
access the billsec variable and thus write it to the logs.

Is there no way to do this within the XML dialplan?

Thanks!

Best regards,
Joel.
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From stszap at gmail.com  Tue Apr 12 12:47:49 2016
From: stszap at gmail.com (=?UTF-8?B?0KHRgtCw0L3QuNGB0LvQsNCyINCX0LDQv9C+0LvRjNGB0LrQuNC5?=)
Date: Tue, 12 Apr 2016 13:47:49 +0500
Subject: [Freeswitch-users] Codec renegotiation on Re-INVITE withot SDP
In-Reply-To: <CANtLK7P2JWntbupPU8m1HfC=X9Q8Vop7E=5BabE2oPUfz2SsQg@mail.gmail.com>
References: <CANtLK7P2JWntbupPU8m1HfC=X9Q8Vop7E=5BabE2oPUfz2SsQg@mail.gmail.com>
Message-ID: <CANtLK7OjwCCDnNBRCgLbDqf69+2=6Ee5U8Ma-resBVeDhOVXXQ@mail.gmail.com>

Anyone? I will try to describe my problem once more. I have default FS
install. I enabled "renegotiate-codec-on-hold",
"renegotiate-codec-on-reinvite" and "enable-3pcc" in internal profile. My
dialplan is:

<include>
  <context name="default">

    <extension name="echo">
      <condition field="destination_number" expression="^.*$">
        <action application="answer"/>
        <action application="echo"/>
      </condition>
    </extension>

  </context>
</include>


I use the following sipp scenario https://pastebin.freeswitch.org/24629
with the following command:
sipp <ip>:<port>  -sf sipp_scenario.xml -m 1

And getting the following siptrace https://pastebin.freeswitch.org/24630

My question is: Is it possible to setup FS in such a way, that in the
second "200 OK" it would send "a=sendrecv" instead of "a=recvonly"?

Thanks.

On Wed, Apr 6, 2016 at 6:16 PM, ????????? ?????????? <stszap at gmail.com>
wrote:

> Hello. We are facing the following problem with our FS setup:
>
> 1. provider sends us INVITE with "a=sendonly" in SDP
> 2. FS picks up the call, playing ivr and replying with 200 and
> "a=recvonly" in SDP
> 3. provider sends Re-INVITE without SDP
> 4. FS replying 200 with already negotiated SDP (single codec and
> "a=recvonly")
>
> Provider wants renegotiation of sdp on step 4, or a least "a=sendrecv"
> parameter. Is there a way to achieve this without source modification? We
> tried setting following profile parameters:
> renegotiate-codec-on-hold=true
> renegotiate-codec-on-reinvite=true
> enable-3pcc=true
> but no luck. The only solution we found was modifing sofia.c like this:
> line 7374 - replace switch_core_media_gen_local_sdp(session,
> SDP_TYPE_RESPONSE, NULL, 0, NULL, 0); with
> switch_core_media_gen_local_sdp(session, SDP_TYPE_RESPONSE, NULL, 0,
> "sendrecv", 0);
> line 7376 - replace nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); with
> nua_respond(tech_pvt->nh, SIP_200_OK,
> SOATAG_USER_SDP_STR(tech_pvt->mparams.local_sdp_str), TAG_END());
>
> Thanks.
>
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From deepikay at iiitd.ac.in  Tue Apr 12 17:29:02 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Tue, 12 Apr 2016 18:59:02 +0530
Subject: [Freeswitch-users] Nested Conference
Message-ID: <CAM99_YkccOiQgim8edR6VcJ94TY2zW769P7QEFB9W1Yyrtg-hQ@mail.gmail.com>

Hi All,

Is there a way to start multiple sub conference calls in an on going
conference of large number of callers.

For example: There a conference going on between 20 callers, then some
point in time, a conference between 3-4 callers spawns based on some
condition so that what they discuss is not audible to rest of the caller
and on termination of the condition these 3-4 people rejoins the original
conference.

Regards,
Deepika
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From s.safarov at gmail.com  Tue Apr 12 17:43:23 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Tue, 12 Apr 2016 13:43:23 +0000
Subject: [Freeswitch-users] Nested Conference
In-Reply-To: <CAM99_YkccOiQgim8edR6VcJ94TY2zW769P7QEFB9W1Yyrtg-hQ@mail.gmail.com>
References: <CAM99_YkccOiQgim8edR6VcJ94TY2zW769P7QEFB9W1Yyrtg-hQ@mail.gmail.com>
Message-ID: <CAHtxdDesFsjp50qdRBVi2NF4788P7O0RBJidvsXi7Vea-jakBg@mail.gmail.com>

I has done simalar via tranfering requred menbers to new conference.

??, 12 ???. 2016 ?. ? 16:30, Deepika Yadav <deepikay at iiitd.ac.in>:

> Hi All,
>
> Is there a way to start multiple sub conference calls in an on going
> conference of large number of callers.
>
> For example: There a conference going on between 20 callers, then some
> point in time, a conference between 3-4 callers spawns based on some
> condition so that what they discuss is not audible to rest of the caller
> and on termination of the condition these 3-4 people rejoins the original
> conference.
>
> Regards,
> Deepika
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From deepikay at iiitd.ac.in  Tue Apr 12 18:03:08 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Tue, 12 Apr 2016 19:33:08 +0530
Subject: [Freeswitch-users] Nested Conference
In-Reply-To: <CAHtxdDesFsjp50qdRBVi2NF4788P7O0RBJidvsXi7Vea-jakBg@mail.gmail.com>
References: <CAM99_YkccOiQgim8edR6VcJ94TY2zW769P7QEFB9W1Yyrtg-hQ@mail.gmail.com>
	<CAHtxdDesFsjp50qdRBVi2NF4788P7O0RBJidvsXi7Vea-jakBg@mail.gmail.com>
Message-ID: <CAM99_Ym0ZrF2N+iRDaju+uTLe6HkDKQN7VSTJtEEH0e2A5_hxg@mail.gmail.com>

Thanks, I will definitely try transfer api.

Regards,
Deepika

On Tue, Apr 12, 2016 at 7:13 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> I has done simalar via tranfering requred menbers to new conference.
>
> ??, 12 ???. 2016 ?. ? 16:30, Deepika Yadav <deepikay at iiitd.ac.in>:
>
>> Hi All,
>>
>> Is there a way to start multiple sub conference calls in an on going
>> conference of large number of callers.
>>
>> For example: There a conference going on between 20 callers, then some
>> point in time, a conference between 3-4 callers spawns based on some
>> condition so that what they discuss is not audible to rest of the caller
>> and on termination of the condition these 3-4 people rejoins the original
>> conference.
>>
>> Regards,
>> Deepika
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From amani.mansour2 at gmail.com  Tue Apr 12 18:06:17 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Tue, 12 Apr 2016 14:06:17 +0000
Subject: [Freeswitch-users] Anonymous call with p-assrted-identity
Message-ID: <CAFETSWgZup+e1ef1RZgbrTD8rm2mn5cVChv84jrOu=jEVUp0yA@mail.gmail.com>

Hi all,
I need to call from an extension A registered in my freeswitch another
extension registered in my soft phone B
i have done this : originate
{origination_caller_id_number=A,origination_caller_id_name=A}user/B A
is this correct ?
also I want to have anonymous  call so i replace
origination_caller_id_number and origination_caller_id_name with anonymous
is this correct ?
last question I need to add P-Asserted-Identity do you know how please ?

thanks,best regards
amani
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From abaci64 at gmail.com  Tue Apr 12 18:39:03 2016
From: abaci64 at gmail.com (Abaci B)
Date: Tue, 12 Apr 2016 10:39:03 -0400
Subject: [Freeswitch-users] Nested Conference
In-Reply-To: <CAM99_Ym0ZrF2N+iRDaju+uTLe6HkDKQN7VSTJtEEH0e2A5_hxg@mail.gmail.com>
References: <CAM99_YkccOiQgim8edR6VcJ94TY2zW769P7QEFB9W1Yyrtg-hQ@mail.gmail.com>
	<CAHtxdDesFsjp50qdRBVi2NF4788P7O0RBJidvsXi7Vea-jakBg@mail.gmail.com>
	<CAM99_Ym0ZrF2N+iRDaju+uTLe6HkDKQN7VSTJtEEH0e2A5_hxg@mail.gmail.com>
Message-ID: <CAFmZFiMqs1c3tTuN--pOQBJ6H-KOiU=h+a_WFOhhLtmtunhmGQ@mail.gmail.com>

or you can try the relate api to isolate those people within the main
conference if that works for you

On Tue, Apr 12, 2016 at 10:03 AM, Deepika Yadav <deepikay at iiitd.ac.in>
wrote:

> Thanks, I will definitely try transfer api.
>
> Regards,
> Deepika
>
> On Tue, Apr 12, 2016 at 7:13 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> I has done simalar via tranfering requred menbers to new conference.
>>
>> ??, 12 ???. 2016 ?. ? 16:30, Deepika Yadav <deepikay at iiitd.ac.in>:
>>
>>> Hi All,
>>>
>>> Is there a way to start multiple sub conference calls in an on going
>>> conference of large number of callers.
>>>
>>> For example: There a conference going on between 20 callers, then some
>>> point in time, a conference between 3-4 callers spawns based on some
>>> condition so that what they discuss is not audible to rest of the caller
>>> and on termination of the condition these 3-4 people rejoins the original
>>> conference.
>>>
>>> Regards,
>>> Deepika
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mervyn.yeo at gmail.com  Tue Apr 12 18:43:26 2016
From: mervyn.yeo at gmail.com (Mervyn Yeo)
Date: Tue, 12 Apr 2016 22:43:26 +0800
Subject: [Freeswitch-users] How to register to a gateway/provider using an
 outbound proxy but have realm value in request URI?
Message-ID: <CA+cDheNjGPf0-kL+R7L9+z0oViC8vuWytPQfmOaGRY7vxERB3A@mail.gmail.com>

Hi all,

I'm trying to register with a provider without success. I'm able to do it
from my snom 300 phone as well as zoiper softphone. On the phones, all I
did was set

<fake substitute values below>

account : username
registrar : sip.provider.com
outbound proxy : 8.8.8.8

and it sends the register to 8.8.8.8 with a register packet like so in
http://pastebin.com/XME4UknQ

in fs, i created a provider with values http://pastebin.com/Q6i6Njxu and
it's not registering. the difference in the sip trace is that although the
request is sent to network address 8.8.8.8 the "Request-URI Host Part" uses
the values from my proxy setting instead of "sip.provider.com". Sip trace
from fs here -> http://pastebin.com/Pfgsu28s (Please ignore the user agent,
this was sent from fs. I was experimenting with the "user-agent-string"
param).

How can I setup fs so that it'll send to the outbound proxy address but the
Request-Line still shows

REGISTER sip:sip.provider.com SIP/2.0

Thanks,
Mervyn
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From mandra at gmail.com  Tue Apr 12 22:08:08 2016
From: mandra at gmail.com (Chris Mandra)
Date: Tue, 12 Apr 2016 14:08:08 -0400
Subject: [Freeswitch-users] curl connect timeout question
In-Reply-To: <F2A75B3F-6627-4809-9AA2-B296B4FBAC7E@jerris.com>
References: <CAJdRzN5MsyG8gcHEsbR3udnt4vGV0QWciypgJbAkwzNypsMJ7A@mail.gmail.com>
	<CAJdRzN4rDu=ZOsgSX-C6qGhwg2iTGdNDj7WrZKPwF+se=8LGBw@mail.gmail.com>
	<F2A75B3F-6627-4809-9AA2-B296B4FBAC7E@jerris.com>
Message-ID: <CAJdRzN5MGqyQvcULixqb=48Gh2OM8tJzN-=adS_E2nGq6yivvA@mail.gmail.com>

Thanks for responding. Yes - I make sure it's not connected to the internet
and run my module with the curl functions. I'd expect it to time out or
something, but instead it's crashing freeswitch.

On Mon, Apr 11, 2016 at 4:01 PM, Michael Jerris <mike at jerris.com> wrote:

> So you are adding code to it and then its crashing?
>
> On Apr 10, 2016, at 10:47 PM, Chris Mandra <mandra at gmail.com> wrote:
>
> Any ideas?
>
> On Saturday, April 9, 2016, Chris Mandra <mandra at gmail.com> wrote:
>
>> Hey guys, if I'm using curl in a module and the connection times out do
>> you have suggestion for how this should be handled?
>>
>> For example, if I disconnect freeswitch from the internet and have the
>> following settings:
>>
>>
>> curl_easy_setopt(curl_handle, CURLOPT_CONNECTTIMEOUT, 5L);
>> curl_easy_setopt(curl_handle, CURLOPT_TIMEOUT, 5L);
>> switch_curl_easy_setopt(curl_handle, CURLOPT_NOSIGNAL, 1L);
>>
>> it never times out and then crashes freeswitch. Is this expected behavior?
>> Is there something I should be doing instead?
>> Thanks, chris
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
mandra
c:410.258.5281
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From mike at jerris.com  Tue Apr 12 23:06:34 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 Apr 2016 15:06:34 -0400
Subject: [Freeswitch-users] curl connect timeout question
In-Reply-To: <CAJdRzN5MGqyQvcULixqb=48Gh2OM8tJzN-=adS_E2nGq6yivvA@mail.gmail.com>
References: <CAJdRzN5MsyG8gcHEsbR3udnt4vGV0QWciypgJbAkwzNypsMJ7A@mail.gmail.com>
	<CAJdRzN4rDu=ZOsgSX-C6qGhwg2iTGdNDj7WrZKPwF+se=8LGBw@mail.gmail.com>
	<F2A75B3F-6627-4809-9AA2-B296B4FBAC7E@jerris.com>
	<CAJdRzN5MGqyQvcULixqb=48Gh2OM8tJzN-=adS_E2nGq6yivvA@mail.gmail.com>
Message-ID: <CAEwTsAhA7qcwVPCekonwAyYOmEeVs-y5c4018nbfTcDz83YGhg@mail.gmail.com>

have you looked at the backtrace to see why it's crashing?  is it in this
new module you made?

On Tuesday, April 12, 2016, Chris Mandra <mandra at gmail.com> wrote:

> Thanks for responding. Yes - I make sure it's not connected to the
> internet and run my module with the curl functions. I'd expect it to time
> out or something, but instead it's crashing freeswitch.
>
> On Mon, Apr 11, 2016 at 4:01 PM, Michael Jerris <mike at jerris.com
> <javascript:_e(%7B%7D,'cvml','mike at jerris.com');>> wrote:
>
>> So you are adding code to it and then its crashing?
>>
>> On Apr 10, 2016, at 10:47 PM, Chris Mandra <mandra at gmail.com
>> <javascript:_e(%7B%7D,'cvml','mandra at gmail.com');>> wrote:
>>
>> Any ideas?
>>
>> On Saturday, April 9, 2016, Chris Mandra <mandra at gmail.com
>> <javascript:_e(%7B%7D,'cvml','mandra at gmail.com');>> wrote:
>>
>>> Hey guys, if I'm using curl in a module and the connection times out do
>>> you have suggestion for how this should be handled?
>>>
>>> For example, if I disconnect freeswitch from the internet and have the
>>> following settings:
>>>
>>>
>>> curl_easy_setopt(curl_handle, CURLOPT_CONNECTTIMEOUT, 5L);
>>> curl_easy_setopt(curl_handle, CURLOPT_TIMEOUT, 5L);
>>> switch_curl_easy_setopt(curl_handle, CURLOPT_NOSIGNAL, 1L);
>>>
>>> it never times out and then crashes freeswitch. Is this expected
>>> behavior?
>>> Is there something I should be doing instead?
>>> Thanks, chris
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> mandra
> c:410.258.5281
>
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From mike at jerris.com  Tue Apr 12 23:22:15 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 Apr 2016 15:22:15 -0400
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
References: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
Message-ID: <CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>

Billsec is set in reporting state, after you are out of dial plan.   What
are you trying to do based on that, I can suggest a few approaches

On Monday, April 11, 2016, Joel Serrano <joel at gogii.net> wrote:

> Hi,
>
> Is it possible to access ${billsec} in the dialplan after the bridge
> application (having hangup_after_bridge=false)??
>
> I want to log the billsec but currently it is always empty.
>
> I have a workaround in a test server using api_hangup_hook to call a lua
> script with session_in_hangup_hook=true, and inside the lua script I can
> access the billsec variable and thus write it to the logs.
>
> Is there no way to do this within the XML dialplan?
>
> Thanks!
>
> Best regards,
> Joel.
>
>
>
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From mike at jerris.com  Tue Apr 12 23:23:29 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 Apr 2016 15:23:29 -0400
Subject: [Freeswitch-users] Nested Conference
In-Reply-To: <CAFmZFiMqs1c3tTuN--pOQBJ6H-KOiU=h+a_WFOhhLtmtunhmGQ@mail.gmail.com>
References: <CAM99_YkccOiQgim8edR6VcJ94TY2zW769P7QEFB9W1Yyrtg-hQ@mail.gmail.com>
	<CAHtxdDesFsjp50qdRBVi2NF4788P7O0RBJidvsXi7Vea-jakBg@mail.gmail.com>
	<CAM99_Ym0ZrF2N+iRDaju+uTLe6HkDKQN7VSTJtEEH0e2A5_hxg@mail.gmail.com>
	<CAFmZFiMqs1c3tTuN--pOQBJ6H-KOiU=h+a_WFOhhLtmtunhmGQ@mail.gmail.com>
Message-ID: <CAEwTsAiO89Os-R-cPVMDH7AynYrj9x46ynaysxy2TJ=N9wsDNw@mail.gmail.com>

relate could do this but conference transfer is a much simpler approach if
you can use it

On Tuesday, April 12, 2016, Abaci B <abaci64 at gmail.com> wrote:

> or you can try the relate api to isolate those people within the main
> conference if that works for you
>
> On Tue, Apr 12, 2016 at 10:03 AM, Deepika Yadav <deepikay at iiitd.ac.in
> <javascript:_e(%7B%7D,'cvml','deepikay at iiitd.ac.in');>> wrote:
>
>> Thanks, I will definitely try transfer api.
>>
>> Regards,
>> Deepika
>>
>> On Tue, Apr 12, 2016 at 7:13 PM, Sergey Safarov <s.safarov at gmail.com
>> <javascript:_e(%7B%7D,'cvml','s.safarov at gmail.com');>> wrote:
>>
>>> I has done simalar via tranfering requred menbers to new conference.
>>>
>>> ??, 12 ???. 2016 ?. ? 16:30, Deepika Yadav <deepikay at iiitd.ac.in
>>> <javascript:_e(%7B%7D,'cvml','deepikay at iiitd.ac.in');>>:
>>>
>>>> Hi All,
>>>>
>>>> Is there a way to start multiple sub conference calls in an on going
>>>> conference of large number of callers.
>>>>
>>>> For example: There a conference going on between 20 callers, then some
>>>> point in time, a conference between 3-4 callers spawns based on some
>>>> condition so that what they discuss is not audible to rest of the caller
>>>> and on termination of the condition these 3-4 people rejoins the original
>>>> conference.
>>>>
>>>> Regards,
>>>> Deepika
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From joel at gogii.net  Tue Apr 12 23:39:45 2016
From: joel at gogii.net (Joel Serrano)
Date: Tue, 12 Apr 2016 12:39:45 -0700
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>
References: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
	<CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>
Message-ID: <CAMtXxQnyeqq8aHNEk8f_EsvraiPrtpF0se91v8dOJx6ehhKwbg@mail.gmail.com>

Hi Michael,

I want to call an API basically with the following info:

sip_from_uri
destination_number
billsec

I found away using "api_hangup_hook=lua script.lua" +
"session_in_hangup_hook=true", but, I don't like it... I would prefer using
curl from within the dialplan.

Can you give me your suggestion? I have also asked in IRC and using the
CDRs looks like the best way to go, so we will probably follow that, but I
would still like to now if it is possible and how.

Thanks!
Joel.



On Tue, Apr 12, 2016 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:

> Billsec is set in reporting state, after you are out of dial plan.   What
> are you trying to do based on that, I can suggest a few approaches
>
>
> On Monday, April 11, 2016, Joel Serrano <joel at gogii.net> wrote:
>
>> Hi,
>>
>> Is it possible to access ${billsec} in the dialplan after the bridge
>> application (having hangup_after_bridge=false)??
>>
>> I want to log the billsec but currently it is always empty.
>>
>> I have a workaround in a test server using api_hangup_hook to call a lua
>> script with session_in_hangup_hook=true, and inside the lua script I can
>> access the billsec variable and thus write it to the logs.
>>
>> Is there no way to do this within the XML dialplan?
>>
>> Thanks!
>>
>> Best regards,
>> 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 phenix at vfemail.net  Wed Apr 13 00:20:25 2016
From: phenix at vfemail.net (Tanguy)
Date: Tue, 12 Apr 2016 22:20:25 +0200
Subject: [Freeswitch-users] nibblebill do not terminate calls properly
In-Reply-To: <CAHtxdDfrtThmDh0GpETSiHDBwRFX4vy+-WhBXW6mb-pH3Q-nfw@mail.gmail.com>
References: <570A3728.6020900@vfemail.net>
	<CAFLo2Qzi5oSHzXMyFEiDExdzHsmew_2a+Wne_f1Z7H6PUNTe4A@mail.gmail.com>
	<B700FF78-C2BA-41C3-8E9C-AD49B658DD60@freeswitch.org>
	<CAFLo2QwD6eM6=B1TUa0ObWa9-3GLraVJxeu0UfaV6QS4zkeLmA@mail.gmail.com>
	<B6118ED6-AAA7-462D-B4CB-06D3713948A8@d-man.org>
	<20160411064240.Horde.6M0V77qRE_MGQRkFnMD4-w1@www.vfemail.net>
	<CAHtxdDeTKfpCrN1snWDTnuCTvvezfrzaKPWR6pf2V8M-NMRdwA@mail.gmail.com>
	<570C15D5.9010000@vfemail.net>
	<CAHtxdDfrtThmDh0GpETSiHDBwRFX4vy+-WhBXW6mb-pH3Q-nfw@mail.gmail.com>
Message-ID: <570D5889.4050407@vfemail.net>

Hello again

I tried to force heartbeat using dialplan and it's works: When the 
balance reach 0, my calls are hanged up and i can't place a new call.

I don't understand why this did not work with *global_heartbeat* but 
it's not really annoying if i can bypass this.

<extension name="international" >
    <condition field="destination_number" expression="^00(\d{6,20})$" >
         <action application="set" data="nibble_account=${accountcode}"/>
         <action application="set" data="nibble_rate=0.1"/>
*<action application="nibblebill" data="heartbeat 30"/>*
         <action application="nibblebill" data="flush"/>
        <action application="bridge" data="sofia/gateway/gw_idt/$1" />
    </condition>
</extension>


On 11/04/2016 23:55, Sergey Safarov wrote:
>
> Probable. Yes.
> On my servers, all work as documented in confluence.
>
>
> On Tue, Apr 12, 2016, 00:24 Tanguy <phenix at vfemail.net 
> <mailto:phenix at vfemail.net>> wrote:
>
>     Hello Sergey
>
>     Thanks for your advice, now i can't place a new call with a
>     negative balance
>
>
>     [DEBUG] mod_nibblebill.c:488 Attempting to bill at $0.1 per minute
>     to account company.voip.domain.com <http://company.voip.domain.com>
>     [DEBUG] mod_nibblebill.c:500 Not billing account
>     company.voip.domain.com <http://company.voip.domain.com> - call is
>     not in answered state
>
>     [DEBUG] mod_nibblebill.c:420 Doing lookup query
>     [DEBUG] mod_nibblebill.c:428 Retrieved current balance for account
>     account company.voip.domain.com <http://company.voip.domain.com>
>     (balance = -0.028433)
>     [DEBUG] mod_nibblebill.c:504 Comparing -0.028433 to hangup balance
>     of 0.000000
>
>     But currents calls are not hanged up when my balance reach 0 and
>     the database is never updated during the call ( but only after
>     hangup ), i probably have a issue with the "global_heartbeat"
>     parameter.
>
>
>
>
>
>
>
>     On 11/04/2016 21:59, Sergey Safarov wrote:
>>
>>     I think problem in order of flush command. To take effect nibble
>>     rate must be set before execution flush.
>>
>>
>>     On Mon, Apr 11, 2016, 22:50 tanguy <phenix at vfemail.net
>>     <mailto:phenix at vfemail.net>> wrote:
>>
>>         > Sergey
>>         Thank you for your piece of dialplan but i don't think it's the
>>         solution. You offer a different solution to set the nibble_rate
>>         variable, but in my sample case this variable is properly set
>>         because
>>         the call is billed.
>>
>>         > Luis
>>         A bug, maybe. I don't remember my freeswitch version ( i
>>         can't connect
>>         to my servers yet ) but i think it's 1.4 series.
>>
>>         I can try the last 1.6 version, i will also try a standard
>>         freeswitch
>>         sample configuration ( without config files provided by
>>         fusionpbx )
>>
>
>     _________________________________________________________________________
>     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 mike at jerris.com  Wed Apr 13 00:42:14 2016
From: mike at jerris.com (Michael Jerris)
Date: Tue, 12 Apr 2016 15:42:14 -0500
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <CAMtXxQnyeqq8aHNEk8f_EsvraiPrtpF0se91v8dOJx6ehhKwbg@mail.gmail.com>
References: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
	<CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>
	<CAMtXxQnyeqq8aHNEk8f_EsvraiPrtpF0se91v8dOJx6ehhKwbg@mail.gmail.com>
Message-ID: <5123CC15-8A19-453E-BB13-A8B230BC3DFC@jerris.com>

I'm probably going to push towards cdr, but what are you trying to actually do with that info?

> On Apr 12, 2016, at 2:39 PM, Joel Serrano <joel at gogii.net> wrote:
> 
> Hi Michael, 
> 
> I want to call an API basically with the following info:
> 
> sip_from_uri
> destination_number
> billsec
> 
> I found away using "api_hangup_hook=lua script.lua" + "session_in_hangup_hook=true", but, I don't like it... I would prefer using curl from within the dialplan.
> 
> Can you give me your suggestion? I have also asked in IRC and using the CDRs looks like the best way to go, so we will probably follow that, but I would still like to now if it is possible and how.
> 
> Thanks!
> Joel.
> 
> 
> 
> On Tue, Apr 12, 2016 at 12:22 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> Billsec is set in reporting state, after you are out of dial plan.   What are you trying to do based on that, I can suggest a few approaches
> 
> 
> On Monday, April 11, 2016, Joel Serrano <joel at gogii.net <mailto:joel at gogii.net>> wrote:
> Hi, 
> 
> Is it possible to access ${billsec} in the dialplan after the bridge application (having hangup_after_bridge=false)??
> 
> I want to log the billsec but currently it is always empty.
> 
> I have a workaround in a test server using api_hangup_hook to call a lua script with session_in_hangup_hook=true, and inside the lua script I can access the billsec variable and thus write it to the logs.
> 
> Is there no way to do this within the XML dialplan?
> 
> Thanks!
> 
> Best regards, 
> Joel.
> 

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From msc at freeswitch.org  Wed Apr 13 01:18:24 2016
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Apr 2016 14:18:24 -0700
Subject: [Freeswitch-users] Cannot get automatic call recording working.
In-Reply-To: <42b77762b2cff94364249083ee8113dd@mail.gmail.com>
References: <42b77762b2cff94364249083ee8113dd@mail.gmail.com>
Message-ID: <CAKzWOxVQv+U3A9CabW72n-YXdv0Crupk_4MhbaakTwOovyq6bg@mail.gmail.com>

Capture the complete log output of a test call and drop it into
pastebin.freeswitch.org. The output will definitely tell you exactly what
the dialplan is doing, so if it is attempting to initiate the
record_session app then you'll see any errors that may have resulted.

-MSC

On Mon, Apr 11, 2016 at 12:54 PM, Piotr Starzyk <
pstarzyk at general-devices.com> wrote:

> I?m running FreeSWITCH in console mode on a Windows 7 machine, using
> unmodified demo setup.  I?m having a hard time getting automatic call
> recording to work.  I?m using the following instructions to record a call
> (by modifying  Local_Extension dialplan):
>
>
>
>
> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+record_session
>
> *Record Calls To Extensions*
>
> To record all phone calls between extensions do the following. Make a
> directory under freeswitch/recordings/archive/. Then edit Local_Extension
> in dialplan in conf/dialplan/default.xml
>
> <extension name="Local_Extension">
>
>     <condition field="destination_number" expression="^(10[01][0-8])$">
>
> Then add the following actions.
>
> <action application="set" data="RECORD_TITLE=Recording
> ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/>
>
> <action application="set" data="RECORD_COPYRIGHT=(c) 2011"/>
>
> <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
>
> <action application="set" data="RECORD_ARTIST=FreeSWITCH"/>
>
> <action application="set" data="RECORD_COMMENT=FreeSWITCH"/>
>
> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>
>
> <action application="set" data="RECORD_STEREO=true"/>
>
> <action application="record_session"
> data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>
>
>
>
>
>
> But when I place a call, no files are getting created in the recordings or
> archive folder.  I tried calling both ways, still nothing.  I am running in
> admin mode, and have firewalls disabled.
>
>
>
> When looking through the logs, the only two lines relevant to the
> recording are the following:
>
>
>
> 892ef279-18c7-4d8d-8795-fa1c366b68c7 EXECUTE sofia/internal/
> 1000 at 10.10.10.77
>  bind_meta_app(2 b s record_session::C:/Program Files/FreeSWITCH/recordings/1000.2016-04-11-15-16-58.wav)
>
>
> 892ef279-18c7-4d8d-8795-fa1c366b68c7 2016-04-11 15:16:58.521344 [INFO] switch_ivr_async.c:4152 Bound B-Leg: *2 record_session::C:/Program Files/FreeSWITCH/recordings/1000.2016-04-11-15-16-58.wav
>
>
>
> From FreeSWITCH cookbook, I found out they allow manual recording by
> pressing *2 on receiving end.  When I tested that, it did work, and the wav
> file was created in the recordings directory, so the recording module seems
> to be working.  But for some reason, I still can?t get the automatic
> session recording to work.  What am I doing wrong?
>
>
>
>
>
> 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 athompson at successos.com  Wed Apr 13 03:02:27 2016
From: athompson at successos.com (Adrian Thompson)
Date: Tue, 12 Apr 2016 17:02:27 -0600
Subject: [Freeswitch-users] Call forced held after answer
Message-ID: <009f01d1950f$62bbadd0$28330970$@successos.com>

 

Hello,

 

Any insight into this would be very helpful!  I have exhausted my search of
Freeswitch and Polycom. I have no idea how to solve this issue L

 

I have a great working setup with VVX 300s 400s 500s and a IP 6000, however
there is a strange issue going on with an IP 7000 I have.

 

When calling to or from IP 7000

1.       It rings fine with audio on both ends

2.       Press answer on either side

3.       Call is immediately "Held" on IP 7000

4.       Green lights flash on IP 7000 as if the call was placed on hold by
the other phone, but call is not on hold by other phone

5.       No audio (probably because on hold)

6.       End call normally

 

NAT settings are correct

 

-          Formatted IP 7000 and reset all config

-          Reinstalled firmware

-          Disconnected from provisioning server

-          Created  manual config from scratch on phone web interface

-          Removed all encryption

 

IP 7000 logs during call:

003219.281|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2.

003219.341|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2.

003219.500|net  |4|03|rtosNetwork: netwTask() - Can't find associated CCB.

003220.080|net  |4|03|rtosNetwork net02: netwSend() - sendto() call failed.
fd 1123346772 errno=130

003220.861|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (0) 2.

003221.600|net  |4|03|rtosNetwork: netwTask() - Can't find associated CCB.

003223.600|net  |4|03|rtosNetwork: netwTask() - Can't find associated CCB.

Then press hang-up:

003227.600|net  |4|03|rtosNetwork: netwTask() - Can't find associated CCB.

003227.985|net  |2|03|NWIF: nw_setlocalhold() - setting local hold (1) 2.

003242.982|sip  |5|03|Can not decode the packet

 

-          Setting local hold (0) should be fine and it works on all other
phones I have.

-          Setting local hold (1) after pressing hang-up is weird but is
happening after hang-up anyway.

-          I wonder why that send error is in the logs. it looks like an
operating system error.. Maybe a driver error?

 

Thanks in advance!!

Adrian

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From jelena at misticnabica.hr  Wed Apr 13 04:09:39 2016
From: jelena at misticnabica.hr (jelena at misticnabica.hr)
Date: Wed, 13 Apr 2016 00:09:39 GMT
Subject: [Freeswitch-users] Call forced held after answer
Message-ID: <878CF0107C5A4AB68FBFBF779222E7C6.MAI@server2.totohost.hr>



From pstarzyk at general-devices.com  Wed Apr 13 15:27:07 2016
From: pstarzyk at general-devices.com (Piotr Starzyk)
Date: Wed, 13 Apr 2016 07:27:07 -0400
Subject: [Freeswitch-users] Cannot get automatic call recording working.
In-Reply-To: <CAKzWOxVQv+U3A9CabW72n-YXdv0Crupk_4MhbaakTwOovyq6bg@mail.gmail.com>
References: <42b77762b2cff94364249083ee8113dd@mail.gmail.com>
	<CAKzWOxVQv+U3A9CabW72n-YXdv0Crupk_4MhbaakTwOovyq6bg@mail.gmail.com>
Message-ID: <6af3cf80dbb08a8bb2dc23376d48ee07@mail.gmail.com>

I found the problem - it was my fault.  Before making changes to the
default.xml dialplan file, I made a copy of it in the same folder and
called it:

default ? before recording.xml



Then I made changes to default xml.



Turns out FreeSWITCH was parsing the backup file I made, instead of the
actual file, so none of my changes were taking effect.



Once I moved the backup file out of the dialplan folder, the recording
started working as expected.









*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
Collins
*Sent:* Tuesday, April 12, 2016 5:18 PM
*To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
*Subject:* Re: [Freeswitch-users] Cannot get automatic call recording
working.



Capture the complete log output of a test call and drop it into
pastebin.freeswitch.org. The output will definitely tell you exactly what
the dialplan is doing, so if it is attempting to initiate the
record_session app then you'll see any errors that may have resulted.



-MSC



On Mon, Apr 11, 2016 at 12:54 PM, Piotr Starzyk <
pstarzyk at general-devices.com> wrote:

I?m running FreeSWITCH in console mode on a Windows 7 machine, using
unmodified demo setup.  I?m having a hard time getting automatic call
recording to work.  I?m using the following instructions to record a call
(by modifying  Local_Extension dialplan):



https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+record_session

*Record Calls To Extensions*

To record all phone calls between extensions do the following. Make a
directory under freeswitch/recordings/archive/. Then edit Local_Extension
in dialplan in conf/dialplan/default.xml

<extension name="Local_Extension">

    <condition field="destination_number" expression="^(10[01][0-8])$">

Then add the following actions.

<action application="set" data="RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/>

<action application="set" data="RECORD_COPYRIGHT=(c) 2011"/>

<action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>

<action application="set" data="RECORD_ARTIST=FreeSWITCH"/>

<action application="set" data="RECORD_COMMENT=FreeSWITCH"/>

<action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>

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

<action application="record_session"
data="$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/>





But when I place a call, no files are getting created in the recordings or
archive folder.  I tried calling both ways, still nothing.  I am running in
admin mode, and have firewalls disabled.



When looking through the logs, the only two lines relevant to the recording
are the following:



892ef279-18c7-4d8d-8795-fa1c366b68c7 EXECUTE sofia/internal/1000 at 10.10.10.77
 bind_meta_app(2 b s record_session::C:/Program
Files/FreeSWITCH/recordings/1000.2016-04-11-15-16-58.wav)

892ef279-18c7-4d8d-8795-fa1c366b68c7 2016-04-11 15:16:58.521344 [INFO]
switch_ivr_async.c:4152 Bound B-Leg: *2 record_session::C:/Program
Files/FreeSWITCH/recordings/1000.2016-04-11-15-16-58.wav



>From FreeSWITCH cookbook, I found out they allow manual recording by
pressing *2 on receiving end.  When I tested that, it did work, and the wav
file was created in the recordings directory, so the recording module seems
to be working.  But for some reason, I still can?t get the automatic
session recording to work.  What am I doing wrong?





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 italo at freeswitch.org  Wed Apr 13 15:46:45 2016
From: italo at freeswitch.org (=?utf-8?q?=C3=8Dtalo_Rossi?=)
Date: Wed, 13 Apr 2016 04:46:45 -0700 (PDT)
Subject: [Freeswitch-users] Thursday FreeSWITCH Bug Hunt - TOMORROW!
Message-ID: <6xky154ht4d71qzlg493of6bx-0@mailer.nylas.com>

FreeSWITCHers,

  

If you have questions, need attention for any JIRA you reported or want to
contribute, join us!

  

Tomorrow 2PM CST for the Thursday FreeSWITCH Bug Hunt!

  

Where? [conference.freeswitch.org/vc/#/?autocall=888](https://conference.frees
witch.org/vc/#/?autocall=888
"https://conference.freeswitch.org/vc/#/?autocall=888" )

  

Chat? <https://hipchat.freeswitch.org/gYzPMg3Oc>

  

What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support!

  

Help us help you, Join the Bug Hunt!

?talo Rossi

italo at freeswitch.org

IRC chat.freenode.net #freeswitch #freeswitch-dev

Bugs? https://freeswitch.org/jira

Docs? https://freeswitch.org/jira

Chat? https://hipchat.freeswitch.org/gUdAgy0m6

  

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From udy786 at gmail.com  Wed Apr 13 17:18:08 2016
From: udy786 at gmail.com (Uday kumar)
Date: Wed, 13 Apr 2016 18:48:08 +0530
Subject: [Freeswitch-users] SMS/SMPP CDR
Message-ID: <CAOfqg5af4h32hoyFcXYqNeME-ViLfkvbmRTUrJOrVNnKwENe0Q@mail.gmail.com>

Hello All,

I am using SMS/SMPP on Freeswitch 1.6 with CentOS 7.2. Is this possible to
mange CDR report for SMS/SMPP in database like from user, to user, message
and time save in database?

Please advise.



-- 
Thanks & Regard
Uday
Site:- www.shareyourknowledge.in
Mobile:- +91-9377579349
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From piotrek.gregor at gmail.com  Wed Apr 13 21:47:02 2016
From: piotrek.gregor at gmail.com (Piotr Gregor)
Date: Wed, 13 Apr 2016 19:47:02 +0200
Subject: [Freeswitch-users] module question
In-Reply-To: <CAJdRzN4QLpjxFPd=LvE32SYgFUN_ekZ0Q3jPEO=vsrCprK5Zbg@mail.gmail.com>
References: <CAJdRzN6C3W4vR6pZBAzDirfC11txdJq=HoCWbPB3EtJQu6Z6Yg@mail.gmail.com>
	<56D6EC6A.3000609@mst.edu>
	<CAJdRzN5u+2xVyRp7EO_cPfD0ZJ7m5-vjjpLd1B8FnrXx4=s7zA@mail.gmail.com>
	<4a6001d17494$1db2bd50$591837f0$@freeswitch.org>
	<56D7031B.8020809@mst.edu>
	<CAJdRzN73YUfik4oguRduj3Lj5JTVVVHxktBRmwgd4NALhP6k8g@mail.gmail.com>
	<4af401d174a4$9a7718d0$cf654a70$@freeswitch.org>
	<CAJdRzN7=ky6im8ER2F2FB7hqjTSnfcmXQhseSK2r2iTMPHmZVw@mail.gmail.com>
	<CAJdRzN72KD3Vfy=AMcpHQcZ1c5tk6d1LM6u+ER-ix6rA2w-BkA@mail.gmail.com>
	<CAJdRzN5GcK2jRMnACnqN97L1zbDMsuYk2HYRLyf5xuCguDyK5g@mail.gmail.com>
	<CAJdRzN6xDZfN_rd_hwJgLEmkR+w4hpEG2du6j1grnzJPYazCwA@mail.gmail.com>
	<CAJdRzN5Ui64U1J+8t2dd2ygcYvPqVB2O5NG2TULh+HzEJ4An4Q@mail.gmail.com>
	<56D998B0.1020305@mst.edu>
	<CAJdRzN7-24zrzvWu7zm12+NnY5Bq_3CSJ69vtU1N-PE7FW=g=w@mail.gmail.com>
	<CAJdRzN7ZP09P4O+Ms13VdVWbQRX8SCVjU9EYcaHBiZ_bW0dkbA@mail.gmail.com>
	<56D9D0AD.1050506@mst.edu>
	<CAJdRzN5huHwdr3i=k31LnC5FKJxh5omQ5jY049skBv+gHwZnvQ@mail.gmail.com>
	<56D9E1B7.7040404@mst.edu>
	<CAJdRzN6YRutiJnMFBHciH6sO=+++f9HBP4=BB0g_qzXqeEqORw@mail.gmail.com>
	<401F3406-FA7C-430A-A8BC-C39DF816ED8E@jerris.com>
	<CAJdRzN7b+xdfoMnYQeP88ma3_cyCpOBqfioEP_sWsOqcAtqRWQ@mail.gmail.com>
	<FCA816DA-69F2-4566-82B6-F463D56DDA6E@jerris.com>
	<CAO3obxaEg3C7oLfiP3xM7rynd3tHD3s0zv5h6vbiOg_TUTUH2w@mail.gmail.com>
	<CAJdRzN7CHWpX8q=YYC3GcNKd8GEuyG_B7GUaT1V4Fh_fLeF=JQ@mail.gmail.com>
	<CAO3obxZQ7B+wkONjLzLCcCjF_vPXmWvN_C-1WqBqEBQzSdfwdg@mail.gmail.com>
	<CAJdRzN4QLpjxFPd=LvE32SYgFUN_ekZ0Q3jPEO=vsrCprK5Zbg@mail.gmail.com>
Message-ID: <CAO3obxZen0Nd6gvt2gHYBTGZdjt8ve3wueNOSX+Je7RabsLY-A@mail.gmail.com>

Hi Chris,

have you fixed your problems with module reloading?

cheers,
Piotr
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From joel at gogii.net  Thu Apr 14 02:52:56 2016
From: joel at gogii.net (Joel Serrano)
Date: Wed, 13 Apr 2016 15:52:56 -0700
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <5123CC15-8A19-453E-BB13-A8B230BC3DFC@jerris.com>
References: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
	<CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>
	<CAMtXxQnyeqq8aHNEk8f_EsvraiPrtpF0se91v8dOJx6ehhKwbg@mail.gmail.com>
	<5123CC15-8A19-453E-BB13-A8B230BC3DFC@jerris.com>
Message-ID: <CAMtXxQ=8Q7iyzx2JvZ=o-nwBnyaxNhkF0rC2rp1cJqt0t4w9gA@mail.gmail.com>

Hi Michael,

The problem is that occasionally we see the following in FS log:

2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:385 Got error [0] posting to
web server [http://cdrs.example.com/queue]
2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:392 Retry will be with url [
http://cdrs.example.com/queue]

When that happens, the CDR gets written to disc (so the CDR is not
processed on time, thus our user is not billed for that call yet).

We have a task that checks every 5 minutes for failed CDRs and we resubmit
them to the processing queue, but I need to find out why FS is failing.

I have checked the access logs on the receiving side, but all requests
return a 200 OK.

Now I have enabled logs in the loadbalancer between FS and the queues to
see if I get more vision from there...


A workaround was to make API requests directly from the dialplan with the
billsec info, but as all suggest going with CDRs, I am now focussed on
finding the root cause for that error.

Any help/suggestion is more than welcome.


Thank you.

Best regards,
Joel.




On Tue, Apr 12, 2016 at 1:42 PM, Michael Jerris <mike at jerris.com> wrote:

> I'm probably going to push towards cdr, but what are you trying to
> actually do with that info?
>
> On Apr 12, 2016, at 2:39 PM, Joel Serrano <joel at gogii.net> wrote:
>
> Hi Michael,
>
> I want to call an API basically with the following info:
>
> sip_from_uri
> destination_number
> billsec
>
> I found away using "api_hangup_hook=lua script.lua" +
> "session_in_hangup_hook=true", but, I don't like it... I would prefer using
> curl from within the dialplan.
>
> Can you give me your suggestion? I have also asked in IRC and using the
> CDRs looks like the best way to go, so we will probably follow that, but I
> would still like to now if it is possible and how.
>
> Thanks!
> Joel.
>
>
>
> On Tue, Apr 12, 2016 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Billsec is set in reporting state, after you are out of dial plan.   What
>> are you trying to do based on that, I can suggest a few approaches
>>
>>
>> On Monday, April 11, 2016, Joel Serrano <joel at gogii.net> wrote:
>>
>>> Hi,
>>>
>>> Is it possible to access ${billsec} in the dialplan after the bridge
>>> application (having hangup_after_bridge=false)??
>>>
>>> I want to log the billsec but currently it is always empty.
>>>
>>> I have a workaround in a test server using api_hangup_hook to call a lua
>>> script with session_in_hangup_hook=true, and inside the lua script I can
>>> access the billsec variable and thus write it to the logs.
>>>
>>> Is there no way to do this within the XML dialplan?
>>>
>>> Thanks!
>>>
>>> Best regards,
>>> 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 jelena at misticnabica.hr  Thu Apr 14 03:03:38 2016
From: jelena at misticnabica.hr (jelena at misticnabica.hr)
Date: Wed, 13 Apr 2016 23:03:38 GMT
Subject: [Freeswitch-users] Billsec in dialplan
Message-ID: <93B147C8C67941FBA807EA8DADA809EF.MAI@server2.totohost.hr>



From joel at gogii.net  Thu Apr 14 03:07:56 2016
From: joel at gogii.net (Joel Serrano)
Date: Wed, 13 Apr 2016 16:07:56 -0700
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <93B147C8C67941FBA807EA8DADA809EF.MAI@server2.totohost.hr>
References: <93B147C8C67941FBA807EA8DADA809EF.MAI@server2.totohost.hr>
Message-ID: <CAMtXxQ=6z2d9MoLDWAFy0sTPZcJbtXnT3KUseqwOJbAzsd0+Yg@mail.gmail.com>

Hi Jelena,

Your answer came empty, can you try to resend it with the text please?

Thanks,
Joel.

On Wed, Apr 13, 2016 at 4:03 PM, <jelena at misticnabica.hr> wrote:

>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Thu Apr 14 03:13:46 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 13 Apr 2016 18:13:46 -0500
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <93B147C8C67941FBA807EA8DADA809EF.MAI@server2.totohost.hr>
References: <93B147C8C67941FBA807EA8DADA809EF.MAI@server2.totohost.hr>
Message-ID: <56CE6855-334B-4BDF-BF69-98EB6C7F6CDA@freeswitch.org>

Please stop sending blank emails to the list 

Sent from my iPhone

> On Apr 13, 2016, at 6:03 PM, "" <jelena at misticnabica.hr> <jelena at misticnabica.hr> wrote:
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From krice at freeswitch.org  Thu Apr 14 03:15:28 2016
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 13 Apr 2016 18:15:28 -0500
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <CAMtXxQ=8Q7iyzx2JvZ=o-nwBnyaxNhkF0rC2rp1cJqt0t4w9gA@mail.gmail.com>
References: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
	<CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>
	<CAMtXxQnyeqq8aHNEk8f_EsvraiPrtpF0se91v8dOJx6ehhKwbg@mail.gmail.com>
	<5123CC15-8A19-453E-BB13-A8B230BC3DFC@jerris.com>
	<CAMtXxQ=8Q7iyzx2JvZ=o-nwBnyaxNhkF0rC2rp1cJqt0t4w9gA@mail.gmail.com>
Message-ID: <EC4896F1-829D-4C95-B7E5-AB9BD66021A1@freeswitch.org>

The correct answer here is you must check more often i would say at a minimum every 15 seconds. Search the list for me discussing cdrs in volume in the past or contact me off list if you need pro help

Sent from my iPhone

> On Apr 13, 2016, at 5:52 PM, Joel Serrano <joel at gogii.net> wrote:
> 
> Hi Michael, 
> 
> The problem is that occasionally we see the following in FS log:
> 
> 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:385 Got error [0] posting to web server [http://cdrs.example.com/queue]
> 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:392 Retry will be with url [http://cdrs.example.com/queue]
> 
> When that happens, the CDR gets written to disc (so the CDR is not processed on time, thus our user is not billed for that call yet).
> 
> We have a task that checks every 5 minutes for failed CDRs and we resubmit them to the processing queue, but I need to find out why FS is failing.
> 
> I have checked the access logs on the receiving side, but all requests return a 200 OK.
> 
> Now I have enabled logs in the loadbalancer between FS and the queues to see if I get more vision from there...
> 
> 
> A workaround was to make API requests directly from the dialplan with the billsec info, but as all suggest going with CDRs, I am now focussed on finding the root cause for that error.
> 
> Any help/suggestion is more than welcome.
> 
> 
> Thank you.
> 
> Best regards, 
> Joel.
> 
> 
> 
> 
>> On Tue, Apr 12, 2016 at 1:42 PM, Michael Jerris <mike at jerris.com> wrote:
>> I'm probably going to push towards cdr, but what are you trying to actually do with that info?
>> 
>>> On Apr 12, 2016, at 2:39 PM, Joel Serrano <joel at gogii.net> wrote:
>>> 
>>> Hi Michael, 
>>> 
>>> I want to call an API basically with the following info:
>>> 
>>> sip_from_uri
>>> destination_number
>>> billsec
>>> 
>>> I found away using "api_hangup_hook=lua script.lua" + "session_in_hangup_hook=true", but, I don't like it... I would prefer using curl from within the dialplan.
>>> 
>>> Can you give me your suggestion? I have also asked in IRC and using the CDRs looks like the best way to go, so we will probably follow that, but I would still like to now if it is possible and how.
>>> 
>>> Thanks!
>>> Joel.
>>> 
>>> 
>>> 
>>>> On Tue, Apr 12, 2016 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:
>>>> Billsec is set in reporting state, after you are out of dial plan.   What are you trying to do based on that, I can suggest a few approaches
>>>> 
>>>> 
>>>>> On Monday, April 11, 2016, Joel Serrano <joel at gogii.net> wrote:
>>>>> Hi, 
>>>>> 
>>>>> Is it possible to access ${billsec} in the dialplan after the bridge application (having hangup_after_bridge=false)??
>>>>> 
>>>>> I want to log the billsec but currently it is always empty.
>>>>> 
>>>>> I have a workaround in a test server using api_hangup_hook to call a lua script with session_in_hangup_hook=true, and inside the lua script I can access the billsec variable and thus write it to the logs.
>>>>> 
>>>>> Is there no way to do this within the XML dialplan?
>>>>> 
>>>>> Thanks!
>>>>> 
>>>>> Best regards, 
>>>>> 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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From shakumarsoftware at gmail.com  Thu Apr 14 03:33:23 2016
From: shakumarsoftware at gmail.com (Sharath Kumar)
Date: Wed, 13 Apr 2016 17:33:23 -0600
Subject: [Freeswitch-users] How to register to a gateway/provider using
 an outbound proxy but have realm value in request URI?
In-Reply-To: <CA+cDheNjGPf0-kL+R7L9+z0oViC8vuWytPQfmOaGRY7vxERB3A@mail.gmail.com>
References: <CA+cDheNjGPf0-kL+R7L9+z0oViC8vuWytPQfmOaGRY7vxERB3A@mail.gmail.com>
Message-ID: <CA+wj8uar0ynkZqnGbNYCJ3dAk9U10+vJjg5U-s661Vg-Xkm8tA@mail.gmail.com>

Can't you not set the "proxy" to "sip.provider.com" ?

On Tue, Apr 12, 2016 at 8:43 AM, Mervyn Yeo <mervyn.yeo at gmail.com> wrote:

> Hi all,
>
> I'm trying to register with a provider without success. I'm able to do it
> from my snom 300 phone as well as zoiper softphone. On the phones, all I
> did was set
>
> <fake substitute values below>
>
> account : username
> registrar : sip.provider.com
> outbound proxy : 8.8.8.8
>
> and it sends the register to 8.8.8.8 with a register packet like so in
> http://pastebin.com/XME4UknQ
>
> in fs, i created a provider with values http://pastebin.com/Q6i6Njxu and
> it's not registering. the difference in the sip trace is that although the
> request is sent to network address 8.8.8.8 the "Request-URI Host Part" uses
> the values from my proxy setting instead of "sip.provider.com". Sip trace
> from fs here -> http://pastebin.com/Pfgsu28s (Please ignore the user
> agent, this was sent from fs. I was experimenting with the
> "user-agent-string" param).
>
> How can I setup fs so that it'll send to the outbound proxy address but
> the Request-Line still shows
>
> REGISTER sip:sip.provider.com SIP/2.0
>
> Thanks,
> Mervyn
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From arsenman at connectto.com  Thu Apr 14 04:29:18 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Wed, 13 Apr 2016 17:29:18 -0700
Subject: [Freeswitch-users] help, with Verto - verto-min.js configuration
Message-ID: <570EE45E.3060008@connectto.com>


Please help, with Verto - verto-min.js configuration
After changing username, password for STUN and TURN server looks like my 
TURN server doesn't receive any messages form Webrtc phone
here is my config:


STUN={url:!moz?'stun:turn.mydomain.com:3478':'stun:204.174.104.38'};var 
TURN={url:'turn:ars at tu
rn.mydomain.com:3478 
<http://rn.mydomain.com:3478>',credential:'ars123'};var 
iceServers=null;if(options.iceServers){var 
tmp=options.iceServers;if(typeof(tmp)==
="boolean"){tmp=null;}
if(tmp&&!(typeof(tmp)=="object"&&tmp.constructor===Array)){console.warn("iceServers 
must be an array, reverting to default ice serve
rs");tmp=null;}
iceServers={iceServers:tmp||[STUN]};if(!moz&&!tmp){if(parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2])>=28)TURN={u
rl:'turn:turn.mydomain.com:3478 
<http://turn.mydomain.com:3478>',credential:'ars123',username:'ars'};
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From hardyanto.donny at gmail.com  Thu Apr 14 04:57:22 2016
From: hardyanto.donny at gmail.com (Donny Hardyanto)
Date: Thu, 14 Apr 2016 07:57:22 +0700
Subject: [Freeswitch-users] How to register to a gateway/provider using
 an outbound proxy but have realm value in request URI?
In-Reply-To: <CA+wj8uar0ynkZqnGbNYCJ3dAk9U10+vJjg5U-s661Vg-Xkm8tA@mail.gmail.com>
References: <CA+cDheNjGPf0-kL+R7L9+z0oViC8vuWytPQfmOaGRY7vxERB3A@mail.gmail.com>
	<CA+wj8uar0ynkZqnGbNYCJ3dAk9U10+vJjg5U-s661Vg-Xkm8tA@mail.gmail.com>
Message-ID: <CAEhEJ-XRhOL4ry4Oz+gN14knjt6JedVPbzi=zMqYq2ibHA6qLQ@mail.gmail.com>

Yes. Set outbound proxy to sip.provider.com
On Apr 14, 2016 06:34, "Sharath Kumar" <shakumarsoftware at gmail.com> wrote:

> Can't you not set the "proxy" to "sip.provider.com" ?
>
> On Tue, Apr 12, 2016 at 8:43 AM, Mervyn Yeo <mervyn.yeo at gmail.com> wrote:
>
>> Hi all,
>>
>> I'm trying to register with a provider without success. I'm able to do it
>> from my snom 300 phone as well as zoiper softphone. On the phones, all I
>> did was set
>>
>> <fake substitute values below>
>>
>> account : username
>> registrar : sip.provider.com
>> outbound proxy : 8.8.8.8
>>
>> and it sends the register to 8.8.8.8 with a register packet like so in
>> http://pastebin.com/XME4UknQ
>>
>> in fs, i created a provider with values http://pastebin.com/Q6i6Njxu and
>> it's not registering. the difference in the sip trace is that although the
>> request is sent to network address 8.8.8.8 the "Request-URI Host Part" uses
>> the values from my proxy setting instead of "sip.provider.com". Sip
>> trace from fs here -> http://pastebin.com/Pfgsu28s (Please ignore the
>> user agent, this was sent from fs. I was experimenting with the
>> "user-agent-string" param).
>>
>> How can I setup fs so that it'll send to the outbound proxy address but
>> the Request-Line still shows
>>
>> REGISTER sip:sip.provider.com SIP/2.0
>>
>> Thanks,
>> Mervyn
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 14 05:16:39 2016
From: joel at gogii.net (Joel Serrano)
Date: Wed, 13 Apr 2016 18:16:39 -0700
Subject: [Freeswitch-users] FS Logs show ERR msgs: not enough buffer space
	for required resample operation!
Message-ID: <CAMtXxQmLrVi8oBUor7N8TMBfK8O=_jdXAUApY7py7d6OTtACmw@mail.gmail.com>

Hi,

We are occasionally seeing in our logs the following error:

7115ef69-3964-469c-b577-3127a9972bf6 2016-04-12 04:54:55.851240 [CRIT]
switch_core_io.c:1255 sofia/-profile_name-/-number- at -carrier- not enough
buffer space for required resample operation!
0b59a195-525e-49d4-8310-877d05c06849 2016-04-12 08:01:54.571182 [CRIT]
switch_core_io.c:1255 sofia/-profile_name-/-number- at -carrier- not enough
buffer space for required resample operation!
c1dc692d-95b9-409d-b51e-75cc85c30e1f 2016-04-13 01:07:39.491176 [CRIT]
switch_core_io.c:1255 sofia/-profile_name-/-number- at -carrier- not enough
buffer space for required resample operation!
44ead897-9b00-48c6-825a-f267c8ac6fbd 2016-04-13 03:35:52.471176 [CRIT]
switch_core_io.c:1255 sofia/-profile_name-/-number- at -carrier- not enough
buffer space for required resample operation!

I have searched and only found this thread:

http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116298.html


Does anyone know what can cause these type of errors? What buffer exactly
is the one that is out-of-space? Can we increase such buffer size?

FYI: FreeSWITCH (Version 1.6.6 -13-d2d0b32 64bit)



Thanks,

Joel.
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From joel at gogii.net  Thu Apr 14 05:19:09 2016
From: joel at gogii.net (Joel Serrano)
Date: Wed, 13 Apr 2016 18:19:09 -0700
Subject: [Freeswitch-users] FS Logs show ERR msgs: not enough buffer
 space for required resample operation!
In-Reply-To: <CAMtXxQmLrVi8oBUor7N8TMBfK8O=_jdXAUApY7py7d6OTtACmw@mail.gmail.com>
References: <CAMtXxQmLrVi8oBUor7N8TMBfK8O=_jdXAUApY7py7d6OTtACmw@mail.gmail.com>
Message-ID: <CAMtXxQm_t5GisLkYE3nCe775gq1T31XhxZvVZRPyygHx4__how@mail.gmail.com>

Hi again,

Just to add more information:


root at freeswitch:~# ps -fe | grep freeswitch | grep -v grep
root      3223     1  0 Feb11 ?        00:00:00 /usr/bin/freeswitch -u
freeswitch -g freeswitch -ncwait -reincarnate
freeswi+ 20239  3223 21 Mar31 ?        2-19:35:14 /usr/bin/freeswitch -u
freeswitch -g freeswitch -ncwait -reincarnate
root at freeswitch:~#

root at freeswitch:~# cat /proc/3223/limits
Limit                     Soft Limit           Hard Limit           Units
Max cpu time              unlimited            unlimited            seconds
Max file size             unlimited            unlimited            bytes
Max data size             unlimited            unlimited            bytes
Max stack size            8388608              unlimited            bytes
Max core file size        unlimited            unlimited            bytes
Max resident set          unlimited            unlimited            bytes
Max processes             60000                60000
 processes
Max open files            100000               100000               files
Max locked memory         65536                65536                bytes
Max address space         unlimited            unlimited            bytes
Max file locks            unlimited            unlimited            locks
Max pending signals       193514               193514               signals
Max msgqueue size         819200               819200               bytes
Max nice priority         0                    0
Max realtime priority     unlimited            unlimited
Max realtime timeout      7000000              7000000              us
root at freeswitch:~#

root at freeswitch:~# cat /proc/20239/limits
Limit                     Soft Limit           Hard Limit           Units
Max cpu time              unlimited            unlimited            seconds
Max file size             unlimited            unlimited            bytes
Max data size             unlimited            unlimited            bytes
Max stack size            245760               8388608              bytes
Max core file size        unlimited            unlimited            bytes
Max resident set          unlimited            unlimited            bytes
Max processes             unlimited            unlimited
 processes
Max open files            999999               999999               files
Max locked memory         65536                65536                bytes
Max address space         unlimited            unlimited            bytes
Max file locks            unlimited            unlimited            locks
Max pending signals       193514               193514               signals
Max msgqueue size         819200               819200               bytes
Max nice priority         0                    0
Max realtime priority     unlimited            unlimited
Max realtime timeout      7000000              7000000              us
root at freeswitch:~#



Thanks!

Joel.

On Wed, Apr 13, 2016 at 6:16 PM, Joel Serrano <joel at gogii.net> wrote:

> Hi,
>
> We are occasionally seeing in our logs the following error:
>
> 7115ef69-3964-469c-b577-3127a9972bf6 2016-04-12 04:54:55.851240 [CRIT]
> switch_core_io.c:1255 sofia/-profile_name-/-number- at -carrier- not enough
> buffer space for required resample operation!
> 0b59a195-525e-49d4-8310-877d05c06849 2016-04-12 08:01:54.571182 [CRIT]
> switch_core_io.c:1255 sofia/-profile_name-/-number- at -carrier- not enough
> buffer space for required resample operation!
> c1dc692d-95b9-409d-b51e-75cc85c30e1f 2016-04-13 01:07:39.491176 [CRIT]
> switch_core_io.c:1255 sofia/-profile_name-/-number- at -carrier- not enough
> buffer space for required resample operation!
> 44ead897-9b00-48c6-825a-f267c8ac6fbd 2016-04-13 03:35:52.471176 [CRIT]
> switch_core_io.c:1255 sofia/-profile_name-/-number- at -carrier- not enough
> buffer space for required resample operation!
>
> I have searched and only found this thread:
>
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116298.html
>
>
> Does anyone know what can cause these type of errors? What buffer exactly
> is the one that is out-of-space? Can we increase such buffer size?
>
> FYI: FreeSWITCH (Version 1.6.6 -13-d2d0b32 64bit)
>
>
>
> Thanks,
>
> Joel.
>
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From s.safarov at gmail.com  Thu Apr 14 08:11:09 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 14 Apr 2016 04:11:09 +0000
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <EC4896F1-829D-4C95-B7E5-AB9BD66021A1@freeswitch.org>
References: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
	<CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>
	<CAMtXxQnyeqq8aHNEk8f_EsvraiPrtpF0se91v8dOJx6ehhKwbg@mail.gmail.com>
	<5123CC15-8A19-453E-BB13-A8B230BC3DFC@jerris.com>
	<CAMtXxQ=8Q7iyzx2JvZ=o-nwBnyaxNhkF0rC2rp1cJqt0t4w9gA@mail.gmail.com>
	<EC4896F1-829D-4C95-B7E5-AB9BD66021A1@freeswitch.org>
Message-ID: <CAHtxdDdb+ULhYRF3shuMDcnvHr1woY7sgD8LxeV1P=GxP9OrNA@mail.gmail.com>

Please also check mod_format

On Thu, Apr 14, 2016, 02:16 Ken Rice <krice at freeswitch.org> wrote:

> The correct answer here is you must check more often i would say at a
> minimum every 15 seconds. Search the list for me discussing cdrs in volume
> in the past or contact me off list if you need pro help
>
> Sent from my iPhone
>
> On Apr 13, 2016, at 5:52 PM, Joel Serrano <joel at gogii.net> wrote:
>
> Hi Michael,
>
> The problem is that occasionally we see the following in FS log:
>
> 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:385 Got error [0] posting
> to web server [http://cdrs.example.com/queue]
> 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:392 Retry will be with url [
> http://cdrs.example.com/queue]
>
> When that happens, the CDR gets written to disc (so the CDR is not
> processed on time, thus our user is not billed for that call yet).
>
> We have a task that checks every 5 minutes for failed CDRs and we resubmit
> them to the processing queue, but I need to find out why FS is failing.
>
> I have checked the access logs on the receiving side, but all requests
> return a 200 OK.
>
> Now I have enabled logs in the loadbalancer between FS and the queues to
> see if I get more vision from there...
>
>
> A workaround was to make API requests directly from the dialplan with the
> billsec info, but as all suggest going with CDRs, I am now focussed on
> finding the root cause for that error.
>
> Any help/suggestion is more than welcome.
>
>
> Thank you.
>
> Best regards,
> Joel.
>
>
>
>
> On Tue, Apr 12, 2016 at 1:42 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I'm probably going to push towards cdr, but what are you trying to
>> actually do with that info?
>>
>> On Apr 12, 2016, at 2:39 PM, Joel Serrano <joel at gogii.net> wrote:
>>
>> Hi Michael,
>>
>> I want to call an API basically with the following info:
>>
>> sip_from_uri
>> destination_number
>> billsec
>>
>> I found away using "api_hangup_hook=lua script.lua" +
>> "session_in_hangup_hook=true", but, I don't like it... I would prefer using
>> curl from within the dialplan.
>>
>> Can you give me your suggestion? I have also asked in IRC and using the
>> CDRs looks like the best way to go, so we will probably follow that, but I
>> would still like to now if it is possible and how.
>>
>> Thanks!
>> Joel.
>>
>>
>>
>> On Tue, Apr 12, 2016 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Billsec is set in reporting state, after you are out of dial plan.
>>> What are you trying to do based on that, I can suggest a few approaches
>>>
>>>
>>> On Monday, April 11, 2016, Joel Serrano <joel at gogii.net> wrote:
>>>
>>>> Hi,
>>>>
>>>> Is it possible to access ${billsec} in the dialplan after the bridge
>>>> application (having hangup_after_bridge=false)??
>>>>
>>>> I want to log the billsec but currently it is always empty.
>>>>
>>>> I have a workaround in a test server using api_hangup_hook to call a
>>>> lua script with session_in_hangup_hook=true, and inside the lua script I
>>>> can access the billsec variable and thus write it to the logs.
>>>>
>>>> Is there no way to do this within the XML dialplan?
>>>>
>>>> Thanks!
>>>>
>>>> Best regards,
>>>> 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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From amani.mansour2 at gmail.com  Thu Apr 14 12:44:58 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Thu, 14 Apr 2016 08:44:58 +0000
Subject: [Freeswitch-users] Originate a call without SDP
Message-ID: <CAFETSWiwYQz_iEa9rv39B_6DdJh=QhKkwud1kNyN0uFewp1Nhg@mail.gmail.com>

Hi all,
I need to make a call from Fs to a sip client, The INVITE must be without
SDP ,is there any one can help me please .
I have enabled 3pcc
FS------------------------(INVITE without
SDP)---------------------------->phone

originate  {?????}sofia/gateway/mygateway/extension  &park()
or
modifie any thing in XML file
thanks

Best regards
Amani
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From hunterj91 at hotmail.com  Thu Apr 14 13:10:35 2016
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Thu, 14 Apr 2016 09:10:35 +0000
Subject: [Freeswitch-users] Voicemail message File Retrieval on Remote Server
Message-ID: <DUB112-W6976FAEF7938D9FA6DD5F7BD970@phx.gbl>


Hi Guys,
Quick question, I am centralising Voicemail storage as I am using multiple FreeSWITCH servers, and I have this working for recorded Voicemail greetings.
I am now doing this for stored voicemail messages.
So in voicemail.conf I am changing the storage-dir to the IP of my web server;

<param name="storage-dir" value="http://1.2.3.4:88"/>
Now this pushes the recorded wav file to the server using Put and works well, I just have 2 questions.
Locally I see FreeSWITCH creates the domain and extension folders in the original storage directory, can it be configured to build them on remove servers or should this be done manually? (which isnt a problem).
And the main question, in terms of when retrieving the voicemail, how is this achieved, as currently I use;
<action application="voicemail" data="check default ]]..call_context..[[ ]]..sip_from_user..[["/>
In my Lua scripts, which still checks locally, where do I let the voicemail module know about retrieving, and will it just perform a GET ?
Many thanks
Jon 		 	   		  
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From piotrek.gregor at gmail.com  Thu Apr 14 17:58:49 2016
From: piotrek.gregor at gmail.com (Piotr Gregor)
Date: Thu, 14 Apr 2016 14:58:49 +0100
Subject: [Freeswitch-users] FS Logs show ERR msgs: not enough buffer
 space for required resample operation!
In-Reply-To: <CAMtXxQm_t5GisLkYE3nCe775gq1T31XhxZvVZRPyygHx4__how@mail.gmail.com>
References: <CAMtXxQmLrVi8oBUor7N8TMBfK8O=_jdXAUApY7py7d6OTtACmw@mail.gmail.com>
	<CAMtXxQm_t5GisLkYE3nCe775gq1T31XhxZvVZRPyygHx4__how@mail.gmail.com>
Message-ID: <CAO3obxYL9J__ZapmX3JVmRU-TkSHyueRHrrwohpxV+Ejt=Tx0g@mail.gmail.com>

Hi Joel,

This might mean you try to resample between formats that differ too much,
e.g. "to" rate is too big compared to "from" rate.

cheers,
Piotr
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From bobjectsfreeswitch at gmail.com  Thu Apr 14 18:06:38 2016
From: bobjectsfreeswitch at gmail.com (Bob Hartwig)
Date: Thu, 14 Apr 2016 09:06:38 -0500
Subject: [Freeswitch-users] Originate a call without SDP
In-Reply-To: <CAFETSWiwYQz_iEa9rv39B_6DdJh=QhKkwud1kNyN0uFewp1Nhg@mail.gmail.com>
References: <CAFETSWiwYQz_iEa9rv39B_6DdJh=QhKkwud1kNyN0uFewp1Nhg@mail.gmail.com>
Message-ID: <CAAemr3yEpZ=S488FTuXqDLyJ28LkYxrnZpDPW2HGsSg4tJdyLg@mail.gmail.com>

I think 3pcc only enables interoperability with inbound empty INVITEs, but
I could be wrong.

Just curious - I understand the motivation behind using empty INVITEs, but
why is it a requirement for your far-end user agent?



On Thu, Apr 14, 2016 at 3:44 AM, amani mansour <amani.mansour2 at gmail.com>
wrote:

> Hi all,
> I need to make a call from Fs to a sip client, The INVITE must be without
> SDP ,is there any one can help me please .
> I have enabled 3pcc
> FS------------------------(INVITE without
> SDP)---------------------------->phone
>
> originate  {?????}sofia/gateway/mygateway/extension  &park()
> or
> modifie any thing in XML file
> thanks
>
> Best regards
> Amani
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From colin.morelli at gmail.com  Thu Apr 14 17:56:55 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Thu, 14 Apr 2016 13:56:55 +0000
Subject: [Freeswitch-users] Custom Authentication
Message-ID: <CAPtU-UokMgo6LqxUs4OfezQLfYXxhs-52qQ14tHiM8dNvy9nMg@mail.gmail.com>

Hey all,

I'm looking for a way to support custom authentication with FreeSwitch.
What I am hoping to get is similar to mod_xml_curl directory, but where FS
will provide both user and password to my service. The use case I'm looking
to achieve is to support token/key based authentication in addition to
standard SIP auth.

Is this something that's possible in FS? Any other suggestions?

Thanks
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From mike at jerris.com  Thu Apr 14 18:32:30 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Apr 2016 10:32:30 -0400
Subject: [Freeswitch-users] Voicemail message File Retrieval on Remote
	Server
In-Reply-To: <DUB112-W6976FAEF7938D9FA6DD5F7BD970@phx.gbl>
References: <DUB112-W6976FAEF7938D9FA6DD5F7BD970@phx.gbl>
Message-ID: <CAEwTsAg-9VrMQqxJUkxJ6V5Z8wpWsDRSQGw-J41KC7PJ1Y-kng@mail.gmail.com>

This is pretty clever but I think there are too many things in the code
that assume it's a local directory for this to work without modifications
to code

On Thursday, April 14, 2016, Jonathan Hunter <hunterj91 at hotmail.com> wrote:

>
> Hi Guys,
>
> Quick question, I am centralising Voicemail storage as I am using multiple
> FreeSWITCH servers, and I have this working for recorded Voicemail
> greetings.
>
> I am now doing this for stored voicemail messages.
>
> So in voicemail.conf I am changing the storage-dir to the IP of my web
> server;
>
>
> <param name="storage-dir" value="http://1.2.3.4:88"/>
>
> Now this pushes the recorded wav file to the server using Put and works
> well, I just have 2 questions.
>
> Locally I see FreeSWITCH creates the domain and extension folders in the
> original storage directory, can it be configured to build them on remove
> servers or should this be done manually? (which isnt a problem).
>
> And the main question, in terms of when retrieving the voicemail, how is
> this achieved, as currently I use;
>
> <action application="voicemail" data="check default ]]..call_context..[[
> ]]..sip_from_user..[["/>
>
> In my Lua scripts, which still checks locally, where do I let the
> voicemail module know about retrieving, and will it just perform a GET ?
>
> Many thanks
>
> Jon
>
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From amani.mansour2 at gmail.com  Thu Apr 14 18:36:30 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Thu, 14 Apr 2016 14:36:30 +0000
Subject: [Freeswitch-users] Originate a call without SDP
In-Reply-To: <CAAemr3yEpZ=S488FTuXqDLyJ28LkYxrnZpDPW2HGsSg4tJdyLg@mail.gmail.com>
References: <CAFETSWiwYQz_iEa9rv39B_6DdJh=QhKkwud1kNyN0uFewp1Nhg@mail.gmail.com>
	<CAAemr3yEpZ=S488FTuXqDLyJ28LkYxrnZpDPW2HGsSg4tJdyLg@mail.gmail.com>
Message-ID: <CAFETSWhWM+4b53JPWG2DnrsEYUmSygR1s9AWinW+oGS+=y5maA@mail.gmail.com>

Hi Bob,
I am sorry but i don't understand  your question ? what do you mean whith
 far-end user agent ?
Actually i am developing a web application that permit to test different
sip scenario,one of them is to configure a number to send invite without
SDP I 'am blocked (1 week in this scenario and also i am failed to resolve
it can you help me please )
best regards
amani

Le jeu. 14 avr. 2016 ? 15:07, Bob Hartwig <bobjectsfreeswitch at gmail.com> a
?crit :

> I think 3pcc only enables interoperability with inbound empty INVITEs, but
> I could be wrong.
>
> Just curious - I understand the motivation behind using empty INVITEs, but
> why is it a requirement for your far-end user agent?
>
>
>
> On Thu, Apr 14, 2016 at 3:44 AM, amani mansour <amani.mansour2 at gmail.com>
> wrote:
>
>> Hi all,
>> I need to make a call from Fs to a sip client, The INVITE must be without
>> SDP ,is there any one can help me please .
>> I have enabled 3pcc
>> FS------------------------(INVITE without
>> SDP)---------------------------->phone
>>
>> originate  {?????}sofia/gateway/mygateway/extension  &park()
>> or
>> modifie any thing in XML file
>> thanks
>>
>> Best regards
>> Amani
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From amani.mansour2 at gmail.com  Thu Apr 14 18:41:09 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Thu, 14 Apr 2016 14:41:09 +0000
Subject: [Freeswitch-users] (no subject)
Message-ID: <CAFETSWjMUcKFADoVoNiMRHp6Py_h4U+DgrRRX-Dof=+MrCw1wQ@mail.gmail.com>

Hey all,

I'm looking for a way to unauthorize an extension to be regestred .

i need to do this scenario ,please can any one help me ?

phone -------------------Register-------------------------->FS
FS---------------------------401------------------------------->phone
phone -------------------Register-------------------------->FS
FS---------------------------401(whith stale =false)------------->phone

Is this something that's possible in FS? Any other suggestions?

Thanks
amani
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From hunterj91 at hotmail.com  Thu Apr 14 19:01:23 2016
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Thu, 14 Apr 2016 15:01:23 +0000
Subject: [Freeswitch-users] Voicemail message File Retrieval on Remote
 Server
In-Reply-To: <CAPQovyRovMm6vCCpHeeyU+Fw+n1Xm=EqMB4n=g4vSG-6kvBH-g@mail.gmail.com>
References: <DUB112-W6976FAEF7938D9FA6DD5F7BD970@phx.gbl>,
	<CAEwTsAg-9VrMQqxJUkxJ6V5Z8wpWsDRSQGw-J41KC7PJ1Y-kng@mail.gmail.com>,
	<CAPQovyRovMm6vCCpHeeyU+Fw+n1Xm=EqMB4n=g4vSG-6kvBH-g@mail.gmail.com>
Message-ID: <DUB112-W1020282C558B19A339B9B13BD970@phx.gbl>

Hi Michael,
I appreciate the response, where in the code would I need to be looking?
Also how are other people doing this? I presume they do some file sync do they to ensure all media servers have the same files?
Many thanks
Jon



---------- Forwarded message ----------
From: Michael Jerris <mike at jerris.com>
Date: Thu, Apr 14, 2016 at 3:32 PM
Subject: Re: [Freeswitch-users] Voicemail message File Retrieval on Remote Server
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>


This is pretty clever but I think there are too many things in the code that assume it's a local directory for this to work without modifications to code
On Thursday, April 14, 2016, Jonathan Hunter <hunterj91 at hotmail.com> wrote:




Hi Guys,
Quick question, I am centralising Voicemail storage as I am using multiple FreeSWITCH servers, and I have this working for recorded Voicemail greetings.
I am now doing this for stored voicemail messages.
So in voicemail.conf I am changing the storage-dir to the IP of my web server;

<param name="storage-dir" value="http://1.2.3.4:88"/>
Now this pushes the recorded wav file to the server using Put and works well, I just have 2 questions.
Locally I see FreeSWITCH creates the domain and extension folders in the original storage directory, can it be configured to build them on remove servers or should this be done manually? (which isnt a problem).
And the main question, in terms of when retrieving the voicemail, how is this achieved, as currently I use;
<action application="voicemail" data="check default ]]..call_context..[[ ]]..sip_from_user..[["/>
In my Lua scripts, which still checks locally, where do I let the voicemail module know about retrieving, and will it just perform a GET ?
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


-- 
Jonathan HunterTechnical Director /Telephony DeveloperM:(+44) 7917 190 438Email:jhunter at voxboxcoms.co.uk

 		 	   		  
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From mike at jerris.com  Thu Apr 14 19:27:52 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Apr 2016 11:27:52 -0400
Subject: [Freeswitch-users] Voicemail message File Retrieval on Remote
	Server
In-Reply-To: <DUB112-W1020282C558B19A339B9B13BD970@phx.gbl>
References: <DUB112-W6976FAEF7938D9FA6DD5F7BD970@phx.gbl>
	<CAEwTsAg-9VrMQqxJUkxJ6V5Z8wpWsDRSQGw-J41KC7PJ1Y-kng@mail.gmail.com>
	<CAPQovyRovMm6vCCpHeeyU+Fw+n1Xm=EqMB4n=g4vSG-6kvBH-g@mail.gmail.com>
	<DUB112-W1020282C558B19A339B9B13BD970@phx.gbl>
Message-ID: <0F514384-D461-4C3B-960E-776662BD0E62@jerris.com>

The code would be in mod_voicemail.  I know there is some stuff in there for creating the directories, that obviously doesn't deal with stream format for files (hence why you get those empty directories), also there are places we are checking if a file exists, I suspect thats whats wrong with playing files.  Its likely there are more issues, those two are off the top of my head.  When we have done this in the past, we have just used nfs.


> On Apr 14, 2016, at 11:01 AM, Jonathan Hunter <hunterj91 at hotmail.com> wrote:
> 
> Hi Michael,
> 
> I appreciate the response, where in the code would I need to be looking?
> 
> Also how are other people doing this? I presume they do some file sync do they to ensure all media servers have the same files?
> 
> Many thanks
> 
> Jon
> 
> 
> 
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>
> Date: Thu, Apr 14, 2016 at 3:32 PM
> Subject: Re: [Freeswitch-users] Voicemail message File Retrieval on Remote Server
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> 
> 
> This is pretty clever but I think there are too many things in the code that assume it's a local directory for this to work without modifications to code
> 
> On Thursday, April 14, 2016, Jonathan Hunter <hunterj91 at hotmail.com <mailto:hunterj91 at hotmail.com>> wrote:
> 
> Hi Guys,
> 
> Quick question, I am centralising Voicemail storage as I am using multiple FreeSWITCH servers, and I have this working for recorded Voicemail greetings.
> 
> I am now doing this for stored voicemail messages.
> 
> So in voicemail.conf I am changing the storage-dir to the IP of my web server;
> 
> 
> <param name="storage-dir" value="http://1.2.3.4:88 <http://1.2.3.4:88/>"/>
> 
> Now this pushes the recorded wav file to the server using Put and works well, I just have 2 questions.
> 
> Locally I see FreeSWITCH creates the domain and extension folders in the original storage directory, can it be configured to build them on remove servers or should this be done manually? (which isnt a problem).
> 
> And the main question, in terms of when retrieving the voicemail, how is this achieved, as currently I use;
> 
> <action application="voicemail" data="check default ]]..call_context..[[ ]]..sip_from_user..[["/>
> 
> In my Lua scripts, which still checks locally, where do I let the voicemail module know about retrieving, and will it just perform a GET ?
> 
> Many thanks
> 
> Jon

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From nneul at mst.edu  Thu Apr 14 20:06:39 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 14 Apr 2016 11:06:39 -0500
Subject: [Freeswitch-users] Voicemail message File Retrieval on Remote
 Server
In-Reply-To: <DUB112-W1020282C558B19A339B9B13BD970@phx.gbl>
References: <DUB112-W6976FAEF7938D9FA6DD5F7BD970@phx.gbl>
	<CAEwTsAg-9VrMQqxJUkxJ6V5Z8wpWsDRSQGw-J41KC7PJ1Y-kng@mail.gmail.com>
	<CAPQovyRovMm6vCCpHeeyU+Fw+n1Xm=EqMB4n=g4vSG-6kvBH-g@mail.gmail.com>
	<DUB112-W1020282C558B19A339B9B13BD970@phx.gbl>
Message-ID: <570FC00F.30205@mst.edu>

I use a gluster filesystem as backend for storage on my FS cluster for media files, logs, etc.

-- Nathan

On 04/14/2016 10:01 AM, Jonathan Hunter wrote:
> Hi Michael,
>
> I appreciate the response, where in the code would I need to be looking?
>
> Also how are other people doing this? I presume they do some file sync do they to ensure all media servers have the same
> files?
>
> Many thanks
>
> Jon
>
>
>
> ---------- Forwarded message ----------
> From: *Michael Jerris* <mike at jerris.com <mailto:mike at jerris.com>>
> Date: Thu, Apr 14, 2016 at 3:32 PM
> Subject: Re: [Freeswitch-users] Voicemail message File Retrieval on Remote Server
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
>
>
> This is pretty clever but I think there are too many things in the code that assume it's a local directory for this to
> work without modifications to code
>
> On Thursday, April 14, 2016, Jonathan Hunter <hunterj91 at hotmail.com <mailto:hunterj91 at hotmail.com>> wrote:
>
>
>     Hi Guys,
>
>     Quick question, I am centralising Voicemail storage as I am using multiple FreeSWITCH servers, and I have this
>     working for recorded Voicemail greetings.
>
>     I am now doing this for stored voicemail messages.
>
>     So in voicemail.conf I am changing the storage-dir to the IP of my web server;
>
>
>     <param name="storage-dir" value="http://1.2.3.4:88"/>
>
>     Now this pushes the recorded wav file to the server using Put and works well, I just have 2 questions.
>
>     Locally I see FreeSWITCH creates the domain and extension folders in the original storage directory, can it be
>     configured to build them on remove servers or should this be done manually? (which isnt a problem).
>
>     And the main question, in terms of when retrieving the voicemail, how is this achieved, as currently I use;
>
>     <action application="voicemail" data="check default ]]..call_context..[[ ]]..sip_from_user..[["/>
>
>     In my Lua scripts, which still checks locally, where do I let the voicemail module know about retrieving, and will
>     it just perform a GET ?
>
>     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
>
>
>
> --
> Jonathan Hunter
> Technical Director /Telephony Developer
> M:(+44) 7917 190 438
> Email:jhunter at voxboxcoms.co.uk <mailto:Email%3Ajhunter 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
>

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


From carlosj.gf at gmail.com  Thu Apr 14 20:24:01 2016
From: carlosj.gf at gmail.com (=?UTF-8?Q?Carlos_Gonz=C3=A1lez_Florido?=)
Date: Thu, 14 Apr 2016 18:24:01 +0200
Subject: [Freeswitch-users] Sending commands over Verto
Message-ID: <CAFKjMAYBM2LgRyauAKs_vhesf633-ACa-Azn9G3Th7+G06uaWw@mail.gmail.com>

Hi list.

Reading "Verto Not just for call signaling" [1] I see you can send API
commands over verto channels, I tested the included example ("command":
"status") and it works. However I cannot make it work for any other
command, FS always replies with {"code": -32602,  "message": "Permission
Denied"}.

How can I enable other commands, even with arguments?

Thank you,
Carlos

[1] https://freeswitch.org/verto-not-just-for-call-signaling/
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From pstarzyk at general-devices.com  Thu Apr 14 20:28:46 2016
From: pstarzyk at general-devices.com (Piotr Starzyk)
Date: Thu, 14 Apr 2016 12:28:46 -0400
Subject: [Freeswitch-users] How to dynamically set up conferences?
Message-ID: <59173f0f8e9631c3712aea0308b7b66d@mail.gmail.com>

Let?s say I have 3 SIP endpoints: Caller_A, Caller_B and Caller_C



Initially Caller_A calls Caller_B, so we end up with a SIP call between
them.



But now, there will be some cases, when Caller_A will want to ?conference
in? Caller_C into the call between Caller_A and Caller_B.



How is it normally done?  Based on my limited understanding of
mod_conference, I would attempt doing it as following:



Approach I:

Tear down the original call between A & B, and set up a conference between
A, B & C:

1.       Caller_A calls Caller_B

2.       SIP call is established between Caller_A and Caller_B.

3.       After a while Caller_A wants to conference in Caller_C, as
following:

a.       Caller_A hangs up the call with Caller_B

b.      Caller_A dials into a conference room.

c.       Caller_B dials into the same conference room.

d.      Caller_C dials into the same conference room.

4.       Caller_A, Caller_B and Caller_C are now in a conference.



Approach II:

Use conferences for all calls from the beginning, even the 2-way calls, to
make it easier to conference in additional callers:

1.       Caller_A dials into a conference room.

2.       Caller_B dials into a conference room.

3.       Calelr_A and Caller_B can now talk to each other.

4.       After a while Caller_A wants to conference in Caller_C

a.       Caller_C dials into the same conference room.

5.       Caller_A, Caller_B and Caller_C are now in a conference.





I?ve also read about eavesdrop tool, but I understand it only allows 3-way
communication, and I need to be able to support N-way.



Thanks.
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From nneul at mst.edu  Thu Apr 14 20:49:15 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 14 Apr 2016 11:49:15 -0500
Subject: [Freeswitch-users] Are there any known memory leaks with reloadxml
 and large XML configs?
Message-ID: <570FCA0B.3020308@mst.edu>

I've got a persistent issue with a slow (sometimes not that slow) memory leak in FS. Our environment does frequent 
reconfigs built around reloadxml and when the leak gets bad enough, I start getting random reports of slipping audio 
sync. I've put in weekly forced failovers, but if we have a week with more changes than typical (we're still in middle 
of mass migration off of CCM) - the leak growth can be too fast:

freeswi+ 17977 12.1 27.2 15247652 2213648 ?    S<l  Apr02 2155:56 /local/freeswitch/server/bin/freeswitch -u freeswitch 
-core -cfgname freeswitch-prod.xml -base /local/freeswitch/server -conf /local/freeswitch/server/conf -db 
/local/freeswitch/server/db -run /local/freeswitch/data -log /local/freeswitch/data -rp -nf -nc -nonat -nonatmap


The reason I ask if the leak is related to reloadxml - when doing some analysis on the resulting core dump on the (no 
longer active) server, I'm finding examples like this when doing a 'strings corefile | sort | uniq -c | sort -n':

...
   74642 !-- type(skinny) --
   76811 /buttons
   76811 "InvalidHash
   76812 "a1-hash
   76836 /skinny
   76853 buttons
...
  105277 !-- password won't be used by cisco devices --
...
  105277 "vm-mailfrom
  105277 "vm-password

[root at freesw-p1 data]# grep -c InvalidHash freeswitch-prod.xml.fsxml
1431

[root at freesw-p1 data]# ls -al *.fsxml
-rw------- 1 freeswitch freeswitch 4985669 Apr 14 08:28 freeswitch-prod.xml.fsxml
[root at freesw-p1 data]# wc -l *.fsxml
105025 freeswitch-prod.xml.fsxml


It's like it's leaking large numbers of complete copies of the XML. When I look directly at the core dump, it looks to 
me like the strings are in the parsed state of the XML. (Below slightly masked copy and paste from viewing dump with less.)


-------------
^@!-- xxx-xxx-xxxx --^@
^@user^@id^@"xxxxxxxxxxxxxxxxxxxxxxxx^@^@
     ^@params^@
         ^@!-- password won't be used by cisco devices --^@
         ^@param^@name^@"password^@ value^@"XXXXXXXXXXXXXXXXX^@^@^@
         ^@param^@name^@"vm-mailfrom^@ value^@"voicemail at mst.edu^@^@^@
-----------

Would really appreciate any ideas on how I might mitigate this leaking or if there is anything that could be done to 
help diagnose it further to help address the underlying issue.


I'm happy to open a JIRA on this, but will NOT be able to test this with latest master as I can't just experiment with 
the live production environment.

-- Nathan

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


From blasterjr at gmail.com  Thu Apr 14 21:02:37 2016
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Thu, 14 Apr 2016 11:02:37 -0600
Subject: [Freeswitch-users] Voicemail message File Retrieval on Remote
	Server
In-Reply-To: <570FC00F.30205@mst.edu>
References: <DUB112-W6976FAEF7938D9FA6DD5F7BD970@phx.gbl>
	<CAEwTsAg-9VrMQqxJUkxJ6V5Z8wpWsDRSQGw-J41KC7PJ1Y-kng@mail.gmail.com>
	<CAPQovyRovMm6vCCpHeeyU+Fw+n1Xm=EqMB4n=g4vSG-6kvBH-g@mail.gmail.com>
	<DUB112-W1020282C558B19A339B9B13BD970@phx.gbl> <570FC00F.30205@mst.edu>
Message-ID: <CAO3hyzFqOVLjfiOU0x0Hz3W3UMwjYAK3on8u_Q74zwRqXey6YA@mail.gmail.com>

Jonathan, i just wanted to give my feedback in regaurds to this, i
recommend going a different route, such as GlusterFS or writing your own
voicemail in lua (really its not that hard!).

The issues that i ran into while trying to use http was that freeswitch
wasn't able to check that the file existed and it would break certain
functions such as the email voicemail and with things such as greetings, it
would end up either creating empty directories or not returning success
when saving.

I actually opened an issue about this a while back:
https://freeswitch.org/jira/browse/FS-7280  there was some discussion on
how it could be handled, however i think the easier route is to either use
a shared filesystem setup like GlusterFS or to write your own voicemail
module.

On Thu, Apr 14, 2016 at 10:06 AM, Nathan Neulinger <nneul at mst.edu> wrote:

> I use a gluster filesystem as backend for storage on my FS cluster for
> media files, logs, etc.
>
> -- Nathan
>
> On 04/14/2016 10:01 AM, Jonathan Hunter wrote:
> > Hi Michael,
> >
> > I appreciate the response, where in the code would I need to be looking?
> >
> > Also how are other people doing this? I presume they do some file sync
> do they to ensure all media servers have the same
> > files?
> >
> > Many thanks
> >
> > Jon
> >
> >
> >
> > ---------- Forwarded message ----------
> > From: *Michael Jerris* <mike at jerris.com <mailto:mike at jerris.com>>
> > Date: Thu, Apr 14, 2016 at 3:32 PM
> > Subject: Re: [Freeswitch-users] Voicemail message File Retrieval on
> Remote Server
> > To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
> <mailto:freeswitch-users at lists.freeswitch.org>>
> >
> >
> > This is pretty clever but I think there are too many things in the code
> that assume it's a local directory for this to
> > work without modifications to code
> >
> > On Thursday, April 14, 2016, Jonathan Hunter <hunterj91 at hotmail.com
> <mailto:hunterj91 at hotmail.com>> wrote:
> >
> >
> >     Hi Guys,
> >
> >     Quick question, I am centralising Voicemail storage as I am using
> multiple FreeSWITCH servers, and I have this
> >     working for recorded Voicemail greetings.
> >
> >     I am now doing this for stored voicemail messages.
> >
> >     So in voicemail.conf I am changing the storage-dir to the IP of my
> web server;
> >
> >
> >     <param name="storage-dir" value="http://1.2.3.4:88"/>
> >
> >     Now this pushes the recorded wav file to the server using Put and
> works well, I just have 2 questions.
> >
> >     Locally I see FreeSWITCH creates the domain and extension folders in
> the original storage directory, can it be
> >     configured to build them on remove servers or should this be done
> manually? (which isnt a problem).
> >
> >     And the main question, in terms of when retrieving the voicemail,
> how is this achieved, as currently I use;
> >
> >     <action application="voicemail" data="check default
> ]]..call_context..[[ ]]..sip_from_user..[["/>
> >
> >     In my Lua scripts, which still checks locally, where do I let the
> voicemail module know about retrieving, and will
> >     it just perform a GET ?
> >
> >     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
> >
> >
> >
> > --
> > Jonathan Hunter
> > Technical Director /Telephony Developer
> > M:(+44) 7917 190 438
> > Email:jhunter at voxboxcoms.co.uk <mailto:Email%3Ajhunter 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
> >
>
> --
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From abaci64 at gmail.com  Thu Apr 14 21:20:58 2016
From: abaci64 at gmail.com (Abaci B)
Date: Thu, 14 Apr 2016 13:20:58 -0400
Subject: [Freeswitch-users] Voicemail message File Retrieval on Remote
	Server
In-Reply-To: <CAO3hyzFqOVLjfiOU0x0Hz3W3UMwjYAK3on8u_Q74zwRqXey6YA@mail.gmail.com>
References: <DUB112-W6976FAEF7938D9FA6DD5F7BD970@phx.gbl>
	<CAEwTsAg-9VrMQqxJUkxJ6V5Z8wpWsDRSQGw-J41KC7PJ1Y-kng@mail.gmail.com>
	<CAPQovyRovMm6vCCpHeeyU+Fw+n1Xm=EqMB4n=g4vSG-6kvBH-g@mail.gmail.com>
	<DUB112-W1020282C558B19A339B9B13BD970@phx.gbl>
	<570FC00F.30205@mst.edu>
	<CAO3hyzFqOVLjfiOU0x0Hz3W3UMwjYAK3on8u_Q74zwRqXey6YA@mail.gmail.com>
Message-ID: <CAFmZFiNrHQ7jMdCh4C4YNS8K4XMrj-7-MhoobYDO8bY-pS-sZA@mail.gmail.com>

if it has to be http there are some fuse filesystems that use http as
backend

On Thu, Apr 14, 2016 at 1:02 PM, Chris Tunbridge <blasterjr at gmail.com>
wrote:

> Jonathan, i just wanted to give my feedback in regaurds to this, i
> recommend going a different route, such as GlusterFS or writing your own
> voicemail in lua (really its not that hard!).
>
> The issues that i ran into while trying to use http was that freeswitch
> wasn't able to check that the file existed and it would break certain
> functions such as the email voicemail and with things such as greetings, it
> would end up either creating empty directories or not returning success
> when saving.
>
> I actually opened an issue about this a while back:
> https://freeswitch.org/jira/browse/FS-7280  there was some discussion on
> how it could be handled, however i think the easier route is to either use
> a shared filesystem setup like GlusterFS or to write your own voicemail
> module.
>
> On Thu, Apr 14, 2016 at 10:06 AM, Nathan Neulinger <nneul at mst.edu> wrote:
>
>> I use a gluster filesystem as backend for storage on my FS cluster for
>> media files, logs, etc.
>>
>> -- Nathan
>>
>> On 04/14/2016 10:01 AM, Jonathan Hunter wrote:
>> > Hi Michael,
>> >
>> > I appreciate the response, where in the code would I need to be looking?
>> >
>> > Also how are other people doing this? I presume they do some file sync
>> do they to ensure all media servers have the same
>> > files?
>> >
>> > Many thanks
>> >
>> > Jon
>> >
>> >
>> >
>> > ---------- Forwarded message ----------
>> > From: *Michael Jerris* <mike at jerris.com <mailto:mike at jerris.com>>
>> > Date: Thu, Apr 14, 2016 at 3:32 PM
>> > Subject: Re: [Freeswitch-users] Voicemail message File Retrieval on
>> Remote Server
>> > To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
>> <mailto:freeswitch-users at lists.freeswitch.org>>
>> >
>> >
>> > This is pretty clever but I think there are too many things in the code
>> that assume it's a local directory for this to
>> > work without modifications to code
>> >
>> > On Thursday, April 14, 2016, Jonathan Hunter <hunterj91 at hotmail.com
>> <mailto:hunterj91 at hotmail.com>> wrote:
>> >
>> >
>> >     Hi Guys,
>> >
>> >     Quick question, I am centralising Voicemail storage as I am using
>> multiple FreeSWITCH servers, and I have this
>> >     working for recorded Voicemail greetings.
>> >
>> >     I am now doing this for stored voicemail messages.
>> >
>> >     So in voicemail.conf I am changing the storage-dir to the IP of my
>> web server;
>> >
>> >
>> >     <param name="storage-dir" value="http://1.2.3.4:88"/>
>> >
>> >     Now this pushes the recorded wav file to the server using Put and
>> works well, I just have 2 questions.
>> >
>> >     Locally I see FreeSWITCH creates the domain and extension folders
>> in the original storage directory, can it be
>> >     configured to build them on remove servers or should this be done
>> manually? (which isnt a problem).
>> >
>> >     And the main question, in terms of when retrieving the voicemail,
>> how is this achieved, as currently I use;
>> >
>> >     <action application="voicemail" data="check default
>> ]]..call_context..[[ ]]..sip_from_user..[["/>
>> >
>> >     In my Lua scripts, which still checks locally, where do I let the
>> voicemail module know about retrieving, and will
>> >     it just perform a GET ?
>> >
>> >     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
>> >
>> >
>> >
>> > --
>> > Jonathan Hunter
>> > Technical Director /Telephony Developer
>> > M:(+44) 7917 190 438
>> > Email:jhunter at voxboxcoms.co.uk <mailto:Email%3Ajhunter 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
>> >
>>
>> --
>> ------------------------------------------------------------
>> 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
>
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From mike at jerris.com  Thu Apr 14 21:27:53 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Apr 2016 13:27:53 -0400
Subject: [Freeswitch-users] Sending commands over Verto
In-Reply-To: <CAFKjMAYBM2LgRyauAKs_vhesf633-ACa-Azn9G3Th7+G06uaWw@mail.gmail.com>
References: <CAFKjMAYBM2LgRyauAKs_vhesf633-ACa-Azn9G3Th7+G06uaWw@mail.gmail.com>
Message-ID: <ACE60D4B-1606-4C5D-97B9-C6749320608E@jerris.com>

check out user directory params:

jsonrpc-allowed-jsapi
jsonrpc-allowed-fsapi


they go the same place as 

jsonrpc-allowed-event-channels"


> On Apr 14, 2016, at 12:24 PM, Carlos Gonz?lez Florido <carlosj.gf at gmail.com> wrote:
> 
> Hi list.
> 
> Reading "Verto Not just for call signaling" [1] I see you can send API commands over verto channels, I tested the included example ("command": "status") and it works. However I cannot make it work for any other command, FS always replies with {"code": -32602,  "message": "Permission Denied"}.
> 
> How can I enable other commands, even with arguments?
> 
> Thank you,
> Carlos
> 
> [1] https://freeswitch.org/verto-not-just-for-call-signaling/ <https://freeswitch.org/verto-not-just-for-call-signaling/>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 14 21:30:34 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Apr 2016 13:30:34 -0400
Subject: [Freeswitch-users] How to dynamically set up conferences?
In-Reply-To: <59173f0f8e9631c3712aea0308b7b66d@mail.gmail.com>
References: <59173f0f8e9631c3712aea0308b7b66d@mail.gmail.com>
Message-ID: <7009D577-4530-45AE-B135-A3365F884CEF@jerris.com>

I've seen a lot of people use Approach II and it is always a huge mess, don't do it.  Approach 1 is actually very simple, but you can do it even simpler than this, you can just transfer both call legs into a conference on the fly, you don't need to hang up and call back.  If you adjust params to not get tones and such, its nearly unnoticeable it happens so quick.


> On Apr 14, 2016, at 12:28 PM, Piotr Starzyk <pstarzyk at general-devices.com> wrote:
> 
> Let?s say I have 3 SIP endpoints: Caller_A, Caller_B and Caller_C
>  
> Initially Caller_A calls Caller_B, so we end up with a SIP call between them.
>  
> But now, there will be some cases, when Caller_A will want to ?conference in? Caller_C into the call between Caller_A and Caller_B.
>  
> How is it normally done?  Based on my limited understanding of mod_conference, I would attempt doing it as following:
>  
> Approach I: 
> Tear down the original call between A & B, and set up a conference between A, B & C:
> 1.       Caller_A calls Caller_B
> 2.       SIP call is established between Caller_A and Caller_B.
> 3.       After a while Caller_A wants to conference in Caller_C, as following:
> a.       Caller_A hangs up the call with Caller_B
> b.      Caller_A dials into a conference room.
> c.       Caller_B dials into the same conference room.
> d.      Caller_C dials into the same conference room.
> 4.       Caller_A, Caller_B and Caller_C are now in a conference.
>  
> Approach II: 
> Use conferences for all calls from the beginning, even the 2-way calls, to make it easier to conference in additional callers:
> 1.       Caller_A dials into a conference room.
> 2.       Caller_B dials into a conference room.
> 3.       Calelr_A and Caller_B can now talk to each other.
> 4.       After a while Caller_A wants to conference in Caller_C
> a.       Caller_C dials into the same conference room.
> 5.       Caller_A, Caller_B and Caller_C are now in a conference.
>  
>  
> I?ve also read about eavesdrop tool, but I understand it only allows 3-way communication, and I need to be able to support N-way.
>  
> Thanks.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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From mike at jerris.com  Thu Apr 14 21:38:13 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Apr 2016 13:38:13 -0400
Subject: [Freeswitch-users] Are there any known memory leaks with
	reloadxml and large XML configs?
In-Reply-To: <570FCA0B.3020308@mst.edu>
References: <570FCA0B.3020308@mst.edu>
Message-ID: <209D8D8A-7D31-4579-8E58-C15A2406EEDE@jerris.com>

A parsed xml tree in a core file would be normal.  We always keep one in memory, thats why you have to reload to get a new one, so it would be in the core file every time, not an indication of a leak.


> On Apr 14, 2016, at 12:49 PM, Nathan Neulinger <nneul at mst.edu> wrote:
> 
> I've got a persistent issue with a slow (sometimes not that slow) memory leak in FS. Our environment does frequent 
> reconfigs built around reloadxml and when the leak gets bad enough, I start getting random reports of slipping audio 
> sync. I've put in weekly forced failovers, but if we have a week with more changes than typical (we're still in middle 
> of mass migration off of CCM) - the leak growth can be too fast:
> 
> freeswi+ 17977 12.1 27.2 15247652 2213648 ?    S<l  Apr02 2155:56 /local/freeswitch/server/bin/freeswitch -u freeswitch 
> -core -cfgname freeswitch-prod.xml -base /local/freeswitch/server -conf /local/freeswitch/server/conf -db 
> /local/freeswitch/server/db -run /local/freeswitch/data -log /local/freeswitch/data -rp -nf -nc -nonat -nonatmap
> 
> 
> The reason I ask if the leak is related to reloadxml - when doing some analysis on the resulting core dump on the (no 
> longer active) server, I'm finding examples like this when doing a 'strings corefile | sort | uniq -c | sort -n':
> 
> ...
>   74642 !-- type(skinny) --
>   76811 /buttons
>   76811 "InvalidHash
>   76812 "a1-hash
>   76836 /skinny
>   76853 buttons
> ...
>  105277 !-- password won't be used by cisco devices --
> ...
>  105277 "vm-mailfrom
>  105277 "vm-password
> 
> [root at freesw-p1 data]# grep -c InvalidHash freeswitch-prod.xml.fsxml
> 1431
> 
> [root at freesw-p1 data]# ls -al *.fsxml
> -rw------- 1 freeswitch freeswitch 4985669 Apr 14 08:28 freeswitch-prod.xml.fsxml
> [root at freesw-p1 data]# wc -l *.fsxml
> 105025 freeswitch-prod.xml.fsxml
> 

this is just a copy of the compiled xml file.  it is 100k lines long.  I'm guessing you have lots of static users in here.

> 

None of this would indicate what you are suggesting necessarily 

> It's like it's leaking large numbers of complete copies of the XML. When I look directly at the core dump, it looks to 
> me like the strings are in the parsed state of the XML. (Below slightly masked copy and paste from viewing dump with less.)

Thats what i would accept.

> 
> 
> -------------
> ^@!-- xxx-xxx-xxxx --^@
> ^@user^@id^@"xxxxxxxxxxxxxxxxxxxxxxxx^@^@
>     ^@params^@
>         ^@!-- password won't be used by cisco devices --^@
>         ^@param^@name^@"password^@ value^@"XXXXXXXXXXXXXXXXX^@^@^@
>         ^@param^@name^@"vm-mailfrom^@ value^@"voicemail at mst.edu^@^@^@
> -----------
> 
> Would really appreciate any ideas on how I might mitigate this leaking or if there is anything that could be done to 
> help diagnose it further to help address the underlying issue.
> 
> 
> I'm happy to open a JIRA on this, but will NOT be able to test this with latest master as I can't just experiment with 
> the live production environment.
> 
> -- 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 anthony.minessale at gmail.com  Thu Apr 14 21:40:54 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Apr 2016 12:40:54 -0500
Subject: [Freeswitch-users] FS Logs show ERR msgs: not enough buffer
 space for required resample operation!
In-Reply-To: <CAO3obxYL9J__ZapmX3JVmRU-TkSHyueRHrrwohpxV+Ejt=Tx0g@mail.gmail.com>
References: <CAMtXxQmLrVi8oBUor7N8TMBfK8O=_jdXAUApY7py7d6OTtACmw@mail.gmail.com>
	<CAMtXxQm_t5GisLkYE3nCe775gq1T31XhxZvVZRPyygHx4__how@mail.gmail.com>
	<CAO3obxYL9J__ZapmX3JVmRU-TkSHyueRHrrwohpxV+Ejt=Tx0g@mail.gmail.com>
Message-ID: <CAKbxfG9_qEz2LrMPASXi6WKYhYTm0HYSCqDXXif17MxgMXA_8w@mail.gmail.com>

We do not field issues on this mailing list.
http://freeswitch.org/jira

Help keep the list about discussions and not support.


On Thu, Apr 14, 2016 at 8:58 AM, Piotr Gregor <piotrek.gregor at gmail.com>
wrote:

> Hi Joel,
>
> This might mean you try to resample between formats that differ too much,
> e.g. "to" rate is too big compared to "from" rate.
>
> cheers,
> Piotr
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From hunterj91 at hotmail.com  Thu Apr 14 21:42:45 2016
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Thu, 14 Apr 2016 17:42:45 +0000
Subject: [Freeswitch-users] Voicemail message File Retrieval on Remote
 Server
In-Reply-To: <CAPQovyQ17_tTY45GtN8ROsAwuFU079KmqeoBZpkKs0JFcsvDhQ@mail.gmail.com>
References: <DUB112-W6976FAEF7938D9FA6DD5F7BD970@phx.gbl>,
	<CAEwTsAg-9VrMQqxJUkxJ6V5Z8wpWsDRSQGw-J41KC7PJ1Y-kng@mail.gmail.com>,
	<CAPQovyRovMm6vCCpHeeyU+Fw+n1Xm=EqMB4n=g4vSG-6kvBH-g@mail.gmail.com>,
	<DUB112-W1020282C558B19A339B9B13BD970@phx.gbl>,
	<570FC00F.30205@mst.edu>,
	<CAO3hyzFqOVLjfiOU0x0Hz3W3UMwjYAK3on8u_Q74zwRqXey6YA@mail.gmail.com>,
	<CAFmZFiNrHQ7jMdCh4C4YNS8K4XMrj-7-MhoobYDO8bY-pS-sZA@mail.gmail.com>,
	<CAPQovyQ17_tTY45GtN8ROsAwuFU079KmqeoBZpkKs0JFcsvDhQ@mail.gmail.com>
Message-ID: <DUB112-W958EF92C3047CA644B45DFBD970@phx.gbl>

Hi Guys,
Great thanks for all the helpful comments, appreciate it!
I think will go the lua route.
Many thanks
Jon


---------- Forwarded message ----------
From: Abaci B <abaci64 at gmail.com>
Date: Thu, Apr 14, 2016 at 6:20 PM
Subject: Re: [Freeswitch-users] Voicemail message File Retrieval on Remote Server
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>


if it has to be http there are some fuse filesystems that use http as backend
On Thu, Apr 14, 2016 at 1:02 PM, Chris Tunbridge <blasterjr at gmail.com> wrote:
Jonathan, i just wanted to give my feedback in regaurds to this, i recommend going a different route, such as GlusterFS or writing your own voicemail in lua (really its not that hard!).

The issues that i ran into while trying to use http was that freeswitch wasn't able to check that the file existed and it would break certain functions such as the email voicemail and with things such as greetings, it would end up either creating empty directories or not returning success when saving.

I actually opened an issue about this a while back: https://freeswitch.org/jira/browse/FS-7280  there was some discussion on how it could be handled, however i think the easier route is to either use a shared filesystem setup like GlusterFS or to write your own voicemail module.

On Thu, Apr 14, 2016 at 10:06 AM, Nathan Neulinger <nneul at mst.edu> wrote:
I use a gluster filesystem as backend for storage on my FS cluster for media files, logs, etc.



-- Nathan



On 04/14/2016 10:01 AM, Jonathan Hunter wrote:

> Hi Michael,

>

> I appreciate the response, where in the code would I need to be looking?

>

> Also how are other people doing this? I presume they do some file sync do they to ensure all media servers have the same

> files?

>

> Many thanks

>

> Jon

>

>

>

> ---------- Forwarded message ----------

> From: *Michael Jerris* <mike at jerris.com <mailto:mike at jerris.com>>

> Date: Thu, Apr 14, 2016 at 3:32 PM

> Subject: Re: [Freeswitch-users] Voicemail message File Retrieval on Remote Server

> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>

>

>

> This is pretty clever but I think there are too many things in the code that assume it's a local directory for this to

> work without modifications to code

>

> On Thursday, April 14, 2016, Jonathan Hunter <hunterj91 at hotmail.com <mailto:hunterj91 at hotmail.com>> wrote:

>

>

>     Hi Guys,

>

>     Quick question, I am centralising Voicemail storage as I am using multiple FreeSWITCH servers, and I have this

>     working for recorded Voicemail greetings.

>

>     I am now doing this for stored voicemail messages.

>

>     So in voicemail.conf I am changing the storage-dir to the IP of my web server;

>

>

>     <param name="storage-dir" value="http://1.2.3.4:88"/>

>

>     Now this pushes the recorded wav file to the server using Put and works well, I just have 2 questions.

>

>     Locally I see FreeSWITCH creates the domain and extension folders in the original storage directory, can it be

>     configured to build them on remove servers or should this be done manually? (which isnt a problem).

>

>     And the main question, in terms of when retrieving the voicemail, how is this achieved, as currently I use;

>

>     <action application="voicemail" data="check default ]]..call_context..[[ ]]..sip_from_user..[["/>

>

>     In my Lua scripts, which still checks locally, where do I let the voicemail module know about retrieving, and will

>     it just perform a GET ?

>

>     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

>

>

>

> --

> Jonathan Hunter

> Technical Director /Telephony Developer

> M:(+44) 7917 190 438

> Email:jhunter at voxboxcoms.co.uk <mailto:Email%3Ajhunter 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

>



--

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

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



_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com



Official FreeSWITCH Sites

http://www.freeswitch.org

http://confluence.freeswitch.org

http://www.cluecon.com



FreeSWITCH-users mailing list

FreeSWITCH-users at 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 HunterTechnical Director /Telephony DeveloperM:(+44) 7917 190 438Email:jhunter at voxboxcoms.co.uk

 		 	   		  
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From nneul at mst.edu  Thu Apr 14 21:50:36 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 14 Apr 2016 12:50:36 -0500
Subject: [Freeswitch-users] Are there any known memory leaks with
 reloadxml and large XML configs?
In-Reply-To: <209D8D8A-7D31-4579-8E58-C15A2406EEDE@jerris.com>
References: <570FCA0B.3020308@mst.edu>
	<209D8D8A-7D31-4579-8E58-C15A2406EEDE@jerris.com>
Message-ID: <570FD86C.5050307@mst.edu>

The indication I'm reacting to of leak isn't "one in memory".... The below is showing FIFTY THREE of them.

-- Nathan

On 04/14/2016 12:38 PM, Michael Jerris wrote:
> A parsed xml tree in a core file would be normal.  We always keep one in memory, thats why you have to reload to get a new one, so it would be in the core file every time, not an indication of a leak.
>
>
>> On Apr 14, 2016, at 12:49 PM, Nathan Neulinger <nneul at mst.edu> wrote:
>>
>> I've got a persistent issue with a slow (sometimes not that slow) memory leak in FS. Our environment does frequent
>> reconfigs built around reloadxml and when the leak gets bad enough, I start getting random reports of slipping audio
>> sync. I've put in weekly forced failovers, but if we have a week with more changes than typical (we're still in middle
>> of mass migration off of CCM) - the leak growth can be too fast:
>>
>> freeswi+ 17977 12.1 27.2 15247652 2213648 ?    S<l  Apr02 2155:56 /local/freeswitch/server/bin/freeswitch -u freeswitch
>> -core -cfgname freeswitch-prod.xml -base /local/freeswitch/server -conf /local/freeswitch/server/conf -db
>> /local/freeswitch/server/db -run /local/freeswitch/data -log /local/freeswitch/data -rp -nf -nc -nonat -nonatmap
>>
>>
>> The reason I ask if the leak is related to reloadxml - when doing some analysis on the resulting core dump on the (no
>> longer active) server, I'm finding examples like this when doing a 'strings corefile | sort | uniq -c | sort -n':
>>
>> ...
>>    74642 !-- type(skinny) --
>>    76811 /buttons
>>    76811 "InvalidHash
>>    76812 "a1-hash
>>    76836 /skinny
>>    76853 buttons
>> ...
>>   105277 !-- password won't be used by cisco devices --
>> ...
>>   105277 "vm-mailfrom
>>   105277 "vm-password
>>
>> [root at freesw-p1 data]# grep -c InvalidHash freeswitch-prod.xml.fsxml
>> 1431
>>
>> [root at freesw-p1 data]# ls -al *.fsxml
>> -rw------- 1 freeswitch freeswitch 4985669 Apr 14 08:28 freeswitch-prod.xml.fsxml
>> [root at freesw-p1 data]# wc -l *.fsxml
>> 105025 freeswitch-prod.xml.fsxml
>>
>
> this is just a copy of the compiled xml file.  it is 100k lines long.  I'm guessing you have lots of static users in here.
>
>>
>
> None of this would indicate what you are suggesting necessarily
>
>> It's like it's leaking large numbers of complete copies of the XML. When I look directly at the core dump, it looks to
>> me like the strings are in the parsed state of the XML. (Below slightly masked copy and paste from viewing dump with less.)
>
> Thats what i would accept.
>
>>
>>
>> -------------
>> ^@!-- xxx-xxx-xxxx --^@
>> ^@user^@id^@"xxxxxxxxxxxxxxxxxxxxxxxx^@^@
>>      ^@params^@
>>          ^@!-- password won't be used by cisco devices --^@
>>          ^@param^@name^@"password^@ value^@"XXXXXXXXXXXXXXXXX^@^@^@
>>          ^@param^@name^@"vm-mailfrom^@ value^@"voicemail at mst.edu^@^@^@
>> -----------
>>
>> Would really appreciate any ideas on how I might mitigate this leaking or if there is anything that could be done to
>> help diagnose it further to help address the underlying issue.
>>
>>
>> I'm happy to open a JIRA on this, but will NOT be able to test this with latest master as I can't just experiment with
>> the live production environment.
>>
>> -- 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 pstarzyk at general-devices.com  Thu Apr 14 21:54:11 2016
From: pstarzyk at general-devices.com (Piotr Starzyk)
Date: Thu, 14 Apr 2016 13:54:11 -0400
Subject: [Freeswitch-users] How to dynamically set up conferences?
In-Reply-To: <7009D577-4530-45AE-B135-A3365F884CEF@jerris.com>
References: <59173f0f8e9631c3712aea0308b7b66d@mail.gmail.com>
	<7009D577-4530-45AE-B135-A3365F884CEF@jerris.com>
Message-ID: <fdaccaed469500ba093ac5f606ebe32e@mail.gmail.com>

I see.  In that case, I guess my question really boils down to converting a
call into a conference.  To ?move? a leg from a call to a conference, do I
need to be looking into ?bridge? or ?transfer? tool?



Found a similar question from two years back, sadly without the final
answer:

http://lists.freeswitch.org/pipermail/freeswitch-users/2015-January/110392.html







*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael Jerris
*Sent:* Thursday, April 14, 2016 1:31 PM
*To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
*Subject:* Re: [Freeswitch-users] How to dynamically set up conferences?



I've seen a lot of people use Approach II and it is always a huge mess,
don't do it.  Approach 1 is actually very simple, but you can do it even
simpler than this, you can just transfer both call legs into a conference
on the fly, you don't need to hang up and call back.  If you adjust params
to not get tones and such, its nearly unnoticeable it happens so quick.





On Apr 14, 2016, at 12:28 PM, Piotr Starzyk <pstarzyk at general-devices.com>
wrote:



Let?s say I have 3 SIP endpoints: Caller_A, Caller_B and Caller_C



Initially Caller_A calls Caller_B, so we end up with a SIP call between
them.



But now, there will be some cases, when Caller_A will want to ?conference
in? Caller_C into the call between Caller_A and Caller_B.



How is it normally done?  Based on my limited understanding of
mod_conference, I would attempt doing it as following:



Approach I:

Tear down the original call between A & B, and set up a conference between
A, B & C:

1.       Caller_A calls Caller_B

2.       SIP call is established between Caller_A and Caller_B.

3.       After a while Caller_A wants to conference in Caller_C, as
following:

a.       Caller_A hangs up the call with Caller_B

b.      Caller_A dials into a conference room.

c.       Caller_B dials into the same conference room.

d.      Caller_C dials into the same conference room.

4.       Caller_A, Caller_B and Caller_C are now in a conference.



Approach II:

Use conferences for all calls from the beginning, even the 2-way calls, to
make it easier to conference in additional callers:

1.       Caller_A dials into a conference room.

2.       Caller_B dials into a conference room.

3.       Calelr_A and Caller_B can now talk to each other.

4.       After a while Caller_A wants to conference in Caller_C

a.       Caller_C dials into the same conference room.

5.       Caller_A, Caller_B and Caller_C are now in a conference.





I?ve also read about eavesdrop tool, but I understand it only allows 3-way
communication, and I need to be able to support N-way.



Thanks.

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Thu Apr 14 21:58:46 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Apr 2016 12:58:46 -0500
Subject: [Freeswitch-users] Are there any known memory leaks with
 reloadxml and large XML configs?
In-Reply-To: <570FD86C.5050307@mst.edu>
References: <570FCA0B.3020308@mst.edu>
	<209D8D8A-7D31-4579-8E58-C15A2406EEDE@jerris.com>
	<570FD86C.5050307@mst.edu>
Message-ID: <CAKbxfG8Je3Y_E9p4Xx3oTakReAVNKRW6kQ+=Kc45Eep+v+gLVg@mail.gmail.com>

Have you done any debugging like running valgrind or ASAN on a server?
Did you try to reproduce it in a lab yet?

We can't really do any leg work for you so we would need a more specific
reproduction case.
We do offer commercial options as you are probably already are aware of
where we could deploy consultants to do more of the leg work for you.

Do you only reloadxml or do you reload any modules.
Also you should ALWAYS file a jira.  We don't want to field issues on the
list.  Its not a problem to close NOT A BUG if it turns out that way.






On Thu, Apr 14, 2016 at 12:50 PM, Nathan Neulinger <nneul at mst.edu> wrote:

> The indication I'm reacting to of leak isn't "one in memory".... The below
> is showing FIFTY THREE of them.
>
> -- Nathan
>
> On 04/14/2016 12:38 PM, Michael Jerris wrote:
> > A parsed xml tree in a core file would be normal.  We always keep one in
> memory, thats why you have to reload to get a new one, so it would be in
> the core file every time, not an indication of a leak.
> >
> >
> >> On Apr 14, 2016, at 12:49 PM, Nathan Neulinger <nneul at mst.edu> wrote:
> >>
> >> I've got a persistent issue with a slow (sometimes not that slow)
> memory leak in FS. Our environment does frequent
> >> reconfigs built around reloadxml and when the leak gets bad enough, I
> start getting random reports of slipping audio
> >> sync. I've put in weekly forced failovers, but if we have a week with
> more changes than typical (we're still in middle
> >> of mass migration off of CCM) - the leak growth can be too fast:
> >>
> >> freeswi+ 17977 12.1 27.2 15247652 2213648 ?    S<l  Apr02 2155:56
> /local/freeswitch/server/bin/freeswitch -u freeswitch
> >> -core -cfgname freeswitch-prod.xml -base /local/freeswitch/server -conf
> /local/freeswitch/server/conf -db
> >> /local/freeswitch/server/db -run /local/freeswitch/data -log
> /local/freeswitch/data -rp -nf -nc -nonat -nonatmap
> >>
> >>
> >> The reason I ask if the leak is related to reloadxml - when doing some
> analysis on the resulting core dump on the (no
> >> longer active) server, I'm finding examples like this when doing a
> 'strings corefile | sort | uniq -c | sort -n':
> >>
> >> ...
> >>    74642 !-- type(skinny) --
> >>    76811 /buttons
> >>    76811 "InvalidHash
> >>    76812 "a1-hash
> >>    76836 /skinny
> >>    76853 buttons
> >> ...
> >>   105277 !-- password won't be used by cisco devices --
> >> ...
> >>   105277 "vm-mailfrom
> >>   105277 "vm-password
> >>
> >> [root at freesw-p1 data]# grep -c InvalidHash freeswitch-prod.xml.fsxml
> >> 1431
> >>
> >> [root at freesw-p1 data]# ls -al *.fsxml
> >> -rw------- 1 freeswitch freeswitch 4985669 Apr 14 08:28
> freeswitch-prod.xml.fsxml
> >> [root at freesw-p1 data]# wc -l *.fsxml
> >> 105025 freeswitch-prod.xml.fsxml
> >>
> >
> > this is just a copy of the compiled xml file.  it is 100k lines long.
> I'm guessing you have lots of static users in here.
> >
> >>
> >
> > None of this would indicate what you are suggesting necessarily
> >
> >> It's like it's leaking large numbers of complete copies of the XML.
> When I look directly at the core dump, it looks to
> >> me like the strings are in the parsed state of the XML. (Below slightly
> masked copy and paste from viewing dump with less.)
> >
> > Thats what i would accept.
> >
> >>
> >>
> >> -------------
> >> ^@!-- xxx-xxx-xxxx --^@
> >> ^@user^@id^@"xxxxxxxxxxxxxxxxxxxxxxxx^@^@
> >>      ^@params^@
> >>          ^@!-- password won't be used by cisco devices --^@
> >>          ^@param^@name^@"password^@ value^@"XXXXXXXXXXXXXXXXX^@^@^@
> >>          ^@param^@name^@"vm-mailfrom^@ value^@"voicemail at mst.edu^@^@^@
> >> -----------
> >>
> >> Would really appreciate any ideas on how I might mitigate this leaking
> or if there is anything that could be done to
> >> help diagnose it further to help address the underlying issue.
> >>
> >>
> >> I'm happy to open a JIRA on this, but will NOT be able to test this
> with latest master as I can't just experiment with
> >> the live production environment.
> >>
> >> -- 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
>



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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From mike at jerris.com  Thu Apr 14 22:02:28 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Apr 2016 14:02:28 -0400
Subject: [Freeswitch-users] How to dynamically set up conferences?
In-Reply-To: <fdaccaed469500ba093ac5f606ebe32e@mail.gmail.com>
References: <59173f0f8e9631c3712aea0308b7b66d@mail.gmail.com>
	<7009D577-4530-45AE-B135-A3365F884CEF@jerris.com>
	<fdaccaed469500ba093ac5f606ebe32e@mail.gmail.com>
Message-ID: <50799E82-2585-4E39-9B54-C648E4FA2810@jerris.com>

if you are remotely controlling, you can uuid_transfer using the both flag

> On Apr 14, 2016, at 1:54 PM, Piotr Starzyk <pstarzyk at general-devices.com> wrote:
> 
> I see.  In that case, I guess my question really boils down to converting a call into a conference.  To ?move? a leg from a call to a conference, do I need to be looking into ?bridge? or ?transfer? tool?
>  
> Found a similar question from two years back, sadly without the final answer:
> http://lists.freeswitch.org/pipermail/freeswitch-users/2015-January/110392.html <http://lists.freeswitch.org/pipermail/freeswitch-users/2015-January/110392.html>
>  
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Thursday, April 14, 2016 1:31 PM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] How to dynamically set up conferences?
>  
> I've seen a lot of people use Approach II and it is always a huge mess, don't do it.  Approach 1 is actually very simple, but you can do it even simpler than this, you can just transfer both call legs into a conference on the fly, you don't need to hang up and call back.  If you adjust params to not get tones and such, its nearly unnoticeable it happens so quick.
>  
>  
>> On Apr 14, 2016, at 12:28 PM, Piotr Starzyk <pstarzyk at general-devices.com <mailto:pstarzyk at general-devices.com>> wrote:
>>  
>> Let?s say I have 3 SIP endpoints: Caller_A, Caller_B and Caller_C
>>  
>> Initially Caller_A calls Caller_B, so we end up with a SIP call between them.
>>  
>> But now, there will be some cases, when Caller_A will want to ?conference in? Caller_C into the call between Caller_A and Caller_B.
>>  
>> How is it normally done?  Based on my limited understanding of mod_conference, I would attempt doing it as following:
>>  
>> Approach I: 
>> Tear down the original call between A & B, and set up a conference between A, B & C:
>> 1.       Caller_A calls Caller_B
>> 2.       SIP call is established between Caller_A and Caller_B.
>> 3.       After a while Caller_A wants to conference in Caller_C, as following:
>> a.       Caller_A hangs up the call with Caller_B
>> b.      Caller_A dials into a conference room.
>> c.       Caller_B dials into the same conference room.
>> d.      Caller_C dials into the same conference room.
>> 4.       Caller_A, Caller_B and Caller_C are now in a conference.
>>  
>> Approach II: 
>> Use conferences for all calls from the beginning, even the 2-way calls, to make it easier to conference in additional callers:
>> 1.       Caller_A dials into a conference room.
>> 2.       Caller_B dials into a conference room.
>> 3.       Calelr_A and Caller_B can now talk to each other.
>> 4.       After a while Caller_A wants to conference in Caller_C
>> a.       Caller_C dials into the same conference room.
>> 5.       Caller_A, Caller_B and Caller_C are now in a conference.
>>  
>>  
>> I?ve also read about eavesdrop tool, but I understand it only allows 3-way communication, and I need to be able to support N-way.
>>  
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From anthony.minessale at gmail.com  Thu Apr 14 22:07:30 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Apr 2016 13:07:30 -0500
Subject: [Freeswitch-users] help, with Verto - verto-min.js configuration
In-Reply-To: <570EE45E.3060008@connectto.com>
References: <570EE45E.3060008@connectto.com>
Message-ID: <CAKbxfG-SDDK6nPwCbByMZgttbRX87CkWf3tE3FGGGVeA9FvK9g@mail.gmail.com>

You don't edit verto.js you pass the iceServers param to new $.verto

an array of js objs where each one has single param url


[{url: stun:1234}]





On Wed, Apr 13, 2016 at 7:29 PM, Arsen Manukyan <arsenman at connectto.com>
wrote:

>
> Please help, with Verto - verto-min.js configuration
> After changing username, password for STUN and TURN server looks like my
> TURN server doesn't receive any messages form Webrtc phone
> here is my config:
>
>
> STUN={url:!moz?'stun:turn.mydomain.com:3478':'stun:204.174.104.38'};var
> TURN={url:'turn:ars at tu
> rn.mydomain.com:3478',credential:'ars123'};var
> iceServers=null;if(options.iceServers){var
> tmp=options.iceServers;if(typeof(tmp)==
> ="boolean"){tmp=null;}
> if(tmp&&!(typeof(tmp)=="object"&&tmp.constructor===Array)){console.warn("iceServers
> must be an array, reverting to default ice serve
> rs");tmp=null;}
>
> iceServers={iceServers:tmp||[STUN]};if(!moz&&!tmp){if(parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2])>=28)TURN={u
> rl:'turn:turn.mydomain.com:3478',credential:'ars123',username:'ars'};
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From pstarzyk at general-devices.com  Thu Apr 14 22:07:36 2016
From: pstarzyk at general-devices.com (Piotr Starzyk)
Date: Thu, 14 Apr 2016 14:07:36 -0400
Subject: [Freeswitch-users] How to dynamically set up conferences?
In-Reply-To: <50799E82-2585-4E39-9B54-C648E4FA2810@jerris.com>
References: <59173f0f8e9631c3712aea0308b7b66d@mail.gmail.com>
	<7009D577-4530-45AE-B135-A3365F884CEF@jerris.com>	<fdaccaed469500ba093ac5f606ebe32e@mail.gmail.com>
	<50799E82-2585-4E39-9B54-C648E4FA2810@jerris.com>
Message-ID: <c7448f91a3d6c5acc13e1fdad522f51a@mail.gmail.com>

Thanks, will give that a try, looks like it should do the trick.



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael Jerris
*Sent:* Thursday, April 14, 2016 2:02 PM
*To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
*Subject:* Re: [Freeswitch-users] How to dynamically set up conferences?



if you are remotely controlling, you can uuid_transfer using the both flag



On Apr 14, 2016, at 1:54 PM, Piotr Starzyk <pstarzyk at general-devices.com>
wrote:



I see.  In that case, I guess my question really boils down to converting a
call into a conference.  To ?move? a leg from a call to a conference, do I
need to be looking into ?bridge? or ?transfer? tool?



Found a similar question from two years back, sadly without the final
answer:

http://lists.freeswitch.org/pipermail/freeswitch-users/2015-January/110392.html







*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael Jerris
*Sent:* Thursday, April 14, 2016 1:31 PM
*To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
*Subject:* Re: [Freeswitch-users] How to dynamically set up conferences?



I've seen a lot of people use Approach II and it is always a huge mess,
don't do it.  Approach 1 is actually very simple, but you can do it even
simpler than this, you can just transfer both call legs into a conference
on the fly, you don't need to hang up and call back.  If you adjust params
to not get tones and such, its nearly unnoticeable it happens so quick.





On Apr 14, 2016, at 12:28 PM, Piotr Starzyk <pstarzyk at general-devices.com>
wrote:



Let?s say I have 3 SIP endpoints: Caller_A, Caller_B and Caller_C



Initially Caller_A calls Caller_B, so we end up with a SIP call between
them.



But now, there will be some cases, when Caller_A will want to ?conference
in? Caller_C into the call between Caller_A and Caller_B.



How is it normally done?  Based on my limited understanding of
mod_conference, I would attempt doing it as following:



Approach I:

Tear down the original call between A & B, and set up a conference between
A, B & C:

1.       Caller_A calls Caller_B

2.       SIP call is established between Caller_A and Caller_B.

3.       After a while Caller_A wants to conference in Caller_C, as
following:

a.       Caller_A hangs up the call with Caller_B

b.      Caller_A dials into a conference room.

c.       Caller_B dials into the same conference room.

d.      Caller_C dials into the same conference room.

4.       Caller_A, Caller_B and Caller_C are now in a conference.



Approach II:

Use conferences for all calls from the beginning, even the 2-way calls, to
make it easier to conference in additional callers:

1.       Caller_A dials into a conference room.

2.       Caller_B dials into a conference room.

3.       Calelr_A and Caller_B can now talk to each other.

4.       After a while Caller_A wants to conference in Caller_C

a.       Caller_C dials into the same conference room.

5.       Caller_A, Caller_B and Caller_C are now in a conference.





I?ve also read about eavesdrop tool, but I understand it only allows 3-way
communication, and I need to be able to support N-way.
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From anthony.minessale at gmail.com  Thu Apr 14 22:18:11 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Apr 2016 13:18:11 -0500
Subject: [Freeswitch-users] Are there any known memory leaks with
 reloadxml and large XML configs?
In-Reply-To: <CAKbxfG8Je3Y_E9p4Xx3oTakReAVNKRW6kQ+=Kc45Eep+v+gLVg@mail.gmail.com>
References: <570FCA0B.3020308@mst.edu>
	<209D8D8A-7D31-4579-8E58-C15A2406EEDE@jerris.com>
	<570FD86C.5050307@mst.edu>
	<CAKbxfG8Je3Y_E9p4Xx3oTakReAVNKRW6kQ+=Kc45Eep+v+gLVg@mail.gmail.com>
Message-ID: <CAKbxfG_9mdSMSezO5XHGgUyn2ihmHJHq6tVPx3UWcJF9X6sjGA@mail.gmail.com>

P.S.
Check all the modules you use to make sure when they open the XML registry
that they close it again.


On Thu, Apr 14, 2016 at 12:58 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Have you done any debugging like running valgrind or ASAN on a server?
> Did you try to reproduce it in a lab yet?
>
> We can't really do any leg work for you so we would need a more specific
> reproduction case.
> We do offer commercial options as you are probably already are aware of
> where we could deploy consultants to do more of the leg work for you.
>
> Do you only reloadxml or do you reload any modules.
> Also you should ALWAYS file a jira.  We don't want to field issues on the
> list.  Its not a problem to close NOT A BUG if it turns out that way.
>
>
>
>
>
>
> On Thu, Apr 14, 2016 at 12:50 PM, Nathan Neulinger <nneul at mst.edu> wrote:
>
>> The indication I'm reacting to of leak isn't "one in memory".... The
>> below is showing FIFTY THREE of them.
>>
>> -- Nathan
>>
>> On 04/14/2016 12:38 PM, Michael Jerris wrote:
>> > A parsed xml tree in a core file would be normal.  We always keep one
>> in memory, thats why you have to reload to get a new one, so it would be in
>> the core file every time, not an indication of a leak.
>> >
>> >
>> >> On Apr 14, 2016, at 12:49 PM, Nathan Neulinger <nneul at mst.edu> wrote:
>> >>
>> >> I've got a persistent issue with a slow (sometimes not that slow)
>> memory leak in FS. Our environment does frequent
>> >> reconfigs built around reloadxml and when the leak gets bad enough, I
>> start getting random reports of slipping audio
>> >> sync. I've put in weekly forced failovers, but if we have a week with
>> more changes than typical (we're still in middle
>> >> of mass migration off of CCM) - the leak growth can be too fast:
>> >>
>> >> freeswi+ 17977 12.1 27.2 15247652 2213648 ?    S<l  Apr02 2155:56
>> /local/freeswitch/server/bin/freeswitch -u freeswitch
>> >> -core -cfgname freeswitch-prod.xml -base /local/freeswitch/server
>> -conf /local/freeswitch/server/conf -db
>> >> /local/freeswitch/server/db -run /local/freeswitch/data -log
>> /local/freeswitch/data -rp -nf -nc -nonat -nonatmap
>> >>
>> >>
>> >> The reason I ask if the leak is related to reloadxml - when doing some
>> analysis on the resulting core dump on the (no
>> >> longer active) server, I'm finding examples like this when doing a
>> 'strings corefile | sort | uniq -c | sort -n':
>> >>
>> >> ...
>> >>    74642 !-- type(skinny) --
>> >>    76811 /buttons
>> >>    76811 "InvalidHash
>> >>    76812 "a1-hash
>> >>    76836 /skinny
>> >>    76853 buttons
>> >> ...
>> >>   105277 !-- password won't be used by cisco devices --
>> >> ...
>> >>   105277 "vm-mailfrom
>> >>   105277 "vm-password
>> >>
>> >> [root at freesw-p1 data]# grep -c InvalidHash freeswitch-prod.xml.fsxml
>> >> 1431
>> >>
>> >> [root at freesw-p1 data]# ls -al *.fsxml
>> >> -rw------- 1 freeswitch freeswitch 4985669 Apr 14 08:28
>> freeswitch-prod.xml.fsxml
>> >> [root at freesw-p1 data]# wc -l *.fsxml
>> >> 105025 freeswitch-prod.xml.fsxml
>> >>
>> >
>> > this is just a copy of the compiled xml file.  it is 100k lines long.
>> I'm guessing you have lots of static users in here.
>> >
>> >>
>> >
>> > None of this would indicate what you are suggesting necessarily
>> >
>> >> It's like it's leaking large numbers of complete copies of the XML.
>> When I look directly at the core dump, it looks to
>> >> me like the strings are in the parsed state of the XML. (Below
>> slightly masked copy and paste from viewing dump with less.)
>> >
>> > Thats what i would accept.
>> >
>> >>
>> >>
>> >> -------------
>> >> ^@!-- xxx-xxx-xxxx --^@
>> >> ^@user^@id^@"xxxxxxxxxxxxxxxxxxxxxxxx^@^@
>> >>      ^@params^@
>> >>          ^@!-- password won't be used by cisco devices --^@
>> >>          ^@param^@name^@"password^@ value^@"XXXXXXXXXXXXXXXXX^@^@^@
>> >>          ^@param^@name^@"vm-mailfrom^@ value^@"voicemail at mst.edu^@^@^@
>> >> -----------
>> >>
>> >> Would really appreciate any ideas on how I might mitigate this leaking
>> or if there is anything that could be done to
>> >> help diagnose it further to help address the underlying issue.
>> >>
>> >>
>> >> I'm happy to open a JIRA on this, but will NOT be able to test this
>> with latest master as I can't just experiment with
>> >> the live production environment.
>> >>
>> >> -- 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
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>



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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From nneul at mst.edu  Thu Apr 14 22:25:28 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 14 Apr 2016 13:25:28 -0500
Subject: [Freeswitch-users] Are there any known memory leaks with
 reloadxml and large XML configs?
In-Reply-To: <CAKbxfG8Je3Y_E9p4Xx3oTakReAVNKRW6kQ+=Kc45Eep+v+gLVg@mail.gmail.com>
References: <570FCA0B.3020308@mst.edu>
	<209D8D8A-7D31-4579-8E58-C15A2406EEDE@jerris.com>
	<570FD86C.5050307@mst.edu>
	<CAKbxfG8Je3Y_E9p4Xx3oTakReAVNKRW6kQ+=Kc45Eep+v+gLVg@mail.gmail.com>
Message-ID: <570FE098.3090002@mst.edu>

FS-9074 opened.

I've attempted to reproduce in lab environment, but not with much success since I can't readily reproduce the 
activity/usage level from the production environment. On production env, I can readily see the resident memory growing 
when I reloadxml on the active server, but it doesn't do it every time, so it's clearly some sort of combination of 
events/timing related.

On the lab setup, I typically see about 3 x duplicates of the strings from the XML in a generated core.

No modules are being reloaded during normal operations. Just a rewrite of the configs followed by a reloadxml.


As far as the commercial option - have asked y'all several times for what our options would be for that sort of service, 
but have never gotten any clear answer particularly given the custom deployment we have. Would be very interested in 
presenting that to management here if you can give me some proposed service offerings and costs.

-- Nathan

On 04/14/2016 12:58 PM, Anthony Minessale wrote:
> Have you done any debugging like running valgrind or ASAN on a server?
> Did you try to reproduce it in a lab yet?
>
> We can't really do any leg work for you so we would need a more specific reproduction case.
> We do offer commercial options as you are probably already are aware of where we could deploy consultants to do more of
> the leg work for you.
>
> Do you only reloadxml or do you reload any modules.
> Also you should ALWAYS file a jira.  We don't want to field issues on the list.  Its not a problem to close NOT A BUG if
> it turns out that way.
>
>
>
>
>
>
> On Thu, Apr 14, 2016 at 12:50 PM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>> wrote:
>
>     The indication I'm reacting to of leak isn't "one in memory".... The below is showing FIFTY THREE of them.
>
>     -- Nathan


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


From arsenman at connectto.com  Thu Apr 14 22:33:36 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Thu, 14 Apr 2016 11:33:36 -0700
Subject: [Freeswitch-users] help, with Verto - verto-min.js configuration
In-Reply-To: <CAKbxfG-SDDK6nPwCbByMZgttbRX87CkWf3tE3FGGGVeA9FvK9g@mail.gmail.com>
References: <570EE45E.3060008@connectto.com>
	<CAKbxfG-SDDK6nPwCbByMZgttbRX87CkWf3tE3FGGGVeA9FvK9g@mail.gmail.com>
Message-ID: <570FE280.7090807@connectto.com>

Hi Antony

Sorry maybe i misunderstand you

Original file is:

var 
STUN={url:!moz?'stun:stun.l.google.com:19302':'stun:23.21.150.121'};var 
TURN={url:'turn:homeo at tur
n.bistri.com:80',credential:'homeo'};var 
iceServers=null;if(options.iceServers){var 
tmp=options.iceServers;if(typeof(tmp)==="boolean
"){tmp=null;}
if(tmp&&!(typeof(tmp)=="object"&&tmp.constructor===Array)){console.warn("iceServers 
must be an array, reverting to default ice serve
rs");tmp=null;}
iceServers={iceServers:tmp||[STUN]};if(!moz&&!tmp){if(parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2])>=28)TURN={u
rl:'turn:turn.bistri.com:80',credential:'homeo',username:'homeo'};iceServers.iceServers=[STUN];}}
var 
optional={optional:[]};if(!moz){optional.optional=[{DtlsSrtpKeyAgreement:true},{RtpDataChannels:options.onChannelMessage?true:fa
lse}];}


My credentials is:

STUN Server = stun.mydomain.com
TURN Server = turn.mydomain.com

TURN user = user1
TURN pass = PassUser1

Can you please show me what exactly configuration should i have?





On 4/14/2016 11:07 AM, Anthony Minessale wrote:
> You don't edit verto.js you pass the iceServers param to new$.verto
>
> an array of js objs where each one has single param url
>
>
> [{url: stun:1234}]
>
>
>
>
>
> On Wed, Apr 13, 2016 at 7:29 PM, Arsen Manukyan 
> <arsenman at connectto.com <mailto:arsenman at connectto.com>> wrote:
>
>
>     Please help, with Verto - verto-min.js configuration
>     After changing username, password for STUN and TURN server looks
>     like my TURN server doesn't receive any messages form Webrtc phone
>     here is my config:
>
>
>     STUN={url:!moz?'stun:turn.mydomain.com:3478':'stun:204.174.104.38'};var
>     TURN={url:'turn:ars at tu
>     rn.mydomain.com:3478
>     <http://rn.mydomain.com:3478>',credential:'ars123'};var
>     iceServers=null;if(options.iceServers){var
>     tmp=options.iceServers;if(typeof(tmp)==
>     ="boolean"){tmp=null;}
>     if(tmp&&!(typeof(tmp)=="object"&&tmp.constructor===Array)){console.warn("iceServers
>     must be an array, reverting to default ice serve
>     rs");tmp=null;}
>     iceServers={iceServers:tmp||[STUN]};if(!moz&&!tmp){if(parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2])>=28)TURN={u
>     rl:'turn:turn.mydomain.com:3478
>     <http://turn.mydomain.com:3478>',credential:'ars123',username:'ars'};
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ? 
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net <http://irc.freenode.net> #freeswitch ? 
> _http://freeswitch.org/g+_
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Arsen Manukyan
ConnectTo Communications Inc.
555 Riverdale Dr., Suite A
Glendale, CA 91204
arsenman at connectto.com
http://www.ConnectTo.com
Tel. 818.546.4636
FAX 818.546.4617
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From mike at jerris.com  Thu Apr 14 22:37:39 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 14 Apr 2016 14:37:39 -0400
Subject: [Freeswitch-users] help, with Verto - verto-min.js configuration
In-Reply-To: <570FE280.7090807@connectto.com>
References: <570EE45E.3060008@connectto.com>
	<CAKbxfG-SDDK6nPwCbByMZgttbRX87CkWf3tE3FGGGVeA9FvK9g@mail.gmail.com>
	<570FE280.7090807@connectto.com>
Message-ID: <F786B5C6-9190-4585-9078-F681614F291E@jerris.com>

You most certainly don't edit the minified version directly as well.  

> On Apr 14, 2016, at 2:33 PM, Arsen Manukyan <arsenman at connectto.com> wrote:
> 
> Hi Antony 
> 
> Sorry maybe i misunderstand you
> 
> Original file is: 
> 
> var STUN={url:!moz?'stun:stun.l.google.com:19302':'stun:23.21.150.121'};var TURN={url:'turn:homeo at tur
> n.bistri.com:80',credential:'homeo'};var iceServers=null;if(options.iceServers){var tmp=options.iceServers;if(typeof(tmp)==="boolean
> "){tmp=null;}
> if(tmp&&!(typeof(tmp)=="object"&&tmp.constructor===Array)){console.warn("iceServers must be an array, reverting to default ice serve
> rs");tmp=null;}
> iceServers={iceServers:tmp||[STUN]};if(!moz&&!tmp){if(parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2])>=28)TURN={u
> rl:'turn:turn.bistri.com:80',credential:'homeo',username:'homeo'};iceServers.iceServers=[STUN];}}
> var optional={optional:[]};if(!moz){optional.optional=[{DtlsSrtpKeyAgreement:true},{RtpDataChannels:options.onChannelMessage?true:fa
> lse}];}
> 
> 
> My credentials is:
> 
> STUN Server = stun.mydomain.com
> TURN Server = turn.mydomain.com
> 
> TURN user = user1
> TURN pass = PassUser1
> 
> Can you please show me what exactly configuration should i have?
> 
> 
> 
> 
> 
> On 4/14/2016 11:07 AM, Anthony Minessale wrote:
>> You don't edit verto.js you pass the iceServers param to new $.verto
>> an array of js objs where each one has single param url
>> 
>> 
>> 
>> [{url: stun:1234}]
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> On Wed, Apr 13, 2016 at 7:29 PM, Arsen Manukyan <arsenman at connectto.com <mailto:arsenman at connectto.com>> wrote:
>> 
>> Please help, with Verto - verto-min.js configuration 
>> After changing username, password for STUN and TURN server looks like my TURN server doesn't receive any messages form Webrtc phone  
>> here is my config:
>> 
>> 
>> STUN={url:!moz?'stun:turn.mydomain.com:3478':'stun:204.174.104.38'};var TURN={url:'turn:ars at tu
>> rn.mydomain.com:3478 <http://rn.mydomain.com:3478/>',credential:'ars123'};var iceServers=null;if(options.iceServers){var tmp=options.iceServers;if(typeof(tmp)==
>> ="boolean"){tmp=null;}
>> if(tmp&&!(typeof(tmp)=="object"&&tmp.constructor===Array)){console.warn("iceServers must be an array, reverting to default ice serve
>> rs");tmp=null;}
>> iceServers={iceServers:tmp||[STUN]};if(!moz&&!tmp){if(parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2])>=28)TURN={u
>> rl:'turn:turn.mydomain.com:3478 <http://turn.mydomain.com:3478/>',credential:'ars123',username:'ars'};
>> 
>> _
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From arsenman at connectto.com  Thu Apr 14 22:42:43 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Thu, 14 Apr 2016 11:42:43 -0700
Subject: [Freeswitch-users] help, with Verto - verto-min.js configuration
In-Reply-To: <F786B5C6-9190-4585-9078-F681614F291E@jerris.com>
References: <570EE45E.3060008@connectto.com>
	<CAKbxfG-SDDK6nPwCbByMZgttbRX87CkWf3tE3FGGGVeA9FvK9g@mail.gmail.com>
	<570FE280.7090807@connectto.com>
	<F786B5C6-9190-4585-9078-F681614F291E@jerris.com>
Message-ID: <570FE4A3.6020804@connectto.com>

how to edit, which version ?

On 4/14/2016 11:37 AM, Michael Jerris wrote:
> You most certainly don't edit the minified version directly as well.

-- 
Arsen Manukyan
ConnectTo Communications Inc.
555 Riverdale Dr., Suite A
Glendale, CA 91204
arsenman at connectto.com
http://www.ConnectTo.com
Tel. 818.546.4636
FAX 818.546.4617
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From nneul at mst.edu  Thu Apr 14 23:24:58 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 14 Apr 2016 14:24:58 -0500
Subject: [Freeswitch-users] Are there any known memory leaks with
 reloadxml and large XML configs?
In-Reply-To: <570FE098.3090002@mst.edu>
References: <570FCA0B.3020308@mst.edu>
	<209D8D8A-7D31-4579-8E58-C15A2406EEDE@jerris.com>
	<570FD86C.5050307@mst.edu>
	<CAKbxfG8Je3Y_E9p4Xx3oTakReAVNKRW6kQ+=Kc45Eep+v+gLVg@mail.gmail.com>
	<570FE098.3090002@mst.edu>
Message-ID: <570FEE8A.9060803@mst.edu>

Running some more post analysis over the raw content of the core dump, and extracting each parsed blob of XML, it looks 
like almost all of them are complete copies (and not partial like if it ran out of room and had to resize+relocate).

I extracted all of the chunks to individual files, and looking at each one in turn based on sequential position in 
memory, I am seeing reasonably expected small/tiny changes from one to the next that would correspond to incremental 
changes in our configuration UI (stuff like individual changes to a line key, button position, voicemail address, 
password, etc.)


What in the system results in a full copy of the parsed XML structure? Does every access do some sort of temporary copy 
like you'd get with SQL and transaction isolation?


-- Nathan

On 04/14/2016 01:25 PM, Nathan Neulinger wrote:
> FS-9074 opened.
>
> I've attempted to reproduce in lab environment, but not with much success since I can't readily reproduce the
> activity/usage level from the production environment. On production env, I can readily see the resident memory growing
> when I reloadxml on the active server, but it doesn't do it every time, so it's clearly some sort of combination of
> events/timing related.
>
> On the lab setup, I typically see about 3 x duplicates of the strings from the XML in a generated core.
>
> No modules are being reloaded during normal operations. Just a rewrite of the configs followed by a reloadxml.
>
>
> As far as the commercial option - have asked y'all several times for what our options would be for that sort of service,
> but have never gotten any clear answer particularly given the custom deployment we have. Would be very interested in
> presenting that to management here if you can give me some proposed service offerings and costs.
>
> -- Nathan
>
> On 04/14/2016 12:58 PM, Anthony Minessale wrote:
>> Have you done any debugging like running valgrind or ASAN on a server?
>> Did you try to reproduce it in a lab yet?
>>
>> We can't really do any leg work for you so we would need a more specific reproduction case.
>> We do offer commercial options as you are probably already are aware of where we could deploy consultants to do more of
>> the leg work for you.
>>
>> Do you only reloadxml or do you reload any modules.
>> Also you should ALWAYS file a jira.  We don't want to field issues on the list.  Its not a problem to close NOT A BUG if
>> it turns out that way.
>>
>>
>>
>>
>>
>>
>> On Thu, Apr 14, 2016 at 12:50 PM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>> wrote:
>>
>>     The indication I'm reacting to of leak isn't "one in memory".... The below is showing FIFTY THREE of them.
>>
>>     -- Nathan
>
>

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


From daveh at beachdognet.com  Fri Apr 15 07:40:14 2016
From: daveh at beachdognet.com (Dave Horton)
Date: Thu, 14 Apr 2016 23:40:14 -0400
Subject: [Freeswitch-users] rfc 2833 not offered on outbound leg in bridge
	scenario with FS 1.6.7 (worked properly in FS 1.6.6)
Message-ID: <9F1A8D0A-19D3-4C6C-84D6-AA8FF4018CF4@beachdognet.com>

I?m running into a problem on v1.6.7 with something that seemed to work properly on v1.6.6, and I am wondering if this is a known bug.

I have an application that receives an incoming call and bridges it to an outbound destination.  Where there is RFC 2833 offered on the A leg for dtmf I want that to offer that on the B leg as well.  
I have defined late-negotiation=true on my sip profile.

In v1.6.6 (specifically, User agent header identifies as 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit) this worked as expected.

In v1.6.7 (specifically, User agent header identifies as 1.6.7+git~20160401T134007Z~f0c3870be3~64bit) it doesn?t.
The SDP offered on the B leg does not include RFC 2833 even though the A leg offered it.

Is this a known bug?  If so, is there a patch?

If not, I can provide more information (pastebin logs, etc) to help troubleshoot.  Please let me know what would be most useful to see.

From s.safarov at gmail.com  Fri Apr 15 07:53:22 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 15 Apr 2016 03:53:22 +0000
Subject: [Freeswitch-users] rfc 2833 not offered on outbound leg in
 bridge scenario with FS 1.6.7 (worked properly in FS 1.6.6)
In-Reply-To: <9F1A8D0A-19D3-4C6C-84D6-AA8FF4018CF4@beachdognet.com>
References: <9F1A8D0A-19D3-4C6C-84D6-AA8FF4018CF4@beachdognet.com>
Message-ID: <CAHtxdDccOnAT-3QyE2SpUtRoGpMMDtQc-np1DR4-ksnZjZDefQ@mail.gmail.com>

Could you please find commit where is bug has been introduced using "git
bisect" commands.

On Fri, Apr 15, 2016, 06:41 Dave Horton <daveh at beachdognet.com> wrote:

> I?m running into a problem on v1.6.7 with something that seemed to work
> properly on v1.6.6, and I am wondering if this is a known bug.
>
> I have an application that receives an incoming call and bridges it to an
> outbound destination.  Where there is RFC 2833 offered on the A leg for
> dtmf I want that to offer that on the B leg as well.
> I have defined late-negotiation=true on my sip profile.
>
> In v1.6.6 (specifically, User agent header identifies as
> 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit) this worked as expected.
>
> In v1.6.7 (specifically, User agent header identifies as
> 1.6.7+git~20160401T134007Z~f0c3870be3~64bit) it doesn?t.
> The SDP offered on the B leg does not include RFC 2833 even though the A
> leg offered it.
>
> Is this a known bug?  If so, is there a patch?
>
> If not, I can provide more information (pastebin logs, etc) to help
> troubleshoot.  Please let me know what would be most useful to see.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 15 08:41:26 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 14 Apr 2016 23:41:26 -0500
Subject: [Freeswitch-users] Are there any known memory leaks with
 reloadxml and large XML configs?
In-Reply-To: <570FEE8A.9060803@mst.edu>
References: <570FCA0B.3020308@mst.edu>
	<209D8D8A-7D31-4579-8E58-C15A2406EEDE@jerris.com>
	<570FD86C.5050307@mst.edu>
	<CAKbxfG8Je3Y_E9p4Xx3oTakReAVNKRW6kQ+=Kc45Eep+v+gLVg@mail.gmail.com>
	<570FE098.3090002@mst.edu> <570FEE8A.9060803@mst.edu>
Message-ID: <CAKbxfG9oi5SrT=-xWNzcGV=WQxZZqiABXySU-nnD6jhDmQ4QXA@mail.gmail.com>

Why are you still discussing this here when you have a jira open?  I am not
sure how to properly explain the IMMENSE difficulty of working on bugs over
email.  There is no right reason to side with email when trying to decide
between jira and email.  If a jira turns out not to be a bug, one click to
close it.  When an email turns out to be a bug, frick, now we have to copy
all the data over.  Pass it on, file jiras when you have a problem.  We
have broadcasted it here for years..:.....

The xml registry is ref counted if it reloads while its open, it probably
means something you use opens the xml root and does not close it.  I said
this on your jira.


On Thursday, April 14, 2016, Nathan Neulinger <nneul at mst.edu> wrote:

> Running some more post analysis over the raw content of the core dump, and
> extracting each parsed blob of XML, it looks
> like almost all of them are complete copies (and not partial like if it
> ran out of room and had to resize+relocate).
>
> I extracted all of the chunks to individual files, and looking at each one
> in turn based on sequential position in
> memory, I am seeing reasonably expected small/tiny changes from one to the
> next that would correspond to incremental
> changes in our configuration UI (stuff like individual changes to a line
> key, button position, voicemail address,
> password, etc.)
>
>
> What in the system results in a full copy of the parsed XML structure?
> Does every access do some sort of temporary copy
> like you'd get with SQL and transaction isolation?
>
>
> -- Nathan
>
> On 04/14/2016 01:25 PM, Nathan Neulinger wrote:
> > FS-9074 opened.
> >
> > I've attempted to reproduce in lab environment, but not with much
> success since I can't readily reproduce the
> > activity/usage level from the production environment. On production env,
> I can readily see the resident memory growing
> > when I reloadxml on the active server, but it doesn't do it every time,
> so it's clearly some sort of combination of
> > events/timing related.
> >
> > On the lab setup, I typically see about 3 x duplicates of the strings
> from the XML in a generated core.
> >
> > No modules are being reloaded during normal operations. Just a rewrite
> of the configs followed by a reloadxml.
> >
> >
> > As far as the commercial option - have asked y'all several times for
> what our options would be for that sort of service,
> > but have never gotten any clear answer particularly given the custom
> deployment we have. Would be very interested in
> > presenting that to management here if you can give me some proposed
> service offerings and costs.
> >
> > -- Nathan
> >
> > On 04/14/2016 12:58 PM, Anthony Minessale wrote:
> >> Have you done any debugging like running valgrind or ASAN on a server?
> >> Did you try to reproduce it in a lab yet?
> >>
> >> We can't really do any leg work for you so we would need a more
> specific reproduction case.
> >> We do offer commercial options as you are probably already are aware of
> where we could deploy consultants to do more of
> >> the leg work for you.
> >>
> >> Do you only reloadxml or do you reload any modules.
> >> Also you should ALWAYS file a jira.  We don't want to field issues on
> the list.  Its not a problem to close NOT A BUG if
> >> it turns out that way.
> >>
> >>
> >>
> >>
> >>
> >>
> >> On Thu, Apr 14, 2016 at 12:50 PM, Nathan Neulinger <nneul at mst.edu
> <javascript:;> <mailto:nneul at mst.edu <javascript:;>>> wrote:
> >>
> >>     The indication I'm reacting to of leak isn't "one in memory"....
> The below is showing FIFTY THREE of them.
> >>
> >>     -- Nathan
> >
> >
>
> --
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu <javascript:;>
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From jaybinks at gmail.com  Fri Apr 15 10:14:55 2016
From: jaybinks at gmail.com (jay binks)
Date: Fri, 15 Apr 2016 16:14:55 +1000
Subject: [Freeswitch-users] rfc 2833 not offered on outbound leg in
 bridge scenario with FS 1.6.7 (worked properly in FS 1.6.6)
In-Reply-To: <CAHtxdDccOnAT-3QyE2SpUtRoGpMMDtQc-np1DR4-ksnZjZDefQ@mail.gmail.com>
References: <9F1A8D0A-19D3-4C6C-84D6-AA8FF4018CF4@beachdognet.com>
	<CAHtxdDccOnAT-3QyE2SpUtRoGpMMDtQc-np1DR4-ksnZjZDefQ@mail.gmail.com>
Message-ID: <CAO47HpdvpfhFr0Qf97=2pM+Qbb9Sacv-AZqFon1hicrp3W5EPg@mail.gmail.com>

Id suggest you guys take a look at FS-9051

also note, that I found that I could work around this bug by explicitly
setting
<param name="dtmf-type" value="rfc2833"/> in the sip profile.

Jay

On 15 April 2016 at 13:53, Sergey Safarov <s.safarov at gmail.com> wrote:

> Could you please find commit where is bug has been introduced using "git
> bisect" commands.
>
> On Fri, Apr 15, 2016, 06:41 Dave Horton <daveh at beachdognet.com> wrote:
>
>> I?m running into a problem on v1.6.7 with something that seemed to work
>> properly on v1.6.6, and I am wondering if this is a known bug.
>>
>> I have an application that receives an incoming call and bridges it to an
>> outbound destination.  Where there is RFC 2833 offered on the A leg for
>> dtmf I want that to offer that on the B leg as well.
>> I have defined late-negotiation=true on my sip profile.
>>
>> In v1.6.6 (specifically, User agent header identifies as
>> 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit) this worked as expected.
>>
>> In v1.6.7 (specifically, User agent header identifies as
>> 1.6.7+git~20160401T134007Z~f0c3870be3~64bit) it doesn?t.
>> The SDP offered on the B leg does not include RFC 2833 even though the A
>> leg offered it.
>>
>> Is this a known bug?  If so, is there a patch?
>>
>> If not, I can provide more information (pastebin logs, etc) to help
>> troubleshoot.  Please let me know what would be most useful to see.
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

Jay
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From david.witham at netsip.com.au  Fri Apr 15 10:18:30 2016
From: david.witham at netsip.com.au (David Witham)
Date: Fri, 15 Apr 2016 16:18:30 +1000
Subject: [Freeswitch-users] rfc 2833 not offered on outbound leg in
 bridge scenario with FS 1.6.7 (worked properly in FS 1.6.6)
In-Reply-To: <CAHtxdDccOnAT-3QyE2SpUtRoGpMMDtQc-np1DR4-ksnZjZDefQ@mail.gmail.com>
References: <9F1A8D0A-19D3-4C6C-84D6-AA8FF4018CF4@beachdognet.com>
	<CAHtxdDccOnAT-3QyE2SpUtRoGpMMDtQc-np1DR4-ksnZjZDefQ@mail.gmail.com>
Message-ID: <CAPY30xbQqywCB0JbfZ+zEADVaxXkpE87QAEgFejOWt0McMMzLg@mail.gmail.com>

Dave,

Looks like this was fixed in FS-9049 so DTMF should work properly in master
and/or when 1.6.8 is released.

regards,
David

On 15 April 2016 at 13:53, Sergey Safarov <s.safarov at gmail.com> wrote:

> Could you please find commit where is bug has been introduced using "git
> bisect" commands.
>
> On Fri, Apr 15, 2016, 06:41 Dave Horton <daveh at beachdognet.com> wrote:
>
>> I?m running into a problem on v1.6.7 with something that seemed to work
>> properly on v1.6.6, and I am wondering if this is a known bug.
>>
>> I have an application that receives an incoming call and bridges it to an
>> outbound destination.  Where there is RFC 2833 offered on the A leg for
>> dtmf I want that to offer that on the B leg as well.
>> I have defined late-negotiation=true on my sip profile.
>>
>> In v1.6.6 (specifically, User agent header identifies as
>> 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit) this worked as expected.
>>
>> In v1.6.7 (specifically, User agent header identifies as
>> 1.6.7+git~20160401T134007Z~f0c3870be3~64bit) it doesn?t.
>> The SDP offered on the B leg does not include RFC 2833 even though the A
>> leg offered it.
>>
>> Is this a known bug?  If so, is there a patch?
>>
>> If not, I can provide more information (pastebin logs, etc) to help
>> troubleshoot.  Please let me know what would be most useful to see.
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Witham
Senior Voice/Systems Engineer

Netsip pty ltd ? An Over the Wire Company
Level 1, 24 Little Edward St, Spring Hill QLD 4000

t    +61 1300 638 747
e   david.witham at netsip.com.au    www.netsip.com.au
<https://macowa.netsip.com.au/owa/redir.aspx?C=95eec4d12cf0400796940d56513ed0f7&URL=http%3a%2f%2fwww.netsip.com.au>

<https://macowa.netsip.com.au/owa/redir.aspx?C=95eec4d12cf0400796940d56513ed0f7&URL=http%3a%2f%2fwww.netsip.com.au>
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From mustafin.aleksandr at gmail.com  Fri Apr 15 13:19:21 2016
From: mustafin.aleksandr at gmail.com (Alexander Mustafin)
Date: Fri, 15 Apr 2016 14:19:21 +0500
Subject: [Freeswitch-users] HTTPS in mod_shout
Message-ID: <DE3AA37F-8E45-47A4-AA58-5707445ABAE3@gmail.com>

Hi there!

Is it possible to use HTTPS URLs in mod_shout some way? I want to test TTS API and it require https. I?ve been tried to send request to 443 port, but it doesn?t work. 


Best regards,
Alexander Mustafin
mustafin.aleksandr at gmail.com




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From carlosj.gf at gmail.com  Fri Apr 15 13:44:09 2016
From: carlosj.gf at gmail.com (=?UTF-8?Q?Carlos_Gonz=C3=A1lez_Florido?=)
Date: Fri, 15 Apr 2016 11:44:09 +0200
Subject: [Freeswitch-users] Sending commands over Verto
In-Reply-To: <ACE60D4B-1606-4C5D-97B9-C6749320608E@jerris.com>
References: <CAFKjMAYBM2LgRyauAKs_vhesf633-ACa-Azn9G3Th7+G06uaWw@mail.gmail.com>
	<ACE60D4B-1606-4C5D-97B9-C6749320608E@jerris.com>
Message-ID: <CAFKjMAZ61SH54GsWt3ov913YosCUGZU4jvRFQe7Q2H+xnK4-Ag@mail.gmail.com>

I tried that without success...
Could you send me an example of use of that variables (are they boolean?
must list the methods to use?) and the json request for a command with
arguments? (Like 'show channels' or any other...).

Also, I think that since we are using  <param name="blind-reg"
value="true"> in verto's config it seems it is not hitting the directory at
all... is that correct?

Thank you,
Carlos
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From dcolombo at voismart.it  Fri Apr 15 16:07:05 2016
From: dcolombo at voismart.it (Davide Colombo)
Date: Fri, 15 Apr 2016 14:07:05 +0200 (CEST)
Subject: [Freeswitch-users] matrix-appservice-verto
Message-ID: <1225063259.48291.1460722025127.JavaMail.zimbra@voismart.it>

Hi all,

anyone knows matrix-appservice-verto, a verto bridge used to hook FreeSWITCH up to Matrix (an open standard for decentralised communication)?

It seems cool!


From nneul at mst.edu  Fri Apr 15 17:37:15 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Fri, 15 Apr 2016 08:37:15 -0500
Subject: [Freeswitch-users] Are there any known memory leaks with
 reloadxml and large XML configs?
In-Reply-To: <CAKbxfG9oi5SrT=-xWNzcGV=WQxZZqiABXySU-nnD6jhDmQ4QXA@mail.gmail.com>
References: <570FCA0B.3020308@mst.edu>
	<209D8D8A-7D31-4579-8E58-C15A2406EEDE@jerris.com>
	<570FD86C.5050307@mst.edu>
	<CAKbxfG8Je3Y_E9p4Xx3oTakReAVNKRW6kQ+=Kc45Eep+v+gLVg@mail.gmail.com>
	<570FE098.3090002@mst.edu> <570FEE8A.9060803@mst.edu>
	<CAKbxfG9oi5SrT=-xWNzcGV=WQxZZqiABXySU-nnD6jhDmQ4QXA@mail.gmail.com>
Message-ID: <CACtPGgWNR-gDY1iedCBCEaLwHieXL4NnDVbqZ6gOwwWzKHdezw@mail.gmail.com>

Thanks for the assistance. Your additional suggestion of where to look led
me right to an additional memory leak in mod_skinny - and was as a result
able to reproduce a scenario that could trigger it. I don't know for sure
that it fixes the symptom I've seen on production, but I suspect this is at
least a significant part of it. A fix has been pushed into master for it.

JIRA issue updated accordingly.

-- Nathan

On Thu, Apr 14, 2016 at 11:41 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Why are you still discussing this here when you have a jira open?  I am
> not sure how to properly explain the IMMENSE difficulty of working on bugs
> over email.  There is no right reason to side with email when trying to
> decide between jira and email.  If a jira turns out not to be a bug, one
> click to close it.  When an email turns out to be a bug, frick, now we have
> to copy all the data over.  Pass it on, file jiras when you have a
> problem.  We have broadcasted it here for years..:.....
>
> The xml registry is ref counted if it reloads while its open, it probably
> means something you use opens the xml root and does not close it.  I said
> this on your jira.
>
>
> On Thursday, April 14, 2016, Nathan Neulinger <nneul at mst.edu> wrote:
>
>> Running some more post analysis over the raw content of the core dump,
>> and extracting each parsed blob of XML, it looks
>> like almost all of them are complete copies (and not partial like if it
>> ran out of room and had to resize+relocate).
>>
>> I extracted all of the chunks to individual files, and looking at each
>> one in turn based on sequential position in
>> memory, I am seeing reasonably expected small/tiny changes from one to
>> the next that would correspond to incremental
>> changes in our configuration UI (stuff like individual changes to a line
>> key, button position, voicemail address,
>> password, etc.)
>>
>>
>> What in the system results in a full copy of the parsed XML structure?
>> Does every access do some sort of temporary copy
>> like you'd get with SQL and transaction isolation?
>>
>>
>> -- Nathan
>>
>> On 04/14/2016 01:25 PM, Nathan Neulinger wrote:
>> > FS-9074 opened.
>> >
>> > I've attempted to reproduce in lab environment, but not with much
>> success since I can't readily reproduce the
>> > activity/usage level from the production environment. On production
>> env, I can readily see the resident memory growing
>> > when I reloadxml on the active server, but it doesn't do it every time,
>> so it's clearly some sort of combination of
>> > events/timing related.
>> >
>> > On the lab setup, I typically see about 3 x duplicates of the strings
>> from the XML in a generated core.
>> >
>> > No modules are being reloaded during normal operations. Just a rewrite
>> of the configs followed by a reloadxml.
>> >
>> >
>> > As far as the commercial option - have asked y'all several times for
>> what our options would be for that sort of service,
>> > but have never gotten any clear answer particularly given the custom
>> deployment we have. Would be very interested in
>> > presenting that to management here if you can give me some proposed
>> service offerings and costs.
>> >
>> > -- Nathan
>> >
>> > On 04/14/2016 12:58 PM, Anthony Minessale wrote:
>> >> Have you done any debugging like running valgrind or ASAN on a server?
>> >> Did you try to reproduce it in a lab yet?
>> >>
>> >> We can't really do any leg work for you so we would need a more
>> specific reproduction case.
>> >> We do offer commercial options as you are probably already are aware
>> of where we could deploy consultants to do more of
>> >> the leg work for you.
>> >>
>> >> Do you only reloadxml or do you reload any modules.
>> >> Also you should ALWAYS file a jira.  We don't want to field issues on
>> the list.  Its not a problem to close NOT A BUG if
>> >> it turns out that way.
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> On Thu, Apr 14, 2016 at 12:50 PM, Nathan Neulinger <nneul at mst.edu
>> <mailto:nneul at mst.edu>> wrote:
>> >>
>> >>     The indication I'm reacting to of leak isn't "one in memory"....
>> The below is showing FIFTY THREE of them.
>> >>
>> >>     -- 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
>>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From amani.mansour2 at gmail.com  Fri Apr 15 18:15:20 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Fri, 15 Apr 2016 14:15:20 +0000
Subject: [Freeswitch-users] Register request problem
Message-ID: <CAFETSWig2AiJsRxZGp8kzbiG6=tPj516ESmiwXng5SnbRvcKiQ@mail.gmail.com>

Hi all ,

Please i need to know the file (directory,dialplan or ........) from which
the request register is send ,I need your help ,
i need to know the origine of the message Register to be able to modify it
i need to create a scenario which is:
FS receive Register
FS send 401
FS receive  Register with authorization header
FS send second 401 with stale = false

please i need your help .
thank you again

with best regards
amani
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From mgg at giagnocavo.net  Fri Apr 15 18:19:12 2016
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 15 Apr 2016 14:19:12 +0000
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <CAHtxdDdb+ULhYRF3shuMDcnvHr1woY7sgD8LxeV1P=GxP9OrNA@mail.gmail.com>
References: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
	<CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>
	<CAMtXxQnyeqq8aHNEk8f_EsvraiPrtpF0se91v8dOJx6ehhKwbg@mail.gmail.com>
	<5123CC15-8A19-453E-BB13-A8B230BC3DFC@jerris.com>
	<CAMtXxQ=8Q7iyzx2JvZ=o-nwBnyaxNhkF0rC2rp1cJqt0t4w9gA@mail.gmail.com>
	<EC4896F1-829D-4C95-B7E5-AB9BD66021A1@freeswitch.org>
	<CAHtxdDdb+ULhYRF3shuMDcnvHr1woY7sgD8LxeV1P=GxP9OrNA@mail.gmail.com>
Message-ID: <DM2PR07MB7023026A02BD78E92821B11D4680@DM2PR07MB702.namprd07.prod.outlook.com>

One easy way, PCAP your HTTP. You can setup a rotating set of files, then when you have an issue, break the capture and hunt for the failed POST. That should give you "ground truth" as to what's going on and may point you to the issue.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
Sent: Wednesday, 13 April, 2016 22:11
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Billsec in dialplan

Please also check mod_format

On Thu, Apr 14, 2016, 02:16 Ken Rice <krice at freeswitch.org> wrote:
The correct answer here is you must check more often i would say at a minimum every 15 seconds. Search the list for me discussing cdrs in volume in the past or contact me off list if you need pro help

Sent from my iPhone

On Apr 13, 2016, at 5:52 PM, Joel Serrano <joel at gogii.net> wrote:
Hi Michael,?

The problem is that occasionally we see the following in FS log:

2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:385 Got error [0] posting to web server [http://cdrs.example.com/queue]
2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:392 Retry will be with url [http://cdrs.example.com/queue]

When that happens, the CDR gets written to disc (so the CDR is not processed on time, thus our user is not billed for that call yet).

We have a task that checks every 5 minutes for failed CDRs and we resubmit them to the processing queue, but I need to find out why FS is failing.

I have checked the access logs on the receiving side, but all requests return a 200 OK.

Now I have enabled logs in the loadbalancer between FS and the queues to see if I get more vision from there...


A workaround was to make API requests directly from the dialplan with the billsec info, but as all suggest going with CDRs, I am now focussed on finding the root cause for that error.

Any help/suggestion is more than welcome.


Thank you.

Best regards,?
Joel.




On Tue, Apr 12, 2016 at 1:42 PM, Michael Jerris <mike at jerris.com> wrote:
I'm probably going to push towards cdr, but what are you trying to actually do with that info?

On Apr 12, 2016, at 2:39 PM, Joel Serrano <joel at gogii.net> wrote:

Hi Michael,?

I want to call an API basically with the following info:

sip_from_uri
destination_number
billsec

I found away using "api_hangup_hook=lua script.lua" + "session_in_hangup_hook=true", but, I don't like it... I would prefer using curl from within the dialplan.

Can you give me your suggestion? I have also asked in IRC and using the CDRs looks like the best way to go, so we will probably follow that, but I would still like to now if it is possible and how.

Thanks!
Joel.



On Tue, Apr 12, 2016 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:
Billsec is set in reporting state, after you are out of dial plan. ? What are you trying to do based on that, I can suggest a few approaches


On Monday, April 11, 2016, Joel Serrano <joel at gogii.net> wrote:
Hi,?

Is it possible to access ${billsec} in the dialplan after the bridge application (having hangup_after_bridge=false)??

I want to log the billsec but currently it is always empty.

I have a workaround in a test server using api_hangup_hook to call a lua script with session_in_hangup_hook=true, and inside the lua script I can access the billsec variable and thus write it to the logs.

Is there no way to do this within the XML dialplan?

Thanks!

Best regards,?
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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 Shawn.Wheeler at interlockconcepts.com  Fri Apr 15 18:45:33 2016
From: Shawn.Wheeler at interlockconcepts.com (Shawn Wheeler)
Date: Fri, 15 Apr 2016 14:45:33 +0000
Subject: [Freeswitch-users] FXO and FXS Devices
Message-ID: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>

What types of FXO and FXS devices does FreeSWITCH use?

Thank you

Shawn



From krice at freeswitch.org  Fri Apr 15 18:50:52 2016
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 15 Apr 2016 09:50:52 -0500
Subject: [Freeswitch-users] FXO and FXS Devices
In-Reply-To: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
References: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
Message-ID: <aa8401d19726$35abfbb0$a103f310$@freeswitch.org>

FreeSWITCH can use Sangoma FXO/FXS  PCI(e) hardware but I recommend using an
external gateway, how many ports are you going to need?

K

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shawn
Wheeler
Sent: Friday, April 15, 2016 9:46 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] FXO and FXS Devices

What types of FXO and FXS devices does FreeSWITCH use?

Thank you

Shawn


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at 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 bobjectsfreeswitch at gmail.com  Fri Apr 15 19:20:06 2016
From: bobjectsfreeswitch at gmail.com (Bob Hartwig)
Date: Fri, 15 Apr 2016 10:20:06 -0500
Subject: [Freeswitch-users] FXO and FXS Devices
In-Reply-To: <aa8401d19726$35abfbb0$a103f310$@freeswitch.org>
References: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<aa8401d19726$35abfbb0$a103f310$@freeswitch.org>
Message-ID: <CAAemr3xLAqzDus1WXyOR_n2vGmcMtqr_e=-uN1BbVVRywOcSuA@mail.gmail.com>

Hey Ken, why the recommendation to use an external gateway instead of
FreeTDM / Sangoma / Dahdi-compatible boards?



On Fri, Apr 15, 2016 at 9:50 AM, Ken Rice <krice at freeswitch.org> wrote:

> FreeSWITCH can use Sangoma FXO/FXS  PCI(e) hardware but I recommend using
> an
> external gateway, how many ports are you going to need?
>
> K
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shawn
> Wheeler
> Sent: Friday, April 15, 2016 9:46 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] FXO and FXS Devices
>
> What types of FXO and FXS devices does FreeSWITCH use?
>
> Thank you
>
> Shawn
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 15 19:20:20 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Apr 2016 11:20:20 -0400
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <DM2PR07MB7023026A02BD78E92821B11D4680@DM2PR07MB702.namprd07.prod.outlook.com>
References: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
	<CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>
	<CAMtXxQnyeqq8aHNEk8f_EsvraiPrtpF0se91v8dOJx6ehhKwbg@mail.gmail.com>
	<5123CC15-8A19-453E-BB13-A8B230BC3DFC@jerris.com>
	<CAMtXxQ=8Q7iyzx2JvZ=o-nwBnyaxNhkF0rC2rp1cJqt0t4w9gA@mail.gmail.com>
	<EC4896F1-829D-4C95-B7E5-AB9BD66021A1@freeswitch.org>
	<CAHtxdDdb+ULhYRF3shuMDcnvHr1woY7sgD8LxeV1P=GxP9OrNA@mail.gmail.com>
	<DM2PR07MB7023026A02BD78E92821B11D4680@DM2PR07MB702.namprd07.prod.outlook.com>
Message-ID: <FA15B116-A7B8-431D-8F26-95A69A09F78C@jerris.com>

Most likely those failures are due to not enough workers on your web server, but yes, the pcap will help find it.

> On Apr 15, 2016, at 10:19 AM, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
> 
> One easy way, PCAP your HTTP. You can setup a rotating set of files, then when you have an issue, break the capture and hunt for the failed POST. That should give you "ground truth" as to what's going on and may point you to the issue.
> 
> -Michael
> 
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
> Sent: Wednesday, 13 April, 2016 22:11
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Billsec in dialplan
> 
> Please also check mod_format
> 
> On Thu, Apr 14, 2016, 02:16 Ken Rice <krice at freeswitch.org> wrote:
> The correct answer here is you must check more often i would say at a minimum every 15 seconds. Search the list for me discussing cdrs in volume in the past or contact me off list if you need pro help
> 
> Sent from my iPhone
> 
> On Apr 13, 2016, at 5:52 PM, Joel Serrano <joel at gogii.net> wrote:
> Hi Michael, 
> 
> The problem is that occasionally we see the following in FS log:
> 
> 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:385 Got error [0] posting to web server [http://cdrs.example.com/queue]
> 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:392 Retry will be with url [http://cdrs.example.com/queue]
> 
> When that happens, the CDR gets written to disc (so the CDR is not processed on time, thus our user is not billed for that call yet).
> 
> We have a task that checks every 5 minutes for failed CDRs and we resubmit them to the processing queue, but I need to find out why FS is failing.
> 
> I have checked the access logs on the receiving side, but all requests return a 200 OK.
> 
> Now I have enabled logs in the loadbalancer between FS and the queues to see if I get more vision from there...
> 
> 
> A workaround was to make API requests directly from the dialplan with the billsec info, but as all suggest going with CDRs, I am now focussed on finding the root cause for that error.
> 
> Any help/suggestion is more than welcome.
> 
> 
> Thank you.
> 
> Best regards, 
> Joel.
> 
> 
> 
> 
> On Tue, Apr 12, 2016 at 1:42 PM, Michael Jerris <mike at jerris.com> wrote:
> I'm probably going to push towards cdr, but what are you trying to actually do with that info?
> 
> On Apr 12, 2016, at 2:39 PM, Joel Serrano <joel at gogii.net> wrote:
> 
> Hi Michael, 
> 
> I want to call an API basically with the following info:
> 
> sip_from_uri
> destination_number
> billsec
> 
> I found away using "api_hangup_hook=lua script.lua" + "session_in_hangup_hook=true", but, I don't like it... I would prefer using curl from within the dialplan.
> 
> Can you give me your suggestion? I have also asked in IRC and using the CDRs looks like the best way to go, so we will probably follow that, but I would still like to now if it is possible and how.
> 
> Thanks!
> Joel.
> 
> 
> 
> On Tue, Apr 12, 2016 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:
> Billsec is set in reporting state, after you are out of dial plan.   What are you trying to do based on that, I can suggest a few approaches
> 
> 
> On Monday, April 11, 2016, Joel Serrano <joel at gogii.net> wrote:
> Hi, 
> 
> Is it possible to access ${billsec} in the dialplan after the bridge application (having hangup_after_bridge=false)??
> 
> I want to log the billsec but currently it is always empty.
> 
> I have a workaround in a test server using api_hangup_hook to call a lua script with session_in_hangup_hook=true, and inside the lua script I can access the billsec variable and thus write it to the logs.
> 
> Is there no way to do this within the XML dialplan?
> 
> Thanks!
> 
> Best regards, 
> Joel.
> 
> 



From Shawn.Wheeler at interlockconcepts.com  Fri Apr 15 19:23:48 2016
From: Shawn.Wheeler at interlockconcepts.com (Shawn Wheeler)
Date: Fri, 15 Apr 2016 15:23:48 +0000
Subject: [Freeswitch-users] FXO and FXS Devices
In-Reply-To: <aa8401d19726$35abfbb0$a103f310$@freeswitch.org>
References: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<aa8401d19726$35abfbb0$a103f310$@freeswitch.org>
Message-ID: <SN1PR0201MB1504F0AAC4A7C171CC7A0183E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>

Two at the most.  The machine doesn't have PCI slots to something USB would be best.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Friday, April 15, 2016 7:51 AM
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FXO and FXS Devices

FreeSWITCH can use Sangoma FXO/FXS  PCI(e) hardware but I recommend using an external gateway, how many ports are you going to need?

K

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shawn Wheeler
Sent: Friday, April 15, 2016 9:46 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] FXO and FXS Devices

What types of FXO and FXS devices does FreeSWITCH use?

Thank you

Shawn


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

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

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


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

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

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


From krice at freeswitch.org  Fri Apr 15 19:29:54 2016
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 15 Apr 2016 10:29:54 -0500
Subject: [Freeswitch-users] FXO and FXS Devices
In-Reply-To: <CAAemr3xLAqzDus1WXyOR_n2vGmcMtqr_e=-uN1BbVVRywOcSuA@mail.gmail.com>
References: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>	<aa8401d19726$35abfbb0$a103f310$@freeswitch.org>
	<CAAemr3xLAqzDus1WXyOR_n2vGmcMtqr_e=-uN1BbVVRywOcSuA@mail.gmail.com>
Message-ID: <aadf01d1972b$a9d87c20$fd897460$@freeswitch.org>

The cost of the boards vs the cost of an external Ethernet gateway are negligible, however, with the gateways you get hardware codecs and if the server crashes you can have the hardware gateways immediate send calls over to the secondary server.

 

As for USB, I don?t know of anything USB that really works for FXO and FXS.

 

Companies like Adtran TA900 series and Sangoma Vega series seem to be very reliable, the later can be a little bit of a pain to configure

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob Hartwig
Sent: Friday, April 15, 2016 10:20 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FXO and FXS Devices

 

Hey Ken, why the recommendation to use an external gateway instead of FreeTDM / Sangoma / Dahdi-compatible boards?

 

 

 

On Fri, Apr 15, 2016 at 9:50 AM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> > wrote:

FreeSWITCH can use Sangoma FXO/FXS  PCI(e) hardware but I recommend using an
external gateway, how many ports are you going to need?

K


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> 
[mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Shawn
Wheeler
Sent: Friday, April 15, 2016 9:46 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: [Freeswitch-users] FXO and FXS Devices

What types of FXO and FXS devices does FreeSWITCH use?

Thank you

Shawn


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

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

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


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

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

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

 

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From mike at jerris.com  Fri Apr 15 19:32:03 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Apr 2016 11:32:03 -0400
Subject: [Freeswitch-users] FXO and FXS Devices
In-Reply-To: <SN1PR0201MB1504F0AAC4A7C171CC7A0183E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
References: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<aa8401d19726$35abfbb0$a103f310$@freeswitch.org>
	<SN1PR0201MB1504F0AAC4A7C171CC7A0183E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
Message-ID: <CDA04177-CD6F-4C9C-AD31-F852BAFE99A3@jerris.com>

I've yet to find a USB device I'm happy with.  I would avoid them. External gateways are relatively inexpensive and make your implementation much simpler.

> On Apr 15, 2016, at 11:23 AM, Shawn Wheeler <Shawn.Wheeler at interlockconcepts.com> wrote:
> 
> Two at the most.  The machine doesn't have PCI slots to something USB would be best.
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
> Sent: Friday, April 15, 2016 7:51 AM
> To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FXO and FXS Devices
> 
> FreeSWITCH can use Sangoma FXO/FXS  PCI(e) hardware but I recommend using an external gateway, how many ports are you going to need?
> 
> K
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shawn Wheeler
> Sent: Friday, April 15, 2016 9:46 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] FXO and FXS Devices
> 
> What types of FXO and FXS devices does FreeSWITCH use?
> 
> Thank you
> 
> Shawn
> 



From mgg at giagnocavo.net  Fri Apr 15 19:40:51 2016
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Fri, 15 Apr 2016 15:40:51 +0000
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <FA15B116-A7B8-431D-8F26-95A69A09F78C@jerris.com>
References: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
	<CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>
	<CAMtXxQnyeqq8aHNEk8f_EsvraiPrtpF0se91v8dOJx6ehhKwbg@mail.gmail.com>
	<5123CC15-8A19-453E-BB13-A8B230BC3DFC@jerris.com>
	<CAMtXxQ=8Q7iyzx2JvZ=o-nwBnyaxNhkF0rC2rp1cJqt0t4w9gA@mail.gmail.com>
	<EC4896F1-829D-4C95-B7E5-AB9BD66021A1@freeswitch.org>
	<CAHtxdDdb+ULhYRF3shuMDcnvHr1woY7sgD8LxeV1P=GxP9OrNA@mail.gmail.com>
	<DM2PR07MB7023026A02BD78E92821B11D4680@DM2PR07MB702.namprd07.prod.outlook.com>
	<FA15B116-A7B8-431D-8F26-95A69A09F78C@jerris.com>
Message-ID: <DM2PR07MB70260E83D6D9D25A998FC15D4680@DM2PR07MB702.namprd07.prod.outlook.com>

Also points to that people should, generally, be using a queuing solution. As much as RabbitMQ has a bad rep, I find that writing to disk then pushing them into a local RabbitMQ instance provides the nicest level of auto-flow-control. You can count on local being up, and CDRs will backlog and flow as the network can handle them. Seems easier to me than having to handle multiple failure modes (http, fail to disk, etc.) Also I think RabbitMQ can do transactional stuff, so you are less likely to run into a scenario where you get an HTTP 200 OK but processing didn't actually happen. Just my repeated 2 cents.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Friday, 15 April, 2016 9:20
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Billsec in dialplan

Most likely those failures are due to not enough workers on your web server, but yes, the pcap will help find it.

> On Apr 15, 2016, at 10:19 AM, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
> 
> One easy way, PCAP your HTTP. You can setup a rotating set of files, then when you have an issue, break the capture and hunt for the failed POST. That should give you "ground truth" as to what's going on and may point you to the issue.
> 
> -Michael
> 
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
> Sent: Wednesday, 13 April, 2016 22:11
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Billsec in dialplan
> 
> Please also check mod_format
> 
> On Thu, Apr 14, 2016, 02:16 Ken Rice <krice at freeswitch.org> wrote:
> The correct answer here is you must check more often i would say at a minimum every 15 seconds. Search the list for me discussing cdrs in volume in the past or contact me off list if you need pro help
> 
> Sent from my iPhone
> 
> On Apr 13, 2016, at 5:52 PM, Joel Serrano <joel at gogii.net> wrote:
> Hi Michael, 
> 
> The problem is that occasionally we see the following in FS log:
> 
> 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:385 Got error [0] posting to web server [http://cdrs.example.com/queue]
> 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:392 Retry will be with url [http://cdrs.example.com/queue]
> 
> When that happens, the CDR gets written to disc (so the CDR is not processed on time, thus our user is not billed for that call yet).
> 
> We have a task that checks every 5 minutes for failed CDRs and we resubmit them to the processing queue, but I need to find out why FS is failing.
> 
> I have checked the access logs on the receiving side, but all requests return a 200 OK.
> 
> Now I have enabled logs in the loadbalancer between FS and the queues to see if I get more vision from there...
> 
> 
> A workaround was to make API requests directly from the dialplan with the billsec info, but as all suggest going with CDRs, I am now focussed on finding the root cause for that error.
> 
> Any help/suggestion is more than welcome.
> 
> 
> Thank you.
> 
> Best regards, 
> Joel.
> 
> 
> 
> 
> On Tue, Apr 12, 2016 at 1:42 PM, Michael Jerris <mike at jerris.com> wrote:
> I'm probably going to push towards cdr, but what are you trying to actually do with that info?
> 
> On Apr 12, 2016, at 2:39 PM, Joel Serrano <joel at gogii.net> wrote:
> 
> Hi Michael, 
> 
> I want to call an API basically with the following info:
> 
> sip_from_uri
> destination_number
> billsec
> 
> I found away using "api_hangup_hook=lua script.lua" + "session_in_hangup_hook=true", but, I don't like it... I would prefer using curl from within the dialplan.
> 
> Can you give me your suggestion? I have also asked in IRC and using the CDRs looks like the best way to go, so we will probably follow that, but I would still like to now if it is possible and how.
> 
> Thanks!
> Joel.
> 
> 
> 
> On Tue, Apr 12, 2016 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:
> Billsec is set in reporting state, after you are out of dial plan.   What are you trying to do based on that, I can suggest a few approaches
> 
> 
> On Monday, April 11, 2016, Joel Serrano <joel at gogii.net> wrote:
> Hi, 
> 
> Is it possible to access ${billsec} in the dialplan after the bridge application (having hangup_after_bridge=false)??
> 
> I want to log the billsec but currently it is always empty.
> 
> I have a workaround in a test server using api_hangup_hook to call a lua script with session_in_hangup_hook=true, and inside the lua script I can access the billsec variable and thus write it to the logs.
> 
> Is there no way to do this within the XML dialplan?
> 
> Thanks!
> 
> Best regards, 
> 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


From fdelawarde at wirelessmundi.com  Fri Apr 15 19:43:59 2016
From: fdelawarde at wirelessmundi.com (=?UTF-8?B?RnJhbsOnb2lz?=)
Date: Fri, 15 Apr 2016 17:43:59 +0200
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <DM2PR07MB7023026A02BD78E92821B11D4680@DM2PR07MB702.namprd07.prod.outlook.com>
References: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
	<CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>
	<CAMtXxQnyeqq8aHNEk8f_EsvraiPrtpF0se91v8dOJx6ehhKwbg@mail.gmail.com>
	<5123CC15-8A19-453E-BB13-A8B230BC3DFC@jerris.com>
	<CAMtXxQ=8Q7iyzx2JvZ=o-nwBnyaxNhkF0rC2rp1cJqt0t4w9gA@mail.gmail.com>
	<EC4896F1-829D-4C95-B7E5-AB9BD66021A1@freeswitch.org>
	<CAHtxdDdb+ULhYRF3shuMDcnvHr1woY7sgD8LxeV1P=GxP9OrNA@mail.gmail.com>
	<DM2PR07MB7023026A02BD78E92821B11D4680@DM2PR07MB702.namprd07.prod.outlook.com>
Message-ID: <57110C3F.2040708@wirelessmundi.com>

Watch out with billsec is the integer part of the "real" billing seconds.

So if a call has duration of 10.9 seconds, billsec will be 10 seconds, 
but your provider will bill you 11 seconds.

One should use billusec!

F

On 2016-04-15 16:19, Michael Giagnocavo wrote:
> One easy way, PCAP your HTTP. You can setup a rotating set of files, then when you have an issue, break the capture and hunt for the failed POST. That should give you "ground truth" as to what's going on and may point you to the issue.
>
> -Michael
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
> Sent: Wednesday, 13 April, 2016 22:11
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Billsec in dialplan
>
> Please also check mod_format
>
> On Thu, Apr 14, 2016, 02:16 Ken Rice <krice at freeswitch.org> wrote:
> The correct answer here is you must check more often i would say at a minimum every 15 seconds. Search the list for me discussing cdrs in volume in the past or contact me off list if you need pro help
>
> Sent from my iPhone
>
> On Apr 13, 2016, at 5:52 PM, Joel Serrano <joel at gogii.net> wrote:
> Hi Michael,
>
> The problem is that occasionally we see the following in FS log:
>
> 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:385 Got error [0] posting to web server [http://cdrs.example.com/queue]
> 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:392 Retry will be with url [http://cdrs.example.com/queue]
>
> When that happens, the CDR gets written to disc (so the CDR is not processed on time, thus our user is not billed for that call yet).
>
> We have a task that checks every 5 minutes for failed CDRs and we resubmit them to the processing queue, but I need to find out why FS is failing.
>
> I have checked the access logs on the receiving side, but all requests return a 200 OK.
>
> Now I have enabled logs in the loadbalancer between FS and the queues to see if I get more vision from there...
>
>
> A workaround was to make API requests directly from the dialplan with the billsec info, but as all suggest going with CDRs, I am now focussed on finding the root cause for that error.
>
> Any help/suggestion is more than welcome.
>
>
> Thank you.
>
> Best regards,
> Joel.
>
>
>
>
> On Tue, Apr 12, 2016 at 1:42 PM, Michael Jerris <mike at jerris.com> wrote:
> I'm probably going to push towards cdr, but what are you trying to actually do with that info?
>
> On Apr 12, 2016, at 2:39 PM, Joel Serrano <joel at gogii.net> wrote:
>
> Hi Michael,
>
> I want to call an API basically with the following info:
>
> sip_from_uri
> destination_number
> billsec
>
> I found away using "api_hangup_hook=lua script.lua" + "session_in_hangup_hook=true", but, I don't like it... I would prefer using curl from within the dialplan.
>
> Can you give me your suggestion? I have also asked in IRC and using the CDRs looks like the best way to go, so we will probably follow that, but I would still like to now if it is possible and how.
>
> Thanks!
> Joel.
>
>
>
> On Tue, Apr 12, 2016 at 12:22 PM, Michael Jerris <mike at jerris.com> wrote:
> Billsec is set in reporting state, after you are out of dial plan.   What are you trying to do based on that, I can suggest a few approaches
>
>
> On Monday, April 11, 2016, Joel Serrano <joel at gogii.net> wrote:
> Hi,
>
> Is it possible to access ${billsec} in the dialplan after the bridge application (having hangup_after_bridge=false)??
>
> I want to log the billsec but currently it is always empty.
>
> I have a workaround in a test server using api_hangup_hook to call a lua script with session_in_hangup_hook=true, and inside the lua script I can access the billsec variable and thus write it to the logs.
>
> Is there no way to do this within the XML dialplan?
>
> Thanks!
>
> Best regards,
> 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Shawn.Wheeler at interlockconcepts.com  Fri Apr 15 20:07:58 2016
From: Shawn.Wheeler at interlockconcepts.com (Shawn Wheeler)
Date: Fri, 15 Apr 2016 16:07:58 +0000
Subject: [Freeswitch-users] FXO and FXS Devices
In-Reply-To: <aadf01d1972b$a9d87c20$fd897460$@freeswitch.org>
References: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<aa8401d19726$35abfbb0$a103f310$@freeswitch.org>
	<CAAemr3xLAqzDus1WXyOR_n2vGmcMtqr_e=-uN1BbVVRywOcSuA@mail.gmail.com>
	<aadf01d1972b$a9d87c20$fd897460$@freeswitch.org>
Message-ID: <SN1PR0201MB150417DFF44C56BC59973234E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>

How about something like a Linksys will any of those work?


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Friday, April 15, 2016 8:30 AM
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FXO and FXS Devices

The cost of the boards vs the cost of an external Ethernet gateway are negligible, however, with the gateways you get hardware codecs and if the server crashes you can have the hardware gateways immediate send calls over to the secondary server.

As for USB, I don?t know of anything USB that really works for FXO and FXS.

Companies like Adtran TA900 series and Sangoma Vega series seem to be very reliable, the later can be a little bit of a pain to configure


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob Hartwig
Sent: Friday, April 15, 2016 10:20 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] FXO and FXS Devices

Hey Ken, why the recommendation to use an external gateway instead of FreeTDM / Sangoma / Dahdi-compatible boards?



On Fri, Apr 15, 2016 at 9:50 AM, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:
FreeSWITCH can use Sangoma FXO/FXS  PCI(e) hardware but I recommend using an
external gateway, how many ports are you going to need?

K

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>
[mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Shawn
Wheeler
Sent: Friday, April 15, 2016 9:46 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] FXO and FXS Devices

What types of FXO and FXS devices does FreeSWITCH use?

Thank you

Shawn


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

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

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


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

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

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

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From mike at jerris.com  Fri Apr 15 21:36:41 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Apr 2016 13:36:41 -0400
Subject: [Freeswitch-users] FXO and FXS Devices
In-Reply-To: <SN1PR0201MB150417DFF44C56BC59973234E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
References: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<aa8401d19726$35abfbb0$a103f310$@freeswitch.org>
	<CAAemr3xLAqzDus1WXyOR_n2vGmcMtqr_e=-uN1BbVVRywOcSuA@mail.gmail.com>
	<aadf01d1972b$a9d87c20$fd897460$@freeswitch.org>
	<SN1PR0201MB150417DFF44C56BC59973234E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
Message-ID: <634B6B04-9E1D-4D8F-96C7-7D1616FDC306@jerris.com>

sure, just make sure the device interops successfully with your solution.


> On Apr 15, 2016, at 12:07 PM, Shawn Wheeler <Shawn.Wheeler at interlockconcepts.com> wrote:
> 
> How about something like a Linksys will any of those work?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
> Sent: Friday, April 15, 2016 8:30 AM
> To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FXO and FXS Devices
>  
> The cost of the boards vs the cost of an external Ethernet gateway are negligible, however, with the gateways you get hardware codecs and if the server crashes you can have the hardware gateways immediate send calls over to the secondary server.
>  
> As for USB, I don?t know of anything USB that really works for FXO and FXS.
>  
> Companies like Adtran TA900 series and Sangoma Vega series seem to be very reliable, the later can be a little bit of a pain to configure
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Bob Hartwig
> Sent: Friday, April 15, 2016 10:20 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] FXO and FXS Devices
>  
> Hey Ken, why the recommendation to use an external gateway instead of FreeTDM / Sangoma / Dahdi-compatible boards?
>  
>  
>  
> On Fri, Apr 15, 2016 at 9:50 AM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
> FreeSWITCH can use Sangoma FXO/FXS  PCI(e) hardware but I recommend using an
> external gateway, how many ports are you going to need?
> 
> K
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Shawn
> Wheeler
> Sent: Friday, April 15, 2016 9:46 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: [Freeswitch-users] FXO and FXS Devices
> 
> What types of FXO and FXS devices does FreeSWITCH use?
> 
> Thank you
> 
> Shawn
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From joel at gogii.net  Fri Apr 15 21:42:32 2016
From: joel at gogii.net (Joel Serrano)
Date: Fri, 15 Apr 2016 10:42:32 -0700
Subject: [Freeswitch-users] Billsec in dialplan
In-Reply-To: <57110C3F.2040708@wirelessmundi.com>
References: <CAMtXxQky_tGX7qfOx2kJr7DnjOHaQ+bR=qAR9a8keqtX0-Mfnw@mail.gmail.com>
	<CAEwTsAhzixoKCC7EDxnV3tmT0uHX6J2CDe9X7M82P3tULNKmxg@mail.gmail.com>
	<CAMtXxQnyeqq8aHNEk8f_EsvraiPrtpF0se91v8dOJx6ehhKwbg@mail.gmail.com>
	<5123CC15-8A19-453E-BB13-A8B230BC3DFC@jerris.com>
	<CAMtXxQ=8Q7iyzx2JvZ=o-nwBnyaxNhkF0rC2rp1cJqt0t4w9gA@mail.gmail.com>
	<EC4896F1-829D-4C95-B7E5-AB9BD66021A1@freeswitch.org>
	<CAHtxdDdb+ULhYRF3shuMDcnvHr1woY7sgD8LxeV1P=GxP9OrNA@mail.gmail.com>
	<DM2PR07MB7023026A02BD78E92821B11D4680@DM2PR07MB702.namprd07.prod.outlook.com>
	<57110C3F.2040708@wirelessmundi.com>
Message-ID: <CAMtXxQkd_CL9hEHmhH=hG4z5S7HUYy4Pq_LKjrWetewqX8855A@mail.gmail.com>

Thank you all for your replies.

1- Definitely going to change from billsec to billusec
2- Going to setup tcpdump to capture traffic to see what it has to say when
it fails

Currently, our FS servers are doing a POST request to 2 ActiveMQ's setup
with a load balancer in front of them.

Will update when I have more info.

Thanks!
Joel.


On Fri, Apr 15, 2016 at 8:43 AM, Fran?ois <fdelawarde at wirelessmundi.com>
wrote:

> Watch out with billsec is the integer part of the "real" billing seconds.
>
> So if a call has duration of 10.9 seconds, billsec will be 10 seconds,
> but your provider will bill you 11 seconds.
>
> One should use billusec!
>
> F
>
> On 2016-04-15 16:19, Michael Giagnocavo wrote:
> > One easy way, PCAP your HTTP. You can setup a rotating set of files,
> then when you have an issue, break the capture and hunt for the failed
> POST. That should give you "ground truth" as to what's going on and may
> point you to the issue.
> >
> > -Michael
> >
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
> > Sent: Wednesday, 13 April, 2016 22:11
> > To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] Billsec in dialplan
> >
> > Please also check mod_format
> >
> > On Thu, Apr 14, 2016, 02:16 Ken Rice <krice at freeswitch.org> wrote:
> > The correct answer here is you must check more often i would say at a
> minimum every 15 seconds. Search the list for me discussing cdrs in volume
> in the past or contact me off list if you need pro help
> >
> > Sent from my iPhone
> >
> > On Apr 13, 2016, at 5:52 PM, Joel Serrano <joel at gogii.net> wrote:
> > Hi Michael,
> >
> > The problem is that occasionally we see the following in FS log:
> >
> > 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:385 Got error [0] posting
> to web server [http://cdrs.example.com/queue]
> > 2016-04-13 02:33:11.143230 [ERR] mod_xml_cdr.c:392 Retry will be with
> url [http://cdrs.example.com/queue]
> >
> > When that happens, the CDR gets written to disc (so the CDR is not
> processed on time, thus our user is not billed for that call yet).
> >
> > We have a task that checks every 5 minutes for failed CDRs and we
> resubmit them to the processing queue, but I need to find out why FS is
> failing.
> >
> > I have checked the access logs on the receiving side, but all requests
> return a 200 OK.
> >
> > Now I have enabled logs in the loadbalancer between FS and the queues to
> see if I get more vision from there...
> >
> >
> > A workaround was to make API requests directly from the dialplan with
> the billsec info, but as all suggest going with CDRs, I am now focussed on
> finding the root cause for that error.
> >
> > Any help/suggestion is more than welcome.
> >
> >
> > Thank you.
> >
> > Best regards,
> > Joel.
> >
> >
> >
> >
> > On Tue, Apr 12, 2016 at 1:42 PM, Michael Jerris <mike at jerris.com> wrote:
> > I'm probably going to push towards cdr, but what are you trying to
> actually do with that info?
> >
> > On Apr 12, 2016, at 2:39 PM, Joel Serrano <joel at gogii.net> wrote:
> >
> > Hi Michael,
> >
> > I want to call an API basically with the following info:
> >
> > sip_from_uri
> > destination_number
> > billsec
> >
> > I found away using "api_hangup_hook=lua script.lua" +
> "session_in_hangup_hook=true", but, I don't like it... I would prefer using
> curl from within the dialplan.
> >
> > Can you give me your suggestion? I have also asked in IRC and using the
> CDRs looks like the best way to go, so we will probably follow that, but I
> would still like to now if it is possible and how.
> >
> > Thanks!
> > Joel.
> >
> >
> >
> > On Tue, Apr 12, 2016 at 12:22 PM, Michael Jerris <mike at jerris.com>
> wrote:
> > Billsec is set in reporting state, after you are out of dial plan.
>  What are you trying to do based on that, I can suggest a few approaches
> >
> >
> > On Monday, April 11, 2016, Joel Serrano <joel at gogii.net> wrote:
> > Hi,
> >
> > Is it possible to access ${billsec} in the dialplan after the bridge
> application (having hangup_after_bridge=false)??
> >
> > I want to log the billsec but currently it is always empty.
> >
> > I have a workaround in a test server using api_hangup_hook to call a lua
> script with session_in_hangup_hook=true, and inside the lua script I can
> access the billsec variable and thus write it to the logs.
> >
> > Is there no way to do this within the XML dialplan?
> >
> > Thanks!
> >
> > Best regards,
> > 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
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Apr 15 21:55:59 2016
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Apr 2016 10:55:59 -0700
Subject: [Freeswitch-users] Register request problem
In-Reply-To: <CAFETSWig2AiJsRxZGp8kzbiG6=tPj516ESmiwXng5SnbRvcKiQ@mail.gmail.com>
References: <CAFETSWig2AiJsRxZGp8kzbiG6=tPj516ESmiwXng5SnbRvcKiQ@mail.gmail.com>
Message-ID: <CAKzWOxVVb3SewDbC=NtO8gLdOjG8RS_OCeZz_i_McPHn+H-zew@mail.gmail.com>

This sounds like a job better suited for a SIP proxy like OpenSIPS or
Kamailio. I suspect that this task is either difficult, impractical, or
impossible with FreeSWITCH. (FreeSWITCH is a B2BUA.)

That being said, if anyone knows how to manipulate SIP registrations at
such granular level in Sofia I would be very curious to hear about it.

-Michael

On Fri, Apr 15, 2016 at 7:15 AM, amani mansour <amani.mansour2 at gmail.com>
wrote:

> Hi all ,
>
> Please i need to know the file (directory,dialplan or ........) from which
> the request register is send ,I need your help ,
> i need to know the origine of the message Register to be able to modify it
> i need to create a scenario which is:
> FS receive Register
> FS send 401
> FS receive  Register with authorization header
> FS send second 401 with stale = false
>
> please i need your help .
> thank you again
>
> with best regards
> amani
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From igorolhovskiy at gmail.com  Fri Apr 15 23:38:17 2016
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Fri, 15 Apr 2016 22:38:17 +0300
Subject: [Freeswitch-users] Asterisk-like hints
Message-ID: <CAJTkRNuDWbKDayTyAzmzHxCkq9WBFgMQoAoYMkMdA3BSdZrQdQ@mail.gmail.com>

Hi!

Is there any type of Asterisk-like hints in Freeswitch. Like a virtual
extension, you can subscribe for and change it state over dialplan?
Main idea - create BLF Day/Night switch.

-- 
Best regards,
Igor
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From Shawn.Wheeler at interlockconcepts.com  Sat Apr 16 00:29:37 2016
From: Shawn.Wheeler at interlockconcepts.com (Shawn Wheeler)
Date: Fri, 15 Apr 2016 20:29:37 +0000
Subject: [Freeswitch-users] FXO and FXS Devices
In-Reply-To: <634B6B04-9E1D-4D8F-96C7-7D1616FDC306@jerris.com>
References: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<aa8401d19726$35abfbb0$a103f310$@freeswitch.org>
	<CAAemr3xLAqzDus1WXyOR_n2vGmcMtqr_e=-uN1BbVVRywOcSuA@mail.gmail.com>
	<aadf01d1972b$a9d87c20$fd897460$@freeswitch.org>
	<SN1PR0201MB150417DFF44C56BC59973234E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<634B6B04-9E1D-4D8F-96C7-7D1616FDC306@jerris.com>
Message-ID: <SN1PR0201MB1504D5AC1487EC2F0343A916E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>

Thank you but this takes me back to the original question with the add that is must be external


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Friday, April 15, 2016 10:37 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FXO and FXS Devices

sure, just make sure the device interops successfully with your solution.


On Apr 15, 2016, at 12:07 PM, Shawn Wheeler <Shawn.Wheeler at interlockconcepts.com<mailto:Shawn.Wheeler at interlockconcepts.com>> wrote:

How about something like a Linksys will any of those work?


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Friday, April 15, 2016 8:30 AM
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] FXO and FXS Devices

The cost of the boards vs the cost of an external Ethernet gateway are negligible, however, with the gateways you get hardware codecs and if the server crashes you can have the hardware gateways immediate send calls over to the secondary server.

As for USB, I don?t know of anything USB that really works for FXO and FXS.

Companies like Adtran TA900 series and Sangoma Vega series seem to be very reliable, the later can be a little bit of a pain to configure


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob Hartwig
Sent: Friday, April 15, 2016 10:20 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] FXO and FXS Devices

Hey Ken, why the recommendation to use an external gateway instead of FreeTDM / Sangoma / Dahdi-compatible boards?



On Fri, Apr 15, 2016 at 9:50 AM, Ken Rice <krice at freeswitch.org<mailto:krice at freeswitch.org>> wrote:
FreeSWITCH can use Sangoma FXO/FXS  PCI(e) hardware but I recommend using an
external gateway, how many ports are you going to need?

K

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>
[mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Shawn
Wheeler
Sent: Friday, April 15, 2016 9:46 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] FXO and FXS Devices

What types of FXO and FXS devices does FreeSWITCH use?

Thank you

Shawn


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

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

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


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

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

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

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

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

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

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From amani.mansour2 at gmail.com  Sat Apr 16 00:29:27 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Fri, 15 Apr 2016 20:29:27 +0000
Subject: [Freeswitch-users] Register request problem
In-Reply-To: <CAKzWOxVVb3SewDbC=NtO8gLdOjG8RS_OCeZz_i_McPHn+H-zew@mail.gmail.com>
References: <CAFETSWig2AiJsRxZGp8kzbiG6=tPj516ESmiwXng5SnbRvcKiQ@mail.gmail.com>
	<CAKzWOxVVb3SewDbC=NtO8gLdOjG8RS_OCeZz_i_McPHn+H-zew@mail.gmail.com>
Message-ID: <CAFETSWjmjtmC=aBEunvNKki3bSmRBs96bMDEW+p49hxYn9dpNQ@mail.gmail.com>

Hi mr Michael ,

Yes i know that is very difficult, may be impossible for that reason  i
proppose another  solution may be also impossible which is:
when the sipphone send the register request i redirect this packet with
iptables to sipp or scapy and this is will return the 401 unauthorized to
the sipphone in the place of FS .

what do you think sir is this possible ???

Thanks a lot ,
Best regards
amani


Le ven. 15 avr. 2016 ? 18:56, Michael Collins <msc at freeswitch.org> a ?crit :

> This sounds like a job better suited for a SIP proxy like OpenSIPS or
> Kamailio. I suspect that this task is either difficult, impractical, or
> impossible with FreeSWITCH. (FreeSWITCH is a B2BUA.)
>
> That being said, if anyone knows how to manipulate SIP registrations at
> such granular level in Sofia I would be very curious to hear about it.
>
> -Michael
>
> On Fri, Apr 15, 2016 at 7:15 AM, amani mansour <amani.mansour2 at gmail.com>
> wrote:
>
>> Hi all ,
>>
>> Please i need to know the file (directory,dialplan or ........) from
>> which the request register is send ,I need your help ,
>> i need to know the origine of the message Register to be able to modify
>> it i need to create a scenario which is:
>> FS receive Register
>> FS send 401
>> FS receive  Register with authorization header
>> FS send second 401 with stale = false
>>
>> please i need your help .
>> thank you again
>>
>> with best regards
>> amani
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Sat Apr 16 00:38:17 2016
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Apr 2016 13:38:17 -0700
Subject: [Freeswitch-users] FXO and FXS Devices
In-Reply-To: <SN1PR0201MB1504D5AC1487EC2F0343A916E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
References: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<aa8401d19726$35abfbb0$a103f310$@freeswitch.org>
	<CAAemr3xLAqzDus1WXyOR_n2vGmcMtqr_e=-uN1BbVVRywOcSuA@mail.gmail.com>
	<aadf01d1972b$a9d87c20$fd897460$@freeswitch.org>
	<SN1PR0201MB150417DFF44C56BC59973234E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<634B6B04-9E1D-4D8F-96C7-7D1616FDC306@jerris.com>
	<SN1PR0201MB1504D5AC1487EC2F0343A916E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
Message-ID: <CAKzWOxURxJ3Q=Wy_fsxtw4pRC4caq0AN1qUtQiRFQsd9Gx+C7w@mail.gmail.com>

There are many gateways that work with FreeSWITCH, from simple
consumer-grade ones to bigger business-class, rack-mounted devices.

I've personally worked with Cisco SPA devices, Sangoma VegaStream (not a
fan of those, YMMV), Audiocodes, and Patton devices. You might want to
check out the specific feedback people have given here:
https://freeswitch.org/confluence/display/FREESWITCH/Gateways

If you have a specific device or a specific scenario I'm sure the folks
here who have experience would be willing to share their thoughts.

-MSC

On Fri, Apr 15, 2016 at 1:29 PM, Shawn Wheeler <
Shawn.Wheeler at interlockconcepts.com> wrote:

> Thank you but this takes me back to the original question with the add
> that is must be external
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Friday, April 15, 2016 10:37 AM
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FXO and FXS Devices
>
>
>
> sure, just make sure the device interops successfully with your solution.
>
>
>
>
>
> On Apr 15, 2016, at 12:07 PM, Shawn Wheeler <
> Shawn.Wheeler at interlockconcepts.com> wrote:
>
>
>
> How about something like a Linksys will any of those work?
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Ken Rice
> *Sent:* Friday, April 15, 2016 8:30 AM
> *To:* 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FXO and FXS Devices
>
>
>
> The cost of the boards vs the cost of an external Ethernet gateway are
> negligible, however, with the gateways you get hardware codecs and if the
> server crashes you can have the hardware gateways immediate send calls over
> to the secondary server.
>
>
>
> As for USB, I don?t know of anything USB that really works for FXO and FXS.
>
>
>
> Companies like Adtran TA900 series and Sangoma Vega series seem to be very
> reliable, the later can be a little bit of a pain to configure
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>] *On Behalf Of *Bob
> Hartwig
> *Sent:* Friday, April 15, 2016 10:20 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FXO and FXS Devices
>
>
>
> Hey Ken, why the recommendation to use an external gateway instead of
> FreeTDM / Sangoma / Dahdi-compatible boards?
>
>
>
>
>
>
>
> On Fri, Apr 15, 2016 at 9:50 AM, Ken Rice <krice at freeswitch.org> wrote:
>
> FreeSWITCH can use Sangoma FXO/FXS  PCI(e) hardware but I recommend using
> an
> external gateway, how many ports are you going to need?
>
> K
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shawn
> Wheeler
> Sent: Friday, April 15, 2016 9:46 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] FXO and FXS Devices
>
> What types of FXO and FXS devices does FreeSWITCH use?
>
> Thank you
>
> Shawn
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Apr 16 01:12:00 2016
From: davidwaf at gmail.com (David Wafula)
Date: Fri, 15 Apr 2016 23:12:00 +0200
Subject: [Freeswitch-users] mp4 Conference recording is always unplayable
	352 bytes file
Message-ID: <CAK9Ws2eTyYPLq9ERxmCu_xHUAaOaa1k7B9EBL9cpWSGpEa_qTg@mail.gmail.com>

I have verto-client  connecting to a conference. I have setup auto-record
pointing to an mp4 file format for mod_conference. I have installed
libavcodec-extra,
enabled mod_av. The recording seems to start and stop just fine, except the
file is just an unplayable 352 bytes. I must have missed something, just
can't to figure out what.

Am running FreeSWITCH Version 1.6.7-14-d38d065~64bit (-14-d38d065 64bit) on
Debian 8.1 Jesse, x64.

Please anyone with pointers help.

-- 
David W
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From mike at jerris.com  Sat Apr 16 01:39:12 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Apr 2016 17:39:12 -0400
Subject: [Freeswitch-users] mp4 Conference recording is always
	unplayable 352 bytes file
In-Reply-To: <CAK9Ws2eTyYPLq9ERxmCu_xHUAaOaa1k7B9EBL9cpWSGpEa_qTg@mail.gmail.com>
References: <CAK9Ws2eTyYPLq9ERxmCu_xHUAaOaa1k7B9EBL9cpWSGpEa_qTg@mail.gmail.com>
Message-ID: <25F63786-A284-48DB-8387-0340F9B3797F@jerris.com>

is it using mod_av?

> On Apr 15, 2016, at 5:12 PM, David Wafula <davidwaf at gmail.com> wrote:
> 
> I have verto-client  connecting to a conference. I have setup auto-record pointing to an mp4 file format for mod_conference. I have installed libavcodec-extra, enabled mod_av.   The recording seems to start and stop just fine, except the file is just an unplayable 352 bytes. I must have missed something, just can't to figure out what.
> 
> Am running FreeSWITCH Version 1.6.7-14-d38d065~64bit (-14-d38d065 64bit)  on Debian 8.1 Jesse, x64.
> 
> Please anyone with pointers help.

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From davidwaf at gmail.com  Sat Apr 16 01:57:28 2016
From: davidwaf at gmail.com (David Wafula)
Date: Fri, 15 Apr 2016 23:57:28 +0200
Subject: [Freeswitch-users] mp4 Conference recording is always
 unplayable 352 bytes file
In-Reply-To: <25F63786-A284-48DB-8387-0340F9B3797F@jerris.com>
References: <CAK9Ws2eTyYPLq9ERxmCu_xHUAaOaa1k7B9EBL9cpWSGpEa_qTg@mail.gmail.com>
	<25F63786-A284-48DB-8387-0340F9B3797F@jerris.com>
Message-ID: <CAK9Ws2e0Y=MUsmeYNQ1apC3a9q3eTN=5eAscOyAEn70KptvzMg@mail.gmail.com>

I installed  mod_av it like:
 apt-get install freeswitch-mod-av

and put  <load module="mod_av"/> in modules.conf.xml

I think that just about the only thing i did with it.

Am assuming  then than means yes, it is using mod_av ?

(just for testing, i tried wav and  it records perfectly fine)

Regards

On Fri, Apr 15, 2016 at 11:39 PM, Michael Jerris <mike at jerris.com> wrote:

> is it using mod_av?
>
> On Apr 15, 2016, at 5:12 PM, David Wafula <davidwaf at gmail.com> wrote:
>
> I have verto-client  connecting to a conference. I have setup auto-record
> pointing to an mp4 file format for mod_conference. I have installed libavcodec-extra,
> enabled mod_av. The recording seems to start and stop just fine, except the
> file is just an unplayable 352 bytes. I must have missed something, just
> can't to figure out what.
>
> Am running FreeSWITCH Version 1.6.7-14-d38d065~64bit (-14-d38d065 64bit)
> on Debian 8.1 Jesse, x64.
>
> Please anyone with pointers help.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
David W
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From mike at jerris.com  Sat Apr 16 03:15:36 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 15 Apr 2016 19:15:36 -0400
Subject: [Freeswitch-users] mp4 Conference recording is always
	unplayable 352 bytes file
In-Reply-To: <CAK9Ws2e0Y=MUsmeYNQ1apC3a9q3eTN=5eAscOyAEn70KptvzMg@mail.gmail.com>
References: <CAK9Ws2eTyYPLq9ERxmCu_xHUAaOaa1k7B9EBL9cpWSGpEa_qTg@mail.gmail.com>
	<25F63786-A284-48DB-8387-0340F9B3797F@jerris.com>
	<CAK9Ws2e0Y=MUsmeYNQ1apC3a9q3eTN=5eAscOyAEn70KptvzMg@mail.gmail.com>
Message-ID: <54250C19-0C34-46AA-9FCC-74EA5A06C5C6@jerris.com>

look at the debug log, there are other modules that support mp4

> On Apr 15, 2016, at 5:57 PM, David Wafula <davidwaf at gmail.com> wrote:
> 
> I installed  mod_av it like:
>  apt-get install freeswitch-mod-av
> 
> and put  <load module="mod_av"/> in modules.conf.xml
> 
> I think that just about the only thing i did with it.
> 
> Am assuming  then than means yes, it is using mod_av ?
> 
> (just for testing, i tried wav and  it records perfectly fine)
> 
> Regards
> 
> On Fri, Apr 15, 2016 at 11:39 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> is it using mod_av?
> 
>> On Apr 15, 2016, at 5:12 PM, David Wafula <davidwaf at gmail.com <mailto:davidwaf at gmail.com>> wrote:
>> 
>> I have verto-client  connecting to a conference. I have setup auto-record pointing to an mp4 file format for mod_conference. I have installed libavcodec-extra, enabled mod_av.   The recording seems to start and stop just fine, except the file is just an unplayable 352 bytes. I must have missed something, just can't to figure out what.
>> 
>> Am running FreeSWITCH Version 1.6.7-14-d38d065~64bit (-14-d38d065 64bit)  on Debian 8.1 Jesse, x64.
>> 
>> Please anyone with pointers help.
> 

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From alhakeem at gmail.com  Sat Apr 16 15:42:41 2016
From: alhakeem at gmail.com (Abdul Hakeem)
Date: Sat, 16 Apr 2016 12:42:41 +0100
Subject: [Freeswitch-users] HTTP2/Websockets support for mod XML CURL
Message-ID: <CAO=waFD+S4cp1dEE4HabaBwB517d2ku5CiL8io_wfwKqtxDQvA@mail.gmail.com>

Hello,
Is there http2/websockets support such as connection pooling, keepalive,
header compression etc inbuilt into mod xml curl ?.

My concern is for the setup teardown roundtrips for each call made by FS.
I am hoping someone has worked on a push technology where a websserver
listen for events, then pushes the xml data to FS.

Does anyone have a clue on how to configure or how to make mod xml curl do
this ?

Cheers,
AH
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From colin.morelli at gmail.com  Sat Apr 16 20:32:34 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sat, 16 Apr 2016 16:32:34 +0000
Subject: [Freeswitch-users] Scaling Freeswitch
Message-ID: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>

Does anyone have any good references for horizontally scaling out large
multi-tenant FS clusters? Most of what I've been able to find involving
load balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no
recent versions of the information available. Has this not changed or is
there a fundamental shift in how people have been tackling this problem?

To clarify, I'm just looking for pointers/references here. Although if
anyone has some personal experience I'd greatly appreciate specific
examples and insight as well.

Thanks in advance.

Best,
Colin
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From mario_fs at mgtech.com  Sat Apr 16 20:46:39 2016
From: mario_fs at mgtech.com (Mario G)
Date: Sat, 16 Apr 2016 09:46:39 -0700
Subject: [Freeswitch-users] FXO and FXS Devices
In-Reply-To: <CAKzWOxURxJ3Q=Wy_fsxtw4pRC4caq0AN1qUtQiRFQsd9Gx+C7w@mail.gmail.com>
References: <SN1PR0201MB1504BD3A6121FA322CDBAB4DE0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<aa8401d19726$35abfbb0$a103f310$@freeswitch.org>
	<CAAemr3xLAqzDus1WXyOR_n2vGmcMtqr_e=-uN1BbVVRywOcSuA@mail.gmail.com>
	<aadf01d1972b$a9d87c20$fd897460$@freeswitch.org>
	<SN1PR0201MB150417DFF44C56BC59973234E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<634B6B04-9E1D-4D8F-96C7-7D1616FDC306@jerris.com>
	<SN1PR0201MB1504D5AC1487EC2F0343A916E0680@SN1PR0201MB1504.namprd02.prod.outlook.com>
	<CAKzWOxURxJ3Q=Wy_fsxtw4pRC4caq0AN1qUtQiRFQsd9Gx+C7w@mail.gmail.com>
Message-ID: <A52AD862-B9B7-4EBC-B562-3C3E0A039766@mgtech.com>

Look at Obihai, they have a wide range of gateways. I am using the OBI110 and wrote the wiki page on it: https://freeswitch.org/confluence/display/FREESWITCH/Obihai

> On Apr 15, 2016, at 1:38 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
> There are many gateways that work with FreeSWITCH, from simple consumer-grade ones to bigger business-class, rack-mounted devices.
> 
> I've personally worked with Cisco SPA devices, Sangoma VegaStream (not a fan of those, YMMV), Audiocodes, and Patton devices. You might want to check out the specific feedback people have given here:
> https://freeswitch.org/confluence/display/FREESWITCH/Gateways <https://freeswitch.org/confluence/display/FREESWITCH/Gateways>
> 
> If you have a specific device or a specific scenario I'm sure the folks here who have experience would be willing to share their thoughts.
> 
> -MSC
> 
> On Fri, Apr 15, 2016 at 1:29 PM, Shawn Wheeler <Shawn.Wheeler at interlockconcepts.com <mailto:Shawn.Wheeler at interlockconcepts.com>> wrote:
> Thank you but this takes me back to the original question with the add that is must be external
> 
>  
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Jerris
> Sent: Friday, April 15, 2016 10:37 AM
> 
> 
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] FXO and FXS Devices
> 
>  
> 
> sure, just make sure the device interops successfully with your solution.
> 
>  
> 
>  
> 
> On Apr 15, 2016, at 12:07 PM, Shawn Wheeler <Shawn.Wheeler at interlockconcepts.com <mailto:Shawn.Wheeler at interlockconcepts.com>> wrote:
> 
>  
> 
> How about something like a Linksys will any of those work?
> 
>  
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Ken Rice
> Sent: Friday, April 15, 2016 8:30 AM
> To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] FXO and FXS Devices
> 
>  
> 
> The cost of the boards vs the cost of an external Ethernet gateway are negligible, however, with the gateways you get hardware codecs and if the server crashes you can have the hardware gateways immediate send calls over to the secondary server.
> 
>  
> 
> As for USB, I don?t know of anything USB that really works for FXO and FXS.
> 
>  
> 
> Companies like Adtran TA900 series and Sangoma Vega series seem to be very reliable, the later can be a little bit of a pain to configure
> 
>  
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Bob Hartwig
> Sent: Friday, April 15, 2016 10:20 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] FXO and FXS Devices
> 
>  
> 
> Hey Ken, why the recommendation to use an external gateway instead of FreeTDM / Sangoma / Dahdi-compatible boards?
> 
>  
> 
>  
> 
>  
> 
> On Fri, Apr 15, 2016 at 9:50 AM, Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org>> wrote:
> 
> FreeSWITCH can use Sangoma FXO/FXS  PCI(e) hardware but I recommend using an
> external gateway, how many ports are you going to need?
> 
> K
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Shawn
> Wheeler
> Sent: Friday, April 15, 2016 9:46 AM
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: [Freeswitch-users] FXO and FXS Devices
> 
> What types of FXO and FXS devices does FreeSWITCH use?
> 
> Thank you
> 
> Shawn
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
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> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From luis.daniel.lucio at gmail.com  Sat Apr 16 20:47:00 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sat, 16 Apr 2016 12:47:00 -0400
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
Message-ID: <CAFLo2QxXCsbeAeYj=AxQ_-ept1nRT47i0bO5bRA351fJiG_oyA@mail.gmail.com>

Explain more what you want to do. I have dinner it without kamalio. Don't
know if that fits your needs
Le 16 avr. 2016 12:41 PM, "Colin Morelli" <colin.morelli at gmail.com> a
?crit :
>
> Does anyone have any good references for horizontally scaling out large
multi-tenant FS clusters? Most of what I've been able to find involving
load balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no
recent versions of the information available. Has this not changed or is
there a fundamental shift in how people have been tackling this problem?
>
> To clarify, I'm just looking for pointers/references here. Although if
anyone has some personal experience I'd greatly appreciate specific
examples and insight as well.
>
> Thanks in advance.
>
> Best,
> Colin
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From colin.morelli at gmail.com  Sat Apr 16 21:01:58 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sat, 16 Apr 2016 17:01:58 +0000
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAFLo2QxXCsbeAeYj=AxQ_-ept1nRT47i0bO5bRA351fJiG_oyA@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAFLo2QxXCsbeAeYj=AxQ_-ept1nRT47i0bO5bRA351fJiG_oyA@mail.gmail.com>
Message-ID: <CAPtU-UoROhpzV90OdrHdJD=Lx-8Nxi9o--wRF4kc1j_-x=iPZg@mail.gmail.com>

I think that's part of what I'm trying to figure out here.

I'm looking to run a SIP platform that will support multiple tenants and
device types (SIP phones, WebRTC clients, etc). Each tenant can be isolated
to a subset of hosts, as there's no need to bridge across multiple tenants.
My initial thought was to run SIP proxies in front of small clusters of FS
servers. Essentially creating cluster A, B, C, and so on, each of which is
made up of a few FS hosts. Then, have a much smaller number of Kamailio
instances in the front that essentially proxy SIP traffic to the
appropriate SIP cluster for the requested domain.

However, I'm not sure how well this scales. I've been reading a lot about
FS being great as a media server, but there being better options for the
signaling portion. My proposal would still push SIP registration and
signaling to FS, just with a proxy in front. The alternative approach is to
have Kamailio do all SIP registration and signaling, using FS as a media
server, but I'm not sure what implications this has on the ability to do
dynamic call routing in FS (for example, how would I use uuid_intercept to
intercept a live call if Kamailio is performing all of the signaling)

Most likely my issue is just a lack of depth in the understanding of the
roles that Kamailio and FS would play in a hybrid scenario (I'll admit I'm
new to this).

Thanks for the response.

Best,
Colin

On Sat, Apr 16, 2016 at 12:54 PM Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> Explain more what you want to do. I have dinner it without kamalio. Don't
> know if that fits your needs
>
>
> Le 16 avr. 2016 12:41 PM, "Colin Morelli" <colin.morelli at gmail.com> a
> ?crit :
> >
> > Does anyone have any good references for horizontally scaling out large
> multi-tenant FS clusters? Most of what I've been able to find involving
> load balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no
> recent versions of the information available. Has this not changed or is
> there a fundamental shift in how people have been tackling this problem?
> >
> > To clarify, I'm just looking for pointers/references here. Although if
> anyone has some personal experience I'd greatly appreciate specific
> examples and insight as well.
> >
> > Thanks in advance.
> >
> > Best,
> > Colin
> >
>
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 16 21:05:35 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 16 Apr 2016 17:05:35 +0000
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
Message-ID: <CAHtxdDfRmNhnOvfw0LZHVW3XhUXmZ17=qFMExOsAdKU8utth8A@mail.gmail.com>

Usage of kamailio as registrator and FreeSwitch as call processing server
is perfect way to create scalable VoIP infrastructure.

Sergey.

??, 16 ???. 2016 ?. ? 19:40, Colin Morelli <colin.morelli at gmail.com>:

> Does anyone have any good references for horizontally scaling out large
> multi-tenant FS clusters? Most of what I've been able to find involving
> load balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no
> recent versions of the information available. Has this not changed or is
> there a fundamental shift in how people have been tackling this problem?
>
> To clarify, I'm just looking for pointers/references here. Although if
> anyone has some personal experience I'd greatly appreciate specific
> examples and insight as well.
>
> Thanks in advance.
>
> Best,
> Colin
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jurijs.ivolga at gmail.com  Sat Apr 16 21:11:33 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Sat, 16 Apr 2016 20:11:33 +0300
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAHtxdDfRmNhnOvfw0LZHVW3XhUXmZ17=qFMExOsAdKU8utth8A@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAHtxdDfRmNhnOvfw0LZHVW3XhUXmZ17=qFMExOsAdKU8utth8A@mail.gmail.com>
Message-ID: <CAOrA2UbBJptVfqmECouV83FmaMcp0QHktrAMrvLS14AMfsMOkg@mail.gmail.com>

Hi,

It is not mandatory to use Kamailio as registrar server, you can use
Freeswitch as registrar server and Kamailio can just load balance all SIP
messages to Freeswitch, including registrations...

With kind regards,

Jurijs

On Sat, Apr 16, 2016 at 8:05 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Usage of kamailio as registrator and FreeSwitch as call processing server
> is perfect way to create scalable VoIP infrastructure.
>
> Sergey.
>
> ??, 16 ???. 2016 ?. ? 19:40, Colin Morelli <colin.morelli at gmail.com>:
>
>> Does anyone have any good references for horizontally scaling out large
>> multi-tenant FS clusters? Most of what I've been able to find involving
>> load balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no
>> recent versions of the information available. Has this not changed or is
>> there a fundamental shift in how people have been tackling this problem?
>>
>> To clarify, I'm just looking for pointers/references here. Although if
>> anyone has some personal experience I'd greatly appreciate specific
>> examples and insight as well.
>>
>> Thanks in advance.
>>
>> Best,
>> Colin
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From colin.morelli at gmail.com  Sat Apr 16 21:17:20 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sat, 16 Apr 2016 17:17:20 +0000
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAOrA2UbBJptVfqmECouV83FmaMcp0QHktrAMrvLS14AMfsMOkg@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAHtxdDfRmNhnOvfw0LZHVW3XhUXmZ17=qFMExOsAdKU8utth8A@mail.gmail.com>
	<CAOrA2UbBJptVfqmECouV83FmaMcp0QHktrAMrvLS14AMfsMOkg@mail.gmail.com>
Message-ID: <CAPtU-Uq6Hpo7ZBHpgJwC3Pab--EHP3rp4d0qUDfscTywf+Bb-w@mail.gmail.com>

Jurijs,

That's good to know, since that's what my original plan was. Is there
particular advantage, then, to using Kamailio for SIP registrations if this
is a possibility? It would seem that you could have virtually "infinite"
scale with this approach. Am I right to assume that you could TCP load
balance traffic to a small number of Kamailio servers up front, which then
have a database mapping SIP domain -> another cluster of FS servers behind
them?

So, tenants 1, 2 and 3 go to cluster A, 4, 5, 6 to cluster B, and so on? If
so, am I missing why you might want Kamailio to perform the registration at
all?

Best,
Colin

On Sat, Apr 16, 2016 at 1:13 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:

> Hi,
>
> It is not mandatory to use Kamailio as registrar server, you can use
> Freeswitch as registrar server and Kamailio can just load balance all SIP
> messages to Freeswitch, including registrations...
>
> With kind regards,
>
> Jurijs
>
> On Sat, Apr 16, 2016 at 8:05 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Usage of kamailio as registrator and FreeSwitch as call processing server
>> is perfect way to create scalable VoIP infrastructure.
>>
>> Sergey.
>>
>> ??, 16 ???. 2016 ?. ? 19:40, Colin Morelli <colin.morelli at gmail.com>:
>>
>>> Does anyone have any good references for horizontally scaling out large
>>> multi-tenant FS clusters? Most of what I've been able to find involving
>>> load balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no
>>> recent versions of the information available. Has this not changed or is
>>> there a fundamental shift in how people have been tackling this problem?
>>>
>>> To clarify, I'm just looking for pointers/references here. Although if
>>> anyone has some personal experience I'd greatly appreciate specific
>>> examples and insight as well.
>>>
>>> Thanks in advance.
>>>
>>> Best,
>>> Colin
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 16 21:25:57 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 16 Apr 2016 17:25:57 +0000
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAPtU-Uq6Hpo7ZBHpgJwC3Pab--EHP3rp4d0qUDfscTywf+Bb-w@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAHtxdDfRmNhnOvfw0LZHVW3XhUXmZ17=qFMExOsAdKU8utth8A@mail.gmail.com>
	<CAOrA2UbBJptVfqmECouV83FmaMcp0QHktrAMrvLS14AMfsMOkg@mail.gmail.com>
	<CAPtU-Uq6Hpo7ZBHpgJwC3Pab--EHP3rp4d0qUDfscTywf+Bb-w@mail.gmail.com>
Message-ID: <CAHtxdDczvWqKFwQ5PtGYztd7-icTva-wtkap6E6rMX83Uo1vOQ@mail.gmail.com>

"Is there particular advantage, then, to using Kamailio for SIP
registrations if this is a possibility?"
This simplifies sending messages to the client in a case where it is behind
NAT.

Sergey


??, 16 ???. 2016 ?. ? 20:18, Colin Morelli <colin.morelli at gmail.com>:

> Jurijs,
>
> That's good to know, since that's what my original plan was. Is there
> particular advantage, then, to using Kamailio for SIP registrations if this
> is a possibility? It would seem that you could have virtually "infinite"
> scale with this approach. Am I right to assume that you could TCP load
> balance traffic to a small number of Kamailio servers up front, which then
> have a database mapping SIP domain -> another cluster of FS servers behind
> them?
>
> So, tenants 1, 2 and 3 go to cluster A, 4, 5, 6 to cluster B, and so on?
> If so, am I missing why you might want Kamailio to perform the registration
> at all?
>
> Best,
> Colin
>
> On Sat, Apr 16, 2016 at 1:13 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
> wrote:
>
>> Hi,
>>
>> It is not mandatory to use Kamailio as registrar server, you can use
>> Freeswitch as registrar server and Kamailio can just load balance all SIP
>> messages to Freeswitch, including registrations...
>>
>> With kind regards,
>>
>> Jurijs
>>
>> On Sat, Apr 16, 2016 at 8:05 PM, Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> Usage of kamailio as registrator and FreeSwitch as call processing
>>> server is perfect way to create scalable VoIP infrastructure.
>>>
>>> Sergey.
>>>
>>> ??, 16 ???. 2016 ?. ? 19:40, Colin Morelli <colin.morelli at gmail.com>:
>>>
>>>> Does anyone have any good references for horizontally scaling out large
>>>> multi-tenant FS clusters? Most of what I've been able to find involving
>>>> load balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no
>>>> recent versions of the information available. Has this not changed or is
>>>> there a fundamental shift in how people have been tackling this problem?
>>>>
>>>> To clarify, I'm just looking for pointers/references here. Although if
>>>> anyone has some personal experience I'd greatly appreciate specific
>>>> examples and insight as well.
>>>>
>>>> Thanks in advance.
>>>>
>>>> Best,
>>>> Colin
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jurijs.ivolga at gmail.com  Sat Apr 16 21:42:51 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Sat, 16 Apr 2016 20:42:51 +0300
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAHtxdDczvWqKFwQ5PtGYztd7-icTva-wtkap6E6rMX83Uo1vOQ@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAHtxdDfRmNhnOvfw0LZHVW3XhUXmZ17=qFMExOsAdKU8utth8A@mail.gmail.com>
	<CAOrA2UbBJptVfqmECouV83FmaMcp0QHktrAMrvLS14AMfsMOkg@mail.gmail.com>
	<CAPtU-Uq6Hpo7ZBHpgJwC3Pab--EHP3rp4d0qUDfscTywf+Bb-w@mail.gmail.com>
	<CAHtxdDczvWqKFwQ5PtGYztd7-icTva-wtkap6E6rMX83Uo1vOQ@mail.gmail.com>
Message-ID: <CAOrA2UbQzNdAFU9e402uUwsH-S5nwOXLc8DPsWADHcSLjUZjPw@mail.gmail.com>

Hi,

Kamailio is SIP proxy, so it was build to support thousands and millions
SIP registrations and it will perform better then freeswitch as registrar
server(no offense guys, freeswitch is brilliant software, but it can't be
1st everywhere :D), so from performance perspective, Kamailio is better to
use for registration, then Freeswitch, but if you are building set-up for
no more then 10K(or even more) extensions, then there is no such huge
difference in performance and you can leave registrar server on Freeswitch
side...

By default I will recommend you to go in standard way - Kamailio as
registrar server, but again I don't know your full requirements and maybe
this is not best way for you...

I would like to emphasize that Freeswitch as registrar server is just one
option from several other options and I really don't know which one is best
for you.

With kind regards,

Jurijs

On Sat, Apr 16, 2016 at 8:25 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> "Is there particular advantage, then, to using Kamailio for SIP
> registrations if this is a possibility?"
> This simplifies sending messages to the client in a case where it is
> behind NAT.
>
> Sergey
>
>
> ??, 16 ???. 2016 ?. ? 20:18, Colin Morelli <colin.morelli at gmail.com>:
>
>> Jurijs,
>>
>> That's good to know, since that's what my original plan was. Is there
>> particular advantage, then, to using Kamailio for SIP registrations if this
>> is a possibility? It would seem that you could have virtually "infinite"
>> scale with this approach. Am I right to assume that you could TCP load
>> balance traffic to a small number of Kamailio servers up front, which then
>> have a database mapping SIP domain -> another cluster of FS servers behind
>> them?
>>
>> So, tenants 1, 2 and 3 go to cluster A, 4, 5, 6 to cluster B, and so on?
>> If so, am I missing why you might want Kamailio to perform the registration
>> at all?
>>
>> Best,
>> Colin
>>
>> On Sat, Apr 16, 2016 at 1:13 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
>> wrote:
>>
>>> Hi,
>>>
>>> It is not mandatory to use Kamailio as registrar server, you can use
>>> Freeswitch as registrar server and Kamailio can just load balance all SIP
>>> messages to Freeswitch, including registrations...
>>>
>>> With kind regards,
>>>
>>> Jurijs
>>>
>>> On Sat, Apr 16, 2016 at 8:05 PM, Sergey Safarov <s.safarov at gmail.com>
>>> wrote:
>>>
>>>> Usage of kamailio as registrator and FreeSwitch as call processing
>>>> server is perfect way to create scalable VoIP infrastructure.
>>>>
>>>> Sergey.
>>>>
>>>> ??, 16 ???. 2016 ?. ? 19:40, Colin Morelli <colin.morelli at gmail.com>:
>>>>
>>>>> Does anyone have any good references for horizontally scaling out
>>>>> large multi-tenant FS clusters? Most of what I've been able to find
>>>>> involving load balancing Kamailio/OpenSIPS is fairly old (2+ years), and
>>>>> there's no recent versions of the information available. Has this not
>>>>> changed or is there a fundamental shift in how people have been tackling
>>>>> this problem?
>>>>>
>>>>> To clarify, I'm just looking for pointers/references here. Although if
>>>>> anyone has some personal experience I'd greatly appreciate specific
>>>>> examples and insight as well.
>>>>>
>>>>> Thanks in advance.
>>>>>
>>>>> Best,
>>>>> Colin
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From colin.morelli at gmail.com  Sat Apr 16 22:27:31 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sat, 16 Apr 2016 18:27:31 +0000
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAOrA2UbQzNdAFU9e402uUwsH-S5nwOXLc8DPsWADHcSLjUZjPw@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAHtxdDfRmNhnOvfw0LZHVW3XhUXmZ17=qFMExOsAdKU8utth8A@mail.gmail.com>
	<CAOrA2UbBJptVfqmECouV83FmaMcp0QHktrAMrvLS14AMfsMOkg@mail.gmail.com>
	<CAPtU-Uq6Hpo7ZBHpgJwC3Pab--EHP3rp4d0qUDfscTywf+Bb-w@mail.gmail.com>
	<CAHtxdDczvWqKFwQ5PtGYztd7-icTva-wtkap6E6rMX83Uo1vOQ@mail.gmail.com>
	<CAOrA2UbQzNdAFU9e402uUwsH-S5nwOXLc8DPsWADHcSLjUZjPw@mail.gmail.com>
Message-ID: <CAPtU-Upui9sT5tWue4FC9mGVBW+raPyjTxD6oh9SJZA2nXGNwg@mail.gmail.com>

Thanks for the additional info, Jurijs. This has been very helpful already.

Sorry I'm a bit slow here (a lot of this is quite new to me, so I'm trying
to absorb as much knowledge as I can). If Kamailio is the registrar, what
role does it play in call routing? For example, if a call hits FS and I
instruct FS to bridge to a user, how is that performed? Or does all call
routing now need to be performed on Kamailio, since it's the one that's
aware of where users are (and their associated presence)?

I'd just like to better understand the roles that each of these plays in
this scenario.

Thanks again.

Best,
Colin

On Sat, Apr 16, 2016 at 1:45 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:

> Hi,
>
> Kamailio is SIP proxy, so it was build to support thousands and millions
> SIP registrations and it will perform better then freeswitch as registrar
> server(no offense guys, freeswitch is brilliant software, but it can't be
> 1st everywhere :D), so from performance perspective, Kamailio is better to
> use for registration, then Freeswitch, but if you are building set-up for
> no more then 10K(or even more) extensions, then there is no such huge
> difference in performance and you can leave registrar server on Freeswitch
> side...
>
> By default I will recommend you to go in standard way - Kamailio as
> registrar server, but again I don't know your full requirements and maybe
> this is not best way for you...
>
> I would like to emphasize that Freeswitch as registrar server is just one
> option from several other options and I really don't know which one is best
> for you.
>
> With kind regards,
>
> Jurijs
>
> On Sat, Apr 16, 2016 at 8:25 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> "Is there particular advantage, then, to using Kamailio for SIP
>> registrations if this is a possibility?"
>> This simplifies sending messages to the client in a case where it is
>> behind NAT.
>>
>> Sergey
>>
>>
>> ??, 16 ???. 2016 ?. ? 20:18, Colin Morelli <colin.morelli at gmail.com>:
>>
>>> Jurijs,
>>>
>>> That's good to know, since that's what my original plan was. Is there
>>> particular advantage, then, to using Kamailio for SIP registrations if this
>>> is a possibility? It would seem that you could have virtually "infinite"
>>> scale with this approach. Am I right to assume that you could TCP load
>>> balance traffic to a small number of Kamailio servers up front, which then
>>> have a database mapping SIP domain -> another cluster of FS servers behind
>>> them?
>>>
>>> So, tenants 1, 2 and 3 go to cluster A, 4, 5, 6 to cluster B, and so on?
>>> If so, am I missing why you might want Kamailio to perform the registration
>>> at all?
>>>
>>> Best,
>>> Colin
>>>
>>> On Sat, Apr 16, 2016 at 1:13 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> It is not mandatory to use Kamailio as registrar server, you can use
>>>> Freeswitch as registrar server and Kamailio can just load balance all SIP
>>>> messages to Freeswitch, including registrations...
>>>>
>>>> With kind regards,
>>>>
>>>> Jurijs
>>>>
>>>> On Sat, Apr 16, 2016 at 8:05 PM, Sergey Safarov <s.safarov at gmail.com>
>>>> wrote:
>>>>
>>>>> Usage of kamailio as registrator and FreeSwitch as call processing
>>>>> server is perfect way to create scalable VoIP infrastructure.
>>>>>
>>>>> Sergey.
>>>>>
>>>>> ??, 16 ???. 2016 ?. ? 19:40, Colin Morelli <colin.morelli at gmail.com>:
>>>>>
>>>>>> Does anyone have any good references for horizontally scaling out
>>>>>> large multi-tenant FS clusters? Most of what I've been able to find
>>>>>> involving load balancing Kamailio/OpenSIPS is fairly old (2+ years), and
>>>>>> there's no recent versions of the information available. Has this not
>>>>>> changed or is there a fundamental shift in how people have been tackling
>>>>>> this problem?
>>>>>>
>>>>>> To clarify, I'm just looking for pointers/references here. Although
>>>>>> if anyone has some personal experience I'd greatly appreciate specific
>>>>>> examples and insight as well.
>>>>>>
>>>>>> Thanks in advance.
>>>>>>
>>>>>> Best,
>>>>>> Colin
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Apr 16 23:23:10 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Sat, 16 Apr 2016 12:23:10 -0700
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
Message-ID: <CAKE2PDuMaAYshxNR8-oqB9O4ZF==K4J1EDKxNHSY7HPriXuBWw@mail.gmail.com>

On 16 April 2016 at 09:32, Colin Morelli <colin.morelli at gmail.com> wrote:
> To clarify, I'm just looking for pointers/references here. Although if
> anyone has some personal experience I'd greatly appreciate specific examples
> and insight as well.

This won't answer all your questions but it will give you an idea of
freeswitch + Kamailio:

https://www.youtube.com/watch?v=6VsuC7-jHc4



-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From jurijs.ivolga at gmail.com  Sun Apr 17 01:16:55 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Sun, 17 Apr 2016 00:16:55 +0300
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAKE2PDuMaAYshxNR8-oqB9O4ZF==K4J1EDKxNHSY7HPriXuBWw@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAKE2PDuMaAYshxNR8-oqB9O4ZF==K4J1EDKxNHSY7HPriXuBWw@mail.gmail.com>
Message-ID: <CAOrA2UZk6-kS58m3hwqR7d+Voxnj55WQM_DWQz+Ai=odtXX-jw@mail.gmail.com>

Hi,

If Kamailio is the registrar, what role does it play in call routing? For
> example, if a call hits FS and I instruct FS to bridge to a user, how is
> that performed? Or does all call routing now need to be performed on
> Kamailio, since it's the one that's aware of where users are (and their
> associated presence)?
>

You can configure Kamailio and Freeswitch in anyway you need, Kamailio can
be call router, freeswitch can be call router and both of them
simultaneously can be call routers :)

But if we will take following manual as starting point:

http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc

Then users location DB will be in Kamilio and information regarding users
will be stored in Kamailio, so Freeswitch will not know where user located
and all calls between extensions will go through Freeswitch and then will
be looped back to Kamailio and Kamailio will decide where to route call.

So in this case Kamailio is call router and it decides where to route
calls, for example 44 prefix is routed to "vbox"(voicemail).

Nevertheless you can add some additional routing on Freeswitch too if
necessary, for example if you need to send calls to PSTN and you need
transcoding.

Nevertheless it is still possible to route PSTN calls directly from
Kamailio too, without freeswitch.

With kind regards,

Jurijs

On Sat, Apr 16, 2016 at 10:23 PM, jungle Boogie <jungleboogie0 at gmail.com>
wrote:

> On 16 April 2016 at 09:32, Colin Morelli <colin.morelli at gmail.com> wrote:
> > To clarify, I'm just looking for pointers/references here. Although if
> > anyone has some personal experience I'd greatly appreciate specific
> examples
> > and insight as well.
>
> This won't answer all your questions but it will give you an idea of
> freeswitch + Kamailio:
>
> https://www.youtube.com/watch?v=6VsuC7-jHc4
>
>
>
> --
> -------
> inum: 883510009027723
> sip: jungleboogie at sip2sip.info
> xmpp: jungle-boogie at jit.si
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From davidwaf at gmail.com  Sun Apr 17 01:28:14 2016
From: davidwaf at gmail.com (David Wafula)
Date: Sat, 16 Apr 2016 23:28:14 +0200
Subject: [Freeswitch-users] mp4 Conference recording is always
 unplayable 352 bytes file
In-Reply-To: <54250C19-0C34-46AA-9FCC-74EA5A06C5C6@jerris.com>
References: <CAK9Ws2eTyYPLq9ERxmCu_xHUAaOaa1k7B9EBL9cpWSGpEa_qTg@mail.gmail.com>
	<25F63786-A284-48DB-8387-0340F9B3797F@jerris.com>
	<CAK9Ws2e0Y=MUsmeYNQ1apC3a9q3eTN=5eAscOyAEn70KptvzMg@mail.gmail.com>
	<54250C19-0C34-46AA-9FCC-74EA5A06C5C6@jerris.com>
Message-ID: <CAK9Ws2ezEDnccH+1JaudtsgvULNNAhBey+zAVs0KpmAFBHoPjQ@mail.gmail.com>

Please see the log here if there is anything am doing wrong:
http://pastebin.com/WGcmg2iX

I noticed one line there:
2016-04-16 21:12:10.381543 [DEBUG] mod_verto.c:1862 BAD READ -1000

is that normal ?




On Sat, Apr 16, 2016 at 1:15 AM, Michael Jerris <mike at jerris.com> wrote:

> look at the debug log, there are other modules that support mp4
>
> On Apr 15, 2016, at 5:57 PM, David Wafula <davidwaf at gmail.com> wrote:
>
> I installed  mod_av it like:
>  apt-get install freeswitch-mod-av
>
> and put  <load module="mod_av"/> in modules.conf.xml
>
> I think that just about the only thing i did with it.
>
> Am assuming  then than means yes, it is using mod_av ?
>
> (just for testing, i tried wav and  it records perfectly fine)
>
> Regards
>
> On Fri, Apr 15, 2016 at 11:39 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> is it using mod_av?
>>
>> On Apr 15, 2016, at 5:12 PM, David Wafula <davidwaf at gmail.com> wrote:
>>
>> I have verto-client  connecting to a conference. I have setup auto-record
>> pointing to an mp4 file format for mod_conference. I have installed libavcodec-extra,
>> enabled mod_av. The recording seems to start and stop just fine, except the
>> file is just an unplayable 352 bytes. I must have missed something, just
>> can't to figure out what.
>>
>> Am running FreeSWITCH Version 1.6.7-14-d38d065~64bit (-14-d38d065 64bit)
>> on Debian 8.1 Jesse, x64.
>>
>> Please anyone with pointers help.
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
David W
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From davidwaf at gmail.com  Sun Apr 17 02:17:54 2016
From: davidwaf at gmail.com (David Wafula)
Date: Sun, 17 Apr 2016 00:17:54 +0200
Subject: [Freeswitch-users] mp4 Conference recording is always
 unplayable 352 bytes file
In-Reply-To: <CAK9Ws2ezEDnccH+1JaudtsgvULNNAhBey+zAVs0KpmAFBHoPjQ@mail.gmail.com>
References: <CAK9Ws2eTyYPLq9ERxmCu_xHUAaOaa1k7B9EBL9cpWSGpEa_qTg@mail.gmail.com>
	<25F63786-A284-48DB-8387-0340F9B3797F@jerris.com>
	<CAK9Ws2e0Y=MUsmeYNQ1apC3a9q3eTN=5eAscOyAEn70KptvzMg@mail.gmail.com>
	<54250C19-0C34-46AA-9FCC-74EA5A06C5C6@jerris.com>
	<CAK9Ws2ezEDnccH+1JaudtsgvULNNAhBey+zAVs0KpmAFBHoPjQ@mail.gmail.com>
Message-ID: <CAK9Ws2df=e3PpffNbikZ9N+yeNsaUfeU1_UsnHxEHSgbAhpb5w@mail.gmail.com>

So i did a comparison between wav recording and mp4 recording, and it seems
BAD READ -1000 only occurs when using mp4.

Here is part  log for mp4

----
3f39d3c5-97cb-f91a-48b5-c016e82175e2 2016-04-16 21:12:10.361551 [NOTICE]
mod_verto.c:2746 Hangup verto.rtc/83789 [CS_EXECUTE] [NORMAL_CLEARING]
2016-04-16 21:12:10.361551 [DEBUG] switch_core_media.c:5459 verto.rtc/83789
Video thread ended
3f39d3c5-97cb-f91a-48b5-c016e82175e2 2016-04-16 21:12:10.381543 [INFO]
conference_loop.c:1400 Channel leaving conference, cause: NORMAL_CLEARING
3f39d3c5-97cb-f91a-48b5-c016e82175e2 2016-04-16 21:12:10.381543 [DEBUG]
mod_conference.c:2209 verto.rtc/83789 skip receive message [UNBRIDGE]
(channel is hungup already)
2016-04-16 21:12:10.381543 [DEBUG] mod_verto.c:1862 BAD READ -1000
2016-04-16 21:12:10.381543 [INFO] mod_verto.c:2005 105.226.104.136:46232
Ending client thread.
2016-04-16 21:12:10.381543 [INFO] mod_verto.c:2013 105.226.104.136:46232
Thread ended

--



Here is the same bit of log with just wav recording:

-------

d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.490132 [NOTICE]
mod_verto.c:2746 Hangup verto.rtc/83789 [CS_EXECUTE] [NORMAL_CLEARING]
2016-04-16 22:06:57.500121 [DEBUG] switch_core_media.c:5459 verto.rtc/83789
Video thread ended
d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [INFO]
conference_loop.c:1400 Channel leaving conference, cause: NORMAL_CLEARING
d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG]
mod_conference.c:2209 verto.rtc/83789 skip receive message [UNBRIDGE]
(channel is hungup already)
d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG]
switch_core_media.c:9147 verto.rtc/83789 skip receive message [HARD_MUTE]
(channel is hungup already)
d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG]
switch_core_codec.c:258 Restore original codec.
d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG]
switch_core_session.c:2796 verto.rtc/83789 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)

------
Thank you

On Sat, Apr 16, 2016 at 11:28 PM, David Wafula <davidwaf at gmail.com> wrote:

> Please see the log here if there is anything am doing wrong:
> http://pastebin.com/WGcmg2iX
>
> I noticed one line there:
> 2016-04-16 21:12:10.381543 [DEBUG] mod_verto.c:1862 BAD READ -1000
>
> is that normal ?
>
>
>
>
> On Sat, Apr 16, 2016 at 1:15 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> look at the debug log, there are other modules that support mp4
>>
>> On Apr 15, 2016, at 5:57 PM, David Wafula <davidwaf at gmail.com> wrote:
>>
>> I installed  mod_av it like:
>>  apt-get install freeswitch-mod-av
>>
>> and put  <load module="mod_av"/> in modules.conf.xml
>>
>> I think that just about the only thing i did with it.
>>
>> Am assuming  then than means yes, it is using mod_av ?
>>
>> (just for testing, i tried wav and  it records perfectly fine)
>>
>> Regards
>>
>> On Fri, Apr 15, 2016 at 11:39 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> is it using mod_av?
>>>
>>> On Apr 15, 2016, at 5:12 PM, David Wafula <davidwaf at gmail.com> wrote:
>>>
>>> I have verto-client  connecting to a conference. I have setup
>>> auto-record pointing to an mp4 file format for mod_conference. I have
>>> installed libavcodec-extra, enabled mod_av. The recording seems to
>>> start and stop just fine, except the file is just an unplayable 352 bytes.
>>> I must have missed something, just can't to figure out what.
>>>
>>> Am running FreeSWITCH Version 1.6.7-14-d38d065~64bit (-14-d38d065 64bit)
>>> on Debian 8.1 Jesse, x64.
>>>
>>> Please anyone with pointers help.
>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> David W
>



-- 
David W
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From bc at iptel.co  Sun Apr 17 04:13:47 2016
From: bc at iptel.co (Brian ::)
Date: Sun, 17 Apr 2016 01:13:47 +0100
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAPtU-UoROhpzV90OdrHdJD=Lx-8Nxi9o--wRF4kc1j_-x=iPZg@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAFLo2QxXCsbeAeYj=AxQ_-ept1nRT47i0bO5bRA351fJiG_oyA@mail.gmail.com>
	<CAPtU-UoROhpzV90OdrHdJD=Lx-8Nxi9o--wRF4kc1j_-x=iPZg@mail.gmail.com>
Message-ID: <CAAr8QowUjrEO9nx8ckKX-za4eJubZJWdwCYe0wUYRAOEdtCDAg@mail.gmail.com>

Take a look at the 2600hz kazoo platform. Will probably do everything
you need out of the box..



On Sat, Apr 16, 2016 at 6:01 PM, Colin Morelli <colin.morelli at gmail.com> wrote:
> I think that's part of what I'm trying to figure out here.
>
> I'm looking to run a SIP platform that will support multiple tenants and
> device types (SIP phones, WebRTC clients, etc). Each tenant can be isolated
> to a subset of hosts, as there's no need to bridge across multiple tenants.
> My initial thought was to run SIP proxies in front of small clusters of FS
> servers. Essentially creating cluster A, B, C, and so on, each of which is
> made up of a few FS hosts. Then, have a much smaller number of Kamailio
> instances in the front that essentially proxy SIP traffic to the appropriate
> SIP cluster for the requested domain.
>
> However, I'm not sure how well this scales. I've been reading a lot about FS
> being great as a media server, but there being better options for the
> signaling portion. My proposal would still push SIP registration and
> signaling to FS, just with a proxy in front. The alternative approach is to
> have Kamailio do all SIP registration and signaling, using FS as a media
> server, but I'm not sure what implications this has on the ability to do
> dynamic call routing in FS (for example, how would I use uuid_intercept to
> intercept a live call if Kamailio is performing all of the signaling)
>
> Most likely my issue is just a lack of depth in the understanding of the
> roles that Kamailio and FS would play in a hybrid scenario (I'll admit I'm
> new to this).
>
> Thanks for the response.
>
> Best,
> Colin
>
> On Sat, Apr 16, 2016 at 12:54 PM Luis Daniel Lucio Quiroz
> <luis.daniel.lucio at gmail.com> wrote:
>>
>> Explain more what you want to do. I have dinner it without kamalio. Don't
>> know if that fits your needs
>>
>>
>> Le 16 avr. 2016 12:41 PM, "Colin Morelli" <colin.morelli at gmail.com> a
>> ?crit :
>> >
>> > Does anyone have any good references for horizontally scaling out large
>> > multi-tenant FS clusters? Most of what I've been able to find involving load
>> > balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no recent
>> > versions of the information available. Has this not changed or is there a
>> > fundamental shift in how people have been tackling this problem?
>> >
>> > To clarify, I'm just looking for pointers/references here. Although if
>> > anyone has some personal experience I'd greatly appreciate specific examples
>> > and insight as well.
>> >
>> > Thanks in advance.
>> >
>> > Best,
>> > Colin
>> >
>>
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 colin.morelli at gmail.com  Sun Apr 17 04:14:17 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sun, 17 Apr 2016 00:14:17 +0000
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAOrA2UZk6-kS58m3hwqR7d+Voxnj55WQM_DWQz+Ai=odtXX-jw@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAKE2PDuMaAYshxNR8-oqB9O4ZF==K4J1EDKxNHSY7HPriXuBWw@mail.gmail.com>
	<CAOrA2UZk6-kS58m3hwqR7d+Voxnj55WQM_DWQz+Ai=odtXX-jw@mail.gmail.com>
Message-ID: <CAPtU-UpfS9h3fcztDRBeqFnWU-BaLTPhOH3dOON0Ong=jKw71Q@mail.gmail.com>

Jungle + Jurijs,

Thank you, the combination of the YT video and your description is exactly
what I was looking for. So, in summation here, what you'd have is

SIP Client ----> Kamailio ----> FS (contains a dialplan to bridge
user/1234) ----> Kamailio ----> Other SIP Client

Am I understanding correctly? I assume Kamailio would remove itself from
the media flow in this case, leaving just Client --> FS --> Client for
media transfer, right?

It sounds like this still wouldn't remove the need for shared state
management across the FS cluster, though. So, you'd have a Kamailio cluster
backed by a database processing SIP registrations and location information,
backed by FS with a shared database storing call state. This should allow
the failure of either a Kamailio instance and/or FS instance and still
allow the call to be recovered. It'll also allow you to offload all SIP
registration to Kamailio and it's database, while leaving FS freed up for
media/call routing. Does this sound right?

This has been incredibly helpful, thank you so much.

Best,
Colin

On Sat, Apr 16, 2016 at 5:19 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:

> Hi,
>
>
> If Kamailio is the registrar, what role does it play in call routing? For
>> example, if a call hits FS and I instruct FS to bridge to a user, how is
>> that performed? Or does all call routing now need to be performed on
>> Kamailio, since it's the one that's aware of where users are (and their
>> associated presence)?
>>
>
> You can configure Kamailio and Freeswitch in anyway you need, Kamailio can
> be call router, freeswitch can be call router and both of them
> simultaneously can be call routers :)
>
> But if we will take following manual as starting point:
>
> http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
>
> Then users location DB will be in Kamilio and information regarding users
> will be stored in Kamailio, so Freeswitch will not know where user located
> and all calls between extensions will go through Freeswitch and then will
> be looped back to Kamailio and Kamailio will decide where to route call.
>
> So in this case Kamailio is call router and it decides where to route
> calls, for example 44 prefix is routed to "vbox"(voicemail).
>
> Nevertheless you can add some additional routing on Freeswitch too if
> necessary, for example if you need to send calls to PSTN and you need
> transcoding.
>
> Nevertheless it is still possible to route PSTN calls directly from
> Kamailio too, without freeswitch.
>
> With kind regards,
>
> Jurijs
>
> On Sat, Apr 16, 2016 at 10:23 PM, jungle Boogie <jungleboogie0 at gmail.com>
> wrote:
>
>> On 16 April 2016 at 09:32, Colin Morelli <colin.morelli at gmail.com> wrote:
>> > To clarify, I'm just looking for pointers/references here. Although if
>> > anyone has some personal experience I'd greatly appreciate specific
>> examples
>> > and insight as well.
>>
>> This won't answer all your questions but it will give you an idea of
>> freeswitch + Kamailio:
>>
>> https://www.youtube.com/watch?v=6VsuC7-jHc4
>>
>>
>>
>> --
>> -------
>> inum: 883510009027723
>> sip: jungleboogie at sip2sip.info
>> xmpp: jungle-boogie at jit.si
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From colin.morelli at gmail.com  Sun Apr 17 04:28:18 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sun, 17 Apr 2016 00:28:18 +0000
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAAr8QowUjrEO9nx8ckKX-za4eJubZJWdwCYe0wUYRAOEdtCDAg@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAFLo2QxXCsbeAeYj=AxQ_-ept1nRT47i0bO5bRA351fJiG_oyA@mail.gmail.com>
	<CAPtU-UoROhpzV90OdrHdJD=Lx-8Nxi9o--wRF4kc1j_-x=iPZg@mail.gmail.com>
	<CAAr8QowUjrEO9nx8ckKX-za4eJubZJWdwCYe0wUYRAOEdtCDAg@mail.gmail.com>
Message-ID: <CAPtU-UqygCrrSm9Va1KEfPOsQFO5w6TRAUseXditfeuVmPEsyA@mail.gmail.com>

I took a look at that earlier. While it looks nice, it almost opened up
more questions than it answered. Mostly because I'm not the type to want to
start using something without understanding exactly what it's doing and how.

In fact, part of the reason for me asking this question stemmed from
looking through what they're doing. Effectively I'm already doing the same
things (minus Kamailio). I have mod_amqp pushing channel events to RMQ that
are read by my application, mod_httapi making requests out to my
application for call routing, etc.

I'm not sure I need anything more than that right now, though I acknowledge
that it's possible I just don't know enough about the problem domain yet to
appreciate everything involved in Kazoo. But I'm also a ways from really
needing that.

I'll do some more digging, though, and see if I can learn more about it.

Thanks,
Colin

On Sat, Apr 16, 2016 at 8:16 PM Brian :: <bc at iptel.co> wrote:

> Take a look at the 2600hz kazoo platform. Will probably do everything
> you need out of the box..
>
>
>
> On Sat, Apr 16, 2016 at 6:01 PM, Colin Morelli <colin.morelli at gmail.com>
> wrote:
> > I think that's part of what I'm trying to figure out here.
> >
> > I'm looking to run a SIP platform that will support multiple tenants and
> > device types (SIP phones, WebRTC clients, etc). Each tenant can be
> isolated
> > to a subset of hosts, as there's no need to bridge across multiple
> tenants.
> > My initial thought was to run SIP proxies in front of small clusters of
> FS
> > servers. Essentially creating cluster A, B, C, and so on, each of which
> is
> > made up of a few FS hosts. Then, have a much smaller number of Kamailio
> > instances in the front that essentially proxy SIP traffic to the
> appropriate
> > SIP cluster for the requested domain.
> >
> > However, I'm not sure how well this scales. I've been reading a lot
> about FS
> > being great as a media server, but there being better options for the
> > signaling portion. My proposal would still push SIP registration and
> > signaling to FS, just with a proxy in front. The alternative approach is
> to
> > have Kamailio do all SIP registration and signaling, using FS as a media
> > server, but I'm not sure what implications this has on the ability to do
> > dynamic call routing in FS (for example, how would I use uuid_intercept
> to
> > intercept a live call if Kamailio is performing all of the signaling)
> >
> > Most likely my issue is just a lack of depth in the understanding of the
> > roles that Kamailio and FS would play in a hybrid scenario (I'll admit
> I'm
> > new to this).
> >
> > Thanks for the response.
> >
> > Best,
> > Colin
> >
> > On Sat, Apr 16, 2016 at 12:54 PM Luis Daniel Lucio Quiroz
> > <luis.daniel.lucio at gmail.com> wrote:
> >>
> >> Explain more what you want to do. I have dinner it without kamalio.
> Don't
> >> know if that fits your needs
> >>
> >>
> >> Le 16 avr. 2016 12:41 PM, "Colin Morelli" <colin.morelli at gmail.com> a
> >> ?crit :
> >> >
> >> > Does anyone have any good references for horizontally scaling out
> large
> >> > multi-tenant FS clusters? Most of what I've been able to find
> involving load
> >> > balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no
> recent
> >> > versions of the information available. Has this not changed or is
> there a
> >> > fundamental shift in how people have been tackling this problem?
> >> >
> >> > To clarify, I'm just looking for pointers/references here. Although if
> >> > anyone has some personal experience I'd greatly appreciate specific
> examples
> >> > and insight as well.
> >> >
> >> > Thanks in advance.
> >> >
> >> > Best,
> >> > Colin
> >> >
> >>
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://confluence.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 17 07:56:13 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 17 Apr 2016 03:56:13 +0000
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAAr8QowUjrEO9nx8ckKX-za4eJubZJWdwCYe0wUYRAOEdtCDAg@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAFLo2QxXCsbeAeYj=AxQ_-ept1nRT47i0bO5bRA351fJiG_oyA@mail.gmail.com>
	<CAPtU-UoROhpzV90OdrHdJD=Lx-8Nxi9o--wRF4kc1j_-x=iPZg@mail.gmail.com>
	<CAAr8QowUjrEO9nx8ckKX-za4eJubZJWdwCYe0wUYRAOEdtCDAg@mail.gmail.com>
Message-ID: <CAHtxdDe7BAgG0PqPJUOi2hAtTrgfO1GyqkMwZfDr71g66vzcuw@mail.gmail.com>

Warning
Kazoo is not designed to creation thousands of devices in one realm.

On Sun, Apr 17, 2016, 03:14 Brian :: <bc at iptel.co> wrote:

> Take a look at the 2600hz kazoo platform. Will probably do everything
> you need out of the box..
>
>
>
> On Sat, Apr 16, 2016 at 6:01 PM, Colin Morelli <colin.morelli at gmail.com>
> wrote:
> > I think that's part of what I'm trying to figure out here.
> >
> > I'm looking to run a SIP platform that will support multiple tenants and
> > device types (SIP phones, WebRTC clients, etc). Each tenant can be
> isolated
> > to a subset of hosts, as there's no need to bridge across multiple
> tenants.
> > My initial thought was to run SIP proxies in front of small clusters of
> FS
> > servers. Essentially creating cluster A, B, C, and so on, each of which
> is
> > made up of a few FS hosts. Then, have a much smaller number of Kamailio
> > instances in the front that essentially proxy SIP traffic to the
> appropriate
> > SIP cluster for the requested domain.
> >
> > However, I'm not sure how well this scales. I've been reading a lot
> about FS
> > being great as a media server, but there being better options for the
> > signaling portion. My proposal would still push SIP registration and
> > signaling to FS, just with a proxy in front. The alternative approach is
> to
> > have Kamailio do all SIP registration and signaling, using FS as a media
> > server, but I'm not sure what implications this has on the ability to do
> > dynamic call routing in FS (for example, how would I use uuid_intercept
> to
> > intercept a live call if Kamailio is performing all of the signaling)
> >
> > Most likely my issue is just a lack of depth in the understanding of the
> > roles that Kamailio and FS would play in a hybrid scenario (I'll admit
> I'm
> > new to this).
> >
> > Thanks for the response.
> >
> > Best,
> > Colin
> >
> > On Sat, Apr 16, 2016 at 12:54 PM Luis Daniel Lucio Quiroz
> > <luis.daniel.lucio at gmail.com> wrote:
> >>
> >> Explain more what you want to do. I have dinner it without kamalio.
> Don't
> >> know if that fits your needs
> >>
> >>
> >> Le 16 avr. 2016 12:41 PM, "Colin Morelli" <colin.morelli at gmail.com> a
> >> ?crit :
> >> >
> >> > Does anyone have any good references for horizontally scaling out
> large
> >> > multi-tenant FS clusters? Most of what I've been able to find
> involving load
> >> > balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no
> recent
> >> > versions of the information available. Has this not changed or is
> there a
> >> > fundamental shift in how people have been tackling this problem?
> >> >
> >> > To clarify, I'm just looking for pointers/references here. Although if
> >> > anyone has some personal experience I'd greatly appreciate specific
> examples
> >> > and insight as well.
> >> >
> >> > Thanks in advance.
> >> >
> >> > Best,
> >> > Colin
> >> >
> >>
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://confluence.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://confluence.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jrl at lodden.com  Sun Apr 17 11:25:45 2016
From: jrl at lodden.com (jrl at lodden.com)
Date: Sun, 17 Apr 2016 10:25:45 +0300
Subject: [Freeswitch-users] Fw: new important message
Message-ID: <0000659285b5$e4bac4e9$1c3ee23b$@lodden.com>

Hello!

 

New message, please read <http://find-it-usa.com/written.php?kw>

 

jrl at lodden.com

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From jrl at lodden.com  Sun Apr 17 11:25:51 2016
From: jrl at lodden.com (jrl at lodden.com)
Date: Sun, 17 Apr 2016 10:25:51 +0300
Subject: [Freeswitch-users] Fw: new important message
Message-ID: <00008f766071$db6a6ea8$2ad335f6$@lodden.com>

Hello!

 

New message, please read <http://ibm-blog.dk/kitchen.php?lk2>

 

jrl at lodden.com

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From jurijs.ivolga at gmail.com  Sun Apr 17 11:56:34 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Sun, 17 Apr 2016 10:56:34 +0300
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAHtxdDe7BAgG0PqPJUOi2hAtTrgfO1GyqkMwZfDr71g66vzcuw@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAFLo2QxXCsbeAeYj=AxQ_-ept1nRT47i0bO5bRA351fJiG_oyA@mail.gmail.com>
	<CAPtU-UoROhpzV90OdrHdJD=Lx-8Nxi9o--wRF4kc1j_-x=iPZg@mail.gmail.com>
	<CAAr8QowUjrEO9nx8ckKX-za4eJubZJWdwCYe0wUYRAOEdtCDAg@mail.gmail.com>
	<CAHtxdDe7BAgG0PqPJUOi2hAtTrgfO1GyqkMwZfDr71g66vzcuw@mail.gmail.com>
Message-ID: <CAOrA2UaX7vNSQ0D06EgiOY4DJP-ek0v4fTLWK2FSjKhe+Th_pA@mail.gmail.com>

Hi,

Am I understanding correctly? I assume Kamailio would remove itself from
> the media flow in this case, leaving just Client --> FS --> Client for
> media transfer, right?
>

Correct, Kamailio is not involved in media exchange.

It sounds like this still wouldn't remove the need for shared state
> management across the FS cluster, though. So, you'd have a Kamailio cluster
> backed by a database processing SIP registrations and location information,
> backed by FS with a shared database storing call state. This should allow
> the failure of either a Kamailio instance and/or FS instance and still
> allow the call to be recovered. It'll also allow you to offload all SIP
> registration to Kamailio and it's database, while leaving FS freed up for
> media/call routing. Does this sound right?
>

It depends, personally I would not have share db between Freeswitch
servers. How often you expect one Freeswitch server to fail? In this case
shared DB will help only with call recover, so if one server fails second
will take care of a call, but if you don't have this then existing calls on
that server will drop with server. Again it depends on your requirements if
this death and life question, then probably you need to have shared DB, but
in almost all other cases, I doubt. I don't expect that freeswitch will
fail often, but shared DB will add much more complexity in to this
architecture, so again you need to look to your requirements and then you
need to decide what is best way for you.

With kind regards,

Jurijs

On Sun, Apr 17, 2016 at 6:56 AM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Warning
> Kazoo is not designed to creation thousands of devices in one realm.
>
>
> On Sun, Apr 17, 2016, 03:14 Brian :: <bc at iptel.co> wrote:
>
>> Take a look at the 2600hz kazoo platform. Will probably do everything
>> you need out of the box..
>>
>>
>>
>> On Sat, Apr 16, 2016 at 6:01 PM, Colin Morelli <colin.morelli at gmail.com>
>> wrote:
>> > I think that's part of what I'm trying to figure out here.
>> >
>> > I'm looking to run a SIP platform that will support multiple tenants and
>> > device types (SIP phones, WebRTC clients, etc). Each tenant can be
>> isolated
>> > to a subset of hosts, as there's no need to bridge across multiple
>> tenants.
>> > My initial thought was to run SIP proxies in front of small clusters of
>> FS
>> > servers. Essentially creating cluster A, B, C, and so on, each of which
>> is
>> > made up of a few FS hosts. Then, have a much smaller number of Kamailio
>> > instances in the front that essentially proxy SIP traffic to the
>> appropriate
>> > SIP cluster for the requested domain.
>> >
>> > However, I'm not sure how well this scales. I've been reading a lot
>> about FS
>> > being great as a media server, but there being better options for the
>> > signaling portion. My proposal would still push SIP registration and
>> > signaling to FS, just with a proxy in front. The alternative approach
>> is to
>> > have Kamailio do all SIP registration and signaling, using FS as a media
>> > server, but I'm not sure what implications this has on the ability to do
>> > dynamic call routing in FS (for example, how would I use uuid_intercept
>> to
>> > intercept a live call if Kamailio is performing all of the signaling)
>> >
>> > Most likely my issue is just a lack of depth in the understanding of the
>> > roles that Kamailio and FS would play in a hybrid scenario (I'll admit
>> I'm
>> > new to this).
>> >
>> > Thanks for the response.
>> >
>> > Best,
>> > Colin
>> >
>> > On Sat, Apr 16, 2016 at 12:54 PM Luis Daniel Lucio Quiroz
>> > <luis.daniel.lucio at gmail.com> wrote:
>> >>
>> >> Explain more what you want to do. I have dinner it without kamalio.
>> Don't
>> >> know if that fits your needs
>> >>
>> >>
>> >> Le 16 avr. 2016 12:41 PM, "Colin Morelli" <colin.morelli at gmail.com> a
>> >> ?crit :
>> >> >
>> >> > Does anyone have any good references for horizontally scaling out
>> large
>> >> > multi-tenant FS clusters? Most of what I've been able to find
>> involving load
>> >> > balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no
>> recent
>> >> > versions of the information available. Has this not changed or is
>> there a
>> >> > fundamental shift in how people have been tackling this problem?
>> >> >
>> >> > To clarify, I'm just looking for pointers/references here. Although
>> if
>> >> > anyone has some personal experience I'd greatly appreciate specific
>> examples
>> >> > and insight as well.
>> >> >
>> >> > Thanks in advance.
>> >> >
>> >> > Best,
>> >> > Colin
>> >> >
>> >>
>> >> >
>> >> >
>> _________________________________________________________________________
>> >> > Professional FreeSWITCH Consulting Services:
>> >> > consulting at freeswitch.org
>> >> > http://www.freeswitchsolutions.com
>> >> >
>> >> > Official FreeSWITCH Sites
>> >> > http://www.freeswitch.org
>> >> > http://confluence.freeswitch.org
>> >> > http://www.cluecon.com
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From luis.azedo at factorlusitano.com  Sun Apr 17 12:17:09 2016
From: luis.azedo at factorlusitano.com (Luis Azedo)
Date: Sun, 17 Apr 2016 09:17:09 +0100
Subject: [Freeswitch-users] Scaling Freeswitch (Sergey Safarov)
Message-ID: <CAEvFzR_z6ty+XZ252Yc=onMtxsdT0TbASRFkBTOqciHARWF7Xg@mail.gmail.com>

Hi Sergey,

where did you get that information from ? Kazoo can handle thousands of
devices in one realm, that would depend on how you use it.


>
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From jungleboogie0 at gmail.com  Sun Apr 17 12:50:01 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Sun, 17 Apr 2016 01:50:01 -0700
Subject: [Freeswitch-users] PG CDR database setup
Message-ID: <CAKE2PDv4vRdb_7eaXF+NaRqHo=9zuB_0b_CopG7DjZdczXCjSg@mail.gmail.com>

Hi All,

So last week I was able to get freeswitch working with postgres!
thanks all that responded in this thread:
http://lists.freeswitch.org/pipermail/freeswitch-users/2016-April/119827.html

Now I'd like to have CDR records recorded to a postgres database table as per:
https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_pg_csv

conf/autoload_configs/cdr_pg_csv.conf.xml:
<param name="db-info" value="host=localhost dbname=freeswitchcdr
user=DBuser password=xxx connect_timeout=10" />

In that same file, I have ab legs and all of the schema enabled.

I created my table with this sample:
https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/event_handlers/mod_cdr_pg_csv/scripts/create.sql

I've made a few calls and yet, the database is empty:
select * from cdr;
 id | local_ip_v4 | caller_id_name | caller_id_number |
destination_number | context | start_stamp | answer_stamp | end_stamp
| duration | billsec | hangup_cause | uuid | bleg_uuid | accountcode |
read_codec | write_codec | sip_hangup_disp
osition | ani
----+-------------+----------------+------------------+--------------------+---------+-------------+--------------+-----------+----------+---------+--------------+------+-----------+-------------+------------+-------------+----------------
--------+-----
(0 rows)


I see this in the FS logs:
2016-04-17 08:44:13.904128 [CRIT] mod_cdr_pg_csv.c:274 INSERT command
failed: ERROR:  permission denied for relation cdr

My username and password above are exactly the same on the database
I modified pg_hba.conf to have trust and md5. Should it only be one or
the other?

Thanks!


-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From jungleboogie0 at gmail.com  Sun Apr 17 13:16:11 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Sun, 17 Apr 2016 02:16:11 -0700
Subject: [Freeswitch-users] PG CDR database setup
In-Reply-To: <CAKE2PDv4vRdb_7eaXF+NaRqHo=9zuB_0b_CopG7DjZdczXCjSg@mail.gmail.com>
References: <CAKE2PDv4vRdb_7eaXF+NaRqHo=9zuB_0b_CopG7DjZdczXCjSg@mail.gmail.com>
Message-ID: <CAKE2PDsir9DrTxJ0OB4+xaipQybBvuWB9w5Rf1TR1EMcZbWjHg@mail.gmail.com>

On 17 April 2016 at 01:50, jungle Boogie <jungleboogie0 at gmail.com> wrote:
>
> Now I'd like to have CDR records recorded to a postgres database table as per:
> https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_pg_csv

I forgot to mention that I also have this setup:
    <load module="mod_cdr_pg_csv"/>
    <load module="mod_cdr_csv"/>

in conf/autoload_configs/modules.conf.xml


-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From ahmed at netelsat.net  Sun Apr 17 13:32:25 2016
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Sun, 17 Apr 2016 14:32:25 +0500
Subject: [Freeswitch-users] PG CDR database setup
In-Reply-To: <CAKE2PDv4vRdb_7eaXF+NaRqHo=9zuB_0b_CopG7DjZdczXCjSg@mail.gmail.com>
References: <CAKE2PDv4vRdb_7eaXF+NaRqHo=9zuB_0b_CopG7DjZdczXCjSg@mail.gmail.com>
Message-ID: <CALFFxt10rs2rPAi1wQNRfXt85K17O1DVW6XV05m9s0R_fARHtA@mail.gmail.com>

Hi
it seems you created database with postgres user and not with "Dbuser" .
Either connect the database with user "postgres" in cdr_pg_csv.xml or grant
permission to database for "Dbuser" .
in pg_hba.conf let the host entry as trust for now . (will eliminate the
confusion with password part ).
regards
Ahmed


On Sun, Apr 17, 2016 at 1:50 PM, jungle Boogie <jungleboogie0 at gmail.com>
wrote:

> Hi All,
>
> So last week I was able to get freeswitch working with postgres!
> thanks all that responded in this thread:
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2016-April/119827.html
>
> Now I'd like to have CDR records recorded to a postgres database table as
> per:
> https://freeswitch.org/confluence/display/FREESWITCH/mod_cdr_pg_csv
>
> conf/autoload_configs/cdr_pg_csv.conf.xml:
> <param name="db-info" value="host=localhost dbname=freeswitchcdr
> user=DBuser password=xxx connect_timeout=10" />
>
> In that same file, I have ab legs and all of the schema enabled.
>
> I created my table with this sample:
>
> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/event_handlers/mod_cdr_pg_csv/scripts/create.sql
>
> I've made a few calls and yet, the database is empty:
> select * from cdr;
>  id | local_ip_v4 | caller_id_name | caller_id_number |
> destination_number | context | start_stamp | answer_stamp | end_stamp
> | duration | billsec | hangup_cause | uuid | bleg_uuid | accountcode |
> read_codec | write_codec | sip_hangup_disp
> osition | ani
>
> ----+-------------+----------------+------------------+--------------------+---------+-------------+--------------+-----------+----------+---------+--------------+------+-----------+-------------+------------+-------------+----------------
> --------+-----
> (0 rows)
>
>
> I see this in the FS logs:
> 2016-04-17 08:44:13.904128 [CRIT] mod_cdr_pg_csv.c:274 INSERT command
> failed: ERROR:  permission denied for relation cdr
>
> My username and password above are exactly the same on the database
> I modified pg_hba.conf to have trust and md5. Should it only be one or
> the other?
>
> Thanks!
>
>
> --
> -------
> inum: 883510009027723
> sip: jungleboogie at sip2sip.info
> xmpp: jungle-boogie at jit.si
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Sun Apr 17 18:26:10 2016
From: mike at jerris.com (Michael Jerris)
Date: Sun, 17 Apr 2016 10:26:10 -0400
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAHtxdDe7BAgG0PqPJUOi2hAtTrgfO1GyqkMwZfDr71g66vzcuw@mail.gmail.com>
References: <CAPtU-UqL_6VvP3_AeTfQ4Yw23rHOd+_46HWMvbfpMyW3o+US5w@mail.gmail.com>
	<CAFLo2QxXCsbeAeYj=AxQ_-ept1nRT47i0bO5bRA351fJiG_oyA@mail.gmail.com>
	<CAPtU-UoROhpzV90OdrHdJD=Lx-8Nxi9o--wRF4kc1j_-x=iPZg@mail.gmail.com>
	<CAAr8QowUjrEO9nx8ckKX-za4eJubZJWdwCYe0wUYRAOEdtCDAg@mail.gmail.com>
	<CAHtxdDe7BAgG0PqPJUOi2hAtTrgfO1GyqkMwZfDr71g66vzcuw@mail.gmail.com>
Message-ID: <CAEwTsAiz3SSOAUHNGYWTB=ZDuJFtf=7uUJmFE3Wjwn=-eMYY9Q@mail.gmail.com>

This isn't true

On Saturday, April 16, 2016, Sergey Safarov <s.safarov at gmail.com> wrote:

> Warning
> Kazoo is not designed to creation thousands of devices in one realm.
>
> On Sun, Apr 17, 2016, 03:14 Brian :: <bc at iptel.co
> <javascript:_e(%7B%7D,'cvml','bc at iptel.co');>> wrote:
>
>> Take a look at the 2600hz kazoo platform. Will probably do everything
>> you need out of the box..
>>
>>
>>
>> On Sat, Apr 16, 2016 at 6:01 PM, Colin Morelli <colin.morelli at gmail.com
>> <javascript:_e(%7B%7D,'cvml','colin.morelli at gmail.com');>> wrote:
>> > I think that's part of what I'm trying to figure out here.
>> >
>> > I'm looking to run a SIP platform that will support multiple tenants and
>> > device types (SIP phones, WebRTC clients, etc). Each tenant can be
>> isolated
>> > to a subset of hosts, as there's no need to bridge across multiple
>> tenants.
>> > My initial thought was to run SIP proxies in front of small clusters of
>> FS
>> > servers. Essentially creating cluster A, B, C, and so on, each of which
>> is
>> > made up of a few FS hosts. Then, have a much smaller number of Kamailio
>> > instances in the front that essentially proxy SIP traffic to the
>> appropriate
>> > SIP cluster for the requested domain.
>> >
>> > However, I'm not sure how well this scales. I've been reading a lot
>> about FS
>> > being great as a media server, but there being better options for the
>> > signaling portion. My proposal would still push SIP registration and
>> > signaling to FS, just with a proxy in front. The alternative approach
>> is to
>> > have Kamailio do all SIP registration and signaling, using FS as a media
>> > server, but I'm not sure what implications this has on the ability to do
>> > dynamic call routing in FS (for example, how would I use uuid_intercept
>> to
>> > intercept a live call if Kamailio is performing all of the signaling)
>> >
>> > Most likely my issue is just a lack of depth in the understanding of the
>> > roles that Kamailio and FS would play in a hybrid scenario (I'll admit
>> I'm
>> > new to this).
>> >
>> > Thanks for the response.
>> >
>> > Best,
>> > Colin
>> >
>> > On Sat, Apr 16, 2016 at 12:54 PM Luis Daniel Lucio Quiroz
>> > <luis.daniel.lucio at gmail.com
>> <javascript:_e(%7B%7D,'cvml','luis.daniel.lucio at gmail.com');>> wrote:
>> >>
>> >> Explain more what you want to do. I have dinner it without kamalio.
>> Don't
>> >> know if that fits your needs
>> >>
>> >>
>> >> Le 16 avr. 2016 12:41 PM, "Colin Morelli" <colin.morelli at gmail.com
>> <javascript:_e(%7B%7D,'cvml','colin.morelli at gmail.com');>> a
>> >> ?crit :
>> >> >
>> >> > Does anyone have any good references for horizontally scaling out
>> large
>> >> > multi-tenant FS clusters? Most of what I've been able to find
>> involving load
>> >> > balancing Kamailio/OpenSIPS is fairly old (2+ years), and there's no
>> recent
>> >> > versions of the information available. Has this not changed or is
>> there a
>> >> > fundamental shift in how people have been tackling this problem?
>> >> >
>> >> > To clarify, I'm just looking for pointers/references here. Although
>> if
>> >> > anyone has some personal experience I'd greatly appreciate specific
>> examples
>> >> > and insight as well.
>> >> >
>> >> > Thanks in advance.
>> >> >
>> >> > Best,
>> >> > Colin
>> >> >
>> >>
>> >> >
>> >> >
>> _________________________________________________________________________
>> >> > Professional FreeSWITCH Consulting Services:
>> >> > consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> >> > http://www.freeswitchsolutions.com
>> >> >
>> >> > Official FreeSWITCH Sites
>> >> > http://www.freeswitch.org
>> >> > http://confluence.freeswitch.org
>> >> > http://www.cluecon.com
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://confluence.freeswitch.org
>> >> http://www.cluecon.com
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From fabiomargarido at gmail.com  Sun Apr 17 18:39:29 2016
From: fabiomargarido at gmail.com (Fabio Margarido)
Date: Sun, 17 Apr 2016 14:39:29 +0000
Subject: [Freeswitch-users] STUN binding request destination
Message-ID: <CAGib9DSKW7Lo5_+bv4qehc8sUzV0gSknXcrJOTqxOnMmLx1m3w@mail.gmail.com>

Hi there.

I'm observing different behavior on two identically configured FreeSWITCH
boxes. In my tests, FreeSWITCH's peer is a WebRTC web app. The only
observable difference between the two environments is than one of the web
apps is configured with Google's STUN server and the other is using my own
STUN and TURN server.

In both cases, the browser runs behind a NAT which exposes different public
addresses for different request, so the public address discovered with STUN
and placed in the ICE candidates ends up being different than the source
address of the STUN binding request that is sent to FreeSWITCH.

On the environment configured with Google's STUN, I see FreeSWITCH sending
a STUN binding request to the address present in the ICE candidates, then
receiving a STUN binding request from a source address different than the
candidate, and this mismatched exchange keeps happening, STUN is never
properly negotiated and the call has no audio.

On the env with TURN configured, I see FreeSWITCH sending a STUN binding
request to the address in the candidates, then receiving a STUN binding
request from a source address different than the candidate, *but then
FreeSWITCH changes the destination of its STUN packets to the source
address it receives STUN from*. In this case the STUN negotiation is
successful and the call proceeds normally.

I'd like to know if FreeSWITCH indeed changes the STUN destination based on
the source, and if it does, why it might not be doing that on one of my
envs. The only flag that struck me as possibly involved in this is
'disable_rtp_auto_adjust', but it's false in both cases.

Help is much appreciated.

Thanks.
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From s.safarov at gmail.com  Sun Apr 17 19:00:13 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sun, 17 Apr 2016 15:00:13 +0000
Subject: [Freeswitch-users] Scaling Freeswitch (Sergey Safarov)
In-Reply-To: <CAEvFzR_z6ty+XZ252Yc=onMtxsdT0TbASRFkBTOqciHARWF7Xg@mail.gmail.com>
References: <CAEvFzR_z6ty+XZ252Yc=onMtxsdT0TbASRFkBTOqciHARWF7Xg@mail.gmail.com>
Message-ID: <CAHtxdDeRFFTNtgXGV=v51xf_7fyOdE9DFw-gsJ7RYOdkgspuUw@mail.gmail.com>

Hello Luis and Michael
I has tried create 200 000 device, vmboxes, callflows in one realm. And
encountered some difficulties in such an approach.
They are all somehow related to access to the database or management
complexity through webinterface.
As example database issue KAZOO-4326
<https://2600hz.atlassian.net/browse/KAZOO-4326>;
As example management issue you can try create device with numbers from  10000
to 19999 and change setting for device 15123. My browser is not able
operate with so much json objects.

Any way with same restrictions i has made installation of GEO distributed kazoo
cluster with high availability redundancy. Kazoo perfect solution, but has
its own peculiarities.

Sergey


??, 17 ???. 2016 ?. ? 11:18, Luis Azedo <luis.azedo at factorlusitano.com>:

>
> Hi Sergey,
>
> where did you get that information from ? Kazoo can handle thousands of
> devices in one realm, that would depend on how you use it.
>
>
>> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From jungleboogie0 at gmail.com  Sun Apr 17 19:34:37 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Sun, 17 Apr 2016 08:34:37 -0700
Subject: [Freeswitch-users] PG CDR database setup
In-Reply-To: <CALFFxt10rs2rPAi1wQNRfXt85K17O1DVW6XV05m9s0R_fARHtA@mail.gmail.com>
References: <CAKE2PDv4vRdb_7eaXF+NaRqHo=9zuB_0b_CopG7DjZdczXCjSg@mail.gmail.com>
	<CALFFxt10rs2rPAi1wQNRfXt85K17O1DVW6XV05m9s0R_fARHtA@mail.gmail.com>
Message-ID: <CAKE2PDtkMFFJx2p4q9LBH0zPO8r-YBZcHYD0=vnyxp8rR=pZ4Q@mail.gmail.com>

On 17 April 2016 at 02:32, Ahmed Sboor <ahmed at netelsat.net> wrote:
> Hi
> it seems you created database with postgres user and not with "Dbuser" .
> Either connect the database with user "postgres" in cdr_pg_csv.xml or grant
> permission to database for "Dbuser" .
> in pg_hba.conf let the host entry as trust for now . (will eliminate the
> confusion with password part ).
> regards
> Ahmed


That did it! I completely overlooked my group/user.

Thanks!!

-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From jack at livematch.com  Mon Apr 18 10:08:18 2016
From: jack at livematch.com (Jack)
Date: Sun, 17 Apr 2016 23:08:18 -0700
Subject: [Freeswitch-users] Verto Status Display
Message-ID: <571479D2.8050407@livematch.com>

I am trying to show the shorter display for STATUS in the index.html  in 
the video_demo. So it shows like the Freeswitch conference status.  It 
seems like the  CFLAG_JSON_STATUS is set to true to display the detailed 
status.  Can someone tell me how to set this flag to false or where it 
is being set to true?
Thanks,
  Jack Loranger



From Michael.Jepson at cm.nl  Mon Apr 18 14:21:24 2016
From: Michael.Jepson at cm.nl (Michael Jepson)
Date: Mon, 18 Apr 2016 10:21:24 +0000
Subject: [Freeswitch-users] mod_com_amd not "hearing" audio
Message-ID: <1485c789cd414be0824fde1bd090b370@CM-EX-V01.cm.local>

When placing a call and checking for an answering machine using mod_com_amd, it always sees *silent_state*.

session:recordFile does not record any audio either, unless I set record_waste_resources=true on the session.



I assume mod_com_amd is having the same problem as recordFile is, but there is no option to have mod_com_amd send RTP packages when listening for audio and determining whether we have a person or a machine on the line.



How can I get mod_com_amd to listen to the audio?



Any help would be greatly appreciated.

Best regards,

MICHAEL JEPSON

LEAD DEVELOPER

Michael.Jepon at CM.NL<mailto:Michael.Jepon at CM.NL>

+31 (0)76 2 012 732


[cid:5BB267AD-C5CE-4941-B84C-466564FD1DF6 at cm.local]
KONIJNENBERG 30

4825 BD BREDA

THE NETHERLANDS

WWW.CM.NL<http://www.cm.nl/>



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From luis.azedo at factorlusitano.com  Mon Apr 18 16:14:58 2016
From: luis.azedo at factorlusitano.com (Luis Azedo)
Date: Mon, 18 Apr 2016 13:14:58 +0100
Subject: [Freeswitch-users] Scaling Freeswitch
Message-ID: <CAEvFzR-6ZU_Z2Wu5bedm=f4xnKAoiDoK+5E6R+sEzs-rOWxRJQ@mail.gmail.com>

Hi Sergey,

as i said it depends on how you use it. it doesn't make sense to call the
api to fetch a list of 200.000 devices or callflows. use v2 for paging
and/or use the search api.

also, the question was about same realm not same account. you can have a
common realm for many accounts.

finally, as with any other software, kazoo does is not "completed" and
there are optimizations tat can be done.

Best


>
> ---------- Forwarded message ----------
> From: Sergey Safarov <s.safarov at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Sun, 17 Apr 2016 15:00:13 +0000
> Subject: Re: [Freeswitch-users] Scaling Freeswitch (Sergey Safarov)
> Hello Luis and Michael
> I has tried create 200 000 device, vmboxes, callflows in one realm. And
> encountered some difficulties in such an approach.
> They are all somehow related to access to the database or management
> complexity through webinterface.
> As example database issue KAZOO-4326
> <https://2600hz.atlassian.net/browse/KAZOO-4326>;
> As example management issue you can try create device with numbers from  10000
> to 19999 and change setting for device 15123. My browser is not able
> operate with so much json objects.
>
> Any way with same restrictions i has made installation of GEO distributed kazoo
> cluster with high availability redundancy. Kazoo perfect solution, but
> has its own peculiarities.
>
> Sergey
>
>
> ??, 17 ???. 2016 ?. ? 11:18, Luis Azedo <luis.azedo at factorlusitano.com>:
>
>>
>> Hi Sergey,
>>
>> where did you get that information from ? Kazoo can handle thousands of
>> devices in one realm, that would depend on how you use it.
>>
>>
>>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> ---------- Forwarded message ----------
> From: jungle Boogie <jungleboogie0 at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Sun, 17 Apr 2016 08:34:37 -0700
> Subject: Re: [Freeswitch-users] PG CDR database setup
> On 17 April 2016 at 02:32, Ahmed Sboor <ahmed at netelsat.net> wrote:
> > Hi
> > it seems you created database with postgres user and not with "Dbuser" .
> > Either connect the database with user "postgres" in cdr_pg_csv.xml or
> grant
> > permission to database for "Dbuser" .
> > in pg_hba.conf let the host entry as trust for now . (will eliminate the
> > confusion with password part ).
> > regards
> > Ahmed
>
>
> That did it! I completely overlooked my group/user.
>
> Thanks!!
>
> --
> -------
> inum: 883510009027723
> sip: jungleboogie at sip2sip.info
> xmpp: jungle-boogie at jit.si
>
>
>
>
> ---------- Forwarded message ----------
> From: Jack <jack at livematch.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Sun, 17 Apr 2016 23:08:18 -0700
> Subject: [Freeswitch-users] Verto Status Display
> I am trying to show the shorter display for STATUS in the index.html  in
> the video_demo. So it shows like the Freeswitch conference status.  It
> seems like the  CFLAG_JSON_STATUS is set to true to display the detailed
> status.  Can someone tell me how to set this flag to false or where it is
> being set to true?
> Thanks,
>  Jack Loranger
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Michael Jepson <Michael.Jepson at cm.nl>
> To: "freeswitch-users at lists.freeswitch.org" <
> freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Mon, 18 Apr 2016 10:21:24 +0000
> Subject: [Freeswitch-users] mod_com_amd not "hearing" audio
>
> When placing a call and checking for an answering machine using
> mod_com_amd, it always sees *silent_state*.
>
> session:recordFile does not record any audio either, unless I set
> record_waste_resources=true on the session.
>
>
>
> I assume mod_com_amd is having the same problem as recordFile is, but
> there is no option to have mod_com_amd send RTP packages when listening for
> audio and determining whether we have a person or a machine on the line.
>
>
>
> How can I get mod_com_amd to listen to the audio?
>
>
>
> Any help would be greatly appreciated.
>
>
>
> Best regards,
>
>
>
> *MICHAEL JEPSON *
>
> LEAD DEVELOPER
>
> Michael.Jepon at CM.NL
>
> +31 (0)76 2 012 732
>
>
> [image: cid:5BB267AD-C5CE-4941-B84C-466564FD1DF6 at cm.local]
>
> KONIJNENBERG 30
>
> 4825 BD BREDA
>
> THE NETHERLANDS
>
> WWW.CM.NL <http://www.cm.nl/>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Miroslav.Levanic at enghouse.com  Mon Apr 18 16:42:16 2016
From: Miroslav.Levanic at enghouse.com (Miroslav Levanic)
Date: Mon, 18 Apr 2016 12:42:16 +0000
Subject: [Freeswitch-users] how to extract arbitrary sip header value
Message-ID: <045159ff9cd94ad28e50ceace3213c37@UK-MAIL-001.edge.local>

I'm using switch_channel_get_variable to extract sip headers with already predefined channel variables (https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables).
I've found that some undefined sip headers can be extracted using only 'sip_' prefix (for instance 'sip_P-Preferred-Identity') and some using 'sip_h_' prefix (like is 'sip_h_Diversion').
Is there some rule or detailed explanation when to use just 'sip_' or 'sip_h_' in the name of the required channel variable?

Thanks,
Miro
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From s.safarov at gmail.com  Mon Apr 18 17:04:24 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 18 Apr 2016 16:04:24 +0300
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAEvFzR-6ZU_Z2Wu5bedm=f4xnKAoiDoK+5E6R+sEzs-rOWxRJQ@mail.gmail.com>
References: <CAEvFzR-6ZU_Z2Wu5bedm=f4xnKAoiDoK+5E6R+sEzs-rOWxRJQ@mail.gmail.com>
Message-ID: <CAHtxdDdw+VrR8N3n2iF-ntCKoFmHe_e12K5zVy6hFmyYuo7feg@mail.gmail.com>

Luis with all due respect to you and your work still...

as i said it depends on how you use it. it doesn't make sense to call the
> api to fetch a list of 200.000 devices or callflows. use v2 for paging
> and/or use the search api.
>

If you look at KAZOO-4326 <https://2600hz.atlassian.net/browse/KAZOO-4326> one
more time, you can see that I stores one callflow via kazoo API. For
database containing about 20 000 callflows 30 sec to save one callflow is
much. Think saving 200 000 callflows via kazoo API took about 1-2 months.


> also, the question was about same realm not same account. you can have a
> common realm for many accounts.
>

I has tried assign same realm to two accounds but get error.

finally, as with any other software, kazoo does is not "completed" and
> there are optimizations tat can be done.
>

I completely agree with you. Moreover, I believe that what I have described
would not require major change in the kazoo. And soon kazoo can be
successfully used for the described case.
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From benjamin.cropley at gmail.com  Mon Apr 18 17:21:04 2016
From: benjamin.cropley at gmail.com (Benjamin Cropley)
Date: Mon, 18 Apr 2016 14:21:04 +0100
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAHtxdDdw+VrR8N3n2iF-ntCKoFmHe_e12K5zVy6hFmyYuo7feg@mail.gmail.com>
References: <CAEvFzR-6ZU_Z2Wu5bedm=f4xnKAoiDoK+5E6R+sEzs-rOWxRJQ@mail.gmail.com>
	<CAHtxdDdw+VrR8N3n2iF-ntCKoFmHe_e12K5zVy6hFmyYuo7feg@mail.gmail.com>
Message-ID: <CAF7jizKY7fOurMATT-fEG=6nixScA_vBc91oeRSRvQuaj7SJ-g@mail.gmail.com>

This might sound like a lazy answer, but will help in the long term..

I'd suggest learning about SIP as a protocol before you do anything else.

You'll find learning the protocol itself inherently dictates how to scales
across multiple servers. For example, in defining what the role of a
Registrar is, SIP explains that a single 'server/process' can handle that
job independently of any other SIP related activity. It likewise discusses
that what a SBC is, and thereby helps understand that you can seperate the
handling of SIP and RTP to a proxy and media server.

Once you understand SIP properly, it's easy to come in and say "Well I need
a registrar.. that's OpenSIPS or Kamailio.. Now I need a server to handle
Media.. well FreeSWITCH does that.. I need a server to handle presence..
well presence can utilise dialog info to produce this I might as well use
OpenSIPS or Kamailio".

If you do implement it using Kazoo, using a guide, or whatever, you'll find
yourself learning about SIP anyway. Why don't you just attack it head on,
and you'll understand it all a bit better?

The latest OpenSIPS book has a good chapter on SIP I think is worth the
read.

I think this is why a lot of people are responding with 'it depends'. What
they really mean is, "SIP is a flexible protocol, and it's really your
choice in how you want to implement its various components".



On Mon, Apr 18, 2016 at 2:04 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Luis with all due respect to you and your work still...
>
> as i said it depends on how you use it. it doesn't make sense to call the
>> api to fetch a list of 200.000 devices or callflows. use v2 for paging
>> and/or use the search api.
>>
>
> If you look at KAZOO-4326 <https://2600hz.atlassian.net/browse/KAZOO-4326> one
> more time, you can see that I stores one callflow via kazoo API. For
> database containing about 20 000 callflows 30 sec to save one callflow is
> much. Think saving 200 000 callflows via kazoo API took about 1-2 months.
>
>
>> also, the question was about same realm not same account. you can have a
>> common realm for many accounts.
>>
>
> I has tried assign same realm to two accounds but get error.
>
> finally, as with any other software, kazoo does is not "completed" and
>> there are optimizations tat can be done.
>>
>
> I completely agree with you. Moreover, I believe that what I have
> described would not require major change in the kazoo. And soon kazoo can
> be successfully used for the described case.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 18 18:37:58 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Apr 2016 10:37:58 -0400
Subject: [Freeswitch-users] mod_com_amd not "hearing" audio
In-Reply-To: <1485c789cd414be0824fde1bd090b370@CM-EX-V01.cm.local>
References: <1485c789cd414be0824fde1bd090b370@CM-EX-V01.cm.local>
Message-ID: <CAEwTsAhFy1Ln=DBkNoVHsUhoC2MhMeiETbjBjJ2wFTkg_49j5Q@mail.gmail.com>

Interesting.  Please file a bug on this, it's likely going to require a
small patch in that module if it's what you say it's doing.

On Monday, April 18, 2016, Michael Jepson <Michael.Jepson at cm.nl> wrote:

> When placing a call and checking for an answering machine using
> mod_com_amd, it always sees *silent_state*.
>
> session:recordFile does not record any audio either, unless I set
> record_waste_resources=true on the session.
>
>
>
> I assume mod_com_amd is having the same problem as recordFile is, but
> there is no option to have mod_com_amd send RTP packages when listening for
> audio and determining whether we have a person or a machine on the line.
>
>
>
> How can I get mod_com_amd to listen to the audio?
>
>
>
> Any help would be greatly appreciated.
>
>
>
> Best regards,
>
>
>
> *MICHAEL JEPSON *
>
> LEAD DEVELOPER
>
> Michael.Jepon at CM.NL <javascript:_e(%7B%7D,'cvml','Michael.Jepon at CM.NL');>
>
> +31 (0)76 2 012 732
>
>
> [image: cid:5BB267AD-C5CE-4941-B84C-466564FD1DF6 at cm.local]
>
> KONIJNENBERG 30
>
> 4825 BD BREDA
>
> THE NETHERLANDS
>
> WWW.CM.NL <http://www.cm.nl/>
>
>
>
>
>
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From mike at jerris.com  Mon Apr 18 18:39:16 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Apr 2016 10:39:16 -0400
Subject: [Freeswitch-users] how to extract arbitrary sip header value
In-Reply-To: <045159ff9cd94ad28e50ceace3213c37@UK-MAIL-001.edge.local>
References: <045159ff9cd94ad28e50ceace3213c37@UK-MAIL-001.edge.local>
Message-ID: <CAEwTsAj5-xz9W7mjvMGE=W6jgT+D6MDVqHHaEmYT+HDjMi9SWQ@mail.gmail.com>

if you uuid_dump the channel
you can see everything that it has. quickest way to get your answer

On Monday, April 18, 2016, Miroslav Levanic <Miroslav.Levanic at enghouse.com>
wrote:

> I?m using switch_channel_get_variable to extract sip headers with already
> predefined channel variables (
> https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables).
>
> I?ve found that some undefined sip headers can be extracted using only
> ?sip_? prefix (for instance ?sip_P-Preferred-Identity?) and some using
> ?sip_h_? prefix (like is ?sip_h_Diversion?).
>
> Is there some rule or detailed explanation when to use just ?sip_? or
> ?sip_h_? in the name of the required channel variable?
>
>
>
> Thanks,
>
> Miro
>
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From colin.morelli at gmail.com  Mon Apr 18 18:46:11 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Mon, 18 Apr 2016 14:46:11 +0000
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAF7jizKY7fOurMATT-fEG=6nixScA_vBc91oeRSRvQuaj7SJ-g@mail.gmail.com>
References: <CAEvFzR-6ZU_Z2Wu5bedm=f4xnKAoiDoK+5E6R+sEzs-rOWxRJQ@mail.gmail.com>
	<CAHtxdDdw+VrR8N3n2iF-ntCKoFmHe_e12K5zVy6hFmyYuo7feg@mail.gmail.com>
	<CAF7jizKY7fOurMATT-fEG=6nixScA_vBc91oeRSRvQuaj7SJ-g@mail.gmail.com>
Message-ID: <CAPtU-UqqsncL3+3zO4JZ37jMVJPXt9zd0rgF7Snh1AAyOjNrQA@mail.gmail.com>

Benjamin,

Doesn't sound like a lazy answer at all - it's perfectly fair. Hopefully I
didn't come off as trying to avoid learning SIP - that wasn't my goal here
at all. I'm actively reading up as much about SIP as I can reasonably
absorb. For me, trying to get a real example of something running so I can
rip it apart and play with it is the best way for me to learn, that's all I
was really going for here.

That said, note taken - I'll be diving back into learning more about the
protocol overall. I agree if I knew more about this it would probably
naturally fall together a lot easier. I picked up the FreeSWITCH book and
will get the OpenSIPS book as well. Thanks for the recommendation.

Thanks again everyone else for all of your input.

Best,
Colin

On Mon, Apr 18, 2016 at 10:39 AM Benjamin Cropley <
benjamin.cropley at gmail.com> wrote:

> This might sound like a lazy answer, but will help in the long term..
>
> I'd suggest learning about SIP as a protocol before you do anything else.
>
> You'll find learning the protocol itself inherently dictates how to scales
> across multiple servers. For example, in defining what the role of a
> Registrar is, SIP explains that a single 'server/process' can handle that
> job independently of any other SIP related activity. It likewise discusses
> that what a SBC is, and thereby helps understand that you can seperate the
> handling of SIP and RTP to a proxy and media server.
>
> Once you understand SIP properly, it's easy to come in and say "Well I
> need a registrar.. that's OpenSIPS or Kamailio.. Now I need a server to
> handle Media.. well FreeSWITCH does that.. I need a server to handle
> presence.. well presence can utilise dialog info to produce this I might as
> well use OpenSIPS or Kamailio".
>
> If you do implement it using Kazoo, using a guide, or whatever, you'll
> find yourself learning about SIP anyway. Why don't you just attack it head
> on, and you'll understand it all a bit better?
>
> The latest OpenSIPS book has a good chapter on SIP I think is worth the
> read.
>
> I think this is why a lot of people are responding with 'it depends'. What
> they really mean is, "SIP is a flexible protocol, and it's really your
> choice in how you want to implement its various components".
>
>
>
> On Mon, Apr 18, 2016 at 2:04 PM, Sergey Safarov <s.safarov at gmail.com>
> wrote:
>
>> Luis with all due respect to you and your work still...
>>
>> as i said it depends on how you use it. it doesn't make sense to call the
>>> api to fetch a list of 200.000 devices or callflows. use v2 for paging
>>> and/or use the search api.
>>>
>>
>> If you look at KAZOO-4326
>> <https://2600hz.atlassian.net/browse/KAZOO-4326> one more time, you can
>> see that I stores one callflow via kazoo API. For database containing about
>> 20 000 callflows 30 sec to save one callflow is much. Think saving 200 000
>> callflows via kazoo API took about 1-2 months.
>>
>>
>>> also, the question was about same realm not same account. you can have a
>>> common realm for many accounts.
>>>
>>
>> I has tried assign same realm to two accounds but get error.
>>
>> finally, as with any other software, kazoo does is not "completed" and
>>> there are optimizations tat can be done.
>>>
>>
>> I completely agree with you. Moreover, I believe that what I have
>> described would not require major change in the kazoo. And soon kazoo can
>> be successfully used for the described case.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From luis.azedo at factorlusitano.com  Mon Apr 18 18:49:15 2016
From: luis.azedo at factorlusitano.com (Luis Azedo)
Date: Mon, 18 Apr 2016 15:49:15 +0100
Subject: [Freeswitch-users] Scaling Freeswitch
Message-ID: <CAEvFzR-sZ9McNEDk6X0iKr2aFE21H0zv9i8Ny_8zqnzWvZJpkQ@mail.gmail.com>

Sergey,

if you have questions regarding KAZOO-4326 then you can send to proper
mailing list or use JIRA.
if you have any other questions concerning kazoo usage (share realm between
accounts) then you can also send to proper list.

i believe this is not the place for posting and responding to kazoo
specific questions.

i only pop in here because you stated that it doesn't scale and i
completely disagree with that, and didn't want the rest of freeswitch
mailing list users to think this statement is true.

Best
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From brian at freeswitch.org  Mon Apr 18 19:04:06 2016
From: brian at freeswitch.org (Brian West)
Date: Mon, 18 Apr 2016 10:04:06 -0500
Subject: [Freeswitch-users] mod_com_amd not "hearing" audio
In-Reply-To: <CAEwTsAhFy1Ln=DBkNoVHsUhoC2MhMeiETbjBjJ2wFTkg_49j5Q@mail.gmail.com>
References: <1485c789cd414be0824fde1bd090b370@CM-EX-V01.cm.local>
	<CAEwTsAhFy1Ln=DBkNoVHsUhoC2MhMeiETbjBjJ2wFTkg_49j5Q@mail.gmail.com>
Message-ID: <CAEJMVkCAiqEk2dGBv5KE2=h30sPerQ2+VgaVbMUj+s5MJzPVrQ@mail.gmail.com>

Set the variable record_waste_resources=true, it will send media when
record is going and its not mod_com_amd that sends media, its FreeSWITCH so
your idea that it needs to send media is incorrect.

/b


On Mon, Apr 18, 2016 at 9:37 AM, Michael Jerris <mike at jerris.com> wrote:

> Interesting.  Please file a bug on this, it's likely going to require a
> small patch in that module if it's what you say it's doing.
>
>
> On Monday, April 18, 2016, Michael Jepson <Michael.Jepson at cm.nl> wrote:
>
>> When placing a call and checking for an answering machine using
>> mod_com_amd, it always sees *silent_state*.
>>
>> session:recordFile does not record any audio either, unless I set
>> record_waste_resources=true on the session.
>>
>>
>>
>> I assume mod_com_amd is having the same problem as recordFile is, but
>> there is no option to have mod_com_amd send RTP packages when listening for
>> audio and determining whether we have a person or a machine on the line.
>>
>>
>>
>> How can I get mod_com_amd to listen to the audio?
>>
>>
>>
>> Any help would be greatly appreciated.
>>
>>
>>
>> Best regards,
>>
>>
>>
>> *MICHAEL JEPSON *
>>
>> LEAD DEVELOPER
>>
>> Michael.Jepon at CM.NL
>>
>> +31 (0)76 2 012 732
>>
>>
>> [image: cid:5BB267AD-C5CE-4941-B84C-466564FD1DF6 at cm.local]
>>
>> KONIJNENBERG 30
>>
>> 4825 BD BREDA
>>
>> THE NETHERLANDS
>>
>> WWW.CM.NL <http://www.cm.nl/>
>>
>>
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
https://www.gofundme.com/freeswitch_ubuntu

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From italo at freeswitch.org  Mon Apr 18 22:14:26 2016
From: italo at freeswitch.org (=?utf-8?q?=C3=8Dtalo_Rossi?=)
Date: Mon, 18 Apr 2016 11:14:26 -0700 (PDT)
Subject: [Freeswitch-users] Thursday FreeSWITCH Bug Hunt
Message-ID: <a4i9iv2ziqih55mhit8ap4zxn-0@mailer.nylas.com>

FreeSWITCHers,

  

Join us thursday 2PM CST for the Thursday FreeSWITCH Bug Hunt!

  

Where? [conference.freeswitch.org/vc/#/?autocall=888](https://conference.frees
witch.org/vc/#/?autocall=888
"https://conference.freeswitch.org/vc/#/?autocall=888" )

  

Chat? <https://hipchat.freeswitch.org/gYzPMg3Oc>

  

What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support!

  

Help us help you, Join the Bug Hunt!

?talo Rossi

italo at freeswitch.org

IRC chat.freenode.net #freeswitch #freeswitch-dev

Bugs? https://freeswitch.org/jira

Docs? https://freeswitch.org/jira

Chat? https://hipchat.freeswitch.org/gUdAgy0m6

  

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From mgg at giagnocavo.net  Mon Apr 18 22:23:33 2016
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Mon, 18 Apr 2016 18:23:33 +0000
Subject: [Freeswitch-users] Scaling Freeswitch
In-Reply-To: <CAPtU-UqqsncL3+3zO4JZ37jMVJPXt9zd0rgF7Snh1AAyOjNrQA@mail.gmail.com>
References: <CAEvFzR-6ZU_Z2Wu5bedm=f4xnKAoiDoK+5E6R+sEzs-rOWxRJQ@mail.gmail.com>
	<CAHtxdDdw+VrR8N3n2iF-ntCKoFmHe_e12K5zVy6hFmyYuo7feg@mail.gmail.com>
	<CAF7jizKY7fOurMATT-fEG=6nixScA_vBc91oeRSRvQuaj7SJ-g@mail.gmail.com>
	<CAPtU-UqqsncL3+3zO4JZ37jMVJPXt9zd0rgF7Snh1AAyOjNrQA@mail.gmail.com>
Message-ID: <DM2PR07MB702040EAD3307A75C3EEC4AD46B0@DM2PR07MB702.namprd07.prod.outlook.com>

There?s some commercial courses. One is called ?SIP Illustrated? but I can?t seem to find it online anymore. It looked a bit cheesy but was quite good. And it came with an annotated copy of RFC3261, all hyperlinked up and with good explanations for interpreting that ? rather creative spec. I wrote a SIP proxy and another SIP processor and that annotated guide was indispensable to keeping any notion of sanity. I see there?s annotated RFC3261 available online, but I?m not sure it?s the same. Email me if interested and I can dig it up.

You really can?t use OpenSIPS without understanding the SIP protocol (and even then it?s easy to mess up on the many edges). Whereas with FS alone, you can often get away with just thinking ?phone call here to there? and having it work. But scaling out is inevitably going to require a SIP proxy and a decent understanding of SIP message flow.

After you get over the hump of learning SIP you?ll find most people end up more-or-less with a similar architecture as described in Benjamin?s answer. It?s proven to work, and straightforward.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Colin Morelli
Sent: Monday, 18 April, 2016 8:46
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Scaling Freeswitch

Benjamin,

Doesn't sound like a lazy answer at all - it's perfectly fair. Hopefully I didn't come off as trying to avoid learning SIP - that wasn't my goal here at all. I'm actively reading up as much about SIP as I can reasonably absorb. For me, trying to get a real example of something running so I can rip it apart and play with it is the best way for me to learn, that's all I was really going for here.

That said, note taken - I'll be diving back into learning more about the protocol overall. I agree if I knew more about this it would probably naturally fall together a lot easier. I picked up the FreeSWITCH book and will get the OpenSIPS book as well. Thanks for the recommendation.

Thanks again everyone else for all of your input.

Best,
Colin

On Mon, Apr 18, 2016 at 10:39 AM Benjamin Cropley <benjamin.cropley at gmail.com<mailto:benjamin.cropley at gmail.com>> wrote:
This might sound like a lazy answer, but will help in the long term..

I'd suggest learning about SIP as a protocol before you do anything else.

You'll find learning the protocol itself inherently dictates how to scales across multiple servers. For example, in defining what the role of a Registrar is, SIP explains that a single 'server/process' can handle that job independently of any other SIP related activity. It likewise discusses that what a SBC is, and thereby helps understand that you can seperate the handling of SIP and RTP to a proxy and media server.

Once you understand SIP properly, it's easy to come in and say "Well I need a registrar.. that's OpenSIPS or Kamailio.. Now I need a server to handle Media.. well FreeSWITCH does that.. I need a server to handle presence.. well presence can utilise dialog info to produce this I might as well use OpenSIPS or Kamailio".

If you do implement it using Kazoo, using a guide, or whatever, you'll find yourself learning about SIP anyway. Why don't you just attack it head on, and you'll understand it all a bit better?

The latest OpenSIPS book has a good chapter on SIP I think is worth the read.

I think this is why a lot of people are responding with 'it depends'. What they really mean is, "SIP is a flexible protocol, and it's really your choice in how you want to implement its various components".



On Mon, Apr 18, 2016 at 2:04 PM, Sergey Safarov <s.safarov at gmail.com<mailto:s.safarov at gmail.com>> wrote:
Luis with all due respect to you and your work still...

as i said it depends on how you use it. it doesn't make sense to call the api to fetch a list of 200.000 devices or callflows. use v2 for paging and/or use the search api.

If you look at KAZOO-4326<https://2600hz.atlassian.net/browse/KAZOO-4326> one more time, you can see that I stores one callflow via kazoo API. For database containing about 20 000 callflows 30 sec to save one callflow is much. Think saving 200 000 callflows via kazoo API took about 1-2 months.

also, the question was about same realm not same account. you can have a common realm for many accounts.

I has tried assign same realm to two accounds but get error.

finally, as with any other software, kazoo does is not "completed" and there are optimizations tat can be done.

I completely agree with you. Moreover, I believe that what I have described would not require major change in the kazoo. And soon kazoo can be successfully used for the described case.


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

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

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

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From arsenman at connectto.com  Tue Apr 19 00:16:31 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Mon, 18 Apr 2016 13:16:31 -0700
Subject: [Freeswitch-users] help, with Verto - verto-min.js configuration
In-Reply-To: <570FE4A3.6020804@connectto.com>
References: <570EE45E.3060008@connectto.com>
	<CAKbxfG-SDDK6nPwCbByMZgttbRX87CkWf3tE3FGGGVeA9FvK9g@mail.gmail.com>
	<570FE280.7090807@connectto.com>
	<F786B5C6-9190-4585-9078-F681614F291E@jerris.com>
	<570FE4A3.6020804@connectto.com>
Message-ID: <5715409F.3030905@connectto.com>

any other help ?
How to fix setting for turn server ?

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From krice at freeswitch.org  Tue Apr 19 00:26:49 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 18 Apr 2016 15:26:49 -0500
Subject: [Freeswitch-users] help, with Verto - verto-min.js configuration
In-Reply-To: <5715409F.3030905@connectto.com>
References: <570EE45E.3060008@connectto.com>	<CAKbxfG-SDDK6nPwCbByMZgttbRX87CkWf3tE3FGGGVeA9FvK9g@mail.gmail.com>	<570FE280.7090807@connectto.com>	<F786B5C6-9190-4585-9078-F681614F291E@jerris.com>	<570FE4A3.6020804@connectto.com>
	<5715409F.3030905@connectto.com>
Message-ID: <b50601d199b0$a379d720$ea6d8560$@freeswitch.org>

Its just part of the settings when setting up a new call,,, this was covered
very recently on this mailing list. Try checking the archives for the thread

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Arsen
Manukyan
Sent: Monday, April 18, 2016 3:17 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] help, with Verto - verto-min.js
configuration

 

any other help ?
How to fix setting for turn server ?

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From davidwaf at gmail.com  Tue Apr 19 02:04:52 2016
From: davidwaf at gmail.com (David Wafula)
Date: Tue, 19 Apr 2016 00:04:52 +0200
Subject: [Freeswitch-users] mp4 Conference recording is always
 unplayable 352 bytes file
In-Reply-To: <CAK9Ws2df=e3PpffNbikZ9N+yeNsaUfeU1_UsnHxEHSgbAhpb5w@mail.gmail.com>
References: <CAK9Ws2eTyYPLq9ERxmCu_xHUAaOaa1k7B9EBL9cpWSGpEa_qTg@mail.gmail.com>
	<25F63786-A284-48DB-8387-0340F9B3797F@jerris.com>
	<CAK9Ws2e0Y=MUsmeYNQ1apC3a9q3eTN=5eAscOyAEn70KptvzMg@mail.gmail.com>
	<54250C19-0C34-46AA-9FCC-74EA5A06C5C6@jerris.com>
	<CAK9Ws2ezEDnccH+1JaudtsgvULNNAhBey+zAVs0KpmAFBHoPjQ@mail.gmail.com>
	<CAK9Ws2df=e3PpffNbikZ9N+yeNsaUfeU1_UsnHxEHSgbAhpb5w@mail.gmail.com>
Message-ID: <CAK9Ws2fp5NNE17uCqQpbfCT-vsu0Oe9iM9Az0V8uWWGo_Hxb6w@mail.gmail.com>

After trying over and over, i just discovered that an mp4 recording in a
conference will only work if
<param name="video-mode" value="mux"/> is set.

Am all sorted now.


On Sun, Apr 17, 2016 at 12:17 AM, David Wafula <davidwaf at gmail.com> wrote:

> So i did a comparison between wav recording and mp4 recording, and it
> seems BAD READ -1000 only occurs when using mp4.
>
> Here is part  log for mp4
>
> ----
> 3f39d3c5-97cb-f91a-48b5-c016e82175e2 2016-04-16 21:12:10.361551 [NOTICE]
> mod_verto.c:2746 Hangup verto.rtc/83789 [CS_EXECUTE] [NORMAL_CLEARING]
> 2016-04-16 21:12:10.361551 [DEBUG] switch_core_media.c:5459
> verto.rtc/83789 Video thread ended
> 3f39d3c5-97cb-f91a-48b5-c016e82175e2 2016-04-16 21:12:10.381543 [INFO]
> conference_loop.c:1400 Channel leaving conference, cause: NORMAL_CLEARING
> 3f39d3c5-97cb-f91a-48b5-c016e82175e2 2016-04-16 21:12:10.381543 [DEBUG]
> mod_conference.c:2209 verto.rtc/83789 skip receive message [UNBRIDGE]
> (channel is hungup already)
> 2016-04-16 21:12:10.381543 [DEBUG] mod_verto.c:1862 BAD READ -1000
> 2016-04-16 21:12:10.381543 [INFO] mod_verto.c:2005 105.226.104.136:46232
> Ending client thread.
> 2016-04-16 21:12:10.381543 [INFO] mod_verto.c:2013 105.226.104.136:46232
> Thread ended
>
> --
>
>
>
> Here is the same bit of log with just wav recording:
>
> -------
>
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.490132 [NOTICE]
> mod_verto.c:2746 Hangup verto.rtc/83789 [CS_EXECUTE] [NORMAL_CLEARING]
> 2016-04-16 22:06:57.500121 [DEBUG] switch_core_media.c:5459
> verto.rtc/83789 Video thread ended
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [INFO]
> conference_loop.c:1400 Channel leaving conference, cause: NORMAL_CLEARING
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG]
> mod_conference.c:2209 verto.rtc/83789 skip receive message [UNBRIDGE]
> (channel is hungup already)
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG]
> switch_core_media.c:9147 verto.rtc/83789 skip receive message [HARD_MUTE]
> (channel is hungup already)
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG]
> switch_core_codec.c:258 Restore original codec.
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG]
> switch_core_session.c:2796 verto.rtc/83789 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>
> ------
> Thank you
>
> On Sat, Apr 16, 2016 at 11:28 PM, David Wafula <davidwaf at gmail.com> wrote:
>
>> Please see the log here if there is anything am doing wrong:
>> http://pastebin.com/WGcmg2iX
>>
>> I noticed one line there:
>> 2016-04-16 21:12:10.381543 [DEBUG] mod_verto.c:1862 BAD READ -1000
>>
>> is that normal ?
>>
>>
>>
>>
>> On Sat, Apr 16, 2016 at 1:15 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> look at the debug log, there are other modules that support mp4
>>>
>>> On Apr 15, 2016, at 5:57 PM, David Wafula <davidwaf at gmail.com> wrote:
>>>
>>> I installed  mod_av it like:
>>>  apt-get install freeswitch-mod-av
>>>
>>> and put  <load module="mod_av"/> in modules.conf.xml
>>>
>>> I think that just about the only thing i did with it.
>>>
>>> Am assuming  then than means yes, it is using mod_av ?
>>>
>>> (just for testing, i tried wav and  it records perfectly fine)
>>>
>>> Regards
>>>
>>> On Fri, Apr 15, 2016 at 11:39 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>>
>>>> is it using mod_av?
>>>>
>>>> On Apr 15, 2016, at 5:12 PM, David Wafula <davidwaf at gmail.com> wrote:
>>>>
>>>> I have verto-client  connecting to a conference. I have setup
>>>> auto-record pointing to an mp4 file format for mod_conference. I have
>>>> installed libavcodec-extra, enabled mod_av. The recording seems to
>>>> start and stop just fine, except the file is just an unplayable 352 bytes.
>>>> I must have missed something, just can't to figure out what.
>>>>
>>>> Am running FreeSWITCH Version 1.6.7-14-d38d065~64bit (-14-d38d065
>>>> 64bit) on Debian 8.1 Jesse, x64.
>>>>
>>>> Please anyone with pointers help.
>>>>
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> David W
>>
>
>
>
> --
> David W
>



-- 
David W
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From arsenman at connectto.com  Tue Apr 19 02:21:50 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Mon, 18 Apr 2016 15:21:50 -0700
Subject: [Freeswitch-users] help, with Verto - verto-min.js configuration
In-Reply-To: <b50601d199b0$a379d720$ea6d8560$@freeswitch.org>
References: <570EE45E.3060008@connectto.com>
	<CAKbxfG-SDDK6nPwCbByMZgttbRX87CkWf3tE3FGGGVeA9FvK9g@mail.gmail.com>
	<570FE280.7090807@connectto.com>
	<F786B5C6-9190-4585-9078-F681614F291E@jerris.com>
	<570FE4A3.6020804@connectto.com> <5715409F.3030905@connectto.com>
	<b50601d199b0$a379d720$ea6d8560$@freeswitch.org>
Message-ID: <57155DFE.2050806@connectto.com>

I really don't know how to search information in mailing list

On 4/18/2016 1:26 PM, Ken Rice wrote:
>
> Its just part of the settings when setting up a new call,,, this was 
> covered very recently on this mailing list. Try checking the archives 
> for the thread
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Arsen Manukyan
> *Sent:* Monday, April 18, 2016 3:17 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] help, with Verto - verto-min.js 
> configuration
>
> any other help ?
> How to fix setting for turn server ?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Apr 19 02:53:14 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 18 Apr 2016 18:53:14 -0400
Subject: [Freeswitch-users] mp4 Conference recording is always
	unplayable 352 bytes file
In-Reply-To: <CAK9Ws2fp5NNE17uCqQpbfCT-vsu0Oe9iM9Az0V8uWWGo_Hxb6w@mail.gmail.com>
References: <CAK9Ws2eTyYPLq9ERxmCu_xHUAaOaa1k7B9EBL9cpWSGpEa_qTg@mail.gmail.com>
	<25F63786-A284-48DB-8387-0340F9B3797F@jerris.com>
	<CAK9Ws2e0Y=MUsmeYNQ1apC3a9q3eTN=5eAscOyAEn70KptvzMg@mail.gmail.com>
	<54250C19-0C34-46AA-9FCC-74EA5A06C5C6@jerris.com>
	<CAK9Ws2ezEDnccH+1JaudtsgvULNNAhBey+zAVs0KpmAFBHoPjQ@mail.gmail.com>
	<CAK9Ws2df=e3PpffNbikZ9N+yeNsaUfeU1_UsnHxEHSgbAhpb5w@mail.gmail.com>
	<CAK9Ws2fp5NNE17uCqQpbfCT-vsu0Oe9iM9Az0V8uWWGo_Hxb6w@mail.gmail.com>
Message-ID: <92E07D67-3D62-4B30-8FA9-7B15A458C227@jerris.com>

or transcode too...

> On Apr 18, 2016, at 6:04 PM, David Wafula <davidwaf at gmail.com> wrote:
> 
> After trying over and over, i just discovered that an mp4 recording in a conference will only work if  
> <param name="video-mode" value="mux"/> is set.
> 
> Am all sorted now.
> 
> 
> On Sun, Apr 17, 2016 at 12:17 AM, David Wafula <davidwaf at gmail.com <mailto:davidwaf at gmail.com>> wrote:
> So i did a comparison between wav recording and mp4 recording, and it seems BAD READ -1000 only occurs when using mp4.
> 
> Here is part  log for mp4 
> 
> ----
> 3f39d3c5-97cb-f91a-48b5-c016e82175e2 2016-04-16 21:12:10.361551 [NOTICE] mod_verto.c:2746 Hangup verto.rtc/83789 [CS_EXECUTE] [NORMAL_CLEARING]
> 2016-04-16 21:12:10.361551 [DEBUG] switch_core_media.c:5459 verto.rtc/83789 Video thread ended
> 3f39d3c5-97cb-f91a-48b5-c016e82175e2 2016-04-16 21:12:10.381543 [INFO] conference_loop.c:1400 Channel leaving conference, cause: NORMAL_CLEARING
> 3f39d3c5-97cb-f91a-48b5-c016e82175e2 2016-04-16 21:12:10.381543 [DEBUG] mod_conference.c:2209 verto.rtc/83789 skip receive message [UNBRIDGE] (channel is hungup already)
> 2016-04-16 21:12:10.381543 [DEBUG] mod_verto.c:1862 BAD READ -1000
> 2016-04-16 21:12:10.381543 [INFO] mod_verto.c:2005 105.226.104.136:46232 <http://105.226.104.136:46232/> Ending client thread.
> 2016-04-16 21:12:10.381543 [INFO] mod_verto.c:2013 105.226.104.136:46232 <http://105.226.104.136:46232/> Thread ended
> 
> --
> 
> 
> 
> Here is the same bit of log with just wav recording:
> 
> -------
> 
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.490132 [NOTICE] mod_verto.c:2746 Hangup verto.rtc/83789 [CS_EXECUTE] [NORMAL_CLEARING]
> 2016-04-16 22:06:57.500121 [DEBUG] switch_core_media.c:5459 verto.rtc/83789 Video thread ended
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [INFO] conference_loop.c:1400 Channel leaving conference, cause: NORMAL_CLEARING
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG] mod_conference.c:2209 verto.rtc/83789 skip receive message [UNBRIDGE] (channel is hungup already)
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG] switch_core_media.c:9147 verto.rtc/83789 skip receive message [HARD_MUTE] (channel is hungup already)
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG] switch_core_codec.c:258 Restore original codec.
> d5f7b099-e268-b149-bd5f-459b15c1a9f4 2016-04-16 22:06:57.500121 [DEBUG] switch_core_session.c:2796 verto.rtc/83789 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 
> ------
> Thank you
> 
> On Sat, Apr 16, 2016 at 11:28 PM, David Wafula <davidwaf at gmail.com <mailto:davidwaf at gmail.com>> wrote:
> Please see the log here if there is anything am doing wrong:
> http://pastebin.com/WGcmg2iX <http://pastebin.com/WGcmg2iX>
> 
> I noticed one line there:
> 2016-04-16 21:12:10.381543 [DEBUG] mod_verto.c:1862 BAD READ -1000
>  
> is that normal ?
> 
> 
> 
> 
> On Sat, Apr 16, 2016 at 1:15 AM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> look at the debug log, there are other modules that support mp4
> 
>> On Apr 15, 2016, at 5:57 PM, David Wafula <davidwaf at gmail.com <mailto:davidwaf at gmail.com>> wrote:
>> 
>> I installed  mod_av it like:
>>  apt-get install freeswitch-mod-av
>> 
>> and put  <load module="mod_av"/> in modules.conf.xml
>> 
>> I think that just about the only thing i did with it.
>> 
>> Am assuming  then than means yes, it is using mod_av ?
>> 
>> (just for testing, i tried wav and  it records perfectly fine)
>> 
>> Regards
>> 
>> On Fri, Apr 15, 2016 at 11:39 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> is it using mod_av?
>> 
>>> On Apr 15, 2016, at 5:12 PM, David Wafula <davidwaf at gmail.com <mailto:davidwaf at gmail.com>> wrote:
>>> 
>>> I have verto-client  connecting to a conference. I have setup auto-record pointing to an mp4 file format for mod_conference. I have installed libavcodec-extra, enabled mod_av.   The recording seems to start and stop just fine, except the file is just an unplayable 352 bytes. I must have missed something, just can't to figure out what.
>>> 
>>> Am running FreeSWITCH Version 1.6.7-14-d38d065~64bit (-14-d38d065 64bit)  on Debian 8.1 Jesse, x64.
>>> 
>>> Please anyone with pointers help.
>> 
> 
> 
> _________________________________________________________________________
> 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/>
> 
> 
> 
> -- 
> David W
> 
> 
> 
> -- 
> David W
> 
> 
> 
> -- 
> David W
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From italo at freeswitch.org  Tue Apr 19 05:03:17 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Mon, 18 Apr 2016 22:03:17 -0300
Subject: [Freeswitch-users] mod_com_amd not "hearing" audio
In-Reply-To: <CAEJMVkCAiqEk2dGBv5KE2=h30sPerQ2+VgaVbMUj+s5MJzPVrQ@mail.gmail.com>
References: <1485c789cd414be0824fde1bd090b370@CM-EX-V01.cm.local>
	<CAEwTsAhFy1Ln=DBkNoVHsUhoC2MhMeiETbjBjJ2wFTkg_49j5Q@mail.gmail.com>
	<CAEJMVkCAiqEk2dGBv5KE2=h30sPerQ2+VgaVbMUj+s5MJzPVrQ@mail.gmail.com>
Message-ID: <CABMX_w99Wjp=ZqZZHrYA3TwEgmB5cVXUH4SW0qXXahAM2KV5rA@mail.gmail.com>

Michael,

What's the name of the installer you tried to use? Are you sure you're
using the latest version? I have it deployed in a few scenarios and it's
working.

On Mon, Apr 18, 2016 at 12:04 PM, Brian West <brian at freeswitch.org> wrote:

> Set the variable record_waste_resources=true, it will send media when
> record is going and its not mod_com_amd that sends media, its FreeSWITCH so
> your idea that it needs to send media is incorrect.
>
> /b
>
>
> On Mon, Apr 18, 2016 at 9:37 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> Interesting.  Please file a bug on this, it's likely going to require a
>> small patch in that module if it's what you say it's doing.
>>
>>
>> On Monday, April 18, 2016, Michael Jepson <Michael.Jepson at cm.nl> wrote:
>>
>>> When placing a call and checking for an answering machine using
>>> mod_com_amd, it always sees *silent_state*.
>>>
>>> session:recordFile does not record any audio either, unless I set
>>> record_waste_resources=true on the session.
>>>
>>>
>>>
>>> I assume mod_com_amd is having the same problem as recordFile is, but
>>> there is no option to have mod_com_amd send RTP packages when listening for
>>> audio and determining whether we have a person or a machine on the line.
>>>
>>>
>>>
>>> How can I get mod_com_amd to listen to the audio?
>>>
>>>
>>>
>>> Any help would be greatly appreciated.
>>>
>>>
>>>
>>> Best regards,
>>>
>>>
>>>
>>> *MICHAEL JEPSON *
>>>
>>> LEAD DEVELOPER
>>>
>>> Michael.Jepon at CM.NL
>>>
>>> +31 (0)76 2 012 732
>>>
>>>
>>> [image: cid:5BB267AD-C5CE-4941-B84C-466564FD1DF6 at cm.local]
>>>
>>> KONIJNENBERG 30
>>>
>>> 4825 BD BREDA
>>>
>>> THE NETHERLANDS
>>>
>>> WWW.CM.NL <http://www.cm.nl/>
>>>
>>>
>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
> https://www.gofundme.com/freeswitch_ubuntu
>
> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit:
> /r/freeswitch <https://www.reddit.com/r/freeswitch>
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
italo at freeswitch.org
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From deepikay at iiitd.ac.in  Tue Apr 19 15:55:56 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Tue, 19 Apr 2016 17:25:56 +0530
Subject: [Freeswitch-users] Conference Setup
Message-ID: <CAM99_YnGD4TaNe6MYuw3Y4NjwJheQg_PM19sEsUtP2xmJuOaWw@mail.gmail.com>

Hi All,

I want to set up a conference initiated from a python esl script.
My Dialplans are:

<extension name="conf_demo">
<condition field="destination_number" expression="^9099$">
<action application="conference"
data="radioHealth_${strftime(%Y-%m-%d)}+flags{endconf}"/>
</condition>

<extension name="conf_demo">
<condition field="destination_number" expression="^9098$">
<action application="conference"
data="radioHealth_${strftime(%Y-%m-%d)}+flags{mute}"/>
</condition>

Python script snippets:

Calling the first person in unmute mode :

freeswitchcon =  ESL.ESLconnection('127.0.0.2', '8021', 'ClueCon')

freeswitchcon.api("originate","sofia/gateway/MySIP/91XXXXXXXXXX+"
&conference(9099)"

Calling the second persion

freeswitchcon.api("originate","sofia/gateway/MySIP/91XXXXXXXXXX+"
&conference(radioHealth_${strftime(%Y-%m-%d)}+flags{mute})"

Two outbound calls are created but these are not bridged, I can't hear the
voice of first person

I tried bridge flags

<extension name="conf_demo">
<condition field="destination_number" expression="^9098$">
<action application="conference" data="bridge
:radioHealth_${strftime(%Y-%m-%d)}+flags{mute}"/>
</condition>

 not sure if it is the correct way.

I also tried calling from SIP internal account :

freeswitchcon.api("originate","sofia/internal/1004 at X>X.X.X.X:5080 4446")

where 4446 transfers a call to the conference dialplan and called other
members from the script to add them in the conference

but in this case, I get logs as:

switch_channel.c:1055 New Channel loopback/app=voicemail:default X.X.X.X
1004-

 loopback/app=voicemail:default X.X.X.X 1004-a setup codec opus/

and conference is not setup

Any help would be appreciated

Regards,
Deepiak
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From 35633 at heb.be  Tue Apr 19 15:59:16 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Tue, 19 Apr 2016 13:59:16 +0200
Subject: [Freeswitch-users] Freeswitch and opensips
In-Reply-To: <CAFSzUtb8hyWgY+sugSsUDd5BNRDVsVTCX5NUkqzq505e9+0rig@mail.gmail.com>
References: <CAFSzUtb8hyWgY+sugSsUDd5BNRDVsVTCX5NUkqzq505e9+0rig@mail.gmail.com>
Message-ID: <CAFSzUtZOjj5NfHbB7ne4QOSkufn76qzXCMniCT_QLjKcsQvzcw@mail.gmail.com>

Hello Younas,
You asked me to add u to skype in order  u to help me confoguring Opensips
and Freeswitch, i sent the invitation since 13 April but u did not
respond.Maybe you are busy or you did not recognize me. My name is Francjos
on opensips mailing list and on skype i'm nduwayezu Joselyne.
Please, i need your help.

NDUWAYEZU Joselyne

2016-04-11 14:25 GMT+02:00 Nduwayezu, Joselyne <35633 at heb.be>:

> I'm using Opensips as load balancer in front of FreeSwitch. To tell
> Freeswitch that calls come from Opensips proxy, do i have to create a new
> external profile in sip_profiles directory or add an extension in
> dialplan/public.xml or both of two?
>
> Second question, in this file
> :/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml , i read that i
> have to specify the CIDR, is the ip address  of Opensips?
>
> Thank you
>
>
>
> NDUWAYEZU Joselyne
>

-- 
Haute ?cole de Bruxelles
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From miha at softnet.si  Tue Apr 19 16:29:03 2016
From: miha at softnet.si (Miha)
Date: Tue, 19 Apr 2016 14:29:03 +0200
Subject: [Freeswitch-users] Freeswitch and opensips
In-Reply-To: <CAFSzUtZOjj5NfHbB7ne4QOSkufn76qzXCMniCT_QLjKcsQvzcw@mail.gmail.com>
References: <CAFSzUtb8hyWgY+sugSsUDd5BNRDVsVTCX5NUkqzq505e9+0rig@mail.gmail.com>
	<CAFSzUtZOjj5NfHbB7ne4QOSkufn76qzXCMniCT_QLjKcsQvzcw@mail.gmail.com>
Message-ID: <5716248F.6060809@softnet.si>

1. you do not have to add another external profile but it would be nice :)
2. you have to add ip to that file.


miha

On 19/04/2016 13:59, Nduwayezu, Joselyne wrote:
> Hello Younas,
> You asked me to add u to skype in order  u to help me confoguring 
> Opensips and Freeswitch, i sent the invitation since 13 April but u 
> did not respond.Maybe you are busy or you did not recognize me. My 
> name is Francjos on opensips mailing list and on skype i'm nduwayezu 
> Joselyne.
> Please, i need your help.
>
> NDUWAYEZU Joselyne
>
> 2016-04-11 14:25 GMT+02:00 Nduwayezu, Joselyne <35633 at heb.be 
> <mailto:35633 at heb.be>>:
>
>     I'm using Opensips as load balancer in front of FreeSwitch. To
>     tell Freeswitch that calls come from Opensips proxy, do i have to
>     create a new external profile in sip_profiles directory or add an
>     extension in dialplan/public.xml or both of two?
>
>     Second question, in this file
>     :/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml , i read
>     that i have to specify the CIDR, is the ip address  of Opensips?
>
>     Thank you
>
>
>
>     NDUWAYEZU Joselyne
>
>
>
> Haute ?cole de Bruxelles
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From arsenman at connectto.com  Tue Apr 19 22:28:21 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Tue, 19 Apr 2016 11:28:21 -0700
Subject: [Freeswitch-users] Verto demo STUN and TURN configs
Message-ID: <571678C5.1050804@connectto.com>

Guys please help with Verto demo STUN and TURN configs edit
I cant found any helpful information

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From luis.daniel.lucio at gmail.com  Wed Apr 20 01:07:13 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Tue, 19 Apr 2016 17:07:13 -0400
Subject: [Freeswitch-users] Verto demo STUN and TURN configs
In-Reply-To: <571678C5.1050804@connectto.com>
References: <571678C5.1050804@connectto.com>
Message-ID: <CAFLo2QxPca8rNQN3HuAysh_LsdJEZWk1mRGsQ0oaSHJ=417M5Q@mail.gmail.com>

Do you have already a turn server?
Le 19 avr. 2016 2:29 PM, "Arsen Manukyan" <arsenman at connectto.com> a ?crit :

> Guys please help with Verto demo STUN and TURN configs edit
> I cant found any helpful information
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Apr 20 02:01:07 2016
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Apr 2016 15:01:07 -0700
Subject: [Freeswitch-users] Conference Setup
In-Reply-To: <CAM99_YnGD4TaNe6MYuw3Y4NjwJheQg_PM19sEsUtP2xmJuOaWw@mail.gmail.com>
References: <CAM99_YnGD4TaNe6MYuw3Y4NjwJheQg_PM19sEsUtP2xmJuOaWw@mail.gmail.com>
Message-ID: <CAKzWOxV-Y_yqX8GVBN2=kc9YRDZPGXEkWqBwOZmwaM8=pv1QAg@mail.gmail.com>

The first thing you might do is verify that both of these call legs are
going into the same conference. Run the script, let the call legs get
answered, then from fs_cli run:
conference list

I suspect that you will have each user in his own conference. Check and see
if you have a conference named "9099". If you do then change the target of
the originate from "&conference(9099)" to just "9099". The former calls the
conference app directly while the latter sends the call through the
dialplan, which is probably what you were trying to do.

-MSC

On Tue, Apr 19, 2016 at 4:55 AM, Deepika Yadav <deepikay at iiitd.ac.in> wrote:

> Hi All,
>
> I want to set up a conference initiated from a python esl script.
> My Dialplans are:
>
> <extension name="conf_demo">
> <condition field="destination_number" expression="^9099$">
> <action application="conference"
> data="radioHealth_${strftime(%Y-%m-%d)}+flags{endconf}"/>
> </condition>
>
> <extension name="conf_demo">
> <condition field="destination_number" expression="^9098$">
> <action application="conference"
> data="radioHealth_${strftime(%Y-%m-%d)}+flags{mute}"/>
> </condition>
>
> Python script snippets:
>
> Calling the first person in unmute mode :
>
> freeswitchcon =  ESL.ESLconnection('127.0.0.2', '8021', 'ClueCon')
>
> freeswitchcon.api("originate","sofia/gateway/MySIP/91XXXXXXXXXX+"
> &conference(9099)"
>
> Calling the second persion
>
> freeswitchcon.api("originate","sofia/gateway/MySIP/91XXXXXXXXXX+"
> &conference(radioHealth_${strftime(%Y-%m-%d)}+flags{mute})"
>
> Two outbound calls are created but these are not bridged, I can't hear the
> voice of first person
>
> I tried bridge flags
>
> <extension name="conf_demo">
> <condition field="destination_number" expression="^9098$">
> <action application="conference" data="bridge
> :radioHealth_${strftime(%Y-%m-%d)}+flags{mute}"/>
> </condition>
>
>  not sure if it is the correct way.
>
> I also tried calling from SIP internal account :
>
> freeswitchcon.api("originate","sofia/internal/1004 at X>X.X.X.X:5080 4446")
>
> where 4446 transfers a call to the conference dialplan and called other
> members from the script to add them in the conference
>
> but in this case, I get logs as:
>
> switch_channel.c:1055 New Channel loopback/app=voicemail:default X.X.X.X
> 1004-
>
>  loopback/app=voicemail:default X.X.X.X 1004-a setup codec opus/
>
> and conference is not setup
>
> Any help would be appreciated
>
> Regards,
> Deepiak
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From arsenman at connectto.com  Wed Apr 20 02:12:49 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Tue, 19 Apr 2016 15:12:49 -0700
Subject: [Freeswitch-users] Verto demo STUN and TURN configs
In-Reply-To: <CAFLo2QxPca8rNQN3HuAysh_LsdJEZWk1mRGsQ0oaSHJ=417M5Q@mail.gmail.com>
References: <571678C5.1050804@connectto.com>
	<CAFLo2QxPca8rNQN3HuAysh_LsdJEZWk1mRGsQ0oaSHJ=417M5Q@mail.gmail.com>
Message-ID: <5716AD61.6030002@connectto.com>

yes

On 4/19/2016 2:07 PM, Luis Daniel Lucio Quiroz wrote:
>
> Do you have already a turn server?
>
> Le 19 avr. 2016 2:29 PM, "Arsen Manukyan" <arsenman at connectto.com 
> <mailto:arsenman at connectto.com>> a ?crit :
>
>     Guys please help with Verto demo STUN and TURN configs edit
>     I cant found any helpful information
>
>
>     _________________________________________________________________________
>     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

-- 
Arsen Manukyan
ConnectTo Communications Inc.
555 Riverdale Dr., Suite A
Glendale, CA 91204
arsenman at connectto.com
http://www.ConnectTo.com
Tel. 818.546.4636
FAX 818.546.4617
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From joel at gogii.net  Wed Apr 20 05:38:38 2016
From: joel at gogii.net (Joel Serrano)
Date: Tue, 19 Apr 2016 18:38:38 -0700
Subject: [Freeswitch-users] JitterBuffer possible problem when configured
	with number of packets.
Message-ID: <CAMtXxQ==0-1GeLik0+4_XfoPYMa7Tq4ue5coP3CsG4pypHQ3tQ@mail.gmail.com>

Hi all,

I'm doing several tests jitterbuffer related. I've read in different posts
to the list that you can specify packet size ("20"), or the number of
packets ("1p"), why is it not being enabled if I use 1p instead of 20?
(Example:
http://lists.freeswitch.org/pipermail/freeswitch-docs/2015-November/000532.html
)


Scenario:

** Add in the sofia profile:

<param name="auto-jitterbuffer-msec" value="20"/>

--- Make a call

** Output in debug log:

[...]
4557180d-579a-4c3c-98a3-287cb078e913 2016-04-20 00:57:03.570889 [DEBUG]
switch_core_media.c:1964 Setting Jitterbuffer to 20ms (1 frames) (50 max
frames)
[...]


Scenario 2:

** Add in the sofia profile:

<param name="auto-jitterbuffer-msec" value="1p"/>

--- Make a call

Log doesn't say that jitterbuffer has been enabled.


Can it be that 1p doesn't output the log but is working anyway?


I have removed the auto-jitterbuffer-msec from the sofia profile, and done
both tests setting jitterbuffer_msec directly in the dialplan with "20" and
with "1p", the result is the same: 20 works, 1p doesn't.

If I use 1p, I don't have to worry about the size as if its 20ms, 40ms or
whatever it will work, using "20", the setting would not be valid for
40ms.. (please correct me if I'm wrong).

I know this can be a bug and thus should be handled in JIRA, I will be
happy to create the ticket etc etc, I first want to make sure that the
problem is not my config.

Tests are with version 1.6.7 -14-d38d065 64bit

Thanks!
Joel.
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From joel at gogii.net  Wed Apr 20 05:40:02 2016
From: joel at gogii.net (Joel Serrano)
Date: Tue, 19 Apr 2016 18:40:02 -0700
Subject: [Freeswitch-users] FS Logs show ERR msgs: not enough buffer
 space for required resample operation!
In-Reply-To: <CAKbxfG9_qEz2LrMPASXi6WKYhYTm0HYSCqDXXif17MxgMXA_8w@mail.gmail.com>
References: <CAMtXxQmLrVi8oBUor7N8TMBfK8O=_jdXAUApY7py7d6OTtACmw@mail.gmail.com>
	<CAMtXxQm_t5GisLkYE3nCe775gq1T31XhxZvVZRPyygHx4__how@mail.gmail.com>
	<CAO3obxYL9J__ZapmX3JVmRU-TkSHyueRHrrwohpxV+Ejt=Tx0g@mail.gmail.com>
	<CAKbxfG9_qEz2LrMPASXi6WKYhYTm0HYSCqDXXif17MxgMXA_8w@mail.gmail.com>
Message-ID: <CAMtXxQnposrLPqMxicRVMm1mVFPye7Bzd3vSrg2ToBLnHni-7A@mail.gmail.com>

Hi Anthony,

I am happy to open a JIRA for this, but I don't really know how to
reproduce it, is that ok?

Thanks,
Joel.

On Thu, Apr 14, 2016 at 10:40 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> We do not field issues on this mailing list.
> http://freeswitch.org/jira
>
> Help keep the list about discussions and not support.
>
>
> On Thu, Apr 14, 2016 at 8:58 AM, Piotr Gregor <piotrek.gregor at gmail.com>
> wrote:
>
>> Hi Joel,
>>
>> This might mean you try to resample between formats that differ too much,
>> e.g. "to" rate is too big compared to "from" rate.
>>
>> cheers,
>> Piotr
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ? @anthmfs  ? @FreeSWITCH  ?
>
> ? http://freeswitch.org/  ? http://cluecon.com/  ?
> http://twitter.com/FreeSWITCH
> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
> ? sip:888 at conference.freeswitch.org  ? +19193869900
>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From deepikay at iiitd.ac.in  Wed Apr 20 07:28:01 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Wed, 20 Apr 2016 08:58:01 +0530
Subject: [Freeswitch-users] Conference Setup
In-Reply-To: <CAKzWOxV-Y_yqX8GVBN2=kc9YRDZPGXEkWqBwOZmwaM8=pv1QAg@mail.gmail.com>
References: <CAM99_YnGD4TaNe6MYuw3Y4NjwJheQg_PM19sEsUtP2xmJuOaWw@mail.gmail.com>
	<CAKzWOxV-Y_yqX8GVBN2=kc9YRDZPGXEkWqBwOZmwaM8=pv1QAg@mail.gmail.com>
Message-ID: <CAM99_YnSEckL2AHUwi7BERvb0mhePY=qADPmh+bX4TH6gg2xYw@mail.gmail.com>

Thanks Michael, using 9099 directly solved the issue.

On Wed, Apr 20, 2016 at 3:31 AM, Michael Collins <msc at freeswitch.org> wrote:

> The first thing you might do is verify that both of these call legs are
> going into the same conference. Run the script, let the call legs get
> answered, then from fs_cli run:
> conference list
>
> I suspect that you will have each user in his own conference. Check and
> see if you have a conference named "9099". If you do then change the target
> of the originate from "&conference(9099)" to just "9099". The former calls
> the conference app directly while the latter sends the call through the
> dialplan, which is probably what you were trying to do.
>
> -MSC
>
> On Tue, Apr 19, 2016 at 4:55 AM, Deepika Yadav <deepikay at iiitd.ac.in>
> wrote:
>
>> Hi All,
>>
>> I want to set up a conference initiated from a python esl script.
>> My Dialplans are:
>>
>> <extension name="conf_demo">
>> <condition field="destination_number" expression="^9099$">
>> <action application="conference"
>> data="radioHealth_${strftime(%Y-%m-%d)}+flags{endconf}"/>
>> </condition>
>>
>> <extension name="conf_demo">
>> <condition field="destination_number" expression="^9098$">
>> <action application="conference"
>> data="radioHealth_${strftime(%Y-%m-%d)}+flags{mute}"/>
>> </condition>
>>
>> Python script snippets:
>>
>> Calling the first person in unmute mode :
>>
>> freeswitchcon =  ESL.ESLconnection('127.0.0.2', '8021', 'ClueCon')
>>
>> freeswitchcon.api("originate","sofia/gateway/MySIP/91XXXXXXXXXX+"
>> &conference(9099)"
>>
>> Calling the second persion
>>
>> freeswitchcon.api("originate","sofia/gateway/MySIP/91XXXXXXXXXX+"
>> &conference(radioHealth_${strftime(%Y-%m-%d)}+flags{mute})"
>>
>> Two outbound calls are created but these are not bridged, I can't hear
>> the voice of first person
>>
>> I tried bridge flags
>>
>> <extension name="conf_demo">
>> <condition field="destination_number" expression="^9098$">
>> <action application="conference" data="bridge
>> :radioHealth_${strftime(%Y-%m-%d)}+flags{mute}"/>
>> </condition>
>>
>>  not sure if it is the correct way.
>>
>> I also tried calling from SIP internal account :
>>
>> freeswitchcon.api("originate","sofia/internal/1004 at X>X.X.X.X:5080 4446")
>>
>> where 4446 transfers a call to the conference dialplan and called other
>> members from the script to add them in the conference
>>
>> but in this case, I get logs as:
>>
>> switch_channel.c:1055 New Channel loopback/app=voicemail:default X.X.X.X
>> 1004-
>>
>>  loopback/app=voicemail:default X.X.X.X 1004-a setup codec opus/
>>
>> and conference is not setup
>>
>> Any help would be appreciated
>>
>> Regards,
>> Deepiak
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 20 09:46:07 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Wed, 20 Apr 2016 11:16:07 +0530
Subject: [Freeswitch-users] No audio Freesswitch
Message-ID: <CAM99_YnWuBGBaO3dg1yjxspM889fw2=J1EMo423GNPZ1Mb5AUg@mail.gmail.com>

Hi,

I am using Amazon ec2 platfrom to run freeswitch application. On restarting
I am not able to hear any audio both sides,

I have configured the settings by following the steps given here:

https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2

I have allowed all traffic for inbound and outbound fro all IPs (Ec2 GUI
security group)

netstat -an | grep ':5060' gives


tcp        0      0 172.31.22.224:5060      0.0.0.0:*               LISTEN
tcp6       0      0 ::1:5060                :::*                    LISTEN
udp        0      0 172.31.22.224:5060      0.0.0.0:*
udp6       0      0 ::1:5060                :::*

netstat -an | grep ':5080'

tcp        0      0 172.31.22.224:5080      0.0.0.0:*               LISTEN
tcp6       0      0 ::1:5080                :::*                    LISTEN
udp        0      0 172.31.22.224:5080      0.0.0.0:*
udp6       0      0 ::1:5080                :::*

Any guidance to proceed would be highly thankful

Regards,
Deepika
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From deepikay at iiitd.ac.in  Wed Apr 20 09:48:56 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Wed, 20 Apr 2016 11:18:56 +0530
Subject: [Freeswitch-users] No audio Freesswitch
In-Reply-To: <CAM99_YnWuBGBaO3dg1yjxspM889fw2=J1EMo423GNPZ1Mb5AUg@mail.gmail.com>
References: <CAM99_YnWuBGBaO3dg1yjxspM889fw2=J1EMo423GNPZ1Mb5AUg@mail.gmail.com>
Message-ID: <CAM99_Yn028GLab=fZQgqF+5YgrZAwUKaMik93qpm_J0cD-h-bg@mail.gmail.com>

On dialling 5000 from a softphone gives the following logs :

 mod_dialplan_xml.c:635 Processing 1019 <1019>->5000 in context default
 switch_core_media.c:5396 Activating RTCP PORT 4001
 switch_rtp.c:3384 Activating Audio Secure RTP SEND
 switch_rtp.c:3362 Activating Audio Secure RTP RECV


On Wed, Apr 20, 2016 at 11:16 AM, Deepika Yadav <deepikay at iiitd.ac.in>
wrote:

> Hi,
>
> I am using Amazon ec2 platfrom to run freeswitch application. On
> restarting I am not able to hear any audio both sides,
>
> I have configured the settings by following the steps given here:
>
> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2
>
> I have allowed all traffic for inbound and outbound fro all IPs (Ec2 GUI
> security group)
>
> netstat -an | grep ':5060' gives
>
>
> tcp        0      0 172.31.22.224:5060      0.0.0.0:*               LISTEN
> tcp6       0      0 ::1:5060                :::*                    LISTEN
> udp        0      0 172.31.22.224:5060      0.0.0.0:*
> udp6       0      0 ::1:5060                :::*
>
> netstat -an | grep ':5080'
>
> tcp        0      0 172.31.22.224:5080      0.0.0.0:*               LISTEN
> tcp6       0      0 ::1:5080                :::*                    LISTEN
> udp        0      0 172.31.22.224:5080      0.0.0.0:*
> udp6       0      0 ::1:5080                :::*
>
> Any guidance to proceed would be highly thankful
>
> Regards,
> Deepika
>
>
>
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From blasterjr at gmail.com  Wed Apr 20 10:05:57 2016
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Wed, 20 Apr 2016 00:05:57 -0600
Subject: [Freeswitch-users] No audio Freesswitch
In-Reply-To: <CAM99_Yn028GLab=fZQgqF+5YgrZAwUKaMik93qpm_J0cD-h-bg@mail.gmail.com>
References: <CAM99_YnWuBGBaO3dg1yjxspM889fw2=J1EMo423GNPZ1Mb5AUg@mail.gmail.com>
	<CAM99_Yn028GLab=fZQgqF+5YgrZAwUKaMik93qpm_J0cD-h-bg@mail.gmail.com>
Message-ID: <CAO3hyzFnrNVEuJ73XBKymExoqyu2smvT7m8vPO8FKG9THE1rfw@mail.gmail.com>

You need to also forward RTP ports (UDP 16384-32768) you can see more
information here:
https://freeswitch.org/confluence/display/FREESWITCH/Firewall

if oyu're not forwarding those ports then no audio can pass through
freeswitch.

On Tue, Apr 19, 2016 at 11:48 PM, Deepika Yadav <deepikay at iiitd.ac.in>
wrote:

> On dialling 5000 from a softphone gives the following logs :
>
>  mod_dialplan_xml.c:635 Processing 1019 <1019>->5000 in context default
>  switch_core_media.c:5396 Activating RTCP PORT 4001
>  switch_rtp.c:3384 Activating Audio Secure RTP SEND
>  switch_rtp.c:3362 Activating Audio Secure RTP RECV
>
>
> On Wed, Apr 20, 2016 at 11:16 AM, Deepika Yadav <deepikay at iiitd.ac.in>
> wrote:
>
>> Hi,
>>
>> I am using Amazon ec2 platfrom to run freeswitch application. On
>> restarting I am not able to hear any audio both sides,
>>
>> I have configured the settings by following the steps given here:
>>
>> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2
>>
>> I have allowed all traffic for inbound and outbound fro all IPs (Ec2 GUI
>> security group)
>>
>> netstat -an | grep ':5060' gives
>>
>>
>> tcp        0      0 172.31.22.224:5060      0.0.0.0:*
>> LISTEN
>> tcp6       0      0 ::1:5060                :::*                    LISTEN
>> udp        0      0 172.31.22.224:5060      0.0.0.0:*
>> udp6       0      0 ::1:5060                :::*
>>
>> netstat -an | grep ':5080'
>>
>> tcp        0      0 172.31.22.224:5080      0.0.0.0:*
>> LISTEN
>> tcp6       0      0 ::1:5080                :::*                    LISTEN
>> udp        0      0 172.31.22.224:5080      0.0.0.0:*
>> udp6       0      0 ::1:5080                :::*
>>
>> Any guidance to proceed would be highly thankful
>>
>> Regards,
>> Deepika
>>
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From blasterjr at gmail.com  Wed Apr 20 10:06:28 2016
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Wed, 20 Apr 2016 00:06:28 -0600
Subject: [Freeswitch-users] No audio Freesswitch
In-Reply-To: <CAO3hyzFnrNVEuJ73XBKymExoqyu2smvT7m8vPO8FKG9THE1rfw@mail.gmail.com>
References: <CAM99_YnWuBGBaO3dg1yjxspM889fw2=J1EMo423GNPZ1Mb5AUg@mail.gmail.com>
	<CAM99_Yn028GLab=fZQgqF+5YgrZAwUKaMik93qpm_J0cD-h-bg@mail.gmail.com>
	<CAO3hyzFnrNVEuJ73XBKymExoqyu2smvT7m8vPO8FKG9THE1rfw@mail.gmail.com>
Message-ID: <CAO3hyzEee6C5Z81mwcsbVB3X8ndASUUy5kdyFMbV6r801jdDgg@mail.gmail.com>

You also need to set your ext-sip-ip and ext-rtp-ip inside your profiles..

On Wed, Apr 20, 2016 at 12:05 AM, Chris Tunbridge <blasterjr at gmail.com>
wrote:

> You need to also forward RTP ports (UDP 16384-32768) you can see more
> information here:
> https://freeswitch.org/confluence/display/FREESWITCH/Firewall
>
> if oyu're not forwarding those ports then no audio can pass through
> freeswitch.
>
> On Tue, Apr 19, 2016 at 11:48 PM, Deepika Yadav <deepikay at iiitd.ac.in>
> wrote:
>
>> On dialling 5000 from a softphone gives the following logs :
>>
>>  mod_dialplan_xml.c:635 Processing 1019 <1019>->5000 in context default
>>  switch_core_media.c:5396 Activating RTCP PORT 4001
>>  switch_rtp.c:3384 Activating Audio Secure RTP SEND
>>  switch_rtp.c:3362 Activating Audio Secure RTP RECV
>>
>>
>> On Wed, Apr 20, 2016 at 11:16 AM, Deepika Yadav <deepikay at iiitd.ac.in>
>> wrote:
>>
>>> Hi,
>>>
>>> I am using Amazon ec2 platfrom to run freeswitch application. On
>>> restarting I am not able to hear any audio both sides,
>>>
>>> I have configured the settings by following the steps given here:
>>>
>>> https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2
>>>
>>> I have allowed all traffic for inbound and outbound fro all IPs (Ec2 GUI
>>> security group)
>>>
>>> netstat -an | grep ':5060' gives
>>>
>>>
>>> tcp        0      0 172.31.22.224:5060      0.0.0.0:*
>>> LISTEN
>>> tcp6       0      0 ::1:5060                :::*
>>>  LISTEN
>>> udp        0      0 172.31.22.224:5060      0.0.0.0:*
>>> udp6       0      0 ::1:5060                :::*
>>>
>>> netstat -an | grep ':5080'
>>>
>>> tcp        0      0 172.31.22.224:5080      0.0.0.0:*
>>> LISTEN
>>> tcp6       0      0 ::1:5080                :::*
>>>  LISTEN
>>> udp        0      0 172.31.22.224:5080      0.0.0.0:*
>>> udp6       0      0 ::1:5080                :::*
>>>
>>> Any guidance to proceed would be highly thankful
>>>
>>> Regards,
>>> Deepika
>>>
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From mylists at polite.se  Wed Apr 20 12:08:41 2016
From: mylists at polite.se (Oivvio Polite)
Date: Wed, 20 Apr 2016 10:08:41 +0200
Subject: [Freeswitch-users] Verto vs SIP
Message-ID: <20160420080841.GA23645@blomma.liberationtech.net>

Summary:

I'm just getting started with VoIP/SIP/WebRTC. Following the FreeSwitch
Cookbook I've built a webclient that uses SIP signalling to establish a 
connection with FreeSwitch and then exchanges media with WebRTC. But I
get the impression from a lot of sources that SIP is a bad fit for
WebRTC. One datapoint that indicates this is the very existance of
mod_verto. Why would you bother writing it if SIP, that has 10+ years of
battle testing, was a good fit right?

What I don't understand yet is why SIP is a bad fit for WebRTC? Would
someone in the know care to muse about that?

regards Oivvio



From deepikay at iiitd.ac.in  Wed Apr 20 12:20:27 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Wed, 20 Apr 2016 13:50:27 +0530
Subject: [Freeswitch-users] mod_python ESL
Message-ID: <CAM99_Y=iq76p77wSL+=qAMd4p6S8zVb1ztkECCZWVJJkem8HtA@mail.gmail.com>

Hi,

I want to change the speaking status of a conference member through a
python script as:

import ESL
freeswitchcon =  ESL.ESLconnection(freeswitch_ip, freeswitch_port,
freeswitch_name)
cmd = conferencename + member_id
freeswitchcon.api("conference", str(cmd))

This api doesn't doesn't fired though "originate" command works

Another option :

import freeswitch

new_api_obj = API()
new_api_obj.executeString("conference "+conferencename+" "+member_id)

but this can't be called, due to module "_freeswitch" import error, instead
need to called via the dialplan that calls the handler, doesn't looks a
good way to do it

I am able to achieve this functionality in mod_java ESL simply using ;

org.freeswitch.swig.API a = new API(new JavaSession());
String rt = a.executeString("conference "+conferencename+" mute
"+member_id);

How can I achieve this in python?

Regards,
Deepika
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From mustafin.aleksandr at gmail.com  Wed Apr 20 16:26:47 2016
From: mustafin.aleksandr at gmail.com (Alexander Mustafin)
Date: Wed, 20 Apr 2016 17:26:47 +0500
Subject: [Freeswitch-users] mod_http_cache doesn't retrieve any file
Message-ID: <7F3D9325-69DC-479C-A642-65F42F311D3C@gmail.com>

Hi there!

I am testing mod_http_cache module and unfortunately it doesn?t work for me properly. I loaded module, and trying console commands such as ?http_prefetch? and others.

A respond of server look like:

freeswitch at fs.sip3.net> http_prefetch http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav
+OK

2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache
2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2
2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:718 Adding http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav(/var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav) to cache index 1
2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache
2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:1069 opening /var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav for URL cache
2016-04-20 12:22:38.810363 [ERR] mod_http_cache.c:1103 open() error: No such file or directory
2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache
2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:675 Failed to download URL http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav
2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache

And server does not try to obtain file from the url, because I listen 80 port and I can?t see anything.

Possibly the problem here '2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2? but I don?t know what  it?s about.

Any clue for this problem?


Best regards,
Alexander Mustafin
mustafin.aleksandr at gmail.com




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From italo at freeswitch.org  Wed Apr 20 16:38:07 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 20 Apr 2016 09:38:07 -0300
Subject: [Freeswitch-users] Verto Status Display
In-Reply-To: <571479D2.8050407@livematch.com>
References: <571479D2.8050407@livematch.com>
Message-ID: <CABMX_w-z5AfKnHZQc7J8GLscD3TXrEgcy=S_TcPb2YPeb9EOYA@mail.gmail.com>

Remove the flag on the conference profile, conference.conf.xml

On Mon, Apr 18, 2016 at 3:08 AM, Jack <jack at livematch.com> wrote:

> I am trying to show the shorter display for STATUS in the index.html  in
> the video_demo. So it shows like the Freeswitch conference status.  It
> seems like the  CFLAG_JSON_STATUS is set to true to display the detailed
> status.  Can someone tell me how to set this flag to false or where it
> is being set to true?
> Thanks,
>   Jack Loranger
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
italo at freeswitch.org
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From 35633 at heb.be  Wed Apr 20 17:02:01 2016
From: 35633 at heb.be (Joselyne Nduwayezu)
Date: Wed, 20 Apr 2016 15:02:01 +0200
Subject: [Freeswitch-users] Freeswitch and opensips
In-Reply-To: <5716248F.6060809@softnet.si>
References: <CAFSzUtb8hyWgY+sugSsUDd5BNRDVsVTCX5NUkqzq505e9+0rig@mail.gmail.com>
	<CAFSzUtZOjj5NfHbB7ne4QOSkufn76qzXCMniCT_QLjKcsQvzcw@mail.gmail.com>
	<5716248F.6060809@softnet.si>
Message-ID: <57177dc7.d3981c0a.91051.549a@mx.google.com>

When i directly register freeswitch at the provider,the incoming call to my line works well,but when i configure opensips in front of freeswitch,the nember rings busy.I thing i'm missing manu things regarding configuration,but i dont know how to solve them.I followed the site of opensipsFreeswitch integrtion wthout no result.Can someone tell me how to proceed or steps i must follow.I'm hopeless

-----Message d'origine-----
De : "Miha" <miha at softnet.si>
Envoy? : ?19-?04-?16 15:46
??: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Objet : Re: [Freeswitch-users] Freeswitch and opensips

1. you do not have to add another external profile but it would be nice :)
2. you have to add ip to that file.


miha


On 19/04/2016 13:59, Nduwayezu, Joselyne wrote:

Hello Younas,
You asked me to add u to skype in order  u to help me confoguring Opensips and Freeswitch, i sent the invitation since 13 April but u did not respond.Maybe you are busy or you did not recognize me. My name is Francjos on opensips mailing list and on skype i'm nduwayezu Joselyne.
Please, i need your help.


NDUWAYEZU Joselyne



2016-04-11 14:25 GMT+02:00 Nduwayezu, Joselyne <35633 at heb.be>:

I'm using Opensips as load balancer in front of FreeSwitch. To tell Freeswitch that calls come from Opensips proxy, do i have to create a new external profile in sip_profiles directory or add an extension in dialplan/public.xml or both of two? 

Second question, in this file :/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml , i read that i have to specify the CIDR, is the ip address  of Opensips? 

Thank you 






NDUWAYEZU Joselyne




Haute ?cole de Bruxelles 
 

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-- 
Haute ?cole de Bruxelles
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From jurijs.ivolga at gmail.com  Wed Apr 20 17:06:36 2016
From: jurijs.ivolga at gmail.com (Jurijs Ivolga)
Date: Wed, 20 Apr 2016 16:06:36 +0300
Subject: [Freeswitch-users] Freeswitch and opensips
In-Reply-To: <57177dc7.d3981c0a.91051.549a@mx.google.com>
References: <CAFSzUtb8hyWgY+sugSsUDd5BNRDVsVTCX5NUkqzq505e9+0rig@mail.gmail.com>
	<CAFSzUtZOjj5NfHbB7ne4QOSkufn76qzXCMniCT_QLjKcsQvzcw@mail.gmail.com>
	<5716248F.6060809@softnet.si>
	<57177dc7.d3981c0a.91051.549a@mx.google.com>
Message-ID: <CAOrA2UYyMzZG7D-_bkjj8HvLxqaELj_1G+99cqKi5Te70ZT8uA@mail.gmail.com>

Hi,

Please check this one:

http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc

It should mainly work out of the box, but it is not opensips, but Kamailio.
Opensips is a fork from Kamailio and they are pretty much similar.

With kind regards,

Jurijs

On Wed, Apr 20, 2016 at 4:02 PM, Joselyne Nduwayezu <35633 at heb.be> wrote:

> When i directly register freeswitch at the provider,the incoming call to
> my line works well,but when i configure opensips in front of freeswitch,the
> nember rings busy.I thing i'm missing manu things regarding
> configuration,but i dont know how to solve them.I followed the site of
> opensipsFreeswitch integrtion wthout no result.Can someone tell me how to
> proceed or steps i must follow.I'm hopeless
> ------------------------------
> De : Miha <miha at softnet.si>
> Envoy? : ?19-?04-?16 15:46
> ? : FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Objet : Re: [Freeswitch-users] Freeswitch and opensips
>
> 1. you do not have to add another external profile but it would be nice :)
> 2. you have to add ip to that file.
>
>
> miha
>
> On 19/04/2016 13:59, Nduwayezu, Joselyne wrote:
>
> Hello Younas,
> You asked me to add u to skype in order  u to help me confoguring Opensips
> and Freeswitch, i sent the invitation since 13 April but u did not
> respond.Maybe you are busy or you did not recognize me. My name is Francjos
> on opensips mailing list and on skype i'm nduwayezu Joselyne.
> Please, i need your help.
>
> NDUWAYEZU Joselyne
>
> 2016-04-11 14:25 GMT+02:00 Nduwayezu, Joselyne <35633 at heb.be>:
>
>> I'm using Opensips as load balancer in front of FreeSwitch. To tell
>> Freeswitch that calls come from Opensips proxy, do i have to create a new
>> external profile in sip_profiles directory or add an extension in
>> dialplan/public.xml or both of two?
>>
>> Second question, in this file
>> :/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml , i read that i
>> have to specify the CIDR, is the ip address  of Opensips?
>>
>> Thank you
>>
>>
>>
>> NDUWAYEZU Joselyne
>>
>
>
> Haute ?cole de Bruxelles
>
> _________________________________________________________________________
> 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
>
>
>
> Haute ?cole de Bruxelles
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From benjamin.cropley at gmail.com  Wed Apr 20 17:10:36 2016
From: benjamin.cropley at gmail.com (Benjamin Cropley)
Date: Wed, 20 Apr 2016 14:10:36 +0100
Subject: [Freeswitch-users] 'Call Pickup',
	replaces headers.. does FS handle them properly?
Message-ID: <CAF7jizK8NJ0+bmQHs4nGy+oDjGa2K+9JWcZE48B3mgb=4hNiFg@mail.gmail.com>

According to sect. 7.1 of  https://www.ietf.org/rfc/rfc3891.txt, when an
INVITE is recieved with a replaces header, containing the call-id of
another dialog, the receiving party needs to respond immediately with a
200OK containing the new INVITES call-id and cancelling the other call-id.
In the below, Alice calls Bobs Desk but he's not there. He's in his lab,
and hears the Desk ring, so does a dialoginfo call pickup..

                              Bob                      Bob
       Alice                  desk                     lab
        |                       |                        |
    *1  |-----INVITE----------->|                        |
    *2  |<----180---------------|  Bob hears desk phone  |
        |                       |  ringing from lab but  |
        |                       |  isn't REGISTERed yet  |
        |                       |                        |
        |                       |<--fetch dialog state --|
        |                       |---response ----------->|
   *3/4 |<-----INVITE with Replaces/200/ACK--------------|
   *5/6 |------CANCEL/200------>|                        |
   *7   |<-----487--------------|                        |
        |------ACK------------->|                        |
        |                       |                        |
        |                       |                        |
I have a setup of OpenSIPS and FreeSWITCH. OpenSIPS loadbalances all
INVITES to FreeSWITCH, and then any subsequent packets are just relayed on
as a stateful proxy. With that in mind, see what happens if I try do the
same..
https://pastebin.freeswitch.org/24660 - sip trace of..
1. Alice calls bobs desk
2. Bobs desk rings
3. Bobs lab tries to pick up the call, so that bobs lab is the one speaking
to alice and bobs desk stops ringing
4. Instead, Alice gets cut off and bobs desk ends up speaking to bob lab

https://pastebin.freeswitch.org/24650 - FS logs of this issue

As you can see, on line 19, Bobs lab sends an INVITE with a callid matching
that of the one Alice established... FS sends a 200OK back.. All good so
far. At this point I logically feel FS would then send a 200OK to the a
dialog/Alice and bridge the two.. but instead it kills the dialog with
alice and takes place directly with bobs desk..

tl;dr is FS handing replaces correctly here?

Any advice would be much appreciated.

Ben
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From igorolhovskiy at gmail.com  Wed Apr 20 17:11:06 2016
From: igorolhovskiy at gmail.com (Igor Olhovskiy)
Date: Wed, 20 Apr 2016 16:11:06 +0300
Subject: [Freeswitch-users] Verto vs SIP
In-Reply-To: <20160420080841.GA23645@blomma.liberationtech.net>
References: <20160420080841.GA23645@blomma.liberationtech.net>
Message-ID: <CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>

Hi!

>From my experience, verto is much more stable and more resource and
browser(js) friendly, than sipjs and other SIP implementations of SIP.
With various SIP libraries I?ve got strange issues, like sound delay (time
to time), not working audio in most fresh browsers, other some strange
issues. But verto was installed over a year ago and still working without
any troubles (with FS 1.5, I believe)
So, my voice is for verto.

2016-04-20 11:08 GMT+03:00 Oivvio Polite <mylists at polite.se>:

> Summary:
>
> I'm just getting started with VoIP/SIP/WebRTC. Following the FreeSwitch
> Cookbook I've built a webclient that uses SIP signalling to establish a
> connection with FreeSwitch and then exchanges media with WebRTC. But I
> get the impression from a lot of sources that SIP is a bad fit for
> WebRTC. One datapoint that indicates this is the very existance of
> mod_verto. Why would you bother writing it if SIP, that has 10+ years of
> battle testing, was a good fit right?
>
> What I don't understand yet is why SIP is a bad fit for WebRTC? Would
> someone in the know care to muse about that?
>
> regards Oivvio
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,
Igor
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From mylists at polite.se  Wed Apr 20 17:24:42 2016
From: mylists at polite.se (Oivvio Polite)
Date: Wed, 20 Apr 2016 15:24:42 +0200
Subject: [Freeswitch-users] Verto vs SIP
In-Reply-To: <CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>
References: <20160420080841.GA23645@blomma.liberationtech.net>
	<CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>
Message-ID: <20160420132442.GA23884@blomma.liberationtech.net>

On ons, apr 20, 2016 at 04:11:06 +0300, Igor Olhovskiy wrote:
> >From my experience, verto is much more stable and more resource and
> browser(js) friendly, than sipjs and other SIP implementations of SIP.
> With various SIP libraries I?ve got strange issues, like sound delay (time
> to time), not working audio in most fresh browsers, other some strange
> issues. But verto was installed over a year ago and still working without
> any troubles (with FS 1.5, I believe)
> So, my voice is for verto.
> 

This is exactly the kind of information I need (The kind that is never
in the docs). Thanks for sharing!

regards Oivvio


From cmrienzo at gmail.com  Wed Apr 20 17:34:47 2016
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Wed, 20 Apr 2016 09:34:47 -0400
Subject: [Freeswitch-users] mod_http_cache doesn't retrieve any file
In-Reply-To: <7F3D9325-69DC-479C-A642-65F42F311D3C@gmail.com>
References: <7F3D9325-69DC-479C-A642-65F42F311D3C@gmail.com>
Message-ID: <D9D14C3C-2B26-43B2-A8D7-10C5723F34A7@gmail.com>

Make sure FreeSWITCH process has permission to open /create directories in /var/cache/freeswitch and that directory exists. 

Chris


> On Apr 20, 2016, at 08:26, Alexander Mustafin <mustafin.aleksandr at gmail.com> wrote:
> 
> Hi there!
> 
> I am testing mod_http_cache module and unfortunately it doesn?t work for me properly. I loaded module, and trying console commands such as ?http_prefetch? and others.
> 
> A respond of server look like:
> 
> freeswitch at fs.sip3.net> http_prefetch http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav
> +OK
> 
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache
> 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:718 Adding http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav(/var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav) to cache index 1
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:1069 opening /var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav for URL cache
> 2016-04-20 12:22:38.810363 [ERR] mod_http_cache.c:1103 open() error: No such file or directory
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache
> 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:675 Failed to download URL http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache
> 
> And server does not try to obtain file from the url, because I listen 80 port and I can?t see anything.
> 
> Possibly the problem here '2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2? but I don?t know what  it?s about.
> 
> Any clue for this problem?
> 
> 
> Best regards,
> Alexander Mustafin
> mustafin.aleksandr at gmail.com
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Wed Apr 20 18:13:42 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Apr 2016 10:13:42 -0400
Subject: [Freeswitch-users] mod_python ESL
In-Reply-To: <CAM99_Y=iq76p77wSL+=qAMd4p6S8zVb1ztkECCZWVJJkem8HtA@mail.gmail.com>
References: <CAM99_Y=iq76p77wSL+=qAMd4p6S8zVb1ztkECCZWVJJkem8HtA@mail.gmail.com>
Message-ID: <CAEwTsAgv=0e5z7KBqp8Vu_p3yQaEu2TnGhszjKt-Mf4n8iGjxg@mail.gmail.com>

your Python example is missing the action,
conference confname memberid
^^^^ that's not a valid command

On Wednesday, April 20, 2016, Deepika Yadav <deepikay at iiitd.ac.in> wrote:

> Hi,
>
> I want to change the speaking status of a conference member through a
> python script as:
>
> import ESL
> freeswitchcon =  ESL.ESLconnection(freeswitch_ip, freeswitch_port,
> freeswitch_name)
> cmd = conferencename + member_id
> freeswitchcon.api("conference", str(cmd))
>
> This api doesn't doesn't fired though "originate" command works
>
> Another option :
>
> import freeswitch
>
> new_api_obj = API()
> new_api_obj.executeString("conference "+conferencename+" "+member_id)
>
> but this can't be called, due to module "_freeswitch" import error,
> instead need to called via the dialplan that calls the handler, doesn't
> looks a good way to do it
>
> I am able to achieve this functionality in mod_java ESL simply using ;
>
> org.freeswitch.swig.API a = new API(new JavaSession());
> String rt = a.executeString("conference "+conferencename+" mute
> "+member_id);
>
> How can I achieve this in python?
>
> Regards,
> Deepika
>
>
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From s.safarov at gmail.com  Wed Apr 20 18:40:32 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Wed, 20 Apr 2016 14:40:32 +0000
Subject: [Freeswitch-users] mod_http_cache doesn't retrieve any file
In-Reply-To: <D9D14C3C-2B26-43B2-A8D7-10C5723F34A7@gmail.com>
References: <7F3D9325-69DC-479C-A642-65F42F311D3C@gmail.com>
	<D9D14C3C-2B26-43B2-A8D7-10C5723F34A7@gmail.com>
Message-ID: <CAHtxdDd96yYb3e7H=q5GHUkQr2TaG4pY2=Y_HfkyUTzT4hCsig@mail.gmail.com>

Also check that host name "download.wavetlan.com" is not configured in
"hosts" file.
FreeSwitch is ignores hosts file.

Sergey

??, 20 ???. 2016 ?. ? 16:35, <cmrienzo at gmail.com>:

> Make sure FreeSWITCH process has permission to open /create directories in
> /var/cache/freeswitch and that directory exists.
>
> Chris
>
>
> On Apr 20, 2016, at 08:26, Alexander Mustafin <
> mustafin.aleksandr at gmail.com> wrote:
>
> Hi there!
>
> I am testing mod_http_cache module and unfortunately it doesn?t work for
> me properly. I loaded module, and trying console commands such as
> ?http_prefetch? and others.
>
> A respond of server look like:
>
> freeswitch at fs.sip3.net> http_prefetch
> http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav
> +OK
>
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache
> 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size =
> 1 (0 MB), hit ratio = 0/2
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:718 Adding
> http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav(/var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav)
> to cache index 1
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:1069 opening
> /var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav for URL
> cache
> 2016-04-20 12:22:38.810363 [ERR] mod_http_cache.c:1103 open() error: No
> such file or directory
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache
> 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:675 Failed to download
> URL
> http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache
>
> And server does not try to obtain file from the url, because I listen 80
> port and I can?t see anything.
>
> Possibly the problem here '2016-04-20 12:22:38.810363 [INFO]
> mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2? but I
> don?t know what  it?s about.
>
> Any clue for this problem?
>
>
> Best regards,
> Alexander Mustafin
> mustafin.aleksandr at gmail.com
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From bobjectsfreeswitch at gmail.com  Wed Apr 20 21:13:32 2016
From: bobjectsfreeswitch at gmail.com (Bob Hartwig)
Date: Wed, 20 Apr 2016 12:13:32 -0500
Subject: [Freeswitch-users] Verto vs SIP
In-Reply-To: <20160420132442.GA23884@blomma.liberationtech.net>
References: <20160420080841.GA23645@blomma.liberationtech.net>
	<CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>
	<20160420132442.GA23884@blomma.liberationtech.net>
Message-ID: <CAAemr3wdz0VbjVBSwS7-EYB8b7bMXPLRWfPQngJVTSth9b55NA@mail.gmail.com>

But stating that one client is more stable and better sounding than another
doesn't really answer the original question.  Why not use SIP over
WebSockets for signaling, when SIP is a proven, standard, flexible protocol?

Bob





On Wed, Apr 20, 2016 at 8:24 AM, Oivvio Polite <mylists at polite.se> wrote:

> On ons, apr 20, 2016 at 04:11:06 +0300, Igor Olhovskiy wrote:
> > >From my experience, verto is much more stable and more resource and
> > browser(js) friendly, than sipjs and other SIP implementations of SIP.
> > With various SIP libraries I?ve got strange issues, like sound delay
> (time
> > to time), not working audio in most fresh browsers, other some strange
> > issues. But verto was installed over a year ago and still working without
> > any troubles (with FS 1.5, I believe)
> > So, my voice is for verto.
> >
>
> This is exactly the kind of information I need (The kind that is never
> in the docs). Thanks for sharing!
>
> regards Oivvio
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Wed Apr 20 21:38:20 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 20 Apr 2016 12:38:20 -0500
Subject: [Freeswitch-users] Verto vs SIP
In-Reply-To: <CAAemr3wdz0VbjVBSwS7-EYB8b7bMXPLRWfPQngJVTSth9b55NA@mail.gmail.com>
References: <20160420080841.GA23645@blomma.liberationtech.net>
	<CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>
	<20160420132442.GA23884@blomma.liberationtech.net>
	<CAAemr3wdz0VbjVBSwS7-EYB8b7bMXPLRWfPQngJVTSth9b55NA@mail.gmail.com>
Message-ID: <CAKbxfG_0h7kV+x7tLMxDDkhgxvQrRwVojuUauunw1ZDr=_JQBA@mail.gmail.com>

SIP over websockets is fine for setting up and tearing down calls but keep
in mind that is mostly the only features it will ever support.
The reason to consider Verto would be that it has more of an event-driven
HTML5 design goal meant to be extensible to UI and gateway Web with media.
I actually implemented a large portion of all of the above and I think
Verto is a lighter more sensible media protocol for the web since I
designed it that way.
Also, it will only progress more from here in upcoming versions of FS.

P.S.
After 12 years implementing SIP I'm still trying to get over the comment
about proven, flexible and standard. ;)



On Wed, Apr 20, 2016 at 12:13 PM, Bob Hartwig <bobjectsfreeswitch at gmail.com>
wrote:

> But stating that one client is more stable and better sounding than
> another doesn't really answer the original question.  Why not use SIP over
> WebSockets for signaling, when SIP is a proven, standard, flexible protocol?
>
> Bob
>
>
>
>
>
> On Wed, Apr 20, 2016 at 8:24 AM, Oivvio Polite <mylists at polite.se> wrote:
>
>> On ons, apr 20, 2016 at 04:11:06 +0300, Igor Olhovskiy wrote:
>> > >From my experience, verto is much more stable and more resource and
>> > browser(js) friendly, than sipjs and other SIP implementations of SIP.
>> > With various SIP libraries I?ve got strange issues, like sound delay
>> (time
>> > to time), not working audio in most fresh browsers, other some strange
>> > issues. But verto was installed over a year ago and still working
>> without
>> > any troubles (with FS 1.5, I believe)
>> > So, my voice is for verto.
>> >
>>
>> This is exactly the kind of information I need (The kind that is never
>> in the docs). Thanks for sharing!
>>
>> regards Oivvio
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From arsenman at connectto.com  Wed Apr 20 22:55:48 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Wed, 20 Apr 2016 11:55:48 -0700
Subject: [Freeswitch-users] Verto communicator errors
In-Reply-To: <CAKbxfG_0h7kV+x7tLMxDDkhgxvQrRwVojuUauunw1ZDr=_JQBA@mail.gmail.com>
References: <20160420080841.GA23645@blomma.liberationtech.net>
	<CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>
	<20160420132442.GA23884@blomma.liberationtech.net>
	<CAAemr3wdz0VbjVBSwS7-EYB8b7bMXPLRWfPQngJVTSth9b55NA@mail.gmail.com>
	<CAKbxfG_0h7kV+x7tLMxDDkhgxvQrRwVojuUauunw1ZDr=_JQBA@mail.gmail.com>
Message-ID: <5717D0B4.2000409@connectto.com>


I installed verto communicator
After ext. registration trying to call
but as result have bunch of errors:
whats wrong ?


2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:59672 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:59672 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:59672 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-20 18:52:24.155150 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:59672 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:59672 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:59672 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:59672 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]


From italo at freeswitch.org  Thu Apr 21 00:46:44 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 20 Apr 2016 17:46:44 -0300
Subject: [Freeswitch-users] Verto communicator errors
In-Reply-To: <5717D0B4.2000409@connectto.com>
References: <20160420080841.GA23645@blomma.liberationtech.net>
	<CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>
	<20160420132442.GA23884@blomma.liberationtech.net>
	<CAAemr3wdz0VbjVBSwS7-EYB8b7bMXPLRWfPQngJVTSth9b55NA@mail.gmail.com>
	<CAKbxfG_0h7kV+x7tLMxDDkhgxvQrRwVojuUauunw1ZDr=_JQBA@mail.gmail.com>
	<5717D0B4.2000409@connectto.com>
Message-ID: <CABMX_w-_krA+HY8Wkrp+UnQ9DPOtVMrynQRXyjHiJrPVMs1Mqw@mail.gmail.com>

How are you serving verto communicator? Which web server? I'm suspecting
you're point your browser to a mod_verto endpoint which is not correct.

Please see
https://freeswitch.org/confluence/display/FREESWITCH/Verto+Communicator#VertoCommunicator-BuildingforProduction

On Wed, Apr 20, 2016 at 3:55 PM, Arsen Manukyan <arsenman at connectto.com>
wrote:

>
> I installed verto communicator
> After ext. registration trying to call
> but as result have bunch of errors:
> whats wrong ?
>
>
> 2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
> 10.110.11.16:59672 [{
>          "jsonrpc":      "2.0",
>          "error":        {
>                  "code": -32600,
>                  "message":      "Invalid Request"
>          },
>          "id":   null
> }]
> 2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
> 10.110.11.16:59672 [{
>          "jsonrpc":      "2.0",
>          "error":        {
>                  "code": -32600,
>                  "message":      "Invalid Request"
>          },
>          "id":   null
> }]
> 2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
> 10.110.11.16:59672 [{
>          "jsonrpc":      "2.0",
>          "error":        {
>                  "code": -32600,
>                  "message":      "Invalid Request"
>          },
>          "id":   null
> }]
> 2016-04-20 18:52:24.155150 [ALERT] mod_verto.c:604 WRITE
> 10.110.11.16:59672 [{
>          "jsonrpc":      "2.0",
>          "error":        {
>                  "code": -32600,
>                  "message":      "Invalid Request"
>          },
>          "id":   null
> }]
> 2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
> 10.110.11.16:59672 [{
>          "jsonrpc":      "2.0",
>          "error":        {
>                  "code": -32600,
>                  "message":      "Invalid Request"
>          },
>          "id":   null
> }]
> 2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
> 10.110.11.16:59672 [{
>          "jsonrpc":      "2.0",
>          "error":        {
>                  "code": -32600,
>                  "message":      "Invalid Request"
>          },
>          "id":   null
> }]
> 2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
> 10.110.11.16:59672 [{
>          "jsonrpc":      "2.0",
>          "error":        {
>                  "code": -32600,
>                  "message":      "Invalid Request"
>          },
>          "id":   null
> }]
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
italo at freeswitch.org
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From arsenman at connectto.com  Thu Apr 21 00:54:37 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Wed, 20 Apr 2016 13:54:37 -0700
Subject: [Freeswitch-users] Verto communicator errors
In-Reply-To: <CABMX_w-_krA+HY8Wkrp+UnQ9DPOtVMrynQRXyjHiJrPVMs1Mqw@mail.gmail.com>
References: <20160420080841.GA23645@blomma.liberationtech.net>
	<CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>
	<20160420132442.GA23884@blomma.liberationtech.net>
	<CAAemr3wdz0VbjVBSwS7-EYB8b7bMXPLRWfPQngJVTSth9b55NA@mail.gmail.com>
	<CAKbxfG_0h7kV+x7tLMxDDkhgxvQrRwVojuUauunw1ZDr=_JQBA@mail.gmail.com>
	<5717D0B4.2000409@connectto.com>
	<CABMX_w-_krA+HY8Wkrp+UnQ9DPOtVMrynQRXyjHiJrPVMs1Mqw@mail.gmail.com>
Message-ID: <5717EC8D.7070908@connectto.com>

its ngnix (fusionpbx)
i copied dist folder to my web server
browser point et to https://myserver/verto/index.html#/

sometimes during Loading i also receiving error: Provision failed

but extension registration is ok:  client 10.110.11.16:64765      
CONN_REG (WSS)





On 4/20/2016 1:46 PM, ?talo Rossi wrote:
> How are you serving verto communicator? Which web server? I'm 
> suspecting you're point your browser to a mod_verto endpoint which is 
> not correct.
>
> Please see 
> https://freeswitch.org/confluence/display/FREESWITCH/Verto+Communicator#VertoCommunicator-BuildingforProduction
>
> On Wed, Apr 20, 2016 at 3:55 PM, Arsen Manukyan 
> <arsenman at connectto.com <mailto:arsenman at connectto.com>> wrote:
>
>
>     I installed verto communicator
>     After ext. registration trying to call
>     but as result have bunch of errors:
>     whats wrong ?
>
>
>     2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
>     10.110.11.16:59672 <http://10.110.11.16:59672> [{
>              "jsonrpc":      "2.0",
>              "error":        {
>                      "code": -32600,
>                      "message":      "Invalid Request"
>              },
>              "id":   null
>     }]
>     2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
>     10.110.11.16:59672 <http://10.110.11.16:59672> [{
>              "jsonrpc":      "2.0",
>              "error":        {
>                      "code": -32600,
>                      "message":      "Invalid Request"
>              },
>              "id":   null
>     }]
>     2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
>     10.110.11.16:59672 <http://10.110.11.16:59672> [{
>              "jsonrpc":      "2.0",
>              "error":        {
>                      "code": -32600,
>                      "message":      "Invalid Request"
>              },
>              "id":   null
>     }]
>     2016-04-20 18:52:24.155150 [ALERT] mod_verto.c:604 WRITE
>     10.110.11.16:59672 <http://10.110.11.16:59672> [{
>              "jsonrpc":      "2.0",
>              "error":        {
>                      "code": -32600,
>                      "message":      "Invalid Request"
>              },
>              "id":   null
>     }]
>     2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
>     10.110.11.16:59672 <http://10.110.11.16:59672> [{
>              "jsonrpc":      "2.0",
>              "error":        {
>                      "code": -32600,
>                      "message":      "Invalid Request"
>              },
>              "id":   null
>     }]
>     2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
>     10.110.11.16:59672 <http://10.110.11.16:59672> [{
>              "jsonrpc":      "2.0",
>              "error":        {
>                      "code": -32600,
>                      "message":      "Invalid Request"
>              },
>              "id":   null
>     }]
>     2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
>     10.110.11.16:59672 <http://10.110.11.16:59672> [{
>              "jsonrpc":      "2.0",
>              "error":        {
>                      "code": -32600,
>                      "message":      "Invalid Request"
>              },
>              "id":   null
>     }]
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> ?talo Rossi
> italo at freeswitch.org <mailto:italo at freeswitch.org>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From clive at lansink.co.nz  Thu Apr 21 05:10:24 2016
From: clive at lansink.co.nz (Clive Lansink)
Date: Thu, 21 Apr 2016 13:10:24 +1200
Subject: [Freeswitch-users] Freeswitch and Ping
Message-ID: <mailman.2.1461201052.8220.freeswitch-users@lists.freeswitch.org>

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From mike at jerris.com  Thu Apr 21 06:52:30 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 20 Apr 2016 22:52:30 -0400
Subject: [Freeswitch-users] Freeswitch and Ping
In-Reply-To: <5718295c.e7c0320a.508e1.1f73SMTPIN_ADDED_MISSING@mx.google.com>
References: <5718295c.e7c0320a.508e1.1f73SMTPIN_ADDED_MISSING@mx.google.com>
Message-ID: <CAEwTsAjWPTHgXt2tnw+8TLa95+o4hE3+YB+3a0NZKeFYQ07kbA@mail.gmail.com>

sounds like a broken router to me

On Wednesday, April 20, 2016, Clive Lansink <clive at lansink.co.nz> wrote:

>
> Hi everyone. I often use Freeswitch set up as a client to provide a soft
> phone on my laptop.
>
> But recently I noticed a significant problem when at the same time I was
> running a standard Windows "ping" in the background. The speech I heard
> from the other end would often stutter, and the remote party complained
> that my speech sometimes sounded gittery.
>
> I reasoned that ping doesn't take up much network bandwidth so while in
> conversation I did something like copy a large file from my laptop to
> another PC on the network and there was no disturbance to the conversation.
> It only seems to be ping that causes this.
>
> Is there something about pinging that might impact on Freeswitch or VOIP?
> What would cause this?
>
> Further information.
>
> the above refers to the standard Windows ping command. I also have a
> Python script that pings a specified host, and running it also caused the
> call quality to degrade. I tried various changes to the source code to no
> effect, including:
> * only creating the socket once and then reusing it instead of it being
> created for each ping.
> * Only creating the data packet once and reusing it instead of it being
> created for each ping.
> * Using a very short packet size.
> * Eliminating the select statement and just sleeping for a period of time
> before reading the response in case the select itself was at fault.
> * Eliminating the step of dissecting and decoding the received packet.
> None of these made any difference so it seems to me that whatever it is
> must be something quite fundamental to something at the lower networking
> level.
>
> Thank you.
>
>
> Clive Lansink
> Email: Clive at Lansink.Co.NZ <javascript:;>
> Phone: +64 9 520-4242
> Mobile: +64 21 663-999
> Fax: +64 21 789-150
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From miha at softnet.si  Thu Apr 21 10:06:43 2016
From: miha at softnet.si (Miha)
Date: Thu, 21 Apr 2016 08:06:43 +0200
Subject: [Freeswitch-users] Nat issue
Message-ID: <57186DF3.4000800@softnet.si>

Hi

our FS is on public ip but the voip server to which trunk is made (SFB) 
is behind NAT.

So the issue is that in SDP is private ip and FS sends RTP to this 
private ip, it does not change to src ip from which sip request was 
received.

Remote SDP:
2016-04-21 08:04:00.954752 [DEBUG] sofia.c:5614 Remote SDP:
v=0
o=- 46176 1 IN IP4 172.16.2.3
s=session
c=IN IP4 172.16.2.3
b=CT:1000
t=0 0
m=audio 31200 RTP/AVP 8 101 13
c=IN IP4 172.16.2.3
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtcp:31201
a=label:Audio


And then:

  xxx.xxx.xxx.xxx (FS public ip) port 17094 -> 172.16.2.3 port 31200 
codec: 8 ms: 30


I tried with:
     <param name="NDLB-force-rport" value="true"/>
     <param name="aggressive-nat-detection" value="true"/>


on internal profile and on trunk profile but the same.


tnx for help!
Miha


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From gregor at infomedia.si  Thu Apr 21 11:49:00 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 21 Apr 2016 09:49:00 +0200
Subject: [Freeswitch-users] Verto
Message-ID: <CAC7ttFE6C_CDkV79vOcgV9YKw+sJALT7bcGf_5RJ5KO8EcXb4A@mail.gmail.com>

?I am building my web phone with verto from scratch and using this tutorial:

http://evoluxbr.github.io/verto-docs/

BAsically everything is ok, I can register and make calls, but I cannot
hear audio. Mic is working, but not speakers. I see error:

Cannot read property 'sinkId' of undefined. This is error line:

$.verto.dialog.prototype.setAudioPlaybackDevice = function(sinkId,
callback, arg) {
var dialog = this;
var element = dialog.audioStream;

* if (typeof element.sinkId !== 'undefined') {*

I get dialog object without audioStream property and hence element is null.

I am using self signed certificate. Could this be cause?

Best regards, Gregor
?
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From mylists at polite.se  Thu Apr 21 12:06:28 2016
From: mylists at polite.se (Oivvio Polite)
Date: Thu, 21 Apr 2016 10:06:28 +0200
Subject: [Freeswitch-users] Verto vs SIP
In-Reply-To: <CAKbxfG_0h7kV+x7tLMxDDkhgxvQrRwVojuUauunw1ZDr=_JQBA@mail.gmail.com>
References: <20160420080841.GA23645@blomma.liberationtech.net>
	<CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>
	<20160420132442.GA23884@blomma.liberationtech.net>
	<CAAemr3wdz0VbjVBSwS7-EYB8b7bMXPLRWfPQngJVTSth9b55NA@mail.gmail.com>
	<CAKbxfG_0h7kV+x7tLMxDDkhgxvQrRwVojuUauunw1ZDr=_JQBA@mail.gmail.com>
Message-ID: <20160421080628.GA29840@blomma.liberationtech.net>

On ons, apr 20, 2016 at 12:38:20 -0500, Anthony Minessale wrote:
> SIP over websockets is fine for setting up and tearing down calls but keep
> in mind that is mostly the only features it will ever support.
> The reason to consider Verto would be that it has more of an event-driven
> HTML5 design goal meant to be extensible to UI and gateway Web with media.
> I actually implemented a large portion of all of the above and I think
> Verto is a lighter more sensible media protocol for the web since I
> designed it that way.

Thanks for the explanation.

> Also, it will only progress more from here in upcoming versions of FS.
> 

That's great to hear! What's on the roadmap for Verto?

Oivvio



From mustafin.aleksandr at gmail.com  Thu Apr 21 13:24:17 2016
From: mustafin.aleksandr at gmail.com (Alexander Mustafin)
Date: Thu, 21 Apr 2016 14:24:17 +0500
Subject: [Freeswitch-users] mod_http_cache doesn't retrieve any file
In-Reply-To: <D9D14C3C-2B26-43B2-A8D7-10C5723F34A7@gmail.com>
References: <7F3D9325-69DC-479C-A642-65F42F311D3C@gmail.com>
	<D9D14C3C-2B26-43B2-A8D7-10C5723F34A7@gmail.com>
Message-ID: <F86AD2B9-AAFD-4CA8-AD9A-0593CF9BE52F@gmail.com>

Thanks for helping me, guys. 

The directory /var/cache/freeswitch didn?t exist (Debian 8.4, debian packages) . I?ve created this one and set appropriate permissions.

Now all works fine!

Best regards,
Alexander Mustafin
mustafin.aleksandr at gmail.com




> 20 ???. 2016 ?., ? 18:34, cmrienzo at gmail.com ???????(?):
> 
> Make sure FreeSWITCH process has permission to open /create directories in /var/cache/freeswitch and that directory exists. 
> 
> Chris
> 
> 
> On Apr 20, 2016, at 08:26, Alexander Mustafin <mustafin.aleksandr at gmail.com <mailto:mustafin.aleksandr at gmail.com>> wrote:
> 
>> Hi there!
>> 
>> I am testing mod_http_cache module and unfortunately it doesn?t work for me properly. I loaded module, and trying console commands such as ?http_prefetch? and others.
>> 
>> A respond of server look like:
>> 
>> freeswitch at fs.sip3.net <mailto:freeswitch at fs.sip3.net>> http_prefetch http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav <http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav>
>> +OK
>> 
>> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache
>> 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2
>> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:718 Adding http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav(/var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav) <http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav(/var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav)> to cache index 1
>> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache
>> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:1069 opening /var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav for URL cache
>> 2016-04-20 12:22:38.810363 [ERR] mod_http_cache.c:1103 open() error: No such file or directory
>> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache
>> 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:675 Failed to download URL http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav <http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav>
>> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache
>> 
>> And server does not try to obtain file from the url, because I listen 80 port and I can?t see anything.
>> 
>> Possibly the problem here '2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2? but I don?t know what  it?s about.
>> 
>> Any clue for this problem?
>> 
>> 
>> Best regards,
>> Alexander Mustafin
>> mustafin.aleksandr at gmail.com <mailto:mustafin.aleksandr at gmail.com>
>> 
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Thu Apr 21 16:42:07 2016
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Apr 2016 07:42:07 -0500
Subject: [Freeswitch-users] mod_http_cache doesn't retrieve any file
In-Reply-To: <F86AD2B9-AAFD-4CA8-AD9A-0593CF9BE52F@gmail.com>
References: <7F3D9325-69DC-479C-A642-65F42F311D3C@gmail.com>
	<D9D14C3C-2B26-43B2-A8D7-10C5723F34A7@gmail.com>
	<F86AD2B9-AAFD-4CA8-AD9A-0593CF9BE52F@gmail.com>
Message-ID: <CAEJMVkABHEdWgZEn8LMgvSk=CHGvy5dZsc4bfVnyh5gj_3dh1w@mail.gmail.com>

Please file a JIRA

On Thursday, April 21, 2016, Alexander Mustafin <
mustafin.aleksandr at gmail.com> wrote:

> Thanks for helping me, guys.
>
> The directory /var/cache/freeswitch didn?t exist (Debian 8.4, debian
> packages) . I?ve created this one and set appropriate permissions.
>
> Now all works fine!
>
> Best regards,
> Alexander Mustafin
> mustafin.aleksandr at gmail.com
> <javascript:_e(%7B%7D,'cvml','mustafin.aleksandr at gmail.com');>
>
>
>
>
> 20 ???. 2016 ?., ? 18:34, cmrienzo at gmail.com
> <javascript:_e(%7B%7D,'cvml','cmrienzo at gmail.com');> ???????(?):
>
> Make sure FreeSWITCH process has permission to open /create directories in
> /var/cache/freeswitch and that directory exists.
>
> Chris
>
>
> On Apr 20, 2016, at 08:26, Alexander Mustafin <
> mustafin.aleksandr at gmail.com
> <javascript:_e(%7B%7D,'cvml','mustafin.aleksandr at gmail.com');>> wrote:
>
> Hi there!
>
> I am testing mod_http_cache module and unfortunately it doesn?t work for
> me properly. I loaded module, and trying console commands such as
> ?http_prefetch? and others.
>
> A respond of server look like:
>
> freeswitch at fs.sip3.net
> <javascript:_e(%7B%7D,'cvml','freeswitch at fs.sip3.net');>> http_prefetch
> http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav
> +OK
>
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache
> 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size =
> 1 (0 MB), hit ratio = 0/2
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:718 Adding
> http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav(/var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav)
> to cache index 1
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:1069 opening
> /var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav for URL
> cache
> 2016-04-20 12:22:38.810363 [ERR] mod_http_cache.c:1103 open() error: No
> such file or directory
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache
> 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:675 Failed to download
> URL
> http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert_test4_Ulaw_Mono_VBR_8SS_22050Hz.wav
> 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache
>
> And server does not try to obtain file from the url, because I listen 80
> port and I can?t see anything.
>
> Possibly the problem here '2016-04-20 12:22:38.810363 [INFO]
> mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2? but I
> don?t know what  it?s about.
>
> Any clue for this problem?
>
>
> Best regards,
> Alexander Mustafin
> mustafin.aleksandr at gmail.com
> <javascript:_e(%7B%7D,'cvml','mustafin.aleksandr at gmail.com');>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>

-- 

*Brian West*
brian at freeswitch.org


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

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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From luis.daniel.lucio at gmail.com  Thu Apr 21 17:01:39 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Thu, 21 Apr 2016 09:01:39 -0400
Subject: [Freeswitch-users] C11 style in libyuv?
In-Reply-To: <CAFLo2QzAKELccfJyfaKrVUFqG5wv7WTeEehrT0hhz335Q8fi8Q@mail.gmail.com>
References: <CAFLo2QxtmtByrBU3K-9tHHGOz2PiroQSos0L8XghTd7BEBNF=w@mail.gmail.com>
	<CAFLo2QyThDn50PMRsgndZZaa2ioe8VLMKtQc45B0-4m2JLwxoA@mail.gmail.com>
	<CAFLo2QzAKELccfJyfaKrVUFqG5wv7WTeEehrT0hhz335Q8fi8Q@mail.gmail.com>
Message-ID: <CAFLo2Qz84L+YOJ7GfKJ3jt4v03iqdyMQ7aFMNNziWV+-57pOQA@mail.gmail.com>

Hello

Is it my imagination of I see c11 style in libyuv (the one bundled in fs
1.6.7)?

This may discard older compilers.

Ld
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From mike at jerris.com  Thu Apr 21 17:17:16 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 21 Apr 2016 09:17:16 -0400
Subject: [Freeswitch-users] C11 style in libyuv?
In-Reply-To: <CAFLo2Qz84L+YOJ7GfKJ3jt4v03iqdyMQ7aFMNNziWV+-57pOQA@mail.gmail.com>
References: <CAFLo2QxtmtByrBU3K-9tHHGOz2PiroQSos0L8XghTd7BEBNF=w@mail.gmail.com>
	<CAFLo2QyThDn50PMRsgndZZaa2ioe8VLMKtQc45B0-4m2JLwxoA@mail.gmail.com>
	<CAFLo2QzAKELccfJyfaKrVUFqG5wv7WTeEehrT0hhz335Q8fi8Q@mail.gmail.com>
	<CAFLo2Qz84L+YOJ7GfKJ3jt4v03iqdyMQ7aFMNNziWV+-57pOQA@mail.gmail.com>
Message-ID: <46ED281E-7B45-4B15-B939-145A8ECF589C@jerris.com>

There may be an issue there, but I've taken a wait and see approach on that and see what it really impacts.  For now you can disable libyuv support, but it will of course disable all the video support.  In practice expecting newer code to use newer toolchains to compile isn't an unreasonable requirement.

> On Apr 21, 2016, at 9:01 AM, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com> wrote:
> 
> Hello
> 
> Is it my imagination of I see c11 style in libyuv (the one bundled in fs 1.6.7)?
> 
> This may discard older compilers.
> 
> Ld
> 

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From rick.jarvis at magicmail.mooo.com  Thu Apr 21 18:11:39 2016
From: rick.jarvis at magicmail.mooo.com (Rick Jarvis)
Date: Thu, 21 Apr 2016 15:11:39 +0100
Subject: [Freeswitch-users] Pulse / portaudio / sharing
Message-ID: <15A10D71-51AB-47BB-ACED-4E7F7C2CF27D@magicmail.mooo.com>

I?d like to monitor the audio levels on an audio interface that?s being used by mod_portaudio

I?m guessing, but I don?t know, that the best way of doing this would be to use Pulse Audio, either directly or indirectly.

I don?t know a huge deal about Pulse Audio though, and while it seems that there?s a mod_pulseaudio available, I can?t find much information on it, and there seem to be quite old unanswered questions about Pulse in relation to FreeSWITCH, on this list.

Can anyone give me any suggestions on which way I might head?

From italo at freeswitch.org  Thu Apr 21 21:57:51 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 21 Apr 2016 14:57:51 -0300
Subject: [Freeswitch-users] Verto communicator errors
In-Reply-To: <5717EC8D.7070908@connectto.com>
References: <20160420080841.GA23645@blomma.liberationtech.net>
	<CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>
	<20160420132442.GA23884@blomma.liberationtech.net>
	<CAAemr3wdz0VbjVBSwS7-EYB8b7bMXPLRWfPQngJVTSth9b55NA@mail.gmail.com>
	<CAKbxfG_0h7kV+x7tLMxDDkhgxvQrRwVojuUauunw1ZDr=_JQBA@mail.gmail.com>
	<5717D0B4.2000409@connectto.com>
	<CABMX_w-_krA+HY8Wkrp+UnQ9DPOtVMrynQRXyjHiJrPVMs1Mqw@mail.gmail.com>
	<5717EC8D.7070908@connectto.com>
Message-ID: <CABMX_w_X5QS6K1SovQr+9YEbQxcpDi5uVpJquSkj4hH2F8Bk9A@mail.gmail.com>

Don't access the index.html directly, just  https://myserver/verto/
<https://myserver/verto/index.html#/> .

On Wed, Apr 20, 2016 at 5:54 PM, Arsen Manukyan <arsenman at connectto.com>
wrote:

> its ngnix (fusionpbx)
> i copied dist folder to my web server
> browser point et to https://myserver/verto/index.html#/
>
> sometimes during Loading i also receiving error: Provision failed
>
> but extension registration is ok:  client
> 10.110.11.16:64765      CONN_REG (WSS)
>
>
>
>
>
>
> On 4/20/2016 1:46 PM, ?talo Rossi wrote:
>
> How are you serving verto communicator? Which web server? I'm suspecting
> you're point your browser to a mod_verto endpoint which is not correct.
>
> Please see
> https://freeswitch.org/confluence/display/FREESWITCH/Verto+Communicator#VertoCommunicator-BuildingforProduction
>
> On Wed, Apr 20, 2016 at 3:55 PM, Arsen Manukyan <arsenman at connectto.com>
> wrote:
>
>>
>> I installed verto communicator
>> After ext. registration trying to call
>> but as result have bunch of errors:
>> whats wrong ?
>>
>>
>> 2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
>> 10.110.11.16:59672 [{
>>          "jsonrpc":      "2.0",
>>          "error":        {
>>                  "code": -32600,
>>                  "message":      "Invalid Request"
>>          },
>>          "id":   null
>> }]
>> 2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
>> 10.110.11.16:59672 [{
>>          "jsonrpc":      "2.0",
>>          "error":        {
>>                  "code": -32600,
>>                  "message":      "Invalid Request"
>>          },
>>          "id":   null
>> }]
>> 2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
>> 10.110.11.16:59672 [{
>>          "jsonrpc":      "2.0",
>>          "error":        {
>>                  "code": -32600,
>>                  "message":      "Invalid Request"
>>          },
>>          "id":   null
>> }]
>> 2016-04-20 18:52:24.155150 [ALERT] mod_verto.c:604 WRITE
>> 10.110.11.16:59672 [{
>>          "jsonrpc":      "2.0",
>>          "error":        {
>>                  "code": -32600,
>>                  "message":      "Invalid Request"
>>          },
>>          "id":   null
>> }]
>> 2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
>> 10.110.11.16:59672 [{
>>          "jsonrpc":      "2.0",
>>          "error":        {
>>                  "code": -32600,
>>                  "message":      "Invalid Request"
>>          },
>>          "id":   null
>> }]
>> 2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
>> 10.110.11.16:59672 [{
>>          "jsonrpc":      "2.0",
>>          "error":        {
>>                  "code": -32600,
>>                  "message":      "Invalid Request"
>>          },
>>          "id":   null
>> }]
>> 2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
>> 10.110.11.16:59672 [{
>>          "jsonrpc":      "2.0",
>>          "error":        {
>>                  "code": -32600,
>>                  "message":      "Invalid Request"
>>          },
>>          "id":   null
>> }]
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?talo Rossi
> italo at freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.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
>



-- 
?talo Rossi
italo at freeswitch.org
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From italo at freeswitch.org  Thu Apr 21 21:59:27 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Thu, 21 Apr 2016 14:59:27 -0300
Subject: [Freeswitch-users] Verto
In-Reply-To: <CAC7ttFE6C_CDkV79vOcgV9YKw+sJALT7bcGf_5RJ5KO8EcXb4A@mail.gmail.com>
References: <CAC7ttFE6C_CDkV79vOcgV9YKw+sJALT7bcGf_5RJ5KO8EcXb4A@mail.gmail.com>
Message-ID: <CABMX_w9vZCmqdkgAZ00FB1WZqT=hRK5hvtHF0tmdsGPWmAjZFQ@mail.gmail.com>

Hey Gregor,

Can you double check if you're using the most recent version of verto js
lib?

On Thu, Apr 21, 2016 at 4:49 AM, Gregor Nanger <gregor at infomedia.si> wrote:

> ?I am building my web phone with verto from scratch and using this
> tutorial:
>
> http://evoluxbr.github.io/verto-docs/
>
> BAsically everything is ok, I can register and make calls, but I cannot
> hear audio. Mic is working, but not speakers. I see error:
>
> Cannot read property 'sinkId' of undefined. This is error line:
>
> $.verto.dialog.prototype.setAudioPlaybackDevice = function(sinkId,
> callback, arg) {
> var dialog = this;
> var element = dialog.audioStream;
>
> * if (typeof element.sinkId !== 'undefined') {*
>
> I get dialog object without audioStream property and hence element is null.
>
> I am using self signed certificate. Could this be cause?
>
> Best regards, Gregor
> ?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
italo at freeswitch.org
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From amani.mansour2 at gmail.com  Fri Apr 22 01:27:40 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Thu, 21 Apr 2016 22:27:40 +0100
Subject: [Freeswitch-users] How to send Re_invite
Message-ID: <CAFETSWjbsGh621pDmSpUM=CFUEop5ka_zpHOzn7r515gogReqw@mail.gmail.com>

Hi all,


I need to configure FS to send 491 request pending message after receiving
Re invite from the caller ,

The problem  that i don't know how to send the re invite request ,which xml
file i whould modifie it ? please help me ,i have done many research but i
didn't find anything .

Thank you ,

Best Regards
Amani
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From abalashov at evaristesys.com  Fri Apr 22 01:30:21 2016
From: abalashov at evaristesys.com (Alex Balashov)
Date: Thu, 21 Apr 2016 17:30:21 -0400
Subject: [Freeswitch-users] How to send Re_invite
In-Reply-To: <CAFETSWjbsGh621pDmSpUM=CFUEop5ka_zpHOzn7r515gogReqw@mail.gmail.com>
References: <CAFETSWjbsGh621pDmSpUM=CFUEop5ka_zpHOzn7r515gogReqw@mail.gmail.com>
Message-ID: <20160421213021.5394500.19050.232651@evaristesys.com>

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From krice at freeswitch.org  Fri Apr 22 02:30:52 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 21 Apr 2016 17:30:52 -0500
Subject: [Freeswitch-users] How to send Re_invite
In-Reply-To: <CAFETSWjbsGh621pDmSpUM=CFUEop5ka_zpHOzn7r515gogReqw@mail.gmail.com>
References: <CAFETSWjbsGh621pDmSpUM=CFUEop5ka_zpHOzn7r515gogReqw@mail.gmail.com>
Message-ID: <047e01d19c1d$777906e0$666b14a0$@freeswitch.org>

What specifically are you trying to accomplish? 

 

While you can configure freeswitch to respond to invites in a specific way, this is intended to use to terminate a call not pause it for continuation later.

 

Most of the SIP handing is intentionally hidden from the user with FreeSWITCH. 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of amani mansour
Sent: Thursday, April 21, 2016 4:28 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] How to send Re_invite

 

Hi all, 

 

 

I need to configure FS to send 491 request pending message after receiving Re invite from the caller ,

 

The problem  that i don't know how to send the re invite request ,which xml file i whould modifie it ? please help me ,i have done many research but i didn't find anything .

 

Thank you ,

 

Best Regards 

Amani

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From gregor at infomedia.si  Fri Apr 22 02:31:31 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Fri, 22 Apr 2016 00:31:31 +0200
Subject: [Freeswitch-users] Verto
In-Reply-To: <CABMX_w9vZCmqdkgAZ00FB1WZqT=hRK5hvtHF0tmdsGPWmAjZFQ@mail.gmail.com>
References: <CAC7ttFE6C_CDkV79vOcgV9YKw+sJALT7bcGf_5RJ5KO8EcXb4A@mail.gmail.com>
	<CABMX_w9vZCmqdkgAZ00FB1WZqT=hRK5hvtHF0tmdsGPWmAjZFQ@mail.gmail.com>
Message-ID: <CAC7ttFGGfpAdcbRfpec3-FRbunRBVGOpvJOgcPQ=Siycsem-gQ@mail.gmail.com>

Thank you for answer.

I think that something is conflicting with other javascript library. I am
integrating verto in other solution (angularjs and bunch of other
javascripts). Demo is also working ok in my environment.

How can I be sure that I am using latest lib? Should I just download latest
branch?







2016-04-21 19:59 GMT+02:00 ?talo Rossi <italo at freeswitch.org>:

> Hey Gregor,
>
> Can you double check if you're using the most recent version of verto js
> lib?
>
> On Thu, Apr 21, 2016 at 4:49 AM, Gregor Nanger <gregor at infomedia.si>
> wrote:
>
>> ?I am building my web phone with verto from scratch and using this
>> tutorial:
>>
>> http://evoluxbr.github.io/verto-docs/
>>
>> BAsically everything is ok, I can register and make calls, but I cannot
>> hear audio. Mic is working, but not speakers. I see error:
>>
>> Cannot read property 'sinkId' of undefined. This is error line:
>>
>> $.verto.dialog.prototype.setAudioPlaybackDevice = function(sinkId,
>> callback, arg) {
>> var dialog = this;
>> var element = dialog.audioStream;
>>
>> * if (typeof element.sinkId !== 'undefined') {*
>>
>> I get dialog object without audioStream property and hence element is
>> null.
>>
>> I am using self signed certificate. Could this be cause?
>>
>> Best regards, Gregor
>> ?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> ?talo Rossi
> italo at freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
? www.infomedia.si
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From gregor at infomedia.si  Fri Apr 22 02:34:37 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Fri, 22 Apr 2016 00:34:37 +0200
Subject: [Freeswitch-users] Verto latency
Message-ID: <CAC7ttFFtZSwejG2_sRzg04ySB+_VCJ5ORVaiZryFJtnA+Pav3w@mail.gmail.com>

If I make call from SIP client everything is ok. But if I make call from
verto demo client, there is 2 second delay of audio in direction from
browser (latest chrome) to callee. Other direction is ok.

Is it posible to tweak lower this latency?
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From mandra at gmail.com  Fri Apr 22 03:26:47 2016
From: mandra at gmail.com (Chris Mandra)
Date: Thu, 21 Apr 2016 19:26:47 -0400
Subject: [Freeswitch-users] module not unloading
Message-ID: <CAJdRzN5hseXNb7ehuEkMFnR6Jy6Un=QRg+aF7tJc-cEDOX=SGg@mail.gmail.com>

Hi Guys - I hope you are well
I never really resolved the issue I was having with modules saying they're
unloading, but not really unloading (unless I restart freeswitch)

I?m having that issue with the module not unloading again when I add in
some flex/bison parser code. It seems the parser is causing the problem
where module doesn?t unload. What's the best way to diagnose this issue.
 freeswitch reports module as unloading but lsof shows it is still being
retained
Thanks!
chris
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From arsenman at connectto.com  Fri Apr 22 04:15:18 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Thu, 21 Apr 2016 17:15:18 -0700
Subject: [Freeswitch-users] Verto communicator errors
In-Reply-To: <CABMX_w_X5QS6K1SovQr+9YEbQxcpDi5uVpJquSkj4hH2F8Bk9A@mail.gmail.com>
References: <20160420080841.GA23645@blomma.liberationtech.net>
	<CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>
	<20160420132442.GA23884@blomma.liberationtech.net>
	<CAAemr3wdz0VbjVBSwS7-EYB8b7bMXPLRWfPQngJVTSth9b55NA@mail.gmail.com>
	<CAKbxfG_0h7kV+x7tLMxDDkhgxvQrRwVojuUauunw1ZDr=_JQBA@mail.gmail.com>
	<5717D0B4.2000409@connectto.com>
	<CABMX_w-_krA+HY8Wkrp+UnQ9DPOtVMrynQRXyjHiJrPVMs1Mqw@mail.gmail.com>
	<5717EC8D.7070908@connectto.com>
	<CABMX_w_X5QS6K1SovQr+9YEbQxcpDi5uVpJquSkj4hH2F8Bk9A@mail.gmail.com>
Message-ID: <babec3e0-2824-4750-28df-ce5fbea612b4@connectto.com>

?talo, thank you

Now i am calling directly https://myserver/verto/ 
<https://myserver/verto/index.html#/>

but now faced with other problem: during loading Verto communicator 
progress bar stock on "Check connection Speed", and extension i see 
registred

again errors in logs:

2016-04-22 00:14:27.475160 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:49615 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-22 00:14:27.475160 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:49615 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-22 00:14:27.475160 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:49615 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-22 00:14:27.475160 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:49615 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-22 00:14:27.475160 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:49615 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-22 00:14:27.475160 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:49615 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-22 00:14:27.475160 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:49615 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-22 00:14:27.475160 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:49615 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-22 00:14:27.475160 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:49615 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-22 00:14:27.475160 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:49615 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]
2016-04-22 00:14:27.475160 [ALERT] mod_verto.c:604 WRITE 
10.110.11.16:49615 [{
         "jsonrpc":      "2.0",
         "error":        {
                 "code": -32600,
                 "message":      "Invalid Request"
         },
         "id":   null
}]





On 4/21/2016 10:57 AM, ?talo Rossi wrote:
> Don't access the index.html directly, just https://myserver/verto/ 
> <https://myserver/verto/index.html#/> .
>
> On Wed, Apr 20, 2016 at 5:54 PM, Arsen Manukyan 
> <arsenman at connectto.com <mailto:arsenman at connectto.com>> wrote:
>
>     its ngnix (fusionpbx)
>     i copied dist folder to my web server
>     browser point et to https://myserver/verto/index.html#/
>
>     sometimes during Loading i also receiving error: Provision failed
>
>     but extension registration is ok: client 10.110.11.16:64765
>     <http://10.110.11.16:64765> CONN_REG (WSS)
>
>
>
>
>
>
>     On 4/20/2016 1:46 PM, ?talo Rossi wrote:
>>     How are you serving verto communicator? Which web server? I'm
>>     suspecting you're point your browser to a mod_verto endpoint
>>     which is not correct.
>>
>>     Please see
>>     https://freeswitch.org/confluence/display/FREESWITCH/Verto+Communicator#VertoCommunicator-BuildingforProduction
>>
>>     On Wed, Apr 20, 2016 at 3:55 PM, Arsen Manukyan
>>     <arsenman at connectto.com <mailto:arsenman at connectto.com>> wrote:
>>
>>
>>         I installed verto communicator
>>         After ext. registration trying to call
>>         but as result have bunch of errors:
>>         whats wrong ?
>>
>>
>>         2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
>>         10.110.11.16:59672 <http://10.110.11.16:59672> [{
>>                  "jsonrpc":      "2.0",
>>                  "error":        {
>>                          "code": -32600,
>>                          "message":      "Invalid Request"
>>                  },
>>                  "id":   null
>>         }]
>>         2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
>>         10.110.11.16:59672 <http://10.110.11.16:59672> [{
>>                  "jsonrpc":      "2.0",
>>                  "error":        {
>>                          "code": -32600,
>>                          "message":      "Invalid Request"
>>                  },
>>                  "id":   null
>>         }]
>>         2016-04-20 18:52:24.135161 [ALERT] mod_verto.c:604 WRITE
>>         10.110.11.16:59672 <http://10.110.11.16:59672> [{
>>                  "jsonrpc":      "2.0",
>>                  "error":        {
>>                          "code": -32600,
>>                          "message":      "Invalid Request"
>>                  },
>>                  "id":   null
>>         }]
>>         2016-04-20 18:52:24.155150 [ALERT] mod_verto.c:604 WRITE
>>         10.110.11.16:59672 <http://10.110.11.16:59672> [{
>>                  "jsonrpc":      "2.0",
>>                  "error":        {
>>                          "code": -32600,
>>                          "message":      "Invalid Request"
>>                  },
>>                  "id":   null
>>         }]
>>         2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
>>         10.110.11.16:59672 <http://10.110.11.16:59672> [{
>>                  "jsonrpc":      "2.0",
>>                  "error":        {
>>                          "code": -32600,
>>                          "message":      "Invalid Request"
>>                  },
>>                  "id":   null
>>         }]
>>         2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
>>         10.110.11.16:59672 <http://10.110.11.16:59672> [{
>>                  "jsonrpc":      "2.0",
>>                  "error":        {
>>                          "code": -32600,
>>                          "message":      "Invalid Request"
>>                  },
>>                  "id":   null
>>         }]
>>         2016-04-20 18:52:24.455157 [ALERT] mod_verto.c:604 WRITE
>>         10.110.11.16:59672 <http://10.110.11.16:59672> [{
>>                  "jsonrpc":      "2.0",
>>                  "error":        {
>>                          "code": -32600,
>>                          "message":      "Invalid Request"
>>                  },
>>                  "id":   null
>>         }]
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://confluence.freeswitch.org
>>         http://www.cluecon.com
>>
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org
>>
>>
>>
>>
>>     -- 
>>     ?talo Rossi
>>     italo at freeswitch.org <mailto:italo at freeswitch.org>
>>
>>
>>     _________________________________________________________________________
>>     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
>
>
>
>
> -- 
> ?talo 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

-- 
Arsen Manukyan
ConnectTo Communications Inc.
555 Riverdale Dr., Suite A
Glendale, CA 91204
arsenman at connectto.com
http://www.ConnectTo.com
Tel. 818.546.4636
FAX 818.546.4617
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From amani.mansour2 at gmail.com  Fri Apr 22 10:19:39 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Fri, 22 Apr 2016 06:19:39 +0000
Subject: [Freeswitch-users] How to send Re_invite
In-Reply-To: <047e01d19c1d$777906e0$666b14a0$@freeswitch.org>
References: <CAFETSWjbsGh621pDmSpUM=CFUEop5ka_zpHOzn7r515gogReqw@mail.gmail.com>
	<047e01d19c1d$777906e0$666b14a0$@freeswitch.org>
Message-ID: <CAFETSWhR-SpeO_s8Qnzh9UQRS8F7yrGX2LYc4kgv3cfxjjM85w@mail.gmail.com>

Good morning sir,

Iam trying to devoloppe a web application which automatise some test
scenarios ,one of those tests is sending 491  message when the freeswitch
send re invite , i don't know why my supervisor need this scenario but i
must do it , first i didn't understand why the freeswitch send a re invite
and how. So my tasck for example i choose two extensions A and B  when A
call B and we open the wireshark i must see re invite and the response
message 491

Do you understand me sir ???

Best regards
Amani

Le jeu. 21 avr. 2016 ? 23:31, Ken Rice <krice at freeswitch.org> a ?crit :

> What specifically are you trying to accomplish?
>
>
>
> While you can configure freeswitch to respond to invites in a specific
> way, this is intended to use to terminate a call not pause it for
> continuation later.
>
>
>
> Most of the SIP handing is intentionally hidden from the user with
> FreeSWITCH.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *amani
> mansour
> *Sent:* Thursday, April 21, 2016 4:28 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> *Subject:* [Freeswitch-users] How to send Re_invite
>
>
>
> Hi all,
>
>
>
>
>
> I need to configure FS to send 491 request pending message after receiving
> Re invite from the caller ,
>
>
>
> The problem  that i don't know how to send the re invite request ,which
> xml file i whould modifie it ? please help me ,i have done many research
> but i didn't find anything .
>
>
>
> Thank you ,
>
>
>
> Best Regards
>
> Amani
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From s.safarov at gmail.com  Fri Apr 22 11:48:50 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 22 Apr 2016 07:48:50 +0000
Subject: [Freeswitch-users] How to send Re_invite
In-Reply-To: <CAFETSWhR-SpeO_s8Qnzh9UQRS8F7yrGX2LYc4kgv3cfxjjM85w@mail.gmail.com>
References: <CAFETSWjbsGh621pDmSpUM=CFUEop5ka_zpHOzn7r515gogReqw@mail.gmail.com>
	<047e01d19c1d$777906e0$666b14a0$@freeswitch.org>
	<CAFETSWhR-SpeO_s8Qnzh9UQRS8F7yrGX2LYc4kgv3cfxjjM85w@mail.gmail.com>
Message-ID: <CAHtxdDdNG4iPVSjb7Unu4Og8XwYxa_5cdyXF3uUKHrCFsCvNhA@mail.gmail.com>

May be do it using test tool like sipp?

??, 22 ???. 2016 ?. ? 9:21, amani mansour <amani.mansour2 at gmail.com>:

> Good morning sir,
>
> Iam trying to devoloppe a web application which automatise some test
> scenarios ,one of those tests is sending 491  message when the freeswitch
> send re invite , i don't know why my supervisor need this scenario but i
> must do it , first i didn't understand why the freeswitch send a re invite
> and how. So my tasck for example i choose two extensions A and B  when A
> call B and we open the wireshark i must see re invite and the response
> message 491
>
> Do you understand me sir ???
>
> Best regards
> Amani
>
> Le jeu. 21 avr. 2016 ? 23:31, Ken Rice <krice at freeswitch.org> a ?crit :
>
>> What specifically are you trying to accomplish?
>>
>>
>>
>> While you can configure freeswitch to respond to invites in a specific
>> way, this is intended to use to terminate a call not pause it for
>> continuation later.
>>
>>
>>
>> Most of the SIP handing is intentionally hidden from the user with
>> FreeSWITCH.
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *amani
>> mansour
>> *Sent:* Thursday, April 21, 2016 4:28 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>
>>
>> *Subject:* [Freeswitch-users] How to send Re_invite
>>
>>
>>
>> Hi all,
>>
>>
>>
>>
>>
>> I need to configure FS to send 491 request pending message after
>> receiving Re invite from the caller ,
>>
>>
>>
>> The problem  that i don't know how to send the re invite request ,which
>> xml file i whould modifie it ? please help me ,i have done many research
>> but i didn't find anything .
>>
>>
>>
>> Thank you ,
>>
>>
>>
>> Best Regards
>>
>> Amani
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From amani.mansour2 at gmail.com  Fri Apr 22 12:10:57 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Fri, 22 Apr 2016 08:10:57 +0000
Subject: [Freeswitch-users] How to send Re_invite
In-Reply-To: <CAHtxdDdNG4iPVSjb7Unu4Og8XwYxa_5cdyXF3uUKHrCFsCvNhA@mail.gmail.com>
References: <CAFETSWjbsGh621pDmSpUM=CFUEop5ka_zpHOzn7r515gogReqw@mail.gmail.com>
	<047e01d19c1d$777906e0$666b14a0$@freeswitch.org>
	<CAFETSWhR-SpeO_s8Qnzh9UQRS8F7yrGX2LYc4kgv3cfxjjM85w@mail.gmail.com>
	<CAHtxdDdNG4iPVSjb7Unu4Og8XwYxa_5cdyXF3uUKHrCFsCvNhA@mail.gmail.com>
Message-ID: <CAFETSWhcrjfOUo-cZOOSWGsHBi-3GrCG=by65Q7CGe+SL5hQ4g@mail.gmail.com>

Thank you sir ,

But why it is impossible such scenario with FreeSWICTH may be the the last
solution wil be the SIPPand then i will modify the ports with iptables like
this is send by FreeSWITCH .

Best Regards
Amani

Le ven. 22 avr. 2016 ? 08:50, Sergey Safarov <s.safarov at gmail.com> a ?crit :

> May be do it using test tool like sipp?
>
> ??, 22 ???. 2016 ?. ? 9:21, amani mansour <amani.mansour2 at gmail.com>:
>
>> Good morning sir,
>>
>> Iam trying to devoloppe a web application which automatise some test
>> scenarios ,one of those tests is sending 491  message when the freeswitch
>> send re invite , i don't know why my supervisor need this scenario but i
>> must do it , first i didn't understand why the freeswitch send a re invite
>> and how. So my tasck for example i choose two extensions A and B  when A
>> call B and we open the wireshark i must see re invite and the response
>> message 491
>>
>> Do you understand me sir ???
>>
>> Best regards
>> Amani
>>
>> Le jeu. 21 avr. 2016 ? 23:31, Ken Rice <krice at freeswitch.org> a ?crit :
>>
>>> What specifically are you trying to accomplish?
>>>
>>>
>>>
>>> While you can configure freeswitch to respond to invites in a specific
>>> way, this is intended to use to terminate a call not pause it for
>>> continuation later.
>>>
>>>
>>>
>>> Most of the SIP handing is intentionally hidden from the user with
>>> FreeSWITCH.
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *amani
>>> mansour
>>> *Sent:* Thursday, April 21, 2016 4:28 PM
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>
>>>
>>> *Subject:* [Freeswitch-users] How to send Re_invite
>>>
>>>
>>>
>>> Hi all,
>>>
>>>
>>>
>>>
>>>
>>> I need to configure FS to send 491 request pending message after
>>> receiving Re invite from the caller ,
>>>
>>>
>>>
>>> The problem  that i don't know how to send the re invite request ,which
>>> xml file i whould modifie it ? please help me ,i have done many research
>>> but i didn't find anything .
>>>
>>>
>>>
>>> Thank you ,
>>>
>>>
>>>
>>> Best Regards
>>>
>>> Amani
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From s.safarov at gmail.com  Fri Apr 22 12:25:53 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 22 Apr 2016 08:25:53 +0000
Subject: [Freeswitch-users] How to send Re_invite
In-Reply-To: <CAFETSWhcrjfOUo-cZOOSWGsHBi-3GrCG=by65Q7CGe+SL5hQ4g@mail.gmail.com>
References: <CAFETSWjbsGh621pDmSpUM=CFUEop5ka_zpHOzn7r515gogReqw@mail.gmail.com>
	<047e01d19c1d$777906e0$666b14a0$@freeswitch.org>
	<CAFETSWhR-SpeO_s8Qnzh9UQRS8F7yrGX2LYc4kgv3cfxjjM85w@mail.gmail.com>
	<CAHtxdDdNG4iPVSjb7Unu4Og8XwYxa_5cdyXF3uUKHrCFsCvNhA@mail.gmail.com>
	<CAFETSWhcrjfOUo-cZOOSWGsHBi-3GrCG=by65Q7CGe+SL5hQ4g@mail.gmail.com>
Message-ID: <CAHtxdDeFz=fOKyQYR9TrF8vmZa2t3LpZoBFXxaGWv4hV9=OrJA@mail.gmail.com>

Because the FS is focused on solving real needs.
And badly doing, then he should not do, but another program developer want
to do.
You can modify FS and create pull request

??, 22 ???. 2016 ?. ? 11:11, amani mansour <amani.mansour2 at gmail.com>:

> Thank you sir ,
>
> But why it is impossible such scenario with FreeSWICTH may be the the last
> solution wil be the SIPPand then i will modify the ports with iptables like
> this is send by FreeSWITCH .
>
> Best Regards
> Amani
>
> Le ven. 22 avr. 2016 ? 08:50, Sergey Safarov <s.safarov at gmail.com> a
> ?crit :
>
>> May be do it using test tool like sipp?
>>
>> ??, 22 ???. 2016 ?. ? 9:21, amani mansour <amani.mansour2 at gmail.com>:
>>
>>> Good morning sir,
>>>
>>> Iam trying to devoloppe a web application which automatise some test
>>> scenarios ,one of those tests is sending 491  message when the freeswitch
>>> send re invite , i don't know why my supervisor need this scenario but i
>>> must do it , first i didn't understand why the freeswitch send a re invite
>>> and how. So my tasck for example i choose two extensions A and B  when A
>>> call B and we open the wireshark i must see re invite and the response
>>> message 491
>>>
>>> Do you understand me sir ???
>>>
>>> Best regards
>>> Amani
>>>
>>> Le jeu. 21 avr. 2016 ? 23:31, Ken Rice <krice at freeswitch.org> a ?crit :
>>>
>>>> What specifically are you trying to accomplish?
>>>>
>>>>
>>>>
>>>> While you can configure freeswitch to respond to invites in a specific
>>>> way, this is intended to use to terminate a call not pause it for
>>>> continuation later.
>>>>
>>>>
>>>>
>>>> Most of the SIP handing is intentionally hidden from the user with
>>>> FreeSWITCH.
>>>>
>>>>
>>>>
>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *amani
>>>> mansour
>>>> *Sent:* Thursday, April 21, 2016 4:28 PM
>>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>
>>>>
>>>> *Subject:* [Freeswitch-users] How to send Re_invite
>>>>
>>>>
>>>>
>>>> Hi all,
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> I need to configure FS to send 491 request pending message after
>>>> receiving Re invite from the caller ,
>>>>
>>>>
>>>>
>>>> The problem  that i don't know how to send the re invite request ,which
>>>> xml file i whould modifie it ? please help me ,i have done many research
>>>> but i didn't find anything .
>>>>
>>>>
>>>>
>>>> Thank you ,
>>>>
>>>>
>>>>
>>>> Best Regards
>>>>
>>>> Amani
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From andrew at cassidywebservices.co.uk  Fri Apr 22 14:17:27 2016
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 22 Apr 2016 11:17:27 +0100
Subject: [Freeswitch-users] London Marathon 2016
Message-ID: <CAMbW9oLGO00KRfVNP0O62K8vKYFKf1B+TpsCb2im6HLsgG-GbA@mail.gmail.com>

Hi Guys,

Not like me to spam this sort of stuff to the list, but it's worth a go!

Sunday is the London Marathon. My other half is running this year for the
charity WhizzKidz. They're a UK based charity who supply equipment to
disabled children, more information can be found here:
http://www.whizz-kidz.org.uk/

Any contributions would be much appreciated. Here's the link to her
justgiving page:

https://www.justgiving.com/Rachel-Wilson39/

Thanks for your time!

Kind regards,

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

03303 880 960 andrew at cassidyweb.co.uk <andrew at cassidywebservices.co.uk>
www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>
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From miha at softnet.si  Fri Apr 22 16:02:16 2016
From: miha at softnet.si (Miha)
Date: Fri, 22 Apr 2016 14:02:16 +0200
Subject: [Freeswitch-users] Nat issue
In-Reply-To: <57186DF3.4000800@softnet.si>
References: <57186DF3.4000800@softnet.si>
Message-ID: <571A12C8.70608@softnet.si>

Hi,

could some help me with this.

tnx
miha

On 21/04/2016 08:06, Miha wrote:
> Hi
>
> our FS is on public ip but the voip server to which trunk is made 
> (SFB) is behind NAT.
>
> So the issue is that in SDP is private ip and FS sends RTP to this 
> private ip, it does not change to src ip from which sip request was 
> received.
>
> Remote SDP:
> 2016-04-21 08:04:00.954752 [DEBUG] sofia.c:5614 Remote SDP:
> v=0
> o=- 46176 1 IN IP4 172.16.2.3
> s=session
> c=IN IP4 172.16.2.3
> b=CT:1000
> t=0 0
> m=audio 31200 RTP/AVP 8 101 13
> c=IN IP4 172.16.2.3
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=rtcp:31201
> a=label:Audio
>
>
> And then:
>
>  xxx.xxx.xxx.xxx (FS public ip) port 17094 -> 172.16.2.3 port 31200 
> codec: 8 ms: 30
>
>
> I tried with:
>     <param name="NDLB-force-rport" value="true"/>
>     <param name="aggressive-nat-detection" value="true"/>
>
>
> on internal profile and on trunk profile but the same.
>
>
> tnx for help!
> Miha
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From iskren.hadzhinedev at ikiji.com  Fri Apr 22 16:15:41 2016
From: iskren.hadzhinedev at ikiji.com (Iskren Hadzhinedev)
Date: Fri, 22 Apr 2016 15:15:41 +0300
Subject: [Freeswitch-users] Nat issue
In-Reply-To: <571A12C8.70608@softnet.si>
References: <57186DF3.4000800@softnet.si> <571A12C8.70608@softnet.si>
Message-ID: <1461327341.11613.44.camel@ikiji.com>

Hi,
Try adding <param name="apply-nat-acl" value="nat.auto"/> to your
external profile.

Regards,
Iskren

-----Original Message-----From: Miha <miha at softnet.si>
Reply-to: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Nat issue
Date: Fri, 22 Apr 2016 14:02:16 +0200

Hi,

could some help me with this.

tnx
miha

On 21/04/2016 08:06, Miha wrote:

> 
> Hi
> 
> our FS is on public ip but the voip server to which trunk is made
> (SFB) is behind NAT.
> 
> So the issue is that in SDP is private ip and FS sends RTP to this
> private ip, it does not change to src ip from which sip request was
> received.
> 
> Remote SDP:
> 2016-04-21 08:04:00.954752 [DEBUG] sofia.c:5614 Remote SDP:
> v=0
> o=- 46176 1 IN IP4 172.16.2.3
> s=session
> c=IN IP4 172.16.2.3
> b=CT:1000
> t=0 0
> m=audio 31200 RTP/AVP 8 101 13
> c=IN IP4 172.16.2.3
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=rtcp:31201
> a=label:Audio
> 
> 
> And then:
> 
>  xxx.xxx.xxx.xxx (FS public ip) port 17094 -> 172.16.2.3 port 31200
> codec: 8 ms: 30
> 
> 
> I tried with:
>     <param name="NDLB-force-rport" value="true"/>
>     <param name="aggressive-nat-detection" value="true"/>
> 
> 
> on internal profile and on trunk profile but the same.
> 
> 
> tnx for help!
> Miha
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From m.hubert at hexanet.fr  Fri Apr 22 16:56:17 2016
From: m.hubert at hexanet.fr (Mickael Hubert)
Date: Fri, 22 Apr 2016 14:56:17 +0200
Subject: [Freeswitch-users] Codec negotiation issue
In-Reply-To: <CAFSwG-vUMZ=FEET4Vwr20UVWGH4T-d4pGjcnwWkH8rZ+cWbpKQ@mail.gmail.com>
References: <CAFSwG-vUMZ=FEET4Vwr20UVWGH4T-d4pGjcnwWkH8rZ+cWbpKQ@mail.gmail.com>
Message-ID: <CAFSwG-trhBpJU98_eoU-_xxzNpGZx__hh1iSySZbm6Qv16+FpQ@mail.gmail.com>

Hi list,
just for info.

It was a bug on this version.

bye

2016-04-05 16:04 GMT+02:00 Mickael Hubert <m.hubert at hexanet.fr>:

> Hi list,
> I have an issue with the negociation codec in my Freeswitch.
>
> I followed this doc: https://wiki.freeswitch.org/wiki/Codec_Negotiation
>
> The call flow:
>
> 1) INVITE: UAC -- (G729, PCMA) --> FS -- (G729, PCMA) --> SVI Asterisk
> 2) 200OK UAC <-- (G729) -- FS <-- (PCMA) -- SVI Asterisk (HOMER)
>
> I have inherit_codec=true in my dialplan and inbound-late-negotiation true
> in sip-profile.
> But freeswitch do not force codec learned from leg B to leg A.
>
> *LEG B (200OK):*
>
>
>
> *2016-04-01 11:28:16.453784 [DEBUG] switch_core_media.c:3194 Audio Codec
> Compare [PCMA:8:8000:150:64000]/[G729:18:8000:20:8000]2016-04-01
> 11:28:16.453784 [DEBUG] switch_core_media.c:3194 Audio Codec Compare
> [PCMA:8:8000:150:64000]/[PCMA:8:8000:20:64000]2016-04-01 11:28:16.453784
> [DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMA:8:8000:20:64000]
> ++++ is saved as a match2016-04-01 11:28:16.453784 [DEBUG]
> switch_core_codec.c:111 sofia/internal/06********@HOMER Original read codec
> set to PCMA:8*
>
> LEG B is in G711A, OK, next:
>
> 2016-04-01 11:28:16.453784 [NOTICE] sofia.c:6727 Channel
> [sofia/internal/06*******@HOMER] has been answered
> 2016-04-01 11:28:16.453784 [DEBUG] switch_channel.c:3686
> (sofia/internal/06*********@HOMER) Callstate Change DOWN -> ACTIVE
> 2016-04-01 11:28:16.473813 [DEBUG] switch_ivr_originate.c:412 *Setting
> codec string on sofia/external/0326793005 at 1.1.1.1 <0326793005 at 1.1.1.1> to
> PCMA at 8000h@20i*
>
> We can see, FS set the correct codec to LEG A; good ! , next...
>
>
>
>
>
>
> *2016-04-01 11:28:16.473813 [DEBUG] switch_core_media.c:3194 Audio Codec
> Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000]2016-04-01
> 11:28:16.473813 [DEBUG] switch_core_media.c:3248 Audio Codec Compare
> [G729:18:8000:20:8000] ++++ is saved as a match2016-04-01 11:28:16.473813
> [DEBUG] switch_core_media.c:3194 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]2016-04-01 11:28:16.473813
> [DEBUG] switch_core_media.c:3248 Audio Codec Compare [PCMA:8:8000:20:64000]
> ++++ is saved as a match2016-04-01 11:28:16.473813 [DEBUG]
> switch_core_media.c:2139 Set Codec sofia/external/0326793005 at 1.1.1.1
> <0326793005 at 1.1.1.1> G729/8000 20 ms 160 samples 8000 bits*
> Why FS compare and re set a new codec (G729) ?
>
> Next:
>
>
> *2016-04-01 11:28:16.613776 [ERR] mod_g729.c:145 This codec is only usable
> in passthrough mode!2016-04-01 11:28:16.613776 [ERR] switch_core_io.c:1245
> Codec G.729 decoder error!*
>
> normal ....
>
> *For informations:*
> - FreeSWITCH Version 1.5.8b+git~20140214T000311Z~fe2a4d6d47~64bit (git
> fe2a4d6 2014-02-14 00:03:11Z 64bit)
>
> Is it a bug ? an configuration error ?
>
> Thanks in advance
>
>
> PS:
>
> *My dialplan:*
>
>
>
>
> *<condition field="${destination_number}" expression="^0([1-79][0-9]{8})$"
> break="on-true"><action application="export"
> data="codec_string=${ep_codec_string}"/><action application="set"
> data="inherit_codec=true"/><action application="bridge"
> data="sofia/internal/${destination_number}@HOMER"/></condition>*
>
>
>
> --
> Cordialement
>
> HUBERT Micka?l
> Ing?nieur VOIP - Hexanet
>



-- 
Cordialement

HUBERT Micka?l
Ing?nieur VOIP - Hexanet

-- 

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From royj at yandex.ru  Fri Apr 22 17:07:18 2016
From: royj at yandex.ru (royj at yandex.ru)
Date: Fri, 22 Apr 2016 16:07:18 +0300
Subject: [Freeswitch-users] Nat issue
In-Reply-To: <1461327341.11613.44.camel@ikiji.com>
References: <57186DF3.4000800@softnet.si> <571A12C8.70608@softnet.si>
	<1461327341.11613.44.camel@ikiji.com>
Message-ID: <5876731461330438@web5m.yandex.ru>

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From shaun.stokes at itec-support.co.uk  Fri Apr 22 17:55:57 2016
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Fri, 22 Apr 2016 13:55:57 +0000
Subject: [Freeswitch-users] Call Recording On Demand for Call Center Queues
	(mod_callcenter)
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB555D@mbx-01.sysconfig.co.uk>

Hi,

We?re trying to setup call recording on demand for call center queues (mod_callcenter) but are experiencing tremendous trouble. Is it possible for the agent to perform pause\resume call recording on mod_callcenter calls which are recorded by default upon answer?

Here?s the important elements of our dialplan:
<condition field="${rec_file}" expression="^(.)+">
                <action application="set" inline="true" data="RECORD_APPEND=true"/>
                <anti-action application="set" inline="true" data="RECORD_APPEND=true"/>
                <anti-action application="export" inline="true" data="rec_file=$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}"/>
</condition>
<condition field="destination_number" expression="^SETUP_RECORDING$">
                <action application="log" data="INFO Configuring bind_digit_action to do recording on this session..."/>
                <action application="bind_digit_action" data="features,*1,exec:execute_extension,START_RECORDING XML ${domain_name}"/>
                <action application="bind_digit_action" data="features,*0,exec:execute_extension,STOP_RECORDING XML ${domain_name}"/>
                <action application="digit_action_set_realm" data=" features"/>
</condition>
<condition field="destination_number" expression="^START_RECORDING$">
                <action application="log" data="INFO Starting recording..."/>
                <action application="record_session" data="${rec_file}"/>
</condition>
<condition field="destination_number" expression="^STOP_RECORDING$">
                <action application="log" data="INFO Stopping recording..."/>
                <action application="stop_record_session" data="${rec_file}"/>
</condition>
<condition field="destination_number" expression="^837$">
<action application="export" data="nolocal:execute_on_answer=record_session ${rec_file}"/>
<action application="set" data="cc_export_vars=rec_file"/>
<action application="set" data="bridge_pre_execute_bleg_app=execute_extension"/>
<action application="set" data="bridge_pre_execute_bleg_data=SETUP_RECORDING XML ${domain_name}"/>
<action application="callcenter" data="Test-bench at tester"/>
</condition>

Here?s the log of the call as it comes in:
SET [RECORD_APPEND]=[true]
Configuring bind_digit_action to do recording on this session...
Digit parser DPTOOLS: binding *1/features/0 callback: 0x7f37720eed10 data: 0x7f367c7dd918
Digit parser DPTOOLS: binding *0/features/0 callback: 0x7f37720eed10 data: 0x7f367c7ddab8
Digit parser DPTOOLS: Setting realm to 'features'
EXPORT (export_vars) (REMOTE ONLY) [execute_on_answer]=[record_session /usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3]

The call is then answered by the agent, we see the recording file has been created, the agent presses *0 but the call continues recording, log as follows:
SET [RECORD_APPEND]=[true]
RTP RECV DTMF *:1536
RTP RECV DTMF 0:1536
Stopping recording...
stop_record_session(/usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3)


The configuration looks as though it should work, any ideas? Is there another method we should be using?

Many Thanks,
Shaun

______________________________________________________________________
This message has been checked for all known viruses by  MessageLabs Virus Scanning Service.
______________________________________________________________________
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From italo at freeswitch.org  Fri Apr 22 18:41:03 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Fri, 22 Apr 2016 11:41:03 -0300
Subject: [Freeswitch-users] Verto
In-Reply-To: <CAC7ttFGGfpAdcbRfpec3-FRbunRBVGOpvJOgcPQ=Siycsem-gQ@mail.gmail.com>
References: <CAC7ttFE6C_CDkV79vOcgV9YKw+sJALT7bcGf_5RJ5KO8EcXb4A@mail.gmail.com>
	<CABMX_w9vZCmqdkgAZ00FB1WZqT=hRK5hvtHF0tmdsGPWmAjZFQ@mail.gmail.com>
	<CAC7ttFGGfpAdcbRfpec3-FRbunRBVGOpvJOgcPQ=Siycsem-gQ@mail.gmail.com>
Message-ID: <CABMX_w-cuK34LfT10U2iG7uuQCN15qzMFsnMBWuvVkdLbOH3_g@mail.gmail.com>

Yeah, get from our latest master.

On Thu, Apr 21, 2016 at 7:31 PM, Gregor Nanger <gregor at infomedia.si> wrote:

> Thank you for answer.
>
> I think that something is conflicting with other javascript library. I am
> integrating verto in other solution (angularjs and bunch of other
> javascripts). Demo is also working ok in my environment.
>
> How can I be sure that I am using latest lib? Should I just download
> latest branch?
>
>
>
>
>
>
>
> 2016-04-21 19:59 GMT+02:00 ?talo Rossi <italo at freeswitch.org>:
>
>> Hey Gregor,
>>
>> Can you double check if you're using the most recent version of verto js
>> lib?
>>
>> On Thu, Apr 21, 2016 at 4:49 AM, Gregor Nanger <gregor at infomedia.si>
>> wrote:
>>
>>> ?I am building my web phone with verto from scratch and using this
>>> tutorial:
>>>
>>> http://evoluxbr.github.io/verto-docs/
>>>
>>> BAsically everything is ok, I can register and make calls, but I cannot
>>> hear audio. Mic is working, but not speakers. I see error:
>>>
>>> Cannot read property 'sinkId' of undefined. This is error line:
>>>
>>> $.verto.dialog.prototype.setAudioPlaybackDevice = function(sinkId,
>>> callback, arg) {
>>> var dialog = this;
>>> var element = dialog.audioStream;
>>>
>>> * if (typeof element.sinkId !== 'undefined') {*
>>>
>>> I get dialog object without audioStream property and hence element is
>>> null.
>>>
>>> I am using self signed certificate. Could this be cause?
>>>
>>> Best regards, Gregor
>>> ?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> ?talo Rossi
>> italo at freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> 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
>



-- 
?talo Rossi
italo at freeswitch.org
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From gregor at infomedia.si  Fri Apr 22 18:50:02 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Fri, 22 Apr 2016 16:50:02 +0200
Subject: [Freeswitch-users] Verto
In-Reply-To: <CAC7ttFGGfpAdcbRfpec3-FRbunRBVGOpvJOgcPQ=Siycsem-gQ@mail.gmail.com>
References: <CAC7ttFE6C_CDkV79vOcgV9YKw+sJALT7bcGf_5RJ5KO8EcXb4A@mail.gmail.com>
	<CABMX_w9vZCmqdkgAZ00FB1WZqT=hRK5hvtHF0tmdsGPWmAjZFQ@mail.gmail.com>
	<CAC7ttFGGfpAdcbRfpec3-FRbunRBVGOpvJOgcPQ=Siycsem-gQ@mail.gmail.com>
Message-ID: <CAC7ttFEf5w6D-qGTtwaARvr9Dhg-m=QN+1LtvRQB4kzRT_FXWA@mail.gmail.com>

Justo for info. I managed to solve problem.

This error:
Cannot read property 'sinkId' of undefined.

is caused if developer make mistake :-)))

I set useSpeak: true, insted objectId of device :-((

But I found that I didn't know and maybe it will help to someone.

Even if you want to use verto only for audio calls, there still needs to be
set video object (can be hidden) in html. Otherwise incoming audio is not
working.

I set: <video width=800 id="webcam" autoplay="autoplay"
hidden="true"></video>
and add tag:'webcam' in verto initialization.

Now everything works as expected, except 2 sec delay in audio stream from
browser to endpoint.

2016-04-22 0:31 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:

> Thank you for answer.
>
> I think that something is conflicting with other javascript library. I am
> integrating verto in other solution (angularjs and bunch of other
> javascripts). Demo is also working ok in my environment.
>
> How can I be sure that I am using latest lib? Should I just download
> latest branch?
>
>
>
>
>
>
>
> 2016-04-21 19:59 GMT+02:00 ?talo Rossi <italo at freeswitch.org>:
>
>> Hey Gregor,
>>
>> Can you double check if you're using the most recent version of verto js
>> lib?
>>
>> On Thu, Apr 21, 2016 at 4:49 AM, Gregor Nanger <gregor at infomedia.si>
>> wrote:
>>
>>> ?I am building my web phone with verto from scratch and using this
>>> tutorial:
>>>
>>> http://evoluxbr.github.io/verto-docs/
>>>
>>> BAsically everything is ok, I can register and make calls, but I cannot
>>> hear audio. Mic is working, but not speakers. I see error:
>>>
>>> Cannot read property 'sinkId' of undefined. This is error line:
>>>
>>> $.verto.dialog.prototype.setAudioPlaybackDevice = function(sinkId,
>>> callback, arg) {
>>> var dialog = this;
>>> var element = dialog.audioStream;
>>>
>>> * if (typeof element.sinkId !== 'undefined') {*
>>>
>>> I get dialog object without audioStream property and hence element is
>>> null.
>>>
>>> I am using self signed certificate. Could this be cause?
>>>
>>> Best regards, Gregor
>>> ?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> ?talo Rossi
>> italo at freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> 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
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
? www.infomedia.si
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From gregor at infomedia.si  Fri Apr 22 18:51:53 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Fri, 22 Apr 2016 16:51:53 +0200
Subject: [Freeswitch-users] Verto latency
In-Reply-To: <CAC7ttFFtZSwejG2_sRzg04ySB+_VCJ5ORVaiZryFJtnA+Pav3w@mail.gmail.com>
References: <CAC7ttFFtZSwejG2_sRzg04ySB+_VCJ5ORVaiZryFJtnA+Pav3w@mail.gmail.com>
Message-ID: <CAC7ttFEmXUjGgJaFiAooz2xervh3Qhixi-YHoPMfuuXgcmS4PQ@mail.gmail.com>

This is last thing that bothers me regarding verto, otherwise it works as
expected. So if you have any hint, you are welcome. Could this be because
of OPUS codec?

2016-04-22 0:34 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:

> If I make call from SIP client everything is ok. But if I make call from
> verto demo client, there is 2 second delay of audio in direction from
> browser (latest chrome) to callee. Other direction is ok.
>
> Is it posible to tweak lower this latency?
>



-- 
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 shaun.stokes at itec-support.co.uk  Fri Apr 22 19:25:34 2016
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Fri, 22 Apr 2016 15:25:34 +0000
Subject: [Freeswitch-users] Call Recording On Demand for Call Center
 Queues	(mod_callcenter)
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB555D@mbx-01.sysconfig.co.uk>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB555D@mbx-01.sysconfig.co.uk>
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB5705@mbx-01.sysconfig.co.uk>

We have been troubleshooting this further, it seems that this line has no effect when using mod_callcenter:
<action application="export" data="nolocal:execute_on_answer=record_session ${rec_file}"/>

This parameter is set in callcenter.conf if we remove this the call doesn?t record:
<param name="record-template" value="=$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}"/>

The file path\name from the above matches that of the call we?re trying to pause, but it seems we can?t pause the recording if this has been started by mod_callcenter, there are no errors. We can only start and pause call recording when this isn?t started using mod_callcenter by default.

Is there any way around this, maybe a different command for mod_callcenter that can be executed on the agent leg using dtmf to stop the recording, or another way we can start recording after being answered by the agent?

Thanks,
Shaun


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes
Sent: 22 April 2016 14:56
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Call Recording On Demand for Call Center Queues (mod_callcenter)

Hi,

We?re trying to setup call recording on demand for call center queues (mod_callcenter) but are experiencing tremendous trouble. Is it possible for the agent to perform pause\resume call recording on mod_callcenter calls which are recorded by default upon answer?

Here?s the important elements of our dialplan:
<condition field="${rec_file}" expression="^(.)+">
                <action application="set" inline="true" data="RECORD_APPEND=true"/>
                <anti-action application="set" inline="true" data="RECORD_APPEND=true"/>
                <anti-action application="export" inline="true" data="rec_file=$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}"/>
</condition>
<condition field="destination_number" expression="^SETUP_RECORDING$">
                <action application="log" data="INFO Configuring bind_digit_action to do recording on this session..."/>
                <action application="bind_digit_action" data="features,*1,exec:execute_extension,START_RECORDING XML ${domain_name}"/>
                <action application="bind_digit_action" data="features,*0,exec:execute_extension,STOP_RECORDING XML ${domain_name}"/>
                <action application="digit_action_set_realm" data=" features"/>
</condition>
<condition field="destination_number" expression="^START_RECORDING$">
                <action application="log" data="INFO Starting recording..."/>
                <action application="record_session" data="${rec_file}"/>
</condition>
<condition field="destination_number" expression="^STOP_RECORDING$">
                <action application="log" data="INFO Stopping recording..."/>
                <action application="stop_record_session" data="${rec_file}"/>
</condition>
<condition field="destination_number" expression="^837$">
<action application="export" data="nolocal:execute_on_answer=record_session ${rec_file}"/>
<action application="set" data="cc_export_vars=rec_file"/>
<action application="set" data="bridge_pre_execute_bleg_app=execute_extension"/>
<action application="set" data="bridge_pre_execute_bleg_data=SETUP_RECORDING XML ${domain_name}"/>
<action application="callcenter" data="Test-bench at tester"/>
</condition>

Here?s the log of the call as it comes in:
SET [RECORD_APPEND]=[true]
Configuring bind_digit_action to do recording on this session...
Digit parser DPTOOLS: binding *1/features/0 callback: 0x7f37720eed10 data: 0x7f367c7dd918
Digit parser DPTOOLS: binding *0/features/0 callback: 0x7f37720eed10 data: 0x7f367c7ddab8
Digit parser DPTOOLS: Setting realm to 'features'
EXPORT (export_vars) (REMOTE ONLY) [execute_on_answer]=[record_session /usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3]

The call is then answered by the agent, we see the recording file has been created, the agent presses *0 but the call continues recording, log as follows:
SET [RECORD_APPEND]=[true]
RTP RECV DTMF *:1536
RTP RECV DTMF 0:1536
Stopping recording...
stop_record_session(/usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3)


The configuration looks as though it should work, any ideas? Is there another method we should be using?

Many Thanks,
Shaun

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

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From regis.freeswitch.org at tornad.net  Fri Apr 22 20:22:35 2016
From: regis.freeswitch.org at tornad.net (Regis M)
Date: Fri, 22 Apr 2016 18:22:35 +0200
Subject: [Freeswitch-users] Verto
In-Reply-To: <CAC7ttFEf5w6D-qGTtwaARvr9Dhg-m=QN+1LtvRQB4kzRT_FXWA@mail.gmail.com>
References: <CAC7ttFE6C_CDkV79vOcgV9YKw+sJALT7bcGf_5RJ5KO8EcXb4A@mail.gmail.com>
	<CABMX_w9vZCmqdkgAZ00FB1WZqT=hRK5hvtHF0tmdsGPWmAjZFQ@mail.gmail.com>
	<CAC7ttFGGfpAdcbRfpec3-FRbunRBVGOpvJOgcPQ=Siycsem-gQ@mail.gmail.com>
	<CAC7ttFEf5w6D-qGTtwaARvr9Dhg-m=QN+1LtvRQB4kzRT_FXWA@mail.gmail.com>
Message-ID: <CAHWtr1QXkSUHjX7-5ejV0cUd2xPRN-4dNDJfagPHHCPQAmJ7zg@mail.gmail.com>

Thx for sharing !!

2016-04-22 16:50 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:

> Justo for info. I managed to solve problem.
>
> This error:
> Cannot read property 'sinkId' of undefined.
>
> is caused if developer make mistake :-)))
>
> I set useSpeak: true, insted objectId of device :-((
>
> But I found that I didn't know and maybe it will help to someone.
>
> Even if you want to use verto only for audio calls, there still needs to
> be set video object (can be hidden) in html. Otherwise incoming audio is
> not working.
>
> I set: <video width=800 id="webcam" autoplay="autoplay"
> hidden="true"></video>
> and add tag:'webcam' in verto initialization.
>
> Now everything works as expected, except 2 sec delay in audio stream from
> browser to endpoint.
>
> 2016-04-22 0:31 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:
>
>> Thank you for answer.
>>
>> I think that something is conflicting with other javascript library. I am
>> integrating verto in other solution (angularjs and bunch of other
>> javascripts). Demo is also working ok in my environment.
>>
>> How can I be sure that I am using latest lib? Should I just download
>> latest branch?
>>
>>
>>
>>
>>
>>
>>
>> 2016-04-21 19:59 GMT+02:00 ?talo Rossi <italo at freeswitch.org>:
>>
>>> Hey Gregor,
>>>
>>> Can you double check if you're using the most recent version of verto js
>>> lib?
>>>
>>> On Thu, Apr 21, 2016 at 4:49 AM, Gregor Nanger <gregor at infomedia.si>
>>> wrote:
>>>
>>>> ?I am building my web phone with verto from scratch and using this
>>>> tutorial:
>>>>
>>>> http://evoluxbr.github.io/verto-docs/
>>>>
>>>> BAsically everything is ok, I can register and make calls, but I cannot
>>>> hear audio. Mic is working, but not speakers. I see error:
>>>>
>>>> Cannot read property 'sinkId' of undefined. This is error line:
>>>>
>>>> $.verto.dialog.prototype.setAudioPlaybackDevice = function(sinkId,
>>>> callback, arg) {
>>>> var dialog = this;
>>>> var element = dialog.audioStream;
>>>>
>>>> * if (typeof element.sinkId !== 'undefined') {*
>>>>
>>>> I get dialog object without audioStream property and hence element is
>>>> null.
>>>>
>>>> I am using self signed certificate. Could this be cause?
>>>>
>>>> Best regards, Gregor
>>>> ?
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> ?talo Rossi
>>> italo at freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> 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
>>
>
>
>
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
> ? www.infomedia.si
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From clive at lansink.co.nz  Fri Apr 22 23:44:50 2016
From: clive at lansink.co.nz (Clive Lansink)
Date: Sat, 23 Apr 2016 07:44:50 +1200
Subject: [Freeswitch-users] Freeswitch and Ping
Message-ID: <mailman.3.1461354344.8220.freeswitch-users@lists.freeswitch.org>

An embedded and charset-unspecified text was scrubbed...
Name: not available
Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160423/44c5066f/attachment.pl 

From italo at freeswitch.org  Fri Apr 22 23:47:26 2016
From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Fri, 22 Apr 2016 16:47:26 -0300
Subject: [Freeswitch-users] Call Recording On Demand for Call Center
 Queues (mod_callcenter)
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB5705@mbx-01.sysconfig.co.uk>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB555D@mbx-01.sysconfig.co.uk>
	<6FD2F8B5BB72834E9939AEDF9FB802A901E7BB5705@mbx-01.sysconfig.co.uk>
Message-ID: <CABMX_w_qtSXYwh_U6ZXD9k43gLOH5j5hYw=yey1b8ktKo72ufw@mail.gmail.com>

Use cc_export_vars
https://freeswitch.org/confluence/display/FREESWITCH/mod_callcenter#mod_callcenter-cc_export_vars

Or you can set variables in the agent's contact field:
{myvar=myvalue}user/number

On Fri, Apr 22, 2016 at 12:25 PM, Shaun Stokes <
shaun.stokes at itec-support.co.uk> wrote:

> We have been troubleshooting this further, it seems that this line has no
> effect when using mod_callcenter:
>
> <action application="export"
> data="nolocal:execute_on_answer=record_session ${rec_file}"/>
>
>
>
> This parameter is set in callcenter.conf if we remove this the call
> doesn?t record:
>
> <param name="record-template" value="
> =$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}
> "/>
>
>
>
> The file path\name from the above matches that of the call we?re trying to
> pause, but it seems we can?t pause the recording if this has been started
> by mod_callcenter, there are no errors. We can only start and pause call
> recording when this isn?t started using mod_callcenter by default.
>
>
>
> Is there any way around this, maybe a different command for mod_callcenter
> that can be executed on the agent leg using dtmf to stop the recording, or
> another way we can start recording after being answered by the agent?
>
>
>
> Thanks,
>
> Shaun
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Shaun Stokes
> *Sent:* 22 April 2016 14:56
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] Call Recording On Demand for Call Center
> Queues (mod_callcenter)
>
>
>
> Hi,
>
>
>
> We?re trying to setup call recording on demand for call center queues
> (mod_callcenter) but are experiencing tremendous trouble. Is it possible
> for the agent to perform pause\resume call recording on mod_callcenter
> calls which are recorded by default upon answer?
>
>
>
> Here?s the important elements of our dialplan:
>
> <condition field="${rec_file}" expression="^(.)+">
>
>                 <action application="set" inline="true"
> data="RECORD_APPEND=true"/>
>
>                 <anti-action application="set" inline="true"
> data="RECORD_APPEND=true"/>
>
>                 <anti-action application="export" inline="true"
> data="rec_file=$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}"/>
>
> </condition>
>
> <condition field="destination_number" expression="^SETUP_RECORDING$">
>
>                 <action application="log" data="INFO Configuring
> bind_digit_action to do recording on this session..."/>
>
>                 <action application="bind_digit_action"
> data="features,*1,exec:execute_extension,START_RECORDING XML
> ${domain_name}"/>
>
>                 <action application="bind_digit_action"
> data="features,*0,exec:execute_extension,STOP_RECORDING XML
> ${domain_name}"/>
>
>                 <action application="digit_action_set_realm" data="
> features"/>
>
> </condition>
>
> <condition field="destination_number" expression="^START_RECORDING$">
>
>                 <action application="log" data="INFO Starting
> recording..."/>
>
>                 <action application="record_session" data="${rec_file}"/>
>
> </condition>
>
> <condition field="destination_number" expression="^STOP_RECORDING$">
>
>                 <action application="log" data="INFO Stopping
> recording..."/>
>
>                 <action application="stop_record_session"
> data="${rec_file}"/>
>
> </condition>
>
> <condition field="destination_number" expression="^837$">
>
> <action application="export"
> data="nolocal:execute_on_answer=record_session ${rec_file}"/>
>
> <action application="set" data="cc_export_vars=rec_file"/>
>
> <action application="set"
> data="bridge_pre_execute_bleg_app=execute_extension"/>
>
> <action application="set"
> data="bridge_pre_execute_bleg_data=SETUP_RECORDING XML ${domain_name}"/>
>
> <action application="callcenter" data="Test-bench at tester"/>
>
> </condition>
>
>
>
> Here?s the log of the call as it comes in:
>
> SET [RECORD_APPEND]=[true]
>
> Configuring bind_digit_action to do recording on this session...
>
> Digit parser DPTOOLS: binding *1/features/0 callback: 0x7f37720eed10 data:
> 0x7f367c7dd918
>
> Digit parser DPTOOLS: binding *0/features/0 callback: 0x7f37720eed10 data:
> 0x7f367c7ddab8
>
> Digit parser DPTOOLS: Setting realm to 'features'
>
> EXPORT (export_vars) (REMOTE ONLY) [execute_on_answer]=[record_session
> /usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3]
>
>
>
> The call is then answered by the agent, we see the recording file has been
> created, the agent presses *0 but the call continues recording, log as
> follows:
>
> SET [RECORD_APPEND]=[true]
>
> RTP RECV DTMF *:1536
>
> RTP RECV DTMF 0:1536
>
> Stopping recording...
>
>
> stop_record_session(/usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3)
>
>
>
>
>
> The configuration looks as though it should work, any ideas? Is there
> another method we should be using?
>
>
>
> Many Thanks,
>
> Shaun
>
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
> ______________________________________________________________________
> 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
>



-- 
?talo Rossi
italo at freeswitch.org
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From jungleboogie0 at gmail.com  Fri Apr 22 23:52:59 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Fri, 22 Apr 2016 12:52:59 -0700
Subject: [Freeswitch-users] Freeswitch and Ping
In-Reply-To: <57182913.53106b0a.ea096.27fdSMTPIN_ADDED_MISSING@mx.google.com>
References: <57182913.53106b0a.ea096.27fdSMTPIN_ADDED_MISSING@mx.google.com>
Message-ID: <CAKE2PDtRtkOo+XqoxXXF+hBfinuNn752QLAyZHqe9a2FsUQTuw@mail.gmail.com>

On 20 April 2016 at 18:10, Clive Lansink <clive at lansink.co.nz> wrote:
>
> Is there something about pinging that might impact on Freeswitch or VOIP? What would cause this?

Were you connecting to an ITSP or using freeswitch for sip calling directly?

If it's the former, have your soft client register directly to the
provider and see what happens while running ping?

What's your soft client anyway?


-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From pstarzyk at general-devices.com  Sat Apr 23 00:16:12 2016
From: pstarzyk at general-devices.com (Piotr Starzyk)
Date: Fri, 22 Apr 2016 16:16:12 -0400
Subject: [Freeswitch-users] Recommendations for a 'demo' PBX setup?
Message-ID: <6c79a3a4d22dba757081c9d8afbf0281@mail.gmail.com>

I?m really liking FreeSWITCH so far, and am seriously considering replacing
our current Toshiba Strata system, with a FreeSWITCH implementation.  To
that end, I?m trying to put together a ?proof of concept? system to demo to
my supervisor.  I?ve done a bit of googling, but couldn?t find a
satisfactory answer.



I basically plan on purchasing two SIP phones, and connecting them via
FreeSWITCH to an analog phone line.  I will then demonstrate different
features, like Voicemail, IVR, Presence, Conferencing (basically a lot of
what FreeSWITCH demo dialplan has to offer).



My question is, what are some relatively cheap components that are
compatible with FreeSWITCH?  From my online searches, it looks like Sangoma
is the right choice, but the prices on their FXS/FXO cards start at a
couple hundred dollars, and while it would be fine for final install, I
can?t justify that for a ?proof of concept? setup.  I?m hoping to spend
less than $100 per each component below (so about $400 total).  The less
the better, but it should obviously be usable.



What I?d like is recommendations for the following, FreeSWITCH compatible
hardware:

-          2 SIP Phones

-          FXS/FXO card (PCI or USB, or gateway or whatever is most
economical)

-          Linux (or Windows) PC capable of running FreeSWITCH and
interfacing with the above card



I?d appreciate any suggestions.  Thanks.
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From luis.daniel.lucio at gmail.com  Sat Apr 23 00:28:49 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Fri, 22 Apr 2016 16:28:49 -0400
Subject: [Freeswitch-users] Recommendations for a 'demo' PBX setup?
In-Reply-To: <6c79a3a4d22dba757081c9d8afbf0281@mail.gmail.com>
References: <6c79a3a4d22dba757081c9d8afbf0281@mail.gmail.com>
Message-ID: <CAFLo2QxmqG7FxuVo5E5qZai-5C28d1wu=MSH9TkFzOgHqEs5yA@mail.gmail.com>

Look for voptech IP phones, they are really cheap and zero problems

Such user fxo/fxs, get any did over sip. Depending the country you are, it
could be cents out 10 usd
Le 22 avr. 2016 4:17 PM, "Piotr Starzyk" <pstarzyk at general-devices.com> a
?crit :

> I?m really liking FreeSWITCH so far, and am seriously considering
> replacing our current Toshiba Strata system, with a FreeSWITCH
> implementation.  To that end, I?m trying to put together a ?proof of
> concept? system to demo to my supervisor.  I?ve done a bit of googling, but
> couldn?t find a satisfactory answer.
>
>
>
> I basically plan on purchasing two SIP phones, and connecting them via
> FreeSWITCH to an analog phone line.  I will then demonstrate different
> features, like Voicemail, IVR, Presence, Conferencing (basically a lot of
> what FreeSWITCH demo dialplan has to offer).
>
>
>
> My question is, what are some relatively cheap components that are
> compatible with FreeSWITCH?  From my online searches, it looks like Sangoma
> is the right choice, but the prices on their FXS/FXO cards start at a
> couple hundred dollars, and while it would be fine for final install, I
> can?t justify that for a ?proof of concept? setup.  I?m hoping to spend
> less than $100 per each component below (so about $400 total).  The less
> the better, but it should obviously be usable.
>
>
>
> What I?d like is recommendations for the following, FreeSWITCH compatible
> hardware:
>
> -          2 SIP Phones
>
> -          FXS/FXO card (PCI or USB, or gateway or whatever is most
> economical)
>
> -          Linux (or Windows) PC capable of running FreeSWITCH and
> interfacing with the above card
>
>
>
> I?d appreciate any suggestions.  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 msc at freeswitch.org  Sat Apr 23 01:10:14 2016
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Apr 2016 14:10:14 -0700
Subject: [Freeswitch-users] Recommendations for a 'demo' PBX setup?
In-Reply-To: <CAFLo2QxmqG7FxuVo5E5qZai-5C28d1wu=MSH9TkFzOgHqEs5yA@mail.gmail.com>
References: <6c79a3a4d22dba757081c9d8afbf0281@mail.gmail.com>
	<CAFLo2QxmqG7FxuVo5E5qZai-5C28d1wu=MSH9TkFzOgHqEs5yA@mail.gmail.com>
Message-ID: <CAKzWOxVN9TGC_kesd=1X9e1OwNCa_2SmT0cXEfiM8czu2abraw@mail.gmail.com>

If analog is an absolute must there are small ATA's that can be used. For
example, there's a Cisco SPA232D and its older cousin, the SPA3102. You
might try eBay to see if you can find something inexpensive.
-MSC

On Fri, Apr 22, 2016 at 1:28 PM, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> Look for voptech IP phones, they are really cheap and zero problems
>
> Such user fxo/fxs, get any did over sip. Depending the country you are, it
> could be cents out 10 usd
> Le 22 avr. 2016 4:17 PM, "Piotr Starzyk" <pstarzyk at general-devices.com> a
> ?crit :
>
>> I?m really liking FreeSWITCH so far, and am seriously considering
>> replacing our current Toshiba Strata system, with a FreeSWITCH
>> implementation.  To that end, I?m trying to put together a ?proof of
>> concept? system to demo to my supervisor.  I?ve done a bit of googling, but
>> couldn?t find a satisfactory answer.
>>
>>
>>
>> I basically plan on purchasing two SIP phones, and connecting them via
>> FreeSWITCH to an analog phone line.  I will then demonstrate different
>> features, like Voicemail, IVR, Presence, Conferencing (basically a lot of
>> what FreeSWITCH demo dialplan has to offer).
>>
>>
>>
>> My question is, what are some relatively cheap components that are
>> compatible with FreeSWITCH?  From my online searches, it looks like Sangoma
>> is the right choice, but the prices on their FXS/FXO cards start at a
>> couple hundred dollars, and while it would be fine for final install, I
>> can?t justify that for a ?proof of concept? setup.  I?m hoping to spend
>> less than $100 per each component below (so about $400 total).  The less
>> the better, but it should obviously be usable.
>>
>>
>>
>> What I?d like is recommendations for the following, FreeSWITCH compatible
>> hardware:
>>
>> -          2 SIP Phones
>>
>> -          FXS/FXO card (PCI or USB, or gateway or whatever is most
>> economical)
>>
>> -          Linux (or Windows) PC capable of running FreeSWITCH and
>> interfacing with the above card
>>
>>
>>
>> I?d appreciate any suggestions.  Thanks.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bobjectsfreeswitch at gmail.com  Sat Apr 23 01:40:48 2016
From: bobjectsfreeswitch at gmail.com (Bob Hartwig)
Date: Fri, 22 Apr 2016 16:40:48 -0500
Subject: [Freeswitch-users] Recommendations for a 'demo' PBX setup?
In-Reply-To: <6c79a3a4d22dba757081c9d8afbf0281@mail.gmail.com>
References: <6c79a3a4d22dba757081c9d8afbf0281@mail.gmail.com>
Message-ID: <CAAemr3zfiUkrSidasj7jZyinVMvPbHkRRfs+qxQ25Gft=wRSgw@mail.gmail.com>

For your proof of concept, consider using a Raspberry Pi for the Freeswitch
box, and a separate, inexpensive FXO adapter for your phone line.



On Fri, Apr 22, 2016 at 3:16 PM, Piotr Starzyk <pstarzyk at general-devices.com
> wrote:

> I?m really liking FreeSWITCH so far, and am seriously considering
> replacing our current Toshiba Strata system, with a FreeSWITCH
> implementation.  To that end, I?m trying to put together a ?proof of
> concept? system to demo to my supervisor.  I?ve done a bit of googling, but
> couldn?t find a satisfactory answer.
>
>
>
> I basically plan on purchasing two SIP phones, and connecting them via
> FreeSWITCH to an analog phone line.  I will then demonstrate different
> features, like Voicemail, IVR, Presence, Conferencing (basically a lot of
> what FreeSWITCH demo dialplan has to offer).
>
>
>
> My question is, what are some relatively cheap components that are
> compatible with FreeSWITCH?  From my online searches, it looks like Sangoma
> is the right choice, but the prices on their FXS/FXO cards start at a
> couple hundred dollars, and while it would be fine for final install, I
> can?t justify that for a ?proof of concept? setup.  I?m hoping to spend
> less than $100 per each component below (so about $400 total).  The less
> the better, but it should obviously be usable.
>
>
>
> What I?d like is recommendations for the following, FreeSWITCH compatible
> hardware:
>
> -          2 SIP Phones
>
> -          FXS/FXO card (PCI or USB, or gateway or whatever is most
> economical)
>
> -          Linux (or Windows) PC capable of running FreeSWITCH and
> interfacing with the above card
>
>
>
> I?d appreciate any suggestions.  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 vagarwal at vertical.com  Sat Apr 23 02:03:04 2016
From: vagarwal at vertical.com (Varsha Agarwal)
Date: Fri, 22 Apr 2016 22:03:04 +0000
Subject: [Freeswitch-users] Freeswitch sound installation is failing on
	CentOS 7
Message-ID: <04AB0A185A23864CA9255FD38901F65D0233786872@SCEX1.vertical.com>

Hi,

I have installed FreeSwitch couple of times on CentOS with no problems. I am doing it on a fresh CentOS 7 VM and running into issues installing sounds.

yum install sox freeswitch-sounds*

This command fails with error 404 not found. I tried yum clean all and yum update but no luck. Can someone help?

Thanks,
Varsha


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From joel at gogii.net  Sat Apr 23 03:13:40 2016
From: joel at gogii.net (Joel Serrano)
Date: Fri, 22 Apr 2016 16:13:40 -0700
Subject: [Freeswitch-users] JitterBuffer possible problem when
	configured with number of packets.
In-Reply-To: <CAMtXxQ==0-1GeLik0+4_XfoPYMa7Tq4ue5coP3CsG4pypHQ3tQ@mail.gmail.com>
References: <CAMtXxQ==0-1GeLik0+4_XfoPYMa7Tq4ue5coP3CsG4pypHQ3tQ@mail.gmail.com>
Message-ID: <CAMtXxQmOQDZDtfsiTQFE3Msfgx0wUjJ1J+9wMzzcmwvS3Y=aOA@mail.gmail.com>

Hi,

Does anyone know what the problem can be using "1p" instead of "20" in the
jitterbuffer settings?

Thanks,
Joel.

On Tue, Apr 19, 2016 at 6:38 PM, Joel Serrano <joel at gogii.net> wrote:

> Hi all,
>
> I'm doing several tests jitterbuffer related. I've read in different posts
> to the list that you can specify packet size ("20"), or the number of
> packets ("1p"), why is it not being enabled if I use 1p instead of 20?
> (Example:
> http://lists.freeswitch.org/pipermail/freeswitch-docs/2015-November/000532.html
> )
>
>
> Scenario:
>
> ** Add in the sofia profile:
>
> <param name="auto-jitterbuffer-msec" value="20"/>
>
> --- Make a call
>
> ** Output in debug log:
>
> [...]
> 4557180d-579a-4c3c-98a3-287cb078e913 2016-04-20 00:57:03.570889 [DEBUG]
> switch_core_media.c:1964 Setting Jitterbuffer to 20ms (1 frames) (50 max
> frames)
> [...]
>
>
> Scenario 2:
>
> ** Add in the sofia profile:
>
> <param name="auto-jitterbuffer-msec" value="1p"/>
>
> --- Make a call
>
> Log doesn't say that jitterbuffer has been enabled.
>
>
> Can it be that 1p doesn't output the log but is working anyway?
>
>
> I have removed the auto-jitterbuffer-msec from the sofia profile, and done
> both tests setting jitterbuffer_msec directly in the dialplan with "20" and
> with "1p", the result is the same: 20 works, 1p doesn't.
>
> If I use 1p, I don't have to worry about the size as if its 20ms, 40ms or
> whatever it will work, using "20", the setting would not be valid for
> 40ms.. (please correct me if I'm wrong).
>
> I know this can be a bug and thus should be handled in JIRA, I will be
> happy to create the ticket etc etc, I first want to make sure that the
> problem is not my config.
>
> Tests are with version 1.6.7 -14-d38d065 64bit
>
> Thanks!
> Joel.
>
>
>
>
>
>
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From anthony.minessale at gmail.com  Sat Apr 23 03:57:59 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Apr 2016 18:57:59 -0500
Subject: [Freeswitch-users] JitterBuffer possible problem when
 configured with number of packets.
In-Reply-To: <CAMtXxQmOQDZDtfsiTQFE3Msfgx0wUjJ1J+9wMzzcmwvS3Y=aOA@mail.gmail.com>
References: <CAMtXxQ==0-1GeLik0+4_XfoPYMa7Tq4ue5coP3CsG4pypHQ3tQ@mail.gmail.com>
	<CAMtXxQmOQDZDtfsiTQFE3Msfgx0wUjJ1J+9wMzzcmwvS3Y=aOA@mail.gmail.com>
Message-ID: <CAKbxfG9S7BHq2SOfxsNWM7MP0J6KMmpy54oy-hQn1zfxGti9Wg@mail.gmail.com>

Last we check it worked but you can't report bugs here.  This is the
mailing list not the bug tracker so nobody will see your issue.
I say about once a week, you don't have to ask if a bug is a bug first then
report it because that costs 2x the resources.
Report bugs, if its not a bug we close it.


On Fri, Apr 22, 2016 at 6:13 PM, Joel Serrano <joel at gogii.net> wrote:

> Hi,
>
> Does anyone know what the problem can be using "1p" instead of "20" in the
> jitterbuffer settings?
>
> Thanks,
> Joel.
>
> On Tue, Apr 19, 2016 at 6:38 PM, Joel Serrano <joel at gogii.net> wrote:
>
>> Hi all,
>>
>> I'm doing several tests jitterbuffer related. I've read in different
>> posts to the list that you can specify packet size ("20"), or the number of
>> packets ("1p"), why is it not being enabled if I use 1p instead of 20?
>> (Example:
>> http://lists.freeswitch.org/pipermail/freeswitch-docs/2015-November/000532.html
>> )
>>
>>
>> Scenario:
>>
>> ** Add in the sofia profile:
>>
>> <param name="auto-jitterbuffer-msec" value="20"/>
>>
>> --- Make a call
>>
>> ** Output in debug log:
>>
>> [...]
>> 4557180d-579a-4c3c-98a3-287cb078e913 2016-04-20 00:57:03.570889 [DEBUG]
>> switch_core_media.c:1964 Setting Jitterbuffer to 20ms (1 frames) (50 max
>> frames)
>> [...]
>>
>>
>> Scenario 2:
>>
>> ** Add in the sofia profile:
>>
>> <param name="auto-jitterbuffer-msec" value="1p"/>
>>
>> --- Make a call
>>
>> Log doesn't say that jitterbuffer has been enabled.
>>
>>
>> Can it be that 1p doesn't output the log but is working anyway?
>>
>>
>> I have removed the auto-jitterbuffer-msec from the sofia profile, and
>> done both tests setting jitterbuffer_msec directly in the dialplan with
>> "20" and with "1p", the result is the same: 20 works, 1p doesn't.
>>
>> If I use 1p, I don't have to worry about the size as if its 20ms, 40ms or
>> whatever it will work, using "20", the setting would not be valid for
>> 40ms.. (please correct me if I'm wrong).
>>
>> I know this can be a bug and thus should be handled in JIRA, I will be
>> happy to create the ticket etc etc, I first want to make sure that the
>> problem is not my config.
>>
>> Tests are with version 1.6.7 -14-d38d065 64bit
>>
>> 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
>



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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From gregor at infomedia.si  Sat Apr 23 04:56:16 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sat, 23 Apr 2016 02:56:16 +0200
Subject: [Freeswitch-users] Verto Caller Id and Caller name
Message-ID: <CAC7ttFFoTN90ADHyTzLYe7i2UotgC1dwLgcSSyDtt2+X_6zptg@mail.gmail.com>

I can set in verto javascript initialization caller id and caller name. But
can I force or override caller id and name in user registration variables
or somewhere else, because I do not want to set caller name in javascript,
since user can tweak it..

Any experience?
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From mike at jerris.com  Sat Apr 23 09:19:06 2016
From: mike at jerris.com (Michael Jerris)
Date: Sat, 23 Apr 2016 01:19:06 -0400
Subject: [Freeswitch-users] Freeswitch sound installation is failing on
 CentOS 7
In-Reply-To: <04AB0A185A23864CA9255FD38901F65D0233786872@SCEX1.vertical.com>
References: <04AB0A185A23864CA9255FD38901F65D0233786872@SCEX1.vertical.com>
Message-ID: <CAEwTsAj4kqWPoJmHoyfOO-XGCfX2g_KVfE_YWp366uizqupweA@mail.gmail.com>

32 bit or 64?  I think we had an issue with those missing from 32 but no
one ever filed a jira so it never got fixed

On Friday, April 22, 2016, Varsha Agarwal <vagarwal at vertical.com> wrote:

> Hi,
>
>
>
> I have installed FreeSwitch couple of times on CentOS with no problems. I
> am doing it on a fresh CentOS 7 VM and running into issues installing
> sounds.
>
>
>
> yum install sox freeswitch-sounds*
>
>
>
> This command fails with error 404 not found. I tried yum clean all and yum
> update but no luck. Can someone help?
>
>
>
> Thanks,
>
> Varsha
>
>
>
>
>
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From mike at jerris.com  Sat Apr 23 09:19:54 2016
From: mike at jerris.com (Michael Jerris)
Date: Sat, 23 Apr 2016 01:19:54 -0400
Subject: [Freeswitch-users] Verto Caller Id and Caller name
In-Reply-To: <CAC7ttFFoTN90ADHyTzLYe7i2UotgC1dwLgcSSyDtt2+X_6zptg@mail.gmail.com>
References: <CAC7ttFFoTN90ADHyTzLYe7i2UotgC1dwLgcSSyDtt2+X_6zptg@mail.gmail.com>
Message-ID: <CAEwTsAiDP1-7MLVxDcSam6kqkK9rBp0utCd3q69ruVHqGkW=uA@mail.gmail.com>

you can set variables from the reg

On Friday, April 22, 2016, Gregor Nanger <gregor at infomedia.si> wrote:

> I can set in verto javascript initialization caller id and caller name.
> But can I force or override caller id and name in user registration
> variables or somewhere else, because I do not want to set caller name in
> javascript, since user can tweak it..
>
> Any experience?
>
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From s.safarov at gmail.com  Sat Apr 23 14:19:12 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 23 Apr 2016 10:19:12 +0000
Subject: [Freeswitch-users] Freeswitch sound installation is failing on
 CentOS 7
In-Reply-To: <CAEwTsAj4kqWPoJmHoyfOO-XGCfX2g_KVfE_YWp366uizqupweA@mail.gmail.com>
References: <04AB0A185A23864CA9255FD38901F65D0233786872@SCEX1.vertical.com>
	<CAEwTsAj4kqWPoJmHoyfOO-XGCfX2g_KVfE_YWp366uizqupweA@mail.gmail.com>
Message-ID: <CAHtxdDfS=Ds+Cq4=h_Jv6BNzt1OmuPx=-BG2kBj166zEWVJi0A@mail.gmail.com>

CentOS 7 is only 64 bit.

On Sat, Apr 23, 2016, 08:20 Michael Jerris <mike at jerris.com> wrote:

> 32 bit or 64?  I think we had an issue with those missing from 32 but no
> one ever filed a jira so it never got fixed
>
> On Friday, April 22, 2016, Varsha Agarwal <vagarwal at vertical.com> wrote:
>
>> Hi,
>>
>>
>>
>> I have installed FreeSwitch couple of times on CentOS with no problems. I
>> am doing it on a fresh CentOS 7 VM and running into issues installing
>> sounds.
>>
>>
>>
>> yum install sox freeswitch-sounds*
>>
>>
>>
>> This command fails with error 404 not found. I tried yum clean all and
>> yum update but no luck. Can someone help?
>>
>>
>>
>> Thanks,
>>
>> Varsha
>>
>>
>>
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From s.safarov at gmail.com  Sat Apr 23 14:21:17 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 23 Apr 2016 10:21:17 +0000
Subject: [Freeswitch-users] Freeswitch sound installation is failing on
 CentOS 7
In-Reply-To: <CAHtxdDfS=Ds+Cq4=h_Jv6BNzt1OmuPx=-BG2kBj166zEWVJi0A@mail.gmail.com>
References: <04AB0A185A23864CA9255FD38901F65D0233786872@SCEX1.vertical.com>
	<CAEwTsAj4kqWPoJmHoyfOO-XGCfX2g_KVfE_YWp366uizqupweA@mail.gmail.com>
	<CAHtxdDfS=Ds+Cq4=h_Jv6BNzt1OmuPx=-BG2kBj166zEWVJi0A@mail.gmail.com>
Message-ID: <CAHtxdDekNBBgZgTPBrfUfk61a=WhUYVA-XBYyd05_+qvCHw_rQ@mail.gmail.com>

I am sorry
32 bit also exist
http://mirror.centos.org/altarch/7/isos/i386/

On Sat, Apr 23, 2016, 13:19 Sergey Safarov <s.safarov at gmail.com> wrote:

> CentOS 7 is only 64 bit.
>
> On Sat, Apr 23, 2016, 08:20 Michael Jerris <mike at jerris.com> wrote:
>
>> 32 bit or 64?  I think we had an issue with those missing from 32 but no
>> one ever filed a jira so it never got fixed
>>
>> On Friday, April 22, 2016, Varsha Agarwal <vagarwal at vertical.com> wrote:
>>
>>> Hi,
>>>
>>>
>>>
>>> I have installed FreeSwitch couple of times on CentOS with no problems.
>>> I am doing it on a fresh CentOS 7 VM and running into issues installing
>>> sounds.
>>>
>>>
>>>
>>> yum install sox freeswitch-sounds*
>>>
>>>
>>>
>>> This command fails with error 404 not found. I tried yum clean all and
>>> yum update but no luck. Can someone help?
>>>
>>>
>>>
>>> Thanks,
>>>
>>> Varsha
>>>
>>>
>>>
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
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From gregor at infomedia.si  Sat Apr 23 14:58:41 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Sat, 23 Apr 2016 10:58:41 +0000
Subject: [Freeswitch-users] Verto Caller Id and Caller name
In-Reply-To: <CAEwTsAiDP1-7MLVxDcSam6kqkK9rBp0utCd3q69ruVHqGkW=uA@mail.gmail.com>
References: <CAC7ttFFoTN90ADHyTzLYe7i2UotgC1dwLgcSSyDtt2+X_6zptg@mail.gmail.com>
	<CAEwTsAiDP1-7MLVxDcSam6kqkK9rBp0utCd3q69ruVHqGkW=uA@mail.gmail.com>
Message-ID: <CAC7ttFHLXx0uLr0qM7eMp-+b3eHevnARtEbSpG9vfb4qN807zQ@mail.gmail.com>

I am using xml_curl for registration and tried to set variables, but
without any succes.

Can you please give me example.

On Sat, Apr 23, 2016, 07:21 Michael Jerris <mike at jerris.com> wrote:

> you can set variables from the reg
>
> On Friday, April 22, 2016, Gregor Nanger <gregor at infomedia.si> wrote:
>
>> I can set in verto javascript initialization caller id and caller name.
>> But can I force or override caller id and name in user registration
>> variables or somewhere else, because I do not want to set caller name in
>> javascript, since user can tweak it..
>>
>> Any experience?
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
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 mandra at gmail.com  Sat Apr 23 18:24:47 2016
From: mandra at gmail.com (Chris Mandra)
Date: Sat, 23 Apr 2016 10:24:47 -0400
Subject: [Freeswitch-users] module not unloading
In-Reply-To: <CAJdRzN5hseXNb7ehuEkMFnR6Jy6Un=QRg+aF7tJc-cEDOX=SGg@mail.gmail.com>
References: <CAJdRzN5hseXNb7ehuEkMFnR6Jy6Un=QRg+aF7tJc-cEDOX=SGg@mail.gmail.com>
Message-ID: <CAJdRzN4u8-Esd4K2jA9cO9BjAA9DOxT1n2-DQ+ROVpgSce31og@mail.gmail.com>

Hi guys, any thoughts about this?

On Thursday, April 21, 2016, Chris Mandra <mandra at gmail.com> wrote:

> Hi Guys - I hope you are well
> I never really resolved the issue I was having with modules saying they're
> unloading, but not really unloading (unless I restart freeswitch)
>
> I?m having that issue with the module not unloading again when I add in
> some flex/bison parser code. It seems the parser is causing the problem
> where module doesn?t unload. What's the best way to diagnose this issue.
>  freeswitch reports module as unloading but lsof shows it is still being
> retained
> Thanks!
> chris
>
>
>

-- 
mandra
c:410.258.5281
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From vagarwal at vertical.com  Sat Apr 23 19:10:40 2016
From: vagarwal at vertical.com (Varsha Agarwal)
Date: Sat, 23 Apr 2016 15:10:40 +0000
Subject: [Freeswitch-users] Freeswitch sound installation is failing on
 CentOS 7
In-Reply-To: <CAHtxdDekNBBgZgTPBrfUfk61a=WhUYVA-XBYyd05_+qvCHw_rQ@mail.gmail.com>
References: <04AB0A185A23864CA9255FD38901F65D0233786872@SCEX1.vertical.com>
	<CAEwTsAj4kqWPoJmHoyfOO-XGCfX2g_KVfE_YWp366uizqupweA@mail.gmail.com>
	<CAHtxdDfS=Ds+Cq4=h_Jv6BNzt1OmuPx=-BG2kBj166zEWVJi0A@mail.gmail.com>,
	<CAHtxdDekNBBgZgTPBrfUfk61a=WhUYVA-XBYyd05_+qvCHw_rQ@mail.gmail.com>
Message-ID: <B8EE5390-7855-4535-9455-70AA2A59E8AA@vertical.com>

64 bit. It looks like some problem with central repository as I have done this install in the past with no problem on CentOS 7.



On Apr 23, 2016, at 3:23 AM, Sergey Safarov <s.safarov at gmail.com<mailto:s.safarov at gmail.com>> wrote:


I am sorry
32 bit also exist
http://mirror.centos.org/altarch/7/isos/i386/

On Sat, Apr 23, 2016, 13:19 Sergey Safarov <s.safarov at gmail.com<mailto:s.safarov at gmail.com>> wrote:

CentOS 7 is only 64 bit.

On Sat, Apr 23, 2016, 08:20 Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
32 bit or 64?  I think we had an issue with those missing from 32 but no one ever filed a jira so it never got fixed

On Friday, April 22, 2016, Varsha Agarwal <vagarwal at vertical.com<mailto:vagarwal at vertical.com>> wrote:
Hi,

I have installed FreeSwitch couple of times on CentOS with no problems. I am doing it on a fresh CentOS 7 VM and running into issues installing sounds.

yum install sox freeswitch-sounds*

This command fails with error 404 not found. I tried yum clean all and yum update but no luck. Can someone help?

Thanks,
Varsha


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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users>
http://www.freeswitch.org
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From krice at freeswitch.org  Sat Apr 23 19:35:09 2016
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 23 Apr 2016 10:35:09 -0500
Subject: [Freeswitch-users] Freeswitch sound installation is failing on
	CentOS 7
In-Reply-To: <B8EE5390-7855-4535-9455-70AA2A59E8AA@vertical.com>
References: <04AB0A185A23864CA9255FD38901F65D0233786872@SCEX1.vertical.com>	<CAEwTsAj4kqWPoJmHoyfOO-XGCfX2g_KVfE_YWp366uizqupweA@mail.gmail.com>	<CAHtxdDfS=Ds+Cq4=h_Jv6BNzt1OmuPx=-BG2kBj166zEWVJi0A@mail.gmail.com>,
	<CAHtxdDekNBBgZgTPBrfUfk61a=WhUYVA-XBYyd05_+qvCHw_rQ@mail.gmail.com>
	<B8EE5390-7855-4535-9455-70AA2A59E8AA@vertical.com>
Message-ID: <098f01d19d75$b8a9b840$29fd28c0$@freeswitch.org>

We do not publish 32bit RPMs for Centos7. Running i386 on x86-64 hardware is
like putting a 100cc Motor Cycle engine in Super Car

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Varsha
Agarwal
Sent: Saturday, April 23, 2016 10:11 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch sound installation is failing on
CentOS 7

 

64 bit. It looks like some problem with central repository as I have done
this install in the past with no problem on CentOS 7. 

 


On Apr 23, 2016, at 3:23 AM, Sergey Safarov <s.safarov at gmail.com
<mailto:s.safarov at gmail.com> > wrote:

I am sorry 
32 bit also exist
http://mirror.centos.org/altarch/7/isos/i386/

 

On Sat, Apr 23, 2016, 13:19 Sergey Safarov <s.safarov at gmail.com
<mailto:s.safarov at gmail.com> > wrote:

CentOS 7 is only 64 bit.

 

On Sat, Apr 23, 2016, 08:20 Michael Jerris <mike at jerris.com
<mailto:mike at jerris.com> > wrote:

32 bit or 64?  I think we had an issue with those missing from 32 but no one
ever filed a jira so it never got fixed

On Friday, April 22, 2016, Varsha Agarwal <vagarwal at vertical.com
<mailto:vagarwal at vertical.com> > wrote:

Hi,

 

I have installed FreeSwitch couple of times on CentOS with no problems. I am
doing it on a fresh CentOS 7 VM and running into issues installing sounds.

 

yum install sox freeswitch-sounds*

 

This command fails with error 404 not found. I tried yum clean all and yum
update but no luck. Can someone help?

 

Thanks,

Varsha

 

 

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

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

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

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

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

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

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From jonlederman at gmail.com  Sat Apr 23 20:25:29 2016
From: jonlederman at gmail.com (Jon Lederman)
Date: Sat, 23 Apr 2016 12:25:29 -0400
Subject: [Freeswitch-users] module not unloading
In-Reply-To: <CAJdRzN4u8-Esd4K2jA9cO9BjAA9DOxT1n2-DQ+ROVpgSce31og@mail.gmail.com>
References: <CAJdRzN5hseXNb7ehuEkMFnR6Jy6Un=QRg+aF7tJc-cEDOX=SGg@mail.gmail.com>
	<CAJdRzN4u8-Esd4K2jA9cO9BjAA9DOxT1n2-DQ+ROVpgSce31og@mail.gmail.com>
Message-ID: <FC6CD1E5-9FDA-4FC4-837E-A26EA8A1D7D2@gmail.com>

I already solved this on my own

Sent from my iPad

> On Apr 23, 2016, at 10:24 AM, Chris Mandra <mandra at gmail.com> wrote:
> 
> Hi guys, any thoughts about this?
> 
>> On Thursday, April 21, 2016, Chris Mandra <mandra at gmail.com> wrote:
>> Hi Guys - I hope you are well
>> I never really resolved the issue I was having with modules saying they're unloading, but not really unloading (unless I restart freeswitch) 
>> I?m having that issue with the module not unloading again when I add in some flex/bison parser code. It seems the parser is causing the problem where module doesn?t unload. What's the best way to diagnose this issue.  freeswitch reports module as unloading but lsof shows it is still being retained
>> 
>> Thanks!
>> chris
> 
> 
> -- 
> mandra
> c:410.258.5281
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From cmrienzo at gmail.com  Sat Apr 23 22:40:47 2016
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Sat, 23 Apr 2016 14:40:47 -0400
Subject: [Freeswitch-users] module not unloading
In-Reply-To: <CAJdRzN4u8-Esd4K2jA9cO9BjAA9DOxT1n2-DQ+ROVpgSce31og@mail.gmail.com>
References: <CAJdRzN5hseXNb7ehuEkMFnR6Jy6Un=QRg+aF7tJc-cEDOX=SGg@mail.gmail.com>
	<CAJdRzN4u8-Esd4K2jA9cO9BjAA9DOxT1n2-DQ+ROVpgSce31og@mail.gmail.com>
Message-ID: <2BE0224F-00D8-4D5A-AD6D-7EB548E4FF9A@gmail.com>

Gdb the process and see if it is stuck.  You can also step through the code as it's unloading. 

Chris


> On Apr 23, 2016, at 10:24, Chris Mandra <mandra at gmail.com> wrote:
> 
> Hi guys, any thoughts about this?
> 
>> On Thursday, April 21, 2016, Chris Mandra <mandra at gmail.com> wrote:
>> Hi Guys - I hope you are well
>> I never really resolved the issue I was having with modules saying they're unloading, but not really unloading (unless I restart freeswitch) 
>> I?m having that issue with the module not unloading again when I add in some flex/bison parser code. It seems the parser is causing the problem where module doesn?t unload. What's the best way to diagnose this issue.  freeswitch reports module as unloading but lsof shows it is still being retained
>> 
>> Thanks!
>> chris
> 
> 
> -- 
> mandra
> c:410.258.5281
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From s.safarov at gmail.com  Sat Apr 23 23:30:51 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 23 Apr 2016 19:30:51 +0000
Subject: [Freeswitch-users] Freeswitch sound installation is failing on
 CentOS 7
In-Reply-To: <04AB0A185A23864CA9255FD38901F65D0233786872@SCEX1.vertical.com>
References: <04AB0A185A23864CA9255FD38901F65D0233786872@SCEX1.vertical.com>
Message-ID: <CAHtxdDdTDmW40YB7cY0LAege_QLxSMS_r0-ob5AXKDUG0=ooqg@mail.gmail.com>

Please display last 10 string before error and rest strings of command "yum
install sox freeswitch-sounds*" output.

Sergey

??, 23 ???. 2016 ?. ? 1:03, Varsha Agarwal <vagarwal at vertical.com>:

> Hi,
>
>
>
> I have installed FreeSwitch couple of times on CentOS with no problems. I
> am doing it on a fresh CentOS 7 VM and running into issues installing
> sounds.
>
>
>
> yum install sox freeswitch-sounds*
>
>
>
> This command fails with error 404 not found. I tried yum clean all and yum
> update but no luck. Can someone help?
>
>
>
> Thanks,
>
> Varsha
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mandra at gmail.com  Sun Apr 24 00:27:08 2016
From: mandra at gmail.com (Chris Mandra)
Date: Sat, 23 Apr 2016 16:27:08 -0400
Subject: [Freeswitch-users] Programmatic way to get BOTH Channel IDs of a
	bridged call?
Message-ID: <CAJdRzN4QLQoruMew7O-LqeZS9_AJckPSWnMcEnLVJPpEzbby-g@mail.gmail.com>

Hey guys - I have another questions:
Is there a programatic way to obtain BOTH the channel ids of a bridged call?
I only have the uuid of one call leg.  How do i get the other uuid?

(similar to what "show calls" or "show channels" shows in cli
but I need to start with one call leg uuid and get the other
from it)

Thanks!
chris
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From s.safarov at gmail.com  Sun Apr 24 01:22:46 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Sat, 23 Apr 2016 21:22:46 +0000
Subject: [Freeswitch-users] Programmatic way to get BOTH Channel IDs of
 a bridged call?
In-Reply-To: <CAJdRzN4QLQoruMew7O-LqeZS9_AJckPSWnMcEnLVJPpEzbby-g@mail.gmail.com>
References: <CAJdRzN4QLQoruMew7O-LqeZS9_AJckPSWnMcEnLVJPpEzbby-g@mail.gmail.com>
Message-ID: <CAHtxdDcUQ3FH5quUWVkUV0xQrRQzSq63stJLM09_H2+MPE_LMA@mail.gmail.com>

Check channel vars. One is contains peer session/channel id.

On Sat, Apr 23, 2016, 23:28 Chris Mandra <mandra at gmail.com> wrote:

> Hey guys - I have another questions:
> Is there a programatic way to obtain BOTH the channel ids of a bridged
> call?
> I only have the uuid of one call leg.  How do i get the other uuid?
>
> (similar to what "show calls" or "show channels" shows in cli
> but I need to start with one call leg uuid and get the other
> from it)
>
> Thanks!
> chris
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From colin.morelli at gmail.com  Sun Apr 24 02:10:21 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Sat, 23 Apr 2016 22:10:21 +0000
Subject: [Freeswitch-users] Programmatic way to get BOTH Channel IDs of
 a bridged call?
In-Reply-To: <CAHtxdDcUQ3FH5quUWVkUV0xQrRQzSq63stJLM09_H2+MPE_LMA@mail.gmail.com>
References: <CAJdRzN4QLQoruMew7O-LqeZS9_AJckPSWnMcEnLVJPpEzbby-g@mail.gmail.com>
	<CAHtxdDcUQ3FH5quUWVkUV0xQrRQzSq63stJLM09_H2+MPE_LMA@mail.gmail.com>
Message-ID: <CAPtU-Uo7FCiKCryPBdu3ZgqmS_Pa8bpmp3gs1a4h9nyWxJBHLw@mail.gmail.com>

I believe the channel variable "call_uuid" (or "originating_leg_uuid")
contains the UUID of the call that the current session was bridged from.
So, from a bridged call it should be easy to get back to the original that
way. Not sure on the other way around.

On Sat, Apr 23, 2016 at 5:25 PM Sergey Safarov <s.safarov at gmail.com> wrote:

> Check channel vars. One is contains peer session/channel id.
>
> On Sat, Apr 23, 2016, 23:28 Chris Mandra <mandra at gmail.com> wrote:
>
>> Hey guys - I have another questions:
>> Is there a programatic way to obtain BOTH the channel ids of a bridged
>> call?
>> I only have the uuid of one call leg.  How do i get the other uuid?
>>
>> (similar to what "show calls" or "show channels" shows in cli
>> but I need to start with one call leg uuid and get the other
>> from it)
>>
>> Thanks!
>> chris
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mandra at gmail.com  Sun Apr 24 05:33:36 2016
From: mandra at gmail.com (Chris Mandra)
Date: Sat, 23 Apr 2016 21:33:36 -0400
Subject: [Freeswitch-users] Programmatic way to get BOTH Channel IDs of
	a bridged call?
In-Reply-To: <CAPtU-Uo7FCiKCryPBdu3ZgqmS_Pa8bpmp3gs1a4h9nyWxJBHLw@mail.gmail.com>
References: <CAJdRzN4QLQoruMew7O-LqeZS9_AJckPSWnMcEnLVJPpEzbby-g@mail.gmail.com>
	<CAHtxdDcUQ3FH5quUWVkUV0xQrRQzSq63stJLM09_H2+MPE_LMA@mail.gmail.com>
	<CAPtU-Uo7FCiKCryPBdu3ZgqmS_Pa8bpmp3gs1a4h9nyWxJBHLw@mail.gmail.com>
Message-ID: <CAJdRzN4EwmyK8-vtbahJgPnTN36Dvt=9JGOp2=D_KpUpzcgd5Q@mail.gmail.com>

Thanks guys

On Saturday, April 23, 2016, Colin Morelli <colin.morelli at gmail.com> wrote:

> I believe the channel variable "call_uuid" (or "originating_leg_uuid")
> contains the UUID of the call that the current session was bridged from.
> So, from a bridged call it should be easy to get back to the original that
> way. Not sure on the other way around.
>
> On Sat, Apr 23, 2016 at 5:25 PM Sergey Safarov <s.safarov at gmail.com
> <javascript:_e(%7B%7D,'cvml','s.safarov at gmail.com');>> wrote:
>
>> Check channel vars. One is contains peer session/channel id.
>>
>> On Sat, Apr 23, 2016, 23:28 Chris Mandra <mandra at gmail.com
>> <javascript:_e(%7B%7D,'cvml','mandra at gmail.com');>> wrote:
>>
>>> Hey guys - I have another questions:
>>> Is there a programatic way to obtain BOTH the channel ids of a bridged
>>> call?
>>> I only have the uuid of one call leg.  How do i get the other uuid?
>>>
>>> (similar to what "show calls" or "show channels" shows in cli
>>> but I need to start with one call leg uuid and get the other
>>> from it)
>>>
>>> Thanks!
>>> chris
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>

-- 
mandra
c:410.258.5281
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From nishadmusthafa at gmail.com  Sun Apr 24 08:47:41 2016
From: nishadmusthafa at gmail.com (Nishad Musthafa)
Date: Sun, 24 Apr 2016 04:47:41 +0000
Subject: [Freeswitch-users] Programmatic way to get BOTH Channel IDs of
 a bridged call?
In-Reply-To: <CAJdRzN4EwmyK8-vtbahJgPnTN36Dvt=9JGOp2=D_KpUpzcgd5Q@mail.gmail.com>
References: <CAJdRzN4QLQoruMew7O-LqeZS9_AJckPSWnMcEnLVJPpEzbby-g@mail.gmail.com>
	<CAHtxdDcUQ3FH5quUWVkUV0xQrRQzSq63stJLM09_H2+MPE_LMA@mail.gmail.com>
	<CAPtU-Uo7FCiKCryPBdu3ZgqmS_Pa8bpmp3gs1a4h9nyWxJBHLw@mail.gmail.com>
	<CAJdRzN4EwmyK8-vtbahJgPnTN36Dvt=9JGOp2=D_KpUpzcgd5Q@mail.gmail.com>
Message-ID: <CADeOUzvo+x6GAbVQh-zVKN74i7h81so1J1eBAA7OBXBfAWHccA@mail.gmail.com>

I believe bridge_uuid is the channel variable that you could use to go the
other way around.
On Sun, Apr 24, 2016 at 7:04 AM Chris Mandra <mandra at gmail.com> wrote:

> Thanks guys
>
> On Saturday, April 23, 2016, Colin Morelli <colin.morelli at gmail.com>
> wrote:
>
>> I believe the channel variable "call_uuid" (or "originating_leg_uuid")
>> contains the UUID of the call that the current session was bridged from.
>> So, from a bridged call it should be easy to get back to the original that
>> way. Not sure on the other way around.
>>
>> On Sat, Apr 23, 2016 at 5:25 PM Sergey Safarov <s.safarov at gmail.com>
>> wrote:
>>
>>> Check channel vars. One is contains peer session/channel id.
>>>
>>> On Sat, Apr 23, 2016, 23:28 Chris Mandra <mandra at gmail.com> wrote:
>>>
>>>> Hey guys - I have another questions:
>>>> Is there a programatic way to obtain BOTH the channel ids of a bridged
>>>> call?
>>>> I only have the uuid of one call leg.  How do i get the other uuid?
>>>>
>>>> (similar to what "show calls" or "show channels" shows in cli
>>>> but I need to start with one call leg uuid and get the other
>>>> from it)
>>>>
>>>> Thanks!
>>>> chris
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>
> --
> mandra
> c:410.258.5281
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From shaun.stokes at itec-support.co.uk  Mon Apr 25 12:10:45 2016
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Mon, 25 Apr 2016 08:10:45 +0000
Subject: [Freeswitch-users] Call Recording On Demand for Call Center
 Queues (mod_callcenter)
In-Reply-To: <CABMX_w_qtSXYwh_U6ZXD9k43gLOH5j5hYw=yey1b8ktKo72ufw@mail.gmail.com>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB555D@mbx-01.sysconfig.co.uk>
	<6FD2F8B5BB72834E9939AEDF9FB802A901E7BB5705@mbx-01.sysconfig.co.uk>
	<CABMX_w_qtSXYwh_U6ZXD9k43gLOH5j5hYw=yey1b8ktKo72ufw@mail.gmail.com>
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB64BE@mbx-01.sysconfig.co.uk>

Hi ?talo,

Unfortunately execute_on_answer doesn?t have any effect if we use this in cc_export_vars or in the agents contact field, with or with-out the nolocal: prefix. We have replaced our variable ${rec_file} with a fixed recording file path for testing purposes so we could rule that out.

Here?s a list of variables we?ve tried in cc_export_vars and the agents contact field for testing purposes:
execute_on_answer=nolocal:record_session /usr/local/freeswitch/recordings/test.mp3
execute_on_answer=record_session /usr/local/freeswitch/recordings/test.mp3
execute_on_answer='record_session /usr/local/freeswitch/recordings/test.mp3'
api_on_answer=nolocal:uuid_record ${uuid} start /usr/local/freeswitch/recordings/test.mp3
api_on_answer=uuid_record ${uuid} start /usr/local/freeswitch/recordings/test.mp3
api_on_answer='uuid_record ${uuid} start /usr/local/freeswitch/recordings/test.mp3'

If you or anyone else here has any other ideas that would be a great help to us.

Many Thanks,
Shaun

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: 22 April 2016 20:47
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Call Recording On Demand for Call Center Queues (mod_callcenter)

Use cc_export_vars https://freeswitch.org/confluence/display/FREESWITCH/mod_callcenter#mod_callcenter-cc_export_vars

Or you can set variables in the agent's contact field: {myvar=myvalue}user/number

On Fri, Apr 22, 2016 at 12:25 PM, Shaun Stokes <shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>> wrote:
We have been troubleshooting this further, it seems that this line has no effect when using mod_callcenter:
<action application="export" data="nolocal:execute_on_answer=record_session ${rec_file}"/>

This parameter is set in callcenter.conf if we remove this the call doesn?t record:
<param name="record-template" value="=$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}"/>

The file path\name from the above matches that of the call we?re trying to pause, but it seems we can?t pause the recording if this has been started by mod_callcenter, there are no errors. We can only start and pause call recording when this isn?t started using mod_callcenter by default.

Is there any way around this, maybe a different command for mod_callcenter that can be executed on the agent leg using dtmf to stop the recording, or another way we can start recording after being answered by the agent?

Thanks,
Shaun


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Shaun Stokes
Sent: 22 April 2016 14:56
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] Call Recording On Demand for Call Center Queues (mod_callcenter)

Hi,

We?re trying to setup call recording on demand for call center queues (mod_callcenter) but are experiencing tremendous trouble. Is it possible for the agent to perform pause\resume call recording on mod_callcenter calls which are recorded by default upon answer?

Here?s the important elements of our dialplan:
<condition field="${rec_file}" expression="^(.)+">
                <action application="set" inline="true" data="RECORD_APPEND=true"/>
                <anti-action application="set" inline="true" data="RECORD_APPEND=true"/>
                <anti-action application="export" inline="true" data="rec_file=$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}"/>
</condition>
<condition field="destination_number" expression="^SETUP_RECORDING$">
                <action application="log" data="INFO Configuring bind_digit_action to do recording on this session..."/>
                <action application="bind_digit_action" data="features,*1,exec:execute_extension,START_RECORDING XML ${domain_name}"/>
                <action application="bind_digit_action" data="features,*0,exec:execute_extension,STOP_RECORDING XML ${domain_name}"/>
                <action application="digit_action_set_realm" data=" features"/>
</condition>
<condition field="destination_number" expression="^START_RECORDING$">
                <action application="log" data="INFO Starting recording..."/>
                <action application="record_session" data="${rec_file}"/>
</condition>
<condition field="destination_number" expression="^STOP_RECORDING$">
                <action application="log" data="INFO Stopping recording..."/>
                <action application="stop_record_session" data="${rec_file}"/>
</condition>
<condition field="destination_number" expression="^837$">
<action application="export" data="nolocal:execute_on_answer=record_session ${rec_file}"/>
<action application="set" data="cc_export_vars=rec_file"/>
<action application="set" data="bridge_pre_execute_bleg_app=execute_extension"/>
<action application="set" data="bridge_pre_execute_bleg_data=SETUP_RECORDING XML ${domain_name}"/>
<action application="callcenter" data="Test-bench at tester"/>
</condition>

Here?s the log of the call as it comes in:
SET [RECORD_APPEND]=[true]
Configuring bind_digit_action to do recording on this session...
Digit parser DPTOOLS: binding *1/features/0 callback: 0x7f37720eed10 data: 0x7f367c7dd918
Digit parser DPTOOLS: binding *0/features/0 callback: 0x7f37720eed10 data: 0x7f367c7ddab8
Digit parser DPTOOLS: Setting realm to 'features'
EXPORT (export_vars) (REMOTE ONLY) [execute_on_answer]=[record_session /usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3]

The call is then answered by the agent, we see the recording file has been created, the agent presses *0 but the call continues recording, log as follows:
SET [RECORD_APPEND]=[true]
RTP RECV DTMF *:1536
RTP RECV DTMF 0:1536
Stopping recording...
stop_record_session(/usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3)


The configuration looks as though it should work, any ideas? Is there another method we should be using?

Many Thanks,
Shaun

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

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

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



--
?talo Rossi
italo at freeswitch.org<mailto:italo at freeswitch.org>

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From telishisheer at gmail.com  Mon Apr 25 14:02:19 2016
From: telishisheer at gmail.com (Shisheer Teli)
Date: Mon, 25 Apr 2016 15:32:19 +0530
Subject: [Freeswitch-users] Extensions not registered on IPv6 address
Message-ID: <CACkRGLw37gi=sJYNOgMy=CgF8xKopOPT=7NpjnGfw8yG4dFZPA@mail.gmail.com>

Dear Team,

I installed FreeSWITCH. Directory and dialplan are default ones.  My phones
are able to register when using just IPv4, and I can call from  one to
another.

When I try the same thing with IPv6, phones not registered.

Am I doing something wrong? Maybe I forgot to configure something, but I
can't see what it is.


-- 
Regards,
Shisheer T
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From shaun.stokes at itec-support.co.uk  Mon Apr 25 14:07:49 2016
From: shaun.stokes at itec-support.co.uk (Shaun Stokes)
Date: Mon, 25 Apr 2016 10:07:49 +0000
Subject: [Freeswitch-users] Call Recording On Demand for Call Center
 Queues (mod_callcenter)
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB64BE@mbx-01.sysconfig.co.uk>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB555D@mbx-01.sysconfig.co.uk>
	<6FD2F8B5BB72834E9939AEDF9FB802A901E7BB5705@mbx-01.sysconfig.co.uk>
	<CABMX_w_qtSXYwh_U6ZXD9k43gLOH5j5hYw=yey1b8ktKo72ufw@mail.gmail.com>
	<6FD2F8B5BB72834E9939AEDF9FB802A901E7BB64BE@mbx-01.sysconfig.co.uk>
Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB6678@mbx-01.sysconfig.co.uk>

I may have come across another solution for call recording on demand when using mod_cellcenter. I?ve tested using the uuid_record API command to stop a Call Center Queue recording and this works.

When using uuid_record I?m not sure if it?s not possible to pause and resume, however it is possible to mask and unmask the audio with silence.

We?re currently testing this further to see if we can make this work in our environment but thus far this looks promising.

Shaun

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes
Sent: 25 April 2016 09:11
To: 'FreeSWITCH Users Help' <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Call Recording On Demand for Call Center Queues (mod_callcenter)

Hi ?talo,

Unfortunately execute_on_answer doesn?t have any effect if we use this in cc_export_vars or in the agents contact field, with or with-out the nolocal: prefix. We have replaced our variable ${rec_file} with a fixed recording file path for testing purposes so we could rule that out.

Here?s a list of variables we?ve tried in cc_export_vars and the agents contact field for testing purposes:
execute_on_answer=nolocal:record_session /usr/local/freeswitch/recordings/test.mp3
execute_on_answer=record_session /usr/local/freeswitch/recordings/test.mp3
execute_on_answer='record_session /usr/local/freeswitch/recordings/test.mp3'
api_on_answer=nolocal:uuid_record ${uuid} start /usr/local/freeswitch/recordings/test.mp3
api_on_answer=uuid_record ${uuid} start /usr/local/freeswitch/recordings/test.mp3
api_on_answer='uuid_record ${uuid} start /usr/local/freeswitch/recordings/test.mp3'

If you or anyone else here has any other ideas that would be a great help to us.

Many Thanks,
Shaun

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ?talo Rossi
Sent: 22 April 2016 20:47
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: Re: [Freeswitch-users] Call Recording On Demand for Call Center Queues (mod_callcenter)

Use cc_export_vars https://freeswitch.org/confluence/display/FREESWITCH/mod_callcenter#mod_callcenter-cc_export_vars

Or you can set variables in the agent's contact field: {myvar=myvalue}user/number

On Fri, Apr 22, 2016 at 12:25 PM, Shaun Stokes <shaun.stokes at itec-support.co.uk<mailto:shaun.stokes at itec-support.co.uk>> wrote:
We have been troubleshooting this further, it seems that this line has no effect when using mod_callcenter:
<action application="export" data="nolocal:execute_on_answer=record_session ${rec_file}"/>

This parameter is set in callcenter.conf if we remove this the call doesn?t record:
<param name="record-template" value="=$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}"/>

The file path\name from the above matches that of the call we?re trying to pause, but it seems we can?t pause the recording if this has been started by mod_callcenter, there are no errors. We can only start and pause call recording when this isn?t started using mod_callcenter by default.

Is there any way around this, maybe a different command for mod_callcenter that can be executed on the agent leg using dtmf to stop the recording, or another way we can start recording after being answered by the agent?

Thanks,
Shaun


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Shaun Stokes
Sent: 22 April 2016 14:56
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] Call Recording On Demand for Call Center Queues (mod_callcenter)

Hi,

We?re trying to setup call recording on demand for call center queues (mod_callcenter) but are experiencing tremendous trouble. Is it possible for the agent to perform pause\resume call recording on mod_callcenter calls which are recorded by default upon answer?

Here?s the important elements of our dialplan:
<condition field="${rec_file}" expression="^(.)+">
                <action application="set" inline="true" data="RECORD_APPEND=true"/>
                <anti-action application="set" inline="true" data="RECORD_APPEND=true"/>
                <anti-action application="export" inline="true" data="rec_file=$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}"/>
</condition>
<condition field="destination_number" expression="^SETUP_RECORDING$">
                <action application="log" data="INFO Configuring bind_digit_action to do recording on this session..."/>
                <action application="bind_digit_action" data="features,*1,exec:execute_extension,START_RECORDING XML ${domain_name}"/>
                <action application="bind_digit_action" data="features,*0,exec:execute_extension,STOP_RECORDING XML ${domain_name}"/>
                <action application="digit_action_set_realm" data=" features"/>
</condition>
<condition field="destination_number" expression="^START_RECORDING$">
                <action application="log" data="INFO Starting recording..."/>
                <action application="record_session" data="${rec_file}"/>
</condition>
<condition field="destination_number" expression="^STOP_RECORDING$">
                <action application="log" data="INFO Stopping recording..."/>
                <action application="stop_record_session" data="${rec_file}"/>
</condition>
<condition field="destination_number" expression="^837$">
<action application="export" data="nolocal:execute_on_answer=record_session ${rec_file}"/>
<action application="set" data="cc_export_vars=rec_file"/>
<action application="set" data="bridge_pre_execute_bleg_app=execute_extension"/>
<action application="set" data="bridge_pre_execute_bleg_data=SETUP_RECORDING XML ${domain_name}"/>
<action application="callcenter" data="Test-bench at tester"/>
</condition>

Here?s the log of the call as it comes in:
SET [RECORD_APPEND]=[true]
Configuring bind_digit_action to do recording on this session...
Digit parser DPTOOLS: binding *1/features/0 callback: 0x7f37720eed10 data: 0x7f367c7dd918
Digit parser DPTOOLS: binding *0/features/0 callback: 0x7f37720eed10 data: 0x7f367c7ddab8
Digit parser DPTOOLS: Setting realm to 'features'
EXPORT (export_vars) (REMOTE ONLY) [execute_on_answer]=[record_session /usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3]

The call is then answered by the agent, we see the recording file has been created, the agent presses *0 but the call continues recording, log as follows:
SET [RECORD_APPEND]=[true]
RTP RECV DTMF *:1536
RTP RECV DTMF 0:1536
Stopping recording...
stop_record_session(/usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3)


The configuration looks as though it should work, any ideas? Is there another method we should be using?

Many Thanks,
Shaun

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



--
?talo Rossi
italo at freeswitch.org<mailto:italo at freeswitch.org>

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

______________________________________________________________________
This message has been checked for all known viruses by MessageLabs Virus Scanning Service.
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From jack at livematch.com  Thu Apr 21 03:25:48 2016
From: jack at livematch.com (Jack Loranger)
Date: Wed, 20 Apr 2016 16:25:48 -0700
Subject: [Freeswitch-users] Verto Status Display
In-Reply-To: <CABMX_w-z5AfKnHZQc7J8GLscD3TXrEgcy=S_TcPb2YPeb9EOYA@mail.gmail.com>
References: <571479D2.8050407@livematch.com>
	<CABMX_w-z5AfKnHZQc7J8GLscD3TXrEgcy=S_TcPb2YPeb9EOYA@mail.gmail.com>
Message-ID: <57180FFC.1060509@livematch.com>

Thank you for your reply ?talo,
  The docs show that the livearray-json-status flag gives the detailed 
display.  I do not see where it is set.
These are the flags that are set in conference.conf.xml :
<param name="conference-flags" 
value="video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding"/>

I can remove livearray-sync and get no status display, but I can't 
figure out how to get the shorter display that I find in the 
conference_members.c  function: conference_member_update_status_field

Thanks again,
  Jack Loranger

On 4/20/2016 5:38 AM, ?talo Rossi wrote:
> Remove the flag on the conference profile, conference.conf.xml
>
> On Mon, Apr 18, 2016 at 3:08 AM, Jack <jack at livematch.com 
> <mailto:jack at livematch.com>> wrote:
>
>     I am trying to show the shorter display for STATUS in the
>     index.html  in
>     the video_demo. So it shows like the Freeswitch conference status.  It
>     seems like the  CFLAG_JSON_STATUS is set to true to display the
>     detailed
>     status.  Can someone tell me how to set this flag to false or where it
>     is being set to true?
>     Thanks,
>       Jack Loranger
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> ?talo Rossi
> italo at freeswitch.org <mailto:italo at freeswitch.org>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2016.0.7539 / Virus Database: 4556/12065 - Release Date: 04/19/16
>

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From fs at hellea.net  Fri Apr 22 10:07:29 2016
From: fs at hellea.net (fs at hellea.net)
Date: Fri, 22 Apr 2016 08:07:29 +0200
Subject: [Freeswitch-users] switch_rtp not changing IP with SRTP
Message-ID: <20160422080729.55541f06@email.my.domain>

Hi,

when calling to a linphone on a VPN without SRTP enabled on the leg-B,
FS does

[INFO] switch_rtp.c:6411 Auto Changing audio port from
192.168.1.35:7076 to 172.16.23.207:7076

But it doesn't when SRTP is enabled on the leg-B.

How to force that behaviour when SRTP is enabled?

Kind regards,
JCh


From jack at livematch.com  Sat Apr 23 06:34:15 2016
From: jack at livematch.com (Jack Loranger)
Date: Fri, 22 Apr 2016 19:34:15 -0700
Subject: [Freeswitch-users] Verto
In-Reply-To: <CAC7ttFEf5w6D-qGTtwaARvr9Dhg-m=QN+1LtvRQB4kzRT_FXWA@mail.gmail.com>
References: <CAC7ttFE6C_CDkV79vOcgV9YKw+sJALT7bcGf_5RJ5KO8EcXb4A@mail.gmail.com>
	<CABMX_w9vZCmqdkgAZ00FB1WZqT=hRK5hvtHF0tmdsGPWmAjZFQ@mail.gmail.com>
	<CAC7ttFGGfpAdcbRfpec3-FRbunRBVGOpvJOgcPQ=Siycsem-gQ@mail.gmail.com>
	<CAC7ttFEf5w6D-qGTtwaARvr9Dhg-m=QN+1LtvRQB4kzRT_FXWA@mail.gmail.com>
Message-ID: <571ADF27.5030109@livematch.com>

Gregor,
   I am experiencing the same thing with the sinkid.
Do you mind being specific about where you changed
I set useSpeak: true, insted objectId of device :-((
Does this mean you had to use a specific objectId or you ended up using 
useSpeak:true ?
and where you added  tag:'webcam' in verto initialization.
Thanks,
  Jack

On 4/22/2016 7:50 AM, Gregor Nanger wrote:
> Justo for info. I managed to solve problem.
>
> This error:
> Cannot read property 'sinkId' of undefined.
>
> is caused if developer make mistake :-)))
>
> I set useSpeak: true, insted objectId of device :-((
>
> But I found that I didn't know and maybe it will help to someone.
>
> Even if you want to use verto only for audio calls, there still needs 
> to be set video object (can be hidden) in html. Otherwise incoming 
> audio is not working.
>
> I set: <video width=800 id="webcam" autoplay="autoplay" 
> hidden="true"></video>
> and add tag:'webcam' in verto initialization.
>
> Now everything works as expected, except 2 sec delay in audio stream 
> from browser to endpoint.
>
> 2016-04-22 0:31 GMT+02:00 Gregor Nanger <gregor at infomedia.si 
> <mailto:gregor at infomedia.si>>:
>
>     Thank you for answer.
>
>     I think that something is conflicting with other javascript
>     library. I am integrating verto in other solution (angularjs and
>     bunch of other javascripts). Demo is also working ok in my
>     environment.
>
>     How can I be sure that I am using latest lib? Should I just
>     download latest branch?
>
>
>
>
>
>
>
>     2016-04-21 19:59 GMT+02:00 ?talo Rossi <italo at freeswitch.org
>     <mailto:italo at freeswitch.org>>:
>
>         Hey Gregor,
>
>         Can you double check if you're using the most recent version
>         of verto js lib?
>
>         On Thu, Apr 21, 2016 at 4:49 AM, Gregor Nanger
>         <gregor at infomedia.si <mailto:gregor at infomedia.si>> wrote:
>
>             ?I am building my web phone with verto from scratch and
>             using this tutorial:
>
>             http://evoluxbr.github.io/verto-docs/
>
>             BAsically everything is ok, I can register and make calls,
>             but I cannot hear audio. Mic is working, but not speakers.
>             I see error:
>
>             Cannot read property 'sinkId' of undefined. This is error
>             line:
>
>             $.verto.dialog.prototype.setAudioPlaybackDevice =
>             function(sinkId, callback, arg) {
>             var dialog = this;
>             var element = dialog.audioStream;
>
>             *if (typeof element.sinkId !== 'undefined') {*
>             *
>             *
>             I get dialog object without audioStream property and hence
>             element is null.
>
>             I am using self signed certificate. Could this be cause?
>
>             Best regards, Gregor
>             ?
>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>             http://www.freeswitchsolutions.com
>
>             Official FreeSWITCH Sites
>             http://www.freeswitch.org
>             http://confluence.freeswitch.org
>             http://www.cluecon.com
>
>             FreeSWITCH-users mailing list
>             FreeSWITCH-users at lists.freeswitch.org
>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>             http://www.freeswitch.org
>
>
>
>
>         -- 
>         ?talo Rossi
>         italo at freeswitch.org <mailto:italo at freeswitch.org>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://confluence.freeswitch.org
>         http://www.cluecon.com
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
>     -- 
>     Gregor Nanger
>     	*CTO*
>     t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
>     ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
>     ? www.infomedia.si <http://www.infomedia.si>
>
>
>
>
> -- 
> Gregor Nanger
> 	*CTO*
> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
> ? www.infomedia.si <http://www.infomedia.si>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2016.0.7539 / Virus Database: 4556/12081 - Release Date: 04/22/16
>

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From jack at livematch.com  Sat Apr 23 06:43:26 2016
From: jack at livematch.com (Jack Loranger)
Date: Fri, 22 Apr 2016 19:43:26 -0700
Subject: [Freeswitch-users] Verto Status Display
In-Reply-To: <CABMX_w-z5AfKnHZQc7J8GLscD3TXrEgcy=S_TcPb2YPeb9EOYA@mail.gmail.com>
References: <571479D2.8050407@livematch.com>
	<CABMX_w-z5AfKnHZQc7J8GLscD3TXrEgcy=S_TcPb2YPeb9EOYA@mail.gmail.com>
Message-ID: <571AE14E.4090201@livematch.com>

This is where I am confused..   here are the flags I have set :
<param name="conference-flags" 
value="video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding"/>
The docs say that it is the livearray-json-status flag that shows the 
detailed display.  I don't see it being set anywhere.
Thanks,
  Jack

On 4/20/2016 5:38 AM, ?talo Rossi wrote:
> Remove the flag on the conference profile, conference.conf.xml
>
> On Mon, Apr 18, 2016 at 3:08 AM, Jack <jack at livematch.com 
> <mailto:jack at livematch.com>> wrote:
>
>     I am trying to show the shorter display for STATUS in the
>     index.html  in
>     the video_demo. So it shows like the Freeswitch conference status.  It
>     seems like the  CFLAG_JSON_STATUS is set to true to display the
>     detailed
>     status.  Can someone tell me how to set this flag to false or where it
>     is being set to true?
>     Thanks,
>       Jack Loranger
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> ?talo Rossi
> italo at freeswitch.org <mailto:italo at freeswitch.org>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2016.0.7539 / Virus Database: 4556/12065 - Release Date: 04/19/16
>

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From jack at livematch.com  Sat Apr 23 23:02:38 2016
From: jack at livematch.com (Jack Loranger)
Date: Sat, 23 Apr 2016 12:02:38 -0700
Subject: [Freeswitch-users] Verto Caller Id and Caller name
In-Reply-To: <CAC7ttFHLXx0uLr0qM7eMp-+b3eHevnARtEbSpG9vfb4qN807zQ@mail.gmail.com>
References: <CAC7ttFFoTN90ADHyTzLYe7i2UotgC1dwLgcSSyDtt2+X_6zptg@mail.gmail.com>
	<CAEwTsAiDP1-7MLVxDcSam6kqkK9rBp0utCd3q69ruVHqGkW=uA@mail.gmail.com>
	<CAC7ttFHLXx0uLr0qM7eMp-+b3eHevnARtEbSpG9vfb4qN807zQ@mail.gmail.com>
Message-ID: <571BC6CE.80903@livematch.com>

Hi Gregor,
Here is the way I create the directory page to return.
The CallerExt. vairiables are being filled from a database query.
The queryString[] variables come from the Freeswitch post to curl

  string userDirectory = "<?xml version=\"1.0\" encoding=\"UTF-8\" 
standalone=\"no\"?>\r\n" +
                 "<document type=\"freeswitch/xml\">\r\n" +
                 "  <section name=\"directory\" description=\"Directory 
For FreeSwitch\">\r\n" +
                 "    <domain name=\"" + queryString["domain"] + "\">\r\n" +
                 "      <user id=\"" + queryString["user"] + "\">\r\n" +
                 "        <params>\r\n" +
                 "          <param name=\"password\" value=\"" + 
CallerExt.Password + "\"/>\r\n" +
                 "           <param name=\"vm-password\" value=\"" + 
CallerExt.Password + "\"/>\r\n" +
                 "           <param name=\"dial-string\" 
value=\"{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}\"/>\r\n" 
+
                         // "           <param name=\"dial-string\" 
value=\"${sofia_contact(${dialed_user}@${dialed_domain})}\"/>\r\n" +
                 "        </params>\r\n" +
                 "        <variables>\r\n" +
                 "          <variable name=\"ani\" value=\"" + 
CallerExt.Number2 + "\"/>\r\n" +
                 "         <variable name=\"user_context\" 
value=\"default\"/>\r\n" +
                 "         <variable name=\"accountcode\" value=\"" + 
queryString["user"] + "\"/>\r\n" +
                 "         <variable name=\"toll_allow\" 
value=\"domestic,local\"/>\r\n" +
                 "         <variable name=\"effective_caller_id_name\" 
value=\"" + queryString["user"] + "\"/>\r\n" +
                 "         <variable name=\"effective_caller_id_number\" 
value=\"" + queryString["user"] + "\"/>\r\n" +
                 "         <variable name=\"outbound_caller_id_name\" 
value=\"$${outbound_caller_name}\"/>\r\n" +
                 "         <variable name=\"outbound_caller_id_number\" 
value=\"$${outbound_caller_id}\"/>\r\n" +
                 "        </variables>\r\n" +
                 "      </user>\r\n" +
                 "    </domain>\r\n" +
                 "  </section>\r\n" +
                 "</document>\r\n";

Hope that helps,
Jack


On 4/23/2016 3:58 AM, Gregor Nanger wrote:
> I am using xml_curl for registration and tried to set variables, but 
> without any succes.
>
> Can you please give me example.
>
> On Sat, Apr 23, 2016, 07:21 Michael Jerris <mike at jerris.com 
> <mailto:mike at jerris.com>> wrote:
>
>     you can set variables from the reg
>
>     On Friday, April 22, 2016, Gregor Nanger <gregor at infomedia.si
>     <mailto:gregor at infomedia.si>> wrote:
>
>         I can set in verto javascript initialization caller id and
>         caller name. But can I force or override caller id and name in
>         user registration variables or somewhere else, because I do
>         not want to set caller name in javascript, since user can
>         tweak it..
>
>         Any experience?
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://confluence.freeswitch.org
>     http://www.cluecon.com
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
> -- 
> Gregor Nanger
> 	*CTO*
> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
> ? www.infomedia.si <http://www.infomedia.si/>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2016.0.7539 / Virus Database: 4556/12084 - Release Date: 04/22/16
>

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From gregor at infomedia.si  Mon Apr 25 17:59:04 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Mon, 25 Apr 2016 15:59:04 +0200
Subject: [Freeswitch-users] Verto
In-Reply-To: <571ADF27.5030109@livematch.com>
References: <CAC7ttFE6C_CDkV79vOcgV9YKw+sJALT7bcGf_5RJ5KO8EcXb4A@mail.gmail.com>
	<CABMX_w9vZCmqdkgAZ00FB1WZqT=hRK5hvtHF0tmdsGPWmAjZFQ@mail.gmail.com>
	<CAC7ttFGGfpAdcbRfpec3-FRbunRBVGOpvJOgcPQ=Siycsem-gQ@mail.gmail.com>
	<CAC7ttFEf5w6D-qGTtwaARvr9Dhg-m=QN+1LtvRQB4kzRT_FXWA@mail.gmail.com>
	<571ADF27.5030109@livematch.com>
Message-ID: <CAC7ttFH6uAnzoi-X7hzdFiT8XmRxagFmguiw=NmV=GeMXtmkrQ@mail.gmail.com>

Yes, if you want to use specific speaker than you need to supply objectid.
But I just remove useSpeak, because verto than use Any. I think that then
default audio device in system is selected.

Regarding tag. Here is initialization code:
new jQuery.verto({
                    login: data.login
                    passwd: data.password,
                    socketUrl: data.wsURL,
                    ringFile: "/sounds/bell_ring2.wav",
                    tag: "webcam",
...........

and you need to add video element in html of page with id="webcam"

Best regards, Gregor




2016-04-23 4:34 GMT+02:00 Jack Loranger <jack at livematch.com>:

> Gregor,
>   I am experiencing the same thing with the sinkid.
> Do you mind being specific about where you changed
> I set useSpeak: true, insted objectId of device :-((
> Does this mean you had to use a specific objectId or you ended up using
> useSpeak:true ?
> and where you added  tag:'webcam' in verto initialization.
> Thanks,
>  Jack
>
>
> On 4/22/2016 7:50 AM, Gregor Nanger wrote:
>
> Justo for info. I managed to solve problem.
>
> This error:
> Cannot read property 'sinkId' of undefined.
>
> is caused if developer make mistake :-)))
>
> I set useSpeak: true, insted objectId of device :-((
>
> But I found that I didn't know and maybe it will help to someone.
>
> Even if you want to use verto only for audio calls, there still needs to
> be set video object (can be hidden) in html. Otherwise incoming audio is
> not working.
>
> I set: <video width=800 id="webcam" autoplay="autoplay"
> hidden="true"></video>
> and add tag:'webcam' in verto initialization.
>
> Now everything works as expected, except 2 sec delay in audio stream from
> browser to endpoint.
>
> 2016-04-22 0:31 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:
>
>> Thank you for answer.
>>
>> I think that something is conflicting with other javascript library. I am
>> integrating verto in other solution (angularjs and bunch of other
>> javascripts). Demo is also working ok in my environment.
>>
>> How can I be sure that I am using latest lib? Should I just download
>> latest branch?
>>
>>
>>
>>
>>
>>
>>
>> 2016-04-21 19:59 GMT+02:00 ?talo Rossi < <italo at freeswitch.org>
>> italo at freeswitch.org>:
>>
>>> Hey Gregor,
>>>
>>> Can you double check if you're using the most recent version of verto js
>>> lib?
>>>
>>> On Thu, Apr 21, 2016 at 4:49 AM, Gregor Nanger < <gregor at infomedia.si>
>>> gregor at infomedia.si> wrote:
>>>
>>>> ?I am building my web phone with verto from scratch and using this
>>>> tutorial:
>>>>
>>>>
>>>> <http://t.sidekickopen04.com/e1t/c/5/f18dQhb0S7lC8dDMPbW2n0x6l2B9nMJN7t5XZsfmT10W5wLWvl4WYKY2W3MhGXn56dLgFf1Tdg-W02?t=http%3A%2F%2Fevoluxbr.github.io%2Fverto-docs%2F&si=5036397111279616&pi=4ee20dcc-e112-4624-b439-d58b5ba336b7>
>>>> http://evoluxbr.github.io/verto-docs/
>>>>
>>>> BAsically everything is ok, I can register and make calls, but I cannot
>>>> hear audio. Mic is working, but not speakers. I see error:
>>>>
>>>> Cannot read property 'sinkId' of undefined. This is error line:
>>>>
>>>> $.verto.dialog.prototype.setAudioPlaybackDevice = function(sinkId,
>>>> callback, arg) {
>>>> var dialog = this;
>>>> var element = dialog.audioStream;
>>>>
>>>> * if (typeof element.sinkId !== 'undefined') {*
>>>>
>>>> I get dialog object without audioStream property and hence element is
>>>> null.
>>>>
>>>> I am using self signed certificate. Could this be cause?
>>>>
>>>> Best regards, Gregor
>>>> ?
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> ?talo Rossi
>>> italo at freeswitch.org
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> 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
>>
>
>
>
> --
> 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.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
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 2016.0.7539 / Virus Database: 4556/12081 - Release Date: 04/22/16
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
? www.infomedia.si
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From gregor at infomedia.si  Mon Apr 25 18:07:44 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Mon, 25 Apr 2016 16:07:44 +0200
Subject: [Freeswitch-users] Verto registrations
Message-ID: <CAC7ttFHVRcuS_PuDi0a52AGBQgg4phxYsVjXR1D1tidMTx+LPg@mail.gmail.com>

Is there a way to flush specific verto registration.

When I show status: verto status, somehow one registration is hanging.

Can I flush specific verto registration from cli or ESL? When does session
expire?
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From veerabhadrarao.kankatala at panamaxil.com  Mon Apr 25 18:16:13 2016
From: veerabhadrarao.kankatala at panamaxil.com (Veerabhadrarao Kankatala)
Date: Mon, 25 Apr 2016 10:16:13 -0400 (EDT)
Subject: [Freeswitch-users] Freeswitch-1.6.5 (video Conference)
Message-ID: <1204632313.104909440.1461593773928.JavaMail.zimbra@panamaxil.com>

Hello, 

-- I am currently working on video conference module in freeswitch-1.6.5 

-- I am using verto client for confernece (i configured just 3 verto users in different hosts) and when i dial 3500 extension from the verto user, the conference is starting but the main problem is 

if 3 users connected with the 3500 conference, the audio is gone(too much noice, no audio clearence) and CPU usage is reached to 150%+ 

and i follow below configuration also 
ulimit -c unlimited # The maximum size of core files created.
ulimit -d unlimited # The maximum size of a process's data segment.
ulimit -f unlimited # The maximum size of files created by the shell (default option)
ulimit -i unlimited # The maximum number of pending signals
ulimit -n 999999    # The maximum number of open file descriptors.
ulimit -q unlimited # The maximum POSIX message queue size
ulimit -u unlimited # The maximum number of processes available to a single user.
ulimit -v unlimited # The maximum amount of virtual memory available to the process.
ulimit -x unlimited #????
ulimit -s 240         # The maximum stack size
ulimit -l unlimited # The maximum size that may be locked into memory.
ulimit -a           # All current limits are reported. 

i am using using Centos 6.5 with 4GB Ram. Could you any one help me to resolve this problem, 

i want to know following thigs 

--> Video confernece configuration 
--> Required system configuration 
--> and also i would like to know how many possible video conferences can be handle by freeswitch-1.6.5 


Please help me. 

-- 
Thanks & Regards 
Veerabhadrarao Kankatala 

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From krice at freeswitch.org  Mon Apr 25 18:24:36 2016
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 25 Apr 2016 09:24:36 -0500
Subject: [Freeswitch-users] Freeswitch-1.6.5 (video Conference)
In-Reply-To: <1204632313.104909440.1461593773928.JavaMail.zimbra@panamaxil.com>
References: <1204632313.104909440.1461593773928.JavaMail.zimbra@panamaxil.com>
Message-ID: <104201d19efe$32e2a110$98a7e330$@freeswitch.org>

Sounds like you just aren?t giving the box enough CPU? FreeSWITCH with Video Conferencing requires a fair bit of CPU. Consider that you are decoding a video stream for each participant to raw video, then potentially scaling and compositing those videos together into 1 video, from there you then have to re-encode the video into the appropriate video codec for the participants and stream it out to them.

 

Doing Audio only conferencing (even with G729 which still requires a transcode for raw slin audio for mixing) is much less CPU intensive. 

 

We routinely do 30 to 50 people video conferences with FreeSWITCH. To accomplish this, we use a 24 thread (dual hexacore Xeon with Hyperthreading enabled) with 48G of Ram. (no we don?t need that much ram it?s just what the box came with and has come in handy during testing)

 

That being said, your individual mileage will vary. The biggest mistake I?ve seen people make is grossly underestimate the CPU resources required to handle the video in real time. (consider how much time it takes even on something as powerful as a i7-4770K when re-encoding your own videos using something like Premiere or other video editing software)

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Veerabhadrarao Kankatala
Sent: Monday, April 25, 2016 9:16 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch-1.6.5 (video Conference)

 

Hello,

 

-- I am currently working on video conference module in freeswitch-1.6.5

 

-- I am using verto client for confernece (i configured just 3 verto users in different hosts) and when i dial 3500 extension from the verto user, the conference is starting but the main problem is 

 

if 3 users connected with the 3500 conference, the audio is gone(too much noice, no audio clearence) and CPU usage is reached to 150%+





and i follow below configuration also

ulimit -c unlimited # The maximum size of core files created.
ulimit -d unlimited # The maximum size of a process's data segment.
ulimit -f unlimited # The maximum size of files created by the shell (default option)
ulimit -i unlimited # The maximum number of pending signals
ulimit -n 999999    # The maximum number of open file descriptors.
ulimit -q unlimited # The maximum POSIX message queue size
ulimit -u unlimited # The maximum number of processes available to a single user.
ulimit -v unlimited # The maximum amount of virtual memory available to the process.
ulimit -x unlimited # ???
ulimit -s 240         # The maximum stack size
ulimit -l unlimited # The maximum size that may be locked into memory.
ulimit -a           # All current limits are reported.





i am using using  Centos 6.5 with 4GB Ram. Could you any one help me to resolve this problem,

 

i want to know following thigs

 

--> Video confernece configuration 

--> Required system configuration

--> and also i would like to know how many possible video conferences can be handle by freeswitch-1.6.5

 

 

Please help me.

 

-- 

Thanks & Regards
Veerabhadrarao Kankatala

 

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From gmaruzz at gmail.com  Mon Apr 25 18:24:21 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 25 Apr 2016 16:24:21 +0200
Subject: [Freeswitch-users] Freeswitch-1.6.5 (video Conference)
In-Reply-To: <1204632313.104909440.1461593773928.JavaMail.zimbra@panamaxil.com>
References: <1204632313.104909440.1461593773928.JavaMail.zimbra@panamaxil.com>
Message-ID: <CALXCt0qLoQq+-Sb6XZbsWm00o6n3Wr+QLGpS-FW02E+o9JqvfQ@mail.gmail.com>

Use Debian 8 64 bit server (Jessie), and latest FreeSWITCH, as directed by
Confluence, and all documentation.

If you use anything else, you will have to find your way by yourself, and
you will waste a lot of time, and at the end you will go to Debian 8.

-giovanni

On Mon, Apr 25, 2016 at 4:16 PM, Veerabhadrarao Kankatala <
veerabhadrarao.kankatala at panamaxil.com> wrote:

> Hello,
>
> -- I am currently working on video conference module in freeswitch-1.6.5
>
> -- I am using verto client for confernece (i configured just 3 verto users
> in different hosts) and when i dial 3500 extension from the verto user, the
> conference is starting but the main problem is
>
> *if 3 users connected with the 3500 conference, the audio is gone(too much
> noice, no audio clearence) and CPU usage is reached to 150%+*
>
> *and i follow below configuration also*
>
> ulimit -c unlimited # The maximum size of core files created.
> ulimit -d unlimited # The maximum size of a process's data segment.
> ulimit -f unlimited # The maximum size of files created by the shell (default option)
> ulimit -i unlimited # The maximum number of pending signals
> ulimit -n 999999    # The maximum number of open file descriptors.
> ulimit -q unlimited # The maximum POSIX message queue size
> ulimit -u unlimited # The maximum number of processes available to a single user.
> ulimit -v unlimited # The maximum amount of virtual memory available to the process.
> ulimit -x unlimited # ???
> ulimit -s 240         # The maximum stack size
> ulimit -l unlimited # The maximum size that may be locked into memory.
> ulimit -a           # All current limits are reported.
>
>
> i am using using  Centos 6.5 with 4GB Ram. Could you any one help me to
> resolve this problem,
>
> i want to know following thigs
>
> --> Video confernece configuration
> --> Required system configuration
> --> and also i would like to know how many possible video conferences can
> be handle by freeswitch-1.6.5
>
>
> Please help me.
>
> --
> Thanks & Regards
> *Veerabhadrarao Kankatala*
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From arsenman at connectto.com  Mon Apr 25 22:14:26 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Mon, 25 Apr 2016 11:14:26 -0700
Subject: [Freeswitch-users] How to change STUN an TURN servers in verto
	communicator ?
In-Reply-To: <CAC7ttFHVRcuS_PuDi0a52AGBQgg4phxYsVjXR1D1tidMTx+LPg@mail.gmail.com>
References: <CAC7ttFHVRcuS_PuDi0a52AGBQgg4phxYsVjXR1D1tidMTx+LPg@mail.gmail.com>
Message-ID: <ef8a937c-5e9c-8871-2bf8-3a0bde92f41b@connectto.com>

I have my own TURN and STUN servers

How to change STUN an TURN servers in verto communicator ?


From anthony.minessale at gmail.com  Mon Apr 25 23:02:37 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Apr 2016 14:02:37 -0500
Subject: [Freeswitch-users] Don't report bugs or issues here please use
	jira.freeswitch.org
Message-ID: <CAKbxfG-oCrcQSui9_vuO+y6jpCzVfqObJyOfo3KchjgJ6qsjng@mail.gmail.com>

Friendly Reminder!

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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From s.safarov at gmail.com  Mon Apr 25 23:13:50 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Mon, 25 Apr 2016 19:13:50 +0000
Subject: [Freeswitch-users] switch_rtp not changing IP with SRTP
In-Reply-To: <20160422080729.55541f06@email.my.domain>
References: <20160422080729.55541f06@email.my.domain>
Message-ID: <CAHtxdDec9nVkN0CDi9NkXshPpi=wzUcmyVvHwrGnMhMq+js8eQ@mail.gmail.com>

Please check that you use UDP transport.
Linphone has issue in in NAT traversal when used TCP or TLS transport.


??, 25 ???. 2016 ?. ? 16:26, <fs at hellea.net>:

> Hi,
>
> when calling to a linphone on a VPN without SRTP enabled on the leg-B,
> FS does
>
> [INFO] switch_rtp.c:6411 Auto Changing audio port from
> 192.168.1.35:7076 to 172.16.23.207:7076
>
> But it doesn't when SRTP is enabled on the leg-B.
>
> How to force that behaviour when SRTP is enabled?
>
> Kind regards,
> JCh
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 25 23:50:51 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Apr 2016 15:50:51 -0400
Subject: [Freeswitch-users] Freeswitch-1.6.5 (video Conference)
In-Reply-To: <104201d19efe$32e2a110$98a7e330$@freeswitch.org>
References: <1204632313.104909440.1461593773928.JavaMail.zimbra@panamaxil.com>
	<104201d19efe$32e2a110$98a7e330$@freeswitch.org>
Message-ID: <CAEwTsAiUi7sp_hgxpSWGrEGkwzJbH_hPN=DRb8h6y0irj6dLqA@mail.gmail.com>

we've also fixed some performance issues in later released since 1.6.5

On Monday, April 25, 2016, Ken Rice <krice at freeswitch.org> wrote:

> Sounds like you just aren?t giving the box enough CPU? FreeSWITCH with
> Video Conferencing requires a fair bit of CPU. Consider that you are
> decoding a video stream for each participant to raw video, then potentially
> scaling and compositing those videos together into 1 video, from there you
> then have to re-encode the video into the appropriate video codec for the
> participants and stream it out to them.
>
>
>
> Doing Audio only conferencing (even with G729 which still requires a
> transcode for raw slin audio for mixing) is much less CPU intensive.
>
>
>
> We routinely do 30 to 50 people video conferences with FreeSWITCH. To
> accomplish this, we use a 24 thread (dual hexacore Xeon with Hyperthreading
> enabled) with 48G of Ram. (no we don?t need that much ram it?s just what
> the box came with and has come in handy during testing)
>
>
>
> That being said, your individual mileage will vary. The biggest mistake
> I?ve seen people make is grossly underestimate the CPU resources required
> to handle the video in real time. (consider how much time it takes even on
> something as powerful as a i7-4770K when re-encoding your own videos using
> something like Premiere or other video editing software)
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *Veerabhadrarao Kankatala
> *Sent:* Monday, April 25, 2016 9:16 AM
> *To:* freeswitch-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>
> *Subject:* [Freeswitch-users] Freeswitch-1.6.5 (video Conference)
>
>
>
> Hello,
>
>
>
> -- I am currently working on video conference module in freeswitch-1.6.5
>
>
>
> -- I am using verto client for confernece (i configured just 3 verto users
> in different hosts) and when i dial 3500 extension from the verto user, the
> conference is starting but the main problem is
>
>
>
> *if 3 users connected with the 3500 conference, the audio is gone(too much
> noice, no audio clearence) and **CPU** usage is reached to **150%+*
>
>
>
> *and i follow below configuration also*
>
> ulimit -c unlimited # The maximum size of core files created.
>
> ulimit -d unlimited # The maximum size of a process's data segment.
>
> ulimit -f unlimited # The maximum size of files created by the shell (default option)
>
> ulimit -i unlimited # The maximum number of pending signals
>
> ulimit -n 999999    # The maximum number of open file descriptors.
>
> ulimit -q unlimited # The maximum POSIX message queue size
>
> ulimit -u unlimited # The maximum number of processes available to a single user.
>
> ulimit -v unlimited # The maximum amount of virtual memory available to the process.
>
> ulimit -x unlimited # ???
>
> ulimit -s 240         # The maximum stack size
>
> ulimit -l unlimited # The maximum size that may be locked into memory.
>
> ulimit -a           # All current limits are reported.
>
>
>
> i am using using  Centos 6.5 with 4GB Ram. Could you any one help me to
> resolve this problem,
>
>
>
> i want to know following thigs
>
>
>
> --> Video confernece configuration
>
> --> Required system configuration
>
> --> and also i would like to know how many possible video conferences can
> be handle by freeswitch-1.6.5
>
>
>
>
>
> Please help me.
>
>
>
> --
>
> Thanks & Regards
> *Veerabhadrarao Kankatala*
>
>
>
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From mike at jerris.com  Mon Apr 25 23:51:57 2016
From: mike at jerris.com (Michael Jerris)
Date: Mon, 25 Apr 2016 15:51:57 -0400
Subject: [Freeswitch-users] Call Recording On Demand for Call Center
 Queues (mod_callcenter)
In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB64BE@mbx-01.sysconfig.co.uk>
References: <6FD2F8B5BB72834E9939AEDF9FB802A901E7BB555D@mbx-01.sysconfig.co.uk>
	<6FD2F8B5BB72834E9939AEDF9FB802A901E7BB5705@mbx-01.sysconfig.co.uk>
	<CABMX_w_qtSXYwh_U6ZXD9k43gLOH5j5hYw=yey1b8ktKo72ufw@mail.gmail.com>
	<6FD2F8B5BB72834E9939AEDF9FB802A901E7BB64BE@mbx-01.sysconfig.co.uk>
Message-ID: <CAEwTsAhqrKGktsrSfnoULxyPN1wuatAgqj9dOLHsux=mx3hLbQ@mail.gmail.com>

it wouldn't because the calls are already answered

On Monday, April 25, 2016, Shaun Stokes <shaun.stokes at itec-support.co.uk>
wrote:

> Hi ?talo,
>
>
>
> Unfortunately execute_on_answer doesn?t have any effect if we use this in cc_export_vars or
> in the agents contact field, with or with-out the nolocal: prefix. We have
> replaced our variable ${rec_file} with a fixed recording file path for
> testing purposes so we could rule that out.
>
>
>
> Here?s a list of variables we?ve tried in cc_export_vars and the agents
> contact field for testing purposes:
>
> execute_on_answer=nolocal:record_session
> /usr/local/freeswitch/recordings/test.mp3
>
> execute_on_answer=record_session /usr/local/freeswitch/recordings/test.mp3
>
> execute_on_answer='record_session
> /usr/local/freeswitch/recordings/test.mp3'
>
> api_on_answer=nolocal:uuid_record ${uuid} start
> /usr/local/freeswitch/recordings/test.mp3
>
> api_on_answer=uuid_record ${uuid} start
> /usr/local/freeswitch/recordings/test.mp3
>
> api_on_answer='uuid_record ${uuid} start
> /usr/local/freeswitch/recordings/test.mp3'
>
>
>
> If you or anyone else here has any other ideas that would be a great help
> to us.
>
>
>
> Many Thanks,
>
> Shaun
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *?talo Rossi
> *Sent:* 22 April 2016 20:47
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>>
> *Subject:* Re: [Freeswitch-users] Call Recording On Demand for Call
> Center Queues (mod_callcenter)
>
>
>
> Use cc_export_vars
> https://freeswitch.org/confluence/display/FREESWITCH/mod_callcenter#mod_callcenter-cc_export_vars
>
>
>
> Or you can set variables in the agent's contact field:
> {myvar=myvalue}user/number
>
>
>
> On Fri, Apr 22, 2016 at 12:25 PM, Shaun Stokes <
> shaun.stokes at itec-support.co.uk
> <javascript:_e(%7B%7D,'cvml','shaun.stokes at itec-support.co.uk');>> wrote:
>
> We have been troubleshooting this further, it seems that this line has no
> effect when using mod_callcenter:
>
> <action application="export"
> data="nolocal:execute_on_answer=record_session ${rec_file}"/>
>
>
>
> This parameter is set in callcenter.conf if we remove this the call
> doesn?t record:
>
> <param name="record-template"
> value="=$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}"/>
>
>
>
> The file path\name from the above matches that of the call we?re trying to
> pause, but it seems we can?t pause the recording if this has been started
> by mod_callcenter, there are no errors. We can only start and pause call
> recording when this isn?t started using mod_callcenter by default.
>
>
>
> Is there any way around this, maybe a different command for mod_callcenter
> that can be executed on the agent leg using dtmf to stop the recording, or
> another way we can start recording after being answered by the agent?
>
>
>
> Thanks,
>
> Shaun
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *Shaun Stokes
> *Sent:* 22 April 2016 14:56
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>>
> *Subject:* [Freeswitch-users] Call Recording On Demand for Call Center
> Queues (mod_callcenter)
>
>
>
> Hi,
>
>
>
> We?re trying to setup call recording on demand for call center queues
> (mod_callcenter) but are experiencing tremendous trouble. Is it possible
> for the agent to perform pause\resume call recording on mod_callcenter
> calls which are recorded by default upon answer?
>
>
>
> Here?s the important elements of our dialplan:
>
> <condition field="${rec_file}" expression="^(.)+">
>
>                 <action application="set" inline="true"
> data="RECORD_APPEND=true"/>
>
>                 <anti-action application="set" inline="true"
> data="RECORD_APPEND=true"/>
>
>                 <anti-action application="export" inline="true"
> data="rec_file=$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}"/>
>
> </condition>
>
> <condition field="destination_number" expression="^SETUP_RECORDING$">
>
>                 <action application="log" data="INFO Configuring
> bind_digit_action to do recording on this session..."/>
>
>                 <action application="bind_digit_action"
> data="features,*1,exec:execute_extension,START_RECORDING XML
> ${domain_name}"/>
>
>                 <action application="bind_digit_action"
> data="features,*0,exec:execute_extension,STOP_RECORDING XML
> ${domain_name}"/>
>
>                 <action application="digit_action_set_realm" data="
> features"/>
>
> </condition>
>
> <condition field="destination_number" expression="^START_RECORDING$">
>
>                 <action application="log" data="INFO Starting
> recording..."/>
>
>                 <action application="record_session" data="${rec_file}"/>
>
> </condition>
>
> <condition field="destination_number" expression="^STOP_RECORDING$">
>
>                 <action application="log" data="INFO Stopping
> recording..."/>
>
>                 <action application="stop_record_session"
> data="${rec_file}"/>
>
> </condition>
>
> <condition field="destination_number" expression="^837$">
>
> <action application="export"
> data="nolocal:execute_on_answer=record_session ${rec_file}"/>
>
> <action application="set" data="cc_export_vars=rec_file"/>
>
> <action application="set"
> data="bridge_pre_execute_bleg_app=execute_extension"/>
>
> <action application="set"
> data="bridge_pre_execute_bleg_data=SETUP_RECORDING XML ${domain_name}"/>
>
> <action application="callcenter" data="Test-bench at tester"/>
>
> </condition>
>
>
>
> Here?s the log of the call as it comes in:
>
> SET [RECORD_APPEND]=[true]
>
> Configuring bind_digit_action to do recording on this session...
>
> Digit parser DPTOOLS: binding *1/features/0 callback: 0x7f37720eed10 data:
> 0x7f367c7dd918
>
> Digit parser DPTOOLS: binding *0/features/0 callback: 0x7f37720eed10 data:
> 0x7f367c7ddab8
>
> Digit parser DPTOOLS: Setting realm to 'features'
>
> EXPORT (export_vars) (REMOTE ONLY) [execute_on_answer]=[record_session
> /usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3]
>
>
>
> The call is then answered by the agent, we see the recording file has been
> created, the agent presses *0 but the call continues recording, log as
> follows:
>
> SET [RECORD_APPEND]=[true]
>
> RTP RECV DTMF *:1536
>
> RTP RECV DTMF 0:1536
>
> Stopping recording...
>
>
> stop_record_session(/usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3)
>
>
>
>
>
> The configuration looks as though it should work, any ideas? Is there
> another method we should be using?
>
>
>
> Many Thanks,
>
> Shaun
>
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> --
>
> ?talo Rossi
>
> italo at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','italo at freeswitch.org');>
>
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
> ______________________________________________________________________
> This message has been checked for all known viruses by MessageLabs Virus
> Scanning Service.
> ______________________________________________________________________
>
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From colin.morelli at gmail.com  Tue Apr 26 00:39:03 2016
From: colin.morelli at gmail.com (Colin Morelli)
Date: Mon, 25 Apr 2016 20:39:03 +0000
Subject: [Freeswitch-users] Using Adhearsion with mod_rayo
Message-ID: <CAPtU-UpLo+qdx7d=LzaDMFrbVgLMpg07T-+OhEVwzvJUvgyCdA@mail.gmail.com>

Hey all,

As I'm looking into better ways to remotely control FS, adhearson and moho
(over rayo) both look very powerful. Right now I'm tracking call state in
the form of URLs and custom-crafted use of httapi. It works, but it's far
from ideal.

Has anyone had experience with using adhearsion and mod_rayo? How do you
handle making sure each FS server in a cluster is actually connected to by
a rayo client? mod_httapi is nice because FS initiates the outbound call,
so as long as your service is up things should work. Rayo, on the other
hand, appears to require that your application be connected to every
running FS instance.

Would love anyones' input/experience here.

Best,
Colin
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From anthony.minessale at gmail.com  Tue Apr 26 00:40:03 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 25 Apr 2016 15:40:03 -0500
Subject: [Freeswitch-users] Verto vs SIP
In-Reply-To: <20160421080628.GA29840@blomma.liberationtech.net>
References: <20160420080841.GA23645@blomma.liberationtech.net>
	<CAJTkRNsbKOMvH_NspCrG-N_XFOZjYL1mS7-XJzYFsJdhTotoDg@mail.gmail.com>
	<20160420132442.GA23884@blomma.liberationtech.net>
	<CAAemr3wdz0VbjVBSwS7-EYB8b7bMXPLRWfPQngJVTSth9b55NA@mail.gmail.com>
	<CAKbxfG_0h7kV+x7tLMxDDkhgxvQrRwVojuUauunw1ZDr=_JQBA@mail.gmail.com>
	<20160421080628.GA29840@blomma.liberationtech.net>
Message-ID: <CAKbxfG-mzWTr8v+nVGGPxeeObhRFZzM-pWHiKgFy6Yyx4QmH8w@mail.gmail.com>

Still working on that so I don't want to start any rumors until I have it
nailed down but we will announce it when the time comes.


On Thu, Apr 21, 2016 at 3:06 AM, Oivvio Polite <mylists at polite.se> wrote:

> On ons, apr 20, 2016 at 12:38:20 -0500, Anthony Minessale wrote:
> > SIP over websockets is fine for setting up and tearing down calls but
> keep
> > in mind that is mostly the only features it will ever support.
> > The reason to consider Verto would be that it has more of an event-driven
> > HTML5 design goal meant to be extensible to UI and gateway Web with
> media.
> > I actually implemented a large portion of all of the above and I think
> > Verto is a lighter more sensible media protocol for the web since I
> > designed it that way.
>
> Thanks for the explanation.
>
> > Also, it will only progress more from here in upcoming versions of FS.
> >
>
> That's great to hear! What's on the roadmap for Verto?
>
> Oivvio
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From hkalyoncu at gmail.com  Tue Apr 26 13:21:10 2016
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Tue, 26 Apr 2016 12:21:10 +0300
Subject: [Freeswitch-users] Codec list
Message-ID: <CAPf+86UYkR_gwfxEOPBXoaFnjyOrB_8jVmVe2tuRNnDkvLG7Tg@mail.gmail.com>

Hello

I have problem with codec list which can be seen with 'show codec' cli
command.

This is what i see after fresh reboot :

freeswitch at fs1> show codec
type,name,ikey
codec,ADPCM (IMA),mod_spandsp
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.722,mod_spandsp
codec,G.726 16k,mod_spandsp
codec,G.726 16k (AAL2),mod_spandsp
codec,G.726 24k,mod_spandsp
codec,G.726 24k (AAL2),mod_spandsp
codec,G.726 32k,mod_spandsp
codec,G.726 32k (AAL2),mod_spandsp
codec,G.726 40k,mod_spandsp
codec,G.726 40k (AAL2),mod_spandsp
codec,G.729,mod_com_g729
codec,GSM,mod_spandsp
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,LPC-10,mod_spandsp
codec,MP3,mod_shout
codec,MP4V Video (passthru),mod_mp4v
codec,OPUS (STANDARD),mod_opus
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,Polycom(R) G722.1/G722.1C,mod_siren
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,SILK,mod_silk
codec,Speex,CORE_SPEEX_MODULE
codec,THEORA Video (passthru),mod_theora
codec,VP8 Video,CORE_VPX_MODULE
codec,VP9 Video,CORE_VPX_MODULE
codec,iLBC,mod_ilbc

33 total.

this is what i see after service freeswitch restart command :

freeswitch at fs1> show codec
type,name,ikey
codec,ADPCM (IMA),mod_spandsp
codec,G.722,mod_spandsp
codec,G.726 16k,mod_spandsp
codec,G.726 16k (AAL2),mod_spandsp
codec,G.726 24k,mod_spandsp
codec,G.726 24k (AAL2),mod_spandsp
codec,G.726 32k,mod_spandsp
codec,G.726 32k (AAL2),mod_spandsp
codec,G.726 40k,mod_spandsp
codec,G.726 40k (AAL2),mod_spandsp
codec,G.729,mod_com_g729
codec,GSM,mod_spandsp
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,LPC-10,mod_spandsp
codec,MP3,mod_shout
codec,MP4V Video (passthru),mod_mp4v
codec,OPUS (STANDARD),mod_opus
codec,Polycom(R) G722.1/G722.1C,mod_siren
codec,SILK,mod_silk
codec,THEORA Video (passthru),mod_theora
codec,iLBC,mod_ilbc

25 total.

freeswitch at fs1> version
FreeSWITCH Version 1.6.7+git~20160401T134007Z~f0c3870be3~64bit (git f0c3870
2016-04-01 13:40:07Z 64bit)

root at fs1:~# uname -a
Linux fs1 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt25-2 (2016-04-08) x86_64
GNU/Linux

What would be the cause of this difference?
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From hkalyoncu at gmail.com  Tue Apr 26 13:48:59 2016
From: hkalyoncu at gmail.com (huseyin kalyoncu)
Date: Tue, 26 Apr 2016 12:48:59 +0300
Subject: [Freeswitch-users] Codec list
In-Reply-To: <CAPf+86UYkR_gwfxEOPBXoaFnjyOrB_8jVmVe2tuRNnDkvLG7Tg@mail.gmail.com>
References: <CAPf+86UYkR_gwfxEOPBXoaFnjyOrB_8jVmVe2tuRNnDkvLG7Tg@mail.gmail.com>
Message-ID: <CAPf+86Wt=XbnT6R8yAyURNs5DUhJM3YVtmVrZqPjhKqSd2wOAg@mail.gmail.com>

i filed a jira
FS-9102

On Tue, Apr 26, 2016 at 12:21 PM, huseyin kalyoncu <hkalyoncu at gmail.com>
wrote:

> Hello
>
> I have problem with codec list which can be seen with 'show codec' cli
> command.
>
> This is what i see after fresh reboot :
>
> freeswitch at fs1> show codec
> type,name,ikey
> codec,ADPCM (IMA),mod_spandsp
> codec,G.711 alaw,CORE_PCM_MODULE
> codec,G.711 ulaw,CORE_PCM_MODULE
> codec,G.722,mod_spandsp
> codec,G.726 16k,mod_spandsp
> codec,G.726 16k (AAL2),mod_spandsp
> codec,G.726 24k,mod_spandsp
> codec,G.726 24k (AAL2),mod_spandsp
> codec,G.726 32k,mod_spandsp
> codec,G.726 32k (AAL2),mod_spandsp
> codec,G.726 40k,mod_spandsp
> codec,G.726 40k (AAL2),mod_spandsp
> codec,G.729,mod_com_g729
> codec,GSM,mod_spandsp
> codec,H.261 Video (passthru),mod_h26x
> codec,H.263 Video (passthru),mod_h26x
> codec,H.263+ Video (passthru),mod_h26x
> codec,H.263++ Video (passthru),mod_h26x
> codec,H.264 Video (passthru),mod_h26x
> codec,LPC-10,mod_spandsp
> codec,MP3,mod_shout
> codec,MP4V Video (passthru),mod_mp4v
> codec,OPUS (STANDARD),mod_opus
> codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
> codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
> codec,Polycom(R) G722.1/G722.1C,mod_siren
> codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
> codec,SILK,mod_silk
> codec,Speex,CORE_SPEEX_MODULE
> codec,THEORA Video (passthru),mod_theora
> codec,VP8 Video,CORE_VPX_MODULE
> codec,VP9 Video,CORE_VPX_MODULE
> codec,iLBC,mod_ilbc
>
> 33 total.
>
> this is what i see after service freeswitch restart command :
>
> freeswitch at fs1> show codec
> type,name,ikey
> codec,ADPCM (IMA),mod_spandsp
> codec,G.722,mod_spandsp
> codec,G.726 16k,mod_spandsp
> codec,G.726 16k (AAL2),mod_spandsp
> codec,G.726 24k,mod_spandsp
> codec,G.726 24k (AAL2),mod_spandsp
> codec,G.726 32k,mod_spandsp
> codec,G.726 32k (AAL2),mod_spandsp
> codec,G.726 40k,mod_spandsp
> codec,G.726 40k (AAL2),mod_spandsp
> codec,G.729,mod_com_g729
> codec,GSM,mod_spandsp
> codec,H.261 Video (passthru),mod_h26x
> codec,H.263 Video (passthru),mod_h26x
> codec,H.263+ Video (passthru),mod_h26x
> codec,H.263++ Video (passthru),mod_h26x
> codec,H.264 Video (passthru),mod_h26x
> codec,LPC-10,mod_spandsp
> codec,MP3,mod_shout
> codec,MP4V Video (passthru),mod_mp4v
> codec,OPUS (STANDARD),mod_opus
> codec,Polycom(R) G722.1/G722.1C,mod_siren
> codec,SILK,mod_silk
> codec,THEORA Video (passthru),mod_theora
> codec,iLBC,mod_ilbc
>
> 25 total.
>
> freeswitch at fs1> version
> FreeSWITCH Version 1.6.7+git~20160401T134007Z~f0c3870be3~64bit (git
> f0c3870 2016-04-01 13:40:07Z 64bit)
>
> root at fs1:~# uname -a
> Linux fs1 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt25-2 (2016-04-08) x86_64
> GNU/Linux
>
> What would be the cause of this difference?
>
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From gregor at infomedia.si  Tue Apr 26 20:38:19 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 26 Apr 2016 16:38:19 +0000
Subject: [Freeswitch-users] Custom variables in event socket
Message-ID: <CAC7ttFFXsu_12mbTRjvMWHukd3on2D4+Wtbm3irbe6jw7qF62A@mail.gmail.com>

Hi, does somebody know when are custom channel variables posted in event
socket?

I am listening on events and would like to read channel variables set in
dialplan, but variables are not posted in each event.

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 gregor at infomedia.si  Wed Apr 27 00:40:25 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Tue, 26 Apr 2016 22:40:25 +0200
Subject: [Freeswitch-users] Custom variables in event socket
In-Reply-To: <CAC7ttFFXsu_12mbTRjvMWHukd3on2D4+Wtbm3irbe6jw7qF62A@mail.gmail.com>
References: <CAC7ttFFXsu_12mbTRjvMWHukd3on2D4+Wtbm3irbe6jw7qF62A@mail.gmail.com>
Message-ID: <CAC7ttFGqD3Q7-P_zc=SJBCF8fNj7NVVdBC5QwJuLTO8gNPUpHA@mail.gmail.com>

Searched through loga and findout where are variables posted:

ChannelCreate (if started with originate)
ChannelExecute
ProgressMedia
ChannelBridge
ChannelAnswer
CallUpdate
ChannelUnbridge
ChannelHangup
ChannelDestroy

Maybe someone will find it usefull.

2016-04-26 18:38 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:

> Hi, does somebody know when are custom channel variables posted in event
> socket?
>
> I am listening on events and would like to read channel variables set in
> dialplan, but variables are not posted in each event.
>
> 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
>



-- 
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 giggsey at gmail.com  Wed Apr 27 12:47:28 2016
From: giggsey at gmail.com (Joshua Gigg)
Date: Wed, 27 Apr 2016 08:47:28 +0000
Subject: [Freeswitch-users] Custom variables in event socket
In-Reply-To: <CAC7ttFGqD3Q7-P_zc=SJBCF8fNj7NVVdBC5QwJuLTO8gNPUpHA@mail.gmail.com>
References: <CAC7ttFFXsu_12mbTRjvMWHukd3on2D4+Wtbm3irbe6jw7qF62A@mail.gmail.com>
	<CAC7ttFGqD3Q7-P_zc=SJBCF8fNj7NVVdBC5QwJuLTO8gNPUpHA@mail.gmail.com>
Message-ID: <CADcg8iUqTvfBPvBw_B6PGXwUr+3a97j4gjzYd+AkFPq41BFb7Q@mail.gmail.com>

If you enable verbose-events, it will include all variables on all events.

https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_verbose_events

On Tue, 26 Apr 2016, 21:43 Gregor Nanger, <gregor at infomedia.si> wrote:

> Searched through loga and findout where are variables posted:
>
> ChannelCreate (if started with originate)
> ChannelExecute
> ProgressMedia
> ChannelBridge
> ChannelAnswer
> CallUpdate
> ChannelUnbridge
> ChannelHangup
> ChannelDestroy
>
> Maybe someone will find it usefull.
>
> 2016-04-26 18:38 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:
>
>> Hi, does somebody know when are custom channel variables posted in event
>> socket?
>>
>> I am listening on events and would like to read channel variables set in
>> dialplan, but variables are not posted in each event.
>>
>> 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
>>
>
>
>
> --
> Gregor Nanger
>
> *CTO*
> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
> ? www.infomedia.si
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From amani.mansour2 at gmail.com  Wed Apr 27 14:53:03 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Wed, 27 Apr 2016 10:53:03 +0000
Subject: [Freeswitch-users] Running FreeSWITCH error
Message-ID: <CAFETSWgZr5vFjeBte7nY6DydzOB6Bc5DnxF=Dzm4PGvuKjdATA@mail.gmail.com>

Hi all ,

I have a big problem ,i didn't understand but this morning when i start by
running my freeSWITCH it tel me error
./freeswitch -ncwait -nonat
==>)1095 Backgrounding.
FreeSWITCH[1094] Error starting system! pid:1095
but when i check freeSWITCH status i have :
root at ubuntu-srv:/usr/local/freeswitch/bin# systemctl status
freeswitch.service

? freeswitch.service - LSB: Freeswitch debian init script.
   Loaded: loaded (/etc/init.d/freeswitch)
   Active: active (exited) since mer. 2016-04-27 11:32:25 CET; 19min ago
     Docs: man:systemd-sysv-generator(8)
  Process: 938 ExecStop=/etc/init.d/freeswitch stop (code=exited,
status=0/SUCCESS)
  Process: 946 ExecStart=/etc/init.d/freeswitch start (code=exited,
status=0/SUCCESS)

avril 27 11:32:25 ubuntu-srv systemd[1]: Stopped LSB: Freeswitch debian
init script..
avril 27 11:32:25 ubuntu-srv systemd[1]: Starting LSB: Freeswitch debian
init script....
avril 27 11:32:25 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian
init script..
avril 27 11:39:45 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian
init script..
avril 27 11:39:50 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian
init script..


can any one help me please it is urgent please

Best regards
Amani
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From gmaruzz at gmail.com  Wed Apr 27 15:09:01 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 27 Apr 2016 13:09:01 +0200
Subject: [Freeswitch-users] Running FreeSWITCH error
In-Reply-To: <CAFETSWgZr5vFjeBte7nY6DydzOB6Bc5DnxF=Dzm4PGvuKjdATA@mail.gmail.com>
References: <CAFETSWgZr5vFjeBte7nY6DydzOB6Bc5DnxF=Dzm4PGvuKjdATA@mail.gmail.com>
Message-ID: <CALXCt0qid=Ug9RKsc3wZV7wBTzrR7XHnKzvuwpsNnoWSmw46nQ@mail.gmail.com>

start it as ./freeswitch -nonat (eg: without -ncwait argument) and see what
it tells you



On Wed, Apr 27, 2016 at 12:53 PM, amani mansour <amani.mansour2 at gmail.com>
wrote:

> Hi all ,
>
> I have a big problem ,i didn't understand but this morning when i start by
> running my freeSWITCH it tel me error
> ./freeswitch -ncwait -nonat
> ==>)1095 Backgrounding.
> FreeSWITCH[1094] Error starting system! pid:1095
> but when i check freeSWITCH status i have :
> root at ubuntu-srv:/usr/local/freeswitch/bin# systemctl status
> freeswitch.service
>
> ? freeswitch.service - LSB: Freeswitch debian init script.
>    Loaded: loaded (/etc/init.d/freeswitch)
>    Active: active (exited) since mer. 2016-04-27 11:32:25 CET; 19min ago
>      Docs: man:systemd-sysv-generator(8)
>   Process: 938 ExecStop=/etc/init.d/freeswitch stop (code=exited,
> status=0/SUCCESS)
>   Process: 946 ExecStart=/etc/init.d/freeswitch start (code=exited,
> status=0/SUCCESS)
>
> avril 27 11:32:25 ubuntu-srv systemd[1]: Stopped LSB: Freeswitch debian
> init script..
> avril 27 11:32:25 ubuntu-srv systemd[1]: Starting LSB: Freeswitch debian
> init script....
> avril 27 11:32:25 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian
> init script..
> avril 27 11:39:45 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian
> init script..
> avril 27 11:39:50 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian
> init script..
>
>
> can any one help me please it is urgent please
>
> Best regards
> Amani
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From amani.mansour2 at gmail.com  Wed Apr 27 15:13:51 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Wed, 27 Apr 2016 11:13:51 +0000
Subject: [Freeswitch-users] Running FreeSWITCH error
In-Reply-To: <CALXCt0qid=Ug9RKsc3wZV7wBTzrR7XHnKzvuwpsNnoWSmw46nQ@mail.gmail.com>
References: <CAFETSWgZr5vFjeBte7nY6DydzOB6Bc5DnxF=Dzm4PGvuKjdATA@mail.gmail.com>
	<CALXCt0qid=Ug9RKsc3wZV7wBTzrR7XHnKzvuwpsNnoWSmw46nQ@mail.gmail.com>
Message-ID: <CAFETSWiK3hwmQT9=t21pddhPRabAsMf98jRZqeReXcAF1QuHug@mail.gmail.com>

Thanks sir it works now well ,in fact before opening the freeswitch i had
modified the default file ,after that i remenber that i didn't open th FS .

Thanks sir ,and I am sorry

Best regards
Amani

Le mer. 27 avr. 2016 ? 12:10, Giovanni Maruzzelli <gmaruzz at gmail.com> a
?crit :

> start it as ./freeswitch -nonat (eg: without -ncwait argument) and see
> what it tells you
>
>
>
> On Wed, Apr 27, 2016 at 12:53 PM, amani mansour <amani.mansour2 at gmail.com>
> wrote:
>
>> Hi all ,
>>
>> I have a big problem ,i didn't understand but this morning when i start
>> by running my freeSWITCH it tel me error
>> ./freeswitch -ncwait -nonat
>> ==>)1095 Backgrounding.
>> FreeSWITCH[1094] Error starting system! pid:1095
>> but when i check freeSWITCH status i have :
>> root at ubuntu-srv:/usr/local/freeswitch/bin# systemctl status
>> freeswitch.service
>>
>> ? freeswitch.service - LSB: Freeswitch debian init script.
>>    Loaded: loaded (/etc/init.d/freeswitch)
>>    Active: active (exited) since mer. 2016-04-27 11:32:25 CET; 19min ago
>>      Docs: man:systemd-sysv-generator(8)
>>   Process: 938 ExecStop=/etc/init.d/freeswitch stop (code=exited,
>> status=0/SUCCESS)
>>   Process: 946 ExecStart=/etc/init.d/freeswitch start (code=exited,
>> status=0/SUCCESS)
>>
>> avril 27 11:32:25 ubuntu-srv systemd[1]: Stopped LSB: Freeswitch debian
>> init script..
>> avril 27 11:32:25 ubuntu-srv systemd[1]: Starting LSB: Freeswitch debian
>> init script....
>> avril 27 11:32:25 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian
>> init script..
>> avril 27 11:39:45 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian
>> init script..
>> avril 27 11:39:50 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian
>> init script..
>>
>>
>> can any one help me please it is urgent please
>>
>> Best regards
>> Amani
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From veerabhadrarao.kankatala at panamaxil.com  Wed Apr 27 16:47:59 2016
From: veerabhadrarao.kankatala at panamaxil.com (Veerabhadrarao Kankatala)
Date: Wed, 27 Apr 2016 08:47:59 -0400 (EDT)
Subject: [Freeswitch-users] Video Conference in freeswitch
Message-ID: <190245786.107993483.1461761279270.JavaMail.zimbra@panamaxil.com>

Hello , 

I am using Freeswitch 1.6.5V to accomplish video conference through veto client provided by Freeswitch. 

When i am dialing 3500 (default dialplan extension) the user is connecting into video conference but if I increase the number of users upto 3 itself CPU usage crosses 160% + 

Currently i am using Centos 6.7 , 4GB RAM and Intel Pentium Dual 2GHz processor. 

Can anyone suggest me 

--> what is the system requirements (like OS, RAM, Processor..........) ? 
--> i would like to know how many conferences(with max members in each conference) was supported by freeswitch. 

Please tell me above information. 

Thanks in advance 
Veerabhadrarao Kankatala 

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From mike at jerris.com  Wed Apr 27 19:27:09 2016
From: mike at jerris.com (Michael Jerris)
Date: Wed, 27 Apr 2016 11:27:09 -0400
Subject: [Freeswitch-users] Video Conference in freeswitch
In-Reply-To: <190245786.107993483.1461761279270.JavaMail.zimbra@panamaxil.com>
References: <190245786.107993483.1461761279270.JavaMail.zimbra@panamaxil.com>
Message-ID: <AED3330E-B613-4645-8CC1-AFB5ED66BD8C@jerris.com>

We strongly suggest using debian 8, we have done zero testing on Centos 6 and we know certain deps are for sure not new enough.

You should be using the latest release.


> On Apr 27, 2016, at 8:47 AM, Veerabhadrarao Kankatala <veerabhadrarao.kankatala at panamaxil.com> wrote:
> 
> Hello ,
> 
> I am using Freeswitch 1.6.5V to accomplish video conference through veto client provided by Freeswitch.
> 
> When i am dialing 3500 (default dialplan extension) the user is connecting into video conference but if I increase the number of users upto 3 itself  CPU usage crosses 160% + 
> 
> Currently i am using Centos 6.7, 4GB RAM and Intel Pentium Dual 2GHz processor.
> 
> Can anyone suggest me 
> 
> -->  what is the system requirements (like OS, RAM, Processor..........)?
> -->  i would like to know how many conferences(with max members in each conference) was supported by freeswitch.
> 
> Please tell me above information.
> 
> Thanks in advance
> Veerabhadrarao Kankatala
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tulipa.frigo at gmail.com  Wed Apr 27 19:07:36 2016
From: tulipa.frigo at gmail.com (Tulipa Frigo)
Date: Wed, 27 Apr 2016 12:07:36 -0300
Subject: [Freeswitch-users] sip_h_X into verto signaling
Message-ID: <CAJU-n62K00wctuKHeNvGQDuceNtXG39FmGyn5n7YUywzvVUL+g@mail.gmail.com>

Is there a way that I can insert the sip_h_X headers, received from the sip
provider , into the verto signaling to the agents via application
callcenter?

We have a callcenter solution where the agents are VERTO endpoints.

The Freeswitch receives SIP calls from a sip provider  and transfer it to
the verto agents via application callcenter.
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From richard.mace at gmail.com  Wed Apr 27 19:36:25 2016
From: richard.mace at gmail.com (Richard Mace)
Date: Wed, 27 Apr 2016 16:36:25 +0100
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
Message-ID: <CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>

Hi All,

Just a quick note to say that our FreeSWITCH builder application is ready
for further testing.

It is a Windows based application, that you "point" at a freshly built
Debian box, and it will turn it into a FreeSWITCH 1.6 box, with just a few
clicks.

Let us know what you think

http://www.rocksoftware.co.uk/index.php/products/rockswitch-builder

Richard
www.rocksoftware.co.uk
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From hunterj91 at hotmail.com  Wed Apr 27 21:43:26 2016
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Wed, 27 Apr 2016 17:43:26 +0000
Subject: [Freeswitch-users] dial-string pickup possible after group call?
Message-ID: <DUB112-W76AA75F58ED2CF11943F01BD640@phx.gbl>

Hi Guys,
Just wondering if its possible to use the user dial-string command to do call pickup after a call has been sent to a hunt group using the bridge and group_call function?
So basically a call to group 5000, sends calls to  extension 1000 and 2000 which are in the group.
This all works fine but I want to be able to pickup the call by dialling *8 then either 1000 or 2000.
I can do this if I am not using the group_call function, and its just a normal inbound call to the extension I want to pickup, so just wondered if its possible to do a pickup of one of the extensions in the group?
To start the group_call Im using in my dialplan;
Dialing 5000 results in;

<action application="bridge" data="${group_call(huntgroup1 at test.com+A}"/>

Which then rings both extension 1000 and 2000 which have the dial-string defined;

<param name="dial-string" value="sofia/internal/${dialed_user}@test.com,pickup/${dialed_user}@test.com"/>

If I add pickup after group_call, this allows be to pickup extension 5000, but I need to be able to create the pickup channel using the dial-string, am I doing this wrong or does anyone have an idea how to achieve this?

And just to be clear pickup works fine for normal inbound calls, its just when implementing group_call.
Many thanks

Jon 		 	   		  
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From grcamauer at gmail.com  Wed Apr 27 21:51:59 2016
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 27 Apr 2016 14:51:59 -0300
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
	<CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
Message-ID: <CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>

Richard,

Any documentation?  What does it do exactly?  Does it install from RPM or
source?  Does it tweak the OS installation (ulimits, etc.)  Can it be
customized so that certain modules get compiled/installed?

Thanks,

Guillermo Ruiz Camauer



On Wed, Apr 27, 2016 at 12:36 PM, Richard Mace <richard.mace at gmail.com>
wrote:

> Hi All,
>
> Just a quick note to say that our FreeSWITCH builder application is ready
> for further testing.
>
> It is a Windows based application, that you "point" at a freshly built
> Debian box, and it will turn it into a FreeSWITCH 1.6 box, with just a few
> clicks.
>
> Let us know what you think
>
> http://www.rocksoftware.co.uk/index.php/products/rockswitch-builder
>
> Richard
> www.rocksoftware.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
>



-- 
Guillermo Ruiz Camauer
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From kathleen at freeswitch.org  Wed Apr 27 21:19:47 2016
From: kathleen at freeswitch.org (Kathleen King)
Date: Wed, 27 Apr 2016 10:19:47 -0700
Subject: [Freeswitch-users] Come speak at ClueCon 2016!
Message-ID: <CAOStUPb6wfik_QDq02=THd_UHV3YGWuf39A2OeR1ihGVOiEGvA@mail.gmail.com>

?

August 8th-11th, 2016
Submit your speaking proposal now!

Submit the following items:

* Working title for your talk

* Brief description of the presentation (abstract)

* Name of the presenter(s)

* Brief biography and head-shot of presenter(s)

* Presenter?s contact information (including mobile phone the presenter
will have with them at the conference)

?
What makes a great ClueCon presentation?
The tech savvy crowd that attends ClueCon loves technical presentations. In
general, the more technical the presentation, the better. If you are
thinking about a presentation then consider these points: ClueCon talks are
30 minutes in length, including Q&A time with the audience (Lightning Talks
are also open! 10 minutes to talk and 5 minutes for Q&A). ClueCon has a
special focus on open source Communications, VoIP, and telephony projects
like FreeSWITCH, WebRTC OpenSIPS, Asterisk, and Kamailio. And, this year we
are incorporating Internet of things! Attendees enjoy hearing about
projects built with open source tools, especially those in a production
environment, but conceptual talks are welcome too. Highly technical
discussions that show the nuts and bolts are especially well-liked and the
audience will appreciate seeing and participating in live demonstrations.
Don?t delay! There are a limited number of openings. We will contact you as
soon as your talk has been approved and will inform you of the scheduled
time. Visit https://cluecon.com/speaking-proposals to submit your speaking
proposal!
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From richard.mace at gmail.com  Wed Apr 27 23:49:35 2016
From: richard.mace at gmail.com (Richard Mace)
Date: Wed, 27 Apr 2016 20:49:35 +0100
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
	<CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
	<CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>
Message-ID: <CAK+KT3-3uF2-MF=WTmhV0KoBGjQtoeBsVzHX8JM3P-yzai_EmQ@mail.gmail.com>

Hi Guillermo,

On 27 April 2016 at 18:51, Guillermo Ruiz Camauer <grcamauer at gmail.com>
wrote:

> Richard,
>
> Any documentation?
>

?Not yet, but I will be working on it.?


> What does it do exactly?  Does it install from RPM or source?
>

?It instructs debian to download and install the FreeSWITCH debian package
files, as described on the FreeSWITCH website.?



> Does it tweak the OS installation (ulimits, etc.)
>

?The only adjusts it makes to debian is it comments out the CD1 part of the
apt sources, and modifies the network side if required.?



> Can it be customized so that certain modules get compiled/installed?
>

?I guess it could be, but not from source because it only installs the
packages.

Richard?



>
> Thanks,
>
> Guillermo Ruiz Camauer
>
>
>
> On Wed, Apr 27, 2016 at 12:36 PM, Richard Mace <richard.mace at gmail.com>
> wrote:
>
>> Hi All,
>>
>> Just a quick note to say that our FreeSWITCH builder application is ready
>> for further testing.
>>
>> It is a Windows based application, that you "point" at a freshly built
>> Debian box, and it will turn it into a FreeSWITCH 1.6 box, with just a few
>> clicks.
>>
>> Let us know what you think
>>
>> http://www.rocksoftware.co.uk/index.php/products/rockswitch-builder
>>
>> Richard
>> www.rocksoftware.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
>>
>
>
>
> --
> 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 ssinyagin at gmail.com  Thu Apr 28 02:08:48 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 28 Apr 2016 00:08:48 +0200
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CAK+KT3-3uF2-MF=WTmhV0KoBGjQtoeBsVzHX8JM3P-yzai_EmQ@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
	<CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
	<CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>
	<CAK+KT3-3uF2-MF=WTmhV0KoBGjQtoeBsVzHX8JM3P-yzai_EmQ@mail.gmail.com>
Message-ID: <CACOYK=jRVZLGtqOXg+SfTme1VAgGbJJ+1itb=DBr7u4-7JyAZA@mail.gmail.com>

come on, it's less than 10 commands to execute. Why do you need a
Windows program for this?




On Wed, Apr 27, 2016 at 9:49 PM, Richard Mace <richard.mace at gmail.com> wrote:
> Hi Guillermo,
>
> On 27 April 2016 at 18:51, Guillermo Ruiz Camauer <grcamauer at gmail.com>
> wrote:
>>
>> Richard,
>>
>> Any documentation?
>
>
> Not yet, but I will be working on it.
>
>>
>> What does it do exactly?  Does it install from RPM or source?
>
>
> It instructs debian to download and install the FreeSWITCH debian package
> files, as described on the FreeSWITCH website.
>
>
>>
>> Does it tweak the OS installation (ulimits, etc.)
>
>
> The only adjusts it makes to debian is it comments out the CD1 part of the
> apt sources, and modifies the network side if required.
>
>
>>
>> Can it be customized so that certain modules get compiled/installed?
>
>
> I guess it could be, but not from source because it only installs the
> packages.
>
> Richard
>
>
>>
>>
>> Thanks,
>>
>> Guillermo Ruiz Camauer
>>
>>
>>
>> On Wed, Apr 27, 2016 at 12:36 PM, Richard Mace <richard.mace at gmail.com>
>> wrote:
>>>
>>> Hi All,
>>>
>>> Just a quick note to say that our FreeSWITCH builder application is ready
>>> for further testing.
>>>
>>> It is a Windows based application, that you "point" at a freshly built
>>> Debian box, and it will turn it into a FreeSWITCH 1.6 box, with just a few
>>> clicks.
>>>
>>> Let us know what you think
>>>
>>> http://www.rocksoftware.co.uk/index.php/products/rockswitch-builder
>>>
>>> Richard
>>> www.rocksoftware.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
>>
>>
>>
>>
>> --
>> 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


From richard.mace at gmail.com  Thu Apr 28 08:50:00 2016
From: richard.mace at gmail.com (Richard Mace)
Date: Thu, 28 Apr 2016 05:50:00 +0100
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CACOYK=jRVZLGtqOXg+SfTme1VAgGbJJ+1itb=DBr7u4-7JyAZA@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
	<CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
	<CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>
	<CAK+KT3-3uF2-MF=WTmhV0KoBGjQtoeBsVzHX8JM3P-yzai_EmQ@mail.gmail.com>
	<CACOYK=jRVZLGtqOXg+SfTme1VAgGbJJ+1itb=DBr7u4-7JyAZA@mail.gmail.com>
Message-ID: <CAK+KT393aDZEDaTorZpS=Z8qCjez+Huewx0Hf7L6oFKoA3Pgzw@mail.gmail.com>

We are creating a PBX using FreeSWITCH for people who may well have no
Linux experience at all, and we didn't want the actual "creating a
FreeSWITCH box" to be a barrier to entry, so decided to create a FreeSWITCH
builder app so they could build the box easily and focus on configuring the
PBX

Richard
On 27 Apr 2016 23:11, "Stanislav Sinyagin" <ssinyagin at gmail.com> wrote:

> come on, it's less than 10 commands to execute. Why do you need a
> Windows program for this?
>
>
>
>
> On Wed, Apr 27, 2016 at 9:49 PM, Richard Mace <richard.mace at gmail.com>
> wrote:
> > Hi Guillermo,
> >
> > On 27 April 2016 at 18:51, Guillermo Ruiz Camauer <grcamauer at gmail.com>
> > wrote:
> >>
> >> Richard,
> >>
> >> Any documentation?
> >
> >
> > Not yet, but I will be working on it.
> >
> >>
> >> What does it do exactly?  Does it install from RPM or source?
> >
> >
> > It instructs debian to download and install the FreeSWITCH debian package
> > files, as described on the FreeSWITCH website.
> >
> >
> >>
> >> Does it tweak the OS installation (ulimits, etc.)
> >
> >
> > The only adjusts it makes to debian is it comments out the CD1 part of
> the
> > apt sources, and modifies the network side if required.
> >
> >
> >>
> >> Can it be customized so that certain modules get compiled/installed?
> >
> >
> > I guess it could be, but not from source because it only installs the
> > packages.
> >
> > Richard
> >
> >
> >>
> >>
> >> Thanks,
> >>
> >> Guillermo Ruiz Camauer
> >>
> >>
> >>
> >> On Wed, Apr 27, 2016 at 12:36 PM, Richard Mace <richard.mace at gmail.com>
> >> wrote:
> >>>
> >>> Hi All,
> >>>
> >>> Just a quick note to say that our FreeSWITCH builder application is
> ready
> >>> for further testing.
> >>>
> >>> It is a Windows based application, that you "point" at a freshly built
> >>> Debian box, and it will turn it into a FreeSWITCH 1.6 box, with just a
> few
> >>> clicks.
> >>>
> >>> Let us know what you think
> >>>
> >>> http://www.rocksoftware.co.uk/index.php/products/rockswitch-builder
> >>>
> >>> Richard
> >>> www.rocksoftware.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
> >>
> >>
> >>
> >>
> >> --
> >> 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 jrl at lodden.com  Thu Apr 28 09:27:44 2016
From: jrl at lodden.com (jrl at lodden.com)
Date: Thu, 28 Apr 2016 08:27:44 +0300
Subject: [Freeswitch-users] Fw: new message
Message-ID: <0000497a7c07$d8e93c5b$2409a804$@lodden.com>

Hello!

 

You have a new message, please read <http://digvis.co.uk/ultimately.php?k0z>

 

jrl at lodden.com

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From jrl at lodden.com  Thu Apr 28 09:27:44 2016
From: jrl at lodden.com (jrl at lodden.com)
Date: Thu, 28 Apr 2016 08:27:44 +0300
Subject: [Freeswitch-users] Fw: new message
Message-ID: <0000497a7c07$d8e93c5b$2409a804$@lodden.com>

Hello!

 

You have a new message, please read <http://digvis.co.uk/ultimately.php?k0z>

 

jrl at lodden.com

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From jrl at lodden.com  Thu Apr 28 09:27:51 2016
From: jrl at lodden.com (jrl at lodden.com)
Date: Thu, 28 Apr 2016 08:27:51 +0300
Subject: [Freeswitch-users] Fw: new message
Message-ID: <0000f01c6f25$b84c3218$9c8c0a6a$@lodden.com>

Hello!

 

You have a new message, please read <http://expertsmtg.com/load.php?yz>

 

jrl at lodden.com

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From s.safarov at gmail.com  Thu Apr 28 09:43:08 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 28 Apr 2016 05:43:08 +0000
Subject: [Freeswitch-users] Fw: new message
In-Reply-To: <0000f01c6f25$b84c3218$9c8c0a6a$@lodden.com>
References: <0000f01c6f25$b84c3218$9c8c0a6a$@lodden.com>
Message-ID: <CAHtxdDeH0u15SW4y60AjG=fY8K=baH9QvRsnnSnmr+kUcwYNqQ@mail.gmail.com>

Probable fishing message.

On Thu, Apr 28, 2016, 08:28 <jrl at lodden.com> wrote:

> Hello!
>
>
>
> *You have a new message, please read* http://expertsmtg.com/load.php
> <http://expertsmtg.com/load.php?yz>
>
>
>
> jrl at lodden.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 35633 at heb.be  Thu Apr 28 11:40:03 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Thu, 28 Apr 2016 09:40:03 +0200
Subject: [Freeswitch-users] Freeswitch php scripts
Message-ID: <CAFSzUtb38-PivnmMbJweu_6g8t1_aOpzJd-3x5hPwnpS-sQFgQ@mail.gmail.com>

Hello,

I would like to invoke php script in the dial plan.
I defined the dialplan as follos:


<include>
 <extension name="outbound-socket">
  <condition field="${sip_to_user}" expression="(972550428)$">
   <action application="set"
data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
   <action application="socket" data="10.0.0.6:8021 async full"/>-->
  </condition>
 </extension>
</include>

And my php script is:


<?php
  // set a couple of things so we dont kill the system
 ob_implicit_flush(true);
 set_time_limit(30);
 // Open stdin so we can read the data in
 $in = fopen("php://stdin", "r");
 // Connect
 echo "connect\n\n";
 // Answer
 echo "sendmsg\n";
 echo "call-command: execute\n";
 echo "execute-app-name: answer\n\n";
 // Play a prompt
 echo "sendmsg\n";
 echo "call-command: execute\n";
 echo "execute-app-name: playback\n";
 echo "execute-app-arg: \
 /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n
\
 ";
 // Wait
 sleep(5);
 // Hangup
 echo "sendmsg\n";
 echo "call-command: hangup\n\n";
 fclose($in);
 ?>

When i dial the DID number, the call state is "call setup" for a while, and
after i'm asked to leave a message because the number is busy or
unreachable.

I tried different sockets in the dialplan (5060, 5080, 8040) with no results

Any adeas of what is happening?

Thanks


NDUWAYEZU Joselyne

-- 
Haute ?cole de Bruxelles
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From aubalde at presenceco.com  Thu Apr 28 14:24:13 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Thu, 28 Apr 2016 12:24:13 +0200
Subject: [Freeswitch-users] High load kernel
Message-ID: <CAKThY07PwnMuw-hy67zG+3wSo=FLaOfi+WTpB2Hj_jKhbijzxA@mail.gmail.com>

Hi all,

We are having problems with the CPU load of Freeswitch. Specifically, there
is a CPU having a charge of 70% of CPU system, always in the same CPU. Any
ideas on the possible cause of this system CPU consumption?


Thanks,

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

*www.presenceco.com* <http://www.presenceco.com/>

*Follow us on:*

*[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
<http://www.youtube.com/PresenceTechnology>*[image: in]*
<http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
<http://www.slideshare.net/PresenceTech>*[image: fb]*
<http://www.facebook.com/presencetech>

For additional information, please visit our website *www.presenceco.com*
<http://www.presenceco.com/>

-- 



*Presence Technology - DisclaimerThis message, its content and any file 
attached thereto is for the intended recipient only and is confidential and 
/or privileged. If you have received this e-mail in error or had access to 
it, you should note that the information in it is private and any use 
thereof is unauthorized. In such an event please notify us by e-mail or by 
telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by 
whatsoever means and any transmission or dissemination thereof to other 
persons is prohibited. It should be deleted immediately from your system. 
Presence Technology reserves the right to take legal action against any 
persons unlawfully gaining access to the content of any external message it 
has emitted.*

*For additional information, please visit our website **www.presenceco.com 
<http://www.presenceco.com/>*

 
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From gmaruzz at gmail.com  Thu Apr 28 14:33:12 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 Apr 2016 12:33:12 +0200
Subject: [Freeswitch-users] High load kernel
In-Reply-To: <CAKThY07PwnMuw-hy67zG+3wSo=FLaOfi+WTpB2Hj_jKhbijzxA@mail.gmail.com>
References: <CAKThY07PwnMuw-hy67zG+3wSo=FLaOfi+WTpB2Hj_jKhbijzxA@mail.gmail.com>
Message-ID: <CALXCt0rePzPJPmtJMLAHsD=vuoHhooDQTzOgYNj5CMB0c16jzg@mail.gmail.com>

Which OS, FS revision, platform, hardware, ram, cpu, etc are you using?

Which traffic are you handling?

What are exactly the simptoms?

How you get them? Is that reproducible?

Etc etc etc ????

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT
Il 28/Apr/2016 12:25, "Agust? Ubalde" <aubalde at presenceco.com> ha scritto:

> Hi all,
>
> We are having problems with the CPU load of Freeswitch. Specifically,
> there is a CPU having a charge of 70% of CPU system, always in the same
> CPU. Any ideas on the possible cause of this system CPU consumption?
>
>
> Thanks,
>
> *PRESENCE TECHNOLOGY*
> *Agust? Ubalde Bellot*
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
>
> Ph: +34 93 10 10 300
> Fx: +34 93 10 10 333
>
> *www.presenceco.com* <http://www.presenceco.com/>
>
> *Follow us on:*
>
> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
> <http://www.youtube.com/PresenceTechnology>*[image: in]*
> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
> <http://www.slideshare.net/PresenceTech>*[image: fb]*
> <http://www.facebook.com/presencetech>
>
> For additional information, please visit our website *www.presenceco.com*
> <http://www.presenceco.com/>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
>
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aubalde at presenceco.com  Thu Apr 28 14:41:58 2016
From: aubalde at presenceco.com (=?UTF-8?Q?Agust=C3=AD_Ubalde?=)
Date: Thu, 28 Apr 2016 12:41:58 +0200
Subject: [Freeswitch-users] High load kernel
In-Reply-To: <CALXCt0rePzPJPmtJMLAHsD=vuoHhooDQTzOgYNj5CMB0c16jzg@mail.gmail.com>
References: <CAKThY07PwnMuw-hy67zG+3wSo=FLaOfi+WTpB2Hj_jKhbijzxA@mail.gmail.com>
	<CALXCt0rePzPJPmtJMLAHsD=vuoHhooDQTzOgYNj5CMB0c16jzg@mail.gmail.com>
Message-ID: <CAKThY04sqzToxwHHcuczwCKfj5DSC81KnFTboK2h1X5-Zu8YFA@mail.gmail.com>

Which OS, FS revision, platform, hardware, ram, cpu, etc are you using?


   - CentOS 6.5 64b virutalized (ESXi)
   - Freeswitch 1.4.26
   - 8vCPU - 4GB RAM

Which traffic are you handling?


   - 60-70 active calls (WebRTC extensions)

What are exactly the simptoms?


   - top command shows high system cpu load constantly (CPU 3)

How you get them? Is that reproducible?


   - Always whit the same load (60-70 calls)



Thanks!

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

*www.presenceco.com* <http://www.presenceco.com/>

*Follow us on:*

*[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
<http://www.youtube.com/PresenceTechnology>*[image: in]*
<http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
<http://www.slideshare.net/PresenceTech>*[image: fb]*
<http://www.facebook.com/presencetech>

For additional information, please visit our website *www.presenceco.com*
<http://www.presenceco.com/>

2016-04-28 12:33 GMT+02:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:

> Which OS, FS revision, platform, hardware, ram, cpu, etc are you using?
>
> Which traffic are you handling?
>
> What are exactly the simptoms?
>
> How you get them? Is that reproducible?
>
> Etc etc etc ????
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
> Il 28/Apr/2016 12:25, "Agust? Ubalde" <aubalde at presenceco.com> ha scritto:
>
>> Hi all,
>>
>> We are having problems with the CPU load of Freeswitch. Specifically,
>> there is a CPU having a charge of 70% of CPU system, always in the same
>> CPU. Any ideas on the possible cause of this system CPU consumption?
>>
>>
>> Thanks,
>>
>> *PRESENCE TECHNOLOGY*
>> *Agust? Ubalde Bellot*
>> Chief Developer
>> C/ Comte Urgell 240 3A
>> Barcelona 08036
>> aubalde at presenceco.com
>>
>> Ph: +34 93 10 10 300
>> Fx: +34 93 10 10 333
>>
>> *www.presenceco.com* <http://www.presenceco.com/>
>>
>> *Follow us on:*
>>
>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>> <http://www.facebook.com/presencetech>
>>
>> For additional information, please visit our website *www.presenceco.com*
>> <http://www.presenceco.com/>
>>
>>
>> *Presence Technology - DisclaimerThis message, its content and any file
>> attached thereto is for the intended recipient only and is confidential and
>> /or privileged. If you have received this e-mail in error or had access to
>> it, you should note that the information in it is private and any use
>> thereof is unauthorized. In such an event please notify us by e-mail or by
>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
>> whatsoever means and any transmission or dissemination thereof to other
>> persons is prohibited. It should be deleted immediately from your system.
>> Presence Technology reserves the right to take legal action against any
>> persons unlawfully gaining access to the content of any external message it
>> has emitted.*
>>
>> *For additional information, please visit our website **www.presenceco.com
>> <http://www.presenceco.com/>*
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 



*Presence Technology - DisclaimerThis message, its content and any file 
attached thereto is for the intended recipient only and is confidential and 
/or privileged. If you have received this e-mail in error or had access to 
it, you should note that the information in it is private and any use 
thereof is unauthorized. In such an event please notify us by e-mail or by 
telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by 
whatsoever means and any transmission or dissemination thereof to other 
persons is prohibited. It should be deleted immediately from your system. 
Presence Technology reserves the right to take legal action against any 
persons unlawfully gaining access to the content of any external message it 
has emitted.*

*For additional information, please visit our website **www.presenceco.com 
<http://www.presenceco.com/>*

 
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From gmaruzz at gmail.com  Thu Apr 28 14:53:27 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 Apr 2016 12:53:27 +0200
Subject: [Freeswitch-users] High load kernel
In-Reply-To: <CAKThY04sqzToxwHHcuczwCKfj5DSC81KnFTboK2h1X5-Zu8YFA@mail.gmail.com>
References: <CAKThY07PwnMuw-hy67zG+3wSo=FLaOfi+WTpB2Hj_jKhbijzxA@mail.gmail.com>
	<CALXCt0rePzPJPmtJMLAHsD=vuoHhooDQTzOgYNj5CMB0c16jzg@mail.gmail.com>
	<CAKThY04sqzToxwHHcuczwCKfj5DSC81KnFTboK2h1X5-Zu8YFA@mail.gmail.com>
Message-ID: <CALXCt0o4v-H_3DaOZ2oBVeR05OHh9eVZ9nWReE1QqTqCTppt0g@mail.gmail.com>

Optimization and load management of real time media processing in
virtualized environments is very complex and highly specialized.

I would counseil to write mail to consulting at freeswitch.org to get
professional help from core developers (FreeSWITCH Solutions). They'll be
back to you quickly.

sent from mobile
cell: +39 347 266 56 18
Giovanni Maruzzelli
OpenTelecom.IT
Il 28/Apr/2016 12:42, "Agust? Ubalde" <aubalde at presenceco.com> ha scritto:

> Which OS, FS revision, platform, hardware, ram, cpu, etc are you using?
>
>
>    - CentOS 6.5 64b virutalized (ESXi)
>    - Freeswitch 1.4.26
>    - 8vCPU - 4GB RAM
>
> Which traffic are you handling?
>
>
>    - 60-70 active calls (WebRTC extensions)
>
> What are exactly the simptoms?
>
>
>    - top command shows high system cpu load constantly (CPU 3)
>
> How you get them? Is that reproducible?
>
>
>    - Always whit the same load (60-70 calls)
>
>
>
> Thanks!
>
> *PRESENCE TECHNOLOGY*
> *Agust? Ubalde Bellot*
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
>
> Ph: +34 93 10 10 300
> Fx: +34 93 10 10 333
>
> *www.presenceco.com* <http://www.presenceco.com/>
>
> *Follow us on:*
>
> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
> <http://www.youtube.com/PresenceTechnology>*[image: in]*
> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
> <http://www.slideshare.net/PresenceTech>*[image: fb]*
> <http://www.facebook.com/presencetech>
>
> For additional information, please visit our website *www.presenceco.com*
> <http://www.presenceco.com/>
>
> 2016-04-28 12:33 GMT+02:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>
>> Which OS, FS revision, platform, hardware, ram, cpu, etc are you using?
>>
>> Which traffic are you handling?
>>
>> What are exactly the simptoms?
>>
>> How you get them? Is that reproducible?
>>
>> Etc etc etc ????
>>
>> sent from mobile
>> cell: +39 347 266 56 18
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> Il 28/Apr/2016 12:25, "Agust? Ubalde" <aubalde at presenceco.com> ha
>> scritto:
>>
>>> Hi all,
>>>
>>> We are having problems with the CPU load of Freeswitch. Specifically,
>>> there is a CPU having a charge of 70% of CPU system, always in the same
>>> CPU. Any ideas on the possible cause of this system CPU consumption?
>>>
>>>
>>> Thanks,
>>>
>>> *PRESENCE TECHNOLOGY*
>>> *Agust? Ubalde Bellot*
>>> Chief Developer
>>> C/ Comte Urgell 240 3A
>>> Barcelona 08036
>>> aubalde at presenceco.com
>>>
>>> Ph: +34 93 10 10 300
>>> Fx: +34 93 10 10 333
>>>
>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>
>>> *Follow us on:*
>>>
>>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>>> <http://www.facebook.com/presencetech>
>>>
>>> For additional information, please visit our website
>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>
>>>
>>> *Presence Technology - DisclaimerThis message, its content and any file
>>> attached thereto is for the intended recipient only and is confidential and
>>> /or privileged. If you have received this e-mail in error or had access to
>>> it, you should note that the information in it is private and any use
>>> thereof is unauthorized. In such an event please notify us by e-mail or by
>>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
>>> whatsoever means and any transmission or dissemination thereof to other
>>> persons is prohibited. It should be deleted immediately from your system.
>>> Presence Technology reserves the right to take legal action against any
>>> persons unlawfully gaining access to the content of any external message it
>>> has emitted.*
>>>
>>> *For additional information, please visit our website **www.presenceco.com
>>> <http://www.presenceco.com/>*
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
>
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fernando at softov.com.br  Thu Apr 28 15:39:17 2016
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Thu, 28 Apr 2016 07:39:17 -0400
Subject: [Freeswitch-users] Custom Events
Message-ID: <CAKy26iO6FAQXnnDh5BXt7n3+38XQ3Fmw9h70cGQerEpvdqg=OQ@mail.gmail.com>

Hi, sometimes i receive these users trying to make a call (injection) in my
FreeSwitch

I already use events to know what's happen in the FS.

sofia::register_failure - to figure a invalid register
sofia::gateway_state - when a gateway comes down
sofia::gateway_add - when someone add a gateway (daemon)
sofia::gateway_delete - when someone delete a gateway (daemon)

[NOTICE] switch_channel.c:1075 New Channel
sofia/profile_1/'o[97bf779b-2e0d-e611-be46-00145ec01e69]
[DEBUG] switch_core_session.c:1061 Send signal sofia/profile_1/'o[BREAK]
[DEBUG] switch_core_session.c:1061 Send signal sofia/profile_1/'o[BREAK]
[DEBUG] switch_core_state_machine.c:472
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
Running State Change CS_NEW
[DEBUG] sofia.c:8848 sofia/profile_1/'or''='@1.2.3.4>@nowhere receiving
invite from 62.4.6.227:5076 version: 1.5.final git 6a2fc5e 2015-05-28
17:35:17Z 64bit
[DEBUG] sofia.c:9015 IP 62.4.6.227 Rejected by acl "domains". Falling back
to Digest auth.
[DEBUG] switch_core_state_machine.c:491
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
State NEW
[DEBUG] switch_core_session.c:1061 Send signal sofia/profile_1/'o[BREAK]
[DEBUG] sofia.c:2065 detaching session 97bf779b-2e0d-e611-be46-00145ec01e69
[WARNING] switch_core_state_machine.c:572
97bf779b-2e0d-e611-be46-00145ec01e69 sofia/profile_1/'or''='@1.2.3.4>@nowhere
Abandoned
[NOTICE] switch_core_state_machine.c:575 Hangup sofia/profile_1/'o[CS_NEW]
[WRONG_CALL_STATE]
[DEBUG] switch_channel.c:3242 Send signal sofia/profile_1/'o[KILL]
[DEBUG] switch_core_session.c:1396 Send signal sofia/profile_1/'o[BREAK]
[DEBUG] switch_core_state_machine.c:472
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
Running State Change CS_HANGUP
[DEBUG] switch_core_state_machine.c:735
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
Callstate Change DOWN -> HANGUP
[DEBUG] switch_core_state_machine.c:737
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
State HANGUP
[DEBUG] mod_sofia.c:413 Channel sofia/profile_1/'or''='@1.2.3.4>@nowhere
hanging up, cause: WRONG_CALL_STATE
[DEBUG] switch_core_state_machine.c:60 sofia/profile_1/'or''='@1.2.3.4>@nowhere
Standard HANGUP, cause: WRONG_CALL_STATE
[DEBUG] switch_core_state_machine.c:737
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
State HANGUP going to sleep
[DEBUG] switch_core_state_machine.c:504
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
State Change CS_HANGUP -> CS_REPORTING
[DEBUG] switch_core_session.c:1396 Send signal sofia/profile_1/'o[BREAK]
[DEBUG] switch_core_state_machine.c:472
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
Running State Change CS_REPORTING
[DEBUG] switch_core_state_machine.c:823
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
State REPORTING
[DEBUG] switch_core_state_machine.c:104
sofia/profile_1/'or''='@1.2.3.4>@nowhere
Standard REPORTING, cause: WRONG_CALL_STATE
[DEBUG] switch_core_state_machine.c:823
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
State REPORTING going to sleep
[DEBUG] switch_core_state_machine.c:498
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
State Change CS_REPORTING -> CS_DESTROY
[DEBUG] switch_core_session.c:1396 Send signal sofia/profile_1/'o[BREAK]
[DEBUG] switch_core_session.c:1623 Session 139 (sofia/profile_1/'or''='@
1.2.3.4>@nowhere) Locked, Waiting on external entities
[NOTICE] switch_core_session.c:1641 Session 139 (sofia/profile_1/'or''='@
1.2.3.4>@nowhere) Ended
[NOTICE] switch_core_session.c:1645 Close Channel
sofia/profile_1/'o[CS_DESTROY]
[DEBUG] switch_core_state_machine.c:626
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
Running State Change CS_DESTROY
[DEBUG] switch_core_state_machine.c:636
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
State DESTROY
[DEBUG] mod_sofia.c:323 sofia/profile_1/'or''='@1.2.3.4>@nowhere SOFIA
DESTROY
[DEBUG] switch_core_state_machine.c:111
sofia/profile_1/'or''='@1.2.3.4>@nowhere
Standard DESTROY
[DEBUG] switch_core_state_machine.c:636
(sofia/profile_1/'or''='@1.2.3.4>@nowhere)
State DESTROY going to sleep


Is there a Event (or another way), to know when someone is trying to make a
call without register, so i can count how many times some IP try to do this
and add it in a firewall.
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From s.safarov at gmail.com  Thu Apr 28 15:47:32 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Thu, 28 Apr 2016 11:47:32 +0000
Subject: [Freeswitch-users] Custom Events
In-Reply-To: <CAKy26iO6FAQXnnDh5BXt7n3+38XQ3Fmw9h70cGQerEpvdqg=OQ@mail.gmail.com>
References: <CAKy26iO6FAQXnnDh5BXt7n3+38XQ3Fmw9h70cGQerEpvdqg=OQ@mail.gmail.com>
Message-ID: <CAHtxdDfkSni_9Hx1o0f4g3L+s-ym-6f3w2+i1A8Umx5o1qD00Q@mail.gmail.com>

mod_fail2ban <https://freeswitch.org/jira/browse/FS-7394> module is that
you want
Also look at PR771
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/771/overview>


??, 28 ???. 2016 ?. ? 14:40, Luiz Fernando Softov <fernando at softov.com.br>:

> Hi, sometimes i receive these users trying to make a call (injection) in
> my FreeSwitch
>
> I already use events to know what's happen in the FS.
>
> sofia::register_failure - to figure a invalid register
> sofia::gateway_state - when a gateway comes down
> sofia::gateway_add - when someone add a gateway (daemon)
> sofia::gateway_delete - when someone delete a gateway (daemon)
>
> [NOTICE] switch_channel.c:1075 New Channel
> sofia/profile_1/'o[97bf779b-2e0d-e611-be46-00145ec01e69]
> [DEBUG] switch_core_session.c:1061 Send signal sofia/profile_1/'o[BREAK]
> [DEBUG] switch_core_session.c:1061 Send signal sofia/profile_1/'o[BREAK]
> [DEBUG] switch_core_state_machine.c:472 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> Running State Change CS_NEW
> [DEBUG] sofia.c:8848 sofia/profile_1/'or''='@1.2.3.4>@nowhere receiving
> invite from 62.4.6.227:5076 version: 1.5.final git 6a2fc5e 2015-05-28
> 17:35:17Z 64bit
> [DEBUG] sofia.c:9015 IP 62.4.6.227 Rejected by acl "domains". Falling back
> to Digest auth.
> [DEBUG] switch_core_state_machine.c:491 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> State NEW
> [DEBUG] switch_core_session.c:1061 Send signal sofia/profile_1/'o[BREAK]
> [DEBUG] sofia.c:2065 detaching session 97bf779b-2e0d-e611-be46-00145ec01e69
> [WARNING] switch_core_state_machine.c:572
> 97bf779b-2e0d-e611-be46-00145ec01e69 sofia/profile_1/'or''='@1.2.3.4>@nowhere
> Abandoned
> [NOTICE] switch_core_state_machine.c:575 Hangup sofia/profile_1/'o[CS_NEW]
> [WRONG_CALL_STATE]
> [DEBUG] switch_channel.c:3242 Send signal sofia/profile_1/'o[KILL]
> [DEBUG] switch_core_session.c:1396 Send signal sofia/profile_1/'o[BREAK]
> [DEBUG] switch_core_state_machine.c:472 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> Running State Change CS_HANGUP
> [DEBUG] switch_core_state_machine.c:735 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> Callstate Change DOWN -> HANGUP
> [DEBUG] switch_core_state_machine.c:737 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> State HANGUP
> [DEBUG] mod_sofia.c:413 Channel sofia/profile_1/'or''='@1.2.3.4>@nowhere
> hanging up, cause: WRONG_CALL_STATE
> [DEBUG] switch_core_state_machine.c:60 sofia/profile_1/'or''='@1.2.3.4>@nowhere
> Standard HANGUP, cause: WRONG_CALL_STATE
> [DEBUG] switch_core_state_machine.c:737 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> State HANGUP going to sleep
> [DEBUG] switch_core_state_machine.c:504 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> State Change CS_HANGUP -> CS_REPORTING
> [DEBUG] switch_core_session.c:1396 Send signal sofia/profile_1/'o[BREAK]
> [DEBUG] switch_core_state_machine.c:472 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> Running State Change CS_REPORTING
> [DEBUG] switch_core_state_machine.c:823 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> State REPORTING
> [DEBUG] switch_core_state_machine.c:104 sofia/profile_1/'or''='@1.2.3.4>@nowhere
> Standard REPORTING, cause: WRONG_CALL_STATE
> [DEBUG] switch_core_state_machine.c:823 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> State REPORTING going to sleep
> [DEBUG] switch_core_state_machine.c:498 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> State Change CS_REPORTING -> CS_DESTROY
> [DEBUG] switch_core_session.c:1396 Send signal sofia/profile_1/'o[BREAK]
> [DEBUG] switch_core_session.c:1623 Session 139 (sofia/profile_1/'or''='@
> 1.2.3.4>@nowhere) Locked, Waiting on external entities
> [NOTICE] switch_core_session.c:1641 Session 139 (sofia/profile_1/'or''='@
> 1.2.3.4>@nowhere) Ended
> [NOTICE] switch_core_session.c:1645 Close Channel
> sofia/profile_1/'o[CS_DESTROY]
> [DEBUG] switch_core_state_machine.c:626 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> Running State Change CS_DESTROY
> [DEBUG] switch_core_state_machine.c:636 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> State DESTROY
> [DEBUG] mod_sofia.c:323 sofia/profile_1/'or''='@1.2.3.4>@nowhere SOFIA
> DESTROY
> [DEBUG] switch_core_state_machine.c:111 sofia/profile_1/'or''='@1.2.3.4>@nowhere
> Standard DESTROY
> [DEBUG] switch_core_state_machine.c:636 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
> State DESTROY going to sleep
>
>
> Is there a Event (or another way), to know when someone is trying to make
> a call without register, so i can count how many times some IP try to do
> this and add it in a firewall.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From fernando at softov.com.br  Thu Apr 28 16:07:45 2016
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Thu, 28 Apr 2016 08:07:45 -0400
Subject: [Freeswitch-users] Custom Events
In-Reply-To: <CAHtxdDfkSni_9Hx1o0f4g3L+s-ym-6f3w2+i1A8Umx5o1qD00Q@mail.gmail.com>
References: <CAKy26iO6FAQXnnDh5BXt7n3+38XQ3Fmw9h70cGQerEpvdqg=OQ@mail.gmail.com>
	<CAHtxdDfkSni_9Hx1o0f4g3L+s-ym-6f3w2+i1A8Umx5o1qD00Q@mail.gmail.com>
Message-ID: <CAKy26iOm2jPnaN_b6HqeSSGOHp=v0JyHstHtZc69ML+MZYh+Vg@mail.gmail.com>

Thanks for the reply, but I don't wanna a module.

I just make a example. I want to receive a event and make some log, or
check a table to do something else, or make a alarm, etc.

I already see this module, i tried to find the source to see how to do it,
but the git is off.

Thanks for the links, i will check this. The PR771
<https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/771/overview>
is
in master.

If i want to clone again FreeSwitch, which one do you recommend?



2016-04-28 7:47 GMT-04:00 Sergey Safarov <s.safarov at gmail.com>:

> mod_fail2ban <https://freeswitch.org/jira/browse/FS-7394> module is that
> you want
> Also look at PR771
> <https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/771/overview>
>
>
> ??, 28 ???. 2016 ?. ? 14:40, Luiz Fernando Softov <fernando at softov.com.br
> >:
>
>> Hi, sometimes i receive these users trying to make a call (injection) in
>> my FreeSwitch
>>
>> I already use events to know what's happen in the FS.
>>
>> sofia::register_failure - to figure a invalid register
>> sofia::gateway_state - when a gateway comes down
>> sofia::gateway_add - when someone add a gateway (daemon)
>> sofia::gateway_delete - when someone delete a gateway (daemon)
>>
>> [NOTICE] switch_channel.c:1075 New Channel
>> sofia/profile_1/'o[97bf779b-2e0d-e611-be46-00145ec01e69]
>> [DEBUG] switch_core_session.c:1061 Send signal sofia/profile_1/'o[BREAK]
>> [DEBUG] switch_core_session.c:1061 Send signal sofia/profile_1/'o[BREAK]
>> [DEBUG] switch_core_state_machine.c:472 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> Running State Change CS_NEW
>> [DEBUG] sofia.c:8848 sofia/profile_1/'or''='@1.2.3.4>@nowhere receiving
>> invite from 62.4.6.227:5076 version: 1.5.final git 6a2fc5e 2015-05-28
>> 17:35:17Z 64bit
>> [DEBUG] sofia.c:9015 IP 62.4.6.227 Rejected by acl "domains". Falling
>> back to Digest auth.
>> [DEBUG] switch_core_state_machine.c:491 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> State NEW
>> [DEBUG] switch_core_session.c:1061 Send signal sofia/profile_1/'o[BREAK]
>> [DEBUG] sofia.c:2065 detaching session
>> 97bf779b-2e0d-e611-be46-00145ec01e69
>> [WARNING] switch_core_state_machine.c:572
>> 97bf779b-2e0d-e611-be46-00145ec01e69 sofia/profile_1/'or''='@1.2.3.4>@nowhere
>> Abandoned
>> [NOTICE] switch_core_state_machine.c:575 Hangup
>> sofia/profile_1/'o[CS_NEW] [WRONG_CALL_STATE]
>> [DEBUG] switch_channel.c:3242 Send signal sofia/profile_1/'o[KILL]
>> [DEBUG] switch_core_session.c:1396 Send signal sofia/profile_1/'o[BREAK]
>> [DEBUG] switch_core_state_machine.c:472 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> Running State Change CS_HANGUP
>> [DEBUG] switch_core_state_machine.c:735 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> Callstate Change DOWN -> HANGUP
>> [DEBUG] switch_core_state_machine.c:737 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> State HANGUP
>> [DEBUG] mod_sofia.c:413 Channel sofia/profile_1/'or''='@1.2.3.4>@nowhere
>> hanging up, cause: WRONG_CALL_STATE
>> [DEBUG] switch_core_state_machine.c:60 sofia/profile_1/'or''='@1.2.3.4>@nowhere
>> Standard HANGUP, cause: WRONG_CALL_STATE
>> [DEBUG] switch_core_state_machine.c:737 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> State HANGUP going to sleep
>> [DEBUG] switch_core_state_machine.c:504 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> State Change CS_HANGUP -> CS_REPORTING
>> [DEBUG] switch_core_session.c:1396 Send signal sofia/profile_1/'o[BREAK]
>> [DEBUG] switch_core_state_machine.c:472 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> Running State Change CS_REPORTING
>> [DEBUG] switch_core_state_machine.c:823 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> State REPORTING
>> [DEBUG] switch_core_state_machine.c:104 sofia/profile_1/'or''='@1.2.3.4>@nowhere
>> Standard REPORTING, cause: WRONG_CALL_STATE
>> [DEBUG] switch_core_state_machine.c:823 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> State REPORTING going to sleep
>> [DEBUG] switch_core_state_machine.c:498 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> State Change CS_REPORTING -> CS_DESTROY
>> [DEBUG] switch_core_session.c:1396 Send signal sofia/profile_1/'o[BREAK]
>> [DEBUG] switch_core_session.c:1623 Session 139 (sofia/profile_1/'or''='@
>> 1.2.3.4>@nowhere) Locked, Waiting on external entities
>> [NOTICE] switch_core_session.c:1641 Session 139 (sofia/profile_1/'or''='@
>> 1.2.3.4>@nowhere) Ended
>> [NOTICE] switch_core_session.c:1645 Close Channel
>> sofia/profile_1/'o[CS_DESTROY]
>> [DEBUG] switch_core_state_machine.c:626 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> Running State Change CS_DESTROY
>> [DEBUG] switch_core_state_machine.c:636 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> State DESTROY
>> [DEBUG] mod_sofia.c:323 sofia/profile_1/'or''='@1.2.3.4>@nowhere SOFIA
>> DESTROY
>> [DEBUG] switch_core_state_machine.c:111 sofia/profile_1/'or''='@1.2.3.4>@nowhere
>> Standard DESTROY
>> [DEBUG] switch_core_state_machine.c:636 (sofia/profile_1/'or''='@1.2.3.4>@nowhere)
>> State DESTROY going to sleep
>>
>>
>> Is there a Event (or another way), to know when someone is trying to make
>> a call without register, so i can count how many times some IP try to do
>> this and add it in a firewall.
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fernando at softov.com.br  Thu Apr 28 16:48:23 2016
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Thu, 28 Apr 2016 08:48:23 -0400
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
Message-ID: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>

Hi,

I don't remember why, but I'm using 1.5.final (branch master) in production
(~1 year), and it's working fine, with 30 simultaneous calls.

I will re-clone the FreeSwitch, because some changes made, that i need.

Someone recommend to clone another tag?
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From italo at freeswitch.org  Thu Apr 28 16:51:44 2016
From: italo at freeswitch.org (=?utf-8?q?=C3=8Dtalo_Rossi?=)
Date: Thu, 28 Apr 2016 05:51:44 -0700 (PDT)
Subject: [Freeswitch-users] Thursday FreeSWITCH Bug Hunt
Message-ID: <5swv6q7ccmx3p4o6b4cgd5g92-0@mailer.nylas.com>

FreeSWITCHers,

  

Join us TODAY 2PM CST for the Thursday FreeSWITCH Bug Hunt!

  

Where? [conference.freeswitch.org/vc/#/?autocall=888](https://conference.frees
witch.org/vc/#/?autocall=888
"https://conference.freeswitch.org/vc/#/?autocall=888" )

  

Chat? <https://hipchat.freeswitch.org/gYzPMg3Oc>

  

What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support!

  

Help us help you, Join the Bug Hunt!

?talo Rossi

italo at freeswitch.org

IRC chat.freenode.net #freeswitch #freeswitch-dev

Bugs? https://freeswitch.org/jira

Docs? https://freeswitch.org/jira

Chat? https://hipchat.freeswitch.org/gUdAgy0m6

  

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From krice at freeswitch.org  Thu Apr 28 17:09:00 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 Apr 2016 08:09:00 -0500
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
Message-ID: <00a101d1a14f$222407d0$666c1770$@freeswitch.org>

Latest master?. There are numerous bugs and potential security issues that have been resolved since then.

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 7:48 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG

 

Hi,

I don't remember why, but I'm using 1.5.final (branch master) in production (~1 year), and it's working fine, with 30 simultaneous calls.


I will re-clone the FreeSwitch, because some changes made, that i need.

Someone recommend to clone another tag?

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From fernando at softov.com.br  Thu Apr 28 17:19:55 2016
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Thu, 28 Apr 2016 09:19:55 -0400
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <00a101d1a14f$222407d0$666c1770$@freeswitch.org>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
Message-ID: <CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>

Yeah, latest master, but witch tag?

1.4.x
1.5.final
1.6.0
1.6.7
1.7.0

There are so many tags, so i have lost my mind witch one.

I already thy to compile 1.6 in FreeBSD, but without success, there are
dependencies in mod_opus, and another thing i can't remember now.

Which is the difference between tags?






2016-04-28 9:09 GMT-04:00 Ken Rice <krice at freeswitch.org>:

> Latest master?. There are numerous bugs and potential security issues that
> have been resolved since then.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 7:48 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> Hi,
>
> I don't remember why, but I'm using 1.5.final (branch master) in
> production (~1 year), and it's working fine, with 30 simultaneous calls.
>
>
> I will re-clone the FreeSwitch, because some changes made, that i need.
>
> Someone recommend to clone another tag?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Thu Apr 28 17:31:32 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 Apr 2016 08:31:32 -0500
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
Message-ID: <00ad01d1a152$4826af20$d8740d60$@freeswitch.org>

There is no tag for latest master. Its just master? git checkout master && git pull?

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 8:20 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSwitch - Clone - TAG

 

Yeah, latest master, but witch tag?

1.4.x
1.5.final
1.6.0
1.6.7
1.7.0

There are so many tags, so i have lost my mind witch one.


I already thy to compile 1.6 in FreeBSD, but without success, there are dependencies in mod_opus, and another thing i can't remember now.

Which is the difference between tags?






 

2016-04-28 9:09 GMT-04:00 Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >:

Latest master?. There are numerous bugs and potential security issues that have been resolved since then.

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 7:48 AM
To: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> 
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG

 

Hi,

I don't remember why, but I'm using 1.5.final (branch master) in production (~1 year), and it's working fine, with 30 simultaneous calls.


I will re-clone the FreeSwitch, because some changes made, that i need.

Someone recommend to clone another tag?


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

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

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

 

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From 35633 at heb.be  Thu Apr 28 17:39:09 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Thu, 28 Apr 2016 15:39:09 +0200
Subject: [Freeswitch-users] ESL.so
Message-ID: <CAFSzUtadEA6HO7Z++QWF4wajBu70ZF=nQcm1vYSJAEhwuuvw0A@mail.gmail.com>

Hello, i m having an error when i start php5.
The error is :
php warnign: php startup: Unable to laod dynamic library
'/usr/lib/php5/20131266/ESL.so': cannot open shared object file: No such
file or directory

Any help?

NDUWAYEZU Joselyne

-- 
Haute ?cole de Bruxelles
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From krice at freeswitch.org  Thu Apr 28 17:41:03 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 Apr 2016 08:41:03 -0500
Subject: [Freeswitch-users] ESL.so
In-Reply-To: <CAFSzUtadEA6HO7Z++QWF4wajBu70ZF=nQcm1vYSJAEhwuuvw0A@mail.gmail.com>
References: <CAFSzUtadEA6HO7Z++QWF4wajBu70ZF=nQcm1vYSJAEhwuuvw0A@mail.gmail.com>
Message-ID: <00b701d1a153$9c7b9f80$d572de80$@freeswitch.org>

Its exactly what it says, it cant open the ESL.so because it doesn?t exist? make sure you installed the .so into the correct location

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nduwayezu, Joselyne
Sent: Thursday, April 28, 2016 8:39 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] ESL.so

 

 

Hello, i m having an error when i start php5.

The error is :

php warnign: php startup: Unable to laod dynamic library '/usr/lib/php5/20131266/ESL.so': cannot open shared object file: No such file or directory 

 

Any help?




NDUWAYEZU Joselyne


Haute ?cole de Bruxelles

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From krice at freeswitch.org  Thu Apr 28 19:21:43 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 Apr 2016 10:21:43 -0500
Subject: [Freeswitch-users] Fw: new message
In-Reply-To: <CAHtxdDeH0u15SW4y60AjG=fY8K=baH9QvRsnnSnmr+kUcwYNqQ@mail.gmail.com>
References: <0000f01c6f25$b84c3218$9c8c0a6a$@lodden.com>
	<CAHtxdDeH0u15SW4y60AjG=fY8K=baH9QvRsnnSnmr+kUcwYNqQ@mail.gmail.com>
Message-ID: <021701d1a161$acac55d0$06050170$@freeswitch.org>

Unfortunately it appears he got hit with an attack. This has been happening lately. In the event this happens again, please don?t quote the message as it only helps propogate it.

 

We are actively monitoring the list for this sort of thing and will moderate users to help keep it at bay.

 

K

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov
Sent: Thursday, April 28, 2016 12:43 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>; Freedom Generator <discrimination at wbobbyd.top>; Freedom Generator <gluttonize at familyix.top>; FreedomGenerator <Off-GridSystem at electriic.info>; Freida Stout <greekx at wordiverse.com>; Fresh Start Tax <FreshStartTax at 2iteemi.top>
Subject: Re: [Freeswitch-users] Fw: new message

 

Probable fishing message. 

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From fernando at softov.com.br  Thu Apr 28 20:28:39 2016
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Thu, 28 Apr 2016 12:28:39 -0400
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
Message-ID: <CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>

I cloned and trying to compile now, in FreeBSD

making all mod_opus
gmake[4]: Entering directory
'/usr/dev/freeswitch_backend/freeswitch/src/mod/codecs/mod_opus'
Makefile:897: *** You must install libopus-dev to build mod_opus.  Stop.

I already compile and install opus from downloads.xiph.org/releases/opus/

since in FreeBSD doesn't have a libopus-dev

Is there another place to get libopus?



2016-04-28 9:31 GMT-04:00 Ken Rice <krice at freeswitch.org>:

> There is no tag for latest master. Its just master? git checkout master &&
> git pull?
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 8:20 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> Yeah, latest master, but witch tag?
>
> 1.4.x
> 1.5.final
> 1.6.0
> 1.6.7
> 1.7.0
>
> There are so many tags, so i have lost my mind witch one.
>
>
> I already thy to compile 1.6 in FreeBSD, but without success, there are
> dependencies in mod_opus, and another thing i can't remember now.
>
> Which is the difference between tags?
>
>
>
>
>
>
> 2016-04-28 9:09 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>
> Latest master?. There are numerous bugs and potential security issues that
> have been resolved since then.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 7:48 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> Hi,
>
> I don't remember why, but I'm using 1.5.final (branch master) in
> production (~1 year), and it's working fine, with 30 simultaneous calls.
>
>
> I will re-clone the FreeSwitch, because some changes made, that i need.
>
> Someone recommend to clone another tag?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Thu Apr 28 20:45:57 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 Apr 2016 11:45:57 -0500
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
Message-ID: <026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>

https://freeswitch.org/stash/projects/SD/repos/opus/browse

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 11:29 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSwitch - Clone - TAG

 

I cloned and trying to compile now, in FreeBSD

making all mod_opus

gmake[4]: Entering directory '/usr/dev/freeswitch_backend/freeswitch/src/mod/codecs/mod_opus'

Makefile:897: *** You must install libopus-dev to build mod_opus.  Stop.

I already compile and install opus from downloads.xiph.org/releases/opus/ <http://downloads.xiph.org/releases/opus/> 

since in FreeBSD doesn't have a libopus-dev

Is there another place to get libopus?



 

2016-04-28 9:31 GMT-04:00 Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >:

There is no tag for latest master. Its just master? git checkout master && git pull?

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 8:20 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] FreeSwitch - Clone - TAG

 

Yeah, latest master, but witch tag?

1.4.x
1.5.final
1.6.0
1.6.7
1.7.0

There are so many tags, so i have lost my mind witch one.


I already thy to compile 1.6 in FreeBSD, but without success, there are dependencies in mod_opus, and another thing i can't remember now.

Which is the difference between tags?





 

2016-04-28 9:09 GMT-04:00 Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >:

Latest master?. There are numerous bugs and potential security issues that have been resolved since then.

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 7:48 AM
To: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> 
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG

 

Hi,

I don't remember why, but I'm using 1.5.final (branch master) in production (~1 year), and it's working fine, with 30 simultaneous calls.


I will re-clone the FreeSwitch, because some changes made, that i need.

Someone recommend to clone another tag?


_________________________________________________________________________
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

 

-------------- next part --------------
An HTML attachment was scrubbed...
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From fernando at softov.com.br  Thu Apr 28 21:09:32 2016
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Thu, 28 Apr 2016 13:09:32 -0400
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
	<026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>
Message-ID: <CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>

It's the same with

making all mod_shout
gmake[4]: Entering directory
'/usr/dev/freeswitch_backend/freeswitch/src/mod/formats/mod_shout'
Makefile:907: *** You must install libshout3-dev to build mod_shout.  Stop.

First I install libshout from ports, reclone, bootstrap, configure e when
make no effect
Then i download from internet, and its the same thing.

I think this will be the same with libmpg123.

About the link, i only need to clone it and make install clean, or need to
clone it in a correct directory?
Like
/usr/dev/freeswitch_backend/freeswitch/libs/opus
/usr/dev/freeswitch_backend/freeswitch/libs/opus-1.1

????





2016-04-28 12:45 GMT-04:00 Ken Rice <krice at freeswitch.org>:

> https://freeswitch.org/stash/projects/SD/repos/opus/browse
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 11:29 AM
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> I cloned and trying to compile now, in FreeBSD
>
> making all mod_opus
>
> gmake[4]: Entering directory
> '/usr/dev/freeswitch_backend/freeswitch/src/mod/codecs/mod_opus'
>
> Makefile:897: *** You must install libopus-dev to build mod_opus.  Stop.
>
> I already compile and install opus from downloads.xiph.org/releases/opus/
>
> since in FreeBSD doesn't have a libopus-dev
>
> Is there another place to get libopus?
>
>
>
> 2016-04-28 9:31 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>
> There is no tag for latest master. Its just master? git checkout master &&
> git pull?
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 8:20 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> Yeah, latest master, but witch tag?
>
> 1.4.x
> 1.5.final
> 1.6.0
> 1.6.7
> 1.7.0
>
> There are so many tags, so i have lost my mind witch one.
>
>
> I already thy to compile 1.6 in FreeBSD, but without success, there are
> dependencies in mod_opus, and another thing i can't remember now.
>
> Which is the difference between tags?
>
>
>
>
>
> 2016-04-28 9:09 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>
> Latest master?. There are numerous bugs and potential security issues that
> have been resolved since then.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 7:48 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> Hi,
>
> I don't remember why, but I'm using 1.5.final (branch master) in
> production (~1 year), and it's working fine, with 30 simultaneous calls.
>
>
> I will re-clone the FreeSwitch, because some changes made, that i need.
>
> Someone recommend to clone another tag?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From krice at freeswitch.org  Thu Apr 28 21:12:18 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 Apr 2016 12:12:18 -0500
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>	<026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>
	<CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>
Message-ID: <029601d1a171$1f5a6810$5e0f3830$@freeswitch.org>

We don?t not provide anything that does MP3 due to patent issues?

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 12:10 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSwitch - Clone - TAG

 

It's the same with

making all mod_shout

gmake[4]: Entering directory '/usr/dev/freeswitch_backend/freeswitch/src/mod/formats/mod_shout'

Makefile:907: *** You must install libshout3-dev to build mod_shout.  Stop.

First I install libshout from ports, reclone, bootstrap, configure e when make no effect

Then i download from internet, and its the same thing.

I think this will be the same with libmpg123.

About the link, i only need to clone it and make install clean, or need to clone it in a correct directory?
Like
/usr/dev/freeswitch_backend/freeswitch/libs/opus
/usr/dev/freeswitch_backend/freeswitch/libs/opus-1.1

????





 

2016-04-28 12:45 GMT-04:00 Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >:

https://freeswitch.org/stash/projects/SD/repos/opus/browse

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 11:29 AM


To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] FreeSwitch - Clone - TAG

 

I cloned and trying to compile now, in FreeBSD

making all mod_opus

gmake[4]: Entering directory '/usr/dev/freeswitch_backend/freeswitch/src/mod/codecs/mod_opus'

Makefile:897: *** You must install libopus-dev to build mod_opus.  Stop.

I already compile and install opus from downloads.xiph.org/releases/opus/ <http://downloads.xiph.org/releases/opus/> 

since in FreeBSD doesn't have a libopus-dev

Is there another place to get libopus?

 

2016-04-28 9:31 GMT-04:00 Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >:

There is no tag for latest master. Its just master? git checkout master && git pull?

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 8:20 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] FreeSwitch - Clone - TAG

 

Yeah, latest master, but witch tag?

1.4.x
1.5.final
1.6.0
1.6.7
1.7.0

There are so many tags, so i have lost my mind witch one.


I already thy to compile 1.6 in FreeBSD, but without success, there are dependencies in mod_opus, and another thing i can't remember now.

Which is the difference between tags?




 

2016-04-28 9:09 GMT-04:00 Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >:

Latest master?. There are numerous bugs and potential security issues that have been resolved since then.

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 7:48 AM
To: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> 
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG

 

Hi,

I don't remember why, but I'm using 1.5.final (branch master) in production (~1 year), and it's working fine, with 30 simultaneous calls.


I will re-clone the FreeSwitch, because some changes made, that i need.

Someone recommend to clone another tag?


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

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

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

 


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

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

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

 


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

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

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

 

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From fernando at softov.com.br  Thu Apr 28 21:26:50 2016
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Thu, 28 Apr 2016 13:26:50 -0400
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <029601d1a171$1f5a6810$5e0f3830$@freeswitch.org>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
	<026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>
	<CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>
	<029601d1a171$1f5a6810$5e0f3830$@freeswitch.org>
Message-ID: <CAKy26iMSooc58roPi9gwx+R3NetRy6YBVS28SJmid8GSvR=YEw@mail.gmail.com>

Yeah.

I know now why I'm using 1.5.final, there i just compile without problems.

About the link, i only need to clone it and make install clean, or need to
clone it in a correct directory?
/usr/dev/freeswitch_backend/freeswitch/libs/opus
/usr/dev/freeswitch_backend/freeswitch/libs/opus-1.1

And the another libraries dependencies, where i can put these libraries?
Since only make; make install don't have effect.

I always get these errors

Makefile:...: *** You must install XXX to build mod_yyyyyyy.  Stop.




2016-04-28 13:12 GMT-04:00 Ken Rice <krice at freeswitch.org>:

> We don?t not provide anything that does MP3 due to patent issues?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 12:10 PM
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> It's the same with
>
> making all mod_shout
>
> gmake[4]: Entering directory
> '/usr/dev/freeswitch_backend/freeswitch/src/mod/formats/mod_shout'
>
> Makefile:907: *** You must install libshout3-dev to build mod_shout.  Stop.
>
> First I install libshout from ports, reclone, bootstrap, configure e when
> make no effect
>
> Then i download from internet, and its the same thing.
>
> I think this will be the same with libmpg123.
>
> About the link, i only need to clone it and make install clean, or need to
> clone it in a correct directory?
> Like
> /usr/dev/freeswitch_backend/freeswitch/libs/opus
> /usr/dev/freeswitch_backend/freeswitch/libs/opus-1.1
>
> ????
>
>
>
>
>
> 2016-04-28 12:45 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>
> https://freeswitch.org/stash/projects/SD/repos/opus/browse
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 11:29 AM
>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> I cloned and trying to compile now, in FreeBSD
>
> making all mod_opus
>
> gmake[4]: Entering directory
> '/usr/dev/freeswitch_backend/freeswitch/src/mod/codecs/mod_opus'
>
> Makefile:897: *** You must install libopus-dev to build mod_opus.  Stop.
>
> I already compile and install opus from downloads.xiph.org/releases/opus/
>
> since in FreeBSD doesn't have a libopus-dev
>
> Is there another place to get libopus?
>
>
>
> 2016-04-28 9:31 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>
> There is no tag for latest master. Its just master? git checkout master &&
> git pull?
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 8:20 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> Yeah, latest master, but witch tag?
>
> 1.4.x
> 1.5.final
> 1.6.0
> 1.6.7
> 1.7.0
>
> There are so many tags, so i have lost my mind witch one.
>
>
> I already thy to compile 1.6 in FreeBSD, but without success, there are
> dependencies in mod_opus, and another thing i can't remember now.
>
> Which is the difference between tags?
>
>
>
>
> 2016-04-28 9:09 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>
> Latest master?. There are numerous bugs and potential security issues that
> have been resolved since then.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 7:48 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> Hi,
>
> I don't remember why, but I'm using 1.5.final (branch master) in
> production (~1 year), and it's working fine, with 30 simultaneous calls.
>
>
> I will re-clone the FreeSwitch, because some changes made, that i need.
>
> Someone recommend to clone another tag?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Apr 28 21:31:00 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Thu, 28 Apr 2016 10:31:00 -0700
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
Message-ID: <CAKE2PDuLrgMu6Y82cMHnKDr1P2m=iz-_r-2gO91d4v_KVifDQQ@mail.gmail.com>

On 28 April 2016 at 09:28, Luiz Fernando Softov <fernando at softov.com.br> wrote:
> Is there another place to get libopus?


Check this page out and see if it helps:
https://freeswitch.org/confluence/display/FREESWITCH/FreeBSD

What FreeBSD version are you using?

Thanks,
Sean



-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From krice at freeswitch.org  Thu Apr 28 21:34:48 2016
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 Apr 2016 12:34:48 -0500
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <CAKy26iMSooc58roPi9gwx+R3NetRy6YBVS28SJmid8GSvR=YEw@mail.gmail.com>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>	<026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>	<CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>	<029601d1a171$1f5a6810$5e0f3830$@freeswitch.org>
	<CAKy26iMSooc58roPi9gwx+R3NetRy6YBVS28SJmid8GSvR=YEw@mail.gmail.com>
Message-ID: <02b601d1a174$43b21e80$cb165b80$@freeswitch.org>

1.5 is long deprecated and no longer supported? the entire 1.5 branch was never even a release.

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 12:27 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSwitch - Clone - TAG

 

Yeah.

I know now why I'm using 1.5.final, there i just compile without problems.

About the link, i only need to clone it and make install clean, or need to clone it in a correct directory?
/usr/dev/freeswitch_backend/freeswitch/libs/opus
/usr/dev/freeswitch_backend/freeswitch/libs/opus-1.1

And the another libraries dependencies, where i can put these libraries? Since only make; make install don't have effect.

I always get these errors

Makefile:...: *** You must install XXX to build mod_yyyyyyy.  Stop.




 

2016-04-28 13:12 GMT-04:00 Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >:

We don?t not provide anything that does MP3 due to patent issues?

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 12:10 PM


To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] FreeSwitch - Clone - TAG

 

It's the same with

making all mod_shout

gmake[4]: Entering directory '/usr/dev/freeswitch_backend/freeswitch/src/mod/formats/mod_shout'

Makefile:907: *** You must install libshout3-dev to build mod_shout.  Stop.

First I install libshout from ports, reclone, bootstrap, configure e when make no effect

Then i download from internet, and its the same thing.

I think this will be the same with libmpg123.

About the link, i only need to clone it and make install clean, or need to clone it in a correct directory?
Like
/usr/dev/freeswitch_backend/freeswitch/libs/opus
/usr/dev/freeswitch_backend/freeswitch/libs/opus-1.1

????




 

2016-04-28 12:45 GMT-04:00 Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >:

https://freeswitch.org/stash/projects/SD/repos/opus/browse

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 11:29 AM


To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] FreeSwitch - Clone - TAG

 

I cloned and trying to compile now, in FreeBSD

making all mod_opus

gmake[4]: Entering directory '/usr/dev/freeswitch_backend/freeswitch/src/mod/codecs/mod_opus'

Makefile:897: *** You must install libopus-dev to build mod_opus.  Stop.

I already compile and install opus from downloads.xiph.org/releases/opus/ <http://downloads.xiph.org/releases/opus/> 

since in FreeBSD doesn't have a libopus-dev

Is there another place to get libopus?

 

2016-04-28 9:31 GMT-04:00 Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >:

There is no tag for latest master. Its just master? git checkout master && git pull?

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 8:20 AM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> >
Subject: Re: [Freeswitch-users] FreeSwitch - Clone - TAG

 

Yeah, latest master, but witch tag?

1.4.x
1.5.final
1.6.0
1.6.7
1.7.0

There are so many tags, so i have lost my mind witch one.


I already thy to compile 1.6 in FreeBSD, but without success, there are dependencies in mod_opus, and another thing i can't remember now.

Which is the difference between tags?



 

2016-04-28 9:09 GMT-04:00 Ken Rice <krice at freeswitch.org <mailto:krice at freeswitch.org> >:

Latest master?. There are numerous bugs and potential security issues that have been resolved since then.

 

From: freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto:freeswitch-users-bounces at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org> ] On Behalf Of Luiz Fernando Softov
Sent: Thursday, April 28, 2016 7:48 AM
To: freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org> 
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG

 

Hi,

I don't remember why, but I'm using 1.5.final (branch master) in production (~1 year), and it's working fine, with 30 simultaneous calls.


I will re-clone the FreeSwitch, because some changes made, that i need.

Someone recommend to clone another tag?


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

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

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

 


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

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

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

 


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

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

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

 


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

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

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

 

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From fernando at softov.com.br  Thu Apr 28 22:10:40 2016
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Thu, 28 Apr 2016 14:10:40 -0400
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <02b601d1a174$43b21e80$cb165b80$@freeswitch.org>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
	<026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>
	<CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>
	<029601d1a171$1f5a6810$5e0f3830$@freeswitch.org>
	<CAKy26iMSooc58roPi9gwx+R3NetRy6YBVS28SJmid8GSvR=YEw@mail.gmail.com>
	<02b601d1a174$43b21e80$cb165b80$@freeswitch.org>
Message-ID: <CAKy26iPJaTC3Wu9pCBX9FbnMTbJ_KaFF6iSOJ=55aN6WVuWeUw@mail.gmail.com>

I am using 10.1

In the same machine that i have already compiled FreeSwitch.

Now i just cloned the master.

But when i tried to compile, i have these problems.

I will go to sleep. I'm wake almost ~26 hours.

Then i will make a fresh install, everything, FreeBSD and FreeSwitch, and
activate the modules (that i need and isn't default active).

Thanks for the help.



2016-04-28 13:34 GMT-04:00 Ken Rice <krice at freeswitch.org>:

> 1.5 is long deprecated and no longer supported? the entire 1.5 branch was
> never even a release.
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 12:27 PM
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> Yeah.
>
> I know now why I'm using 1.5.final, there i just compile without problems.
>
> About the link, i only need to clone it and make install clean, or need to
> clone it in a correct directory?
> /usr/dev/freeswitch_backend/freeswitch/libs/opus
> /usr/dev/freeswitch_backend/freeswitch/libs/opus-1.1
>
> And the another libraries dependencies, where i can put these libraries?
> Since only make; make install don't have effect.
>
> I always get these errors
>
> Makefile:...: *** You must install XXX to build mod_yyyyyyy.  Stop.
>
>
>
>
> 2016-04-28 13:12 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>
> We don?t not provide anything that does MP3 due to patent issues?
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 12:10 PM
>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> It's the same with
>
> making all mod_shout
>
> gmake[4]: Entering directory
> '/usr/dev/freeswitch_backend/freeswitch/src/mod/formats/mod_shout'
>
> Makefile:907: *** You must install libshout3-dev to build mod_shout.  Stop.
>
> First I install libshout from ports, reclone, bootstrap, configure e when
> make no effect
>
> Then i download from internet, and its the same thing.
>
> I think this will be the same with libmpg123.
>
> About the link, i only need to clone it and make install clean, or need to
> clone it in a correct directory?
> Like
> /usr/dev/freeswitch_backend/freeswitch/libs/opus
> /usr/dev/freeswitch_backend/freeswitch/libs/opus-1.1
>
> ????
>
>
>
>
> 2016-04-28 12:45 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>
> https://freeswitch.org/stash/projects/SD/repos/opus/browse
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 11:29 AM
>
>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> I cloned and trying to compile now, in FreeBSD
>
> making all mod_opus
>
> gmake[4]: Entering directory
> '/usr/dev/freeswitch_backend/freeswitch/src/mod/codecs/mod_opus'
>
> Makefile:897: *** You must install libopus-dev to build mod_opus.  Stop.
>
> I already compile and install opus from downloads.xiph.org/releases/opus/
>
> since in FreeBSD doesn't have a libopus-dev
>
> Is there another place to get libopus?
>
>
>
> 2016-04-28 9:31 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>
> There is no tag for latest master. Its just master? git checkout master &&
> git pull?
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 8:20 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> Yeah, latest master, but witch tag?
>
> 1.4.x
> 1.5.final
> 1.6.0
> 1.6.7
> 1.7.0
>
> There are so many tags, so i have lost my mind witch one.
>
>
> I already thy to compile 1.6 in FreeBSD, but without success, there are
> dependencies in mod_opus, and another thing i can't remember now.
>
> Which is the difference between tags?
>
>
>
> 2016-04-28 9:09 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>
> Latest master?. There are numerous bugs and potential security issues that
> have been resolved since then.
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
> Fernando Softov
> *Sent:* Thursday, April 28, 2016 7:48 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] FreeSwitch - Clone - TAG
>
>
>
> Hi,
>
> I don't remember why, but I'm using 1.5.final (branch master) in
> production (~1 year), and it's working fine, with 30 simultaneous calls.
>
>
> I will re-clone the FreeSwitch, because some changes made, that i need.
>
> Someone recommend to clone another tag?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From mike at jerris.com  Thu Apr 28 22:15:00 2016
From: mike at jerris.com (Michael Jerris)
Date: Thu, 28 Apr 2016 14:15:00 -0400
Subject: [Freeswitch-users] High load kernel
In-Reply-To: <CAKThY04sqzToxwHHcuczwCKfj5DSC81KnFTboK2h1X5-Zu8YFA@mail.gmail.com>
References: <CAKThY07PwnMuw-hy67zG+3wSo=FLaOfi+WTpB2Hj_jKhbijzxA@mail.gmail.com>
	<CALXCt0rePzPJPmtJMLAHsD=vuoHhooDQTzOgYNj5CMB0c16jzg@mail.gmail.com>
	<CAKThY04sqzToxwHHcuczwCKfj5DSC81KnFTboK2h1X5-Zu8YFA@mail.gmail.com>
Message-ID: <90331067-44BB-4109-9866-C9538730C145@jerris.com>

Strongly advise to be using latest release in 1.6, particularly for anything webrtc related.
Strongly advice to use Debian 8, there were significant known performance problems on CentOS 6

> On Apr 28, 2016, at 6:41 AM, Agust? Ubalde <aubalde at presenceco.com> wrote:
> 
> Which OS, FS revision, platform, hardware, ram, cpu, etc are you using?
> 
> 
> CentOS 6.5 64b virutalized (ESXi)
> Freeswitch 1.4.26
> 8vCPU - 4GB RAM
> 
> Which traffic are you handling?
> 
> 
> 60-70 active calls (WebRTC extensions)
> 
> What are exactly the simptoms?
> 
> 
> top command shows high system cpu load constantly (CPU 3)
> 
> How you get them? Is that reproducible?
> 
> 
> Always whit the same load (60-70 calls)
> 
> 
> Thanks!
> PRESENCE TECHNOLOGY
> Agust? Ubalde Bellot
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com <mailto:aubalde at presenceco.com>
> Ph: +34 93 10 10 300 
> Fx: +34 93 10 10 333
> 
> 
> www.presenceco.com <http://www.presenceco.com/>
> Follow us on:
> 
>  <http://twitter.com/presencetech> <http://www.youtube.com/PresenceTechnology> <http://www.linkedin.com/groups?mostPopular=&gid=2898558> <http://www.slideshare.net/PresenceTech> <http://www.facebook.com/presencetech>
> For additional information, please visit our website www.presenceco.com <http://www.presenceco.com/>
> 
> 2016-04-28 12:33 GMT+02:00 Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>>:
> Which OS, FS revision, platform, hardware, ram, cpu, etc are you using?
> 
> Which traffic are you handling?
> 
> What are exactly the simptoms?
> 
> How you get them? Is that reproducible?
> 
> Etc etc etc ????
> 
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
> 
> Il 28/Apr/2016 12:25, "Agust? Ubalde" <aubalde at presenceco.com <mailto:aubalde at presenceco.com>> ha scritto:
> Hi all,
> 
> We are having problems with the CPU load of Freeswitch. Specifically, there is a CPU having a charge of 70% of CPU system, always in the same CPU. Any ideas on the possible cause of this system CPU consumption?
> 
> 
> Thanks,
> PRESENCE TECHNOLOGY
> Agust? Ubalde Bellot
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com <mailto:aubalde at presenceco.com>
> Ph: +34 93 10 10 300 
> Fx: +34 93 10 10 333
> 
> 
> www.presenceco.com <http://www.presenceco.com/>
> Follow us on:
> 
>  <http://twitter.com/presencetech> <http://www.youtube.com/PresenceTechnology> <http://www.linkedin.com/groups?mostPopular=&gid=2898558> <http://www.slideshare.net/PresenceTech> <http://www.facebook.com/presencetech>
> For additional information, please visit our website www.presenceco.com <http://www.presenceco.com/>
> 
> Presence Technology - Disclaimer
> This 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 <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
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> 
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> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
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> 
> 
> Presence Technology - Disclaimer
> This 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
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> http://www.cluecon.com
> 
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From luis.daniel.lucio at gmail.com  Thu Apr 28 22:21:21 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Thu, 28 Apr 2016 14:21:21 -0400
Subject: [Freeswitch-users] High load kernel
In-Reply-To: <CAFLo2Qz48Ak6Fb2=r8Wtr6gzCeE5BQs4bqVupecp=Noar=6kCA@mail.gmail.com>
References: <CAKThY07PwnMuw-hy67zG+3wSo=FLaOfi+WTpB2Hj_jKhbijzxA@mail.gmail.com>
	<CALXCt0rePzPJPmtJMLAHsD=vuoHhooDQTzOgYNj5CMB0c16jzg@mail.gmail.com>
	<CAKThY04sqzToxwHHcuczwCKfj5DSC81KnFTboK2h1X5-Zu8YFA@mail.gmail.com>
	<90331067-44BB-4109-9866-C9538730C145@jerris.com>
	<CAFLo2QzZgtSMij9=O=BWRUuOdQsWvoOHoi01ZTH1OZaq=zsJ_w@mail.gmail.com>
	<CAFLo2Qz48Ak6Fb2=r8Wtr6gzCeE5BQs4bqVupecp=Noar=6kCA@mail.gmail.com>
Message-ID: <CAFLo2QxgiyWo3o=d_82Vi092ACUxOzPJ1r-tUxUc6WEqVT0Hew@mail.gmail.com>

I think you need to revise the system. Instead of asking like that. For
example, find exactly when the CPU rises.

Check what the  shows you. You may have a good idea. Check the syslog and
the dmesg output.

You will find answers there.
Le 28 avr. 2016 2:15 PM, "Michael Jerris" <mike at jerris.com> a ?crit :

Strongly advise to be using latest release in 1.6, particularly for
anything webrtc related.
Strongly advice to use Debian 8, there were significant known performance
problems on CentOS 6

On Apr 28, 2016, at 6:41 AM, Agust? Ubalde <aubalde at presenceco.com> wrote:

Which OS, FS revision, platform, hardware, ram, cpu, etc are you using?


   - CentOS 6.5 64b virutalized (ESXi)
   - Freeswitch 1.4.26
   - 8vCPU - 4GB RAM


Which traffic are you handling?


   - 60-70 active calls (WebRTC extensions)


What are exactly the simptoms?


   - top command shows high system cpu load constantly (CPU 3)


How you get them? Is that reproducible?


   - Always whit the same load (60-70 calls)



Thanks!

*PRESENCE TECHNOLOGY*
*Agust? Ubalde Bellot*
Chief Developer
C/ Comte Urgell 240 3A
Barcelona 08036
aubalde at presenceco.com

Ph: +34 93 10 10 300
Fx: +34 93 10 10 333

*www.presenceco.com* <http://www.presenceco.com/>

*Follow us on:*

*[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
<http://www.youtube.com/PresenceTechnology>*[image: in]*
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<http://www.slideshare.net/PresenceTech>*[image: fb]*
<http://www.facebook.com/presencetech>

For additional information, please visit our website *www.presenceco.com*
<http://www.presenceco.com/>

2016-04-28 12:33 GMT+02:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:

> Which OS, FS revision, platform, hardware, ram, cpu, etc are you using?
>
> Which traffic are you handling?
>
> What are exactly the simptoms?
>
> How you get them? Is that reproducible?
>
> Etc etc etc ????
>
> sent from mobile
> cell: +39 347 266 56 18
> Giovanni Maruzzelli
> OpenTelecom.IT
> Il 28/Apr/2016 12:25, "Agust? Ubalde" <aubalde at presenceco.com> ha scritto:
>
>> Hi all,
>>
>> We are having problems with the CPU load of Freeswitch. Specifically,
>> there is a CPU having a charge of 70% of CPU system, always in the same
>> CPU. Any ideas on the possible cause of this system CPU consumption?
>>
>>
>> Thanks,
>>
>> *PRESENCE TECHNOLOGY*
>> *Agust? Ubalde Bellot*
>> Chief Developer
>> C/ Comte Urgell 240 3A
>> Barcelona 08036
>> aubalde at presenceco.com
>>
>> Ph: +34 93 10 10 300
>> Fx: +34 93 10 10 333
>>
>> *www.presenceco.com* <http://www.presenceco.com/>
>>
>> *Follow us on:*
>>
>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>> <http://www.facebook.com/presencetech>
>>
>> For additional information, please visit our website *www.presenceco.com*
>> <http://www.presenceco.com/>
>>
>>
>> *Presence Technology - DisclaimerThis message, its content and any file
>> attached thereto is for the intended recipient only and is confidential and
>> /or privileged. If you have received this e-mail in error or had access to
>> it, you should note that the information in it is private and any use
>> thereof is unauthorized. In such an event please notify us by e-mail or by
>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
>> whatsoever means and any transmission or dissemination thereof to other
>> persons is prohibited. It should be deleted immediately from your system.
>> Presence Technology reserves the right to take legal action against any
>> persons unlawfully gaining access to the content of any external message it
>> has emitted.*
>> *For additional information, please visit our website **www.presenceco.com
>> <http://www.presenceco.com/>*
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



*Presence Technology - DisclaimerThis message, its content and any file
attached thereto is for the intended recipient only and is confidential and
/or privileged. If you have received this e-mail in error or had access to
it, you should note that the information in it is private and any use
thereof is unauthorized. In such an event please notify us by e-mail or by
telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
whatsoever means and any transmission or dissemination thereof to other
persons is prohibited. It should be deleted immediately from your system.
Presence Technology reserves the right to take legal action against any
persons unlawfully gaining access to the content of any external message it
has emitted.*
*For additional information, please visit our website **www.presenceco.com
<http://www.presenceco.com/>*


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

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

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



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

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

FreeSWITCH-users mailing 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 jungleboogie0 at gmail.com  Thu Apr 28 22:22:25 2016
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Thu, 28 Apr 2016 11:22:25 -0700
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <CAKy26iPJaTC3Wu9pCBX9FbnMTbJ_KaFF6iSOJ=55aN6WVuWeUw@mail.gmail.com>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
	<026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>
	<CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>
	<029601d1a171$1f5a6810$5e0f3830$@freeswitch.org>
	<CAKy26iMSooc58roPi9gwx+R3NetRy6YBVS28SJmid8GSvR=YEw@mail.gmail.com>
	<02b601d1a174$43b21e80$cb165b80$@freeswitch.org>
	<CAKy26iPJaTC3Wu9pCBX9FbnMTbJ_KaFF6iSOJ=55aN6WVuWeUw@mail.gmail.com>
Message-ID: <CAKE2PDtS-FEiEitYzshmcx2U70yXLALsXSGTLxfQXWocgEQFRA@mail.gmail.com>

On 28 April 2016 at 11:10, Luiz Fernando Softov <fernando at softov.com.br> wrote:
> Then i will make a fresh install, everything, FreeBSD and FreeSwitch, and
> activate the modules (that i need and isn't default active).


Good plan. 10.1 is EOL end of the year so consider 10.3, released a
few weeks ago:
https://www.freebsd.org/releases/10.3R/announce.html

ftp://ftp.freebsd.org/pub/FreeBSD/releases/ISO-IMAGES/10.3/



-- 
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si


From fernando at softov.com.br  Thu Apr 28 22:31:55 2016
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Thu, 28 Apr 2016 14:31:55 -0400
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <CAKE2PDtS-FEiEitYzshmcx2U70yXLALsXSGTLxfQXWocgEQFRA@mail.gmail.com>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
	<026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>
	<CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>
	<029601d1a171$1f5a6810$5e0f3830$@freeswitch.org>
	<CAKy26iMSooc58roPi9gwx+R3NetRy6YBVS28SJmid8GSvR=YEw@mail.gmail.com>
	<02b601d1a174$43b21e80$cb165b80$@freeswitch.org>
	<CAKy26iPJaTC3Wu9pCBX9FbnMTbJ_KaFF6iSOJ=55aN6WVuWeUw@mail.gmail.com>
	<CAKE2PDtS-FEiEitYzshmcx2U70yXLALsXSGTLxfQXWocgEQFRA@mail.gmail.com>
Message-ID: <CAKy26iOX3rZ5NYgvCKCDfM8EZYHSy7urBLdXBvmwPqvk2cJ-8A@mail.gmail.com>

I can't user another FreeBSD, since i have a lot of patches in the kernel.

And i have ~3000 clients using my system (~ 150) using the system with
FreeSwitch.

But, to install and test, i will download the new one (10.3).


2016-04-28 14:22 GMT-04:00 jungle Boogie <jungleboogie0 at gmail.com>:

> On 28 April 2016 at 11:10, Luiz Fernando Softov <fernando at softov.com.br>
> wrote:
> > Then i will make a fresh install, everything, FreeBSD and FreeSwitch, and
> > activate the modules (that i need and isn't default active).
>
>
> Good plan. 10.1 is EOL end of the year so consider 10.3, released a
> few weeks ago:
> https://www.freebsd.org/releases/10.3R/announce.html
>
> ftp://ftp.freebsd.org/pub/FreeBSD/releases/ISO-IMAGES/10.3/
>
>
>
> --
> -------
> inum: 883510009027723
> sip: jungleboogie at sip2sip.info
> xmpp: jungle-boogie at jit.si
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From arsenman at connectto.com  Thu Apr 28 22:47:41 2016
From: arsenman at connectto.com (Arsen Manukyan)
Date: Thu, 28 Apr 2016 11:47:41 -0700
Subject: [Freeswitch-users] Verto Communicator STUN and TURN settings
In-Reply-To: <CAKy26iOX3rZ5NYgvCKCDfM8EZYHSy7urBLdXBvmwPqvk2cJ-8A@mail.gmail.com>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
	<026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>
	<CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>
	<029601d1a171$1f5a6810$5e0f3830$@freeswitch.org>
	<CAKy26iMSooc58roPi9gwx+R3NetRy6YBVS28SJmid8GSvR=YEw@mail.gmail.com>
	<02b601d1a174$43b21e80$cb165b80$@freeswitch.org>
	<CAKy26iPJaTC3Wu9pCBX9FbnMTbJ_KaFF6iSOJ=55aN6WVuWeUw@mail.gmail.com>
	<CAKE2PDtS-FEiEitYzshmcx2U70yXLALsXSGTLxfQXWocgEQFRA@mail.gmail.com>
	<CAKy26iOX3rZ5NYgvCKCDfM8EZYHSy7urBLdXBvmwPqvk2cJ-8A@mail.gmail.com>
Message-ID: <0a4a68ab-b69b-0972-a088-17dccfc4ee52@connectto.com>

Verto Communicator *STUN* and *TURN* settings
need use my own TURN and STUN servers
I cant find any documentation how to change defalut configs for Verto 
Communicator
How contact with my TURN ?

thank you
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From chad at apartmentlines.com  Thu Apr 28 23:13:24 2016
From: chad at apartmentlines.com (Chad Phillips)
Date: Thu, 28 Apr 2016 12:13:24 -0700
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CAK+KT393aDZEDaTorZpS=Z8qCjez+Huewx0Hf7L6oFKoA3Pgzw@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
	<CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
	<CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>
	<CAK+KT3-3uF2-MF=WTmhV0KoBGjQtoeBsVzHX8JM3P-yzai_EmQ@mail.gmail.com>
	<CACOYK=jRVZLGtqOXg+SfTme1VAgGbJJ+1itb=DBr7u4-7JyAZA@mail.gmail.com>
	<CAK+KT393aDZEDaTorZpS=Z8qCjez+Huewx0Hf7L6oFKoA3Pgzw@mail.gmail.com>
Message-ID: <CAJ0sghQDkZFMtvOG_s0nMt_8TLg-03hV-NJdqnqbk009h98y8A@mail.gmail.com>

Stanislav, there are numerous advantages to automating server build
processes.

Also, having built https://github.com/thehunmonkgroup/freeswitch-kickstart,
I can assure you it's a bit more complicated than 'less than 10 commands to
execute'. :)

Richard, interesting that our projects were released in such close
proximity. Wonder why automation is in the air all of the sudden!

Chad

On Wed, Apr 27, 2016 at 9:50 PM, Richard Mace <richard.mace at gmail.com>
wrote:

> We are creating a PBX using FreeSWITCH for people who may well have no
> Linux experience at all, and we didn't want the actual "creating a
> FreeSWITCH box" to be a barrier to entry, so decided to create a FreeSWITCH
> builder app so they could build the box easily and focus on configuring the
> PBX
>
> Richard
> On 27 Apr 2016 23:11, "Stanislav Sinyagin" <ssinyagin at gmail.com> wrote:
>
>> come on, it's less than 10 commands to execute. Why do you need a
>> Windows program for this?
>>
>>
>>
>>
>> On Wed, Apr 27, 2016 at 9:49 PM, Richard Mace <richard.mace at gmail.com>
>> wrote:
>> > Hi Guillermo,
>> >
>> > On 27 April 2016 at 18:51, Guillermo Ruiz Camauer <grcamauer at gmail.com>
>> > wrote:
>> >>
>> >> Richard,
>> >>
>> >> Any documentation?
>> >
>> >
>> > Not yet, but I will be working on it.
>> >
>> >>
>> >> What does it do exactly?  Does it install from RPM or source?
>> >
>> >
>> > It instructs debian to download and install the FreeSWITCH debian
>> package
>> > files, as described on the FreeSWITCH website.
>> >
>> >
>> >>
>> >> Does it tweak the OS installation (ulimits, etc.)
>> >
>> >
>> > The only adjusts it makes to debian is it comments out the CD1 part of
>> the
>> > apt sources, and modifies the network side if required.
>> >
>> >
>> >>
>> >> Can it be customized so that certain modules get compiled/installed?
>> >
>> >
>> > I guess it could be, but not from source because it only installs the
>> > packages.
>> >
>> > Richard
>> >
>> >
>> >>
>> >>
>> >> Thanks,
>> >>
>> >> Guillermo Ruiz Camauer
>> >>
>> >>
>> >>
>> >> On Wed, Apr 27, 2016 at 12:36 PM, Richard Mace <richard.mace at gmail.com
>> >
>> >> wrote:
>> >>>
>> >>> Hi All,
>> >>>
>> >>> Just a quick note to say that our FreeSWITCH builder application is
>> ready
>> >>> for further testing.
>> >>>
>> >>> It is a Windows based application, that you "point" at a freshly built
>> >>> Debian box, and it will turn it into a FreeSWITCH 1.6 box, with just
>> a few
>> >>> clicks.
>> >>>
>> >>> Let us know what you think
>> >>>
>> >>> http://www.rocksoftware.co.uk/index.php/products/rockswitch-builder
>> >>>
>> >>> Richard
>> >>> www.rocksoftware.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
>> >>
>> >>
>> >>
>> >>
>> >> --
>> >> 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 richard.mace at gmail.com  Thu Apr 28 23:23:17 2016
From: richard.mace at gmail.com (Richard Mace)
Date: Thu, 28 Apr 2016 20:23:17 +0100
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CAJ0sghQDkZFMtvOG_s0nMt_8TLg-03hV-NJdqnqbk009h98y8A@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
	<CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
	<CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>
	<CAK+KT3-3uF2-MF=WTmhV0KoBGjQtoeBsVzHX8JM3P-yzai_EmQ@mail.gmail.com>
	<CACOYK=jRVZLGtqOXg+SfTme1VAgGbJJ+1itb=DBr7u4-7JyAZA@mail.gmail.com>
	<CAK+KT393aDZEDaTorZpS=Z8qCjez+Huewx0Hf7L6oFKoA3Pgzw@mail.gmail.com>
	<CAJ0sghQDkZFMtvOG_s0nMt_8TLg-03hV-NJdqnqbk009h98y8A@mail.gmail.com>
Message-ID: <CAK+KT3_gmfYHK13oh1gstS_Legh63PgxxKZeo7jJwg543b=K8A@mail.gmail.com>

>
>
> Richard, interesting that our projects were released in such close
> proximity. Wonder why automation is in the air all of the sudden!
>
> Chad
>
>
?Hi Chad,
I recognise you from a recent ClueCon weekly :)?
?Our PBX is nearly ready for Alpha testing, and we wanted to make it as
easy to install FreeSWITCH as possible. I remember when I first started
playing around with Asterisk, and I was coming straight from Windows and it
was a very big learning curve! Also, RoCKSwitch Builder is the first Open
Source app that we have written, so we're happy about that as well?.

Richard
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From gregor at infomedia.si  Fri Apr 29 00:06:04 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Thu, 28 Apr 2016 22:06:04 +0200
Subject: [Freeswitch-users] Custom variables in event socket
In-Reply-To: <CADcg8iUqTvfBPvBw_B6PGXwUr+3a97j4gjzYd+AkFPq41BFb7Q@mail.gmail.com>
References: <CAC7ttFFXsu_12mbTRjvMWHukd3on2D4+Wtbm3irbe6jw7qF62A@mail.gmail.com>
	<CAC7ttFGqD3Q7-P_zc=SJBCF8fNj7NVVdBC5QwJuLTO8gNPUpHA@mail.gmail.com>
	<CADcg8iUqTvfBPvBw_B6PGXwUr+3a97j4gjzYd+AkFPq41BFb7Q@mail.gmail.com>
Message-ID: <CAC7ttFFmx0kB=u_c57WpyVdAi+vuv_VsVL5TwSPqktW+3fmS5A@mail.gmail.com>

Oh, I see... Thank you, Joshua.

2016-04-27 10:47 GMT+02:00 Joshua Gigg <giggsey at gmail.com>:

> If you enable verbose-events, it will include all variables on all events.
>
> https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_verbose_events
>
> On Tue, 26 Apr 2016, 21:43 Gregor Nanger, <gregor at infomedia.si> wrote:
>
>> Searched through loga and findout where are variables posted:
>>
>> ChannelCreate (if started with originate)
>> ChannelExecute
>> ProgressMedia
>> ChannelBridge
>> ChannelAnswer
>> CallUpdate
>> ChannelUnbridge
>> ChannelHangup
>> ChannelDestroy
>>
>> Maybe someone will find it usefull.
>>
>> 2016-04-26 18:38 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:
>>
>>> Hi, does somebody know when are custom channel variables posted in event
>>> socket?
>>>
>>> I am listening on events and would like to read channel variables set in
>>> dialplan, but variables are not posted in each event.
>>>
>>> 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
>>>
>>
>>
>>
>> --
>> Gregor Nanger
>>
>> *CTO*
>> t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
>> ? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
>> ? www.infomedia.si
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Gregor Nanger

*CTO*
t./f.: 00386 (0) 7 6000 308/309 ? m:. 00386 (0)41 756485
? Infomedia d.o.o. ? Jerebova 3, Novo mesto, Slovenia
? www.infomedia.si
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From jaybinks at gmail.com  Fri Apr 29 02:32:58 2016
From: jaybinks at gmail.com (jay binks)
Date: Fri, 29 Apr 2016 08:32:58 +1000
Subject: [Freeswitch-users] High load kernel
In-Reply-To: <CAFLo2QxgiyWo3o=d_82Vi092ACUxOzPJ1r-tUxUc6WEqVT0Hew@mail.gmail.com>
References: <CAKThY07PwnMuw-hy67zG+3wSo=FLaOfi+WTpB2Hj_jKhbijzxA@mail.gmail.com>
	<CALXCt0rePzPJPmtJMLAHsD=vuoHhooDQTzOgYNj5CMB0c16jzg@mail.gmail.com>
	<CAKThY04sqzToxwHHcuczwCKfj5DSC81KnFTboK2h1X5-Zu8YFA@mail.gmail.com>
	<90331067-44BB-4109-9866-C9538730C145@jerris.com>
	<CAFLo2QzZgtSMij9=O=BWRUuOdQsWvoOHoi01ZTH1OZaq=zsJ_w@mail.gmail.com>
	<CAFLo2Qz48Ak6Fb2=r8Wtr6gzCeE5BQs4bqVupecp=Noar=6kCA@mail.gmail.com>
	<CAFLo2QxgiyWo3o=d_82Vi092ACUxOzPJ1r-tUxUc6WEqVT0Hew@mail.gmail.com>
Message-ID: <CAO47HpffErsKERbO5LE2NYPwdY=hqKkw82V5CtwySa200WfjXA@mail.gmail.com>

One other thing ( and this may not relate to WebRTC as I dont use it )
check MSI-X and your NIC Interrupts.

I have seen it before where every interrupt ( one per RTP packet ) was
being handled by the one CPU core, which means one core is used
significantly more than the others.

Jay

On 29 April 2016 at 04:21, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> I think you need to revise the system. Instead of asking like that. For
> example, find exactly when the CPU rises.
>
> Check what the  shows you. You may have a good idea. Check the syslog and
> the dmesg output.
>
> You will find answers there.
> Le 28 avr. 2016 2:15 PM, "Michael Jerris" <mike at jerris.com> a ?crit :
>
> Strongly advise to be using latest release in 1.6, particularly for
> anything webrtc related.
> Strongly advice to use Debian 8, there were significant known performance
> problems on CentOS 6
>
> On Apr 28, 2016, at 6:41 AM, Agust? Ubalde <aubalde at presenceco.com> wrote:
>
> Which OS, FS revision, platform, hardware, ram, cpu, etc are you using?
>
>
>    - CentOS 6.5 64b virutalized (ESXi)
>    - Freeswitch 1.4.26
>    - 8vCPU - 4GB RAM
>
>
> Which traffic are you handling?
>
>
>    - 60-70 active calls (WebRTC extensions)
>
>
> What are exactly the simptoms?
>
>
>    - top command shows high system cpu load constantly (CPU 3)
>
>
> How you get them? Is that reproducible?
>
>
>    - Always whit the same load (60-70 calls)
>
>
>
> Thanks!
>
> *PRESENCE TECHNOLOGY*
> *Agust? Ubalde Bellot*
> Chief Developer
> C/ Comte Urgell 240 3A
> Barcelona 08036
> aubalde at presenceco.com
>
> Ph: +34 93 10 10 300
> Fx: +34 93 10 10 333
>
> *www.presenceco.com* <http://www.presenceco.com/>
>
> *Follow us on:*
>
> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
> <http://www.youtube.com/PresenceTechnology>*[image: in]*
> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
> <http://www.slideshare.net/PresenceTech>*[image: fb]*
> <http://www.facebook.com/presencetech>
>
> For additional information, please visit our website *www.presenceco.com*
> <http://www.presenceco.com/>
>
> 2016-04-28 12:33 GMT+02:00 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>
>> Which OS, FS revision, platform, hardware, ram, cpu, etc are you using?
>>
>> Which traffic are you handling?
>>
>> What are exactly the simptoms?
>>
>> How you get them? Is that reproducible?
>>
>> Etc etc etc ????
>>
>> sent from mobile
>> cell: +39 347 266 56 18
>> Giovanni Maruzzelli
>> OpenTelecom.IT
>> Il 28/Apr/2016 12:25, "Agust? Ubalde" <aubalde at presenceco.com> ha
>> scritto:
>>
>>> Hi all,
>>>
>>> We are having problems with the CPU load of Freeswitch. Specifically,
>>> there is a CPU having a charge of 70% of CPU system, always in the same
>>> CPU. Any ideas on the possible cause of this system CPU consumption?
>>>
>>>
>>> Thanks,
>>>
>>> *PRESENCE TECHNOLOGY*
>>> *Agust? Ubalde Bellot*
>>> Chief Developer
>>> C/ Comte Urgell 240 3A
>>> Barcelona 08036
>>> aubalde at presenceco.com
>>>
>>> Ph: +34 93 10 10 300
>>> Fx: +34 93 10 10 333
>>>
>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>
>>> *Follow us on:*
>>>
>>> *[image: tw]* <http://twitter.com/presencetech>*[image: yt]*
>>> <http://www.youtube.com/PresenceTechnology>*[image: in]*
>>> <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]*
>>> <http://www.slideshare.net/PresenceTech>*[image: fb]*
>>> <http://www.facebook.com/presencetech>
>>>
>>> For additional information, please visit our website
>>> *www.presenceco.com* <http://www.presenceco.com/>
>>>
>>>
>>> *Presence Technology - DisclaimerThis message, its content and any file
>>> attached thereto is for the intended recipient only and is confidential and
>>> /or privileged. If you have received this e-mail in error or had access to
>>> it, you should note that the information in it is private and any use
>>> thereof is unauthorized. In such an event please notify us by e-mail or by
>>> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
>>> whatsoever means and any transmission or dissemination thereof to other
>>> persons is prohibited. It should be deleted immediately from your system.
>>> Presence Technology reserves the right to take legal action against any
>>> persons unlawfully gaining access to the content of any external message it
>>> has emitted.*
>>> *For additional information, please visit our website **www.presenceco.com
>>> <http://www.presenceco.com/>*
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> *Presence Technology - DisclaimerThis message, its content and any file
> attached thereto is for the intended recipient only and is confidential and
> /or privileged. If you have received this e-mail in error or had access to
> it, you should note that the information in it is private and any use
> thereof is unauthorized. In such an event please notify us by e-mail or by
> telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by
> whatsoever means and any transmission or dissemination thereof to other
> persons is prohibited. It should be deleted immediately from your system.
> Presence Technology reserves the right to take legal action against any
> persons unlawfully gaining access to the content of any external message it
> has emitted.*
> *For additional information, please visit our website **www.presenceco.com
> <http://www.presenceco.com/>*
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

Jay
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From msc at freeswitch.org  Fri Apr 29 02:58:39 2016
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Apr 2016 15:58:39 -0700
Subject: [Freeswitch-users] Freeswitch php scripts
In-Reply-To: <CAFSzUtb38-PivnmMbJweu_6g8t1_aOpzJd-3x5hPwnpS-sQFgQ@mail.gmail.com>
References: <CAFSzUtb38-PivnmMbJweu_6g8t1_aOpzJd-3x5hPwnpS-sQFgQ@mail.gmail.com>
Message-ID: <CAKzWOxUnu-SGrAHBpYCzoVQF0A_cez8Q+o0UEdiwGSR+r_Fnag@mail.gmail.com>

It sounds like the dialplan never makes it to your extension. Get a debug
log of the call and put it on pastebin.freeswitch.org. We can probably help
you figure it out from there.

-MSC

On Thu, Apr 28, 2016 at 12:40 AM, Nduwayezu, Joselyne <35633 at heb.be> wrote:

>
> Hello,
>
> I would like to invoke php script in the dial plan.
> I defined the dialplan as follos:
>
>
> <include>
>  <extension name="outbound-socket">
>   <condition field="${sip_to_user}" expression="(972550428)$">
>    <action application="set"
> data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
>    <action application="socket" data="10.0.0.6:8021 async full"/>-->
>   </condition>
>  </extension>
> </include>
>
> And my php script is:
>
>
> <?php
>   // set a couple of things so we dont kill the system
>  ob_implicit_flush(true);
>  set_time_limit(30);
>  // Open stdin so we can read the data in
>  $in = fopen("php://stdin", "r");
>  // Connect
>  echo "connect\n\n";
>  // Answer
>  echo "sendmsg\n";
>  echo "call-command: execute\n";
>  echo "execute-app-name: answer\n\n";
>  // Play a prompt
>  echo "sendmsg\n";
>  echo "call-command: execute\n";
>  echo "execute-app-name: playback\n";
>  echo "execute-app-arg: \
>  /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n
> \
>  ";
>  // Wait
>  sleep(5);
>  // Hangup
>  echo "sendmsg\n";
>  echo "call-command: hangup\n\n";
>  fclose($in);
>  ?>
>
> When i dial the DID number, the call state is "call setup" for a while,
> and after i'm asked to leave a message because the number is busy or
> unreachable.
>
> I tried different sockets in the dialplan (5060, 5080, 8040) with no
> results
>
> Any adeas of what is happening?
>
> Thanks
>
>
> NDUWAYEZU Joselyne
>
> Haute ?cole de Bruxelles
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Apr 29 02:58:25 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Fri, 29 Apr 2016 00:58:25 +0200
Subject: [Freeswitch-users] Bridge campon
Message-ID: <CAC7ttFG_NYWBvnU_Ut8D3GdjomTNwKz44-h0eaZ98K-xD4vHCw@mail.gmail.com>

I have strange problem with campon. Somehow FS  doesn't obey what I set as
campon_timeout variable.

Whatever I set, it is always 60 second. It is not even default of 10
seconds. But funny is that, campon_pause works as expected.

Any hints what could be wrong. I set dialplan with xml_curl, so dialplan is
short and nothing is injected.

Best regards, Gregor
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From ssinyagin at gmail.com  Fri Apr 29 04:29:32 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 29 Apr 2016 02:29:32 +0200
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CAK+KT3_gmfYHK13oh1gstS_Legh63PgxxKZeo7jJwg543b=K8A@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
	<CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
	<CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>
	<CAK+KT3-3uF2-MF=WTmhV0KoBGjQtoeBsVzHX8JM3P-yzai_EmQ@mail.gmail.com>
	<CACOYK=jRVZLGtqOXg+SfTme1VAgGbJJ+1itb=DBr7u4-7JyAZA@mail.gmail.com>
	<CAK+KT393aDZEDaTorZpS=Z8qCjez+Huewx0Hf7L6oFKoA3Pgzw@mail.gmail.com>
	<CAJ0sghQDkZFMtvOG_s0nMt_8TLg-03hV-NJdqnqbk009h98y8A@mail.gmail.com>
	<CAK+KT3_gmfYHK13oh1gstS_Legh63PgxxKZeo7jJwg543b=K8A@mail.gmail.com>
Message-ID: <CACOYK=jwm6qUgsYtcfC27zQa0tF=EGhXdBhgaFjphDJCo+tcjw@mail.gmail.com>

Open source without an easy way to browse the code? I couldn't find any
links apart from the binary downloads.
On Apr 28, 2016 21:23, "Richard Mace" <richard.mace at gmail.com> wrote:

>
>> Richard, interesting that our projects were released in such close
>> proximity. Wonder why automation is in the air all of the sudden!
>>
>> Chad
>>
>>
> ?Hi Chad,
> I recognise you from a recent ClueCon weekly :)?
> ?Our PBX is nearly ready for Alpha testing, and we wanted to make it as
> easy to install FreeSWITCH as possible. I remember when I first started
> playing around with Asterisk, and I was coming straight from Windows and it
> was a very big learning curve! Also, RoCKSwitch Builder is the first Open
> Source app that we have written, so we're happy about that as well?.
>
> Richard
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Apr 29 04:54:05 2016
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 28 Apr 2016 19:54:05 -0500
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CACOYK=jwm6qUgsYtcfC27zQa0tF=EGhXdBhgaFjphDJCo+tcjw@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
	<CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
	<CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>
	<CAK+KT3-3uF2-MF=WTmhV0KoBGjQtoeBsVzHX8JM3P-yzai_EmQ@mail.gmail.com>
	<CACOYK=jRVZLGtqOXg+SfTme1VAgGbJJ+1itb=DBr7u4-7JyAZA@mail.gmail.com>
	<CAK+KT393aDZEDaTorZpS=Z8qCjez+Huewx0Hf7L6oFKoA3Pgzw@mail.gmail.com>
	<CAJ0sghQDkZFMtvOG_s0nMt_8TLg-03hV-NJdqnqbk009h98y8A@mail.gmail.com>
	<CAK+KT3_gmfYHK13oh1gstS_Legh63PgxxKZeo7jJwg543b=K8A@mail.gmail.com>
	<CACOYK=jwm6qUgsYtcfC27zQa0tF=EGhXdBhgaFjphDJCo+tcjw@mail.gmail.com>
Message-ID: <CAKbxfG8bH2A84Ng08RWU6QgwM9gPgB2s5JQa9nG=1jHtaP2Vtg@mail.gmail.com>

Hopefully its adding our repo and installing the packages as the method of
setting up the box?



On Thursday, April 28, 2016, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> Open source without an easy way to browse the code? I couldn't find any
> links apart from the binary downloads.
> On Apr 28, 2016 21:23, "Richard Mace" <richard.mace at gmail.com
> <javascript:_e(%7B%7D,'cvml','richard.mace at gmail.com');>> wrote:
>
>>
>>> Richard, interesting that our projects were released in such close
>>> proximity. Wonder why automation is in the air all of the sudden!
>>>
>>> Chad
>>>
>>>
>> ?Hi Chad,
>> I recognise you from a recent ClueCon weekly :)?
>> ?Our PBX is nearly ready for Alpha testing, and we wanted to make it as
>> easy to install FreeSWITCH as possible. I remember when I first started
>> playing around with Asterisk, and I was coming straight from Windows and it
>> was a very big learning curve! Also, RoCKSwitch Builder is the first Open
>> Source app that we have written, so we're happy about that as well?.
>>
>> Richard
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

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

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

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

https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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From nneul at mst.edu  Fri Apr 29 07:49:03 2016
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 28 Apr 2016 22:49:03 -0500
Subject: [Freeswitch-users] fyi - today's free Packt ebook is WebRTC
	Integrators guide
Message-ID: <5722D9AF.5070805@mst.edu>

Might be of interest to those on this list...

https://www.packtpub.com/packt/offers/free-learning


-- Nathan

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


From manpower13.cse at gmail.com  Fri Apr 29 08:59:38 2016
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Fri, 29 Apr 2016 10:29:38 +0530
Subject: [Freeswitch-users] mod_nibblebill issue with recording
Message-ID: <CAMRuXGgAJn_7S_HhAVk6QfwPcgQXhono9Lc9Kt7nB5NxoqBjFQ@mail.gmail.com>

HI,

   We  are using nibllebill for billing and AWS S3 for call recording
storage ,When we record the call using record_session it will upload
recorded WAV file to s3(AWS).

Issue:

          1.mod_nibblebill bill ammount include record_session wav file
upload time


Example:

  if I talk 2 mins and record_session take 20/sec to upload WAV file to
S3(AWS) the billed amount 2.20/mins (0.1 per mins total amount 0.3 )   but
billed time only shows 2 mins
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From joel at gogii.net  Fri Apr 29 09:55:04 2016
From: joel at gogii.net (Joel Serrano)
Date: Thu, 28 Apr 2016 22:55:04 -0700
Subject: [Freeswitch-users] fyi - today's free Packt ebook is WebRTC
 Integrators guide
In-Reply-To: <5722D9AF.5070805@mst.edu>
References: <5722D9AF.5070805@mst.edu>
Message-ID: <CAMtXxQn0LjMTgdhf5ZnZKUf8gr7wDDsqTXXz5WJ8Znovg25V8Q@mail.gmail.com>

Thanks for sharing!


On Thu, Apr 28, 2016 at 8:49 PM, Nathan Neulinger <nneul at mst.edu> wrote:

> Might be of interest to those on this list...
>
> https://www.packtpub.com/packt/offers/free-learning
>
>
> -- Nathan
>
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From amani.mansour2 at gmail.com  Fri Apr 29 10:47:58 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Fri, 29 Apr 2016 06:47:58 +0000
Subject: [Freeswitch-users] FreeSWITCH informations
Message-ID: <CAFETSWjeDshcm90vfBV88f-y95oqF=mCA9ynFg+NSD1OPjGUHA@mail.gmail.com>

Good morning all,
I have some questions please .
          1/how can i use FreeSWITCH interfece GUI (in my researsh i found
that WIKIPBX is abondoned )
            2/we can consider FreeSWITCH us a test tools ?? likeSIPp
because we can make some sc?nario with FreeSWITCH
             3/In the definition of  FreeSWITCH ,it say that FreeSWITCH can
be user us  simple commutateur ,a PBX, a gateway or IVR server
 (Interactive Voice Response) but i didn't find in which case is used in
each time


Sorry for those questions ,may be  is stupid to ask such questions but i
didn't find a clear response

please can any one help me .

Thanks
Best regards
Amani
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From benjamin.cropley at gmail.com  Fri Apr 29 11:20:08 2016
From: benjamin.cropley at gmail.com (Benjamin Cropley)
Date: Fri, 29 Apr 2016 08:20:08 +0100
Subject: [Freeswitch-users] FreeSWITCH informations
In-Reply-To: <CAFETSWjeDshcm90vfBV88f-y95oqF=mCA9ynFg+NSD1OPjGUHA@mail.gmail.com>
References: <CAFETSWjeDshcm90vfBV88f-y95oqF=mCA9ynFg+NSD1OPjGUHA@mail.gmail.com>
Message-ID: <CAF7jizJFJtfG7zG2ufMsXXJAQS1M8BVV2NejLk8v04eEHDwcHA@mail.gmail.com>

* 1/how can i use FreeSWITCH interfece GUI (in my researsh i found that
WIKIPBX is abondoned )*
There isn't an official one. But you can find a list of options here
https://freeswitch.org/confluence/display/FREESWITCH/Freeswitch+GUI

*2/we can consider FreeSWITCH us a test tools ?? likeSIPp because we can
make some sc?nario with FreeSWITCH *
Yes. Look into the ESL
https://freeswitch.org/confluence/display/FREESWITCH/Event+Socket+Library

*3/In the definition of  FreeSWITCH ,it say that FreeSWITCH can be user us
 simple commutateur ,a PBX, a gateway or IVR server  (Interactive Voice
Response) but i didn't find in which case is used in each time *
You can set it up to act as one or all of the above.

Hope that helps,
Ben


On Fri, Apr 29, 2016 at 7:47 AM, amani mansour <amani.mansour2 at gmail.com>
wrote:

> Good morning all,
> I have some questions please .
>           1/how can i use FreeSWITCH interfece GUI (in my researsh i found
> that WIKIPBX is abondoned )
>             2/we can consider FreeSWITCH us a test tools ?? likeSIPp
> because we can make some sc?nario with FreeSWITCH
>              3/In the definition of  FreeSWITCH ,it say that FreeSWITCH
> can be user us  simple commutateur ,a PBX, a gateway or IVR server
>  (Interactive Voice Response) but i didn't find in which case is used in
> each time
>
>
> Sorry for those questions ,may be  is stupid to ask such questions but i
> didn't find a clear response
>
> please can any one help me .
>
> Thanks
> Best regards
> Amani
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From s.safarov at gmail.com  Fri Apr 29 11:28:11 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 29 Apr 2016 07:28:11 +0000
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
	<026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>
	<CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>
Message-ID: <CAHtxdDehdwBRSQVRjZF524HBnyZZ-DGs=Fzrrdrec5tPW6Qn8Q@mail.gmail.com>

Please look

   1. FS-7705 <https://freeswitch.org/jira/browse/FS-7705>
   2. PR293
   <https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/293/diff>

This
Requires: libshout >= 2.3.1
Requires: libmpg123 >= 1.20.1

Check your ports.

??, 28 ???. 2016 ?. ? 20:10, Luiz Fernando Softov <fernando at softov.com.br>:

> It's the same with
>
> making all mod_shout
> gmake[4]: Entering directory
> '/usr/dev/freeswitch_backend/freeswitch/src/mod/formats/mod_shout'
> Makefile:907: *** You must install libshout3-dev to build mod_shout.  Stop.
>
> First I install libshout from ports, reclone, bootstrap, configure e when
> make no effect
> Then i download from internet, and its the same thing.
>
> I think this will be the same with libmpg123.
>
> About the link, i only need to clone it and make install clean, or need to
> clone it in a correct directory?
> Like
> /usr/dev/freeswitch_backend/freeswitch/libs/opus
> /usr/dev/freeswitch_backend/freeswitch/libs/opus-1.1
>
> ????
>
>
>
>
>
> 2016-04-28 12:45 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>
>> https://freeswitch.org/stash/projects/SD/repos/opus/browse
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
>> Fernando Softov
>> *Sent:* Thursday, April 28, 2016 11:29 AM
>>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>>
>>
>>
>> I cloned and trying to compile now, in FreeBSD
>>
>> making all mod_opus
>>
>> gmake[4]: Entering directory
>> '/usr/dev/freeswitch_backend/freeswitch/src/mod/codecs/mod_opus'
>>
>> Makefile:897: *** You must install libopus-dev to build mod_opus.  Stop.
>>
>> I already compile and install opus from downloads.xiph.org/releases/opus/
>>
>> since in FreeBSD doesn't have a libopus-dev
>>
>> Is there another place to get libopus?
>>
>>
>>
>> 2016-04-28 9:31 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>>
>> There is no tag for latest master. Its just master? git checkout master
>> && git pull?
>>
>>
>>
>>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
>> Fernando Softov
>> *Sent:* Thursday, April 28, 2016 8:20 AM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>>
>>
>>
>> Yeah, latest master, but witch tag?
>>
>> 1.4.x
>> 1.5.final
>> 1.6.0
>> 1.6.7
>> 1.7.0
>>
>> There are so many tags, so i have lost my mind witch one.
>>
>>
>> I already thy to compile 1.6 in FreeBSD, but without success, there are
>> dependencies in mod_opus, and another thing i can't remember now.
>>
>> Which is the difference between tags?
>>
>>
>>
>>
>>
>> 2016-04-28 9:09 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>>
>> Latest master?. There are numerous bugs and potential security issues
>> that have been resolved since then.
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
>> Fernando Softov
>> *Sent:* Thursday, April 28, 2016 7:48 AM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] FreeSwitch - Clone - TAG
>>
>>
>>
>> Hi,
>>
>> I don't remember why, but I'm using 1.5.final (branch master) in
>> production (~1 year), and it's working fine, with 30 simultaneous calls.
>>
>>
>> I will re-clone the FreeSwitch, because some changes made, that i need.
>>
>> Someone recommend to clone another tag?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From 35633 at heb.be  Fri Apr 29 11:56:38 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Fri, 29 Apr 2016 09:56:38 +0200
Subject: [Freeswitch-users] ESL.so
In-Reply-To: <00b701d1a153$9c7b9f80$d572de80$@freeswitch.org>
References: <CAFSzUtadEA6HO7Z++QWF4wajBu70ZF=nQcm1vYSJAEhwuuvw0A@mail.gmail.com>
	<00b701d1a153$9c7b9f80$d572de80$@freeswitch.org>
Message-ID: <CAFSzUtZpchL97goF7_yEyxoJhnkQg2fFg3Gg4odvrEwU1u3-jA@mail.gmail.com>

When i look for ESL.so, i can't find it, so it is not installes. How can i
install it? I tried to do "make and make phpmod" in
/usr/local/src/freeswitch/libs/esl, but i get the error 'No rule to make
target 'phpmode' when i run make phpmode

NDUWAYEZU Joselyne

2016-04-28 15:41 GMT+02:00 Ken Rice <krice at freeswitch.org>:

> Its exactly what it says, it cant open the ESL.so because it doesn?t
> exist? make sure you installed the .so into the correct location
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nduwayezu,
> Joselyne
> *Sent:* Thursday, April 28, 2016 8:39 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* [Freeswitch-users] ESL.so
>
>
>
>
>
> Hello, i m having an error when i start php5.
>
> The error is :
>
> php warnign: php startup: Unable to laod dynamic library
> '/usr/lib/php5/20131266/ESL.so': cannot open shared object file: No such
> file or directory
>
>
>
> Any help?
>
>
> NDUWAYEZU Joselyne
>
>
> Haute ?cole de Bruxelles
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Haute ?cole de Bruxelles
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From amani.mansour2 at gmail.com  Fri Apr 29 13:30:04 2016
From: amani.mansour2 at gmail.com (amani mansour)
Date: Fri, 29 Apr 2016 09:30:04 +0000
Subject: [Freeswitch-users] Internal profile doesn't exist
Message-ID: <CAFETSWgCLAyrOagQoNdOUqG4AcfNhOJUdD-Tq08K-WvrU-DyRg@mail.gmail.com>

Hi all,
I don't know what is the problem ,

Remarque  ,my IP address is correct in vars.xml (static)
when i tape sofia status
it affiche external,gateways,internal_ipv6 but the internal profile isn't
exist
is there someone who know the solusion please ?

[image: pasted1]
Best regards
Amani
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From s.safarov at gmail.com  Fri Apr 29 13:33:36 2016
From: s.safarov at gmail.com (Sergey Safarov)
Date: Fri, 29 Apr 2016 09:33:36 +0000
Subject: [Freeswitch-users] Internal profile doesn't exist
In-Reply-To: <CAFETSWgCLAyrOagQoNdOUqG4AcfNhOJUdD-Tq08K-WvrU-DyRg@mail.gmail.com>
References: <CAFETSWgCLAyrOagQoNdOUqG4AcfNhOJUdD-Tq08K-WvrU-DyRg@mail.gmail.com>
Message-ID: <CAHtxdDdbKSN-3dz5v02XJEB9VoZ_VLbz7S7HF4O3vrBsA-ybPw@mail.gmail.com>

Probable;
1) other appication is allready binded to profile ip:port
2) you try configure TLS and cert is to found.

Sergey

??, 29 ???. 2016 ?. ? 12:31, amani mansour <amani.mansour2 at gmail.com>:

> Hi all,
> I don't know what is the problem ,
>
> Remarque  ,my IP address is correct in vars.xml (static)
> when i tape sofia status
> it affiche external,gateways,internal_ipv6 but the internal profile isn't
> exist
> is there someone who know the solusion please ?
>
> [image: pasted1]
> Best regards
> Amani
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Fri Apr 29 13:55:55 2016
From: deepikay at iiitd.ac.in (Deepika Yadav)
Date: Fri, 29 Apr 2016 15:25:55 +0530
Subject: [Freeswitch-users] Internal profile doesn't exist
In-Reply-To: <CAHtxdDdbKSN-3dz5v02XJEB9VoZ_VLbz7S7HF4O3vrBsA-ybPw@mail.gmail.com>
References: <CAFETSWgCLAyrOagQoNdOUqG4AcfNhOJUdD-Tq08K-WvrU-DyRg@mail.gmail.com>
	<CAHtxdDdbKSN-3dz5v02XJEB9VoZ_VLbz7S7HF4O3vrBsA-ybPw@mail.gmail.com>
Message-ID: <CAM99_YnS==0MRe_+hxakfAPp3RJP5z8QS_szL41yvO_4+j1GNA@mail.gmail.com>

Please check which application is running at port 5060 and 5080, if it is
other than Freeswitch, stop it and restart the profile:

"sofia profile internal restart"

On Fri, Apr 29, 2016 at 3:03 PM, Sergey Safarov <s.safarov at gmail.com> wrote:

> Probable;
> 1) other appication is allready binded to profile ip:port
> 2) you try configure TLS and cert is to found.
>
> Sergey
>
> ??, 29 ???. 2016 ?. ? 12:31, amani mansour <amani.mansour2 at gmail.com>:
>
>> Hi all,
>> I don't know what is the problem ,
>>
>> Remarque  ,my IP address is correct in vars.xml (static)
>> when i tape sofia status
>> it affiche external,gateways,internal_ipv6 but the internal profile isn't
>> exist
>> is there someone who know the solusion please ?
>>
>> [image: pasted1]
>> Best regards
>> Amani
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From 35633 at heb.be  Fri Apr 29 15:22:34 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Fri, 29 Apr 2016 13:22:34 +0200
Subject: [Freeswitch-users] Problem with ESL.so
Message-ID: <CAFSzUtY24z0=Gzr6CzUH9iNd6AQpXPkA9rdtUHcuWbhxLTK8Qw@mail.gmail.com>

I,ve followed the freeswitch cookbook in order to user the ESL. I
installed  all the libraries needed.
But when i run ESL.so, i have a segmentation fault which result in error
when i run a php script. the error: unable to load dynamic library
/usr/lib/php5/20131226.
 Any help?

NDUWAYEZU Joselyne

-- 
Haute ?cole de Bruxelles
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From gregor at infomedia.si  Fri Apr 29 16:14:11 2016
From: gregor at infomedia.si (Gregor Nanger)
Date: Fri, 29 Apr 2016 14:14:11 +0200
Subject: [Freeswitch-users] Bridge campon
In-Reply-To: <CAC7ttFG_NYWBvnU_Ut8D3GdjomTNwKz44-h0eaZ98K-xD4vHCw@mail.gmail.com>
References: <CAC7ttFG_NYWBvnU_Ut8D3GdjomTNwKz44-h0eaZ98K-xD4vHCw@mail.gmail.com>
Message-ID: <CAC7ttFGwrxS11ixp3R27Wg-b8Gcr2Z+3EvYMd9qseNFK8R43yw@mail.gmail.com>

Whatever I did, campon_timeout is always 60.

leg_timeout on b leg saved my day :-)))

Other campon variables works as expected.

Am I onlyone with such problem?

2016-04-29 0:58 GMT+02:00 Gregor Nanger <gregor at infomedia.si>:

> I have strange problem with campon. Somehow FS  doesn't obey what I set as
> campon_timeout variable.
>
> Whatever I set, it is always 60 second. It is not even default of 10
> seconds. But funny is that, campon_pause works as expected.
>
> Any hints what could be wrong. I set dialplan with xml_curl, so dialplan
> is short and nothing is injected.
>
> 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 benjamin.cropley at gmail.com  Fri Apr 29 16:37:31 2016
From: benjamin.cropley at gmail.com (Benjamin Cropley)
Date: Fri, 29 Apr 2016 13:37:31 +0100
Subject: [Freeswitch-users] Problem with ESL.so
In-Reply-To: <CAFSzUtY24z0=Gzr6CzUH9iNd6AQpXPkA9rdtUHcuWbhxLTK8Qw@mail.gmail.com>
References: <CAFSzUtY24z0=Gzr6CzUH9iNd6AQpXPkA9rdtUHcuWbhxLTK8Qw@mail.gmail.com>
Message-ID: <CAF7jizKFDPp-8=Q4+4bHbV62pYQMbMVohHjCNeMXG-9ZDLU7fg@mail.gmail.com>

Did you upgrade PHP? Are your .ini files still pointing to old locations?

On Fri, Apr 29, 2016 at 12:22 PM, Nduwayezu, Joselyne <35633 at heb.be> wrote:

>
> I,ve followed the freeswitch cookbook in order to user the ESL. I
> installed  all the libraries needed.
> But when i run ESL.so, i have a segmentation fault which result in error
> when i run a php script. the error: unable to load dynamic library
> /usr/lib/php5/20131226.
>  Any help?
>
> NDUWAYEZU Joselyne
>
> Haute ?cole de Bruxelles
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From thomas.peterseil at gmail.com  Fri Apr 29 16:47:06 2016
From: thomas.peterseil at gmail.com (thomas peterseil)
Date: Fri, 29 Apr 2016 14:47:06 +0200
Subject: [Freeswitch-users] "To field" in sip header
In-Reply-To: <CAH5f_dvQ-4Tt9pnkd0GyYMu3QzXswvRREp==u_=37L+LQ_zxQw@mail.gmail.com>
References: <CAH5f_dvTgwNftgqCx0TSYhdmfjUF5B+0=gaDWNVb1jVYHdX5tA@mail.gmail.com>
	<CAH5f_dvQ-4Tt9pnkd0GyYMu3QzXswvRREp==u_=37L+LQ_zxQw@mail.gmail.com>
Message-ID: <CAH5f_ds2t9ur7sNoF-6pA8r6pBs6C-OUMVcsO-yX+k0rRxtojw@mail.gmail.com>

hello,
i have a sip provider with multiple DID?s and they send me all calls with
the same invite, only the To field in the SIP header is different. how can
i read the To field in the SIP header to be able to make a call routing
based on the To field? i tried the following rule but it doesnt work:

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

<include>

<extension
name="public_test1">

        <condition
regex="all">

        <regex  field="destination_number"
expression="^provider1$"/>
        <regex  field="${sip_h_To}"
expression="43717072377"/>


                <action application="set"
data="effective_caller_id_number=7021"/>
                <action application="set"
data="effective_caller_id_name=thomas"/>
                <action
application="answer"/>

                <action application="transfer" data="6011 XML inbound1
"/>

</condition>

</extension>

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

the To field in the SIP header looks like this:

To: <sip:43717072377 at 193.43.148.37>

thanks a lot for help!

best regards,
thomas
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From abalashov at evaristesys.com  Fri Apr 29 17:03:43 2016
From: abalashov at evaristesys.com (Alex Balashov)
Date: Fri, 29 Apr 2016 09:03:43 -0400
Subject: [Freeswitch-users] "To field" in sip header
In-Reply-To: <CAH5f_ds2t9ur7sNoF-6pA8r6pBs6C-OUMVcsO-yX+k0rRxtojw@mail.gmail.com>
References: <CAH5f_dvTgwNftgqCx0TSYhdmfjUF5B+0=gaDWNVb1jVYHdX5tA@mail.gmail.com>
	<CAH5f_dvQ-4Tt9pnkd0GyYMu3QzXswvRREp==u_=37L+LQ_zxQw@mail.gmail.com>
	<CAH5f_ds2t9ur7sNoF-6pA8r6pBs6C-OUMVcsO-yX+k0rRxtojw@mail.gmail.com>
Message-ID: <20160429130343.5394500.87280.235762@evaristesys.com>

The standards are quite clear that To is a cosmetic header and is not to be used for any routing.

Only the Request URI provides "DNIS". If your SIP provider isn't differentiating it based on the number dialed, ask them to enable that.
?
--
Alex?Balashov?|?Principal?|?Evariste?Systems?LLC
1447?Peachtree?Street?NE,?Suite?700
Atlanta,?GA?30309
United?States

Tel:?+1-800-250-5920?(toll-free)?/?+1-678-954-0671 (direct)
Web:?http://www.evaristesys.com/, http://www.csrpswitch.com/

Sent?from?my?BlackBerry.
? Original Message ?
From: thomas peterseil
Sent: Friday, April 29, 2016 08:48
To: FreeSWITCH Users Help
Reply To: FreeSWITCH Users Help
Subject: [Freeswitch-users] "To field" in sip header

hello,
i have a sip provider with multiple DID?s and they send me all calls with the same invite, only the To field in the SIP header is different. how can i read the To field in the SIP header to be able to make a call routing based on the To field? i tried the following rule but it doesnt work:

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

<include>???????????????????????????????????????????????????????????????????????????????????????????????? 
<extension name="public_test1">?????????????????????????????????????????????????????
??????? <condition regex="all">?????????????????????????????????????????????????????????????????????
??????? <regex? field="destination_number" expression="^provider1$"/>?????????????????????????????? 
??????? <regex? field="${sip_h_To}" expression="43717072377"/>??????????????????????????????????????????? 
????????????????????????????????????????????????????????????????????????????????????????????????????????? 
??????????????? <action application="set" data="effective_caller_id_number=7021"/>??????????????????????? 
??????????????? <action application="set" data="effective_caller_id_name=thomas"/>????????????? 
??????????????? <action application="answer"/>??????????????????????????????????????????
??????????????? <action application="transfer" data="6011 XML inbound1 "/>????????????????????????????? 
??????? </condition>????????????????????????????????????????????????????????????????????????????????????? 
</extension>

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

the To field in the SIP header looks like this:

To: <sip:43717072377 at 193.43.148.37>

thanks a lot for help!

best regards,
thomas




From luis.daniel.lucio at gmail.com  Fri Apr 29 17:13:28 2016
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Fri, 29 Apr 2016 09:13:28 -0400
Subject: [Freeswitch-users] "To field" in sip header
In-Reply-To: <20160429130343.5394500.87280.235762@evaristesys.com>
References: <CAH5f_dvTgwNftgqCx0TSYhdmfjUF5B+0=gaDWNVb1jVYHdX5tA@mail.gmail.com>
	<CAH5f_dvQ-4Tt9pnkd0GyYMu3QzXswvRREp==u_=37L+LQ_zxQw@mail.gmail.com>
	<CAH5f_ds2t9ur7sNoF-6pA8r6pBs6C-OUMVcsO-yX+k0rRxtojw@mail.gmail.com>
	<20160429130343.5394500.87280.235762@evaristesys.com>
Message-ID: <CAFLo2QznqHxnmDDQbSTJrNy0yao+8qSOgOYyeW5+zE1MqFyRHQ@mail.gmail.com>

Tell us your provider, so we can avoid it :)
Le 29 avr. 2016 9:04 AM, "Alex Balashov" <abalashov at evaristesys.com> a
?crit :

> The standards are quite clear that To is a cosmetic header and is not to
> be used for any routing.
>
> Only the Request URI provides "DNIS". If your SIP provider isn't
> differentiating it based on the number dialed, ask them to enable that.
> ?
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 1447 Peachtree Street NE, Suite 700
> Atlanta, GA 30309
> United States
>
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
> Sent from my BlackBerry.
>   Original Message
> From: thomas peterseil
> Sent: Friday, April 29, 2016 08:48
> To: FreeSWITCH Users Help
> Reply To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] "To field" in sip header
>
> hello,
> i have a sip provider with multiple DID?s and they send me all calls with
> the same invite, only the To field in the SIP header is different. how can
> i read the To field in the SIP header to be able to make a call routing
> based on the To field? i tried the following rule but it doesnt work:
>
> ---------------------
>
>
> <include>
> <extension
> name="public_test1">
>         <condition
> regex="all">
>         <regex  field="destination_number"
> expression="^provider1$"/>
>         <regex  field="${sip_h_To}"
> expression="43717072377"/>
>
>
>                 <action application="set"
> data="effective_caller_id_number=7021"/>
>                 <action application="set"
> data="effective_caller_id_name=thomas"/>
>                 <action
> application="answer"/>
>                 <action application="transfer" data="6011 XML inbound1
> "/>
>
> </condition>
> </extension>
>
> ----------------
>
> the To field in the SIP header looks like this:
>
> To: <sip:43717072377 at 193.43.148.37>
>
> thanks a lot for help!
>
> best regards,
> thomas
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From 35633 at heb.be  Fri Apr 29 17:29:49 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Fri, 29 Apr 2016 15:29:49 +0200
Subject: [Freeswitch-users] Freeswitch php scripts
In-Reply-To: <CAKzWOxUnu-SGrAHBpYCzoVQF0A_cez8Q+o0UEdiwGSR+r_Fnag@mail.gmail.com>
References: <CAFSzUtb38-PivnmMbJweu_6g8t1_aOpzJd-3x5hPwnpS-sQFgQ@mail.gmail.com>
	<CAKzWOxUnu-SGrAHBpYCzoVQF0A_cez8Q+o0UEdiwGSR+r_Fnag@mail.gmail.com>
Message-ID: <CAFSzUtbbJD=1D_Zpo6d9-8OawuLAEb9p9VouxgfNY3QfN08ANA@mail.gmail.com>

Hello,
I can not log on pastebin.freeswitch.org
This is the log when i make a call

I still having the problem


root at back-1:/usr/local/freeswitch/conf/dialplan/public# ngrep -d eth0 -t -W
byline "$1" port 5060 -q
interface: eth0 (10.0.0.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
U 2016/04/29 13:14:47.605641 10.0.0.5:5060 -> 10.0.0.6:5080
OPTIONS sip:10.0.0.6:5080 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bKf8a4.0f4ccc25.0.
To: sip:10.0.0.6:5080.
From: <sip:opensips at 10.0.0.5>;tag=5f71290c3118e5f12e8e1e2bda5b046a-b78e.
CSeq: 10 OPTIONS.
Call-ID: 6d19857b67133c70-6120 at 10.0.0.5.
Max-Forwards: 70.
Content-Length: 0.
User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
.

U 2016/04/29 13:14:47.606003 10.0.0.6:5080 -> 10.0.0.5:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bKf8a4.0f4ccc25.0.
From: <sip:opensips at 10.0.0.5>;tag=5f71290c3118e5f12e8e1e2bda5b046a-b78e.
To: <sip:10.0.0.6:5080>;tag=accp7Bpp1m4vN.
Call-ID: 6d19857b67133c70-6120 at 10.0.0.5.
CSeq: 10 OPTIONS.
Contact: <sip:10.0.0.6:5080>.
User-Agent:
FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Length: 0.
.

U 2016/04/29 13:14:50.064942 10.0.0.4:5060 -> 10.0.0.6:5080
INVITE sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
Record-Route: <sip:10.0.0.4;lr;did=ba7.1907a81>.
Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
Contact: <sip:91.121.129.159:5060>.
Content-Type: application/sdp.
CSeq: 493433999 INVITE.
From: "003228800555" <sip:003228800555 at sip3.ovh.fr
;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
Max-Forwards: 26.
Record-Route: <sip:91.121.129.159:5060;lr>.
To: <sip:0972550428 at 10.7.1.68;user=phone>.
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd.
Allow:
REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
User-Agent: Cirpack/v4.70 (gw_sip).
Content-Length: 445.
.
v=0.
o=cp10 146193569025 146193569025 IN IP4 10.7.1.122.
s=SIP Call.
c=IN IP4 91.121.129.155.
t=0 0.
m=audio 30178 RTP/AVP 18 4 0 8 125 111 101.
b=AS:21.
a=rtpmap:18 G729/8000/1.
a=fmtp:18 annexb=no.
a=rtpmap:4 G723/8000/1.
a=fmtp:4 annexa=no.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:8 PCMA/8000/1.
a=rtpmap:125 CLEARMODE/8000/1.
a=rtpmap:111 iLBC/8000/1.
a=fmtp:111 mode=30.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendrecv.

U 2016/04/29 13:14:50.065335 10.0.0.6:5080 -> 10.0.0.4:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd.
Record-Route: <sip:10.0.0.4;lr;did=ba7.1907a81>.
Record-Route: <sip:91.121.129.159:5060;lr>.
From: "003228800555" <sip:003228800555 at sip3.ovh.fr
;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
To: <sip:0972550428 at 10.7.1.68;user=phone>.
Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
CSeq: 493433999 INVITE.
User-Agent:
FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
Content-Length: 0.
.

U 2016/04/29 13:14:50.067423 10.0.0.6:5080 -> 10.0.0.4:5060
SIP/2.0 480 Temporarily Unavailable.
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd.
Max-Forwards: 26.
From: "003228800555" <sip:003228800555 at sip3.ovh.fr
;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=BN5e966SyXtFH.
Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
CSeq: 493433999 INVITE.
User-Agent:
FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
Remote-Party-ID: "opensips" <sip:opensips at 10.7.1.68
>;party=calling;privacy=off;screen=no.
.

U 2016/04/29 13:14:50.068695 10.0.0.4:5060 -> 10.0.0.6:5080
ACK sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
From: "003228800555" <sip:003228800555 at sip3.ovh.fr
;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=BN5e966SyXtFH.
CSeq: 493433999 ACK.
Max-Forwards: 70.
User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.
.

U 2016/04/29 13:14:52.030194 10.0.0.4:5060 -> 10.0.0.6:5080
INVITE sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
Record-Route: <sip:10.0.0.4;lr;did=8df.224b90a>.
Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
Contact: <sip:91.121.129.159:5060>.
Content-Type: application/sdp.
CSeq: 493434154 INVITE.
From: "003228800555" <sip:003228800555 at sip3.ovh.fr
;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
Max-Forwards: 26.
Record-Route: <sip:91.121.129.159:5060;lr>.
To: <sip:0972550428 at 10.7.1.68;user=phone>.
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-FZAL-34d2c970-111a1872.
Allow:
REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
User-Agent: Cirpack/v4.70 (gw_sip).
Content-Length: 315.
.
v=0.
o=cp10 146193569231 146193569231 IN IP4 10.7.16.156.
s=SIP Call.
c=IN IP4 91.121.129.144.
t=0 0.
m=audio 33498 RTP/AVP 18 0 8 101.
b=AS:21.
a=rtpmap:18 G729/8000/1.
a=fmtp:18 annexb=no.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:8 PCMA/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:30.
a=sendrecv.

U 2016/04/29 13:14:52.030438 10.0.0.6:5080 -> 10.0.0.4:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-FZAL-34d2c970-111a1872.
Record-Route: <sip:10.0.0.4;lr;did=8df.224b90a>.
Record-Route: <sip:91.121.129.159:5060;lr>.
From: "003228800555" <sip:003228800555 at sip3.ovh.fr
;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
To: <sip:0972550428 at 10.7.1.68;user=phone>.
Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
CSeq: 493434154 INVITE.
User-Agent:
FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
Content-Length: 0.
.

U 2016/04/29 13:14:52.032086 10.0.0.6:5080 -> 10.0.0.4:5060
SIP/2.0 480 Temporarily Unavailable.
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
Via: SIP/2.0/UDP 91.121.129.159:5060;branch=z9hG4bK-FZAL-34d2c970-111a1872.
Max-Forwards: 26.
From: "003228800555" <sip:003228800555 at sip3.ovh.fr
;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=cyy7a2QXU6g2c.
Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
CSeq: 493434154 INVITE.
User-Agent:
FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
Remote-Party-ID: "opensips" <sip:opensips at 10.7.1.68
>;party=calling;privacy=off;screen=no.
.

U 2016/04/29 13:14:52.032599 10.0.0.4:5060 -> 10.0.0.6:5080
ACK sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
From: "003228800555" <sip:003228800555 at sip3.ovh.fr
;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=cyy7a2QXU6g2c.
CSeq: 493434154 ACK.
Max-Forwards: 70.
User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.
.

U 2016/04/29 13:15:00.670131 10.0.0.4:5060 -> 10.0.0.6:5080
OPTIONS sip:10.0.0.6:5080 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK03bd.4fd62c01.0.
To: sip:10.0.0.6:5080.
From: <sip:opensips at 10.0.0.4>;tag=ea1fc35981b38c23d4118e13e0c2a171-fd59.
CSeq: 10 OPTIONS.
Call-ID: 24a5dabd71563ec7-8608 at 10.0.0.4.
Max-Forwards: 70.
Content-Length: 0.
User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
.

U 2016/04/29 13:15:00.670514 10.0.0.6:5080 -> 10.0.0.4:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK03bd.4fd62c01.0.
From: <sip:opensips at 10.0.0.4>;tag=ea1fc35981b38c23d4118e13e0c2a171-fd59.
To: <sip:10.0.0.6:5080>;tag=D7Q0cX80rF7mr.
Call-ID: 24a5dabd71563ec7-8608 at 10.0.0.4.
CSeq: 10 OPTIONS.
Contact: <sip:10.0.0.6:5080>.
User-Agent:
FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Length: 0.
.

U 2016/04/29 13:15:09.009870 10.0.0.6:5080 -> 10.0.0.5:5060
OPTIONS sip:10.0.0.5;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.6:5080;rport;branch=z9hG4bK2vXj0Nc68SXXQ.
Max-Forwards: 70.
From: <sip:10.0.0.5>;tag=egHSerS4NrX7K.
To: <sip:10.0.0.5>.
Call-ID: 3d12aa3e-88af-1234-b492-000d3a2233e0.
CSeq: 90631748 OPTIONS.
User-Agent:
FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Length: 0.
.

U 2016/04/29 13:15:09.009969 10.0.0.6:5080 -> 10.0.0.4:5060
OPTIONS sip:10.0.0.4;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.6:5080;rport;branch=z9hG4bK35pB2gX952KgK.
Max-Forwards: 70.
From: <sip:10.0.0.4>;tag=FSajgKa8j1KtF.
To: <sip:10.0.0.4>.
Call-ID: 3d12b080-88af-1234-b492-000d3a2233e0.
CSeq: 90631749 OPTIONS.
User-Agent:
FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY.
Supported: timer, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Length: 0.
.

U 2016/04/29 13:15:09.010629 10.0.0.4:5060 -> 10.0.0.6:5080
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP 10.0.0.6:5080
;received=10.0.0.6;rport=5080;branch=z9hG4bK35pB2gX952KgK.
From: <sip:10.0.0.4>;tag=FSajgKa8j1KtF.
To: <sip:10.0.0.4>;tag=61890dad1e908c702027bf054a266115.990f.
Call-ID: 3d12b080-88af-1234-b492-000d3a2233e0.
CSeq: 90631749 OPTIONS.
Server: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.
.

U 2016/04/29 13:15:09.011254 10.0.0.5:5060 -> 10.0.0.6:5080
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP 10.0.0.6:5080
;received=10.0.0.6;rport=5080;branch=z9hG4bK2vXj0Nc68SXXQ.
From: <sip:10.0.0.5>;tag=egHSerS4NrX7K.
To: <sip:10.0.0.5>;tag=2bf6dd58f2c27bac032ef66b671d14ff.52bb.
Call-ID: 3d12aa3e-88af-1234-b492-000d3a2233e0.
CSeq: 90631748 OPTIONS.
Server: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.
.


NDUWAYEZU Joselyne

2016-04-29 0:58 GMT+02:00 Michael Collins <msc at freeswitch.org>:

> It sounds like the dialplan never makes it to your extension. Get a debug
> log of the call and put it on pastebin.freeswitch.org. We can probably
> help you figure it out from there.
>
> -MSC
>
> On Thu, Apr 28, 2016 at 12:40 AM, Nduwayezu, Joselyne <35633 at heb.be>
> wrote:
>
>>
>> Hello,
>>
>> I would like to invoke php script in the dial plan.
>> I defined the dialplan as follos:
>>
>>
>> <include>
>>  <extension name="outbound-socket">
>>   <condition field="${sip_to_user}" expression="(972550428)$">
>>    <action application="set"
>> data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
>>    <action application="socket" data="10.0.0.6:8021 async full"/>-->
>>   </condition>
>>  </extension>
>> </include>
>>
>> And my php script is:
>>
>>
>> <?php
>>   // set a couple of things so we dont kill the system
>>  ob_implicit_flush(true);
>>  set_time_limit(30);
>>  // Open stdin so we can read the data in
>>  $in = fopen("php://stdin", "r");
>>  // Connect
>>  echo "connect\n\n";
>>  // Answer
>>  echo "sendmsg\n";
>>  echo "call-command: execute\n";
>>  echo "execute-app-name: answer\n\n";
>>  // Play a prompt
>>  echo "sendmsg\n";
>>  echo "call-command: execute\n";
>>  echo "execute-app-name: playback\n";
>>  echo "execute-app-arg: \
>>  /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n
>> \
>>  ";
>>  // Wait
>>  sleep(5);
>>  // Hangup
>>  echo "sendmsg\n";
>>  echo "call-command: hangup\n\n";
>>  fclose($in);
>>  ?>
>>
>> When i dial the DID number, the call state is "call setup" for a while,
>> and after i'm asked to leave a message because the number is busy or
>> unreachable.
>>
>> I tried different sockets in the dialplan (5060, 5080, 8040) with no
>> results
>>
>> Any adeas of what is happening?
>>
>> Thanks
>>
>>
>> NDUWAYEZU Joselyne
>>
>> Haute ?cole de Bruxelles
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Haute ?cole de Bruxelles
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From abalashov at evaristesys.com  Fri Apr 29 18:35:06 2016
From: abalashov at evaristesys.com (Alex Balashov)
Date: Fri, 29 Apr 2016 10:35:06 -0400
Subject: [Freeswitch-users] "To field" in sip header
In-Reply-To: <CAFLo2QznqHxnmDDQbSTJrNy0yao+8qSOgOYyeW5+zE1MqFyRHQ@mail.gmail.com>
References: <CAH5f_dvTgwNftgqCx0TSYhdmfjUF5B+0=gaDWNVb1jVYHdX5tA@mail.gmail.com>
	<CAH5f_dvQ-4Tt9pnkd0GyYMu3QzXswvRREp==u_=37L+LQ_zxQw@mail.gmail.com>
	<CAH5f_ds2t9ur7sNoF-6pA8r6pBs6C-OUMVcsO-yX+k0rRxtojw@mail.gmail.com>
	<20160429130343.5394500.87280.235762@evaristesys.com>
	<CAFLo2QznqHxnmDDQbSTJrNy0yao+8qSOgOYyeW5+zE1MqFyRHQ@mail.gmail.com>
Message-ID: <20160429143506.5394500.98441.235791@evaristesys.com>

?:-)

If the FS instance is registering with the provider, their registrar may be doing the very RFC-compliant thing of sending the original AOR contact binding in the RURI, e.g. something like sip:line1 at my.ip.
?
However, any origination ITSP should be able to override this and put the dialed number in the RURI instead.?

--
Alex?Balashov?|?Principal?|?Evariste?Systems?LLC
1447?Peachtree?Street?NE,?Suite?700
Atlanta,?GA?30309
United?States

Tel:?+1-800-250-5920?(toll-free)?/?+1-678-954-0671 (direct)
Web:?http://www.evaristesys.com/, http://www.csrpswitch.com/

Sent?from?my?BlackBerry.
? Original Message ?
From: Luis Daniel Lucio Quiroz
Sent: Friday, April 29, 2016 09:14
To: FreeSWITCH Users Help
Reply To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] "To field" in sip header



From richard.mace at gmail.com  Fri Apr 29 19:41:58 2016
From: richard.mace at gmail.com (Richard Mace)
Date: Fri, 29 Apr 2016 16:41:58 +0100
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CAKbxfG8bH2A84Ng08RWU6QgwM9gPgB2s5JQa9nG=1jHtaP2Vtg@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
	<CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
	<CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>
	<CAK+KT3-3uF2-MF=WTmhV0KoBGjQtoeBsVzHX8JM3P-yzai_EmQ@mail.gmail.com>
	<CACOYK=jRVZLGtqOXg+SfTme1VAgGbJJ+1itb=DBr7u4-7JyAZA@mail.gmail.com>
	<CAK+KT393aDZEDaTorZpS=Z8qCjez+Huewx0Hf7L6oFKoA3Pgzw@mail.gmail.com>
	<CAJ0sghQDkZFMtvOG_s0nMt_8TLg-03hV-NJdqnqbk009h98y8A@mail.gmail.com>
	<CAK+KT3_gmfYHK13oh1gstS_Legh63PgxxKZeo7jJwg543b=K8A@mail.gmail.com>
	<CACOYK=jwm6qUgsYtcfC27zQa0tF=EGhXdBhgaFjphDJCo+tcjw@mail.gmail.com>
	<CAKbxfG8bH2A84Ng08RWU6QgwM9gPgB2s5JQa9nG=1jHtaP2Vtg@mail.gmail.com>
Message-ID: <CAK+KT3-kMj0p+i+t0ApzjYUdpKPaOTqE1F-aN5pJos2oyrf_9A@mail.gmail.com>

On 29 April 2016 at 01:54, Anthony Minessale <anthony.minessale at gmail.com>
wrote:

>

> Hopefully its adding our repo and installing the packages as the method
of setting up the box?
>

?Hi Tony,

Yes, that's exactly what it's doing.

Richard?
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From richard.mace at gmail.com  Fri Apr 29 19:43:36 2016
From: richard.mace at gmail.com (Richard Mace)
Date: Fri, 29 Apr 2016 16:43:36 +0100
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CACOYK=jwm6qUgsYtcfC27zQa0tF=EGhXdBhgaFjphDJCo+tcjw@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
	<CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
	<CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>
	<CAK+KT3-3uF2-MF=WTmhV0KoBGjQtoeBsVzHX8JM3P-yzai_EmQ@mail.gmail.com>
	<CACOYK=jRVZLGtqOXg+SfTme1VAgGbJJ+1itb=DBr7u4-7JyAZA@mail.gmail.com>
	<CAK+KT393aDZEDaTorZpS=Z8qCjez+Huewx0Hf7L6oFKoA3Pgzw@mail.gmail.com>
	<CAJ0sghQDkZFMtvOG_s0nMt_8TLg-03hV-NJdqnqbk009h98y8A@mail.gmail.com>
	<CAK+KT3_gmfYHK13oh1gstS_Legh63PgxxKZeo7jJwg543b=K8A@mail.gmail.com>
	<CACOYK=jwm6qUgsYtcfC27zQa0tF=EGhXdBhgaFjphDJCo+tcjw@mail.gmail.com>
Message-ID: <CAK+KT38dap+nBqZ92Fs9QNTXr8jPdZ5keH3XDhmeS5DGQY_Gyw@mail.gmail.com>

On 29 April 2016 at 01:29, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

>

> Open source without an easy way to browse the code? I couldn't find any
links apart from the binary downloads.

?The source will be available soon, and certainly by the time it's actually
released as non beta code.

Richard?
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From ssinyagin at gmail.com  Fri Apr 29 20:11:08 2016
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 29 Apr 2016 18:11:08 +0200
Subject: [Freeswitch-users] [ANN] RoCKSwitch Builder Beta 1 released
In-Reply-To: <CAK+KT38dap+nBqZ92Fs9QNTXr8jPdZ5keH3XDhmeS5DGQY_Gyw@mail.gmail.com>
References: <CAK+KT39HLgn4iwMi-7PPbrskQa1p5txO9FrWfPvKruGQFuO+cQ@mail.gmail.com>
	<CAK+KT39=Bf_+A2uJzejBjQdvNOH6CcT83084xXGqnjWaHgepzQ@mail.gmail.com>
	<CAPNTnnp9XFX_9Kh9fEWa0Wc0sdOJETjeP0_5Cyq-Tqkj+C1txQ@mail.gmail.com>
	<CAK+KT3-3uF2-MF=WTmhV0KoBGjQtoeBsVzHX8JM3P-yzai_EmQ@mail.gmail.com>
	<CACOYK=jRVZLGtqOXg+SfTme1VAgGbJJ+1itb=DBr7u4-7JyAZA@mail.gmail.com>
	<CAK+KT393aDZEDaTorZpS=Z8qCjez+Huewx0Hf7L6oFKoA3Pgzw@mail.gmail.com>
	<CAJ0sghQDkZFMtvOG_s0nMt_8TLg-03hV-NJdqnqbk009h98y8A@mail.gmail.com>
	<CAK+KT3_gmfYHK13oh1gstS_Legh63PgxxKZeo7jJwg543b=K8A@mail.gmail.com>
	<CACOYK=jwm6qUgsYtcfC27zQa0tF=EGhXdBhgaFjphDJCo+tcjw@mail.gmail.com>
	<CAK+KT38dap+nBqZ92Fs9QNTXr8jPdZ5keH3XDhmeS5DGQY_Gyw@mail.gmail.com>
Message-ID: <CACOYK=gYa67TMkjWVDQLxgnkC--WTC3xvJHB1S1Hi-aqwTsMPg@mail.gmail.com>

I think that's the opposite of the idea of open source: by publishing your
code, you give a chance to get feedback before the product is released.

But hey, installing from packages IS less than 10 commands in a shell
prompt, so what the heck are we talking about.
On Apr 29, 2016 17:44, "Richard Mace" <richard.mace at gmail.com> wrote:

> On 29 April 2016 at 01:29, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
>
> >
>
> > Open source without an easy way to browse the code? I couldn't find any
> links apart from the binary downloads.
>
> ?The source will be available soon, and certainly by the time it's
> actually released as non beta code.
>
> Richard?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From arun.kan at gmail.com  Fri Apr 29 00:31:30 2016
From: arun.kan at gmail.com (Arun K)
Date: Fri, 29 Apr 2016 02:01:30 +0530
Subject: [Freeswitch-users] FreeSWITCH external Gateway registration Issue
Message-ID: <CAA229ioUKWVdxa24nNioYZM-mpBMa2-NMf9p5aTVU=hXXpo-Bw@mail.gmail.com>

Dear Group,

I'm a novice with FreeSwitch and the below doubt is very basic. But I could
not able to resolve it for the past whole week. Kindly assist me.


   1. I'm trying to register my Free switch gateway with my SIP provider.
   the external profile is configured at  C:\Program
   Files\FreeSWITCH\conf\sip_profiles\external\example.xml as below:
   2.
   3. <include>
   4. <gateway name="Rcom">
   5.     <param name="username" value="+914438000100"/>
   6.     <param name="password" value="xxxxxxx"/>
   7.     <param name="from-domain" value="bangalore.relianceims.in"/>
   8.     <param name="proxy" value="10.237.246.225"/>
   9. </gateway>
   10. </include>
   11.
   12. Registering attempts failed with below logs:
   13.
   14.
   15. 2016-04-28 23:08:27.905875 [NOTICE] sofia_reg.c:448 Registering
   bangalore.relianceims.in
   16. 2016-04-28 23:08:28.905875 [NOTICE] sofia_reg.c:448 Registering Rcom
   17. send 596 bytes to udp/[10.237.246.225]:5060 at 17:38:28.921500:
   18.
    ------------------------------------------------------------------------
   19.    REGISTER sip:10.237.246.225;transport=udp SIP/2.0
   20.    Via: SIP/2.0/UDP 10.16.83.21;rport;branch=z9hG4bK11QceD1pmcUjN
   21.    Max-Forwards: 70
   22.    From: <sip:+914438000100 at bangalore.relianceims.in
   >;tag=1H4KN7By10r8D
   23.    To: <sip:+914438000100 at bangalore.relianceims.in>
   24.    Call-ID: e6089137-443f-4b24-8375-37fa7617e1c2
   25.    CSeq: 90596490 REGISTER
   26.    Contact: <sip:gw+Rcom at 10.16.83.21:5060;transport=udp;gw=Rcom>
   27.    Expires: 3600
   28.    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~32bit
   29.   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
   REGISTER, REFER, NOTIFY
   30.    Supported: timer, path, replaces
   31.    Content-Length: 0
   32.
   33.
    ------------------------------------------------------------------------
   34. recv 456 bytes from udp/[10.237.246.225]:5060 at 17:38:28.937125:
   35.
    ------------------------------------------------------------------------
   36.    SIP/2.0 404 Not Found
   37.    Via: SIP/2.0/UDP 10.16.83.21;received=10.16.83.21;rport=5060
   ;branch=z9hG4bK11QceD1pmcUjN
   38.    To: <sip:+914438000100 at bangalore.relianceims.in
   >;tag=ztesipZcoVXTKT0*1-7-16648*ijib.1
   39.    From: <sip:+914438000100 at bangalore.relianceims.in
   >;tag=1H4KN7By10r8D
   40.    Call-ID: e6089137-443f-4b24-8375-37fa7617e1c2
   41.    CSeq: 90596490 REGISTER
   42.    X-ZTE-Cause: "CSCF-1.7.
   BC005000-BC00DA26-BC005319-BC00521F.icscf1.bangalore.relianceims.in"
   43.    Content-Length: 0
   44.
   45.
    ------------------------------------------------------------------------
   46. 2016-04-28 23:08:28.937125 [ERR] sofia_reg.c:2367 Rcom Failed
   Registration with status Not Found [404]. failure #4
   47.
   48.
   49. IF i try to register x-lite with SIP Provider with the following
   parameters it is getting registered successfully.


   1. username               : +914438000100
   2. password               : xxxxxx
   3. domain                 : bangalore.relianceims.in
   4. Authorisation user name: +914438000100 at bangalore.relianceims.in
   5. Proxy                  : 10.237.246.225
   6.
   7. Only thing i do not use at Free switch configuration that i use at
   X-lite is Authorisation user name. without this parameter, x-lite
   registeration also fails. So can any of the experts assist me as where
   should i provide this parameter at free switch. Else how to solve this
   problem. Kindly advice.
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From mike at jerris.com  Fri Apr 29 20:49:12 2016
From: mike at jerris.com (Michael Jerris)
Date: Fri, 29 Apr 2016 12:49:12 -0400
Subject: [Freeswitch-users] mod_nibblebill issue with recording
In-Reply-To: <CAMRuXGgAJn_7S_HhAVk6QfwPcgQXhono9Lc9Kt7nB5NxoqBjFQ@mail.gmail.com>
References: <CAMRuXGgAJn_7S_HhAVk6QfwPcgQXhono9Lc9Kt7nB5NxoqBjFQ@mail.gmail.com>
Message-ID: <3884B57C-046C-49B8-A921-144D073E1DC3@jerris.com>

how are you uploading to amazon?  sounds like whatever that is is blocking execute state.

> On Apr 29, 2016, at 12:59 AM, Murugan Pandian <manpower13.cse at gmail.com> wrote:
> 
> HI,
> 
>    We  are using nibllebill for billing and AWS S3 for call recording storage ,When we record the call using record_session it will upload recorded WAV file to s3(AWS).
> 
> Issue:
> 
>           1.mod_nibblebill bill ammount include record_session wav file upload time
> 
> 
> Example:
> 
>   if I talk 2 mins and record_session take 20/sec to upload WAV file to S3(AWS) the billed amount 2.20/mins (0.1 per mins total amount 0.3 )   but billed time only shows 2 mins  



From phenix at vfemail.net  Fri Apr 29 22:53:18 2016
From: phenix at vfemail.net (Tanguy)
Date: Fri, 29 Apr 2016 20:53:18 +0200
Subject: [Freeswitch-users] "To field" in sip header
In-Reply-To: <CAH5f_ds2t9ur7sNoF-6pA8r6pBs6C-OUMVcsO-yX+k0rRxtojw@mail.gmail.com>
References: <CAH5f_dvTgwNftgqCx0TSYhdmfjUF5B+0=gaDWNVb1jVYHdX5tA@mail.gmail.com>
	<CAH5f_dvQ-4Tt9pnkd0GyYMu3QzXswvRREp==u_=37L+LQ_zxQw@mail.gmail.com>
	<CAH5f_ds2t9ur7sNoF-6pA8r6pBs6C-OUMVcsO-yX+k0rRxtojw@mail.gmail.com>
Message-ID: <5723AD9E.5000202@vfemail.net>

Hello

I think you may add this line in your gateway configuration.

<param name="extension" value="auto_to_user"/>

Then you will be able to use a standard dialplan with destination_number



On 29/04/2016 14:47, thomas peterseil wrote:
>
> hello,
> i have a sip provider with multiple DID?s and they send me all calls 
> with the same invite, only the To field in the SIP header is 
> different. how can i read the To field in the SIP header to be able to 
> make a call routing based on the To field? i tried the following rule 
> but it doesnt work:
>
> ---------------------
>
> <include>
> <extension name="public_test1">
>         <condition regex="all">
>         <regex  field="destination_number" expression="^provider1$"/>
>         <regex  field="${sip_h_To}" expression="43717072377"/>
>
>                 <action application="set" 
> data="effective_caller_id_number=7021"/>
>                 <action application="set" 
> data="effective_caller_id_name=thomas"/>
>                 <action application="answer"/>
>                 <action application="transfer" data="6011 XML inbound1 
> "/>
> </condition>
> </extension>
>
> ----------------
>
> the To field in the SIP header looks like this:
>
> To: <sip:43717072377 at 193.43.148.37 
> <mailto:sip%3A43717072377 at 193.43.148.37>>
>
> thanks a lot for help!
>
> best regards,
> thomas
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Sat Apr 30 00:15:28 2016
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Apr 2016 13:15:28 -0700
Subject: [Freeswitch-users] Freeswitch php scripts
In-Reply-To: <CAFSzUtbbJD=1D_Zpo6d9-8OawuLAEb9p9VouxgfNY3QfN08ANA@mail.gmail.com>
References: <CAFSzUtb38-PivnmMbJweu_6g8t1_aOpzJd-3x5hPwnpS-sQFgQ@mail.gmail.com>
	<CAKzWOxUnu-SGrAHBpYCzoVQF0A_cez8Q+o0UEdiwGSR+r_Fnag@mail.gmail.com>
	<CAFSzUtbbJD=1D_Zpo6d9-8OawuLAEb9p9VouxgfNY3QfN08ANA@mail.gmail.com>
Message-ID: <CAKzWOxXi=gpmcTO6Y4NJ+zc7G7vpeQjF-J1273WiWBSXy2Snzw@mail.gmail.com>

Hello,

That is a SIP trace, which in this case isn't going to help you. You need
to see the output from fs_cli or /var/log/freeswitch.log on a failed call.
Capture the output from your terminal program or copy out of
freeswitch.log. Copy that into pastebin.freeswitch.org. Try username of
"pastebin" and password "freeswitch"

-MSC

On Fri, Apr 29, 2016 at 6:29 AM, Nduwayezu, Joselyne <35633 at heb.be> wrote:

> Hello,
> I can not log on pastebin.freeswitch.org
> This is the log when i make a call
>
> I still having the problem
>
>
> root at back-1:/usr/local/freeswitch/conf/dialplan/public# ngrep -d eth0 -t
> -W byline "$1" port 5060 -q
> interface: eth0 (10.0.0.0/255.255.255.0)
> filter: (ip or ip6) and ( port 5060 )
> U 2016/04/29 13:14:47.605641 10.0.0.5:5060 -> 10.0.0.6:5080
> OPTIONS sip:10.0.0.6:5080 SIP/2.0.
> Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bKf8a4.0f4ccc25.0.
> To: sip:10.0.0.6:5080.
> From: <sip:opensips at 10.0.0.5>;tag=5f71290c3118e5f12e8e1e2bda5b046a-b78e.
> CSeq: 10 OPTIONS.
> Call-ID: 6d19857b67133c70-6120 at 10.0.0.5.
> Max-Forwards: 70.
> Content-Length: 0.
> User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
> .
>
> U 2016/04/29 13:14:47.606003 10.0.0.6:5080 -> 10.0.0.5:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bKf8a4.0f4ccc25.0.
> From: <sip:opensips at 10.0.0.5>;tag=5f71290c3118e5f12e8e1e2bda5b046a-b78e.
> To: <sip:10.0.0.6:5080>;tag=accp7Bpp1m4vN.
> Call-ID: 6d19857b67133c70-6120 at 10.0.0.5.
> CSeq: 10 OPTIONS.
> Contact: <sip:10.0.0.6:5080>.
> User-Agent:
> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Length: 0.
> .
>
> U 2016/04/29 13:14:50.064942 10.0.0.4:5060 -> 10.0.0.6:5080
> INVITE sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
> Record-Route: <sip:10.0.0.4;lr;did=ba7.1907a81>.
> Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
> Contact: <sip:91.121.129.159:5060>.
> Content-Type: application/sdp.
> CSeq: 493433999 INVITE.
> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
> ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
> Max-Forwards: 26.
> Record-Route: <sip:91.121.129.159:5060;lr>.
> To: <sip:0972550428 at 10.7.1.68;user=phone>.
> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd.
> Allow:
> REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
> User-Agent: Cirpack/v4.70 (gw_sip).
> Content-Length: 445.
> .
> v=0.
> o=cp10 146193569025 146193569025 IN IP4 10.7.1.122.
> s=SIP Call.
> c=IN IP4 91.121.129.155.
> t=0 0.
> m=audio 30178 RTP/AVP 18 4 0 8 125 111 101.
> b=AS:21.
> a=rtpmap:18 G729/8000/1.
> a=fmtp:18 annexb=no.
> a=rtpmap:4 G723/8000/1.
> a=fmtp:4 annexa=no.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:8 PCMA/8000/1.
> a=rtpmap:125 CLEARMODE/8000/1.
> a=rtpmap:111 iLBC/8000/1.
> a=fmtp:111 mode=30.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:30.
> a=sendrecv.
>
> U 2016/04/29 13:14:50.065335 10.0.0.6:5080 -> 10.0.0.4:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd.
> Record-Route: <sip:10.0.0.4;lr;did=ba7.1907a81>.
> Record-Route: <sip:91.121.129.159:5060;lr>.
> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
> ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
> To: <sip:0972550428 at 10.7.1.68;user=phone>.
> Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
> CSeq: 493433999 INVITE.
> User-Agent:
> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
> Content-Length: 0.
> .
>
> U 2016/04/29 13:14:50.067423 10.0.0.6:5080 -> 10.0.0.4:5060
> SIP/2.0 480 Temporarily Unavailable.
> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd.
> Max-Forwards: 26.
> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
> ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
> To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=BN5e966SyXtFH.
> Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
> CSeq: 493433999 INVITE.
> User-Agent:
> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> Remote-Party-ID: "opensips" <sip:opensips at 10.7.1.68
> >;party=calling;privacy=off;screen=no.
> .
>
> U 2016/04/29 13:14:50.068695 10.0.0.4:5060 -> 10.0.0.6:5080
> ACK sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
> ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
> Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
> To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=BN5e966SyXtFH.
> CSeq: 493433999 ACK.
> Max-Forwards: 70.
> User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
> Content-Length: 0.
> .
>
> U 2016/04/29 13:14:52.030194 10.0.0.4:5060 -> 10.0.0.6:5080
> INVITE sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
> Record-Route: <sip:10.0.0.4;lr;did=8df.224b90a>.
> Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
> Contact: <sip:91.121.129.159:5060>.
> Content-Type: application/sdp.
> CSeq: 493434154 INVITE.
> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
> ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
> Max-Forwards: 26.
> Record-Route: <sip:91.121.129.159:5060;lr>.
> To: <sip:0972550428 at 10.7.1.68;user=phone>.
> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-FZAL-34d2c970-111a1872.
> Allow:
> REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
> User-Agent: Cirpack/v4.70 (gw_sip).
> Content-Length: 315.
> .
> v=0.
> o=cp10 146193569231 146193569231 IN IP4 10.7.16.156.
> s=SIP Call.
> c=IN IP4 91.121.129.144.
> t=0 0.
> m=audio 33498 RTP/AVP 18 0 8 101.
> b=AS:21.
> a=rtpmap:18 G729/8000/1.
> a=fmtp:18 annexb=no.
> a=rtpmap:0 PCMU/8000/1.
> a=rtpmap:8 PCMA/8000/1.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:30.
> a=sendrecv.
>
> U 2016/04/29 13:14:52.030438 10.0.0.6:5080 -> 10.0.0.4:5060
> SIP/2.0 100 Trying.
> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-FZAL-34d2c970-111a1872.
> Record-Route: <sip:10.0.0.4;lr;did=8df.224b90a>.
> Record-Route: <sip:91.121.129.159:5060;lr>.
> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
> ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
> To: <sip:0972550428 at 10.7.1.68;user=phone>.
> Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
> CSeq: 493434154 INVITE.
> User-Agent:
> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
> Content-Length: 0.
> .
>
> U 2016/04/29 13:14:52.032086 10.0.0.6:5080 -> 10.0.0.4:5060
> SIP/2.0 480 Temporarily Unavailable.
> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
> Via: SIP/2.0/UDP 91.121.129.159:5060
> ;branch=z9hG4bK-FZAL-34d2c970-111a1872.
> Max-Forwards: 26.
> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
> ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
> To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=cyy7a2QXU6g2c.
> Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
> CSeq: 493434154 INVITE.
> User-Agent:
> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> Remote-Party-ID: "opensips" <sip:opensips at 10.7.1.68
> >;party=calling;privacy=off;screen=no.
> .
>
> U 2016/04/29 13:14:52.032599 10.0.0.4:5060 -> 10.0.0.6:5080
> ACK sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
> ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
> Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
> To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=cyy7a2QXU6g2c.
> CSeq: 493434154 ACK.
> Max-Forwards: 70.
> User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
> Content-Length: 0.
> .
>
> U 2016/04/29 13:15:00.670131 10.0.0.4:5060 -> 10.0.0.6:5080
> OPTIONS sip:10.0.0.6:5080 SIP/2.0.
> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK03bd.4fd62c01.0.
> To: sip:10.0.0.6:5080.
> From: <sip:opensips at 10.0.0.4>;tag=ea1fc35981b38c23d4118e13e0c2a171-fd59.
> CSeq: 10 OPTIONS.
> Call-ID: 24a5dabd71563ec7-8608 at 10.0.0.4.
> Max-Forwards: 70.
> Content-Length: 0.
> User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
> .
>
> U 2016/04/29 13:15:00.670514 10.0.0.6:5080 -> 10.0.0.4:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK03bd.4fd62c01.0.
> From: <sip:opensips at 10.0.0.4>;tag=ea1fc35981b38c23d4118e13e0c2a171-fd59.
> To: <sip:10.0.0.6:5080>;tag=D7Q0cX80rF7mr.
> Call-ID: 24a5dabd71563ec7-8608 at 10.0.0.4.
> CSeq: 10 OPTIONS.
> Contact: <sip:10.0.0.6:5080>.
> User-Agent:
> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
> Accept: application/sdp.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Length: 0.
> .
>
> U 2016/04/29 13:15:09.009870 10.0.0.6:5080 -> 10.0.0.5:5060
> OPTIONS sip:10.0.0.5;transport=udp SIP/2.0.
> Via: SIP/2.0/UDP 10.0.0.6:5080;rport;branch=z9hG4bK2vXj0Nc68SXXQ.
> Max-Forwards: 70.
> From: <sip:10.0.0.5>;tag=egHSerS4NrX7K.
> To: <sip:10.0.0.5>.
> Call-ID: 3d12aa3e-88af-1234-b492-000d3a2233e0.
> CSeq: 90631748 OPTIONS.
> User-Agent:
> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Length: 0.
> .
>
> U 2016/04/29 13:15:09.009969 10.0.0.6:5080 -> 10.0.0.4:5060
> OPTIONS sip:10.0.0.4;transport=udp SIP/2.0.
> Via: SIP/2.0/UDP 10.0.0.6:5080;rport;branch=z9hG4bK35pB2gX952KgK.
> Max-Forwards: 70.
> From: <sip:10.0.0.4>;tag=FSajgKa8j1KtF.
> To: <sip:10.0.0.4>.
> Call-ID: 3d12b080-88af-1234-b492-000d3a2233e0.
> CSeq: 90631749 OPTIONS.
> User-Agent:
> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> REFER, NOTIFY.
> Supported: timer, path, replaces.
> Allow-Events: talk, hold, conference, refer.
> Content-Length: 0.
> .
>
> U 2016/04/29 13:15:09.010629 10.0.0.4:5060 -> 10.0.0.6:5080
> SIP/2.0 484 Address Incomplete.
> Via: SIP/2.0/UDP 10.0.0.6:5080
> ;received=10.0.0.6;rport=5080;branch=z9hG4bK35pB2gX952KgK.
> From: <sip:10.0.0.4>;tag=FSajgKa8j1KtF.
> To: <sip:10.0.0.4>;tag=61890dad1e908c702027bf054a266115.990f.
> Call-ID: 3d12b080-88af-1234-b492-000d3a2233e0.
> CSeq: 90631749 OPTIONS.
> Server: OpenSIPS (2.1.2 (x86_64/linux)).
> Content-Length: 0.
> .
>
> U 2016/04/29 13:15:09.011254 10.0.0.5:5060 -> 10.0.0.6:5080
> SIP/2.0 484 Address Incomplete.
> Via: SIP/2.0/UDP 10.0.0.6:5080
> ;received=10.0.0.6;rport=5080;branch=z9hG4bK2vXj0Nc68SXXQ.
> From: <sip:10.0.0.5>;tag=egHSerS4NrX7K.
> To: <sip:10.0.0.5>;tag=2bf6dd58f2c27bac032ef66b671d14ff.52bb.
> Call-ID: 3d12aa3e-88af-1234-b492-000d3a2233e0.
> CSeq: 90631748 OPTIONS.
> Server: OpenSIPS (2.1.2 (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> NDUWAYEZU Joselyne
>
> 2016-04-29 0:58 GMT+02:00 Michael Collins <msc at freeswitch.org>:
>
>> It sounds like the dialplan never makes it to your extension. Get a debug
>> log of the call and put it on pastebin.freeswitch.org. We can probably
>> help you figure it out from there.
>>
>> -MSC
>>
>> On Thu, Apr 28, 2016 at 12:40 AM, Nduwayezu, Joselyne <35633 at heb.be>
>> wrote:
>>
>>>
>>> Hello,
>>>
>>> I would like to invoke php script in the dial plan.
>>> I defined the dialplan as follos:
>>>
>>>
>>> <include>
>>>  <extension name="outbound-socket">
>>>   <condition field="${sip_to_user}" expression="(972550428)$">
>>>    <action application="set"
>>> data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
>>>    <action application="socket" data="10.0.0.6:8021 async full"/>-->
>>>   </condition>
>>>  </extension>
>>> </include>
>>>
>>> And my php script is:
>>>
>>>
>>> <?php
>>>   // set a couple of things so we dont kill the system
>>>  ob_implicit_flush(true);
>>>  set_time_limit(30);
>>>  // Open stdin so we can read the data in
>>>  $in = fopen("php://stdin", "r");
>>>  // Connect
>>>  echo "connect\n\n";
>>>  // Answer
>>>  echo "sendmsg\n";
>>>  echo "call-command: execute\n";
>>>  echo "execute-app-name: answer\n\n";
>>>  // Play a prompt
>>>  echo "sendmsg\n";
>>>  echo "call-command: execute\n";
>>>  echo "execute-app-name: playback\n";
>>>  echo "execute-app-arg: \
>>>  /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n
>>> \
>>>  ";
>>>  // Wait
>>>  sleep(5);
>>>  // Hangup
>>>  echo "sendmsg\n";
>>>  echo "call-command: hangup\n\n";
>>>  fclose($in);
>>>  ?>
>>>
>>> When i dial the DID number, the call state is "call setup" for a while,
>>> and after i'm asked to leave a message because the number is busy or
>>> unreachable.
>>>
>>> I tried different sockets in the dialplan (5060, 5080, 8040) with no
>>> results
>>>
>>> Any adeas of what is happening?
>>>
>>> Thanks
>>>
>>>
>>> NDUWAYEZU Joselyne
>>>
>>> Haute ?cole de Bruxelles
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> Haute ?cole de Bruxelles
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
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From fernando at softov.com.br  Sat Apr 30 03:07:36 2016
From: fernando at softov.com.br (Luiz Fernando Softov)
Date: Fri, 29 Apr 2016 19:07:36 -0400
Subject: [Freeswitch-users] FreeSwitch - Clone - TAG
In-Reply-To: <CAHtxdDehdwBRSQVRjZF524HBnyZZ-DGs=Fzrrdrec5tPW6Qn8Q@mail.gmail.com>
References: <CAKy26iPnLG36h4O=BOMdQ_rtWDtCnyaQdRXopMk6WHst7s+KVg@mail.gmail.com>
	<00a101d1a14f$222407d0$666c1770$@freeswitch.org>
	<CAKy26iOWBMO+i86BKD3eKsCv2t_XkN9bzYnGP5uPhD9cxORG8A@mail.gmail.com>
	<00ad01d1a152$4826af20$d8740d60$@freeswitch.org>
	<CAKy26iP5TzxfUdK_K1LPZ=R9e4oXwA1G+BeSETiTvwAF5AeuuQ@mail.gmail.com>
	<026a01d1a16d$70a6ecb0$51f4c610$@freeswitch.org>
	<CAKy26iPNvQtPRtfcfGu95YZhUuuik+u5YyvpB3956MhoJAJRLw@mail.gmail.com>
	<CAHtxdDehdwBRSQVRjZF524HBnyZZ-DGs=Fzrrdrec5tPW6Qn8Q@mail.gmail.com>
Message-ID: <CAKy26iNCbcWGe95xgmBaj3Pz5awBj6AwshweH+dgGTQxXODm+A@mail.gmail.com>

??
Done.

In FreeBSD I needed to install

pkg install opus /* compile manual with --enable-shared */
pkg install mpg123
pkg install mp3


But I need to remove the FreeSwitch repository directory and clone again.


git clone https://stash.freeswitch.org/scm/fs/freeswitch.git
cd freeswitch
git checkout master

./bootstrap.sh -j

./configure --prefix=/brb_main/freeswitch --disable-fhs
--enable-core-pgsql-support CC=/usr/bin/clang CXX=/usr/bin/clang++
gmake
gmake install


- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - -
ps, to compile mod_gsmopen
pkg install gsm
pkg install gsmlib

And of course, apply the patch, but my mod_gsmopen isn't the same (i have
made some changes), so i don't need anymore.
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
- - - - - -

now I think all installation is OK, i need to test this in production.

?I only need now to edit the source of sofia to add some parameters in
event WRONG_CALL_STATE?

?Thanks a lot for the help.?

2016-04-29 3:28 GMT-04:00 Sergey Safarov <s.safarov at gmail.com>:

> Please look
>
>    1. FS-7705 <https://freeswitch.org/jira/browse/FS-7705>
>    2. PR293
>    <https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/293/diff>
>
> This
> Requires: libshout >= 2.3.1
> Requires: libmpg123 >= 1.20.1
>
> Check your ports.
>
> ??, 28 ???. 2016 ?. ? 20:10, Luiz Fernando Softov <fernando at softov.com.br
> >:
>
>> It's the same with
>>
>> making all mod_shout
>> gmake[4]: Entering directory
>> '/usr/dev/freeswitch_backend/freeswitch/src/mod/formats/mod_shout'
>> Makefile:907: *** You must install libshout3-dev to build mod_shout.
>> Stop.
>>
>> First I install libshout from ports, reclone, bootstrap, configure e when
>> make no effect
>> Then i download from internet, and its the same thing.
>>
>> I think this will be the same with libmpg123.
>>
>> About the link, i only need to clone it and make install clean, or need
>> to clone it in a correct directory?
>> Like
>> /usr/dev/freeswitch_backend/freeswitch/libs/opus
>> /usr/dev/freeswitch_backend/freeswitch/libs/opus-1.1
>>
>> ????
>>
>>
>>
>>
>>
>> 2016-04-28 12:45 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>>
>>> https://freeswitch.org/stash/projects/SD/repos/opus/browse
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
>>> Fernando Softov
>>> *Sent:* Thursday, April 28, 2016 11:29 AM
>>>
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>>>
>>>
>>>
>>> I cloned and trying to compile now, in FreeBSD
>>>
>>> making all mod_opus
>>>
>>> gmake[4]: Entering directory
>>> '/usr/dev/freeswitch_backend/freeswitch/src/mod/codecs/mod_opus'
>>>
>>> Makefile:897: *** You must install libopus-dev to build mod_opus.  Stop.
>>>
>>> I already compile and install opus from
>>> downloads.xiph.org/releases/opus/
>>>
>>> since in FreeBSD doesn't have a libopus-dev
>>>
>>> Is there another place to get libopus?
>>>
>>>
>>>
>>> 2016-04-28 9:31 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>>>
>>> There is no tag for latest master. Its just master? git checkout master
>>> && git pull?
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
>>> Fernando Softov
>>> *Sent:* Thursday, April 28, 2016 8:20 AM
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Subject:* Re: [Freeswitch-users] FreeSwitch - Clone - TAG
>>>
>>>
>>>
>>> Yeah, latest master, but witch tag?
>>>
>>> 1.4.x
>>> 1.5.final
>>> 1.6.0
>>> 1.6.7
>>> 1.7.0
>>>
>>> There are so many tags, so i have lost my mind witch one.
>>>
>>>
>>> I already thy to compile 1.6 in FreeBSD, but without success, there are
>>> dependencies in mod_opus, and another thing i can't remember now.
>>>
>>> Which is the difference between tags?
>>>
>>>
>>>
>>>
>>>
>>> 2016-04-28 9:09 GMT-04:00 Ken Rice <krice at freeswitch.org>:
>>>
>>> Latest master?. There are numerous bugs and potential security issues
>>> that have been resolved since then.
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Luiz
>>> Fernando Softov
>>> *Sent:* Thursday, April 28, 2016 7:48 AM
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Subject:* [Freeswitch-users] FreeSwitch - Clone - TAG
>>>
>>>
>>>
>>> Hi,
>>>
>>> I don't remember why, but I'm using 1.5.final (branch master) in
>>> production (~1 year), and it's working fine, with 30 simultaneous calls.
>>>
>>>
>>> I will re-clone the FreeSwitch, because some changes made, that i need.
>>>
>>> Someone recommend to clone another tag?
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From manpower13.cse at gmail.com  Sat Apr 30 09:43:21 2016
From: manpower13.cse at gmail.com (Murugan Pandian)
Date: Sat, 30 Apr 2016 11:13:21 +0530
Subject: [Freeswitch-users] mod_nibblebill issue with recording
In-Reply-To: <3884B57C-046C-49B8-A921-144D073E1DC3@jerris.com>
References: <CAMRuXGgAJn_7S_HhAVk6QfwPcgQXhono9Lc9Kt7nB5NxoqBjFQ@mail.gmail.com>
	<3884B57C-046C-49B8-A921-144D073E1DC3@jerris.com>
Message-ID: <CAMRuXGi0P7yh=j5=s-xDV17zeSgphfCtMsjK=3CANCHnGdqYog@mail.gmail.com>

Michael,


  We are using mod_http_cache to upload recorted WAV file to S3(AWS)

 Bellow is my billing and upload lua script

  session:execute('set','nibble_account='..acount);
  session:execute('set','nibble_rate='..rate);
  session:execute('export','execute_on_answer=record_session http_cache://
http://bucketname.s3-ap-southeast-1.amazonaws.com/'..file_name)


Thanks



On Fri, Apr 29, 2016 at 10:19 PM, Michael Jerris <mike at jerris.com> wrote:

> how are you uploading to amazon?  sounds like whatever that is is blocking
> execute state.
>
> > On Apr 29, 2016, at 12:59 AM, Murugan Pandian <manpower13.cse at gmail.com>
> wrote:
> >
> > HI,
> >
> >    We  are using nibllebill for billing and AWS S3 for call recording
> storage ,When we record the call using record_session it will upload
> recorded WAV file to s3(AWS).
> >
> > Issue:
> >
> >           1.mod_nibblebill bill ammount include record_session wav file
> upload time
> >
> >
> > Example:
> >
> >   if I talk 2 mins and record_session take 20/sec to upload WAV file to
> S3(AWS) the billed amount 2.20/mins (0.1 per mins total amount 0.3 )   but
> billed time only shows 2 mins
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Sat Apr 30 12:13:20 2016
From: asilva at wirelessmundi.com (Antonio Silva)
Date: Sat, 30 Apr 2016 10:13:20 +0200
Subject: [Freeswitch-users] FreeSWITCH external Gateway registration
	Issue
In-Reply-To: <CAA229ioUKWVdxa24nNioYZM-mpBMa2-NMf9p5aTVU=hXXpo-Bw@mail.gmail.com>
References: <CAA229ioUKWVdxa24nNioYZM-mpBMa2-NMf9p5aTVU=hXXpo-Bw@mail.gmail.com>
Message-ID: <ca85acaf-ed03-88ce-9485-9a57b6f36482@wirelessmundi.com>

hi,

did you try *auth-username *? or *extension*? you can check the gateway 
parameters here:

https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Gateway+Authentication+Params

also take a look at: 
https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration


On 04/28/2016 10:31 PM, Arun K wrote:
> Dear Group,
>
> I'm a novice with FreeSwitch and the below doubt is very basic. But I 
> could not able to resolve it for the past whole week. Kindly assist me.
>
> 1.
>     I'm trying to register my Free switch gateway with my SIP
>     provider. the external profile is configured at  C:\Program
>     Files\FreeSWITCH\conf\sip_profiles\external\example.xml as below:
> 2.
> 3.
>     <include>
> 4.
>     <gateway name="Rcom">
> 5.
>       <param name="username" value="+914438000100"/>
> 6.
>       <param name="password" value="xxxxxxx"/>
> 7.
>       <param name="from-domain" value="bangalore.relianceims.in
>     <http://bangalore.relianceims.in>"/>
> 8.
>       <param name="proxy" value="10.237.246.225"/>
> 9.
>     </gateway>
>10.
>     </include>
>11.
>12.
>     Registering attempts failed with below logs:
>13.
>14.
>15.
>     2016-04-28 23:08:27.905875 [NOTICE] sofia_reg.c:448 Registering
>     bangalore.relianceims.in <http://bangalore.relianceims.in>
>16.
>     2016-04-28 23:08:28.905875 [NOTICE] sofia_reg.c:448 Registering Rcom
>17.
>     send 596 bytes to udp/[10.237.246.225]:5060 at 17:38:28.921500:
>18.
>      ------------------------------------------------------------------------
>19.
>      REGISTER sip:10.237.246.225;transport=udp SIP/2.0
>20.
>      Via: SIP/2.0/UDP 10.16.83.21;rport;branch=z9hG4bK11QceD1pmcUjN
>21.
>      Max-Forwards: 70
>22.
>      From: <sip:+914438000100 at bangalore.relianceims.in
>     <http://bangalore.relianceims.in>>;tag=1H4KN7By10r8D
>23.
>      To: <sip:+914438000100 at bangalore.relianceims.in
>     <http://bangalore.relianceims.in>>
>24.
>      Call-ID: e6089137-443f-4b24-8375-37fa7617e1c2
>25.
>      CSeq: 90596490 REGISTER
>26.
>      Contact: <sip:gw+Rcom at 10.16.83.21
>     <mailto:sip%3Agw%2BRcom at 10.16.83.21>:5060;transport=udp;gw=Rcom>
>27.
>      Expires: 3600
>28.
>      User-Agent: FreeSWITCH-mod_sofia/1.5.15b~32bit
>29.
>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>     REGISTER, REFER, NOTIFY
>30.
>      Supported: timer, path, replaces
>31.
>      Content-Length: 0
>32.
>33.
>      ------------------------------------------------------------------------
>34.
>     recv 456 bytes from udp/[10.237.246.225]:5060 at 17:38:28.937125:
>35.
>      ------------------------------------------------------------------------
>36.
>      SIP/2.0 404 Not Found
>37.
>      Via: SIP/2.0/UDP
>     10.16.83.21;received=10.16.83.21;rport=5060;branch=z9hG4bK11QceD1pmcUjN
>38.
>      To: <sip:+914438000100 at bangalore.relianceims.in
>     <http://bangalore.relianceims.in>>;tag=ztesipZcoVXTKT0*1-7-16648*ijib.1
>39.
>      From: <sip:+914438000100 at bangalore.relianceims.in
>     <http://bangalore.relianceims.in>>;tag=1H4KN7By10r8D
>40.
>      Call-ID: e6089137-443f-4b24-8375-37fa7617e1c2
>41.
>      CSeq: 90596490 REGISTER
>42.
>      X-ZTE-Cause:
>     "CSCF-1.7.BC005000-BC00DA26-BC005319-BC00521F.icscf1.bangalore.relianceims.in
>     <http://BC005000-BC00DA26-BC005319-BC00521F.icscf1.bangalore.relianceims.in>"
>43.
>      Content-Length: 0
>44.
>45.
>      ------------------------------------------------------------------------
>46.
>     2016-04-28 23:08:28.937125 [ERR] sofia_reg.c:2367 Rcom Failed
>     Registration with status Not Found [404]. failure #4
>47.
>48.
>49.
>     IF i try to register x-lite with SIP Provider with the following
>     parameters it is getting registered successfully.
>
> 1.
>     username               : +914438000100
> 2.
>     password               : xxxxxx
> 3.
>     domain                 : bangalore.relianceims.in
>     <http://bangalore.relianceims.in>
> 4.
>     Authorisation user name: +914438000100 at bangalore.relianceims.in
>     <http://bangalore.relianceims.in>
> 5.
>     Proxy                  : 10.237.246.225
> 6.
> 7.
>     Only thing i do not use at Free switch configuration that i use at
>     X-lite is Authorisation user name. without this parameter, x-lite
>     registeration also fails. So can any of the experts assist me as
>     where should i provide this parameter at free switch. Else how to
>     solve this problem. Kindly advice.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From thomas.peterseil at gmail.com  Sat Apr 30 13:51:26 2016
From: thomas.peterseil at gmail.com (thomas peterseil)
Date: Sat, 30 Apr 2016 11:51:26 +0200
Subject: [Freeswitch-users] "To field" in sip header
In-Reply-To: <5723AD9E.5000202@vfemail.net>
References: <CAH5f_dvTgwNftgqCx0TSYhdmfjUF5B+0=gaDWNVb1jVYHdX5tA@mail.gmail.com>
	<CAH5f_dvQ-4Tt9pnkd0GyYMu3QzXswvRREp==u_=37L+LQ_zxQw@mail.gmail.com>
	<CAH5f_ds2t9ur7sNoF-6pA8r6pBs6C-OUMVcsO-yX+k0rRxtojw@mail.gmail.com>
	<5723AD9E.5000202@vfemail.net>
Message-ID: <CAH5f_duvTA8hhu00q2-vQ3G0rFugqeTbfYAA6Yv3BgfePdNmgw@mail.gmail.com>

hi,
thanks a lot, that solved my problem!
but i am still wondering if it?s possible to read and modify sip header
fields with FS at inbound calls.

best,
thomas
Am 29.04.2016 8:54 nachm. schrieb "Tanguy" <phenix at vfemail.net>:

> Hello
>
> I think you may add this line in your gateway configuration.
>
> <param name="extension" value="auto_to_user"/>
>
> Then you will be able to use a standard dialplan with destination_number
>
>
>
> On 29/04/2016 14:47, thomas peterseil wrote:
>
> hello,
> i have a sip provider with multiple DID?s and they send me all calls with
> the same invite, only the To field in the SIP header is different. how can
> i read the To field in the SIP header to be able to make a call routing
> based on the To field? i tried the following rule but it doesnt work:
>
> ---------------------
>
> <include>
>
> <extension
> name="public_test1">
>
>         <condition
> regex="all">
>
>         <regex  field="destination_number"
> expression="^provider1$"/>
>         <regex  field="${sip_h_To}"
> expression="43717072377"/>
>
>
>                 <action application="set"
> data="effective_caller_id_number=7021"/>
>                 <action application="set"
> data="effective_caller_id_name=thomas"/>
>                 <action
> application="answer"/>
>
>                 <action application="transfer" data="6011 XML inbound1
> "/>
>
> </condition>
>
> </extension>
>
> ----------------
>
> the To field in the SIP header looks like this:
>
> To: <sip:43717072377 at 193.43.148.37>
>
> thanks a lot for help!
>
> best regards,
> thomas
>
>
> _________________________________________________________________________
> 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 35633 at heb.be  Sat Apr 30 15:09:10 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Sat, 30 Apr 2016 13:09:10 +0200
Subject: [Freeswitch-users] Internal profile doesn't exist
In-Reply-To: <CAFETSWgCLAyrOagQoNdOUqG4AcfNhOJUdD-Tq08K-WvrU-DyRg@mail.gmail.com>
References: <CAFETSWgCLAyrOagQoNdOUqG4AcfNhOJUdD-Tq08K-WvrU-DyRg@mail.gmail.com>
Message-ID: <CAFSzUtb26dbbERUs0DwYnaKf8OEocwoMKcHAKk7brMuA=hsaAA@mail.gmail.com>

Try to desactivate the files concerning IPv6 in sip_profiles directory

NDUWAYEZU Joselyne

2016-04-29 11:30 GMT+02:00 amani mansour <amani.mansour2 at gmail.com>:

> Hi all,
> I don't know what is the problem ,
>
> Remarque  ,my IP address is correct in vars.xml (static)
> when i tape sofia status
> it affiche external,gateways,internal_ipv6 but the internal profile isn't
> exist
> is there someone who know the solusion please ?
>
> [image: pasted1]
> Best regards
> Amani
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Haute ?cole de Bruxelles
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From 35633 at heb.be  Sat Apr 30 15:14:51 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Sat, 30 Apr 2016 13:14:51 +0200
Subject: [Freeswitch-users] Problem with ESL.so
In-Reply-To: <CAF7jizKFDPp-8=Q4+4bHbV62pYQMbMVohHjCNeMXG-9ZDLU7fg@mail.gmail.com>
References: <CAFSzUtY24z0=Gzr6CzUH9iNd6AQpXPkA9rdtUHcuWbhxLTK8Qw@mail.gmail.com>
	<CAF7jizKFDPp-8=Q4+4bHbV62pYQMbMVohHjCNeMXG-9ZDLU7fg@mail.gmail.com>
Message-ID: <CAFSzUtZ8w48FxnuDLADB89vKCznbWye3o5Kb2fr6tyTe=5wfcw@mail.gmail.com>

Yes i upgraded PHP. I've noticed that php.ini is present in two differents
locations:
/etc/php5/apache2/php.ini and  /etc/php5/cli/php.ini. Which one do i have
to deal with?

And when you say if my .ini are still pointing to old location, what do you
mean by that?

NDUWAYEZU Joselyne

2016-04-29 14:37 GMT+02:00 Benjamin Cropley <benjamin.cropley at gmail.com>:

> Did you upgrade PHP? Are your .ini files still pointing to old locations?
>
> On Fri, Apr 29, 2016 at 12:22 PM, Nduwayezu, Joselyne <35633 at heb.be>
> wrote:
>
>>
>> I,ve followed the freeswitch cookbook in order to user the ESL. I
>> installed  all the libraries needed.
>> But when i run ESL.so, i have a segmentation fault which result in error
>> when i run a php script. the error: unable to load dynamic library
>> /usr/lib/php5/20131226.
>>  Any help?
>>
>> NDUWAYEZU Joselyne
>>
>> Haute ?cole de Bruxelles
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Haute ?cole de Bruxelles
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From brian at linuxpenguins.xyz  Sat Apr 30 15:32:54 2016
From: brian at linuxpenguins.xyz (Brian May)
Date: Sat, 30 Apr 2016 21:32:54 +1000
Subject: [Freeswitch-users] Message is less than minimum record length
In-Reply-To: <87wpoai62u.fsf@prune.linuxpenguins.xyz>
References: <87wpoai62u.fsf@prune.linuxpenguins.xyz>
Message-ID: <87mvobwcux.fsf@prune.linuxpenguins.xyz>

Nobody got any ideas on the following problem? I would rather not have
to discard my freeswitch solution because the voicemail is not working,
however this is somewhat important feature...

Or should I file a bug report?

This only happens for calls via my external SIP provider. Local calls to
voice mail work fine. Also calls received via the external SIP provider
work fine if not diverted to voice mail.

Thanks


Brian May <brian at linuxpenguins.xyz> writes:

> Some reason my freeswitch system has suddenly stopped listening for
> messages. Instead it immediately cuts the talker off and says the
> message is too short. Any ideas? How do I debug this?
>
> i am going to conduct an audio test later on to make sure this is
> working, last I checked phone calls were working fine however.
>
>
> 2016-04-07 10:09:31.172293 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16 at 8000hz 1 channels 20ms
> 2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:1910 done playing file /var/lib/freeswitch/storage/voicemail/default/microcomaustralia.com.au/2000/greeting_1.wav
> 2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en]
> 2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-record_message.wav] (en:en)
> 2016-04-07 10:09:40.232300 [DEBUG] mod_loopback.c:601 loopback/voicemail-b CHANNEL KILL
> 2016-04-07 10:09:40.232300 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16 at 8000hz 1 channels 20ms
> 2016-04-07 10:09:44.792345 [DEBUG] switch_ivr_play_say.c:1910 done playing file /usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.wav
> 2016-04-07 10:09:45.912334 [DEBUG] switch_ivr_play_say.c:559 Raw Codec Activated, ready to waste resources!
> 2016-04-07 10:09:45.912334 [DEBUG] switch_ivr_play_say.c:673 Raw Codec Activated
> 2016-04-07 10:09:45.912334 [DEBUG] switch_core_codec.c:221 loopback/voicemail-b Push codec L16:100
> 2016-04-07 10:09:47.912303 [DEBUG] switch_core_codec.c:246 loopback/voicemail-b Restore previous codec PCMA:8.
> 2016-04-07 10:09:47.932384 [DEBUG] mod_voicemail.c:1245 Message is less than minimum record length: 3, discarding it.
> 2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en]
> 2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-too-small.wav] (en:en)
> 2016-04-07 10:09:47.932384 [DEBUG] mod_loopback.c:601 loopback/voicemail-b CHANNEL KILL
> 2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16 at 8000hz 1 channels 20ms
> 2016-04-07 10:09:51.492336 [DEBUG] sofia.c:6858 Channel sofia/external/61419503953 at 203.31.79.18 entering state [terminated][487]
> 2016-04-07 10:09:51.492336 [NOTICE] sofia.c:7879 Hangup sofia/external/61419503953 at 203.31.79.18 [CS_EXECUTE] [ORIGINATOR_CANCEL]
> 2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:707 sofia/external/61419503953 at 203.31.79.18 ending bridge by request from read function
> 2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:780 BRIDGE THREAD DONE [sofia/external/61419503953 at 203.31.79.18]
> 2016-04-07 10:09:51.492336 [DEBUG] mod_loopback.c:601 loopback/voicemail-a CHANNEL KILL
> 2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:701 sofia/external/61419503953 at 203.31.79.18 ending bridge by request from write function
>
> -- 
> Brian May <brian at linuxpenguins.xyz>
> https://linuxpenguins.xyz/brian/
-- 
Brian May <brian at linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/


From 35633 at heb.be  Sat Apr 30 16:10:50 2016
From: 35633 at heb.be (Nduwayezu, Joselyne)
Date: Sat, 30 Apr 2016 14:10:50 +0200
Subject: [Freeswitch-users] Freeswitch php scripts
In-Reply-To: <CAKzWOxXi=gpmcTO6Y4NJ+zc7G7vpeQjF-J1273WiWBSXy2Snzw@mail.gmail.com>
References: <CAFSzUtb38-PivnmMbJweu_6g8t1_aOpzJd-3x5hPwnpS-sQFgQ@mail.gmail.com>
	<CAKzWOxUnu-SGrAHBpYCzoVQF0A_cez8Q+o0UEdiwGSR+r_Fnag@mail.gmail.com>
	<CAFSzUtbbJD=1D_Zpo6d9-8OawuLAEb9p9VouxgfNY3QfN08ANA@mail.gmail.com>
	<CAKzWOxXi=gpmcTO6Y4NJ+zc7G7vpeQjF-J1273WiWBSXy2Snzw@mail.gmail.com>
Message-ID: <CAFSzUtam33kz=ZXTnHk+_x6LCUG=+ZH+=4--Re7Dgh+_Ds45AQ@mail.gmail.com>

I've just posted the output of the call from fs_cli, you can have a look on
patebin.freeswitch.org. The post is from fransjos.
Thank you

NDUWAYEZU Joselyne

2016-04-29 22:15 GMT+02:00 Michael Collins <msc at freeswitch.org>:

> Hello,
>
> That is a SIP trace, which in this case isn't going to help you. You need
> to see the output from fs_cli or /var/log/freeswitch.log on a failed call.
> Capture the output from your terminal program or copy out of
> freeswitch.log. Copy that into pastebin.freeswitch.org. Try username of
> "pastebin" and password "freeswitch"
>
> -MSC
>
> On Fri, Apr 29, 2016 at 6:29 AM, Nduwayezu, Joselyne <35633 at heb.be> wrote:
>
>> Hello,
>> I can not log on pastebin.freeswitch.org
>> This is the log when i make a call
>>
>> I still having the problem
>>
>>
>> root at back-1:/usr/local/freeswitch/conf/dialplan/public# ngrep -d eth0 -t
>> -W byline "$1" port 5060 -q
>> interface: eth0 (10.0.0.0/255.255.255.0)
>> filter: (ip or ip6) and ( port 5060 )
>> U 2016/04/29 13:14:47.605641 10.0.0.5:5060 -> 10.0.0.6:5080
>> OPTIONS sip:10.0.0.6:5080 SIP/2.0.
>> Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bKf8a4.0f4ccc25.0.
>> To: sip:10.0.0.6:5080.
>> From: <sip:opensips at 10.0.0.5>;tag=5f71290c3118e5f12e8e1e2bda5b046a-b78e.
>> CSeq: 10 OPTIONS.
>> Call-ID: 6d19857b67133c70-6120 at 10.0.0.5.
>> Max-Forwards: 70.
>> Content-Length: 0.
>> User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
>> .
>>
>> U 2016/04/29 13:14:47.606003 10.0.0.6:5080 -> 10.0.0.5:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bKf8a4.0f4ccc25.0.
>> From: <sip:opensips at 10.0.0.5>;tag=5f71290c3118e5f12e8e1e2bda5b046a-b78e.
>> To: <sip:10.0.0.6:5080>;tag=accp7Bpp1m4vN.
>> Call-ID: 6d19857b67133c70-6120 at 10.0.0.5.
>> CSeq: 10 OPTIONS.
>> Contact: <sip:10.0.0.6:5080>.
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
>> Accept: application/sdp.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Content-Length: 0.
>> .
>>
>> U 2016/04/29 13:14:50.064942 10.0.0.4:5060 -> 10.0.0.6:5080
>> INVITE sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
>> Record-Route: <sip:10.0.0.4;lr;did=ba7.1907a81>.
>> Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
>> Contact: <sip:91.121.129.159:5060>.
>> Content-Type: application/sdp.
>> CSeq: 493433999 INVITE.
>> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
>> ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
>> Max-Forwards: 26.
>> Record-Route: <sip:91.121.129.159:5060;lr>.
>> To: <sip:0972550428 at 10.7.1.68;user=phone>.
>> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
>> Via: SIP/2.0/UDP 91.121.129.159:5060
>> ;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd.
>> Allow:
>> REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
>> User-Agent: Cirpack/v4.70 (gw_sip).
>> Content-Length: 445.
>> .
>> v=0.
>> o=cp10 146193569025 146193569025 IN IP4 10.7.1.122.
>> s=SIP Call.
>> c=IN IP4 91.121.129.155.
>> t=0 0.
>> m=audio 30178 RTP/AVP 18 4 0 8 125 111 101.
>> b=AS:21.
>> a=rtpmap:18 G729/8000/1.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:4 G723/8000/1.
>> a=fmtp:4 annexa=no.
>> a=rtpmap:0 PCMU/8000/1.
>> a=rtpmap:8 PCMA/8000/1.
>> a=rtpmap:125 CLEARMODE/8000/1.
>> a=rtpmap:111 iLBC/8000/1.
>> a=fmtp:111 mode=30.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> a=ptime:30.
>> a=sendrecv.
>>
>> U 2016/04/29 13:14:50.065335 10.0.0.6:5080 -> 10.0.0.4:5060
>> SIP/2.0 100 Trying.
>> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
>> Via: SIP/2.0/UDP 91.121.129.159:5060
>> ;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd.
>> Record-Route: <sip:10.0.0.4;lr;did=ba7.1907a81>.
>> Record-Route: <sip:91.121.129.159:5060;lr>.
>> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
>> ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
>> To: <sip:0972550428 at 10.7.1.68;user=phone>.
>> Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
>> CSeq: 493433999 INVITE.
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
>> Content-Length: 0.
>> .
>>
>> U 2016/04/29 13:14:50.067423 10.0.0.6:5080 -> 10.0.0.4:5060
>> SIP/2.0 480 Temporarily Unavailable.
>> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
>> Via: SIP/2.0/UDP 91.121.129.159:5060
>> ;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd.
>> Max-Forwards: 26.
>> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
>> ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
>> To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=BN5e966SyXtFH.
>> Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
>> CSeq: 493433999 INVITE.
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
>> Accept: application/sdp.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
>> Content-Length: 0.
>> Remote-Party-ID: "opensips" <sip:opensips at 10.7.1.68
>> >;party=calling;privacy=off;screen=no.
>> .
>>
>> U 2016/04/29 13:14:50.068695 10.0.0.4:5060 -> 10.0.0.6:5080
>> ACK sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
>> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0.
>> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
>> ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06.
>> Call-ID: 03545-MR-1fc39fe6-23cc798d3 at sip3.ovh.fr.
>> To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=BN5e966SyXtFH.
>> CSeq: 493433999 ACK.
>> Max-Forwards: 70.
>> User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
>> Content-Length: 0.
>> .
>>
>> U 2016/04/29 13:14:52.030194 10.0.0.4:5060 -> 10.0.0.6:5080
>> INVITE sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
>> Record-Route: <sip:10.0.0.4;lr;did=8df.224b90a>.
>> Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
>> Contact: <sip:91.121.129.159:5060>.
>> Content-Type: application/sdp.
>> CSeq: 493434154 INVITE.
>> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
>> ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
>> Max-Forwards: 26.
>> Record-Route: <sip:91.121.129.159:5060;lr>.
>> To: <sip:0972550428 at 10.7.1.68;user=phone>.
>> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
>> Via: SIP/2.0/UDP 91.121.129.159:5060
>> ;branch=z9hG4bK-FZAL-34d2c970-111a1872.
>> Allow:
>> REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK.
>> User-Agent: Cirpack/v4.70 (gw_sip).
>> Content-Length: 315.
>> .
>> v=0.
>> o=cp10 146193569231 146193569231 IN IP4 10.7.16.156.
>> s=SIP Call.
>> c=IN IP4 91.121.129.144.
>> t=0 0.
>> m=audio 33498 RTP/AVP 18 0 8 101.
>> b=AS:21.
>> a=rtpmap:18 G729/8000/1.
>> a=fmtp:18 annexb=no.
>> a=rtpmap:0 PCMU/8000/1.
>> a=rtpmap:8 PCMA/8000/1.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> a=ptime:30.
>> a=sendrecv.
>>
>> U 2016/04/29 13:14:52.030438 10.0.0.6:5080 -> 10.0.0.4:5060
>> SIP/2.0 100 Trying.
>> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
>> Via: SIP/2.0/UDP 91.121.129.159:5060
>> ;branch=z9hG4bK-FZAL-34d2c970-111a1872.
>> Record-Route: <sip:10.0.0.4;lr;did=8df.224b90a>.
>> Record-Route: <sip:91.121.129.159:5060;lr>.
>> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
>> ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
>> To: <sip:0972550428 at 10.7.1.68;user=phone>.
>> Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
>> CSeq: 493434154 INVITE.
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
>> Content-Length: 0.
>> .
>>
>> U 2016/04/29 13:14:52.032086 10.0.0.6:5080 -> 10.0.0.4:5060
>> SIP/2.0 480 Temporarily Unavailable.
>> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
>> Via: SIP/2.0/UDP 91.121.129.159:5060
>> ;branch=z9hG4bK-FZAL-34d2c970-111a1872.
>> Max-Forwards: 26.
>> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
>> ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
>> To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=cyy7a2QXU6g2c.
>> Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
>> CSeq: 493434154 INVITE.
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
>> Accept: application/sdp.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
>> Content-Length: 0.
>> Remote-Party-ID: "opensips" <sip:opensips at 10.7.1.68
>> >;party=calling;privacy=off;screen=no.
>> .
>>
>> U 2016/04/29 13:14:52.032599 10.0.0.4:5060 -> 10.0.0.6:5080
>> ACK sip:opensips at 10.0.0.4:5060;transport=udp SIP/2.0.
>> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0.
>> From: "003228800555" <sip:003228800555 at sip3.ovh.fr
>> ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46.
>> Call-ID: 13530-HA-1fc3a093-020bb53b6 at sip3.ovh.fr.
>> To: <sip:0972550428 at 10.7.1.68;user=phone>;tag=cyy7a2QXU6g2c.
>> CSeq: 493434154 ACK.
>> Max-Forwards: 70.
>> User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
>> Content-Length: 0.
>> .
>>
>> U 2016/04/29 13:15:00.670131 10.0.0.4:5060 -> 10.0.0.6:5080
>> OPTIONS sip:10.0.0.6:5080 SIP/2.0.
>> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK03bd.4fd62c01.0.
>> To: sip:10.0.0.6:5080.
>> From: <sip:opensips at 10.0.0.4>;tag=ea1fc35981b38c23d4118e13e0c2a171-fd59.
>> CSeq: 10 OPTIONS.
>> Call-ID: 24a5dabd71563ec7-8608 at 10.0.0.4.
>> Max-Forwards: 70.
>> Content-Length: 0.
>> User-Agent: OpenSIPS (2.1.2 (x86_64/linux)).
>> .
>>
>> U 2016/04/29 13:15:00.670514 10.0.0.6:5080 -> 10.0.0.4:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK03bd.4fd62c01.0.
>> From: <sip:opensips at 10.0.0.4>;tag=ea1fc35981b38c23d4118e13e0c2a171-fd59.
>> To: <sip:10.0.0.6:5080>;tag=D7Q0cX80rF7mr.
>> Call-ID: 24a5dabd71563ec7-8608 at 10.0.0.4.
>> CSeq: 10 OPTIONS.
>> Contact: <sip:10.0.0.6:5080>.
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
>> Accept: application/sdp.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Content-Length: 0.
>> .
>>
>> U 2016/04/29 13:15:09.009870 10.0.0.6:5080 -> 10.0.0.5:5060
>> OPTIONS sip:10.0.0.5;transport=udp SIP/2.0.
>> Via: SIP/2.0/UDP 10.0.0.6:5080;rport;branch=z9hG4bK2vXj0Nc68SXXQ.
>> Max-Forwards: 70.
>> From: <sip:10.0.0.5>;tag=egHSerS4NrX7K.
>> To: <sip:10.0.0.5>.
>> Call-ID: 3d12aa3e-88af-1234-b492-000d3a2233e0.
>> CSeq: 90631748 OPTIONS.
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Content-Length: 0.
>> .
>>
>> U 2016/04/29 13:15:09.009969 10.0.0.6:5080 -> 10.0.0.4:5060
>> OPTIONS sip:10.0.0.4;transport=udp SIP/2.0.
>> Via: SIP/2.0/UDP 10.0.0.6:5080;rport;branch=z9hG4bK35pB2gX952KgK.
>> Max-Forwards: 70.
>> From: <sip:10.0.0.4>;tag=FSajgKa8j1KtF.
>> To: <sip:10.0.0.4>.
>> Call-ID: 3d12b080-88af-1234-b492-000d3a2233e0.
>> CSeq: 90631749 OPTIONS.
>> User-Agent:
>> FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY.
>> Supported: timer, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Content-Length: 0.
>> .
>>
>> U 2016/04/29 13:15:09.010629 10.0.0.4:5060 -> 10.0.0.6:5080
>> SIP/2.0 484 Address Incomplete.
>> Via: SIP/2.0/UDP 10.0.0.6:5080
>> ;received=10.0.0.6;rport=5080;branch=z9hG4bK35pB2gX952KgK.
>> From: <sip:10.0.0.4>;tag=FSajgKa8j1KtF.
>> To: <sip:10.0.0.4>;tag=61890dad1e908c702027bf054a266115.990f.
>> Call-ID: 3d12b080-88af-1234-b492-000d3a2233e0.
>> CSeq: 90631749 OPTIONS.
>> Server: OpenSIPS (2.1.2 (x86_64/linux)).
>> Content-Length: 0.
>> .
>>
>> U 2016/04/29 13:15:09.011254 10.0.0.5:5060 -> 10.0.0.6:5080
>> SIP/2.0 484 Address Incomplete.
>> Via: SIP/2.0/UDP 10.0.0.6:5080
>> ;received=10.0.0.6;rport=5080;branch=z9hG4bK2vXj0Nc68SXXQ.
>> From: <sip:10.0.0.5>;tag=egHSerS4NrX7K.
>> To: <sip:10.0.0.5>;tag=2bf6dd58f2c27bac032ef66b671d14ff.52bb.
>> Call-ID: 3d12aa3e-88af-1234-b492-000d3a2233e0.
>> CSeq: 90631748 OPTIONS.
>> Server: OpenSIPS (2.1.2 (x86_64/linux)).
>> Content-Length: 0.
>> .
>>
>>
>> NDUWAYEZU Joselyne
>>
>> 2016-04-29 0:58 GMT+02:00 Michael Collins <msc at freeswitch.org>:
>>
>>> It sounds like the dialplan never makes it to your extension. Get a
>>> debug log of the call and put it on pastebin.freeswitch.org. We can
>>> probably help you figure it out from there.
>>>
>>> -MSC
>>>
>>> On Thu, Apr 28, 2016 at 12:40 AM, Nduwayezu, Joselyne <35633 at heb.be>
>>> wrote:
>>>
>>>>
>>>> Hello,
>>>>
>>>> I would like to invoke php script in the dial plan.
>>>> I defined the dialplan as follos:
>>>>
>>>>
>>>> <include>
>>>>  <extension name="outbound-socket">
>>>>   <condition field="${sip_to_user}" expression="(972550428)$">
>>>>    <action application="set"
>>>> data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
>>>>    <action application="socket" data="10.0.0.6:8021 async full"/>-->
>>>>   </condition>
>>>>  </extension>
>>>> </include>
>>>>
>>>> And my php script is:
>>>>
>>>>
>>>> <?php
>>>>   // set a couple of things so we dont kill the system
>>>>  ob_implicit_flush(true);
>>>>  set_time_limit(30);
>>>>  // Open stdin so we can read the data in
>>>>  $in = fopen("php://stdin", "r");
>>>>  // Connect
>>>>  echo "connect\n\n";
>>>>  // Answer
>>>>  echo "sendmsg\n";
>>>>  echo "call-command: execute\n";
>>>>  echo "execute-app-name: answer\n\n";
>>>>  // Play a prompt
>>>>  echo "sendmsg\n";
>>>>  echo "call-command: execute\n";
>>>>  echo "execute-app-name: playback\n";
>>>>  echo "execute-app-arg: \
>>>>  /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n
>>>> \
>>>>  ";
>>>>  // Wait
>>>>  sleep(5);
>>>>  // Hangup
>>>>  echo "sendmsg\n";
>>>>  echo "call-command: hangup\n\n";
>>>>  fclose($in);
>>>>  ?>
>>>>
>>>> When i dial the DID number, the call state is "call setup" for a while,
>>>> and after i'm asked to leave a message because the number is busy or
>>>> unreachable.
>>>>
>>>> I tried different sockets in the dialplan (5060, 5080, 8040) with no
>>>> results
>>>>
>>>> Any adeas of what is happening?
>>>>
>>>> Thanks
>>>>
>>>>
>>>> NDUWAYEZU Joselyne
>>>>
>>>> Haute ?cole de Bruxelles
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> Haute ?cole de Bruxelles
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Haute ?cole de Bruxelles
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