From shlomi.agiv at cell-buddy.com  Mon Dec  1 12:34:43 2014
From: shlomi.agiv at cell-buddy.com (Shlomi Agiv)
Date: Mon, 1 Dec 2014 11:34:43 +0200
Subject: [Freeswitch-users] USB GSM Dongle with FreeSwitch for voice
	calls
In-Reply-To: <CAMeZp62WTeFrSuh_OCKj+fE_soptn0TtCNL1Xrvf3=Jo2R9TuA@mail.gmail.com>
References: <CABUrQGuhh9Gt5xRc9b1OEdfENt=Lv2Nz1KSiH333BH6trnDujw@mail.gmail.com>
	<D34D36B7-9580-47F0-ABA0-42AF2465AEC4@gmail.com>
	<CAMeZp6218jcbHU6d7chaARx41j=ZhCM5aHZqxE6ANQQv5_c=ng@mail.gmail.com>
	<CAL1p8tqgwH=uUwkJ2eGS1E8ojOukwNNbPmH8HU0Zpw-TEQ_GHA@mail.gmail.com>
	<CAMeZp631xAjOu0RD+K1yNHzsjFPXWhagM4o0BC2CX6LWgStKgg@mail.gmail.com>
	<CANba_atdPxkbcVq=rpmV8sr5sUx6NMX8qTmnoyz4mAfzKxCwDQ@mail.gmail.com>
	<CAL1p8to8yoQSiiJFL89RfKnUYK2KH1Ht8ZcQpRyoPizDUBhn+g@mail.gmail.com>
	<CALXCt0pwfssS3suWsLZBw0MeNqNDZG+UFTRRSZw46a16hZS+cw@mail.gmail.com>
	<CAL1p8tqtHAyiOeFdv7Zxh-RG=5GkGtUpbaQVYXwjfgkER3ZWeA@mail.gmail.com>
	<CALXCt0pTaAsvFjC1w7NsWOfgHqRKrEeEq1uaEifHjoVKDRJ+wQ@mail.gmail.com>
	<CAL1p8tq1ijrK1qNJ8JEtreYzJBf5HVKPhmYB=SRFZ8iC4=qnCw@mail.gmail.com>
	<CALXCt0po23pW2S3vOy491m_wo7mVp63imkHCtLtfFVO=Niy61w@mail.gmail.com>
	<CAMeZp600KGSFN-DPfvq=-14vqPLzLZc4cB2SDbMwFB+hPQ40PQ@mail.gmail.com>
	<CALXCt0o14eYNVsci6VyyKdo0fGgRYX8c8Rt=ygojUZiCc0ytgw@mail.gmail.com>
	<CAMeZp62Q-EzeP1zGXez3oufxTVinKnGTSVDd=cZoZ=T+RHPnNg@mail.gmail.com>
	<CALXCt0oOb059VSWt-hdA12yTndFAc-2GnG76jAPifQ0cxdjgTA@mail.gmail.com>
	<CANba_asmVu1PYgd2YcKTJzD1JVY9qCauYHpsD_7c8PZ8jDRMtw@mail.gmail.com>
	<CAPHNFcoq3A4n=PDgWO155bb8ZZFS4aMW7JOXtcGROT6YhX+a5w@mail.gmail.com>
	<CANba_au17viA6xDRKSazwAJytU9dueuReUSwWYwMNbW2xCu4RA@mail.gmail.com>
	<CAL1p8tqNP37GCHvzAWh6ZpmKLD8wFABRCAWUj87=sLSMDnvb1Q@mail.gmail.com>
	<CAMeZp62WTeFrSuh_OCKj+fE_soptn0TtCNL1Xrvf3=Jo2R9TuA@mail.gmail.com>
Message-ID: <CANba_atbTQ9hgPzpPLBtoSAriiH22OnFacHsd9_rqix_KU1jdg@mail.gmail.com>

Hi,

I will be leaving my current job soon, therefore will leave the freeswitch
project.

I put all my changes in https://pastebin.freeswitch.org/23657. It's done
vs. version 1.4.4. The serial reconnect works pretty well. It also contains
a libctb change to immediately detect errors using select, that also works
well.
I put my other email in the CC since I don't know for how long I will keep
the current email, for any issues you can contact me there.

Thanks and best regards,
Shlomi Agiv


On Tue, Sep 30, 2014 at 1:48 PM, Daniel Ivanov <sertys at gmail.com> wrote:

> Yes, it's in the protected zone message  : pastebin / freeswitch
>
> On Tue, Sep 30, 2014 at 1:40 PM, rumodigital <
> webmaster.rumodigital at gmail.com> wrote:
>
>> hello Shlomi
>>
>> is protected with password?
>>
>> 2014-09-30 12:03 GMT+01:00 Shlomi Agiv <shlomi.agiv at cell-buddy.com>:
>>
>>> http://pastebin.freeswitch.org/23339
>>>
>>> The code is based on gsmopen version 1.4.7, if I recall correctly.
>>>
>>> On Mon, Sep 29, 2014 at 9:33 PM, Du?an Dragi? <dragic.dusan at gmail.com>
>>> wrote:
>>>
>>>> If you've been working with git follow instructions in
>>>> docs/SubmittingPatches (in fs repo) and
>>>> https://confluence.freeswitch.org/display/FREESWITCH/Pull+Requests.
>>>> Just follow the steps, push your branch to your stash repo and skip the
>>>> last part (creating the pull request). Afterwards anyone can take a look at
>>>> your changes (your stash repo is public).
>>>> Or if it's easier just use "git diff" to create a patch and put it
>>>> somewhere.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On 29 September 2014 14:45, Shlomi Agiv <shlomi.agiv at cell-buddy.com>
>>>> wrote:
>>>>
>>>>>
>>>>> Du?an,
>>>>> The code I made works pretty well, though I can't say it works
>>>>> perfectly. I just recently quashed a nasty bug there
>>>>> I can post it , no problems.
>>>>> Haven't posted any changes yet, what's the easiest way to do it?
>>>>>
>>>>> On Mon, Sep 29, 2014 at 3:16 PM, Giovanni Maruzzelli <
>>>>> gmaruzz at gmail.com> wrote:
>>>>>
>>>>>> When you get more info, open a new, specific thread.
>>>>>>
>>>>>> I'm sure many persons will find it of interest.
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>> Sent from Mobile
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> +39 347 266 56 18
>>>>>> On Sep 29, 2014 2:14 PM, "Daniel Ivanov" <sertys at gmail.com> wrote:
>>>>>>
>>>>>>> I would rather not hijack this thread with rpi specific issues. I
>>>>>>> will investigate further as from tomorrow, truth is panics started
>>>>>>> happening since i updated to the latest kern & picode.
>>>>>>>
>>>>>>> On Mon, Sep 29, 2014 at 2:02 PM, Giovanni Maruzzelli <
>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>
>>>>>>>> Kernel panics have probably to do with drivers, udev and kernel
>>>>>>>> versions...
>>>>>>>>
>>>>>>>> What distro, kernel, drivers, udev, etc uses when successful and
>>>>>>>> when panics?
>>>>>>>>
>>>>>>>> Please you all, list them here...
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Sent from Mobile
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli
>>>>>>>> +39 347 266 56 18
>>>>>>>> On Sep 29, 2014 2:00 PM, "Daniel Ivanov" <sertys at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> I also use a hub with external power supply, but that hardly cures
>>>>>>>>> the problem. On a normal x64 system, i seldom get usb errors, but the
>>>>>>>>> kernel doesn't panic. On the Pi i get the same bus or mem errors and it
>>>>>>>>> panics to need restart.
>>>>>>>>>
>>>>>>>>> On Mon, Sep 29, 2014 at 12:17 PM, Giovanni Maruzzelli <
>>>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> nice!!!
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Sincerely,
>>>>>>>>>>
>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>
>>>>>>>>>> On Mon, Sep 29, 2014 at 12:14 PM, rumodigital <
>>>>>>>>>> webmaster.rumodigital at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> I decided not to add more than five to avoid heat and maintain
>>>>>>>>>>> stability, so working with a margin of safety devices and distance to allow
>>>>>>>>>>> for ventilation.
>>>>>>>>>>>
>>>>>>>>>>> Not heats and works very well.
>>>>>>>>>>>
>>>>>>>>>>> 2014-09-29 11:01 GMT+01:00 Giovanni Maruzzelli <
>>>>>>>>>>> gmaruzz at gmail.com>:
>>>>>>>>>>>
>>>>>>>>>>>> seems a good one, with power enough to add another additional
>>>>>>>>>>>> dongle, for a total of 5 concurrent.
>>>>>>>>>>>>
>>>>>>>>>>>> Do it get too much warm?
>>>>>>>>>>>>
>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>
>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>
>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>
>>>>>>>>>>>> On Mon, Sep 29, 2014 at 11:57 AM, rumodigital <
>>>>>>>>>>>> webmaster.rumodigital at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Hi Giovanni,
>>>>>>>>>>>>>
>>>>>>>>>>>>> I'm in the testing phase in a Mini ITX board (Atom D2550 +
>>>>>>>>>>>>> debian 7.6 32b) and DUB-H7 USB HUB?. I use 4 devices simultaneously (4x
>>>>>>>>>>>>> 500ma = 2A) and yet everything perfect!
>>>>>>>>>>>>>
>>>>>>>>>>>>> 7-Port USB 2.0 Hub (500ma on all seven front ports, but 3A in
>>>>>>>>>>>>> total)
>>>>>>>>>>>>>
>>>>>>>>>>>>> http://www.dlink.com/uk/en/home-solutions/connect/usb/dub-h7-7-port-usb-2-0-hub
>>>>>>>>>>>>>
>>>>>>>>>>>>> Do you recommend some other HUB?
>>>>>>>>>>>>>
>>>>>>>>>>>>> Doug
>>>>>>>>>>>>>
>>>>>>>>>>>>> 2014-09-29 10:23 GMT+01:00 Giovanni Maruzzelli <
>>>>>>>>>>>>> gmaruzz at gmail.com>:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> all this kind of problems come from lack of USB bus
>>>>>>>>>>>>>> electrical power.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Use a wall powered (eg: with a "brick" power supply) USB hub
>>>>>>>>>>>>>> between FreeSWITCH and dongle.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> No software way to go around this.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> And be sure the hub deliver enough power! (check the specs
>>>>>>>>>>>>>> for your dongles power needs, add them up and use an USB hub with more than
>>>>>>>>>>>>>> that continous power delivery).
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Internet dongles are power hungry !
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> -giovanni
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Mon, Sep 29, 2014 at 11:12 AM, rumodigital <
>>>>>>>>>>>>>> webmaster.rumodigital at gmail.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Hi,
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Really, can be a major headache, many problems can be solved
>>>>>>>>>>>>>>> with the use of a self powered USB hub... yet I had no problems, but for
>>>>>>>>>>>>>>> example, I tested with an old pc I have, and when he
>>>>>>>>>>>>>>> received a call the module disconnected and blocked the USB port.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Which versions of Pi and what model of hardware that use?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Doug
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>>>>>>>>>>>> http://www.cudatel.com
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>>>>>>>>>>> http://www.cudatel.com
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>>
>>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>>>>
>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>>>>>>>>>> http://www.cudatel.com
>>>>>>>>>>>>>
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>
>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>>>>>>>>> http://www.cudatel.com
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>>>>>>>> http://www.cudatel.com
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>>> http://www.cluecon.com
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>>>>>>> http://www.cudatel.com
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://confluence.freeswitch.org
>>>>>>>>> http://www.cluecon.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>>>>>> http://www.cudatel.com
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://confluence.freeswitch.org
>>>>>>>> http://www.cluecon.com
>>>>>>>>
>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>>>>> http://www.cudatel.com
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://confluence.freeswitch.org
>>>>>>> http://www.cluecon.com
>>>>>>>
>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>>>> http://www.cudatel.com
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>>> http://www.cudatel.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>>> http://www.cudatel.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Du?an Dragi?
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>>> http://www.cudatel.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>>> http://www.cudatel.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
>> http://www.cudatel.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vipkilla at gmail.com  Mon Dec  1 16:04:26 2014
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 1 Dec 2014 08:04:26 -0500
Subject: [Freeswitch-users] multi tenet user directory management best
	practice
In-Reply-To: <CAOccsKDwjD4hWYAFxJuU_8D0ifVjni2jXsQdjBBTv2cEPdqGVw@mail.gmail.com>
References: <CAOccsKDwjD4hWYAFxJuU_8D0ifVjni2jXsQdjBBTv2cEPdqGVw@mail.gmail.com>
Message-ID: <CAC-LwPPfa8Ph-wuku==JF5OStXyKL6vCbcmgXeJAf3YRq73N6A@mail.gmail.com>

Use XML_CURL to push same configs dynamically to all FS servers.

On Sun, Nov 30, 2014 at 6:25 AM, ik <idokan at gmail.com> wrote:

> Hello,
>
> This question raises a lot, but no simple answer, so I'm trying to ask it
> differently.
>
> Let's say I have 5 FS servers and Opensips/Kamailio as a load balancer.
>
>
>
> I'm afraid that creating it using xml files is not dynamic enough, and
> requires duplications.
> mod_xml_rpc does not store that information on each update of the server
> itself, and there is no API for providing database management that I could
> find.
>
> So what is the best practice regarding of creating a user directory that
> all of the PBX will be able to handle ?
>
> Thanks
> Ido
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec  1 17:17:58 2014
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Dec 2014 08:17:58 -0600
Subject: [Freeswitch-users] echo cancellation and mod_conference
In-Reply-To: <21728.1417347781@ccs.covici.com>
References: <21728.1417347781@ccs.covici.com>
Message-ID: <CAEJMVkCxsSFVEbzirfDOsdK-V5OySFUGyq6pYQK6kyGYW7OZ8w@mail.gmail.com>

Acoustical Echo Cancel would be rather resource intensive, Why not instruct
your users to acquire proper endpoints? :)  Most Optimal... probably least
favorable.  I for one advocate for endpoints that handle their own nat
traversal, echo cancel leaving the registrar/freeswitch no avoid that sort
of heavy lifting.

On Sun, Nov 30, 2014 at 5:43 AM, <covici at ccs.covici.com> wrote:

> Hi.  I have some sip callers who call in on what are apparently bad
> lines and get echo when calling into conferences -- not freeswitch ones
> yet.  So, I am wondering in mod_conference  is there any attempt to do
> echo cancellation, if so I will try to have them call into an fs one
> instead.  Its intermittent, so I wanted to find out first before I test.
>
> Thanks in advance for any suggestions.
>
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici
>          covici at ccs.covici.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From brian at freeswitch.org  Mon Dec  1 17:19:30 2014
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Dec 2014 08:19:30 -0600
Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive calls
In-Reply-To: <069E914C-296C-495C-8A49-54E1A74A866E@gmail.com>
References: <069E914C-296C-495C-8A49-54E1A74A866E@gmail.com>
Message-ID: <CAEJMVkD=zJqG_0rsZGr1QtT7wAkJy2_2fiQwhg-9KhD0ifyb1g@mail.gmail.com>

What endpoints are involved?  have you looked at 'sofia loglevel all 9'
output and see if it gives you a clue?

On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba <ahabiba at gmail.com> wrote:

> Dears,
>
> I?ve configured FreeSWITCH with the below version with TLS/SRTPas per the
> recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS? and it
> was strait forward, and I was able to connect and make make calls using
> zoiper, but I was not able to receive any calls after enabling the TLS/SRTP.
>
> *"FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git
> b942d0f 2014-11-03 19:53:00Z 64bit)**?*
>
> Your kind feedback will be appreciate.
>
> 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
>



-- 

*Brian West*
brian at freeswitch.org


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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From brian at freeswitch.org  Mon Dec  1 17:25:53 2014
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Dec 2014 08:25:53 -0600
Subject: [Freeswitch-users] mod_verto behind NAT
In-Reply-To: <CAEKk1MBEn-0+uCTi21EVR=-aSU_UVEC54PtAdzV09g6AXV0y5A@mail.gmail.com>
References: <CAEKk1MBEn-0+uCTi21EVR=-aSU_UVEC54PtAdzV09g6AXV0y5A@mail.gmail.com>
Message-ID: <CAEJMVkCPndPs+Yu223vBMZe5EhzXbRJ9f6=7Knu4YWaiLQtSwA@mail.gmail.com>

Please contact consulting at freeswitch.org for a quote to build this into
mod_verto, It would appear mod_verto has no concept of NAT like sofia does.

On Wed, Nov 26, 2014 at 10:13 AM, Ciprian Dosoftei <
ciprian.dosoftei at gmail.com> wrote:

> Hi,
>
> I am dealing with a NAT situation related to mod_verto. The switch is
> hosted in EC2 and the configuration works great with mod_sofia (both for
> receiving calls from our SIP providers and via webRTC clients).
>
> However, the counterpart configuration for mod_verto fails to expose the
> external IP address in the SDP. Here's the mod_verto configuration we're
> using:
>
> http://pastebin.com/4PDMXvW6
>
> I can confirm the external RTP IP is set properly, however here's the
> returned SDP:
>
> http://pastebin.com/3vzyihRp
>
> The verto client is running behind NAT as well, but this doesn't seem to
> be the root of the problem as Sofia's own implementation of webRTC works
> fine. It is effectively sending its media to 10.178.x.x rather than the
> switch's public IP address.
>
> Please advise, thank you.
>
> --
> Best Regards,
> Ciprian Dosoftei
>
> The information transmitted is intended only for the addressee and may
> contain privileged and/or confidential material. If you are not the
> intended recipient, kindly contact the sender and delete the message.
>
> Any disclosure, distribution or copying of this message is strictly
> prohibited without the expressed permission of the sender.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From covici at ccs.covici.com  Mon Dec  1 18:15:08 2014
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 01 Dec 2014 10:15:08 -0500
Subject: [Freeswitch-users] echo cancellation and mod_conference
In-Reply-To: <CAEJMVkCxsSFVEbzirfDOsdK-V5OySFUGyq6pYQK6kyGYW7OZ8w@mail.gmail.com>
References: <21728.1417347781@ccs.covici.com>
	<CAEJMVkCxsSFVEbzirfDOsdK-V5OySFUGyq6pYQK6kyGYW7OZ8w@mail.gmail.com>
Message-ID: <19723.1417446908@ccs.covici.com>

Its not the endpoint itself, but something in the pstn -- if they call
in say a  number of times, the echo will stop, it seems to be something
in the Verizon equipment, this is why I was hoping for fs to do
something.

Thanks.

Brian West <brian at freeswitch.org> wrote:

> Acoustical Echo Cancel would be rather resource intensive, Why not instruct
> your users to acquire proper endpoints? :)  Most Optimal... probably least
> favorable.  I for one advocate for endpoints that handle their own nat
> traversal, echo cancel leaving the registrar/freeswitch no avoid that sort
> of heavy lifting.
> 
> On Sun, Nov 30, 2014 at 5:43 AM, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I have some sip callers who call in on what are apparently bad
> > lines and get echo when calling into conferences -- not freeswitch ones
> > yet.  So, I am wondering in mod_conference  is there any attempt to do
> > echo cancellation, if so I will try to have them call into an fs one
> > instead.  Its intermittent, so I wanted to find out first before I test.
> >
> > Thanks in advance for any suggestions.
> >
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >          John Covici
> >          covici at ccs.covici.com
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> 
> *Brian West*
> brian at freeswitch.org
> 
> 
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> 
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From brian at freeswitch.org  Mon Dec  1 18:45:54 2014
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Dec 2014 09:45:54 -0600
Subject: [Freeswitch-users] echo cancellation and mod_conference
In-Reply-To: <19723.1417446908@ccs.covici.com>
References: <21728.1417347781@ccs.covici.com>
	<CAEJMVkCxsSFVEbzirfDOsdK-V5OySFUGyq6pYQK6kyGYW7OZ8w@mail.gmail.com>
	<19723.1417446908@ccs.covici.com>
Message-ID: <CAEJMVkDRSZYRv2GFkL--Lod75e0N8oRnhV=iKdD-w6r8-_jGqw@mail.gmail.com>

Sadly there probably doesn't exist a silver bullet to solve this problem...
it will always be an uphill battle without a resolution... the real
solution is to open tickets with upstream providers till its resolved, but
I'm sure you could retire before they'd actually find/fix it unless
complain daily for weeks on end.

On Mon, Dec 1, 2014 at 9:15 AM, <covici at ccs.covici.com> wrote:

> Its not the endpoint itself, but something in the pstn -- if they call
> in say a  number of times, the echo will stop, it seems to be something
> in the Verizon equipment, this is why I was hoping for fs to do
> something.
>
> Thanks.
>
> Brian West <brian at freeswitch.org> wrote:
>
> > Acoustical Echo Cancel would be rather resource intensive, Why not
> instruct
> > your users to acquire proper endpoints? :)  Most Optimal... probably
> least
> > favorable.  I for one advocate for endpoints that handle their own nat
> > traversal, echo cancel leaving the registrar/freeswitch no avoid that
> sort
> > of heavy lifting.
> >
> > On Sun, Nov 30, 2014 at 5:43 AM, <covici at ccs.covici.com> wrote:
> >
> > > Hi.  I have some sip callers who call in on what are apparently bad
> > > lines and get echo when calling into conferences -- not freeswitch ones
> > > yet.  So, I am wondering in mod_conference  is there any attempt to do
> > > echo cancellation, if so I will try to have them call into an fs one
> > > instead.  Its intermittent, so I wanted to find out first before I
> test.
> > >
> > > Thanks in advance for any suggestions.
> > >
> > >
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > >
> > >          John Covici
> > >          covici at ccs.covici.com
> > >
> > >
> _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org
> > > http://www.freeswitchsolutions.com
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://confluence.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> >
> >
> >
> > --
> >
> > *Brian West*
> > brian at freeswitch.org
> >
> >
> > *Twitter: @FreeSWITCH , @briankwest*
> > http://www.freeswitchbook.com
> > http://www.freeswitchcookbook.com
> >
> > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici
>          covici at ccs.covici.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

*Brian West*
brian at freeswitch.org


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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From netcentrica at gmail.com  Mon Dec  1 18:46:39 2014
From: netcentrica at gmail.com (Mateusz Bartczak)
Date: Mon, 1 Dec 2014 16:46:39 +0100
Subject: [Freeswitch-users] Donate creation of mod_gearman module
Message-ID: <CAJWEF2baBimM-HpvF2btkvqqhorUQ3Ve6rQJqRdoRCUx4JfJSQ@mail.gmail.com>

Hi


I?m looking for experienced FreeSWITCH developer to build new module which
will allow to send FS events using Gearman library. Something like existing
mod_zmq module, but for other protocol.


Additionally module should support of RPC calls from Gearman clients and
invoke them as API (blocking) and BGAPI (non-blocking) commands inside FS.


I have detailed specification of needed functionalities. Looking for
someone ready to start promptly and make it fast.


If someone from core FS dev team will respond, then I can declare to push
back this module as open source available for all of the community


Anybody interested please PM me: netcentrica at gmail.com
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From ssinyagin at gmail.com  Mon Dec  1 18:55:03 2014
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Mon, 1 Dec 2014 16:55:03 +0100
Subject: [Freeswitch-users] Donate creation of mod_gearman module
In-Reply-To: <CAJWEF2baBimM-HpvF2btkvqqhorUQ3Ve6rQJqRdoRCUx4JfJSQ@mail.gmail.com>
References: <CAJWEF2baBimM-HpvF2btkvqqhorUQ3Ve6rQJqRdoRCUx4JfJSQ@mail.gmail.com>
Message-ID: <CACOYK=jd4H86ON7J33nyNia6Ba5mdOEEKrZNqz6JrygfHwm_oA@mail.gmail.com>

why does it have to be a FreeSWITCH module? Why not making a separate
daemon which talks to FreeSWITCH via ESL and communicates with
Gearman?

The benefit would be that you don't need anything new inside
FreeSWITCH, so your development woudl be completely independent, and
always compatible with future releases of FreeSWITCH.




On Mon, Dec 1, 2014 at 4:46 PM, Mateusz Bartczak <netcentrica at gmail.com> wrote:
> Hi
>
>
> I?m looking for experienced FreeSWITCH developer to build new module which
> will allow to send FS events using Gearman library. Something like existing
> mod_zmq module, but for other protocol.
>
>
> Additionally module should support of RPC calls from Gearman clients and
> invoke them as API (blocking) and BGAPI (non-blocking) commands inside FS.
>
>
> I have detailed specification of needed functionalities. Looking for someone
> ready to start promptly and make it fast.
>
>
> If someone from core FS dev team will respond, then I can declare to push
> back this module as open source available for all of the community
>
>
> Anybody interested please PM me: netcentrica 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


From brian at freeswitch.org  Mon Dec  1 19:03:53 2014
From: brian at freeswitch.org (Brian West)
Date: Mon, 1 Dec 2014 10:03:53 -0600
Subject: [Freeswitch-users] Donate creation of mod_gearman module
In-Reply-To: <CACOYK=jd4H86ON7J33nyNia6Ba5mdOEEKrZNqz6JrygfHwm_oA@mail.gmail.com>
References: <CAJWEF2baBimM-HpvF2btkvqqhorUQ3Ve6rQJqRdoRCUx4JfJSQ@mail.gmail.com>
	<CACOYK=jd4H86ON7J33nyNia6Ba5mdOEEKrZNqz6JrygfHwm_oA@mail.gmail.com>
Message-ID: <CAEJMVkAphhMmFbrYQtiGh-goe=FiDgKq_ecsJFzsXZOX7b63iw@mail.gmail.com>

well either way can work, depends on the functionality desired, can you
outline or write up what exactly the use cases are?

On Mon, Dec 1, 2014 at 9:55 AM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> why does it have to be a FreeSWITCH module? Why not making a separate
> daemon which talks to FreeSWITCH via ESL and communicates with
> Gearman?
>
> The benefit would be that you don't need anything new inside
> FreeSWITCH, so your development woudl be completely independent, and
> always compatible with future releases of FreeSWITCH.
>
>
>
>
> On Mon, Dec 1, 2014 at 4:46 PM, Mateusz Bartczak <netcentrica at gmail.com>
> wrote:
> > Hi
> >
> >
> > I?m looking for experienced FreeSWITCH developer to build new module
> which
> > will allow to send FS events using Gearman library. Something like
> existing
> > mod_zmq module, but for other protocol.
> >
> >
> > Additionally module should support of RPC calls from Gearman clients and
> > invoke them as API (blocking) and BGAPI (non-blocking) commands inside
> FS.
> >
> >
> > I have detailed specification of needed functionalities. Looking for
> someone
> > ready to start promptly and make it fast.
> >
> >
> > If someone from core FS dev team will respond, then I can declare to push
> > back this module as open source available for all of the community
> >
> >
> > Anybody interested please PM me: netcentrica 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




-- 

*Brian West*
brian at freeswitch.org


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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From netcentrica at gmail.com  Mon Dec  1 19:10:21 2014
From: netcentrica at gmail.com (Mateusz Bartczak)
Date: Mon, 1 Dec 2014 17:10:21 +0100
Subject: [Freeswitch-users] Donate creation of mod_gearman module
In-Reply-To: <CACOYK=jd4H86ON7J33nyNia6Ba5mdOEEKrZNqz6JrygfHwm_oA@mail.gmail.com>
References: <CAJWEF2baBimM-HpvF2btkvqqhorUQ3Ve6rQJqRdoRCUx4JfJSQ@mail.gmail.com>
	<CACOYK=jd4H86ON7J33nyNia6Ba5mdOEEKrZNqz6JrygfHwm_oA@mail.gmail.com>
Message-ID: <CAJWEF2abH=9fRRrQmhvsBsrDMnZXE2NG60Nx1r+qPOM_mfJ52Q@mail.gmail.com>

The main reason is performance. Some other reasons are specific to my
company internal requirements and the architecture of project we are
working on. I can't discuss on this public forum.
As stated in the first message I'll be happy to pay for work needed to
create this module

2014-12-01 16:55 GMT+01:00 Stanislav Sinyagin <ssinyagin at gmail.com>:

> why does it have to be a FreeSWITCH module? Why not making a separate
> daemon which talks to FreeSWITCH via ESL and communicates with
> Gearman?
>
> The benefit would be that you don't need anything new inside
> FreeSWITCH, so your development woudl be completely independent, and
> always compatible with future releases of FreeSWITCH.
>
>
>
>
> On Mon, Dec 1, 2014 at 4:46 PM, Mateusz Bartczak <netcentrica at gmail.com>
> wrote:
> > Hi
> >
> >
> > I?m looking for experienced FreeSWITCH developer to build new module
> which
> > will allow to send FS events using Gearman library. Something like
> existing
> > mod_zmq module, but for other protocol.
> >
> >
> > Additionally module should support of RPC calls from Gearman clients and
> > invoke them as API (blocking) and BGAPI (non-blocking) commands inside
> FS.
> >
> >
> > I have detailed specification of needed functionalities. Looking for
> someone
> > ready to start promptly and make it fast.
> >
> >
> > If someone from core FS dev team will respond, then I can declare to push
> > back this module as open source available for all of the community
> >
> >
> > Anybody interested please PM me: netcentrica 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 mike at jerris.com  Mon Dec  1 19:54:11 2014
From: mike at jerris.com (Michael Jerris)
Date: Mon, 1 Dec 2014 11:54:11 -0500
Subject: [Freeswitch-users] Donate creation of mod_gearman module
In-Reply-To: <CAJWEF2abH=9fRRrQmhvsBsrDMnZXE2NG60Nx1r+qPOM_mfJ52Q@mail.gmail.com>
References: <CAJWEF2baBimM-HpvF2btkvqqhorUQ3Ve6rQJqRdoRCUx4JfJSQ@mail.gmail.com>
	<CACOYK=jd4H86ON7J33nyNia6Ba5mdOEEKrZNqz6JrygfHwm_oA@mail.gmail.com>
	<CAJWEF2abH=9fRRrQmhvsBsrDMnZXE2NG60Nx1r+qPOM_mfJ52Q@mail.gmail.com>
Message-ID: <155F1183-78B4-4A3F-A8FA-F880E8E2CBE7@jerris.com>

we can help you at consulting at freeswitch.org.


> On Dec 1, 2014, at 11:10 AM, Mateusz Bartczak <netcentrica at gmail.com> wrote:
> 
> The main reason is performance. Some other reasons are specific to my company internal requirements and the architecture of project we are working on. I can't discuss on this public forum.
> As stated in the first message I'll be happy to pay for work needed to create this module
> 
> 2014-12-01 16:55 GMT+01:00 Stanislav Sinyagin <ssinyagin at gmail.com <mailto:ssinyagin at gmail.com>>:
> why does it have to be a FreeSWITCH module? Why not making a separate
> daemon which talks to FreeSWITCH via ESL and communicates with
> Gearman?
> 
> The benefit would be that you don't need anything new inside
> FreeSWITCH, so your development woudl be completely independent, and
> always compatible with future releases of FreeSWITCH.
> 
> 
> 
> 
> On Mon, Dec 1, 2014 at 4:46 PM, Mateusz Bartczak <netcentrica at gmail.com <mailto:netcentrica at gmail.com>> wrote:
> > Hi
> >
> >
> > I?m looking for experienced FreeSWITCH developer to build new module which
> > will allow to send FS events using Gearman library. Something like existing
> > mod_zmq module, but for other protocol.
> >
> >
> > Additionally module should support of RPC calls from Gearman clients and
> > invoke them as API (blocking) and BGAPI (non-blocking) commands inside FS.
> >
> >
> > I have detailed specification of needed functionalities. Looking for someone
> > ready to start promptly and make it fast.
> >
> >
> > If someone from core FS dev team will respond, then I can declare to push
> > back this module as open source available for all of the community
> >
> >
> > Anybody interested please PM me: netcentrica at gmail.com <mailto:netcentrica at gmail.com>
> >
> >
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From krice at freeswitch.org  Mon Dec  1 21:53:40 2014
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 01 Dec 2014 18:53:40 +0000
Subject: [Freeswitch-users] =?utf-8?q?Verto=E2=84=A2_Not_just_for_call_sig?=
	=?utf-8?q?naling?=
Message-ID: <547cb9344e5a0_d385faf3342313e@ip-10-156-165-183.mail>

New Post on freeswitch.org from anthm
check it out at http://ift.tt/1vL90Zr
Verto? Not just for call signaling

Verto is not only focused on call signaling; There are a few other key components that are important to understand.

One key component of Verto is event channels. Event channels are bi-directional subscription-based logical data channels used to relay information in an organized fashion. This is a low level building block for many features. For instance, call or chat presence. Event channels can be subscribed to from JS and granular permissions may be set from FS on-the-fly or ahead of time. Modules inside FreeSWITCH fire json events and every connection who has a subscription will get a copy of the data allowing action to be taken. At the same time each browser connection is able to send a request back to the server on a particular event channel which can trigger action varying from sending more events, a full sync of saved data or even pushing the events into another event bus to send to another external application.

Our demo site http://ift.tt/12g7Iff and call 3500 you can see an example of the data synchronization functionality as it will generate events that sync and update in real time a data structure in FreeSWITCH and one in the page of anyone subscribed. Because 3500 is a conference, the conference module will grant you access to the data about who else is called into the conference. It sends an event scoped to the exact uuid of the connection of the calling verto client. This event says that there is an available data synchronization for the particular conference. Using the data from this event, you can initialize an object that uses event channels to subscribe to the conference data and request a full sync and then keeps the object in sync with update events as long as you are on the call. This allows a table to be drawn in real time reflecting the changes going on in the conference.

When you hangup, it sends a similar event telling you that you should unregister from this service and removes your permission. If you wanted, you could allow permissions to one or all conference data from your FS config and use verto to connect and provide info about who?s in the conference even when you are not connected directly.

Another important feature of the module is the ability to communicate with FreeSWITCH and invent your own api modules. The lower level json-rpc in FS contains 2 important commands, fsapi and jsapi. fsapi allows you to execute any FreeSWITCH FSAPI command (basically all the commands you can type at the cli) This is set on a per command permission basis and is of course best used with security in mind. The other command, jsapi, is used to interface with a new type of module interface in FreeSWITCH called the JSON API Interface (JSONAPI) This interface is similar to FSAPI only its designed with a JSON in JSON out structure. One could write a C module defining a series of JSONAPI designed to allow your html5 application to communicate directly with your module in the same fashion that mod_verto does. This allows mod_verto to host the server logic and just pass commands up to your module and communicate in basic json exchanges.

There some examples in FreeSWITCH now that you can test from your cli There is a FSAPI command called json that just assumes the input is a chunk of json and passes it right into JSONAPI then gets the results and converts it back to text so you can see it on the screen.

Paste this command into your terminal:

json {?command?: ?status?, ?format?: ?pretty?, ?data?: {}}

Had it been over jsonrpc it would have been sent with method of jsapi and the json below would have been the params element:

{
?jsonrpc?: ?2.0?,
?method?: ?jsapi?,
?params?:{?command?: ?status?, ?format?: ?pretty?, ?data?: {}}

?id?: 2
}

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From k.presler at megafit.su  Mon Dec  1 22:51:41 2014
From: k.presler at megafit.su (=?koi8-r?B?68nSyczMIPDSxdPMxdI=?=)
Date: Mon, 1 Dec 2014 19:51:41 +0000
Subject: [Freeswitch-users] Freeswitch Python scripting
Message-ID: <20905bdd9aa94c639fbcf2e901f0290c@AM3PR06MB433.eurprd06.prod.outlook.com>

We want to use dialplan python script.
Script working but call setup is lasting too long ( <10sec to >40sec when 60 concurent calls on switch)
Script just route call from user to gateway.
I've not a lot of experience in FS and programming, so I need your help.
Ps
I'm not good in English, excuse me.
Code part

                        sessionData = 'sofia/gateway/gwname/dest'
                       new_session = Session(sessionData,session)

                        if new_session.ready():
                                 bridge(session,new_session)
                                 if new_session.ready():
                                      new_session.hangup()
                        hcause = new_session.hangupCause()

My native language is

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From ahabiba at gmail.com  Mon Dec  1 23:56:24 2014
From: ahabiba at gmail.com (Ahmed habiba)
Date: Mon, 1 Dec 2014 23:56:24 +0300
Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive calls
Message-ID: <CAGDxTYThLaBEyrO-3R9tYdSVXNPLF+M0yDL3DJb379=tQmuRDQ@mail.gmail.com>

I?m using "zoiper 1.19.7? on android, the issue appears only one I enable
SRTP, for TLS only it is working very fine.

The issue as shown below it keeps registering till finally zipper restart
on mobile.

One more point I?m configuring Zoiper without using RPORT for either SRTP
or TLS.

*?2014-12-01 20:57:24.003230 [NOTICE] switch_ivr_originate.c:3467 Hangup
sofia/internal/sip:1000 at 222.248.102.244 <1000 at 222.248.102.244>:38614
[CS_CONSUME_MEDIA] [NO_ANSWER]"*



recv 1290 bytes from tls/[222.215.195.234]:2896 at 20:56:54.344450:
   ------------------------------------------------------------------------
   INVITE sip:1000 at abc-xyz.com SIP/2.0
   v: SIP/2.0/TLS 192.168.8.109:41659
;rport;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias
   Max-Forwards: 70
   f: <sip:1006 at abc-xyz.com>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU
   t: <sip:1000 at abc-xyz.com>
   m: <sip:1006 at 222.215.195.234:2896;transport=TLS;ob>
   i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX
   CSeq: 32737 INVITE
   Route: <sip:abc-xyz.com;transport=tls;lr>
   k: replaces, 100rel, timer, norefersub
   x: 1800
   Min-SE: 90
   User-Agent: CSipSimple_GT-S5830-10/r2450
   Proxy-Authorization: Digest username="1006", realm="abc-xyz.com",
nonce="329456ee-7994-11e4-bf79-1182ebca2429", uri="sip:1000 at abc-xyz.com",
response="ab3a0f2fe9207ced62eebd8c9b8c32b4", algorithm=MD5,
cnonce="iLa-C.wpnBSeTQjWiXZK3H3PGYhi6EGt", qop=auth, nc=00000001
   c: application/sdp
   l:   469

   v=0
   o=- 3626452487 3626452487 IN IP4 192.168.8.109
   s=pjmedia
   c=IN IP4 192.168.8.109
   t=0 0
   m=audio 4000 RTP/SAVP 98 0 8 101
   c=IN IP4 192.168.8.109
   a=rtcp:4001 IN IP4 192.168.8.109
   a=sendrecv
   a=rtpmap:98 SILK/16000
   a=fmtp:98 useinbandfec=0
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:Vci7cNdSFTkkLnqz+qRkqCctPvRT6jIOrNc5KMbz
   a=crypto:2 AES_CM_128_HMAC_SHA1_32
inline:nW3ECq5YHHnrY2q5u7XZE4fvPfMOAWKQ+ehOp7zV
   ------------------------------------------------------------------------

   send 405 bytes to tls/[222.215.195.234]:2896 at 20:56:54.344687:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/TLS 192.168.8.109:41659
;rport=2896;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias;received=222.215.195.234
   f: <sip:1006 at abc-xyz.com>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU
   t: <sip:1000 at abc-xyz.com>
   i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX
   CSeq: 32737 INVITE
   User-Agent:
FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Content-Length: 0






send 2133 bytes to tls/[222.248.102.244]:38614 at 20:56:54.360608:
   ------------------------------------------------------------------------
   INVITE
sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS
SIP/2.0
   Via: SIP/2.0/TLS 176.31.245.17;branch=z9hG4bKjXZvg3cX76Krj
   Max-Forwards: 69
   From: "Extension 1006" <sip:1006 at 176.31.245.17>;tag=15DcKN9tUZeaK
   To: <
sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS>
   Call-ID: 0a3e07c5-f437-1232-9bba-d2ab2784dc6a
   CSeq: 68395843 INVITE
   Contact: <sip:mod_sofia at 176.31.245.17:5061;transport=tls>
   User-Agent:
FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   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: 1076
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "Extension 1006" <sip:1006 at 176.31.245.17
>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1417445672 1417445673 IN IP4 176.31.245.17
   s=FreeSWITCH
   c=IN IP4 176.31.245.17
   t=0 0
   m=audio 18142 RTP/SAVP 0 8 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=crypto:1 AEAD_AES_256_GCM_8
inline:6p/uXB7tN5SG9vHP/DMurij6BaZIOKcP4Tle0kjo2lwL1Oa4z9WGyZtk4zs
   a=crypto:2 AEAD_AES_128_GCM_8
inline:hGeJA6XQV1V35oTu2141o7mD64dFDRbeCbcf6g
   a=crypto:3 AES_CM_256_HMAC_SHA1_80
inline:XdZwwcl6kMRrWY7Vkrbozf/o1awka9lef0UjQRwvnOnnrbAgzXIm8s3yu7o8DQ
   a=crypto:4 AES_CM_192_HMAC_SHA1_80
inline:KtM92m0ac2musYPoXLDJb4rUbEMYAwMegPH9RMtzwzPb+0O93e0
   a=crypto:5 AES_CM_128_HMAC_SHA1_80
inline:PtOpdIk2LTSCSDVzPaSN6AV2xosBrJUTd4mvKIGh
   a=crypto:6 AES_CM_256_HMAC_SHA1_32
inline:ZHzWxajRlQp8IuzX5CuFM1SRkL0huC62ukX583+Vg0LBiK9t2NrpP8FNwdkckQ
   a=crypto:7 AES_CM_192_HMAC_SHA1_32
inline:S9BWrDxGFYpU3P9o3HD0Z97W5jwo8vD+E1sJi821r4/B+cdukTw
   a=crypto:8 AES_CM_128_HMAC_SHA1_32
inline:l7C0wU29AjF2ZMnCTvks+1ytd/UiXHqN8UIsM6vf
   a=crypto:9 AES_CM_128_NULL_AUTH
inline:tv97QsO7+S9NLTXrD02omyWoyM7I2KaWaimUgq99
   a=ptime:20
   ------------------------------------------------------------------------


      REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
   Via: SIP/2.0/TLS 192.168.10.2:41120
;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <
sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS>
   To: <sip:1000 at abc-xyz.com;transport=TLS>
   From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 1 REGISTER
   Expires: 60
   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
   Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri
   User-Agent: Zoiper r27147
   Allow-Events: presence, kpml
   Content-Length: 0



------------------------------------------------------------------------
send 716 bytes to tls/[222.248.102.244]:38620 at 20:57:01.806433:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/TLS 192.168.10.2:41120
;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport=38620;received=222.248.102.244
   From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
   To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=2e74mgtyr84ve
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 1 REGISTER
   User-Agent:
FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   WWW-Authenticate: Digest realm="abc-xyz.com",
nonce="374fdd98-7994-11e4-bf9c-1182ebca2429", algorithm=MD5, qop="auth"
   Content-Length: 0



------------------------------------------------------------------------
recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:01.995662:
   ------------------------------------------------------------------------
   REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
   Via: SIP/2.0/TLS 192.168.10.2:41120
;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <
sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS>
   To: <sip:1000 at abc-xyz.com;transport=TLS>
   From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 2 REGISTER
   Expires: 60
   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
   Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri
   User-Agent: Zoiper r27147
   Authorization: Digest username="1000",realm="abc-xyz.com
",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri="
sip:abc-xyz.com;transport=TLS
",response="8d7e108ff98035d37424c32166fe0253",cnonce="107cf281c3b57888a4bfd1a9c3776098",nc=00000001,qop=auth,algorithm=MD5
   Allow-Events: presence, kpml
   Content-Length: 0



------------------------------------------------------------------------
send 712 bytes to tls/[222.248.102.244]:38620 at 20:57:01.997931:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS 192.168.10.2:41120
;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport=38620;received=222.248.102.244
   From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
   To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=3Q0XpBB2NHUFa
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 2 REGISTER
   Contact: <
sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS
>;expires=60
   Date: Mon, 01 Dec 2014 19:57:01 GMT
   User-Agent:
FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:02.205719:
   ------------------------------------------------------------------------
   REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
   Via: SIP/2.0/TLS 222.248.102.244:38620
;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <
sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS
>;expires=0
   To: <sip:1000 at abc-xyz.com;transport=TLS>
   From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 3 REGISTER
   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
   Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri
   User-Agent: Zoiper r27147
   Authorization: Digest username="1000",realm="abc-xyz.com
",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri="
sip:abc-xyz.com;transport=TLS
",response="5168bf9afffe9adacef7d898bd6bec9b",cnonce="80108fa519303990cfd2d46639d767ee",nc=00000002,qop=auth,algorithm=MD5
   Allow-Events: presence, kpml
   Content-Length: 0


------------------------------------------------------------------------
send 598 bytes to tls/[222.248.102.244]:38620 at 20:57:02.207249:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS 222.248.102.244:38620
;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport=38620
   From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
   To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=40Spr6U5jtH2N
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 3 REGISTER
   Date: Mon, 01 Dec 2014 19:57:02 GMT
   User-Agent:
FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Content-Length: 0

   ------------------------------------------------------------------------
recv 681 bytes from tls/[222.248.102.244]:38620 at 20:57:02.400564:
   ------------------------------------------------------------------------
   REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
   Via: SIP/2.0/TLS 222.248.102.244:38620
;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <
sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS>
   To: <sip:1000 at abc-xyz.com;transport=TLS>
   From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
   Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
   CSeq: 1 REGISTER
   Expires: 60
   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
   Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri
   User-Agent: Zoiper r27147
   Allow-Events: presence, kpml
   Content-Length: 0



------------------------------------------------------------------------
send 694 bytes to tls/[222.248.102.244]:38620 at 20:57:02.401167:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/TLS 222.248.102.244:38620
;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport=38620
   From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
   To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=59jFt1c9F37mH
   Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
   CSeq: 1 REGISTER
   User-Agent:
FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   WWW-Authenticate: Digest realm="abc-xyz.com",
nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429", algorithm=MD5, qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 950 bytes from tls/[222.248.102.244]:38620 at 20:57:02.655864:
   ------------------------------------------------------------------------
   REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
   Via: SIP/2.0/TLS 222.248.102.244:38620
;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <
sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS>
   To: <sip:1000 at abc-xyz.com;transport=TLS>
   From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
   Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
   CSeq: 2 REGISTER
   Expires: 60
   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
   Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri
   User-Agent: Zoiper r27147
   Authorization: Digest username="1000",realm="abc-xyz.com
",nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429",uri="
sip:abc-xyz.com;transport=TLS
",response="f1667094626fc10ca76a070438497e6f",cnonce="dc790bffe285311f0b903bd515a3e741",nc=00000001,qop=auth,algorithm=MD5
   Allow-Events: presence, kpml
   Content-Length: 0

   ------------------------------------------------------------------------
send 693 bytes to tls/[222.248.102.244]:38620 at 20:57:02.657688:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS 222.248.102.244:38620
;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport=38620
   From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
   To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=6jc8UvXcDcy7c
   Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
   CSeq: 2 REGISTER
   Contact: <
sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS
>;expires=60
   Date: Mon, 01 Dec 2014 19:57:02 GMT
   User-Agent:
FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Content-Length: 0






*From: *Brian West <brian at freeswitch.org>
*To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
*Date: *December 1, 2014 at 5:19:30 PM GMT+3
*Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
*Subject: **Re: [Freeswitch-users] FreeSWITCH TLS not able to receive calls*


What endpoints are involved?  have you looked at 'sofia loglevel all 9'
output and see if it gives you a clue?

On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba <ahabiba at gmail.com> wrote:

> Dears,
>
> I?ve configured FreeSWITCH with the below version with TLS/SRTPas per the
> recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS? and it
> was strait forward, and I was able to connect and make make calls using
> zoiper, but I was not able to receive any calls after enabling the TLS/SRTP.
>
> *"FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git
> b942d0f 2014-11-03 19:53:00Z 64bit)**?*
>
> Your kind feedback will be appreciate.
>
> 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
>



-- 

*Brian West*
brian at freeswitch.org


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

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






-- 
Thanks and Best Regards,

Ahmed Habiba

Mob: +20 10 37 82 970

*Success: believe (Vision) plus commitment (Action)*
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From AWalters at cawst.org  Tue Dec  2 02:50:32 2014
From: AWalters at cawst.org (Allison Walters)
Date: Mon, 1 Dec 2014 23:50:32 +0000
Subject: [Freeswitch-users] Questions before starting new install
Message-ID: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com>

We're wanting to replace our current freeswitch server (it's fairly old), and I have a few things I'm curious about before I start trying to build the new one.  I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience.


1.       I was told by the consultant that set up the current box that freeswitch doesn't play well with fusionpbx or really any of the GUI's - there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion.  Also I was told it doesn't provide nice clean XML, etc...  I was strongly encouraged to go with no GUI.  If it were just me, I'd be okay with that, but we have a couple of other people who might be helping to maintain the server and they're far more comfortable with GUIs.  Has anyone else had any of these issues, and/or have any recommendations on the subject?

2.       I'm trying to install on Ubuntu.  When I go to the wiki, I get this page:  https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu.  However, that line just seems to have pre-reqs and not freeswitch itself.  Anyone know where I could find current Ubuntu instructions?  I googled but the most recent pages I could find were from 2012, and most were from 2011.  Is that the most up to date?

3.       Are there any guidelines for migration from an old server running an old version to a new server with a current version?

Thank you!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.

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From max at nysolutions.com  Tue Dec  2 03:31:58 2014
From: max at nysolutions.com (Moishe Grunstein)
Date: Tue, 2 Dec 2014 00:31:58 +0000
Subject: [Freeswitch-users] Questions before starting new install
In-Reply-To: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com>
References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA103157920A8@ex2.connex.local>

1)      You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface.

2)      Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off with one of their scripts.

3)      If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from?


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>
Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 Allison Walters
Sent: Monday, December 1, 2014 6:51 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Questions before starting new install

We're wanting to replace our current freeswitch server (it's fairly old), and I have a few things I'm curious about before I start trying to build the new one.  I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience.


1.       I was told by the consultant that set up the current box that freeswitch doesn't play well with fusionpbx or really any of the GUI's - there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion.  Also I was told it doesn't provide nice clean XML, etc...  I was strongly encouraged to go with no GUI.  If it were just me, I'd be okay with that, but we have a couple of other people who might be helping to maintain the server and they're far more comfortable with GUIs.  Has anyone else had any of these issues, and/or have any recommendations on the subject?

2.       I'm trying to install on Ubuntu.  When I go to the wiki, I get this page:  https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu.  However, that line just seems to have pre-reqs and not freeswitch itself.  Anyone know where I could find current Ubuntu instructions?  I googled but the most recent pages I could find were from 2012, and most were from 2011.  Is that the most up to date?

3.       Are there any guidelines for migration from an old server running an old version to a new server with a current version?

Thank you!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.

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From aademattia at comcast.net  Tue Dec  2 04:37:18 2014
From: aademattia at comcast.net (Andrew)
Date: Mon, 1 Dec 2014 20:37:18 -0500
Subject: [Freeswitch-users] Send_DTMF
Message-ID: <01f601d00dd0$86da3370$948e9a50$@comcast.net>

Hi,

I am making an outbound call then I am doing a bridge.

Once I bridge the call I need to send dtmf to second call (bridge call)

When doing a trace it looks like the DTMF is going to my first call and not
the bridge call.

I am using UUID send dtmf using both UUID but both don't seem to work.

 

I do see the DTMF in wire shark but again looks like its going the wrong
direction.

 

Andrew

 

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From AWalters at cawst.org  Tue Dec  2 04:51:06 2014
From: AWalters at cawst.org (Allison Walters)
Date: Tue, 2 Dec 2014 01:51:06 +0000
Subject: [Freeswitch-users] Questions before starting new install
In-Reply-To: <CAAq2U1DtMBa2txADfgPw+bAMaSPBXCDT2gt9qBxB44abUwYPHQ@mail.gmail.com>
References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com>
	<CAAq2U1DtMBa2txADfgPw+bAMaSPBXCDT2gt9qBxB44abUwYPHQ@mail.gmail.com>
Message-ID: <78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com>

Whoops ? accidentally had mail delivery disabled.  So I?ll respond to Moishe?s points with a manual reply/quote:

> 1)      You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface.

I didn?t change anything ? they were both already installed.  Mind you, I don?t know how the person who installed it did it; maybe that?s related to the issues they were seeing?  Regardless, the Fusion interface seems to read all of the correct info, including when things are changed in the XML files (which is how I was encouraged to do changes?).  Now, I?m confused ? Fusion keeps its own copy of the config files?  It?s not just a wrapper that modifies the existing freeswitch ones?  (Just going off what I was told?)


> 2)      Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off with one of their scripts.

I looked at their script, but it didn?t seem to have been updated since 2012, which makes me a bit nervous.  I?ll take another peek, though.


> 3)      If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from?

The current server is running 1.3.0.

Thanks!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.


From: Allison Walters <AWalters at cawst.org<mailto:AWalters at cawst.org>>
Date: Mon, Dec 1, 2014 at 3:50 PM
Subject: [Freeswitch-users] Questions before starting new install
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>

We?re wanting to replace our current freeswitch server (it?s fairly old), and I have a few things I?m curious about before I start trying to build the new one.  I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience.


1.       I was told by the consultant that set up the current box that freeswitch doesn?t play well with fusionpbx or really any of the GUI?s ? there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion.  Also I was told it doesn?t provide nice clean XML, etc?  I was strongly encouraged to go with no GUI.  If it were just me, I?d be okay with that, but we have a couple of other people who might be helping to maintain the server and they?re far more comfortable with GUIs.  Has anyone else had any of these issues, and/or have any recommendations on the subject?

2.       I?m trying to install on Ubuntu.  When I go to the wiki, I get this page:  https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu.  However, that line just seems to have pre-reqs and not freeswitch itself.  Anyone know where I could find current Ubuntu instructions?  I googled but the most recent pages I could find were from 2012, and most were from 2011.  Is that the most up to date?

3.       Are there any guidelines for migration from an old server running an old version to a new server with a current version?

Thank you!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264<tel:%2B1-403-243-3285%20ext.264>
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.


_________________________________________________________________________
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 max at nysolutions.com  Tue Dec  2 07:37:15 2014
From: max at nysolutions.com (Moishe Grunstein)
Date: Tue, 2 Dec 2014 04:37:15 +0000
Subject: [Freeswitch-users] Questions before starting new install
In-Reply-To: <78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com>
References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com>
	<CAAq2U1DtMBa2txADfgPw+bAMaSPBXCDT2gt9qBxB44abUwYPHQ@mail.gmail.com>
	<78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315792D03@ex2.connex.local>

1)      FusionPBX renames Freeswitch default configs after installation, it uses and then creates its own XML files (unless you use their XML handler which writes to db instead). If you then modify XML files directly, the next time you make a change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn and issue tracker..

2)      Not sure what scripts you are referring to be dated, FusionPBX scripts have been last modified this month http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/install_fusionpbx.sh



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>
Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 Allison Walters
Sent: Monday, December 1, 2014 8:51 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Questions before starting new install

Whoops ? accidentally had mail delivery disabled.  So I?ll respond to Moishe?s points with a manual reply/quote:

> 1)      You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface.

I didn?t change anything ? they were both already installed.  Mind you, I don?t know how the person who installed it did it; maybe that?s related to the issues they were seeing?  Regardless, the Fusion interface seems to read all of the correct info, including when things are changed in the XML files (which is how I was encouraged to do changes?).  Now, I?m confused ? Fusion keeps its own copy of the config files?  It?s not just a wrapper that modifies the existing freeswitch ones?  (Just going off what I was told?)


> 2)      Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off with one of their scripts.

I looked at their script, but it didn?t seem to have been updated since 2012, which makes me a bit nervous.  I?ll take another peek, though.


> 3)      If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from?

The current server is running 1.3.0.

Thanks!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.


From: Allison Walters <AWalters at cawst.org<mailto:AWalters at cawst.org>>
Date: Mon, Dec 1, 2014 at 3:50 PM
Subject: [Freeswitch-users] Questions before starting new install
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
We?re wanting to replace our current freeswitch server (it?s fairly old), and I have a few things I?m curious about before I start trying to build the new one.  I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience.


1.       I was told by the consultant that set up the current box that freeswitch doesn?t play well with fusionpbx or really any of the GUI?s ? there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion.  Also I was told it doesn?t provide nice clean XML, etc?  I was strongly encouraged to go with no GUI.  If it were just me, I?d be okay with that, but we have a couple of other people who might be helping to maintain the server and they?re far more comfortable with GUIs.  Has anyone else had any of these issues, and/or have any recommendations on the subject?

2.       I?m trying to install on Ubuntu.  When I go to the wiki, I get this page:  https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu.  However, that line just seems to have pre-reqs and not freeswitch itself.  Anyone know where I could find current Ubuntu instructions?  I googled but the most recent pages I could find were from 2012, and most were from 2011.  Is that the most up to date?

3.       Are there any guidelines for migration from an old server running an old version to a new server with a current version?

Thank you!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264<tel:%2B1-403-243-3285%20ext.264>
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.


_________________________________________________________________________
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 jaybinks at gmail.com  Tue Dec  2 07:54:31 2014
From: jaybinks at gmail.com (jay binks)
Date: Tue, 2 Dec 2014 14:54:31 +1000
Subject: [Freeswitch-users] Crazy Cause Code with "INVALID PROFILE"
Message-ID: <CAO47Hpf5iqWJcfHQRQgQbEFMXWNvPK2Zn2_DnKSiZPY453xaoQ@mail.gmail.com>

So im doing some integration testing, I am sending a call to sofia profile
A , and its routed out profile B.

if I take profile B down,  I can see freeswitch logs "mod_sofia.c:4417
Invalid Profile", which is fine..  the profile is stopped.

but what I did not expect is that Freeswitch responds ( on profile A )
with :

SIP/2.0 502 Bad Gateway
Reason: SIP;cause=611;text="INVALID_PROFILE"


which isnt even listed in
https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table
as a valid hangup cause !!

What I want to know is,  how can I make freeswitch send a Q850 cause 41 in
this case ??
( without patching mod_sofia :P )

-- 
Sincerely

Jay
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From steveayre at gmail.com  Tue Dec  2 12:10:51 2014
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 2 Dec 2014 09:10:51 +0000
Subject: [Freeswitch-users] Crazy Cause Code with "INVALID PROFILE"
In-Reply-To: <CAO47Hpf5iqWJcfHQRQgQbEFMXWNvPK2Zn2_DnKSiZPY453xaoQ@mail.gmail.com>
References: <CAO47Hpf5iqWJcfHQRQgQbEFMXWNvPK2Zn2_DnKSiZPY453xaoQ@mail.gmail.com>
Message-ID: <CAFiqYumRED6heJvuSFsdLq5p0dhrO8zrjxrnGs+6m=CaMNkA5Q@mail.gmail.com>

You can have it return to the dialplan on failure and (conditionally?)
hangup with a 41 from there.

On 2 December 2014 at 04:54, jay binks <jaybinks at gmail.com> wrote:

> So im doing some integration testing, I am sending a call to sofia profile
> A , and its routed out profile B.
>
> if I take profile B down,  I can see freeswitch logs "mod_sofia.c:4417
> Invalid Profile", which is fine..  the profile is stopped.
>
> but what I did not expect is that Freeswitch responds ( on profile A )
> with :
>
> SIP/2.0 502 Bad Gateway
> Reason: SIP;cause=611;text="INVALID_PROFILE"
>
>
> which isnt even listed in
> https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table
> as a valid hangup cause !!
>
> What I want to know is,  how can I make freeswitch send a Q850 cause 41 in
> this case ??
> ( without patching mod_sofia :P )
>
> --
> Sincerely
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From telishisheer at gmail.com  Tue Dec  2 07:25:53 2014
From: telishisheer at gmail.com (Shisheer Teli)
Date: Tue, 2 Dec 2014 09:55:53 +0530
Subject: [Freeswitch-users] Ldap alise and freeswitch integration problem
Message-ID: <CACkRGLyAT_HwST0zGPfJX2pEsAJhYPZ9dqWMdkvnTZbyG8hbiA@mail.gmail.com>

Hi,

I am able to bind with any alise on ldap server except userPassword (MD5)
alise.

when i bind password with userPassword , authentication fails.

any MD5 authentication configuration in freeswitch?

Configuration file:
<configuration name="xml_ldap.conf">

<bindings>
<binding name="directory">
  <param name="basedn" value="dc=example,dc=com"/>
  <param name="filter" value="(telephoneNumber=%d)" bindings="directory"/>
  <param name="url" value="ldap://ldap.example.com"/>
  <param name="binddn" value="cn=use,dc=example,dc=com"/>
  <param name="bindpass" value="XXXX"/>
   <trans>
   <tran name="id" mapfrom="uid"/>
   <tran name="password" mapfrom="userPassword"/>
   </trans>
</binding>
</bindings>

</configuration>



-- 
Regards,
Shisheer T
Phone: +91-022 2278 2763
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From brian at freeswitch.org  Tue Dec  2 15:12:03 2014
From: brian at freeswitch.org (Brian West)
Date: Tue, 2 Dec 2014 06:12:03 -0600
Subject: [Freeswitch-users] Crazy Cause Code with "INVALID PROFILE"
In-Reply-To: <CAO47Hpf5iqWJcfHQRQgQbEFMXWNvPK2Zn2_DnKSiZPY453xaoQ@mail.gmail.com>
References: <CAO47Hpf5iqWJcfHQRQgQbEFMXWNvPK2Zn2_DnKSiZPY453xaoQ@mail.gmail.com>
Message-ID: <CAEJMVkC42GpPGFHE98zOahw5wnReEoa74SSR-ONZNp+94z0Vyw@mail.gmail.com>

Those were added as part of FS-3321, Try sending your calls out a valid
profile? :P

On Mon, Dec 1, 2014 at 10:54 PM, jay binks <jaybinks at gmail.com> wrote:

> So im doing some integration testing, I am sending a call to sofia profile
> A , and its routed out profile B.
>
> if I take profile B down,  I can see freeswitch logs "mod_sofia.c:4417
> Invalid Profile", which is fine..  the profile is stopped.
>
> but what I did not expect is that Freeswitch responds ( on profile A )
> with :
>
> SIP/2.0 502 Bad Gateway
> Reason: SIP;cause=611;text="INVALID_PROFILE"
>
>
> which isnt even listed in
> https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table
> as a valid hangup cause !!
>
> What I want to know is,  how can I make freeswitch send a Q850 cause 41 in
> this case ??
> ( without patching mod_sofia :P )
>
> --
> Sincerely
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From jaybinks at gmail.com  Tue Dec  2 15:37:36 2014
From: jaybinks at gmail.com (jay binks)
Date: Tue, 2 Dec 2014 22:37:36 +1000
Subject: [Freeswitch-users] Crazy Cause Code with "INVALID PROFILE"
In-Reply-To: <CAEJMVkC42GpPGFHE98zOahw5wnReEoa74SSR-ONZNp+94z0Vyw@mail.gmail.com>
References: <CAO47Hpf5iqWJcfHQRQgQbEFMXWNvPK2Zn2_DnKSiZPY453xaoQ@mail.gmail.com>
	<CAEJMVkC42GpPGFHE98zOahw5wnReEoa74SSR-ONZNp+94z0Vyw@mail.gmail.com>
Message-ID: <CAO47Hpcuz7AXnSv+ph5=T=uVqPHatvfkaJm9=095txWVBXmDcA@mail.gmail.com>

yea your probably right that im never really going to have that scenario...
however I wanted to test what would happen ( on profile A side ) if link
was down on the interface for profile B.

hmmm have to test that better :P

On 2 December 2014 at 22:12, Brian West <brian at freeswitch.org> wrote:

> Those were added as part of FS-3321, Try sending your calls out a valid
> profile? :P
>
> On Mon, Dec 1, 2014 at 10:54 PM, jay binks <jaybinks at gmail.com> wrote:
>
>> So im doing some integration testing, I am sending a call to sofia
>> profile A , and its routed out profile B.
>>
>> if I take profile B down,  I can see freeswitch logs "mod_sofia.c:4417
>> Invalid Profile", which is fine..  the profile is stopped.
>>
>> but what I did not expect is that Freeswitch responds ( on profile A )
>> with :
>>
>> SIP/2.0 502 Bad Gateway
>> Reason: SIP;cause=611;text="INVALID_PROFILE"
>>
>>
>> which isnt even listed in
>> https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table
>> as a valid hangup cause !!
>>
>> What I want to know is,  how can I make freeswitch send a Q850 cause 41
>> in this case ??
>> ( without patching mod_sofia :P )
>>
>> --
>> Sincerely
>>
>> Jay
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> *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
>



-- 
Sincerely

Jay
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From achinthau at gmail.com  Tue Dec  2 16:11:24 2014
From: achinthau at gmail.com (Achintha)
Date: Tue, 2 Dec 2014 18:41:24 +0530
Subject: [Freeswitch-users] Fwd: mod_verto configuration
Message-ID: <CA+JwzxtFGUw0XAv6x_mxxr5wsXqZdguYLA++PW5STvGwUNNWeA@mail.gmail.com>

Hi

I used freeswitch  on centos 6 .

I downloaded freeswitch from ?git clone
https://freeswitch.org/stash/scm/fs/freeswitch.git

And  configure properly without errors

Then I try to configure mod_verto ,

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

First I compile the module and then I did configurations from mentioned
link below,

But I couldn?t create ssl certificate from that instructions

Then I used following command to create the certificate and key



openssl req -x509 -nodes -days 365 -newkey rsa:2048 -keyout
/usr/local/freeswitch/certs/wss.key -out
/usr/local/freeswitch/certs/wss.crt



cat  wss.crt  wss.key  >  /usr/local/freeswitch/certs/wss.pem

cat  wss.crt   >  /usr/local/freeswitch/tls/dtls-srtp.crt

then I changed ssl.conf file (/etc/httpd/conf.d/ssl.conf) as follows


ServerName  myserverip:443



SSLEngine on

SSLCertificateFile  /usr/local/freeswitch/certs/wss.crt

SSLCertificateKeyFile  /usr/local/freeswitch/certs/wss.key

Then restarted httpd server



Get the firefox and put https:// myserverip/verto

Ask certificate and after add the certificate

Loaded login page but can?t log in

freeswitch console print  following message



2014-12-02 18:11:40.150201 [INFO] mod_verto.c:3688 192.168.1.195:4870
Client Connect.

2014-12-02 18:11:40.150201 [INFO] mod_verto.c:1827 192.168.1.195:4870
Starting client thread.

2014-12-02 18:11:41.169676 [DEBUG] mod_verto.c:1739 192.168.1.195:4870 WS
SETUP FAILED []

2014-12-02 18:11:41.169676 [INFO] mod_verto.c:1853 192.168.1.195:4870
Ending client thread.

2014-12-02 18:11:41.169676 [INFO] mod_verto.c:1860 192.168.1.195:4870
Thread ended



Kindly help me to configure this

thankyou

ach
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From avi at avimarcus.net  Tue Dec  2 16:22:22 2014
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 2 Dec 2014 13:22:22 +0000
Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound
 Caller ID to external SIP Gateway URI
In-Reply-To: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com>
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com>
Message-ID: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com>

Hi - did you figure this out yet?

One comment:
 <action application="bridge" data="sofia/gateway/othersipgw/+234$1"/> -- I
don't know if $1 is available anymore. You might want to just set that as
part of the actual number to route, e.g. add a 1212 prefix and match it
again or set it as a channel variable. You can see in the logs if the $1 is
resolving correctly.

-Avi
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From dujinfang at gmail.com  Tue Dec  2 16:52:10 2014
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 2 Dec 2014 21:52:10 +0800
Subject: [Freeswitch-users] Send_DTMF
In-Reply-To: <01f601d00dd0$86da3370$948e9a50$@comcast.net>
References: <01f601d00dd0$86da3370$948e9a50$@comcast.net>
Message-ID: <855745AA993749539E5AF564EA1F61A4@gmail.com>

try uuid_recv_dtmf  

--  
Seven Du
http://about.me/dujinfang
http://www.dujinfang.com

http://www.freeswitch.org.cn


Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Tuesday, December 2, 2014 at 9:37 AM, Andrew wrote:

> Hi,
> I am making an outbound call then I am doing a bridge.
> Once I bridge the call I need to send dtmf to second call (bridge call)
> When doing a trace it looks like the DTMF is going to my first call and not the bridge call.
> I am using UUID send dtmf using both UUID but both don?t seem to work.
>   
> I do see the DTMF in wire shark but again looks like its going the wrong direction.
>   
> Andrew
>   
>  
>  
> _________________________________________________________________________
> 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 aademattia at comcast.net  Tue Dec  2 18:04:31 2014
From: aademattia at comcast.net (=?utf-8?B?YWFkZW1hdHRpYUBjb21jYXN0Lm5ldA==?=)
Date: Tue, 02 Dec 2014 10:04:31 -0500
Subject: [Freeswitch-users] =?utf-8?q?Send=5FDTMF?=
Message-ID: <mailman.1.1417532706.578.freeswitch-users@lists.freeswitch.org>

I still see the dtmf going to the wrong destination.
I am sending dtmf on both uuid and can't seem to get the ivr I am calling to see my dtmf. But if I hit dtmf on my phone it works.

Sent from my HTC

----- Reply message -----
From: "Seven Du" <dujinfang at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Send_DTMF
Date: Tue, Dec 2, 2014 8:52 AM

try uuid_recv_dtmf



-- 
Seven Du
http://about.me/dujinfang
http://www.dujinfang.com
http://www.freeswitch.org.cn


Sent with Sparrow




On Tuesday, December 2, 2014 at 9:37 AM, Andrew wrote:
Hi,I am making an outbound call then I am doing a bridge.Once I bridge the call I need to send dtmf to second call (bridge call)When doing a trace it looks like the DTMF is going to my first call and not the bridge call.I am using UUID send dtmf using both UUID but both don?t seem to work. I do see the DTMF in wire shark but again looks like its going the wrong direction. Andrew 
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch..org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing 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 steveayre at gmail.com  Tue Dec  2 18:08:13 2014
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 2 Dec 2014 15:08:13 +0000
Subject: [Freeswitch-users] Crazy Cause Code with "INVALID PROFILE"
In-Reply-To: <CAO47Hpcuz7AXnSv+ph5=T=uVqPHatvfkaJm9=095txWVBXmDcA@mail.gmail.com>
References: <CAO47Hpf5iqWJcfHQRQgQbEFMXWNvPK2Zn2_DnKSiZPY453xaoQ@mail.gmail.com>
	<CAEJMVkC42GpPGFHE98zOahw5wnReEoa74SSR-ONZNp+94z0Vyw@mail.gmail.com>
	<CAO47Hpcuz7AXnSv+ph5=T=uVqPHatvfkaJm9=095txWVBXmDcA@mail.gmail.com>
Message-ID: <CAFiqYukH-LO-1aGB26XP9isJNQ5WpMrZEiROwJr+oGSnJN7+PA@mail.gmail.com>

If the link is down the profile will still exist but packets won't route.
Unless you mean when it's down on startup?

On 2 December 2014 at 12:37, jay binks <jaybinks at gmail.com> wrote:

> yea your probably right that im never really going to have that scenario...
> however I wanted to test what would happen ( on profile A side ) if link
> was down on the interface for profile B.
>
> hmmm have to test that better :P
>
> On 2 December 2014 at 22:12, Brian West <brian at freeswitch.org> wrote:
>
>> Those were added as part of FS-3321, Try sending your calls out a valid
>> profile? :P
>>
>> On Mon, Dec 1, 2014 at 10:54 PM, jay binks <jaybinks at gmail.com> wrote:
>>
>>> So im doing some integration testing, I am sending a call to sofia
>>> profile A , and its routed out profile B.
>>>
>>> if I take profile B down,  I can see freeswitch logs "mod_sofia.c:4417
>>> Invalid Profile", which is fine..  the profile is stopped.
>>>
>>> but what I did not expect is that Freeswitch responds ( on profile A )
>>> with :
>>>
>>> SIP/2.0 502 Bad Gateway
>>> Reason: SIP;cause=611;text="INVALID_PROFILE"
>>>
>>>
>>> which isnt even listed in
>>> https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table
>>> as a valid hangup cause !!
>>>
>>> What I want to know is,  how can I make freeswitch send a Q850 cause 41
>>> in this case ??
>>> ( without patching mod_sofia :P )
>>>
>>> --
>>> Sincerely
>>>
>>> Jay
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> *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
>>
>
>
>
> --
> Sincerely
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Tue Dec  2 18:57:54 2014
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Dec 2014 07:57:54 -0800
Subject: [Freeswitch-users] UniFi phones and FreeSWITCH
Message-ID: <CAKzWOxW4DytBiJ3HY5ePuL9Y8VRS_y_P_hm0WvDo=4Q=35aAGg@mail.gmail.com>

Hello FreeSWITCHers!

Does anybody have any experience with these phones?
http://www.ubnt.com/unifi-voip/uvp/

I'm curious to know how they work in a FS environment.

-MC
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From max at nysolutions.com  Tue Dec  2 19:12:56 2014
From: max at nysolutions.com (Moishe Grunstein)
Date: Tue, 2 Dec 2014 16:12:56 +0000
Subject: [Freeswitch-users] UniFi phones and FreeSWITCH
In-Reply-To: <CAKzWOxW4DytBiJ3HY5ePuL9Y8VRS_y_P_hm0WvDo=4Q=35aAGg@mail.gmail.com>
References: <CAKzWOxW4DytBiJ3HY5ePuL9Y8VRS_y_P_hm0WvDo=4Q=35aAGg@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315794DA5@ex2.connex.local>

They are only shipping the basic model at the moment.

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>
Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 Michael Collins
Sent: Tuesday, December 2, 2014 10:58 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] UniFi phones and FreeSWITCH

Hello FreeSWITCHers!
Does anybody have any experience with these phones?
http://www.ubnt.com/unifi-voip/uvp/
I'm curious to know how they work in a FS environment.
-MC
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From AWalters at cawst.org  Tue Dec  2 19:41:46 2014
From: AWalters at cawst.org (Allison Walters)
Date: Tue, 2 Dec 2014 16:41:46 +0000
Subject: [Freeswitch-users] Questions before starting new install
In-Reply-To: <BA6A05D48E282C4C9989496521266BA10315792D03@ex2.connex.local>
References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com>
	<CAAq2U1DtMBa2txADfgPw+bAMaSPBXCDT2gt9qBxB44abUwYPHQ@mail.gmail.com>
	<78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com>
	<BA6A05D48E282C4C9989496521266BA10315792D03@ex2.connex.local>
Message-ID: <70694fb57953496b88aae1666accac7e@BN1PR0101MB0817.prod.exchangelabs.com>

Hmmm?that?s a different version of the script than the link I found, I think.  Thanks!

So it?s starting to sound like the problem may be that Fusion was installed after the fact on the current server, and so things don?t integrate quite properly (or at least as expected).  Maybe I?ll give a test install of both a go, then.  Thanks!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moishe Grunstein
Sent: Monday, December 01, 2014 9:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Questions before starting new install


1)      FusionPBX renames Freeswitch default configs after installation, it uses and then creates its own XML files (unless you use their XML handler which writes to db instead). If you then modify XML files directly, the next time you make a change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn and issue tracker..

2)      Not sure what scripts you are referring to be dated, FusionPBX scripts have been last modified this month http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/install_fusionpbx.sh



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>
Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified
[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> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Allison Walters
Sent: Monday, December 1, 2014 8:51 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Questions before starting new install

Whoops ? accidentally had mail delivery disabled.  So I?ll respond to Moishe?s points with a manual reply/quote:

> 1)      You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface.

I didn?t change anything ? they were both already installed.  Mind you, I don?t know how the person who installed it did it; maybe that?s related to the issues they were seeing?  Regardless, the Fusion interface seems to read all of the correct info, including when things are changed in the XML files (which is how I was encouraged to do changes?).  Now, I?m confused ? Fusion keeps its own copy of the config files?  It?s not just a wrapper that modifies the existing freeswitch ones?  (Just going off what I was told?)


> 2)      Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off with one of their scripts.

I looked at their script, but it didn?t seem to have been updated since 2012, which makes me a bit nervous.  I?ll take another peek, though.


> 3)      If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from?

The current server is running 1.3.0.

Thanks!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.


From: Allison Walters <AWalters at cawst.org<mailto:AWalters at cawst.org>>
Date: Mon, Dec 1, 2014 at 3:50 PM
Subject: [Freeswitch-users] Questions before starting new install
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
We?re wanting to replace our current freeswitch server (it?s fairly old), and I have a few things I?m curious about before I start trying to build the new one.  I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience.


1.       I was told by the consultant that set up the current box that freeswitch doesn?t play well with fusionpbx or really any of the GUI?s ? there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion.  Also I was told it doesn?t provide nice clean XML, etc?  I was strongly encouraged to go with no GUI.  If it were just me, I?d be okay with that, but we have a couple of other people who might be helping to maintain the server and they?re far more comfortable with GUIs.  Has anyone else had any of these issues, and/or have any recommendations on the subject?

2.       I?m trying to install on Ubuntu.  When I go to the wiki, I get this page:  https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu.  However, that line just seems to have pre-reqs and not freeswitch itself.  Anyone know where I could find current Ubuntu instructions?  I googled but the most recent pages I could find were from 2012, and most were from 2011.  Is that the most up to date?

3.       Are there any guidelines for migration from an old server running an old version to a new server with a current version?

Thank you!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264<tel:%2B1-403-243-3285%20ext.264>
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.


_________________________________________________________________________
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 jlamanna at gmail.com  Tue Dec  2 19:49:10 2014
From: jlamanna at gmail.com (James Lamanna)
Date: Tue, 2 Dec 2014 11:49:10 -0500
Subject: [Freeswitch-users] Missing CALLID occasionally on SpanDSP modems
Message-ID: <CADScKLx+39FdKDv8sK-PFRSF13rUom4W+9kwJ+P+1ZDFVBivww@mail.gmail.com>

Hi,
I'm using FreeSwitch SpanDSP Modems in conjunction with Hylafax, and I'm
noticing that occasionally Hylafax gets no CALLID information:

Dec 02 08:13:22.45: [ 7439]: SESSION BEGIN 000000080 2292330412
Dec 02 08:13:22.45: [ 7439]: HylaFAX (tm) Version 5.5.5
Dec 02 08:13:22.45: [ 7439]: CallID: '' '' '' ''

This doesn't happen consistently, but I've already seen it on around 3
faxes out of 40.
Freeswitch prints out an appropriate callerId to my logs.

This is what my Hylafax config file looks like:

CountryCode:        1
AreaCode:       000
FAXNumber:      +1.999.555.1212
LongDistancePrefix: 1
InternationalPrefix:    011
DialStringRules:    etc/dialrules
ServerTracing:      1
SessionTracing:     0xFFF
RecvFileMode:       0660
LogFileMode:        0660
DeviceMode:     0660
RingsBeforeAnswer:  2
SpeakerVolume:      off
GettyArgs:      "-h %l dx_%s"
LocalIdentifier:    "NothingSetup"
TagLineFont:        etc/LiberationSans-25.pcf
TagLineFormat:      "From %%l|%c|Page %%P of %%T"
MaxRecvPages:       500
ModemType:      Class1      # use this to supply a hint
ModemResetCmds:     AT+VCID=1   # enables CallID display
PagerTTYParity:     none

Class1AdaptRecvCmd: AT+FAR=1
Class1TMConnectDelay:   400     # counteract quick CONNECT response
CallIDPattern:          "NMBR="
CallIDPattern:          "NAME="
CallIDPattern:      "ANID="
CallIDPattern:          "NDID="
DynamicConfig: "/usr/local/bin/faxreceiveconfig"
MaxBatchJobs: "1"
BadPageHandlingMethod: DCN

Any ideas?

Thanks.

(please CC me directly, I am viewing in digest mode)

-- James
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From raphael.lechner at gmail.com  Tue Dec  2 19:49:57 2014
From: raphael.lechner at gmail.com (Raphael Lechner)
Date: Tue, 2 Dec 2014 17:49:57 +0100
Subject: [Freeswitch-users] table sip_presence is always empty
Message-ID: <AE177ACE-B1A5-4245-9648-87080A94AE11@gmail.com>

Hi,

I?m trying to enable presence but for now the sip_presence table is always empty and I don?t know If that is ok or not.
This is my internal.xml https://pastebin.freeswitch.org/23666 <https://pastebin.freeswitch.org/23666> 
I tried with the sqlite db and postgresql but nothing has changed.

The table exists but is always empty:
select * from sip_presence ;
 sip_user | sip_host | status | rpid | expires | user_agent | profile_name | hostname | network_ip | network_port | open_closed 
----------+----------+--------+------+---------+------------+--------------+----------+------------+--------------+-------------
(0 rows)

The table sip_subscriptions contains entries like:
 sip   | 99       | 192.168.130.130 | 12          | 192.168.130.130 | 192.168.130.130,192.168.130.130,test-pbx | presence | "Raphael" <sip:99 at 192.168.130.131:27050> | N2RiMDQ2OTkwMWVhZjYzMjIwMDM0MGZlZjU4NDM1ZDY | "Raphael" <sip:99 at 192.168.130.130>;tag=a4cb4203 | SIP/2.0/UDP 192.168.130.131:27050;branch=z9hG4bK-d8754z-53ed9718e7866617-1---d8754z-;rport=27050 | 1417537731 | Bria 4 4.1.1 74256-688c7d71-M | multipart/related, application/rlmi+xml, application/pidf+xml  | internal     | test-pbx | 27050        | 192.168.130.131 |      16 |            | <sip:12 at 192.168.130.130>;tag=wkBl3G43vfhs

and with sofia_presence_data the status is always unknown:
freeswitch at internal> sofia_presence_data list */99
status|rpid|user_agent|network_ip|network_port
unknown|unknown|Bria 4 4.1.1 74256-688c7d71-M|192.168.130.131|27050
+OK

I have enabled debugging for presence:
https://pastebin.freeswitch.org/23667 <https://pastebin.freeswitch.org/23667>

FreeSWITCH Version:
FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)

Any hint what I have configured wrong?

Thank you,
Raphael
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From krice at freeswitch.org  Tue Dec  2 20:09:26 2014
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 02 Dec 2014 17:09:26 +0000
Subject: [Freeswitch-users] Freeswitch Week in Review (Master Branch)
	November 23rd-29th
Message-ID: <547df2467a147_3aa175b32c65017@ip-10-16-129-130.mail>

New Post on freeswitch.org from kathleen
check it out at http://ift.tt/11P0e1O
Freeswitch Week in Review (Master Branch) November 23rd-29th
Hello, again. This week in the FreeSWITCH master branch we had 7 commits. It was a quiet week with the Thanksgiving holiday falling on Thursday, and the majority of the work done was geared toward build support for unimrcp.

Improvements in cross platform build supports:

32c27b3 Added a Debian dependency to the CentOS6 makefile

f4876d5 FS-7031 [unimrcp] update sofia-sip.m4 so that it can build when relative path is used in configure.gnu ?with-sofia-sip

[Jira: http://ift.tt/1wkDnZk]

061f3cb FS-7031 #resolve #comment [unimrcp] update library again to pull in upstream fix for ?with-sofia-sip=../sofia-sip

[Jira: http://ift.tt/1wkDnZk]

382e683 Use FTDM_UINT64_FMT macro to log uint64_t values, in order to not break x86 builds.

The following bugs were squashed:

5bbef7f FS-7015 Fix for inbound call on hold issue in mod_sofia

[Jira: http://ift.tt/1wkDp3C]

Miscellaneous commits:

0d636af FS-7031 [unimrcp] revert configure.gnu change- doesn?t work when using non-source build dir.

[Jira: http://ift.tt/1wkDnZk]

The complete list of commits can be found here:2014_11_23-2014_11_30new

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From AWalters at cawst.org  Tue Dec  2 21:40:47 2014
From: AWalters at cawst.org (Allison Walters)
Date: Tue, 2 Dec 2014 18:40:47 +0000
Subject: [Freeswitch-users] Questions before starting new install
In-Reply-To: <BA6A05D48E282C4C9989496521266BA10315792D03@ex2.connex.local>
References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com>
	<CAAq2U1DtMBa2txADfgPw+bAMaSPBXCDT2gt9qBxB44abUwYPHQ@mail.gmail.com>
	<78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com>
	<BA6A05D48E282C4C9989496521266BA10315792D03@ex2.connex.local>
Message-ID: <fea24868db514600b1680df34b3fcdc3@BN1PR0101MB0817.prod.exchangelabs.com>

I tried that version of the script and it says it was written for 10.04 and 12.04, and it says it doesn?t always work outside of that, but it?ll ?give it a go?.  And then after a bit, it failed out with a git error:

installing stable v1.4 of FreeSWITCH
/usr/bin/time: cannot run /usr/bin/git: No such file or directory
Command exited with non-zero status 127
0.00user 0.00system 0:00.00elapsed 0%CPU (0avgtext+0avgdata 1220maxresident)k
0inputs+0outputs (0major+27minor)pagefaults 0swaps
./install_fusionpbx.sh: line 1114: cd: /usr/src/freeswitch: No such file or directory
./install_fusionpbx.sh: line 1115: /usr/bin/git: No such file or directory
GIT ERROR

I suspect this is because freeswitch moved to stash in July, apparently.  Anyway, I?ll see if I can find anyone on the Fusion forum or IRC or somewhere that can help with this part.  Thanks!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moishe Grunstein
Sent: Monday, December 01, 2014 9:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Questions before starting new install


1)      FusionPBX renames Freeswitch default configs after installation, it uses and then creates its own XML files (unless you use their XML handler which writes to db instead). If you then modify XML files directly, the next time you make a change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn and issue tracker..

2)      Not sure what scripts you are referring to be dated, FusionPBX scripts have been last modified this month http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/install_fusionpbx.sh



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>
Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified
[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> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Allison Walters
Sent: Monday, December 1, 2014 8:51 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Questions before starting new install

Whoops ? accidentally had mail delivery disabled.  So I?ll respond to Moishe?s points with a manual reply/quote:

> 1)      You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface.

I didn?t change anything ? they were both already installed.  Mind you, I don?t know how the person who installed it did it; maybe that?s related to the issues they were seeing?  Regardless, the Fusion interface seems to read all of the correct info, including when things are changed in the XML files (which is how I was encouraged to do changes?).  Now, I?m confused ? Fusion keeps its own copy of the config files?  It?s not just a wrapper that modifies the existing freeswitch ones?  (Just going off what I was told?)


> 2)      Ubuntu install is the same as debian https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off with one of their scripts.

I looked at their script, but it didn?t seem to have been updated since 2012, which makes me a bit nervous.  I?ll take another peek, though.


> 3)      If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from?

The current server is running 1.3.0.

Thanks!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.


From: Allison Walters <AWalters at cawst.org<mailto:AWalters at cawst.org>>
Date: Mon, Dec 1, 2014 at 3:50 PM
Subject: [Freeswitch-users] Questions before starting new install
To: "freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
We?re wanting to replace our current freeswitch server (it?s fairly old), and I have a few things I?m curious about before I start trying to build the new one.  I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience.


1.       I was told by the consultant that set up the current box that freeswitch doesn?t play well with fusionpbx or really any of the GUI?s ? there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion.  Also I was told it doesn?t provide nice clean XML, etc?  I was strongly encouraged to go with no GUI.  If it were just me, I?d be okay with that, but we have a couple of other people who might be helping to maintain the server and they?re far more comfortable with GUIs.  Has anyone else had any of these issues, and/or have any recommendations on the subject?

2.       I?m trying to install on Ubuntu.  When I go to the wiki, I get this page:  https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu.  However, that line just seems to have pre-reqs and not freeswitch itself.  Anyone know where I could find current Ubuntu instructions?  I googled but the most recent pages I could find were from 2012, and most were from 2011.  Is that the most up to date?

3.       Are there any guidelines for migration from an old server running an old version to a new server with a current version?

Thank you!


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264<tel:%2B1-403-243-3285%20ext.264>
web: www.cawst.org<http://www.cawst.org/>

Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.


_________________________________________________________________________
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 nneul at mst.edu  Tue Dec  2 21:48:05 2014
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 02 Dec 2014 12:48:05 -0600
Subject: [Freeswitch-users] Questions before starting new install
In-Reply-To: <fea24868db514600b1680df34b3fcdc3@BN1PR0101MB0817.prod.exchangelabs.com>
References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com>	<CAAq2U1DtMBa2txADfgPw+bAMaSPBXCDT2gt9qBxB44abUwYPHQ@mail.gmail.com>	<78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com>	<BA6A05D48E282C4C9989496521266BA10315792D03@ex2.connex.local>
	<fea24868db514600b1680df34b3fcdc3@BN1PR0101MB0817.prod.exchangelabs.com>
Message-ID: <547E0965.6050909@mst.edu>

Do you have git installed on your system? Sure looks like it's complaining that you don't have it installed.

-- Nathan

On 12/02/2014 12:40 PM, Allison Walters wrote:
> I tried that version of the script and it says it was written for 10.04 and 12.04, and it says it doesn?t always work
> outside of that, but it?ll ?give it a go?.  And then after a bit, it failed out with a git error:
>
> installing stable v1.4 of FreeSWITCH
>
> /usr/bin/time: cannot run /usr/bin/git: No such file or directory
>
> Command exited with non-zero status 127
>
> 0.00user 0.00system 0:00.00elapsed 0%CPU (0avgtext+0avgdata 1220maxresident)k
>
> 0inputs+0outputs (0major+27minor)pagefaults 0swaps
>
> ./install_fusionpbx.sh: line 1114: cd: /usr/src/freeswitch: No such file or directory
>
> ./install_fusionpbx.sh: line 1115: /usr/bin/git: No such file or directory
>
> GIT ERROR
>
> I suspect this is because freeswitch moved to stash in July, apparently.  Anyway, I?ll see if I can find anyone on the
> Fusion forum or IRC or somewhere that can help with this part.  Thanks!
>
> --
>
> *Allison Walters*
>
> /IT Coordinator/
>
> CAWST - Centre for Affordable Water and Sanitation Technology
>
> 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
>
> tel: +1-403-243-3285 ext.264
>
> web: www.cawst.org <http://www.cawst.org/>
>
> *//*
>
> */Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. /*
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf
> Of *Moishe Grunstein
> *Sent:* Monday, December 01, 2014 9:37 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Questions before starting new install
>
> 1)FusionPBX renames Freeswitch default configs after installation, it uses and then creates its own XML files (unless
> you use their XML handler which writes to db instead). If you then modify XML files directly, the next time you make a
> change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn
> and issue tracker..
>
> 2)Not sure what scripts you are referring to be dated, FusionPBX scripts have been last modified this month
> http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/install_fusionpbx.sh
>
> 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> *
>
> Polycom Certified VAR
> Microsoft Small Business Specialist, Cisco SMB Select Certified
>
> 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>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Allison Walters
> *Sent:* Monday, December 1, 2014 8:51 PM
> *To:* freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Questions before starting new install
>
> Whoops ? accidentally had mail delivery disabled.  So I?ll respond to Moishe?s points with a manual reply/quote:
>
>> 1)      You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface.
>
> I didn?t change anything ? they were both already installed.  Mind you, I don?t know how the person who installed it did
> it; maybe that?s related to the issues they were seeing?  Regardless, the Fusion interface seems to read all of the
> correct info, including when things are changed in the XML files (which is how I was encouraged to do changes?).  Now,
> I?m confused ? Fusion keeps its own copy of the config files? It?s not just a wrapper that modifies the existing
> freeswitch ones?  (Just going off what I was told?)
>
>> 2)      Ubuntu install is the same as debian  https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off
> with one of their scripts.
>
> I looked at their script, but it didn?t seem to have been updated since 2012, which makes me a bit nervous.  I?ll take
> another peek, though.
>
>> 3)      If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from?
>
> The current server is running 1.3.0.
>
> Thanks!
>
> --
>
> *Allison Walters*
>
> /IT Coordinator/
>
> CAWST - Centre for Affordable Water and Sanitation Technology
>
> 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
>
> tel: +1-403-243-3285 ext.264
>
> web: www.cawst.org <http://www.cawst.org/>
>
> *//*
>
> */Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. /*
>
> From: *Allison Walters* <AWalters at cawst.org <mailto:AWalters at cawst.org>>
> Date: Mon, Dec 1, 2014 at 3:50 PM
> Subject: [Freeswitch-users] Questions before starting new install
> To: "freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>"
> <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
>
> We?re wanting to replace our current freeswitch server (it?s fairly old), and I have a few things I?m curious about
> before I start trying to build the new one.  I inherited the current one and am relatively unfamiliar to all of this, so
> I appreciate your patience.
>
> 1.I was told by the consultant that set up the current box that freeswitch doesn?t play well with fusionpbx or really
> any of the GUI?s ? there have been instances (and indeed one happened a couple of months ago while I was here) of the
> config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion.
> Also I was told it doesn?t provide nice clean XML, etc?  I was strongly encouraged to go with no GUI.  If it were just
> me, I?d be okay with that, but we have a couple of other people who might be helping to maintain the server and they?re
> far more comfortable with GUIs.  Has anyone else had any of these issues, and/or have any recommendations on the subject?
>
> 2.I?m trying to install on Ubuntu.  When I go to the wiki, I get this page:
> https://freeswitch.org/confluence/display/FREESWITCH/Linux , which just gives an apt-get line for Ubuntu.  However, that
> line just seems to have pre-reqs and not freeswitch itself.  Anyone know where I could find current Ubuntu
> instructions?  I googled but the most recent pages I could find were from 2012, and most were from 2011.  Is that the
> most up to date?
>
> 3.Are there any guidelines for migration from an old server running an old version to a new server with a current version?
>
> Thank you!
>
> --
>
> *Allison Walters*
>
> /IT Coordinator/
>
> CAWST - Centre for Affordable Water and Sanitation Technology
>
> 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
>
> tel: +1-403-243-3285 ext.264 <tel:%2B1-403-243-3285%20ext.264>
>
> web: www.cawst.org <http://www.cawst.org/>
>
> *//*
>
> */Join us in reaching 20 million people by 2020 with safe drinking water and sanitation. /*
>
>
> _________________________________________________________________________
> 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
>

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


From AWalters at cawst.org  Tue Dec  2 21:55:34 2014
From: AWalters at cawst.org (Allison Walters)
Date: Tue, 2 Dec 2014 18:55:34 +0000
Subject: [Freeswitch-users] Questions before starting new install
In-Reply-To: <547E0965.6050909@mst.edu>
References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com>
	<CAAq2U1DtMBa2txADfgPw+bAMaSPBXCDT2gt9qBxB44abUwYPHQ@mail.gmail.com>
	<78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com>
	<BA6A05D48E282C4C9989496521266BA10315792D03@ex2.connex.local>
	<fea24868db514600b1680df34b3fcdc3@BN1PR0101MB0817.prod.exchangelabs.com>
	<547E0965.6050909@mst.edu>
Message-ID: <cc8b0e4e86f04626803d0f4e6a78a0c1@BN1PR0101MB0817.prod.exchangelabs.com>

*facepalm*  Sorry - I was installing on the wrong test VM.  Ugh....long morning.  I was kind of prepared for it to not work after the warning message so jumped to wrong conclusion.  :-(  My bad!  Time to grab some food and come back to this, I think...

--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264
web:?www.cawst.org
?
Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nathan Neulinger
Sent: Tuesday, December 02, 2014 11:48 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Questions before starting new install

Do you have git installed on your system? Sure looks like it's complaining that you don't have it installed.

-- Nathan

On 12/02/2014 12:40 PM, Allison Walters wrote:
> I tried that version of the script and it says it was written for 
> 10.04 and 12.04, and it says it doesn't always work outside of that, but it'll "give it a go".  And then after a bit, it failed out with a git error:
>
> installing stable v1.4 of FreeSWITCH
>
> /usr/bin/time: cannot run /usr/bin/git: No such file or directory
>
> Command exited with non-zero status 127
>
> 0.00user 0.00system 0:00.00elapsed 0%CPU (0avgtext+0avgdata 
> 1220maxresident)k
>
> 0inputs+0outputs (0major+27minor)pagefaults 0swaps
>
> ./install_fusionpbx.sh: line 1114: cd: /usr/src/freeswitch: No such 
> file or directory
>
> ./install_fusionpbx.sh: line 1115: /usr/bin/git: No such file or 
> directory
>
> GIT ERROR
>
> I suspect this is because freeswitch moved to stash in July, 
> apparently.  Anyway, I'll see if I can find anyone on the Fusion forum or IRC or somewhere that can help with this part.  Thanks!
>
> --
>
> *Allison Walters*
>
> /IT Coordinator/
>
> CAWST - Centre for Affordable Water and Sanitation Technology
>
> 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
>
> tel: +1-403-243-3285 ext.264
>
> web: www.cawst.org <http://www.cawst.org/>
>
> *//*
>
> */Join us in reaching 20 million people by 2020 with safe drinking 
> water and sanitation. /*
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Moishe Grunstein
> *Sent:* Monday, December 01, 2014 9:37 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Questions before starting new 
> install
>
> 1)FusionPBX renames Freeswitch default configs after installation, it 
> uses and then creates its own XML files (unless you use their XML 
> handler which writes to db instead). If you then modify XML files 
> directly, the next time you make a change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn and issue tracker..
>
> 2)Not sure what scripts you are referring to be dated, FusionPBX 
> scripts have been last modified this month 
> http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/insta
> ll_fusionpbx.sh
>
> 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> *
>
> Polycom Certified VAR
> Microsoft Small Business Specialist, Cisco SMB Select Certified
>
> 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>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Allison Walters
> *Sent:* Monday, December 1, 2014 8:51 PM
> *To:* freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Questions before starting new 
> install
>
> Whoops - accidentally had mail delivery disabled.  So I'll respond to Moishe's points with a manual reply/quote:
>
>> 1)      You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface.
>
> I didn't change anything - they were both already installed.  Mind 
> you, I don't know how the person who installed it did it; maybe that's 
> related to the issues they were seeing?  Regardless, the Fusion 
> interface seems to read all of the correct info, including when things 
> are changed in the XML files (which is how I was encouraged to do 
> changes.).  Now, I'm confused - Fusion keeps its own copy of the 
> config files? It's not just a wrapper that modifies the existing 
> freeswitch ones?  (Just going off what I was told.)
>
>> 2)      Ubuntu install is the same as debian  https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off
> with one of their scripts.
>
> I looked at their script, but it didn't seem to have been updated 
> since 2012, which makes me a bit nervous.  I'll take another peek, though.
>
>> 3)      If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from?
>
> The current server is running 1.3.0.
>
> Thanks!
>
> --
>
> *Allison Walters*
>
> /IT Coordinator/
>
> CAWST - Centre for Affordable Water and Sanitation Technology
>
> 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
>
> tel: +1-403-243-3285 ext.264
>
> web: www.cawst.org <http://www.cawst.org/>
>
> *//*
>
> */Join us in reaching 20 million people by 2020 with safe drinking 
> water and sanitation. /*
>
> From: *Allison Walters* <AWalters at cawst.org 
> <mailto:AWalters at cawst.org>>
> Date: Mon, Dec 1, 2014 at 3:50 PM
> Subject: [Freeswitch-users] Questions before starting new install
> To: "freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>"
> <freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>>
>
> We're wanting to replace our current freeswitch server (it's fairly 
> old), and I have a few things I'm curious about before I start trying 
> to build the new one.  I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience.
>
> 1.I was told by the consultant that set up the current box that 
> freeswitch doesn't play well with fusionpbx or really any of the GUI's 
> - there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion.
> Also I was told it doesn't provide nice clean XML, etc.  I was 
> strongly encouraged to go with no GUI.  If it were just me, I'd be 
> okay with that, but we have a couple of other people who might be helping to maintain the server and they're far more comfortable with GUIs.  Has anyone else had any of these issues, and/or have any recommendations on the subject?
>
> 2.I'm trying to install on Ubuntu.  When I go to the wiki, I get this page:
> https://freeswitch.org/confluence/display/FREESWITCH/Linux , which 
> just gives an apt-get line for Ubuntu.  However, that line just seems 
> to have pre-reqs and not freeswitch itself.  Anyone know where I could 
> find current Ubuntu instructions?  I googled but the most recent pages I could find were from 2012, and most were from 2011.  Is that the most up to date?
>
> 3.Are there any guidelines for migration from an old server running an old version to a new server with a current version?
>
> Thank you!
>
> --
>
> *Allison Walters*
>
> /IT Coordinator/
>
> CAWST - Centre for Affordable Water and Sanitation Technology
>
> 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
>
> tel: +1-403-243-3285 ext.264 <tel:%2B1-403-243-3285%20ext.264>
>
> web: www.cawst.org <http://www.cawst.org/>
>
> *//*
>
> */Join us in reaching 20 million people by 2020 with safe drinking 
> water and sanitation. /*
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org 
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>

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

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

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

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


From aademattia at comcast.net  Tue Dec  2 23:25:20 2014
From: aademattia at comcast.net (=?utf-8?B?YWFkZW1hdHRpYUBjb21jYXN0Lm5ldA==?=)
Date: Tue, 02 Dec 2014 15:25:20 -0500
Subject: [Freeswitch-users] =?utf-8?q?Send=5FDTMF?=
Message-ID: <mailman.2.1417551954.578.freeswitch-users@lists.freeswitch.org>

After looking at another sdk and wireshark it looks like the only difference is Marker:false not working but on a working dtmf it's true. Not sure if this means anything.

Sent from my HTC

----- Reply message -----
From: "aademattia at comcast.net" <aademattia at comcast.net>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users]Send_DTMF
Date: Tue, Dec 2, 2014 10:04 AM

I still see the dtmf going to the wrong destination.
I am sending dtmf on both uuid and can't seem to get the ivr I am calling to see my dtmf. But if I hit dtmf on my phone it works.

Sent from my HTC

----- Reply message -----
From: "Seven Du" <dujinfang at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Send_DTMF
Date: Tue, Dec 2, 2014 8:52 AM



try uuid_recv_dtmf



-- 
Seven Du
http://about.me/dujinfang
http://www.dujinfang.com
http://www.freeswitch.org.cn


Sent with Sparrow




On Tuesday, December 2, 2014 at 9:37 AM, Andrew wrote:
Hi,I am making an outbound call then I am doing a bridge.Once I bridge the call I need to send dtmf to second call (bridge call)When doing a trace it looks like the DTMF is going to my first call and not the bridge call.I am using UUID send dtmf using both UUID but both don?t seem to work. I do see the DTMF in wire shark but again looks like its going the wrong direction. Andrew 
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting at freeswitch..org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From AWalters at cawst.org  Tue Dec  2 23:34:40 2014
From: AWalters at cawst.org (Allison Walters)
Date: Tue, 2 Dec 2014 20:34:40 +0000
Subject: [Freeswitch-users] Questions before starting new install
In-Reply-To: <547E0965.6050909@mst.edu>
References: <20b779910393484881146823fb74258f@BN1PR0101MB0817.prod.exchangelabs.com>
	<CAAq2U1DtMBa2txADfgPw+bAMaSPBXCDT2gt9qBxB44abUwYPHQ@mail.gmail.com>
	<78cb8a4d983743bc94c0f63a732a1c10@BN1PR0101MB0817.prod.exchangelabs.com>
	<BA6A05D48E282C4C9989496521266BA10315792D03@ex2.connex.local>
	<fea24868db514600b1680df34b3fcdc3@BN1PR0101MB0817.prod.exchangelabs.com>
	<547E0965.6050909@mst.edu>
Message-ID: <c277acb277c54c199dfb2ef285f7ea71@BN1PR0101MB0817.prod.exchangelabs.com>

Okay, this half-asleep mistake rectified, I ran it on the right system and got further.  However, partway through the make it exited, complaining about various missing pre-reqs.  (The beginning of this script claims to check for prereqs but it only seems to check for 4 or so, and I've had to re-run a bunch of other times for lots it apparently doesn't check for).  Now after installing a bunch of the things it asked for, I ran into one that apparently I have to basically reset the entire installation process before it will recognize that it's there (libmemcached-dev).  I tried deleting the /usr/src/freeswitch dir, which made it re-fetch everything from git.  Sadly, it didn't force it to reconfigure, and now I'm getting an error " make: *** No targets specified and no makefile found.  Stop."

Anyone have any ideas how I can force the entire process to start over?  Thanks, and sorry for the questions - this is just getting frustrating.  (Removing my own bone-headed mistake from the pile, I've still been at this for 3 hours or so now...)


--
Allison Walters
IT Coordinator
CAWST - Centre for Affordable Water and Sanitation Technology
424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
tel: +1-403-243-3285 ext.264
web:?www.cawst.org
?
Join us in reaching 20 million people by 2020 with safe drinking water and sanitation.?


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nathan Neulinger
Sent: Tuesday, December 02, 2014 11:48 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Questions before starting new install

Do you have git installed on your system? Sure looks like it's complaining that you don't have it installed.

-- Nathan

On 12/02/2014 12:40 PM, Allison Walters wrote:
> I tried that version of the script and it says it was written for 
> 10.04 and 12.04, and it says it doesn't always work outside of that, but it'll "give it a go".  And then after a bit, it failed out with a git error:
>
> installing stable v1.4 of FreeSWITCH
>
> /usr/bin/time: cannot run /usr/bin/git: No such file or directory
>
> Command exited with non-zero status 127
>
> 0.00user 0.00system 0:00.00elapsed 0%CPU (0avgtext+0avgdata 
> 1220maxresident)k
>
> 0inputs+0outputs (0major+27minor)pagefaults 0swaps
>
> ./install_fusionpbx.sh: line 1114: cd: /usr/src/freeswitch: No such 
> file or directory
>
> ./install_fusionpbx.sh: line 1115: /usr/bin/git: No such file or 
> directory
>
> GIT ERROR
>
> I suspect this is because freeswitch moved to stash in July, 
> apparently.  Anyway, I'll see if I can find anyone on the Fusion forum or IRC or somewhere that can help with this part.  Thanks!
>
> --
>
> *Allison Walters*
>
> /IT Coordinator/
>
> CAWST - Centre for Affordable Water and Sanitation Technology
>
> 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
>
> tel: +1-403-243-3285 ext.264
>
> web: www.cawst.org <http://www.cawst.org/>
>
> *//*
>
> */Join us in reaching 20 million people by 2020 with safe drinking 
> water and sanitation. /*
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Moishe Grunstein
> *Sent:* Monday, December 01, 2014 9:37 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Questions before starting new 
> install
>
> 1)FusionPBX renames Freeswitch default configs after installation, it 
> uses and then creates its own XML files (unless you use their XML 
> handler which writes to db instead). If you then modify XML files 
> directly, the next time you make a change in the GUI they will overwrite your changes. They have an IRC channel #FusionPBX and they use googlecode for svn and issue tracker..
>
> 2)Not sure what scripts you are referring to be dated, FusionPBX 
> scripts have been last modified this month 
> http://fusionpbx.googlecode.com/svn/trunk/scripts/install/ubuntu/insta
> ll_fusionpbx.sh
>
> 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> *
>
> Polycom Certified VAR
> Microsoft Small Business Specialist, Cisco SMB Select Certified
>
> 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>
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Allison Walters
> *Sent:* Monday, December 1, 2014 8:51 PM
> *To:* freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Questions before starting new 
> install
>
> Whoops - accidentally had mail delivery disabled.  So I'll respond to Moishe's points with a manual reply/quote:
>
>> 1)      You cannot migrate a working freeswitch to FusionPBX. You would need to recreate all the configs in the FusionPBX web interface.
>
> I didn't change anything - they were both already installed.  Mind 
> you, I don't know how the person who installed it did it; maybe that's 
> related to the issues they were seeing?  Regardless, the Fusion 
> interface seems to read all of the correct info, including when things 
> are changed in the XML files (which is how I was encouraged to do 
> changes.).  Now, I'm confused - Fusion keeps its own copy of the 
> config files? It's not just a wrapper that modifies the existing 
> freeswitch ones?  (Just going off what I was told.)
>
>> 2)      Ubuntu install is the same as debian  https://freeswitch.org/confluence/display/FREESWITCH/Debian if you are going with fusionpbx you would be better off
> with one of their scripts.
>
> I looked at their script, but it didn't seem to have been updated 
> since 2012, which makes me a bit nervous.  I'll take another peek, though.
>
>> 3)      If you are staying with native freeswitch, you can copy over almost all configs to the new server. What version are you migrating from?
>
> The current server is running 1.3.0.
>
> Thanks!
>
> --
>
> *Allison Walters*
>
> /IT Coordinator/
>
> CAWST - Centre for Affordable Water and Sanitation Technology
>
> 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
>
> tel: +1-403-243-3285 ext.264
>
> web: www.cawst.org <http://www.cawst.org/>
>
> *//*
>
> */Join us in reaching 20 million people by 2020 with safe drinking 
> water and sanitation. /*
>
> From: *Allison Walters* <AWalters at cawst.org 
> <mailto:AWalters at cawst.org>>
> Date: Mon, Dec 1, 2014 at 3:50 PM
> Subject: [Freeswitch-users] Questions before starting new install
> To: "freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>"
> <freeswitch-users at lists.freeswitch.org 
> <mailto:freeswitch-users at lists.freeswitch.org>>
>
> We're wanting to replace our current freeswitch server (it's fairly 
> old), and I have a few things I'm curious about before I start trying 
> to build the new one.  I inherited the current one and am relatively unfamiliar to all of this, so I appreciate your patience.
>
> 1.I was told by the consultant that set up the current box that 
> freeswitch doesn't play well with fusionpbx or really any of the GUI's 
> - there have been instances (and indeed one happened a couple of months ago while I was here) of the config files getting randomly overwritten with old settings, which supposedly is the fault of trying to run fusion.
> Also I was told it doesn't provide nice clean XML, etc.  I was 
> strongly encouraged to go with no GUI.  If it were just me, I'd be 
> okay with that, but we have a couple of other people who might be helping to maintain the server and they're far more comfortable with GUIs.  Has anyone else had any of these issues, and/or have any recommendations on the subject?
>
> 2.I'm trying to install on Ubuntu.  When I go to the wiki, I get this page:
> https://freeswitch.org/confluence/display/FREESWITCH/Linux , which 
> just gives an apt-get line for Ubuntu.  However, that line just seems 
> to have pre-reqs and not freeswitch itself.  Anyone know where I could 
> find current Ubuntu instructions?  I googled but the most recent pages I could find were from 2012, and most were from 2011.  Is that the most up to date?
>
> 3.Are there any guidelines for migration from an old server running an old version to a new server with a current version?
>
> Thank you!
>
> --
>
> *Allison Walters*
>
> /IT Coordinator/
>
> CAWST - Centre for Affordable Water and Sanitation Technology
>
> 424 Aviation Road NE, Calgary, Alberta, T2E 8H6, Canada
>
> tel: +1-403-243-3285 ext.264 <tel:%2B1-403-243-3285%20ext.264>
>
> web: www.cawst.org <http://www.cawst.org/>
>
> *//*
>
> */Join us in reaching 20 million people by 2020 with safe drinking 
> water and sanitation. /*
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org 
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>

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

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

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

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


From ssinyagin at gmail.com  Wed Dec  3 02:30:18 2014
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Wed, 3 Dec 2014 00:30:18 +0100
Subject: [Freeswitch-users] UniFi phones and FreeSWITCH
In-Reply-To: <CAKzWOxW4DytBiJ3HY5ePuL9Y8VRS_y_P_hm0WvDo=4Q=35aAGg@mail.gmail.com>
References: <CAKzWOxW4DytBiJ3HY5ePuL9Y8VRS_y_P_hm0WvDo=4Q=35aAGg@mail.gmail.com>
Message-ID: <CACOYK=jFx56NxpA8ga550g4OUdQ9mXNwnx3xuaaHpXe+qVGf3A@mail.gmail.com>

Our local reseller expects them only by end of January. But I guess there
shouldn't be anything too bad in regards to compatibility :)
On Dec 2, 2014 4:59 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> Hello FreeSWITCHers!
>
> Does anybody have any experience with these phones?
> http://www.ubnt.com/unifi-voip/uvp/
>
> I'm curious to know how they work in a FS environment.
>
> -MC
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From AHadjiefstathiou at IronFX.com  Wed Dec  3 18:11:06 2014
From: AHadjiefstathiou at IronFX.com (Andri Hadjiefstathiou [IronFX Global])
Date: Wed, 3 Dec 2014 15:11:06 +0000
Subject: [Freeswitch-users] Freeswitch
Message-ID: <C442A00D7C61D34489F8DDC4A769C6729D836A0F@EXCH-DB-2.IronFX.Local>

I got those errors recently from our freeswitch server

[ERR] mod_commands.c:4063 Error connecting
[ERR] sofia_glue.c:6464 Error connecting
[ERR] sofia_glue.c:6504 Error Opening DB
[WARNING] sofia_glue.c:6464 Max handles 50

What does this means?

I have load recently the mod_xml_rpc is this relate with our issue?
Also how I can check the correct open DB open connections ?


Andri Hadjiefstathiou | Network and Security Engineer
t  +357 25027607 | m  +357 96860600 | e  AHadjiefstathiou at IronFX.com<mailto:AHadjiefstathiou at IronFX.com>
IronFX Global Limited
t  +357 25027000 | f  +357 25027001 | w  www.IronFX.com<http://www.IronFX.com>
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From AHadjiefstathiou at IronFX.com  Wed Dec  3 18:39:56 2014
From: AHadjiefstathiou at IronFX.com (Andri Hadjiefstathiou [IronFX Global])
Date: Wed, 3 Dec 2014 15:39:56 +0000
Subject: [Freeswitch-users] Freeswitch
In-Reply-To: <C442A00D7C61D34489F8DDC4A769C6729D836A0F@EXCH-DB-2.IronFX.Local>
References: <C442A00D7C61D34489F8DDC4A769C6729D836A0F@EXCH-DB-2.IronFX.Local>
Message-ID: <C442A00D7C61D34489F8DDC4A769C6729D836AEF@EXCH-DB-2.IronFX.Local>

I got those errors recently from our freeswitch server

2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6376 Error connecting
2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6416 Error Opening DB
2014-11-28 21:19:19.806910 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking....
2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6376 Error connecting
2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6416 Error Opening DB
2014-11-28 21:19:21.476882 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking....

What does this means?

I have load recently the mod_xml_rpc is this relate with our issue?
Also how I can check the correct open DB open connections ?

Andri Hadjiefstathiou | Network and Security Engineer
t  +357 25027607 | m  +357 96860600 | e  AHadjiefstathiou at IronFX.com<mailto:AHadjiefstathiou at IronFX.com>
IronFX Global Limited
t  +357 25027000 | f  +357 25027001 | w  www.IronFX.com<http://www.IronFX.com>
[cid:image001.jpg at 01D00F20.26E275D0]<https://www.facebook.com/IronFX> [cid:image002.gif at 01D00F20.26E275D0] <https://twitter.com/ironfxglobal>  [cid:image003.gif at 01D00F20.26E275D0] <http://www.youtube.com/ironfxmarketing>  [cid:image004.gif at 01D00F20.26E275D0] <http://blog.ironfx.com/>  [cid:image005.gif at 01D00F20.26E275D0] <http://www.linkedin.com/company/ironfxglobal>  [cid:image006.gif at 01D00F20.26E275D0] <https://plus.google.com/+Ironfxglobal>
London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol
[cid:image007.jpg at 01D00F20.26E275D0]
-----------------------------------------------------------------------------------------------------------------------
IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561)
IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482)
IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276)
IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966)
IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10)
IronFX Global Limited is a Member of Eurex Exchange
-----------------------------------------------------------------------------------------------------------------------
Consider the environment before printing this email.
From: Andri Hadjiefstathiou [IronFX Global]
Sent: Wednesday, December 03, 2014 5:11 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Freeswitch

I got those errors recently from our freeswitch server

[ERR] mod_commands.c:4063 Error connecting
[ERR] sofia_glue.c:6464 Error connecting
[ERR] sofia_glue.c:6504 Error Opening DB
[WARNING] sofia_glue.c:6464 Max handles 50

What does this means?

I have load recently the mod_xml_rpc is this relate with our issue?
Also how I can check the correct open DB open connections ?


Andri Hadjiefstathiou | Network and Security Engineer
t  +357 25027607 | m  +357 96860600 | e  AHadjiefstathiou at IronFX.com<mailto:AHadjiefstathiou at IronFX.com>
IronFX Global Limited
t  +357 25027000 | f  +357 25027001 | w  www.IronFX.com<http://www.IronFX.com>
[cid:image001.jpg at 01D00F20.26E275D0]<https://www.facebook.com/IronFX> [cid:image002.gif at 01D00F20.26E275D0] <https://twitter.com/ironfxglobal>  [cid:image003.gif at 01D00F20.26E275D0] <http://www.youtube.com/ironfxmarketing>  [cid:image004.gif at 01D00F20.26E275D0] <http://blog.ironfx.com/>  [cid:image005.gif at 01D00F20.26E275D0] <http://www.linkedin.com/company/ironfxglobal>  [cid:image006.gif at 01D00F20.26E275D0] <https://plus.google.com/+Ironfxglobal>
London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol
[cid:image007.jpg at 01D00F20.26E275D0]
-----------------------------------------------------------------------------------------------------------------------
IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561)
IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482)
IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276)
IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966)
IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10)
IronFX Global Limited is a Member of Eurex Exchange
-----------------------------------------------------------------------------------------------------------------------
This email has been sent from IronFX Global Limited. The information in this email is confidential and may be
legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not
the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is
prohibited and may be unlawful.
Consider the environment before printing this email.

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From cmrienzo at gmail.com  Wed Dec  3 18:59:51 2014
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 3 Dec 2014 10:59:51 -0500
Subject: [Freeswitch-users] Freeswitch
In-Reply-To: <C442A00D7C61D34489F8DDC4A769C6729D836AEF@EXCH-DB-2.IronFX.Local>
References: <C442A00D7C61D34489F8DDC4A769C6729D836A0F@EXCH-DB-2.IronFX.Local>
	<C442A00D7C61D34489F8DDC4A769C6729D836AEF@EXCH-DB-2.IronFX.Local>
Message-ID: <CAD-qRkiywzX3=ZJTdAM=YgUeh_jyzi24RRN6r1Y3iYp=gmZF9w@mail.gmail.com>

Check your sofia profile(s)- it has the database settings in there (dbname,
odbc-dsn, db-*) .  Then, verify your database is running and can accept
connections.


On Wed, Dec 3, 2014 at 10:39 AM, Andri Hadjiefstathiou [IronFX Global] <
AHadjiefstathiou at ironfx.com> wrote:

>  I got those errors recently from our freeswitch server
>
>
>
> 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6376 Error connecting
>
> 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6416 Error Opening DB
>
> 2014-11-28 21:19:19.806910 [WARNING] sofia_glue.c:6376 Max handles 50
> exceeded, blocking....
>
> 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6376 Error connecting
>
> 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6416 Error Opening DB
>
> 2014-11-28 21:19:21.476882 [WARNING] sofia_glue.c:6376 Max handles 50
> exceeded, blocking....
>
>
>
> What does this means?
>
>
>
> I have load recently the mod_xml_rpc is this relate with our issue?
>
> Also how I can check the correct open DB open connections ?
>
>
>
> *Andri* *Hadjiefstathiou* | Network and Security Engineer
> *t  *+357 25027607 | *m  *+357 96860600 | *e*  AHadjiefstathiou at IronFX.com
>
>
> *IronFX Global Limited t*  +357 25027000 | *f*  +357 25027001 | *w  *
> www.IronFX.com
>
>  <https://www.facebook.com/IronFX>  <https://twitter.com/ironfxglobal>
> <http://www.youtube.com/ironfxmarketing>  <http://blog.ironfx.com/>
> <http://www.linkedin.com/company/ironfxglobal>
> <https://plus.google.com/+Ironfxglobal>
>
> *London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt
> | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev
> | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh
> City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo |
> Buenos Aires | Lima | Santiago | Montevideo | Limassol*
>
>
> -----------------------------------------------------------------------------------------------------------------------
> IronFX Global UK Limited is authorised and regulated by the Financial
> Conduct Authority (FCA no. 585561)
> IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC
> (AFSL no. 417482)
> IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial
> Services Board (FSP No 45276)
> IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no.
> 298966)
> IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
> IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
> IronFX Global Limited is authorised and regulated by CySEC (Licence no.
> 125/10)
>
> IronFX Global Limited is a Member of Eurex Exchange
>
> -----------------------------------------------------------------------------------------------------------------------
>
> *Consider the environment before printing this email.*
>
> *From:* Andri Hadjiefstathiou [IronFX Global]
> *Sent:* Wednesday, December 03, 2014 5:11 PM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* Freeswitch
>
>
>
> I got those errors recently from our freeswitch server
>
>
>
> [ERR] mod_commands.c:4063 Error connecting
>
> [ERR] sofia_glue.c:6464 Error connecting
>
> [ERR] sofia_glue.c:6504 Error Opening DB
>
> [WARNING] sofia_glue.c:6464 Max handles 50
>
>
>
> What does this means?
>
>
>
> I have load recently the mod_xml_rpc is this relate with our issue?
>
> Also how I can check the correct open DB open connections ?
>
>
>
>
>
> *Andri* *Hadjiefstathiou* | Network and Security Engineer
> *t  *+357 25027607 | *m  *+357 96860600 | *e*  AHadjiefstathiou at IronFX.com
>
>
> *IronFX Global Limited t*  +357 25027000 | *f*  +357 25027001 | *w  *
> www.IronFX.com
>
> <https://www.facebook.com/IronFX>  <https://twitter.com/ironfxglobal>
> <http://www.youtube.com/ironfxmarketing>  <http://blog.ironfx.com/>
> <http://www.linkedin.com/company/ironfxglobal>
> <https://plus.google.com/+Ironfxglobal>
>
> *London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt
> | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev
> | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh
> City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo |
> Buenos Aires | Lima | Santiago | Montevideo | Limassol*
>
>
> -----------------------------------------------------------------------------------------------------------------------
> IronFX Global UK Limited is authorised and regulated by the Financial
> Conduct Authority (FCA no. 585561)
> IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC
> (AFSL no. 417482)
> IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial
> Services Board (FSP No 45276)
> IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no.
> 298966)
> IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
> IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
> IronFX Global Limited is authorised and regulated by CySEC (Licence no.
> 125/10)
>
> IronFX Global Limited is a Member of Eurex Exchange
>
> -----------------------------------------------------------------------------------------------------------------------
>
>
>
>
> *This email has been sent from IronFX Global Limited. The information in
> this email is confidential and may be legally privileged. It is intended
> solely for the addressee. Access to this email by anyone else is
> unauthorised. If you are not the intended recipient, any disclosure,
> copying, distribution or any action taken or omitted to be taken in
> reliance on it, is prohibited and may be unlawful.*
>
> *Consider the environment before printing this email.*
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From AHadjiefstathiou at IronFX.com  Wed Dec  3 19:03:36 2014
From: AHadjiefstathiou at IronFX.com (Andri Hadjiefstathiou [IronFX Global])
Date: Wed, 3 Dec 2014 16:03:36 +0000
Subject: [Freeswitch-users] Freeswitch
In-Reply-To: <CAD-qRkiywzX3=ZJTdAM=YgUeh_jyzi24RRN6r1Y3iYp=gmZF9w@mail.gmail.com>
References: <C442A00D7C61D34489F8DDC4A769C6729D836A0F@EXCH-DB-2.IronFX.Local>
	<C442A00D7C61D34489F8DDC4A769C6729D836AEF@EXCH-DB-2.IronFX.Local>
	<CAD-qRkiywzX3=ZJTdAM=YgUeh_jyzi24RRN6r1Y3iYp=gmZF9w@mail.gmail.com>
Message-ID: <C442A00D7C61D34489F8DDC4A769C6729D836BE7@EXCH-DB-2.IronFX.Local>

After we restart the server all fine.

But I want to know the root of the problem.. why we got this error???

Recently I have enable mod_xml_rpc is this relate with our issue?

Andri Hadjiefstathiou | Network and Security Engineer
t  +357 25027607 | m  +357 96860600 | e  AHadjiefstathiou at IronFX.com<mailto:AHadjiefstathiou at IronFX.com>
IronFX Global Limited
t  +357 25027000 | f  +357 25027001 | w  www.IronFX.com<http://www.IronFX.com>
[cid:image001.jpg at 01D00F23.751A8A50]<https://www.facebook.com/IronFX> [cid:image002.gif at 01D00F23.751A8A50] <https://twitter.com/ironfxglobal>  [cid:image003.gif at 01D00F23.751A8A50] <http://www.youtube.com/ironfxmarketing>  [cid:image004.gif at 01D00F23.751A8A50] <http://blog.ironfx.com/>  [cid:image005.gif at 01D00F23.751A8A50] <http://www.linkedin.com/company/ironfxglobal>  [cid:image006.gif at 01D00F23.751A8A50] <https://plus.google.com/+Ironfxglobal>
London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol
[cid:image007.jpg at 01D00F23.751A8A50]
-----------------------------------------------------------------------------------------------------------------------
IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561)
IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482)
IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276)
IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966)
IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10)
IronFX Global Limited is a Member of Eurex Exchange
-----------------------------------------------------------------------------------------------------------------------
Consider the environment before printing this email.
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christopher Rienzo
Sent: Wednesday, December 03, 2014 6:00 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch

Check your sofia profile(s)- it has the database settings in there (dbname, odbc-dsn, db-*) .  Then, verify your database is running and can accept connections.

On Wed, Dec 3, 2014 at 10:39 AM, Andri Hadjiefstathiou [IronFX Global] <AHadjiefstathiou at ironfx.com<mailto:AHadjiefstathiou at ironfx.com>> wrote:
I got those errors recently from our freeswitch server

2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6376 Error connecting
2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6416 Error Opening DB
2014-11-28 21:19:19.806910 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking....
2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6376 Error connecting
2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6416 Error Opening DB
2014-11-28 21:19:21.476882 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking....

What does this means?

I have load recently the mod_xml_rpc is this relate with our issue?
Also how I can check the correct open DB open connections ?

Andri Hadjiefstathiou | Network and Security Engineer
t  +357 25027607<tel:%2B357%2025027607> | m  +357 96860600<tel:%2B357%2096860600> | e  AHadjiefstathiou at IronFX.com<mailto:AHadjiefstathiou at IronFX.com>
IronFX Global Limited
t  +357 25027000<tel:%2B357%2025027000> | f  +357 25027001<tel:%2B357%2025027001> | w  www.IronFX.com<http://www.IronFX.com>
       <https://www.facebook.com/IronFX>
London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol
-----------------------------------------------------------------------------------------------------------------------
IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561)
IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482)
IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276)
IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966)
IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10)
IronFX Global Limited is a Member of Eurex Exchange
-----------------------------------------------------------------------------------------------------------------------
Consider the environment before printing this email.
From: Andri Hadjiefstathiou [IronFX Global]
Sent: Wednesday, December 03, 2014 5:11 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Freeswitch

I got those errors recently from our freeswitch server

[ERR] mod_commands.c:4063 Error connecting
[ERR] sofia_glue.c:6464 Error connecting
[ERR] sofia_glue.c:6504 Error Opening DB
[WARNING] sofia_glue.c:6464 Max handles 50

What does this means?

I have load recently the mod_xml_rpc is this relate with our issue?
Also how I can check the correct open DB open connections ?


Andri Hadjiefstathiou | Network and Security Engineer
t  +357 25027607 | m  +357 96860600 | e  AHadjiefstathiou at IronFX.com
IronFX Global Limited
t  +357 25027000 | f  +357 25027001 | w  www.IronFX.com

London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol
-----------------------------------------------------------------------------------------------------------------------
IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561)
IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482)
IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276)
IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966)
IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10)
IronFX Global Limited is a Member of Eurex Exchange
-----------------------------------------------------------------------------------------------------------------------
This email has been sent from IronFX Global Limited. The information in this email is confidential and may be
legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not
the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is
prohibited and may be unlawful.
Consider the environment before printing this email.


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

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

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

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From cmrienzo at gmail.com  Wed Dec  3 19:18:01 2014
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 3 Dec 2014 11:18:01 -0500
Subject: [Freeswitch-users] Freeswitch
In-Reply-To: <C442A00D7C61D34489F8DDC4A769C6729D836BE7@EXCH-DB-2.IronFX.Local>
References: <C442A00D7C61D34489F8DDC4A769C6729D836A0F@EXCH-DB-2.IronFX.Local>
	<C442A00D7C61D34489F8DDC4A769C6729D836AEF@EXCH-DB-2.IronFX.Local>
	<CAD-qRkiywzX3=ZJTdAM=YgUeh_jyzi24RRN6r1Y3iYp=gmZF9w@mail.gmail.com>
	<C442A00D7C61D34489F8DDC4A769C6729D836BE7@EXCH-DB-2.IronFX.Local>
Message-ID: <CAD-qRkhAUSwEkZuJ0fpKacmhAev56k76gkjXh4zNehMLA3NPKQ@mail.gmail.com>

I don't have any ideas for this particular issue other than general db
connection troubleshooting tips.

Are you using SQLite or odbc w/ mysql or postgresql?  Are database handles
being released?  Is a new long / inefficient query consuming all your
connections?  Does the issue go away if you disable mod_xml_rpc?  Find out
these things to troubleshoot your issue.

Also- I noticed you are running an old version of FS based on the line
numbers in your logs, so your likelihood of getting free support is not
good.

On Wed, Dec 3, 2014 at 11:03 AM, Andri Hadjiefstathiou [IronFX Global] <
AHadjiefstathiou at ironfx.com> wrote:

>  After we restart the server all fine.
>
>
>
> But I want to know the root of the problem.. why we got this error???
>
>
>
> Recently I have enable mod_xml_rpc is this relate with our issue?
>
>
>
> *Andri* *Hadjiefstathiou* | Network and Security Engineer
> *t  *+357 25027607 | *m  *+357 96860600 | *e*  AHadjiefstathiou at IronFX.com
>
>
> *IronFX Global Limited t*  +357 25027000 | *f*  +357 25027001 | *w  *
> www.IronFX.com
>
>  <https://www.facebook.com/IronFX>  <https://twitter.com/ironfxglobal>
> <http://www.youtube.com/ironfxmarketing>  <http://blog.ironfx.com/>
> <http://www.linkedin.com/company/ironfxglobal>
> <https://plus.google.com/+Ironfxglobal>
>
> *London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt
> | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev
> | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh
> City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo |
> Buenos Aires | Lima | Santiago | Montevideo | Limassol*
>
>
> -----------------------------------------------------------------------------------------------------------------------
> IronFX Global UK Limited is authorised and regulated by the Financial
> Conduct Authority (FCA no. 585561)
> IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC
> (AFSL no. 417482)
> IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial
> Services Board (FSP No 45276)
> IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no.
> 298966)
> IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
> IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
> IronFX Global Limited is authorised and regulated by CySEC (Licence no.
> 125/10)
>
> IronFX Global Limited is a Member of Eurex Exchange
>
> -----------------------------------------------------------------------------------------------------------------------
>
> *Consider the environment before printing this email.*
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Christopher
> Rienzo
> *Sent:* Wednesday, December 03, 2014 6:00 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Freeswitch
>
>
>
> Check your sofia profile(s)- it has the database settings in there
> (dbname, odbc-dsn, db-*) .  Then, verify your database is running and can
> accept connections.
>
>
>
> On Wed, Dec 3, 2014 at 10:39 AM, Andri Hadjiefstathiou [IronFX Global] <
> AHadjiefstathiou at ironfx.com> wrote:
>
>  I got those errors recently from our freeswitch server
>
>
>
> 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6376 Error connecting
>
> 2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6416 Error Opening DB
>
> 2014-11-28 21:19:19.806910 [WARNING] sofia_glue.c:6376 Max handles 50
> exceeded, blocking....
>
> 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6376 Error connecting
>
> 2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6416 Error Opening DB
>
> 2014-11-28 21:19:21.476882 [WARNING] sofia_glue.c:6376 Max handles 50
> exceeded, blocking....
>
>
>
> What does this means?
>
>
>
> I have load recently the mod_xml_rpc is this relate with our issue?
>
> Also how I can check the correct open DB open connections ?
>
>
>
> *Andri* *Hadjiefstathiou* | Network and Security Engineer
> *t  *+357 25027607 | * m  *+357 96860600 | *e*
> AHadjiefstathiou at IronFX.com
>
>
> *IronFX Global Limited t*  +357 25027000 | *f*  +357 25027001 | *w  *
> www.IronFX.com
>
>         <https://www.facebook.com/IronFX>
>
> *London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt
> | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev
> | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh
> City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo |
> Buenos Aires | Lima | Santiago | Montevideo | Limassol*
>
>
> -----------------------------------------------------------------------------------------------------------------------
>
> IronFX Global UK Limited is authorised and regulated by the Financial
> Conduct Authority (FCA no. 585561)
> IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC
> (AFSL no. 417482)
> IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial
> Services Board (FSP No 45276)
> IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no.
> 298966)
> IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
> IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
> IronFX Global Limited is authorised and regulated by CySEC (Licence no.
> 125/10)
>
> IronFX Global Limited is a Member of Eurex Exchange
>
> -----------------------------------------------------------------------------------------------------------------------
>
> *Consider the environment before printing this email.*
>
> *From:* Andri Hadjiefstathiou [IronFX Global]
> *Sent:* Wednesday, December 03, 2014 5:11 PM
> *To:* '*freeswitch-users at lists.freeswitch.org
> <freeswitch-users at lists.freeswitch.org>*'
> *Subject:* Freeswitch
>
>
>
> I got those errors recently from our freeswitch server
>
>
>
> [ERR] mod_commands.c:4063 Error connecting
>
> [ERR] sofia_glue.c:6464 Error connecting
>
> [ERR] sofia_glue.c:6504 Error Opening DB
>
> [WARNING] sofia_glue.c:6464 Max handles 50
>
>
>
> What does this means?
>
>
>
> I have load recently the mod_xml_rpc is this relate with our issue?
>
> Also how I can check the correct open DB open connections ?
>
>
>
>
>
> *Andri* *Hadjiefstathiou* | Network and Security Engineer
> *t  **+357 25027607* | *m  **+357 96860600* | *e*
> *AHadjiefstathiou at IronFX.com*
>
>
> *IronFX Global Limited t*  *+357 25027000* | *f*  *+357 25027001* | *w  **www.IronFX.com
> <http://www.IronFX.com>*
>
>
>
> *London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt
> | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev
> | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh
> City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo |
> Buenos Aires | Lima | Santiago | Montevideo | Limassol*
>
>
> -----------------------------------------------------------------------------------------------------------------------
>
> IronFX Global UK Limited is authorised and regulated by the Financial
> Conduct Authority (FCA no. 585561)
> IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC
> (AFSL no. 417482)
> IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial
> Services Board (FSP No 45276)
> IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no.
> 298966)
> IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
> IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
> IronFX Global Limited is authorised and regulated by CySEC (Licence no.
> 125/10)
>
> IronFX Global Limited is a Member of Eurex Exchange
>
> -----------------------------------------------------------------------------------------------------------------------
>
>
>
>
> *This email has been sent from IronFX Global Limited. The information in
> this email is confidential and may be legally privileged. It is intended
> solely for the addressee. Access to this email by anyone else is
> unauthorised. If you are not the intended recipient, any disclosure,
> copying, distribution or any action taken or omitted to be taken in
> reliance on it, is prohibited and may be unlawful.*
>
> *Consider the environment before printing this email.*
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> *consulting at freeswitch.org <consulting at freeswitch.org>*
> *http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com>*
>
> Official FreeSWITCH Sites
> *http://www.freeswitch.org <http://www.freeswitch.org>*
> *http://confluence.freeswitch.org <http://confluence.freeswitch.org>*
> *http://www.cluecon.com <http://www.cluecon.com>*
>
> FreeSWITCH-users mailing list
> *FreeSWITCH-users at lists.freeswitch.org
> <FreeSWITCH-users at lists.freeswitch.org>*
> *http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>*
> UNSUBSCRIBE:*http://lists.freeswitch.org/mailman/options/freeswitch-users
> <http://lists.freeswitch.org/mailman/options/freeswitch-users>*
> *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 AHadjiefstathiou at IronFX.com  Wed Dec  3 19:48:45 2014
From: AHadjiefstathiou at IronFX.com (Andri Hadjiefstathiou [IronFX Global])
Date: Wed, 3 Dec 2014 16:48:45 +0000
Subject: [Freeswitch-users] Freeswitch
In-Reply-To: <CAD-qRkhAUSwEkZuJ0fpKacmhAev56k76gkjXh4zNehMLA3NPKQ@mail.gmail.com>
References: <C442A00D7C61D34489F8DDC4A769C6729D836A0F@EXCH-DB-2.IronFX.Local>
	<C442A00D7C61D34489F8DDC4A769C6729D836AEF@EXCH-DB-2.IronFX.Local>
	<CAD-qRkiywzX3=ZJTdAM=YgUeh_jyzi24RRN6r1Y3iYp=gmZF9w@mail.gmail.com>
	<C442A00D7C61D34489F8DDC4A769C6729D836BE7@EXCH-DB-2.IronFX.Local>
	<CAD-qRkhAUSwEkZuJ0fpKacmhAev56k76gkjXh4zNehMLA3NPKQ@mail.gmail.com>
Message-ID: <C442A00D7C61D34489F8DDC4A769C6729D836D51@EXCH-DB-2.IronFX.Local>

We have mysql.

I have enable mod_xml_rpc and after 3 days the issue appeared. The issue go away when I restart the freeswitch .. but to be sure I have disable mod_xml_rpc.

How I can check the below:

Are database handles being released?  Is a new long / inefficient query consuming all your connections?

What do you mean about the below and what do you suggest?

Also- I noticed you are running an old version of FS based on the line numbers in your logs, so your likelihood of getting free support is not good.

Thanks

Andri Hadjiefstathiou | Network and Security Engineer
t  +357 25027607 | m  +357 96860600 | e  AHadjiefstathiou at IronFX.com<mailto:AHadjiefstathiou at IronFX.com>
IronFX Global Limited
t  +357 25027000 | f  +357 25027001 | w  www.IronFX.com<http://www.IronFX.com>
[cid:image001.jpg at 01D00F29.25424440]<https://www.facebook.com/IronFX> [cid:image002.gif at 01D00F29.25424440] <https://twitter.com/ironfxglobal>  [cid:image003.gif at 01D00F29.25424440] <http://www.youtube.com/ironfxmarketing>  [cid:image004.gif at 01D00F29.25424440] <http://blog.ironfx.com/>  [cid:image005.gif at 01D00F29.25424440] <http://www.linkedin.com/company/ironfxglobal>  [cid:image006.gif at 01D00F29.25424440] <https://plus.google.com/+Ironfxglobal>
London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol
[cid:image007.jpg at 01D00F29.25424440]
-----------------------------------------------------------------------------------------------------------------------
IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561)
IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482)
IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276)
IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966)
IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10)
IronFX Global Limited is a Member of Eurex Exchange
-----------------------------------------------------------------------------------------------------------------------
Consider the environment before printing this email.
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christopher Rienzo
Sent: Wednesday, December 03, 2014 6:18 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch

I don't have any ideas for this particular issue other than general db connection troubleshooting tips.

Are you using SQLite or odbc w/ mysql or postgresql?  Are database handles being released?  Is a new long / inefficient query consuming all your connections?  Does the issue go away if you disable mod_xml_rpc?  Find out these things to troubleshoot your issue.
Also- I noticed you are running an old version of FS based on the line numbers in your logs, so your likelihood of getting free support is not good.

On Wed, Dec 3, 2014 at 11:03 AM, Andri Hadjiefstathiou [IronFX Global] <AHadjiefstathiou at ironfx.com<mailto:AHadjiefstathiou at ironfx.com>> wrote:
After we restart the server all fine.

But I want to know the root of the problem.. why we got this error???

Recently I have enable mod_xml_rpc is this relate with our issue?

Andri Hadjiefstathiou | Network and Security Engineer
t  +357 25027607<tel:%2B357%2025027607> | m  +357 96860600<tel:%2B357%2096860600> | e  AHadjiefstathiou at IronFX.com<mailto:AHadjiefstathiou at IronFX.com>
IronFX Global Limited
t  +357 25027000<tel:%2B357%2025027000> | f  +357 25027001<tel:%2B357%2025027001> | w  www.IronFX.com<http://www.IronFX.com>
       <https://www.facebook.com/IronFX>
London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol
-----------------------------------------------------------------------------------------------------------------------
IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561)
IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482)
IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276)
IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966)
IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10)
IronFX Global Limited is a Member of Eurex Exchange
-----------------------------------------------------------------------------------------------------------------------
Consider the environment before printing this email.
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christopher Rienzo
Sent: Wednesday, December 03, 2014 6:00 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch

Check your sofia profile(s)- it has the database settings in there (dbname, odbc-dsn, db-*) .  Then, verify your database is running and can accept connections.

On Wed, Dec 3, 2014 at 10:39 AM, Andri Hadjiefstathiou [IronFX Global] <AHadjiefstathiou at ironfx.com> wrote:
I got those errors recently from our freeswitch server

2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6376 Error connecting
2014-11-28 21:19:19.806910 [ERR] sofia_glue.c:6416 Error Opening DB
2014-11-28 21:19:19.806910 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking....
2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6376 Error connecting
2014-11-28 21:19:21.476882 [ERR] sofia_glue.c:6416 Error Opening DB
2014-11-28 21:19:21.476882 [WARNING] sofia_glue.c:6376 Max handles 50 exceeded, blocking....

What does this means?

I have load recently the mod_xml_rpc is this relate with our issue?
Also how I can check the correct open DB open connections ?

Andri Hadjiefstathiou | Network and Security Engineer
t  +357 25027607 | m  +357 96860600 | e  AHadjiefstathiou at IronFX.com
IronFX Global Limited
t  +357 25027000 | f  +357 25027001 | w  www.IronFX.com

London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol
-----------------------------------------------------------------------------------------------------------------------

IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561)
IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482)
IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276)
IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966)
IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10)
IronFX Global Limited is a Member of Eurex Exchange
-----------------------------------------------------------------------------------------------------------------------
Consider the environment before printing this email.
From: Andri Hadjiefstathiou [IronFX Global]
Sent: Wednesday, December 03, 2014 5:11 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Freeswitch

I got those errors recently from our freeswitch server

[ERR] mod_commands.c:4063 Error connecting
[ERR] sofia_glue.c:6464 Error connecting
[ERR] sofia_glue.c:6504 Error Opening DB
[WARNING] sofia_glue.c:6464 Max handles 50

What does this means?

I have load recently the mod_xml_rpc is this relate with our issue?
Also how I can check the correct open DB open connections ?


Andri Hadjiefstathiou | Network and Security Engineer
t  +357 25027607 | m  +357 96860600 | e  AHadjiefstathiou at IronFX.com
IronFX Global Limited
t  +357 25027000 | f  +357 25027001 | w  www.IronFX.com

London | Hong Kong | Sydney | Shanghai | Shenzhen | Shenyang | Frankfurt | Madrid | Warsaw | Lisbon | Prague | Budapest | Bucharest | Athens | Kiev | Minsk | Moscow | St Petersburg | Mumbai | Jakarta | Manila | Ho Chi Minh City | Kuala Lumpur | Baku | Lagos | Johannesburg | Auckland | Sao Paulo | Buenos Aires | Lima | Santiago | Montevideo | Limassol
-----------------------------------------------------------------------------------------------------------------------

IronFX Global UK Limited is authorised and regulated by the Financial Conduct Authority (FCA no. 585561)
IronFX Global (Australia) Pty Limited is authorised and regulated by ASIC (AFSL no. 417482)
IronFX Global (South Africa) (Pty) Ltd is authorized by the Financial Services Board (FSP No 45276)
IronFX Global NZ Limited is authorised and regulated by FSP (FSPR no. 298966)
IronFX Global (Russia) LLC is a member of CRFIN (Membership no. A-8)
IronFX Global (Ukraine) LLC is a member of UCRFIN (Membership no. 5)
IronFX Global Limited is authorised and regulated by CySEC (Licence no. 125/10)
IronFX Global Limited is a Member of Eurex Exchange
-----------------------------------------------------------------------------------------------------------------------
This email has been sent from IronFX Global Limited. The information in this email is confidential and may be
legally privileged. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not
the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is
prohibited and may be unlawful.
Consider the environment before printing this email.


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

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

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


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

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

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

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From ahabiba at gmail.com  Wed Dec  3 21:36:56 2014
From: ahabiba at gmail.com (Ahmed Habiba)
Date: Wed, 3 Dec 2014 21:36:56 +0300
Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive calls
In-Reply-To: <CAGDxTYThLaBEyrO-3R9tYdSVXNPLF+M0yDL3DJb379=tQmuRDQ@mail.gmail.com>
References: <CAGDxTYThLaBEyrO-3R9tYdSVXNPLF+M0yDL3DJb379=tQmuRDQ@mail.gmail.com>
Message-ID: <1662947A-0E07-448C-9BF9-8AF2F99D3FED@gmail.com>

Dear Brian,

I?m using "zoiper 1.19.7? on android, the issue appears only one I enable SRTP, for TLS only it is working very fine.

The issue as shown below it keeps registering till finally zipper restart on mobile.

One more point I?m configuring Zoiper without using RPORT for either SRTP or TLS.

?2014-12-01 20:57:24.003230 [NOTICE] switch_ivr_originate.c:3467 Hangup sofia/internal/sip:1000 at 222.248.102.244 <mailto:1000 at 222.248.102.244>:38614 [CS_CONSUME_MEDIA] [NO_ANSWER]"



recv 1290 bytes from tls/[222.215.195.234]:2896 at 20:56:54.344450:
   ------------------------------------------------------------------------
   INVITE  <>sip:1000 at abc-xyz.com <sip:1000 at abc-xyz.com> SIP/2.0
   v: SIP/2.0/TLS 192.168.8.109:41659;rport;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias
   Max-Forwards: 70
   f: < <>sip:1006 at abc-xyz.com <sip:1006 at abc-xyz.com>>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU
   t: < <>sip:1000 at abc-xyz.com <sip:1000 at abc-xyz.com>>
   m: < <>sip:1006 at 222.215.195.234:2896;transport=TLS;ob <sip:1006 at 222.215.195.234:2896;transport=TLS;ob>>
   i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX
   CSeq: 32737 INVITE
   Route: < <>sip:abc-xyz.com;transport=tls;lr <sip:abc-xyz.com;transport=tls;lr>>
   k: replaces, 100rel, timer, norefersub
   x: 1800
   Min-SE: 90
   User-Agent: CSipSimple_GT-S5830-10/r2450
   Proxy-Authorization: Digest username="1006", realm="abc-xyz.com <http://abc-xyz.com/>", nonce="329456ee-7994-11e4-bf79-1182ebca2429", uri=" <>sip:1000 at abc-xyz.com <sip:1000 at abc-xyz.com>", response="ab3a0f2fe9207ced62eebd8c9b8c32b4", algorithm=MD5, cnonce="iLa-C.wpnBSeTQjWiXZK3H3PGYhi6EGt", qop=auth, nc=00000001
   c: application/sdp
   l:   469
   
   v=0
   o=- 3626452487 3626452487 IN IP4 192.168.8.109
   s=pjmedia
   c=IN IP4 192.168.8.109
   t=0 0
   m=audio 4000 RTP/SAVP 98 0 8 101
   c=IN IP4 192.168.8.109
   a=rtcp:4001 IN IP4 192.168.8.109
   a=sendrecv
   a=rtpmap:98 SILK/16000
   a=fmtp:98 useinbandfec=0
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Vci7cNdSFTkkLnqz+qRkqCctPvRT6jIOrNc5KMbz
   a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:nW3ECq5YHHnrY2q5u7XZE4fvPfMOAWKQ+ehOp7zV
   ------------------------------------------------------------------------
   
   send 405 bytes to tls/[222.215.195.234]:2896 at 20:56:54.344687:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/TLS 192.168.8.109:41659;rport=2896;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias;received=222.215.195.234
   f: < <>sip:1006 at abc-xyz.com <sip:1006 at abc-xyz.com>>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU
   t: < <>sip:1000 at abc-xyz.com <sip:1000 at abc-xyz.com>>
   i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX
   CSeq: 32737 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Content-Length: 0
   





send 2133 bytes to tls/[222.248.102.244]:38614 at 20:56:54.360608:
   ------------------------------------------------------------------------
   INVITE  <>sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS <sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS> SIP/2.0
   Via: SIP/2.0/TLS 176.31.245.17;branch=z9hG4bKjXZvg3cX76Krj
   Max-Forwards: 69
   From: "Extension 1006" < <>sip:1006 at 176.31.245.17 <sip:1006 at 176.31.245.17>>;tag=15DcKN9tUZeaK
   To: < <>sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS <sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS>>
   Call-ID: 0a3e07c5-f437-1232-9bba-d2ab2784dc6a
   CSeq: 68395843 INVITE
   Contact: < <>sip:mod_sofia at 176.31.245.17:5061;transport=tls <sip:mod_sofia at 176.31.245.17:5061;transport=tls>>
   User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   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: 1076
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "Extension 1006" < <>sip:1006 at 176.31.245.17 <sip:1006 at 176.31.245.17>>;party=calling;screen=yes;privacy=off
   
   v=0
   o=FreeSWITCH 1417445672 1417445673 IN IP4 176.31.245.17
   s=FreeSWITCH
   c=IN IP4 176.31.245.17
   t=0 0
   m=audio 18142 RTP/SAVP 0 8 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=crypto:1 AEAD_AES_256_GCM_8 inline:6p/uXB7tN5SG9vHP/DMurij6BaZIOKcP4Tle0kjo2lwL1Oa4z9WGyZtk4zs
   a=crypto:2 AEAD_AES_128_GCM_8 inline:hGeJA6XQV1V35oTu2141o7mD64dFDRbeCbcf6g
   a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:XdZwwcl6kMRrWY7Vkrbozf/o1awka9lef0UjQRwvnOnnrbAgzXIm8s3yu7o8DQ
   a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:KtM92m0ac2musYPoXLDJb4rUbEMYAwMegPH9RMtzwzPb+0O93e0
   a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:PtOpdIk2LTSCSDVzPaSN6AV2xosBrJUTd4mvKIGh
   a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:ZHzWxajRlQp8IuzX5CuFM1SRkL0huC62ukX583+Vg0LBiK9t2NrpP8FNwdkckQ
   a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:S9BWrDxGFYpU3P9o3HD0Z97W5jwo8vD+E1sJi821r4/B+cdukTw
   a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:l7C0wU29AjF2ZMnCTvks+1ytd/UiXHqN8UIsM6vf
   a=crypto:9 AES_CM_128_NULL_AUTH inline:tv97QsO7+S9NLTXrD02omyWoyM7I2KaWaimUgq99
   a=ptime:20
   ------------------------------------------------------------------------
   
   
      REGISTER  <>sip:abc-xyz.com;transport=TLS <sip:abc-xyz.com;transport=TLS> SIP/2.0
   Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport
   Max-Forwards: 70
   Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS>>
   To: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>
   From: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=4a2eae22
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 1 REGISTER
   Expires: 60
   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
   Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
   User-Agent: Zoiper r27147
   Allow-Events: presence, kpml
   Content-Length: 0
   
   
      ------------------------------------------------------------------------
send 716 bytes to tls/[222.248.102.244]:38620 at 20:57:01.806433:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport=38620;received=222.248.102.244
   From: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=4a2eae22
   To: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=2e74mgtyr84ve
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 1 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   WWW-Authenticate: Digest realm="abc-xyz.com <http://abc-xyz.com/>", nonce="374fdd98-7994-11e4-bf9c-1182ebca2429", algorithm=MD5, qop="auth"
   Content-Length: 0
   
   
      ------------------------------------------------------------------------
recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:01.995662:
   ------------------------------------------------------------------------
   REGISTER  <>sip:abc-xyz.com;transport=TLS <sip:abc-xyz.com;transport=TLS> SIP/2.0
   Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport
   Max-Forwards: 70
   Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS>>
   To: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>
   From: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=4a2eae22
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 2 REGISTER
   Expires: 60
   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
   Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
   User-Agent: Zoiper r27147
   Authorization: Digest username="1000",realm="abc-xyz.com <http://abc-xyz.com/>",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS <sip:abc-xyz.com;transport=TLS>",response="8d7e108ff98035d37424c32166fe0253",cnonce="107cf281c3b57888a4bfd1a9c3776098",nc=00000001,qop=auth,algorithm=MD5
   Allow-Events: presence, kpml
   Content-Length: 0
   
   
      ------------------------------------------------------------------------
send 712 bytes to tls/[222.248.102.244]:38620 at 20:57:01.997931:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport=38620;received=222.248.102.244
   From: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=4a2eae22
   To: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=3Q0XpBB2NHUFa
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 2 REGISTER
   Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS>>;expires=60
   Date: Mon, 01 Dec 2014 19:57:01 GMT
   User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:02.205719:
   ------------------------------------------------------------------------
   REGISTER  <>sip:abc-xyz.com;transport=TLS <sip:abc-xyz.com;transport=TLS> SIP/2.0
   Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport
   Max-Forwards: 70
   Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS>>;expires=0
   To: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>
   From: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=4a2eae22
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 3 REGISTER
   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
   Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
   User-Agent: Zoiper r27147
   Authorization: Digest username="1000",realm="abc-xyz.com <http://abc-xyz.com/>",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS <sip:abc-xyz.com;transport=TLS>",response="5168bf9afffe9adacef7d898bd6bec9b",cnonce="80108fa519303990cfd2d46639d767ee",nc=00000002,qop=auth,algorithm=MD5
   Allow-Events: presence, kpml
   Content-Length: 0
   
      ------------------------------------------------------------------------
send 598 bytes to tls/[222.248.102.244]:38620 at 20:57:02.207249:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport=38620
   From: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=4a2eae22
   To: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=40Spr6U5jtH2N
   Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
   CSeq: 3 REGISTER
   Date: Mon, 01 Dec 2014 19:57:02 GMT
   User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 681 bytes from tls/[222.248.102.244]:38620 at 20:57:02.400564:
   ------------------------------------------------------------------------
   REGISTER  <>sip:abc-xyz.com;transport=TLS <sip:abc-xyz.com;transport=TLS> SIP/2.0
   Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport
   Max-Forwards: 70
   Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS <sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS>>
   To: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>
   From: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=979e5f46
   Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
   CSeq: 1 REGISTER
   Expires: 60
   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
   Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
   User-Agent: Zoiper r27147
   Allow-Events: presence, kpml
   Content-Length: 0
   
   
      ------------------------------------------------------------------------
send 694 bytes to tls/[222.248.102.244]:38620 at 20:57:02.401167:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport=38620
   From: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=979e5f46
   To: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=59jFt1c9F37mH
   Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
   CSeq: 1 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   WWW-Authenticate: Digest realm="abc-xyz.com <http://abc-xyz.com/>", nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429", algorithm=MD5, qop="auth"
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 950 bytes from tls/[222.248.102.244]:38620 at 20:57:02.655864:
   ------------------------------------------------------------------------
   REGISTER  <>sip:abc-xyz.com;transport=TLS <sip:abc-xyz.com;transport=TLS> SIP/2.0
   Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport
   Max-Forwards: 70
   Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS <sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS>>
   To: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>
   From: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=979e5f46
   Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
   CSeq: 2 REGISTER
   Expires: 60
   Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
   Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
   User-Agent: Zoiper r27147
   Authorization: Digest username="1000",realm="abc-xyz.com <http://abc-xyz.com/>",nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS <sip:abc-xyz.com;transport=TLS>",response="f1667094626fc10ca76a070438497e6f",cnonce="dc790bffe285311f0b903bd515a3e741",nc=00000001,qop=auth,algorithm=MD5
   Allow-Events: presence, kpml
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 693 bytes to tls/[222.248.102.244]:38620 at 20:57:02.657688:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport=38620
   From: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=979e5f46
   To: < <>sip:1000 at abc-xyz.com;transport=TLS <sip:1000 at abc-xyz.com;transport=TLS>>;tag=6jc8UvXcDcy7c
   Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
   CSeq: 2 REGISTER
   Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS <sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS>>;expires=60
   Date: Mon, 01 Dec 2014 19:57:02 GMT
   User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, path, replaces
   Content-Length: 0




> 
> 
> From: Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Date: December 1, 2014 at 5:19:30 PM GMT+3
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
> Subject: Re: [Freeswitch-users] FreeSWITCH TLS not able to receive calls
> 
> 
> What endpoints are involved?  have you looked at 'sofia loglevel all 9' output and see if it gives you a clue?
> 
> On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba <ahabiba at gmail.com <mailto:ahabiba at gmail.com>> wrote:
> Dears,
> 
> I?ve configured FreeSWITCH with the below version with TLS/SRTPas per the recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS <https://wiki.freeswitch.org/wiki/SIP_TLS>? and it was strait forward, and I was able to connect and make make calls using zoiper, but I was not able to receive any calls after enabling the TLS/SRTP.
> 
> "FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)?
> 
> Your kind feedback will be appreciate.
> 
> Thanks,
> 
> Ahmed Habiba.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
> T:+19184209001 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype: <skype:briankwest>briankwest <skype:briankwest>
> 




-- 
Thanks and Best Regards,
 
Ahmed Habiba
 
Mob: +20 10 37 82 970
 
Success: believe (Vision) plus commitment (Action)
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From mike at jerris.com  Wed Dec  3 23:31:32 2014
From: mike at jerris.com (Michael Jerris)
Date: Wed, 3 Dec 2014 15:31:32 -0500
Subject: [Freeswitch-users] Windows build system
Message-ID: <E706BBBE-829E-4A71-A299-5C61C5011CF0@jerris.com>

Given the recent announcements by Microsoft about the community edition 2013 being available, we are working to migrate the build system towards using that as our primary build.  As part of this process we will be very soon dropping support for any version of Visual Studio prior to 2012.  If you feel strongly about needing support for these older versions, please speak up now with an offer to maintain these legacy build systems.  We are also investigating moving to using chocolatey as a new system to manage dependencies on windows instead of maintaining the build for all our deps ourselves.   It is also possible we will drop support for the 2012 build system in the not so distant future.  Could the community chime in here as to what their needs are, and what they are willing to do to help support the windows builds so we can determine what we plan to support going forward.

Thanks
Mike

https://chocolatey.org/ <https://chocolatey.org/>
http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx <http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx>
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From brian at freeswitch.org  Wed Dec  3 23:44:31 2014
From: brian at freeswitch.org (Brian West)
Date: Wed, 3 Dec 2014 14:44:31 -0600
Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive calls
In-Reply-To: <1662947A-0E07-448C-9BF9-8AF2F99D3FED@gmail.com>
References: <CAGDxTYThLaBEyrO-3R9tYdSVXNPLF+M0yDL3DJb379=tQmuRDQ@mail.gmail.com>
	<1662947A-0E07-448C-9BF9-8AF2F99D3FED@gmail.com>
Message-ID: <CAEJMVkBJAwGRQa8=oVA5Cn81BZdZhvKxv2QnQhLd7avMQ1hEpw@mail.gmail.com>

I would try limiting the crypto suites on the FS side, Look in vars.xml for
details, I know if we send the whole crypt suite list at a polycom on
pretty much any firmware the phone goes rum dumb and starts to echo our
SDP's back at us.  :P

On Wed, Dec 3, 2014 at 12:36 PM, Ahmed Habiba <ahabiba at gmail.com> wrote:

> Dear Brian,
>
>
> I?m using "zoiper 1.19.7? on android, the issue appears only one I enable
> SRTP, for TLS only it is working very fine.
>
> The issue as shown below it keeps registering till finally zipper restart
> on mobile.
>
> One more point I?m configuring Zoiper without using RPORT for either SRTP
> or TLS.
>
> *?2014-12-01 20:57:24.003230 [NOTICE] switch_ivr_originate.c:3467 Hangup
> sofia/internal/sip:1000 at 222.248.102.244 <1000 at 222.248.102.244>:38614
> [CS_CONSUME_MEDIA] [NO_ANSWER]"*
>
>
>
> recv 1290 bytes from tls/[222.215.195.234]:2896 at 20:56:54.344450:
>    ------------------------------------------------------------------------
>    INVITE sip:1000 at abc-xyz.com SIP/2.0
>    v: SIP/2.0/TLS 192.168.8.109:41659
> ;rport;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias
>    Max-Forwards: 70
>    f: <sip:1006 at abc-xyz.com>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU
>    t: <sip:1000 at abc-xyz.com>
>    m: <sip:1006 at 222.215.195.234:2896;transport=TLS;ob>
>    i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX
>    CSeq: 32737 INVITE
>    Route: <sip:abc-xyz.com;transport=tls;lr>
>    k: replaces, 100rel, timer, norefersub
>    x: 1800
>    Min-SE: 90
>    User-Agent: CSipSimple_GT-S5830-10/r2450
>    Proxy-Authorization: Digest username="1006", realm="abc-xyz.com",
> nonce="329456ee-7994-11e4-bf79-1182ebca2429", uri="sip:1000 at abc-xyz.com",
> response="ab3a0f2fe9207ced62eebd8c9b8c32b4", algorithm=MD5,
> cnonce="iLa-C.wpnBSeTQjWiXZK3H3PGYhi6EGt", qop=auth, nc=00000001
>    c: application/sdp
>    l:   469
>
>    v=0
>    o=- 3626452487 3626452487 IN IP4 192.168.8.109
>    s=pjmedia
>    c=IN IP4 192.168.8.109
>    t=0 0
>    m=audio 4000 RTP/SAVP 98 0 8 101
>    c=IN IP4 192.168.8.109
>    a=rtcp:4001 IN IP4 192.168.8.109
>    a=sendrecv
>    a=rtpmap:98 SILK/16000
>    a=fmtp:98 useinbandfec=0
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:Vci7cNdSFTkkLnqz+qRkqCctPvRT6jIOrNc5KMbz
>    a=crypto:2 AES_CM_128_HMAC_SHA1_32
> inline:nW3ECq5YHHnrY2q5u7XZE4fvPfMOAWKQ+ehOp7zV
>    ------------------------------------------------------------------------
>
>    send 405 bytes to tls/[222.215.195.234]:2896 at 20:56:54.344687:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/TLS 192.168.8.109:41659
> ;rport=2896;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias;received=222.215.195.234
>    f: <sip:1006 at abc-xyz.com>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU
>    t: <sip:1000 at abc-xyz.com>
>    i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX
>    CSeq: 32737 INVITE
>    User-Agent:
> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Content-Length: 0
>
>
>
>
>
>
> send 2133 bytes to tls/[222.248.102.244]:38614 at 20:56:54.360608:
>    ------------------------------------------------------------------------
>    INVITE
> sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS
>  SIP/2.0
>    Via: SIP/2.0/TLS 176.31.245.17;branch=z9hG4bKjXZvg3cX76Krj
>    Max-Forwards: 69
>    From: "Extension 1006" <sip:1006 at 176.31.245.17>;tag=15DcKN9tUZeaK
>    To: <
> sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS>
>    Call-ID: 0a3e07c5-f437-1232-9bba-d2ab2784dc6a
>    CSeq: 68395843 INVITE
>    Contact: <sip:mod_sofia at 176.31.245.17:5061;transport=tls>
>    User-Agent:
> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    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: 1076
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "Extension 1006" <sip:1006 at 176.31.245.17
> >;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1417445672 1417445673 IN IP4 176.31.245.17
>    s=FreeSWITCH
>    c=IN IP4 176.31.245.17
>    t=0 0
>    m=audio 18142 RTP/SAVP 0 8 101 13
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=crypto:1 AEAD_AES_256_GCM_8
> inline:6p/uXB7tN5SG9vHP/DMurij6BaZIOKcP4Tle0kjo2lwL1Oa4z9WGyZtk4zs
>    a=crypto:2 AEAD_AES_128_GCM_8
> inline:hGeJA6XQV1V35oTu2141o7mD64dFDRbeCbcf6g
>    a=crypto:3 AES_CM_256_HMAC_SHA1_80
> inline:XdZwwcl6kMRrWY7Vkrbozf/o1awka9lef0UjQRwvnOnnrbAgzXIm8s3yu7o8DQ
>    a=crypto:4 AES_CM_192_HMAC_SHA1_80
> inline:KtM92m0ac2musYPoXLDJb4rUbEMYAwMegPH9RMtzwzPb+0O93e0
>    a=crypto:5 AES_CM_128_HMAC_SHA1_80
> inline:PtOpdIk2LTSCSDVzPaSN6AV2xosBrJUTd4mvKIGh
>    a=crypto:6 AES_CM_256_HMAC_SHA1_32
> inline:ZHzWxajRlQp8IuzX5CuFM1SRkL0huC62ukX583+Vg0LBiK9t2NrpP8FNwdkckQ
>    a=crypto:7 AES_CM_192_HMAC_SHA1_32
> inline:S9BWrDxGFYpU3P9o3HD0Z97W5jwo8vD+E1sJi821r4/B+cdukTw
>    a=crypto:8 AES_CM_128_HMAC_SHA1_32
> inline:l7C0wU29AjF2ZMnCTvks+1ytd/UiXHqN8UIsM6vf
>    a=crypto:9 AES_CM_128_NULL_AUTH
> inline:tv97QsO7+S9NLTXrD02omyWoyM7I2KaWaimUgq99
>    a=ptime:20
>    ------------------------------------------------------------------------
>
>
>       REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
>    Via: SIP/2.0/TLS 192.168.10.2:41120
> ;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport
>    Max-Forwards: 70
>    Contact: <
> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS>
>    To: <sip:1000 at abc-xyz.com;transport=TLS>
>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 1 REGISTER
>    Expires: 60
>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE
>    Supported: replaces, norefersub, extended-refer, timer,
> X-cisco-serviceuri
>    User-Agent: Zoiper r27147
>    Allow-Events: presence, kpml
>    Content-Length: 0
>
>
>
>   ------------------------------------------------------------------------
> send 716 bytes to tls/[222.248.102.244]:38620 at 20:57:01.806433:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/TLS 192.168.10.2:41120
> ;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport=38620;received=222.248.102.244
>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>    To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=2e74mgtyr84ve
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 1 REGISTER
>    User-Agent:
> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    WWW-Authenticate: Digest realm="abc-xyz.com",
> nonce="374fdd98-7994-11e4-bf9c-1182ebca2429", algorithm=MD5, qop="auth"
>    Content-Length: 0
>
>
>
>   ------------------------------------------------------------------------
> recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:01.995662:
>    ------------------------------------------------------------------------
>    REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
>    Via: SIP/2.0/TLS 192.168.10.2:41120
> ;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport
>    Max-Forwards: 70
>    Contact: <
> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS>
>    To: <sip:1000 at abc-xyz.com;transport=TLS>
>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 2 REGISTER
>    Expires: 60
>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE
>    Supported: replaces, norefersub, extended-refer, timer,
> X-cisco-serviceuri
>    User-Agent: Zoiper r27147
>    Authorization: Digest username="1000",realm="abc-xyz.com
> ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri="
> sip:abc-xyz.com;transport=TLS
> ",response="8d7e108ff98035d37424c32166fe0253",cnonce="107cf281c3b57888a4bfd1a9c3776098",nc=00000001,qop=auth,algorithm=MD5
>    Allow-Events: presence, kpml
>    Content-Length: 0
>
>
>
>   ------------------------------------------------------------------------
> send 712 bytes to tls/[222.248.102.244]:38620 at 20:57:01.997931:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/TLS 192.168.10.2:41120
> ;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport=38620;received=222.248.102.244
>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>    To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=3Q0XpBB2NHUFa
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 2 REGISTER
>    Contact: <
> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS
> >;expires=60
>    Date: Mon, 01 Dec 2014 19:57:01 GMT
>    User-Agent:
> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:02.205719:
>    ------------------------------------------------------------------------
>    REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
>    Via: SIP/2.0/TLS 222.248.102.244:38620
> ;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport
>    Max-Forwards: 70
>    Contact: <
> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS
> >;expires=0
>    To: <sip:1000 at abc-xyz.com;transport=TLS>
>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 3 REGISTER
>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE
>    Supported: replaces, norefersub, extended-refer, timer,
> X-cisco-serviceuri
>    User-Agent: Zoiper r27147
>    Authorization: Digest username="1000",realm="abc-xyz.com
> ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri="
> sip:abc-xyz.com;transport=TLS
> ",response="5168bf9afffe9adacef7d898bd6bec9b",cnonce="80108fa519303990cfd2d46639d767ee",nc=00000002,qop=auth,algorithm=MD5
>    Allow-Events: presence, kpml
>    Content-Length: 0
>
>
>   ------------------------------------------------------------------------
> send 598 bytes to tls/[222.248.102.244]:38620 at 20:57:02.207249:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/TLS 222.248.102.244:38620
> ;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport=38620
>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>    To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=40Spr6U5jtH2N
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 3 REGISTER
>    Date: Mon, 01 Dec 2014 19:57:02 GMT
>    User-Agent:
> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 681 bytes from tls/[222.248.102.244]:38620 at 20:57:02.400564:
>    ------------------------------------------------------------------------
>    REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
>    Via: SIP/2.0/TLS 222.248.102.244:38620
> ;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport
>    Max-Forwards: 70
>    Contact: <
> sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS>
>    To: <sip:1000 at abc-xyz.com;transport=TLS>
>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>    CSeq: 1 REGISTER
>    Expires: 60
>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE
>    Supported: replaces, norefersub, extended-refer, timer,
> X-cisco-serviceuri
>    User-Agent: Zoiper r27147
>    Allow-Events: presence, kpml
>    Content-Length: 0
>
>
>
>   ------------------------------------------------------------------------
> send 694 bytes to tls/[222.248.102.244]:38620 at 20:57:02.401167:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/TLS 222.248.102.244:38620
> ;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport=38620
>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
>    To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=59jFt1c9F37mH
>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>    CSeq: 1 REGISTER
>    User-Agent:
> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    WWW-Authenticate: Digest realm="abc-xyz.com",
> nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429", algorithm=MD5, qop="auth"
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 950 bytes from tls/[222.248.102.244]:38620 at 20:57:02.655864:
>    ------------------------------------------------------------------------
>    REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
>    Via: SIP/2.0/TLS 222.248.102.244:38620
> ;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport
>    Max-Forwards: 70
>    Contact: <
> sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS>
>    To: <sip:1000 at abc-xyz.com;transport=TLS>
>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>    CSeq: 2 REGISTER
>    Expires: 60
>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE
>    Supported: replaces, norefersub, extended-refer, timer,
> X-cisco-serviceuri
>    User-Agent: Zoiper r27147
>    Authorization: Digest username="1000",realm="abc-xyz.com
> ",nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429",uri="
> sip:abc-xyz.com;transport=TLS
> ",response="f1667094626fc10ca76a070438497e6f",cnonce="dc790bffe285311f0b903bd515a3e741",nc=00000001,qop=auth,algorithm=MD5
>    Allow-Events: presence, kpml
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 693 bytes to tls/[222.248.102.244]:38620 at 20:57:02.657688:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/TLS 222.248.102.244:38620
> ;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport=38620
>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
>    To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=6jc8UvXcDcy7c
>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>    CSeq: 2 REGISTER
>    Contact: <
> sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS
> >;expires=60
>    Date: Mon, 01 Dec 2014 19:57:02 GMT
>    User-Agent:
> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Content-Length: 0
>
>
>
>
>
>
> *From: *Brian West <brian at freeswitch.org>
> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Date: *December 1, 2014 at 5:19:30 PM GMT+3
> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject: **Re: [Freeswitch-users] FreeSWITCH TLS not able to receive
> calls*
>
>
> What endpoints are involved?  have you looked at 'sofia loglevel all 9'
> output and see if it gives you a clue?
>
> On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba <ahabiba at gmail.com> wrote:
>
>> Dears,
>>
>> I?ve configured FreeSWITCH with the below version with TLS/SRTPas per the
>> recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS? and it
>> was strait forward, and I was able to connect and make make calls using
>> zoiper, but I was not able to receive any calls after enabling the TLS/SRTP.
>>
>> *"FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git
>> b942d0f 2014-11-03 19:53:00Z 64bit)**?*
>>
>> Your kind feedback will be appreciate.
>>
>> 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
>>
>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
>
>
>
>
> --
> Thanks and Best Regards,
>
> Ahmed Habiba
>
> Mob: +20 10 37 82 970
>
> *Success: believe (Vision) plus commitment (Action)*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From ahabiba at gmail.com  Thu Dec  4 00:31:16 2014
From: ahabiba at gmail.com (Ahmed Habiba)
Date: Thu, 4 Dec 2014 00:31:16 +0300
Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive call
In-Reply-To: <mailman.1954.1417639503.581.freeswitch-users@lists.freeswitch.org>
References: <mailman.1954.1417639503.581.freeswitch-users@lists.freeswitch.org>
Message-ID: <BE13BCC9-F1DA-4538-A866-C6B679FE8356@gmail.com>

you mean the below one, to cut some of it? :

<X-PRE-PROCESS cmd="set" data="rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM    _256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH?/>



> 
> 
> 
> From: Brian West <brian at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: December 3, 2014 at 11:44:31 PM GMT+3
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FreeSWITCH TLS not able to receive calls
> 
> 
> I would try limiting the crypto suites on the FS side, Look in vars.xml for details, I know if we send the whole crypt suite list at a polycom on pretty much any firmware the phone goes rum dumb and starts to echo our SDP's back at us.  :P
> 
> On Wed, Dec 3, 2014 at 12:36 PM, Ahmed Habiba <ahabiba at gmail.com <mailto:ahabiba at gmail.com>> wrote:
> Dear Brian,
> 
> 
> I?m using "zoiper 1.19.7? on android, the issue appears only one I enable SRTP, for TLS only it is working very fine.
> 
> The issue as shown below it keeps registering till finally zipper restart on mobile.
> 
> One more point I?m configuring Zoiper without using RPORT for either SRTP or TLS.
> 
> ?2014-12-01 20:57:24.003230 [NOTICE] switch_ivr_originate.c:3467 Hangup sofia/internal/sip:1000 at 222.248.102.244 <mailto:1000 at 222.248.102.244>:38614 [CS_CONSUME_MEDIA] [NO_ANSWER]"
> 
> 
> 
> recv 1290 bytes from tls/[222.215.195.234]:2896 at 20:56:54.344450:
>    ------------------------------------------------------------------------
>    INVITE  <>sip:1000 at abc-xyz.com <> SIP/2.0
>    v: SIP/2.0/TLS 192.168.8.109:41659;rport;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias
>    Max-Forwards: 70
>    f: < <>sip:1006 at abc-xyz.com <>>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU
>    t: < <>sip:1000 at abc-xyz.com <>>
>    m: < <>sip:1006 at 222.215.195.234:2896;transport=TLS;ob <>>
>    i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX
>    CSeq: 32737 INVITE
>    Route: < <>sip:abc-xyz.com;transport=tls;lr <>>
>    k: replaces, 100rel, timer, norefersub
>    x: 1800
>    Min-SE: 90
>    User-Agent: CSipSimple_GT-S5830-10/r2450
>    Proxy-Authorization: Digest username="1006", realm="abc-xyz.com <http://abc-xyz.com/>", nonce="329456ee-7994-11e4-bf79-1182ebca2429", uri=" <>sip:1000 at abc-xyz.com <>", response="ab3a0f2fe9207ced62eebd8c9b8c32b4", algorithm=MD5, cnonce="iLa-C.wpnBSeTQjWiXZK3H3PGYhi6EGt", qop=auth, nc=00000001
>    c: application/sdp
>    l:   469
>    
>    v=0
>    o=- 3626452487 3626452487 IN IP4 192.168.8.109
>    s=pjmedia
>    c=IN IP4 192.168.8.109
>    t=0 0
>    m=audio 4000 RTP/SAVP 98 0 8 101
>    c=IN IP4 192.168.8.109
>    a=rtcp:4001 IN IP4 192.168.8.109
>    a=sendrecv
>    a=rtpmap:98 SILK/16000
>    a=fmtp:98 useinbandfec=0
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Vci7cNdSFTkkLnqz+qRkqCctPvRT6jIOrNc5KMbz
>    a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:nW3ECq5YHHnrY2q5u7XZE4fvPfMOAWKQ+ehOp7zV
>    ------------------------------------------------------------------------
>    
>    send 405 bytes to tls/[222.215.195.234]:2896 at 20:56:54.344687:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/TLS 192.168.8.109:41659;rport=2896;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias;received=222.215.195.234
>    f: < <>sip:1006 at abc-xyz.com <>>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU
>    t: < <>sip:1000 at abc-xyz.com <>>
>    i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX
>    CSeq: 32737 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Content-Length: 0
>    
> 
> 
> 
> 
> 
> send 2133 bytes to tls/[222.248.102.244]:38614 at 20:56:54.360608:
>    ------------------------------------------------------------------------
>    INVITE  <>sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS <> SIP/2.0
>    Via: SIP/2.0/TLS 176.31.245.17;branch=z9hG4bKjXZvg3cX76Krj
>    Max-Forwards: 69
>    From: "Extension 1006" < <>sip:1006 at 176.31.245.17 <>>;tag=15DcKN9tUZeaK
>    To: < <>sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS <>>
>    Call-ID: 0a3e07c5-f437-1232-9bba-d2ab2784dc6a
>    CSeq: 68395843 INVITE
>    Contact: < <>sip:mod_sofia at 176.31.245.17:5061;transport=tls <>>
>    User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    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: 1076
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "Extension 1006" < <>sip:1006 at 176.31.245.17 <>>;party=calling;screen=yes;privacy=off
>    
>    v=0
>    o=FreeSWITCH 1417445672 1417445673 IN IP4 176.31.245.17
>    s=FreeSWITCH
>    c=IN IP4 176.31.245.17
>    t=0 0
>    m=audio 18142 RTP/SAVP 0 8 101 13
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=crypto:1 AEAD_AES_256_GCM_8 inline:6p/uXB7tN5SG9vHP/DMurij6BaZIOKcP4Tle0kjo2lwL1Oa4z9WGyZtk4zs
>    a=crypto:2 AEAD_AES_128_GCM_8 inline:hGeJA6XQV1V35oTu2141o7mD64dFDRbeCbcf6g
>    a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:XdZwwcl6kMRrWY7Vkrbozf/o1awka9lef0UjQRwvnOnnrbAgzXIm8s3yu7o8DQ
>    a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:KtM92m0ac2musYPoXLDJb4rUbEMYAwMegPH9RMtzwzPb+0O93e0
>    a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:PtOpdIk2LTSCSDVzPaSN6AV2xosBrJUTd4mvKIGh
>    a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:ZHzWxajRlQp8IuzX5CuFM1SRkL0huC62ukX583+Vg0LBiK9t2NrpP8FNwdkckQ
>    a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:S9BWrDxGFYpU3P9o3HD0Z97W5jwo8vD+E1sJi821r4/B+cdukTw
>    a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:l7C0wU29AjF2ZMnCTvks+1ytd/UiXHqN8UIsM6vf
>    a=crypto:9 AES_CM_128_NULL_AUTH inline:tv97QsO7+S9NLTXrD02omyWoyM7I2KaWaimUgq99
>    a=ptime:20
>    ------------------------------------------------------------------------
>    
>    
>       REGISTER  <>sip:abc-xyz.com;transport=TLS <> SIP/2.0
>    Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport
>    Max-Forwards: 70
>    Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <>>
>    To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>
>    From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 1 REGISTER
>    Expires: 60
>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
>    Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
>    User-Agent: Zoiper r27147
>    Allow-Events: presence, kpml
>    Content-Length: 0
>    
>    
>       ------------------------------------------------------------------------
> send 716 bytes to tls/[222.248.102.244]:38620 at 20:57:01.806433:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport=38620;received=222.248.102.244
>    From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22
>    To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=2e74mgtyr84ve
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    WWW-Authenticate: Digest realm="abc-xyz.com <http://abc-xyz.com/>", nonce="374fdd98-7994-11e4-bf9c-1182ebca2429", algorithm=MD5, qop="auth"
>    Content-Length: 0
>    
>    
>       ------------------------------------------------------------------------
> recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:01.995662:
>    ------------------------------------------------------------------------
>    REGISTER  <>sip:abc-xyz.com;transport=TLS <> SIP/2.0
>    Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport
>    Max-Forwards: 70
>    Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <>>
>    To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>
>    From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 2 REGISTER
>    Expires: 60
>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
>    Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
>    User-Agent: Zoiper r27147
>    Authorization: Digest username="1000",realm="abc-xyz.com <http://abc-xyz.com/>",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS <>",response="8d7e108ff98035d37424c32166fe0253",cnonce="107cf281c3b57888a4bfd1a9c3776098",nc=00000001,qop=auth,algorithm=MD5
>    Allow-Events: presence, kpml
>    Content-Length: 0
>    
>    
>       ------------------------------------------------------------------------
> send 712 bytes to tls/[222.248.102.244]:38620 at 20:57:01.997931:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/TLS 192.168.10.2:41120;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport=38620;received=222.248.102.244
>    From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22
>    To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=3Q0XpBB2NHUFa
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 2 REGISTER
>    Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <>>;expires=60
>    Date: Mon, 01 Dec 2014 19:57:01 GMT
>    User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Content-Length: 0
>    
>    ------------------------------------------------------------------------
> recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:02.205719:
>    ------------------------------------------------------------------------
>    REGISTER  <>sip:abc-xyz.com;transport=TLS <> SIP/2.0
>    Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport
>    Max-Forwards: 70
>    Contact: < <>sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS <>>;expires=0
>    To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>
>    From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 3 REGISTER
>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
>    Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
>    User-Agent: Zoiper r27147
>    Authorization: Digest username="1000",realm="abc-xyz.com <http://abc-xyz.com/>",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS <>",response="5168bf9afffe9adacef7d898bd6bec9b",cnonce="80108fa519303990cfd2d46639d767ee",nc=00000002,qop=auth,algorithm=MD5
>    Allow-Events: presence, kpml
>    Content-Length: 0
>    
>       ------------------------------------------------------------------------
> send 598 bytes to tls/[222.248.102.244]:38620 at 20:57:02.207249:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport=38620
>    From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=4a2eae22
>    To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=40Spr6U5jtH2N
>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>    CSeq: 3 REGISTER
>    Date: Mon, 01 Dec 2014 19:57:02 GMT
>    User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Content-Length: 0
>    
>    ------------------------------------------------------------------------
> recv 681 bytes from tls/[222.248.102.244]:38620 at 20:57:02.400564:
>    ------------------------------------------------------------------------
>    REGISTER  <>sip:abc-xyz.com;transport=TLS <> SIP/2.0
>    Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport
>    Max-Forwards: 70
>    Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS <>>
>    To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>
>    From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=979e5f46
>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>    CSeq: 1 REGISTER
>    Expires: 60
>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
>    Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
>    User-Agent: Zoiper r27147
>    Allow-Events: presence, kpml
>    Content-Length: 0
>    
>    
>       ------------------------------------------------------------------------
> send 694 bytes to tls/[222.248.102.244]:38620 at 20:57:02.401167:
>    ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport=38620
>    From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=979e5f46
>    To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=59jFt1c9F37mH
>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>    CSeq: 1 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    WWW-Authenticate: Digest realm="abc-xyz.com <http://abc-xyz.com/>", nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429", algorithm=MD5, qop="auth"
>    Content-Length: 0
>    
>    ------------------------------------------------------------------------
> recv 950 bytes from tls/[222.248.102.244]:38620 at 20:57:02.655864:
>    ------------------------------------------------------------------------
>    REGISTER  <>sip:abc-xyz.com;transport=TLS <> SIP/2.0
>    Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport
>    Max-Forwards: 70
>    Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS <>>
>    To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>
>    From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=979e5f46
>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>    CSeq: 2 REGISTER
>    Expires: 60
>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
>    Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
>    User-Agent: Zoiper r27147
>    Authorization: Digest username="1000",realm="abc-xyz.com <http://abc-xyz.com/>",nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429",uri=" <>sip:abc-xyz.com;transport=TLS <>",response="f1667094626fc10ca76a070438497e6f",cnonce="dc790bffe285311f0b903bd515a3e741",nc=00000001,qop=auth,algorithm=MD5
>    Allow-Events: presence, kpml
>    Content-Length: 0
>    
>    ------------------------------------------------------------------------
> send 693 bytes to tls/[222.248.102.244]:38620 at 20:57:02.657688:
>    ------------------------------------------------------------------------
>    SIP/2.0 200 OK
>    Via: SIP/2.0/TLS 222.248.102.244:38620;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport=38620
>    From: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=979e5f46
>    To: < <>sip:1000 at abc-xyz.com;transport=TLS <>>;tag=6jc8UvXcDcy7c
>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>    CSeq: 2 REGISTER
>    Contact: < <>sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS <>>;expires=60
>    Date: Mon, 01 Dec 2014 19:57:02 GMT
>    User-Agent: FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Content-Length: 0
> 
> 
> 
> 
>> 
>> 
>> From: Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org>>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
>> Date: December 1, 2014 at 5:19:30 PM GMT+3
>> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
>> Subject: Re: [Freeswitch-users] FreeSWITCH TLS not able to receive calls
>> 
>> 
>> What endpoints are involved?  have you looked at 'sofia loglevel all 9' output and see if it gives you a clue?
>> 
>> On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba <ahabiba at gmail.com <mailto:ahabiba at gmail.com>> wrote:
>> Dears,
>> 
>> I?ve configured FreeSWITCH with the below version with TLS/SRTPas per the recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS <https://wiki.freeswitch.org/wiki/SIP_TLS>? and it was strait forward, and I was able to connect and make make calls using zoiper, but I was not able to receive any calls after enabling the TLS/SRTP.
>> 
>> "FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)?
>> 
>> Your kind feedback will be appreciate.
>> 
>> Thanks,
>> 
>> Ahmed Habiba.
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
>> http://www.cluecon.com <http://www.cluecon.com/>
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://www.freeswitch.org <http://www.freeswitch.org/>
>> 
>> 
>> 
>> -- 
>> Brian West
>> brian at freeswitch.org <mailto:brian at freeswitch.org>
>> 
>> Twitter: @FreeSWITCH , @briankwest
>> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
>> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
>> T:+19184209001 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378)
>> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype: <>briankwest <>
>> 
> 
> 
> 
> 
> -- 
> Thanks and Best Regards,
>  
> Ahmed Habiba
>  
> Mob: +20 10 37 82 970
>  
> Success: believe (Vision) plus commitment (Action)
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://confluence.freeswitch.org <http://confluence.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.com/>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://www.freeswitch.org <http://www.freeswitch.org/>
> 
> 
> 
> -- 
> Brian West
> brian at freeswitch.org <mailto:brian at freeswitch.org>
> 
> Twitter: @FreeSWITCH , @briankwest
> http://www.freeswitchbook.com <http://www.freeswitchbook.com/>
> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/>
> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ing.antonyam at gmail.com  Thu Dec  4 01:28:22 2014
From: ing.antonyam at gmail.com (Antony Aguirre Morales)
Date: Wed, 3 Dec 2014 16:28:22 -0600
Subject: [Freeswitch-users] Interconect 3 box freeswitch.
Message-ID: <CABFbAiXL63YF2XJVw0Bq=Cv5gzXEzg=czy_=+_MF6=T_aaXyLA@mail.gmail.com>

Hi,

I would like to configure three freeswitch servers but I need that these
servers have the same directory of extensions, is this escenario possible?

If this is not possible, there a way where two freeswitch servers can query
the same database. For example: (graphic)

[10.30.0.0]Freeswitch 1 ---- |
[10.20.0.0]Freeswitch 2 ---- | ----- Directory of extention
[10.10.0.0]Freeswitch 3 ---- |
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From t.mahe at b-and-c.net  Thu Dec  4 01:38:26 2014
From: t.mahe at b-and-c.net (=?windows-1252?Q?Tristan_Mah=E9?=)
Date: Wed, 03 Dec 2014 14:38:26 -0800
Subject: [Freeswitch-users] Interconect 3 box freeswitch.
In-Reply-To: <CABFbAiXL63YF2XJVw0Bq=Cv5gzXEzg=czy_=+_MF6=T_aaXyLA@mail.gmail.com>
References: <CABFbAiXL63YF2XJVw0Bq=Cv5gzXEzg=czy_=+_MF6=T_aaXyLA@mail.gmail.com>
Message-ID: <547F90E2.6080503@b-and-c.net>

Hi Antony,

You may look at xml_curl for these needs.

Le 03/12/2014 14:28, Antony Aguirre Morales a ?crit :
> Hi,
> 
> I would like to configure three freeswitch servers but I need that these
> servers have the same directory of extensions, is this escenario possible?
> 
> If this is not possible, there a way where two freeswitch servers can
> query the same database. For example: (graphic)
> 
> [10.30.0.0]Freeswitch 1 ---- |
> [10.20.0.0]Freeswitch 2 ---- | ----- Directory of extention
> [10.10.0.0]Freeswitch 3 ---- |
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


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From andretodd at verizon.net  Thu Dec  4 05:00:13 2014
From: andretodd at verizon.net (Andre DeMattia)
Date: Wed, 03 Dec 2014 21:00:13 -0500
Subject: [Freeswitch-users] emails not coming
Message-ID: <0fe501d00f66$0ba778b0$22f66a10$@verizon.net>

Hi, I no longer get the FreeSWITCH emails. Can someone check on that for me?

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From iwada.bassey at gmail.com  Thu Dec  4 11:06:56 2014
From: iwada.bassey at gmail.com (Iwada Eja)
Date: Thu, 4 Dec 2014 03:06:56 -0500
Subject: [Freeswitch-users] emails not coming
In-Reply-To: <0fe501d00f66$0ba778b0$22f66a10$@verizon.net>
References: <0fe501d00f66$0ba778b0$22f66a10$@verizon.net>
Message-ID: <CAHaUJVY5S1539ifcD=sc8VhUhYQViBQCG1yiuT5L7LC_aMtPKA@mail.gmail.com>

Hi Andre,
You still subscribed to the list. Checked your Spam? Probably your
Firewall, Content Filter, or Email Security Policy?

On Wed, Dec 3, 2014 at 9:00 PM, Andre DeMattia <andretodd at verizon.net>
wrote:

> Hi, I no longer get the FreeSWITCH emails. Can someone check on that for
> me?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kind Regards
Iwada
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From iwada.bassey at gmail.com  Thu Dec  4 11:43:31 2014
From: iwada.bassey at gmail.com (Iwada Eja)
Date: Thu, 4 Dec 2014 03:43:31 -0500
Subject: [Freeswitch-users] emails not coming
In-Reply-To: <CAHaUJVY5S1539ifcD=sc8VhUhYQViBQCG1yiuT5L7LC_aMtPKA@mail.gmail.com>
References: <0fe501d00f66$0ba778b0$22f66a10$@verizon.net>
	<CAHaUJVY5S1539ifcD=sc8VhUhYQViBQCG1yiuT5L7LC_aMtPKA@mail.gmail.com>
Message-ID: <CAHaUJVZU0a9QMxJzu=14BFixnpPiXJCHYxUAUv6moQxuHvKOVw@mail.gmail.com>

... On a Second Look, You have list delivery disabled. You can Log in to
your options page and re enable it.

On Thu, Dec 4, 2014 at 3:06 AM, Iwada Eja <iwada.bassey at gmail.com> wrote:

> Hi Andre,
> You still subscribed to the list. Checked your Spam? Probably your
> Firewall, Content Filter, or Email Security Policy?
>
> On Wed, Dec 3, 2014 at 9:00 PM, Andre DeMattia <andretodd at verizon.net>
> wrote:
>
>> Hi, I no longer get the FreeSWITCH emails. Can someone check on that for
>> me?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Kind Regards
> Iwada
>



-- 
Kind Regards
Iwada
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From mbodbg at gmx.net  Thu Dec  4 16:24:59 2014
From: mbodbg at gmx.net (mbo)
Date: Thu, 4 Dec 2014 14:24:59 +0100
Subject: [Freeswitch-users] Status of chat in mod_verto
Message-ID: <350D7621-8242-4C63-A1BB-575E1397D46D@gmx.net>

Hello,

we are playing around with mod_verto. The audio part is working fine, but what is the status with the chat implementation. On the demo page on the internet (bbb.freeswitch.org), I can see that chat is working at least in a conference. But does chat also work between 2 users logged in via mod_verto? 
On the demo page I can see that code handling chat is commented out, isn?t it ready yet? We modified the script a bit and we are at least able to see that chat messages are received on the server like:

2014-12-04 12:10:18.228695 [ALERT] mod_verto.c:1284 READ 192.168.0.214:56164 [{
    "jsonrpc":      "2.0",
    "method":       "verto.info",
    "params":       {
            "msg":  {
                    "to":   "1009",
                    "body": "test"
            },
            "sessid":       "f7d3c02c-71ac-1e42-9a6b-0d7cc0d355ad"
    },
    "id":   14
}]
2014-12-04 12:10:18.228695 [ALERT] mod_verto.c:594 WRITE 192.168.0.214:56164 [{
    "jsonrpc":      "2.0",
    "id":   14,
    "result":       {
            "message":      "SENT",
            "sessid":       "f7d3c02c-71ac-1e42-9a6b-0d7cc0d355ad"
    }
}]

but we do not see anything on the destination client. Does anyone else have chat already working?

Thanks and Regards

Markus






From areski at gmail.com  Thu Dec  4 21:20:47 2014
From: areski at gmail.com (Areski)
Date: Thu, 4 Dec 2014 19:20:47 +0100
Subject: [Freeswitch-users] Doc-Sprint Friday 12 December 2014
Message-ID: <CAMf7LBiAKzHdsML1mGoPPn-ykKd8Zu72m2JrwMfka_Aedx-BDA@mail.gmail.com>

Hi everyone,

We are planning to organize an other doc sprint on *Friday 12 December at
10am CT*.
It will be 4 hours long but you can join for less time.

The Doc-sprint will focus on migrating the remaining pages from MediaWiki (
https://wiki.freeswitch.org) to Confluence Wiki (
https://freeswitch.org/confluence).

We will use an FS IRC channel during the sprint: *#freeswitch-docs*
and will track our work on the spreadsheet:
https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333

During the sprint, please change the URL's "Status" you are working on to
"Editing" with your name next to it so we don't duplicate work.

Some extra information:
- https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration
-
https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation

We hope to get a maximum number of people signed up!

Peoples confirmed so far:
- Italo Rossi (+4)
- Areski Belaid


So, who is in?
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From iwada.bassey at gmail.com  Thu Dec  4 23:00:47 2014
From: iwada.bassey at gmail.com (Iwada Eja)
Date: Thu, 4 Dec 2014 15:00:47 -0500
Subject: [Freeswitch-users] Doc-Sprint Friday 12 December 2014
In-Reply-To: <CAMf7LBiAKzHdsML1mGoPPn-ykKd8Zu72m2JrwMfka_Aedx-BDA@mail.gmail.com>
References: <CAMf7LBiAKzHdsML1mGoPPn-ykKd8Zu72m2JrwMfka_Aedx-BDA@mail.gmail.com>
Message-ID: <CAHaUJVbNaOzCUiwYN-ka5OCY=OsY9M2Ri_zKwyoRKrGuoGVwgw@mail.gmail.com>

I'm In

On Thu, Dec 4, 2014 at 1:20 PM, Areski <areski at gmail.com> wrote:

> Hi everyone,
>
> We are planning to organize an other doc sprint on *Friday 12 December at
> 10am CT*.
> It will be 4 hours long but you can join for less time.
>
> The Doc-sprint will focus on migrating the remaining pages from MediaWiki (
> https://wiki.freeswitch.org) to Confluence Wiki (
> https://freeswitch.org/confluence).
>
> We will use an FS IRC channel during the sprint: *#freeswitch-docs*
> and will track our work on the spreadsheet:
> https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333
>
> During the sprint, please change the URL's "Status" you are working on to
> "Editing" with your name next to it so we don't duplicate work.
>
> Some extra information:
> - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration
> -
> https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation
>
> We hope to get a maximum number of people signed up!
>
> Peoples confirmed so far:
> - Italo Rossi (+4)
> - Areski Belaid
>
>
> So, who is in?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting 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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>



-- 
Kind Regards
Iwada
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From brian at freeswitch.org  Thu Dec  4 23:22:15 2014
From: brian at freeswitch.org (Brian West)
Date: Thu, 4 Dec 2014 14:22:15 -0600
Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive call
In-Reply-To: <BE13BCC9-F1DA-4538-A866-C6B679FE8356@gmail.com>
References: <mailman.1954.1417639503.581.freeswitch-users@lists.freeswitch.org>
	<BE13BCC9-F1DA-4538-A866-C6B679FE8356@gmail.com>
Message-ID: <CAEJMVkA_k5tKTObEn6AjSXqdZpSvcmu2PFYDAqHRmQfOopYv7A@mail.gmail.com>

Yes.

On Wed, Dec 3, 2014 at 3:31 PM, Ahmed Habiba <ahabiba at gmail.com> wrote:

> you mean the below one, to cut some of it? :
>
> <X-PRE-PROCESS cmd="set" data=
> "rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM
>
> _256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH?
> />
>
>
>
>
>
>
> *From: *Brian West <brian at freeswitch.org>
> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Date: *December 3, 2014 at 11:44:31 PM GMT+3
> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject: **Re: [Freeswitch-users] FreeSWITCH TLS not able to receive
> calls*
>
>
> I would try limiting the crypto suites on the FS side, Look in vars.xml
> for details, I know if we send the whole crypt suite list at a polycom on
> pretty much any firmware the phone goes rum dumb and starts to echo our
> SDP's back at us.  :P
>
> On Wed, Dec 3, 2014 at 12:36 PM, Ahmed Habiba <ahabiba at gmail.com> wrote:
>
>> Dear Brian,
>>
>>
>> I?m using "zoiper 1.19.7? on android, the issue appears only one I enable
>> SRTP, for TLS only it is working very fine.
>>
>> The issue as shown below it keeps registering till finally zipper restart
>> on mobile.
>>
>> One more point I?m configuring Zoiper without using RPORT for either SRTP
>> or TLS.
>>
>> *?2014-12-01 20:57:24.003230 [NOTICE] switch_ivr_originate.c:3467 Hangup
>> sofia/internal/sip:1000 at 222.248.102.244 <1000 at 222.248.102.244>:38614
>> [CS_CONSUME_MEDIA] [NO_ANSWER]"*
>>
>>
>>
>> recv 1290 bytes from tls/[222.215.195.234]:2896 at 20:56:54.344450:
>>
>>  ------------------------------------------------------------------------
>>    INVITE sip:1000 at abc-xyz.com SIP/2.0
>>    v: SIP/2.0/TLS 192.168.8.109:41659
>> ;rport;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias
>>    Max-Forwards: 70
>>    f: <sip:1006 at abc-xyz.com>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU
>>    t: <sip:1000 at abc-xyz.com>
>>    m: <sip:1006 at 222.215.195.234:2896;transport=TLS;ob>
>>    i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX
>>    CSeq: 32737 INVITE
>>    Route: <sip:abc-xyz.com;transport=tls;lr>
>>    k: replaces, 100rel, timer, norefersub
>>    x: 1800
>>    Min-SE: 90
>>    User-Agent: CSipSimple_GT-S5830-10/r2450
>>    Proxy-Authorization: Digest username="1006", realm="abc-xyz.com",
>> nonce="329456ee-7994-11e4-bf79-1182ebca2429", uri="sip:1000 at abc-xyz.com",
>> response="ab3a0f2fe9207ced62eebd8c9b8c32b4", algorithm=MD5,
>> cnonce="iLa-C.wpnBSeTQjWiXZK3H3PGYhi6EGt", qop=auth, nc=00000001
>>    c: application/sdp
>>    l:   469
>>
>>    v=0
>>    o=- 3626452487 3626452487 IN IP4 192.168.8.109
>>    s=pjmedia
>>    c=IN IP4 192.168.8.109
>>    t=0 0
>>    m=audio 4000 RTP/SAVP 98 0 8 101
>>    c=IN IP4 192.168.8.109
>>    a=rtcp:4001 IN IP4 192.168.8.109
>>    a=sendrecv
>>    a=rtpmap:98 SILK/16000
>>    a=fmtp:98 useinbandfec=0
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-16
>>    a=crypto:1 AES_CM_128_HMAC_SHA1_80
>> inline:Vci7cNdSFTkkLnqz+qRkqCctPvRT6jIOrNc5KMbz
>>    a=crypto:2 AES_CM_128_HMAC_SHA1_32
>> inline:nW3ECq5YHHnrY2q5u7XZE4fvPfMOAWKQ+ehOp7zV
>>
>>  ------------------------------------------------------------------------
>>
>>    send 405 bytes to tls/[222.215.195.234]:2896 at 20:56:54.344687:
>>
>>  ------------------------------------------------------------------------
>>    SIP/2.0 100 Trying
>>    Via: SIP/2.0/TLS 192.168.8.109:41659
>> ;rport=2896;branch=z9hG4bKPjvnvRZjpAXahzIV7S8SRgS5vwKmYI9WFS;alias;received=222.215.195.234
>>    f: <sip:1006 at abc-xyz.com>;tag=weeMqzInPcNLPd6s5IX63c8be3DMvGkU
>>    t: <sip:1000 at abc-xyz.com>
>>    i: sjprJQszC2LHvY6jRr-lm9rau9V1PlYX
>>    CSeq: 32737 INVITE
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>>    Content-Length: 0
>>
>>
>>
>>
>>
>>
>> send 2133 bytes to tls/[222.248.102.244]:38614 at 20:56:54.360608:
>>
>>  ------------------------------------------------------------------------
>>    INVITE
>> sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS
>>  SIP/2.0
>>    Via: SIP/2.0/TLS 176.31.245.17;branch=z9hG4bKjXZvg3cX76Krj
>>    Max-Forwards: 69
>>    From: "Extension 1006" <sip:1006 at 176.31.245.17>;tag=15DcKN9tUZeaK
>>    To: <
>> sip:1000 at 222.248.102.244:38614;rinstance=98c51296f2d531cc;transport=TLS>
>>    Call-ID: 0a3e07c5-f437-1232-9bba-d2ab2784dc6a
>>    CSeq: 68395843 INVITE
>>    Contact: <sip:mod_sofia at 176.31.245.17:5061;transport=tls>
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>>    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: 1076
>>    X-FS-Support: update_display,send_info
>>    Remote-Party-ID: "Extension 1006" <sip:1006 at 176.31.245.17
>> >;party=calling;screen=yes;privacy=off
>>
>>    v=0
>>    o=FreeSWITCH 1417445672 1417445673 IN IP4 176.31.245.17
>>    s=FreeSWITCH
>>    c=IN IP4 176.31.245.17
>>    t=0 0
>>    m=audio 18142 RTP/SAVP 0 8 101 13
>>    a=rtpmap:0 PCMU/8000
>>    a=rtpmap:8 PCMA/8000
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-16
>>    a=crypto:1 AEAD_AES_256_GCM_8
>> inline:6p/uXB7tN5SG9vHP/DMurij6BaZIOKcP4Tle0kjo2lwL1Oa4z9WGyZtk4zs
>>    a=crypto:2 AEAD_AES_128_GCM_8
>> inline:hGeJA6XQV1V35oTu2141o7mD64dFDRbeCbcf6g
>>    a=crypto:3 AES_CM_256_HMAC_SHA1_80
>> inline:XdZwwcl6kMRrWY7Vkrbozf/o1awka9lef0UjQRwvnOnnrbAgzXIm8s3yu7o8DQ
>>    a=crypto:4 AES_CM_192_HMAC_SHA1_80
>> inline:KtM92m0ac2musYPoXLDJb4rUbEMYAwMegPH9RMtzwzPb+0O93e0
>>    a=crypto:5 AES_CM_128_HMAC_SHA1_80
>> inline:PtOpdIk2LTSCSDVzPaSN6AV2xosBrJUTd4mvKIGh
>>    a=crypto:6 AES_CM_256_HMAC_SHA1_32
>> inline:ZHzWxajRlQp8IuzX5CuFM1SRkL0huC62ukX583+Vg0LBiK9t2NrpP8FNwdkckQ
>>    a=crypto:7 AES_CM_192_HMAC_SHA1_32
>> inline:S9BWrDxGFYpU3P9o3HD0Z97W5jwo8vD+E1sJi821r4/B+cdukTw
>>    a=crypto:8 AES_CM_128_HMAC_SHA1_32
>> inline:l7C0wU29AjF2ZMnCTvks+1ytd/UiXHqN8UIsM6vf
>>    a=crypto:9 AES_CM_128_NULL_AUTH
>> inline:tv97QsO7+S9NLTXrD02omyWoyM7I2KaWaimUgq99
>>    a=ptime:20
>>
>>  ------------------------------------------------------------------------
>>
>>
>>       REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
>>    Via: SIP/2.0/TLS 192.168.10.2:41120
>> ;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport
>>    Max-Forwards: 70
>>    Contact: <
>> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS>
>>    To: <sip:1000 at abc-xyz.com;transport=TLS>
>>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>>    CSeq: 1 REGISTER
>>    Expires: 60
>>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
>> INFO, SUBSCRIBE
>>    Supported: replaces, norefersub, extended-refer, timer,
>> X-cisco-serviceuri
>>    User-Agent: Zoiper r27147
>>    Allow-Events: presence, kpml
>>    Content-Length: 0
>>
>>
>>
>>   ------------------------------------------------------------------------
>> send 716 bytes to tls/[222.248.102.244]:38620 at 20:57:01.806433:
>>
>>  ------------------------------------------------------------------------
>>    SIP/2.0 401 Unauthorized
>>    Via: SIP/2.0/TLS 192.168.10.2:41120
>> ;branch=z9hG4bK-d8754z-903949acb1d10083-1---d8754z-;rport=38620;received=222.248.102.244
>>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>>    To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=2e74mgtyr84ve
>>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>>    CSeq: 1 REGISTER
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, path, replaces
>>    WWW-Authenticate: Digest realm="abc-xyz.com",
>> nonce="374fdd98-7994-11e4-bf9c-1182ebca2429", algorithm=MD5, qop="auth"
>>    Content-Length: 0
>>
>>
>>
>>   ------------------------------------------------------------------------
>> recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:01.995662:
>>
>>  ------------------------------------------------------------------------
>>    REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
>>    Via: SIP/2.0/TLS 192.168.10.2:41120
>> ;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport
>>    Max-Forwards: 70
>>    Contact: <
>> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS>
>>    To: <sip:1000 at abc-xyz.com;transport=TLS>
>>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>>    CSeq: 2 REGISTER
>>    Expires: 60
>>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
>> INFO, SUBSCRIBE
>>    Supported: replaces, norefersub, extended-refer, timer,
>> X-cisco-serviceuri
>>    User-Agent: Zoiper r27147
>>    Authorization: Digest username="1000",realm="abc-xyz.com
>> ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri="
>> sip:abc-xyz.com;transport=TLS
>> ",response="8d7e108ff98035d37424c32166fe0253",cnonce="107cf281c3b57888a4bfd1a9c3776098",nc=00000001,qop=auth,algorithm=MD5
>>    Allow-Events: presence, kpml
>>    Content-Length: 0
>>
>>
>>
>>   ------------------------------------------------------------------------
>> send 712 bytes to tls/[222.248.102.244]:38620 at 20:57:01.997931:
>>
>>  ------------------------------------------------------------------------
>>    SIP/2.0 200 OK
>>    Via: SIP/2.0/TLS 192.168.10.2:41120
>> ;branch=z9hG4bK-d8754z-8ca0e782336414eb-1---d8754z-;rport=38620;received=222.248.102.244
>>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>>    To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=3Q0XpBB2NHUFa
>>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>>    CSeq: 2 REGISTER
>>    Contact: <
>> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS
>> >;expires=60
>>    Date: Mon, 01 Dec 2014 19:57:01 GMT
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, path, replaces
>>    Content-Length: 0
>>
>>
>>  ------------------------------------------------------------------------
>> recv 944 bytes from tls/[222.248.102.244]:38620 at 20:57:02.205719:
>>
>>  ------------------------------------------------------------------------
>>    REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
>>    Via: SIP/2.0/TLS 222.248.102.244:38620
>> ;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport
>>    Max-Forwards: 70
>>    Contact: <
>> sip:1000 at 192.168.10.2:41120;rinstance=57d4e16c725a51e6;transport=TLS
>> >;expires=0
>>    To: <sip:1000 at abc-xyz.com;transport=TLS>
>>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>>    CSeq: 3 REGISTER
>>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
>> INFO, SUBSCRIBE
>>    Supported: replaces, norefersub, extended-refer, timer,
>> X-cisco-serviceuri
>>    User-Agent: Zoiper r27147
>>    Authorization: Digest username="1000",realm="abc-xyz.com
>> ",nonce="374fdd98-7994-11e4-bf9c-1182ebca2429",uri="
>> sip:abc-xyz.com;transport=TLS
>> ",response="5168bf9afffe9adacef7d898bd6bec9b",cnonce="80108fa519303990cfd2d46639d767ee",nc=00000002,qop=auth,algorithm=MD5
>>    Allow-Events: presence, kpml
>>    Content-Length: 0
>>
>>
>>   ------------------------------------------------------------------------
>> send 598 bytes to tls/[222.248.102.244]:38620 at 20:57:02.207249:
>>
>>  ------------------------------------------------------------------------
>>    SIP/2.0 200 OK
>>    Via: SIP/2.0/TLS 222.248.102.244:38620
>> ;branch=z9hG4bK-d8754z-59d493abef215091-1---d8754z-;rport=38620
>>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=4a2eae22
>>    To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=40Spr6U5jtH2N
>>    Call-ID: N2RiMjE2ZTQxNzYzMGY3ZmM3ZGJhODM4NTMzZmQ1YTA.
>>    CSeq: 3 REGISTER
>>    Date: Mon, 01 Dec 2014 19:57:02 GMT
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, path, replaces
>>    Content-Length: 0
>>
>>
>>  ------------------------------------------------------------------------
>> recv 681 bytes from tls/[222.248.102.244]:38620 at 20:57:02.400564:
>>
>>  ------------------------------------------------------------------------
>>    REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
>>    Via: SIP/2.0/TLS 222.248.102.244:38620
>> ;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport
>>    Max-Forwards: 70
>>    Contact: <
>> sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS>
>>    To: <sip:1000 at abc-xyz.com;transport=TLS>
>>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
>>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>>    CSeq: 1 REGISTER
>>    Expires: 60
>>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
>> INFO, SUBSCRIBE
>>    Supported: replaces, norefersub, extended-refer, timer,
>> X-cisco-serviceuri
>>    User-Agent: Zoiper r27147
>>    Allow-Events: presence, kpml
>>    Content-Length: 0
>>
>>
>>
>>   ------------------------------------------------------------------------
>> send 694 bytes to tls/[222.248.102.244]:38620 at 20:57:02.401167:
>>
>>  ------------------------------------------------------------------------
>>    SIP/2.0 401 Unauthorized
>>    Via: SIP/2.0/TLS 222.248.102.244:38620
>> ;branch=z9hG4bK-d8754z-d738a1a836608006-1---d8754z-;rport=38620
>>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
>>    To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=59jFt1c9F37mH
>>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>>    CSeq: 1 REGISTER
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, path, replaces
>>    WWW-Authenticate: Digest realm="abc-xyz.com",
>> nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429", algorithm=MD5, qop="auth"
>>    Content-Length: 0
>>
>>
>>  ------------------------------------------------------------------------
>> recv 950 bytes from tls/[222.248.102.244]:38620 at 20:57:02.655864:
>>
>>  ------------------------------------------------------------------------
>>    REGISTER sip:abc-xyz.com;transport=TLS SIP/2.0
>>    Via: SIP/2.0/TLS 222.248.102.244:38620
>> ;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport
>>    Max-Forwards: 70
>>    Contact: <
>> sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS>
>>    To: <sip:1000 at abc-xyz.com;transport=TLS>
>>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
>>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>>    CSeq: 2 REGISTER
>>    Expires: 60
>>    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
>> INFO, SUBSCRIBE
>>    Supported: replaces, norefersub, extended-refer, timer,
>> X-cisco-serviceuri
>>    User-Agent: Zoiper r27147
>>    Authorization: Digest username="1000",realm="abc-xyz.com
>> ",nonce="37aaa44e-7994-11e4-bf9d-1182ebca2429",uri="
>> sip:abc-xyz.com;transport=TLS
>> ",response="f1667094626fc10ca76a070438497e6f",cnonce="dc790bffe285311f0b903bd515a3e741",nc=00000001,qop=auth,algorithm=MD5
>>    Allow-Events: presence, kpml
>>    Content-Length: 0
>>
>>
>>  ------------------------------------------------------------------------
>> send 693 bytes to tls/[222.248.102.244]:38620 at 20:57:02.657688:
>>
>>  ------------------------------------------------------------------------
>>    SIP/2.0 200 OK
>>    Via: SIP/2.0/TLS 222.248.102.244:38620
>> ;branch=z9hG4bK-d8754z-dfaa7130675d6cfa-1---d8754z-;rport=38620
>>    From: <sip:1000 at abc-xyz.com;transport=TLS>;tag=979e5f46
>>    To: <sip:1000 at abc-xyz.com;transport=TLS>;tag=6jc8UvXcDcy7c
>>    Call-ID: MTJhNjIzMDM1MDEwMDVjZDRkOGMyYjBlMWU4ZDEzY2Y.
>>    CSeq: 2 REGISTER
>>    Contact: <
>> sip:1000 at 222.248.102.244:38620;rinstance=43166fee0aaac1bc;transport=TLS
>> >;expires=60
>>    Date: Mon, 01 Dec 2014 19:57:02 GMT
>>    User-Agent:
>> FreeSWITCH-mod_sofia/1.4.13+git~20141103T195300Z~b942d0faa8~64bit
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>    Supported: timer, path, replaces
>>    Content-Length: 0
>>
>>
>>
>>
>>
>>
>> *From: *Brian West <brian at freeswitch.org>
>> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Date: *December 1, 2014 at 5:19:30 PM GMT+3
>> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject: **Re: [Freeswitch-users] FreeSWITCH TLS not able to receive
>> calls*
>>
>>
>> What endpoints are involved?  have you looked at 'sofia loglevel all 9'
>> output and see if it gives you a clue?
>>
>> On Fri, Nov 28, 2014 at 6:24 PM, Ahmed Habiba <ahabiba at gmail.com> wrote:
>>
>>> Dears,
>>>
>>> I?ve configured FreeSWITCH with the below version with TLS/SRTPas per
>>> the recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS?
>>> and it was strait forward, and I was able to connect and make make calls
>>> using zoiper, but I was not able to receive any calls after enabling the
>>> TLS/SRTP.
>>>
>>> *"FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git
>>> b942d0f 2014-11-03 19:53:00Z 64bit)**?*
>>>
>>> Your kind feedback will be appreciate.
>>>
>>> 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
>>>
>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>>
>>
>>
>>
>>
>> --
>> Thanks and Best Regards,
>>
>> Ahmed Habiba
>>
>> Mob: +20 10 37 82 970
>>
>> *Success: believe (Vision) plus commitment (Action)*
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From anthony.minessale at gmail.com  Thu Dec  4 23:23:11 2014
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 4 Dec 2014 14:23:11 -0600
Subject: [Freeswitch-users] Windows build system
In-Reply-To: <E706BBBE-829E-4A71-A299-5C61C5011CF0@jerris.com>
References: <E706BBBE-829E-4A71-A299-5C61C5011CF0@jerris.com>
Message-ID: <CAKbxfG-uBZmDj+wgex_dyE5FSaW6Xd0rRSa1QNJNV=Cc_+q0nQ@mail.gmail.com>

*crickets* ?

On Wed, Dec 3, 2014 at 2:31 PM, Michael Jerris <mike at jerris.com> wrote:

> Given the recent announcements by Microsoft about the community edition
> 2013 being available, we are working to migrate the build system towards
> using that as our primary build.  As part of this process we will be very
> soon dropping support for any version of Visual Studio prior to 2012.  If
> you feel strongly about needing support for these older versions, please
> speak up now with an offer to maintain these legacy build systems.  We are
> also investigating moving to using chocolatey as a new system to manage
> dependencies on windows instead of maintaining the build for all our deps
> ourselves.   It is also possible we will drop support for the 2012 build
> system in the not so distant future.  Could the community chime in here as
> to what their needs are, and what they are willing to do to help support
> the windows builds so we can determine what we plan to support going
> forward.
>
> Thanks
> Mike
>
> https://chocolatey.org/
> http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From aademattia at comcast.net  Fri Dec  5 00:21:08 2014
From: aademattia at comcast.net (Andrew)
Date: Thu, 4 Dec 2014 16:21:08 -0500
Subject: [Freeswitch-users] Windows build system
In-Reply-To: <CAKbxfG-uBZmDj+wgex_dyE5FSaW6Xd0rRSa1QNJNV=Cc_+q0nQ@mail.gmail.com>
References: <E706BBBE-829E-4A71-A299-5C61C5011CF0@jerris.com>
	<CAKbxfG-uBZmDj+wgex_dyE5FSaW6Xd0rRSa1QNJNV=Cc_+q0nQ@mail.gmail.com>
Message-ID: <086001d01008$3c66b090$b53411b0$@comcast.net>

Hi,

I use Visual Studio prior to 2013.  I am sure community edition will still work.

I will  help for the windows builds anywhere I can.

 

 

Andrew

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Thursday, December 4, 2014 3:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Windows build system

 

*crickets* ?  

 

On Wed, Dec 3, 2014 at 2:31 PM, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com> > wrote:

Given the recent announcements by Microsoft about the community edition 2013 being available, we are working to migrate the build system towards using that as our primary build.  As part of this process we will be very soon dropping support for any version of Visual Studio prior to 2012.  If you feel strongly about needing support for these older versions, please speak up now with an offer to maintain these legacy build systems.  We are also investigating moving to using chocolatey as a new system to manage dependencies on windows instead of maintaining the build for all our deps ourselves.   It is also possible we will drop support for the 2012 build system in the not so distant future.  Could the community chime in here as to what their needs are, and what they are willing to do to help support the windows builds so we can determine what we plan to support going forward.

 

Thanks

Mike

 

https://chocolatey.org/

http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx


_________________________________________________________________________
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 

 

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From jcabezas at inovax.com.br  Fri Dec  5 01:53:16 2014
From: jcabezas at inovax.com.br (Julio Cesar Esteves Cabezas)
Date: Thu, 4 Dec 2014 20:53:16 -0200
Subject: [Freeswitch-users] delay_echo with half of time
Message-ID: <039001d01015$17b4d8f0$471e8ad0$@inovax.com.br>

Hi,

                The delay  I am getting from delay_echo is roughly half of
what I pass as parameter in its data.

 

                This is the snippet, and the log shows on screen that I am
passing what I intend to 

 

    <extension name="delay_echo">

      <condition field="destination_number" expression="^\*7(\d)$">

                <action application="answer"/>

                <action application="set" data="delta=${expr(1+1000*$1)}"/>

                <action application="log" data=" *** delta = ${delta} "/>

                <action application="delay_echo" data="${delta}"/>

      </condition>

    </extension>

 

                I am running FreeSWITCH Version 1.5.12b~64bit ( 64bit)  (as
per command "version" in fs_cli)  in a VMware VM with OS=Windows Server
2012.

 

Thanks.

 

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From markus.klenk at googlemail.com  Fri Dec  5 13:22:20 2014
From: markus.klenk at googlemail.com (Paul Klenk)
Date: Fri, 5 Dec 2014 11:22:20 +0100
Subject: [Freeswitch-users] remove country code from destination_number
Message-ID: <CABR18gdjXQXRJTP2TxWmB6RtKSxd=pLh44WxPBpzSPyJWhLORA@mail.gmail.com>

Hi,

I'd like to have the country code "+49" removed from
destination_number and changed to "0"

I tried:
    <extension name="international" continue="true">
      <condition field="destination_number" expression="^\+49(.*)">
        <action inline="true" application="set" data="destination_number=0$1"/>
      </condition>
    </extension>

but after correctly executing this extension-statement it reverts to
the original version of destination_number


From avi at avimarcus.net  Fri Dec  5 13:34:03 2014
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 5 Dec 2014 10:34:03 +0000
Subject: [Freeswitch-users] remove country code from destination_number
In-Reply-To: <CABR18gdjXQXRJTP2TxWmB6RtKSxd=pLh44WxPBpzSPyJWhLORA@mail.gmail.com>
References: <CABR18gdjXQXRJTP2TxWmB6RtKSxd=pLh44WxPBpzSPyJWhLORA@mail.gmail.com>
Message-ID: <0000014a1a037bde-8bdda21d-1ca7-4a75-bbf5-3fccfed8bc30-000000@email.amazonses.com>

You can't/shouldn't modify destination_number, it's part of the caller
profile.
You can either transfer it to 0$1 to have it hit the dialplan again, or
bridge to your gateway and use 0$1 as the destination.

-Avi

On Fri, Dec 5, 2014 at 12:22 PM, Paul Klenk <markus.klenk at googlemail.com>
wrote:

> Hi,
>
> I'd like to have the country code "+49" removed from
> destination_number and changed to "0"
>
> I tried:
>     <extension name="international" continue="true">
>       <condition field="destination_number" expression="^\+49(.*)">
>         <action inline="true" application="set"
> data="destination_number=0$1"/>
>       </condition>
>     </extension>
>
> but after correctly executing this extension-statement it reverts to
> the original version of destination_number
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ssinyagin at gmail.com  Fri Dec  5 14:52:44 2014
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Fri, 5 Dec 2014 12:52:44 +0100
Subject: [Freeswitch-users] remove country code from destination_number
In-Reply-To: <CABR18gdjXQXRJTP2TxWmB6RtKSxd=pLh44WxPBpzSPyJWhLORA@mail.gmail.com>
References: <CABR18gdjXQXRJTP2TxWmB6RtKSxd=pLh44WxPBpzSPyJWhLORA@mail.gmail.com>
Message-ID: <CACOYK=jo9Tu4LY0zG7PTpwfjmgkmUAwgsHKE2R5P_hO8YiuHPw@mail.gmail.com>

you just need to use a different variable to store the modified
destination number.
here are a couple of examples from my blog:

http://txlab.wordpress.com/2014/10/21/using-voxbeam-for-outbound-calls-with-freeswitch/
http://txlab.wordpress.com/2013/08/03/free-us-number-and-caller-id-manipulation/




On Fri, Dec 5, 2014 at 11:22 AM, Paul Klenk <markus.klenk at googlemail.com> wrote:
> Hi,
>
> I'd like to have the country code "+49" removed from
> destination_number and changed to "0"
>
> I tried:
>     <extension name="international" continue="true">
>       <condition field="destination_number" expression="^\+49(.*)">
>         <action inline="true" application="set" data="destination_number=0$1"/>
>       </condition>
>     </extension>
>
> but after correctly executing this extension-statement it reverts to
> the original version of destination_number
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Kamil.Frydryszek at ilim.poznan.pl  Fri Dec  5 15:14:40 2014
From: Kamil.Frydryszek at ilim.poznan.pl (Kamil Frydryszek)
Date: Fri, 5 Dec 2014 13:14:40 +0100
Subject: [Freeswitch-users] BigBlueButton/FreeSwitch - external SIP account
 - Registration Failed with status Service Unavailable [503]
Message-ID: <OF53FE222D.BABCBDD5-ONC1257DA5.00420FAC-C1257DA5.00434335@ilim.poznan.pl>

Hello,

I use BigBlueButton, 0.81 and we must enable dial into conference via 
phone.
Several months ago everything was good (working), but now it  does not 
work ( I was working on thuesday for short time).
I use external SIP account and there is problem with registration .
There is error "2014-12-05 10:38:11.171892 [WARNING] sofia_reg.c:472 
Bezposredni Failed Registration [503], setting retry to 60 seconds."

I have just tried in different networks, on different servers and with two 
independent SIP operators.
The usual voip client (X-Lite) is working ok with SIP accounts.

I have attached a detailed debug log in file freeswitch.log.

Please give any suggestions. I am in very difficult situation, because it 
is my work and I must repair it, but after several days I have no idea.

Thank you in advance.

Best regards,
Kamil

____________________________________________________________________________________________________________________________________

Instytut Logistyki i Magazynowania                     www.ilim.poznan.pl 
| office at ilim.poznan.pl
ul. Estkowskiego 6 | 61-755 Pozna?, Poland | tel. +48 (61) 850 48 90 | fax 
+48 (61) 852 63 76 | REGON: 000018603 | NIP: 777-00-20-410
Zarejestrowany pod nr KRS 0000052866 - Wydzia? VIII Gospodarczy Krajowego 
Rejestru S?dowego Pozna? - Nowe Miasto i Wilda
_____________________________________________________________________________________________________________________________________

Instytut Logistyki i Magazynowania z siedzib? w Poznaniu dzia?a na 
podstawie ustawy z dnia 30 kwietnia 2010 r. o instytutach badawczych
(Dz. U. z 2010 r., Nr 96, poz. 618). Zgodnie z art. 24 ust. 1 punkt 5 
ustawy o instytutach badawczych oraz ? 4 ust. 2 
Statutu Instytutu Logistyki i Magazynowania, do reprezentowania Instytutu 
Logistyki i Magazynowania uprawniony jest jego dyrektor. 

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From vittico at gmail.com  Fri Dec  5 15:46:15 2014
From: vittico at gmail.com (Victor Medina)
Date: Fri, 5 Dec 2014 08:16:15 -0430
Subject: [Freeswitch-users] Building FS with mod_java support
Message-ID: <CAEA-bfHo6KOrMresGtG8oDpJGEOwQUp2f5=8PmrNsBD=D8M=WQ@mail.gmail.com>

Hi all!

I?ve been trying to build Freeswitch 1.4 with Java support. It
compiles just fine, installation fails with:

making install mod_java
make[4]: Entering directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make  install-am
make[5]: Entering directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[6]: Entering directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[6]: Nothing to be done for `install-exec-am'.
cp freeswitch.jar /opt/CloudVoice-vPBX/fs/scripts
cp: cannot stat ?freeswitch.jar?: No such file or directory
make[6]: *** [install-data-local] Error 1
make[6]: Leaving directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[5]: *** [install-am] Error 2
make[5]: Leaving directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[4]: *** [install] Error 2
make[4]: Leaving directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[3]: *** [mod_java-install] Error 1
make[3]: Leaving directory `/home/administrador/freeswitch/src/mod'
make[2]: *** [install-recursive] Error 1
make[2]: Leaving directory `/home/administrador/freeswitch/src'

Can someone help?

Sin mas a que hacer referencia,

Victor Medina


From vittico at gmail.com  Fri Dec  5 15:59:01 2014
From: vittico at gmail.com (Victor Medina)
Date: Fri, 5 Dec 2014 08:29:01 -0430
Subject: [Freeswitch-users] Building FS with mod_java support
In-Reply-To: <CAEA-bfHo6KOrMresGtG8oDpJGEOwQUp2f5=8PmrNsBD=D8M=WQ@mail.gmail.com>
References: <CAEA-bfHo6KOrMresGtG8oDpJGEOwQUp2f5=8PmrNsBD=D8M=WQ@mail.gmail.com>
Message-ID: <CAEA-bfEOagZ2s-5YFfwWuZzQe5jaj+Q9A1Gq21se+BSF4P=8Gg@mail.gmail.com>

More on the same issue...

root at ubuntu:/home/administrador/freeswitch# make mod_java-install
make[1]: Entering directory `/home/administrador/freeswitch'
 /bin/mkdir -p '/opt/CloudVoice-vPBX/fs/lib'
 /bin/bash /home/administrador/freeswitch/libtool   --mode=install
/usr/bin/install -c   libfreeswitch.la '/opt/CloudVoice-vPBX/fs/lib'
libtool: install: /usr/bin/install -c .libs/libfreeswitch.so.1.0.0
/opt/CloudVoice-vPBX/fs/lib/libfreeswitch.so.1.0.0
libtool: install: (cd /opt/CloudVoice-vPBX/fs/lib && { ln -s -f
libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f
libfreeswitch.so.1 && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so.1;
}; })
libtool: install: (cd /opt/CloudVoice-vPBX/fs/lib && { ln -s -f
libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f libfreeswitch.so &&
ln -s libfreeswitch.so.1.0.0 libfreeswitch.so; }; })
libtool: install: /usr/bin/install -c .libs/libfreeswitch.lai
/opt/CloudVoice-vPBX/fs/lib/libfreeswitch.la
libtool: install: /usr/bin/install -c .libs/libfreeswitch.a
/opt/CloudVoice-vPBX/fs/lib/libfreeswitch.a
libtool: install: chmod 644 /opt/CloudVoice-vPBX/fs/lib/libfreeswitch.a
libtool: install: ranlib /opt/CloudVoice-vPBX/fs/lib/libfreeswitch.a
libtool: finish:
PATH="/opt/CloudVoice-vPBX/jdk/bin/:/usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin:/usr/games:/usr/local/games:/sbin"
ldconfig -n /opt/CloudVoice-vPBX/fs/lib
----------------------------------------------------------------------
Libraries have been installed in:
   /opt/CloudVoice-vPBX/fs/lib

If you ever happen to want to link against installed libraries
in a given directory, LIBDIR, you must either use libtool, and
specify the full pathname of the library, or use the `-LLIBDIR'
flag during linking and do at least one of the following:
   - add LIBDIR to the `LD_LIBRARY_PATH' environment variable
     during execution
   - add LIBDIR to the `LD_RUN_PATH' environment variable
     during linking
   - use the `-Wl,-rpath -Wl,LIBDIR' linker flag
   - have your system administrator add LIBDIR to `/etc/ld.so.conf'

See any operating system documentation about shared libraries for
more information, such as the ld(1) and ld.so(8) manual pages.
----------------------------------------------------------------------
make[1]: Leaving directory `/home/administrador/freeswitch'
make[1]: Entering directory `/home/administrador/freeswitch/src/mod'

making install mod_java
make[2]: Entering directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make  install-am
make[3]: Entering directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
  CXX      mod_java_la-mod_java.lo
  CXX      mod_java_la-freeswitch_java.lo
freeswitch_java.cpp: In function 'void setOriginateStateHandler(jobject)':
freeswitch_java.cpp:17:85: warning: format not a string literal and no
format arguments [-Wformat-security]
                 switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_ERROR, errorMessage);

              ^
freeswitch_java.cpp:27:93: warning: format not a string literal and no
format arguments [-Wformat-security]
                         switch_log_printf(SWITCH_CHANNEL_LOG,
SWITCH_LOG_ERROR, errorMessage);

                      ^
  CXX      mod_java_la-switch_swig_wrap.lo
  CC       mod_java_la-modjava.lo
  CXXLD    mod_java.la
make[4]: Entering directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[4]: Nothing to be done for `install-exec-am'.
cp freeswitch.jar /opt/CloudVoice-vPBX/fs/scripts
cp: cannot stat ?freeswitch.jar?: No such file or directory
make[4]: *** [install-data-local] Error 1
make[4]: Leaving directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[3]: *** [install-am] Error 2
make[3]: Leaving directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[2]: *** [install] Error 2
make[2]: Leaving directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[1]: *** [mod_java-install] Error 1
make[1]: Leaving directory `/home/administrador/freeswitch/src/mod'
make: *** [mod_java-install] Error 2
root at ubuntu:/home/administrador/freeswitch#

Sin mas a que hacer referencia,

Victor Medina


On Fri, Dec 5, 2014 at 8:16 AM, Victor Medina <vittico at gmail.com> wrote:
> Hi all!
>
> I?ve been trying to build Freeswitch 1.4 with Java support. It
> compiles just fine, installation fails with:
>
> making install mod_java
> make[4]: Entering directory
> `/home/administrador/freeswitch/src/mod/languages/mod_java'
> make  install-am
> make[5]: Entering directory
> `/home/administrador/freeswitch/src/mod/languages/mod_java'
> make[6]: Entering directory
> `/home/administrador/freeswitch/src/mod/languages/mod_java'
> make[6]: Nothing to be done for `install-exec-am'.
> cp freeswitch.jar /opt/CloudVoice-vPBX/fs/scripts
> cp: cannot stat ?freeswitch.jar?: No such file or directory
> make[6]: *** [install-data-local] Error 1
> make[6]: Leaving directory
> `/home/administrador/freeswitch/src/mod/languages/mod_java'
> make[5]: *** [install-am] Error 2
> make[5]: Leaving directory
> `/home/administrador/freeswitch/src/mod/languages/mod_java'
> make[4]: *** [install] Error 2
> make[4]: Leaving directory
> `/home/administrador/freeswitch/src/mod/languages/mod_java'
> make[3]: *** [mod_java-install] Error 1
> make[3]: Leaving directory `/home/administrador/freeswitch/src/mod'
> make[2]: *** [install-recursive] Error 1
> make[2]: Leaving directory `/home/administrador/freeswitch/src'
>
> Can someone help?
>
> Sin mas a que hacer referencia,
>
> Victor Medina


From krice at freeswitch.org  Fri Dec  5 18:03:12 2014
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 05 Dec 2014 15:03:12 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <5481c930de4d_a5f1bc733449235@ip-10-58-73-253.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From acheraime at gmail.com  Fri Dec  5 18:14:10 2014
From: acheraime at gmail.com (Adolphe Cher-Aime)
Date: Fri, 5 Dec 2014 10:14:10 -0500
Subject: [Freeswitch-users] Doc-Sprint Friday 12 December 2014
In-Reply-To: <CAHaUJVbNaOzCUiwYN-ka5OCY=OsY9M2Ri_zKwyoRKrGuoGVwgw@mail.gmail.com>
References: <CAMf7LBiAKzHdsML1mGoPPn-ykKd8Zu72m2JrwMfka_Aedx-BDA@mail.gmail.com>
	<CAHaUJVbNaOzCUiwYN-ka5OCY=OsY9M2Ri_zKwyoRKrGuoGVwgw@mail.gmail.com>
Message-ID: <CA+fkitLK8w7h1a+kiQr=vQ-wt0oSGKaQ7q4HWhbep52FS65kMA@mail.gmail.com>

I'm in.


--
Adolphe
On Thu, Dec 4, 2014 at 3:00 PM, Iwada Eja <iwada.bassey at gmail.com> wrote:

> I'm In
>
> On Thu, Dec 4, 2014 at 1:20 PM, Areski <areski at gmail.com> wrote:
>
>> Hi everyone,
>>
>> We are planning to organize an other doc sprint on *Friday 12 December
>> at 10am CT*.
>> It will be 4 hours long but you can join for less time.
>>
>> The Doc-sprint will focus on migrating the remaining pages from MediaWiki
>> (https://wiki.freeswitch.org) to Confluence Wiki (
>> https://freeswitch.org/confluence).
>>
>> We will use an FS IRC channel during the sprint: *#freeswitch-docs*
>> and will track our work on the spreadsheet:
>> https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333
>>
>> During the sprint, please change the URL's "Status" you are working on to
>> "Editing" with your name next to it so we don't duplicate work.
>>
>> Some extra information:
>> - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration
>> -
>> https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation
>>
>> We hope to get a maximum number of people signed up!
>>
>> Peoples confirmed so far:
>> - Italo Rossi (+4)
>> - Areski Belaid
>>
>>
>> So, who is in?
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Kind Regards
> Iwada
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From notify.sina at gmail.com  Fri Dec  5 19:47:56 2014
From: notify.sina at gmail.com (Notify Me)
Date: Fri, 5 Dec 2014 17:47:56 +0100
Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound
 Caller ID to external SIP Gateway URI
In-Reply-To: <0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com>
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com>
	<0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com>
Message-ID: <CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com>

Hi Avi!

Thanks for the reply, I did figure it out, thanks! This is what I did,

<include>
 <extension name="firstsipgw-inbound">
  <condition field="destination_number"  expression="^0123450(\d{2})$"/>
  <condition field="caller_id_number"  expression="^0(\d{10})$"
require-nested="true">
     <action application="set" data="effective_caller_id_number=+234$1"/>
     <action application="set"
data="caller_id_number=+234${effective_caller_id_number}"/>
     <action application="set" data="transfer_ringback=$${us-ring}"/>
     <action application="set"
data="transfer_ringback=file_string://$${hold_music}"/>
     <action application="set"
data="transfer_ringback=local_stream://connecting"/>
       <action application="bridge"
data="sofia/external/sip:user.name at sip.othersipgw.com;${effective_caller_id_number}"/>
  </condition>
 </extension>
</include>

I can see from the logs that dials the othersipgw.com URI, and I can
see tcpdumps that corresponds to the traffic being sent,  and the
other side gets it. The other side immediately drops the call and does
some processing that returns an SMS to the dialler.

I dont know if I can trouble you for more help.. there is no ringing
when the transaction happens, and the dialler is unsure what is
happening, if the call actually connected. As you might have guessed
from the above I am trying to make it ring for a few seconds before
the call drops. Is this possible? I would like for the dialler to be
able to hear it ring a few times before the call is cut.

Any help gratefully accepted.

On Tue, Dec 2, 2014 at 2:22 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Hi - did you figure this out yet?
>
> One comment:
>  <action application="bridge" data="sofia/gateway/othersipgw/+234$1"/> -- I
> don't know if $1 is available anymore. You might want to just set that as
> part of the actual number to route, e.g. add a 1212 prefix and match it
> again or set it as a channel variable. You can see in the logs if the $1 is
> resolving correctly.
>
> -Avi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Dec  5 22:33:25 2014
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 5 Dec 2014 13:33:25 -0600
Subject: [Freeswitch-users] delay_echo with half of time
In-Reply-To: <039001d01015$17b4d8f0$471e8ad0$@inovax.com.br>
References: <039001d01015$17b4d8f0$471e8ad0$@inovax.com.br>
Message-ID: <CAKbxfG_zE0ZBLcF3Q4Kf3bHr4DPS_4fYY6zx6r44n_37p4Jw2w@mail.gmail.com>

Report BUGS to JIRA, Not the mailing list.
I fixed it for you but I cannot push the commit until you properly report
it.



On Thu, Dec 4, 2014 at 4:53 PM, Julio Cesar Esteves Cabezas <
jcabezas at inovax.com.br> wrote:

> Hi,
>
>                 The delay  I am getting from delay_echo is roughly half
> of what I pass as parameter in its data.
>
>
>
>                 This is the snippet, and the log shows on screen that I am
> passing what I intend to
>
>
>
>     <extension name="delay_echo">
>
>       <condition field="destination_number" expression="^\*7(\d)$">
>
>                 <action application="answer"/>
>
>                 <action application="set" data="delta=${expr(1+1000*$1)}"/>
>
>                 <action application="log" data=" *** delta = ${delta} "/>
>
>                 <action application="delay_echo" data="${delta}"/>
>
>       </condition>
>
>     </extension>
>
>
>
>                 I am running FreeSWITCH Version 1.5.12b~64bit ( 64bit)
>  (as per command ?version? in fs_cli)  in a VMware VM with OS=Windows
> Server 2012.
>
>
>
> 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
>



-- 
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
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From aqsyounas at gmail.com  Fri Dec  5 22:42:41 2014
From: aqsyounas at gmail.com (Aqs Younas)
Date: Sat, 6 Dec 2014 00:42:41 +0500
Subject: [Freeswitch-users] How to reset Freeswitch variable
	${read_terminator_used}
Message-ID: <CAGVOjVWdBm7Ary_vB-q=b49fsuG7OYfB+B=Z4sSr6a0hsvsRPg@mail.gmail.com>

Hi, All

I have a scenario in which next file is played when user press # and
previous file when user press *.

I want to reset ${read_terminator_used} after user presses either # or *.

How can i do so.?

Thanks.
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From t.mahe at b-and-c.net  Sat Dec  6 06:46:37 2014
From: t.mahe at b-and-c.net (=?windows-1252?Q?Tristan_Mah=E9?=)
Date: Fri, 05 Dec 2014 19:46:37 -0800
Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and 1.4.14
In-Reply-To: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail>
References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail>
Message-ID: <54827C1D.5000401@b-and-c.net>

Hi guys,

While trying to find out the root cause of this, I just want to know if
someone is also seeing a huge cpu load with freeswitch 1.4.14, which was
not present in 1.4.12.

OS is debian wheezy, packages are official ones.

Same config, same trafic pattern, just an upgrade from 1.4.12 ( no other
packages updated ).

We were previously seeing load around 0.5, with pikes up to around 2.

We now are seeing a constant load of around 2, and pikes up to 20...

If someone is also seeing this, that would be interesting to exchange,
helping find out what is causing this !

I'll file a jira once I find the root of this problem ( and no, this is
production servers, we can't put master there, we need stable ).

Best,

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From shabbirabbasi92 at gmail.com  Sat Dec  6 08:31:33 2014
From: shabbirabbasi92 at gmail.com (Shabbir abbasi)
Date: Sat, 6 Dec 2014 10:31:33 +0500
Subject: [Freeswitch-users] how to select 2nd row of sql query result in lua
Message-ID: <CAGpk5BTzeoPF4mDzXeN8bagmgqTfSSzWSk8Ktku9VOhFJZk9ew@mail.gmail.com>

if i execute a query in mysql select dialprefix,rateinitial  from
cc_ratecard where  (dialprefix='001' or dialprefix='00123')   ORDER BY
LENGTH(dialprefix) DESC;

result
row1    |  001 |     0.01000 |
row22  | 00123         |     0.02000 |

and this is   lua code

local rates = {}
local count = "";

     assert(dbh:query(query, function(qrow)
        for key, val in pairs(qrow) do
            rates[key] = val
freeswitch.consoleLog("INFO","    rates   :"..key.. "..="..val.." \n")
        end
count = qrow.count;
        end))

questios  is
1 how to select 2nd row from result  ?
2 how to count total rows   ?
3 how to print all 2 rows in console   ?

any suggestios wellcome
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From steveayre at gmail.com  Sat Dec  6 13:17:40 2014
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 6 Dec 2014 10:17:40 +0000
Subject: [Freeswitch-users] how to select 2nd row of sql query result in
	lua
In-Reply-To: <CAGpk5BTzeoPF4mDzXeN8bagmgqTfSSzWSk8Ktku9VOhFJZk9ew@mail.gmail.com>
References: <CAGpk5BTzeoPF4mDzXeN8bagmgqTfSSzWSk8Ktku9VOhFJZk9ew@mail.gmail.com>
Message-ID: <CAFiqYum320kNjN3zkC7_f9JyU6S8YYjZQ2rAjaKwHdzmRfZqxg@mail.gmail.com>

The function you pass to dbh:query is a callback. It is run for every row
in the result set, in order.

1) Will be qrow on 2nd invocation
2) 'count = count + 1' on each invocation
3) Print on all invocations. You should be seeing that already, so perhaps
the 2nd row isn't selected by your SQL query?

On 6 December 2014 at 05:31, Shabbir abbasi <shabbirabbasi92 at gmail.com>
wrote:

> if i execute a query in mysql select dialprefix,rateinitial  from
> cc_ratecard where  (dialprefix='001' or dialprefix='00123')   ORDER BY
> LENGTH(dialprefix) DESC;
>
> result
> row1    |  001 |     0.01000 |
> row22  | 00123         |     0.02000 |
>
> and this is   lua code
>
> local rates = {}
> local count = "";
>
>      assert(dbh:query(query, function(qrow)
>         for key, val in pairs(qrow) do
>             rates[key] = val
> freeswitch.consoleLog("INFO","    rates   :"..key.. "..="..val.." \n")
>         end
> count = qrow.count;
>         end))
>
> questios  is
> 1 how to select 2nd row from result  ?
> 2 how to count total rows   ?
> 3 how to print all 2 rows in console   ?
>
> any suggestios wellcome
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From notify.sina at gmail.com  Sun Dec  7 14:47:24 2014
From: notify.sina at gmail.com (Notify Me)
Date: Sun, 7 Dec 2014 12:47:24 +0100
Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound
 Caller ID to external SIP Gateway URI
In-Reply-To: <0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com>
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com>
	<0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com>
	<CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com>
	<0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com>
Message-ID: <CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com>

The delay is very short, they send a100 Trying  and then a 480 temporarily
Unavailable back, as I can see in tcpdump. After they send the fail
and its acknowledged, they end it and send an SMS back.
This is a full dump

12:41:27.169928 IP (tos 0x0, ttl 64, id 34092, offset 0, flags [none],
proto UDP (17), length 1146)
    10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 1118
        INVITE sip:sina.nowahala at sip.othersipgw.com;+234diallednumber SIP/2.0
        Via: SIP/2.0/UDP
my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q
        Max-Forwards: 5
        From: "234diallednumber"
<sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
        To: <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>
        Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
        CSeq: 68640179 INVITE
        Contact: <sip:mod_sofia at my.public.ip.address:5080>
        User-Agent:
FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~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: 247
        X-FS-Support: update_display,send_info
        Remote-Party-ID: "234diallednumber"
<sip:+234diallednumber at my.public.ip.address>;party=calling;screen=yes;privacy=off

        v=0
        o=FreeSWITCH 1417930197 1417930198 IN IP4 my.public.ip.address
        s=FreeSWITCH
        c=IN IP4 my.public.ip.address
        t=0 0
        m=audio 22290 RTP/AVP 8 0 101 13
        a=rtpmap:8 PCMA/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20

12:41:27.385419 IP (tos 0x0, ttl 53, id 5736, offset 0, flags [none],
proto UDP (17), length 384)
    othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 356
        SIP/2.0 100 Trying
        Via: SIP/2.0/UDP
my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q
        From: "234diallednumber"
<sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
        To: <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>
        Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
        CSeq: 68640179 INVITE
        User-Agent: FreeSWITCH-mod_sofia/1.2.14
        Content-Length: 0


12:41:27.424422 IP (tos 0x0, ttl 53, id 5737, offset 0, flags [none],
proto UDP (17), length 1015)
    othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 987
        SIP/2.0 480 Temporarily Unavailable
        Via: SIP/2.0/UDP
my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q
        Max-Forwards: 4
        From: "234diallednumber"
<sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
        To: <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>;tag=8c8DZK1aK9mKN
        Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
        CSeq: 68640179 INVITE
        User-Agent: FreeSWITCH-mod_sofia/1.2.14
        Accept: application/sdp
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
        Supported: timer, precondition, path, replaces
        Allow-Events: talk, hold, conference, presence, dialog,
line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer
        Reason: Q.850;cause=16;text="NORMAL_CLEARING"
        Content-Length: 0
        X-FS-Display-Name: sina.nowahala
        X-FS-Display-Number: sip:sina.nowahala at sip.othersipgw.com
        Remote-Party-ID: "sina.nowahala"
<sip:sina.nowahala at sip.othersipgw.com>;party=calling;privacy=off;screen=no


12:41:27.424789 IP (tos 0x0, ttl 64, id 34093, offset 0, flags [none],
proto UDP (17), length 412)
    10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 384
        ACK sip:sina.nowahala at sip.othersipgw.com;+234diallednumber SIP/2.0
        Via: SIP/2.0/UDP
my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q
        Max-Forwards: 5
        From: "234diallednumber"
<sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
        To: <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>;tag=8c8DZK1aK9mKN
        Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
        CSeq: 68640179 ACK
        Content-Length: 0


So you recommend a transfer on fail wav to be played? I have been
tearing my hair out trying to play sounds before the call is
transferred to them, so that the dialler hears something.

On Sun, Dec 7, 2014 at 4:44 AM, Avi Marcus <avi at avimarcus.net> wrote:
> How long is the delay? I assume they return a normal_clearing or fail 4XX?
> If the delay is short and they have a "real" status return, then.. for
> success, you can just queue a "success.wav" to play after.
> And for fails, you can use
> https://wiki.freeswitch.org/wiki/Variable_transfer_on_fail and transfer to
> an extension that plays an error message.
>
> This SMS sending sounds like a normal API call.. which if it was, could be
> done with curl or inside a lua/js script.
>
> -Avi
>
> On Fri, Dec 5, 2014 at 6:47 PM, Notify Me <notify.sina at gmail.com> wrote:
>>
>> Hi Avi!
>>
>> Thanks for the reply, I did figure it out, thanks! This is what I did,
>>
>> <include>
>>  <extension name="firstsipgw-inbound">
>>   <condition field="destination_number"  expression="^0123450(\d{2})$"/>
>>   <condition field="caller_id_number"  expression="^0(\d{10})$"
>> require-nested="true">
>>      <action application="set" data="effective_caller_id_number=+234$1"/>
>>      <action application="set"
>> data="caller_id_number=+234${effective_caller_id_number}"/>
>>      <action application="set" data="transfer_ringback=$${us-ring}"/>
>>      <action application="set"
>> data="transfer_ringback=file_string://$${hold_music}"/>
>>      <action application="set"
>> data="transfer_ringback=local_stream://connecting"/>
>>        <action application="bridge"
>>
>> data="sofia/external/sip:user.name at sip.othersipgw.com;${effective_caller_id_number}"/>
>>   </condition>
>>  </extension>
>> </include>
>>
>> I can see from the logs that dials the othersipgw.com URI, and I can
>> see tcpdumps that corresponds to the traffic being sent,  and the
>> other side gets it. The other side immediately drops the call and does
>> some processing that returns an SMS to the dialler.
>>
>> I dont know if I can trouble you for more help.. there is no ringing
>> when the transaction happens, and the dialler is unsure what is
>> happening, if the call actually connected. As you might have guessed
>> from the above I am trying to make it ring for a few seconds before
>> the call drops. Is this possible? I would like for the dialler to be
>> able to hear it ring a few times before the call is cut.
>>
>> Any help gratefully accepted.
>>
>> On Tue, Dec 2, 2014 at 2:22 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> > Hi - did you figure this out yet?
>> >
>> > One comment:
>> >  <action application="bridge" data="sofia/gateway/othersipgw/+234$1"/>
>> > -- I
>> > don't know if $1 is available anymore. You might want to just set that
>> > as
>> > part of the actual number to route, e.g. add a 1212 prefix and match it
>> > again or set it as a channel variable. You can see in the logs if the $1
>> > is
>> > resolving correctly.
>> >
>> > -Avi
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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 luis.daniel.lucio at gmail.com  Sun Dec  7 17:49:51 2014
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sun, 7 Dec 2014 09:49:51 -0500
Subject: [Freeswitch-users] Recommendation for Billing for Hosted PBX
	Platform
In-Reply-To: <CAHgk4BWcJ0i4piXn7tgzWsKtNcLwfrsv-qrX45J6i=6LRA3Mbg@mail.gmail.com>
References: <519A0253.3060009@gmail.com>
	<CAHKQagH_R7vNK14Pntb1GgbhV6EHULzDUp9Nff12F8c11TMzLw@mail.gmail.com>
	<CAHgk4BWcJ0i4piXn7tgzWsKtNcLwfrsv-qrX45J6i=6LRA3Mbg@mail.gmail.com>
Message-ID: <CAFLo2QxioW2Gguf-1tHB0ZUOTqA_r9zNe6yNwKqJav9xzAF2uw@mail.gmail.com>

Maybe a little late,

But yes, Deon, you are right, FusionPBX is now near to be the best
frontend for Freeswitch.  You can do almost anything with it.

Gladly, after some years of using FusionPBX I decide to develop a
native Billing application for FusionPBX with Freeswitch.  It
integrates with FusionPBX in many ways, not an external instance of
freeswitch, but inside the FusionPBX itself.

If you are interested, you can go to my website www.okay.com.mx/en  or
contact me (gtalk prefered)

2013-05-20 12:43 GMT-04:00 Brian Foster <bdfoster at davri.com>:
> Jbilling uses CDRs and events to do mediation. So using your guide, you
> should have an idea of what info is required from freeswitch. If you can
> give us some examples of what jbilling is asking for and we can give you
> ideas. Please start a new thread specifically for your jbilling issues.
>
> Thanks,
>
> - BDF
>
> On May 20, 2013 10:02 AM, "Cal Leeming [Simplicity Media Ltd]"
> <cal.leeming at simplicitymedialtd.co.uk> wrote:
>>
>> Take a look at chargify.com - they are extremely good.
>>
>> Cal
>>
>> On Mon, May 20, 2013 at 12:00 PM, Deon Vermeulen
>> <vermeulen.deon at gmail.com> wrote:
>>>
>>> Hi,
>>>
>>>
>>> I've been looking at the Kazoo as a very viable option as a HostedPBX
>>> platform, but don't find any information with regards to Billing.
>>>
>>> I've also been tracking the thread about recommended GUI where 2600Hz are
>>> in the process for updated the code in git for Blue.box.
>>> This also looks like a possibility for a HostedPBX platform, but then
>>> again I have no idea about the Billing software to use for this.
>>>
>>> FusionPBX is a top contender if no the number 1 for a HostedPBX platform
>>> but again information for Billing is lacking.
>>>
>>> I would like to know what the feel out there is for Billing on a
>>> HostedPBX Platform, i.e. Cloud.
>>>
>>> If would be great if there is an opensource Billing system.
>>>
>>> I've purchased the jBilling Telecom guide, but no where can I find
>>> information about integrating with FreeSWITCH nor implementing it for a
>>> HostedPBX platform in the Cloud.
>>>
>>>
>>> Kind Regards
>>>
>>>
>>> _________________________________________________________________________
>>> 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
>


From luis.daniel.lucio at gmail.com  Sun Dec  7 17:58:04 2014
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Sun, 7 Dec 2014 09:58:04 -0500
Subject: [Freeswitch-users] Billing software
In-Reply-To: <518CA7DD.8020802@gmail.com>
References: <CAFLo2QzXJxyEDQ5YeNB3Engu4_LcTbdFELfXGhzgVChS4UoVbw@mail.gmail.com>
	<E84CC2DA-E8F2-4B02-8F09-B60D4406407E@gmail.com>
	<CAFLo2QyynMEY=NzJCZYmVAGFjiwNg7MgSGFrWCTWJYpyRnyCRQ@mail.gmail.com>
	<518CA7DD.8020802@gmail.com>
Message-ID: <CAFLo2Qyd1uq+TOhc_sX5hahDku_DTHrQXdH3hXC+H5QFRLi4mg@mail.gmail.com>

Maybe a little late,

Thanks everyone, but I am really against to install an extra freeswitch
instance for the billing. It makes very complex the call flow,
troubleshoting and in most cases FAX wont work.

I did try the ASTPP before, horrible, hard to configure. I consider myself
good on freeswitch, i know to code on perl, and a good sysadmin.  But after
watching how complex is to get this to work I abort.

So I decide to code my own billing and gladly, I can tell you that if you
have all information needed (such as carrier rates and about 15 mins to get
an explanation on how it works, you can have it working in less than an
hour).  I started to study FusionPBX code, and after some months I have a
working nice FusionPBX LCR+Billing native application.  It fully integrates
in the Fusion, not out side, so you are able to bill for anything,
incoming, internal (ext to ext, ext to conference, ext to ivr), or outgoing
calls and it is able to get money by Paypal, Stripe (a good gw for
CreditCards) and offline.  Currently, I have just release version 1.0.2

If someone is interested, you can go to my website www.okay.com.mx/en  or
contact me (gtalk prefered)

LD

2013-05-10 3:55 GMT-04:00 Deon Vermeulen <vermeulen.deon at gmail.com>:

> Hi Luis
>
> We just bought Commercial services from ASTPP.
> When it comes to comparing price, features this is the BEST solution out
> there at the moment.
>
> This is a very active project and you can contact Samir Doshi, Project
> Maintainer & Developer,  directly for more information wrt your technical
> questions.
>
> samir at astpp.org
>
>
> Kind Regards
> Deon
>
>  Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com>
>  May 9, 2013 11:46 PM
>
>
> Deon, thank you. Can you talk me about your exerience and how fast they
> are on fixing bugs. I have a really good or bad luck to find bugs.
> _________________________________________________________________________
> 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
>  Deon Vermeulen <vermeulen.deon at gmail.com>
>  May 5, 2013 4:52 PM
> Check out ASTPP
>
>
> Kind Regards
> Deon Vermeulen
>
> Sent from my iPhone
>
>  Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com>
>  May 5, 2013 3:36 PM
> What other options for FS compatible software other than vBilling do you
> recommend me?
> _________________________________________________________________________
> 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
>
>
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From avi at avimarcus.net  Sun Dec  7 18:12:22 2014
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 7 Dec 2014 15:12:22 +0000
Subject: [Freeswitch-users] Recommendation for Billing for Hosted PBX
	Platform
In-Reply-To: <CAHKQagH_R7vNK14Pntb1GgbhV6EHULzDUp9Nff12F8c11TMzLw@mail.gmail.com>
References: <519A0253.3060009@gmail.com>
	<CAHKQagH_R7vNK14Pntb1GgbhV6EHULzDUp9Nff12F8c11TMzLw@mail.gmail.com>
Message-ID: <0000014a254f0100-ddeb3eb5-c532-4c5e-9507-1e199a208bbf-000000@email.amazonses.com>

BTW - Stripe has recurring billing and the ability to create add-ons for
the monthly bills, via API.
They don't charge anything extra for these features beyond the normal
credit card processing fees.
-Avi

On Mon, May 20, 2013 at 4:54 PM, Cal Leeming [Simplicity Media Ltd] <
cal.leeming at simplicitymedialtd.co.uk> wrote:

> Take a look at chargify.com - they are extremely good.
>
> Cal
>
>
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From andrew at cassidywebservices.co.uk  Sun Dec  7 18:12:53 2014
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sun, 7 Dec 2014 15:12:53 +0000
Subject: [Freeswitch-users] remove country code from destination_number
In-Reply-To: <CACOYK=jo9Tu4LY0zG7PTpwfjmgkmUAwgsHKE2R5P_hO8YiuHPw@mail.gmail.com>
References: <CABR18gdjXQXRJTP2TxWmB6RtKSxd=pLh44WxPBpzSPyJWhLORA@mail.gmail.com>
	<CACOYK=jo9Tu4LY0zG7PTpwfjmgkmUAwgsHKE2R5P_hO8YiuHPw@mail.gmail.com>
Message-ID: <CAMbW9oLsMyQbyE=b6-x3oUqTAjwqBux=uWwH0GBuwdsmtsB_RQ@mail.gmail.com>

mod_translate

On 5 December 2014 at 11:52, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> you just need to use a different variable to store the modified
> destination number.
> here are a couple of examples from my blog:
>
>
> http://txlab.wordpress.com/2014/10/21/using-voxbeam-for-outbound-calls-with-freeswitch/
>
> http://txlab.wordpress.com/2013/08/03/free-us-number-and-caller-id-manipulation/
>
>
>
>
> On Fri, Dec 5, 2014 at 11:22 AM, Paul Klenk <markus.klenk at googlemail.com>
> wrote:
> > Hi,
> >
> > I'd like to have the country code "+49" removed from
> > destination_number and changed to "0"
> >
> > I tried:
> >     <extension name="international" continue="true">
> >       <condition field="destination_number" expression="^\+49(.*)">
> >         <action inline="true" application="set"
> data="destination_number=0$1"/>
> >       </condition>
> >     </extension>
> >
> > but after correctly executing this extension-statement it reverts to
> > the original version of destination_number
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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


*T <info at cassidywebservices.co.uk> *03300 100 960  *F
<info at cassidywebservices.co.uk> *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From andrew.keil at visytel.com  Mon Dec  8 05:17:45 2014
From: andrew.keil at visytel.com (Andrew Keil)
Date: Mon, 8 Dec 2014 02:17:45 +0000
Subject: [Freeswitch-users] Re- LUA audio playback on Windows 64bit not
	working?
Message-ID: <HKXPR04MB0456465ADBAF1B37E0883CA7F3640@HKXPR04MB0456.apcprd04.prod.outlook.com>

To FreeSWITCH users,

I am just starting to use Lua inside FreeSWITCH and I am having difficulty with playing back audio to the caller.  It seems the Media is not ready or available even though I have answered the caller using session:answer().
It also seems that session:ready() always returns false after session:answer().  Very strange.

Version: FreeSWITCH 1.5.12b(64bit) on Windows 8.1 Pro  (installed from freeswitch.msi) - default installation except for conf\vars.xml (default_password changed)

conf\dialplan\default\00_ASKTestLua.xml:
<include>
    <!-- To test Lua script  -->
    <extension name="ivr_demo">
      <condition field="destination_number" expression="^(9101)$">
                <!-- <action application="answer"/> -->
                <!-- <action application="sleep" data="1000"/> -->
<!-- <action application="playback" data="ivr/ivr-welcome_to_freeswitch.wav"/> -->
                <action inline="true" application="lua" data="ASKTestLua.lua $2"/>
      </condition>
    </extension>
</include>

scripts\ASKTestLua.lua:
-- Test Lua application

-- answer the call
session:answer();

session:sleep(1000);

prompt = "ivr/ivr-welcome_to_freeswitch.wav";
freeswitch.consoleLog("INFO", string.format("prompt: %s\n", prompt));

session:execute("playback",prompt);
--session:streamFile(prompt);

freeswitch.consoleLog("INFO", "Prompt played\n")

-- hangup
session:hangup();


Logs inside FS_CLI: 2014-12-08 12:44:15.002800 [DEBUG] switch_core_session.c:2725 Application playback Requires media! pre_answering channel sofia/internal/1000 at 192.168.15.74
EXECUTE sofia/internal/1000 at 192.168.15.74 playback(ivr/ivr-welcome_to_freeswitch.wav)

vs.

conf\dialplan\default\00_ASKTestLua.xml:
<include>
    <!-- To test Lua script  -->
    <extension name="ivr_demo">
      <condition field="destination_number" expression="^(9101)$">
                <action application="answer"/>
                <action application="sleep" data="1000"/>
<action application="playback" data="ivr/ivr-welcome_to_freeswitch.wav"/>
                <!-- <action inline="true" application="lua" data="ASKTestLua.lua $2"/> -->
      </condition>
    </extension>
</include>

Which works!!  However this is not using Lua.

I have tried using "session:streamFile(prompt);" instead of "session:execute("playback",prompt);" and this also does not work.

I would appreciate any insight into why this is happening.

Kind Regards,

Andrew Keil
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From andrew.keil at visytel.com  Mon Dec  8 09:12:59 2014
From: andrew.keil at visytel.com (Andrew Keil)
Date: Mon, 8 Dec 2014 06:12:59 +0000
Subject: [Freeswitch-users] Re- LUA audio playback on Windows 64bit not
	working? (FOUND & FIXED)
Message-ID: <HKXPR04MB04568FD4BE213278E491EAFFF3640@HKXPR04MB0456.apcprd04.prod.outlook.com>

To all,

Found the issue below:

Removed the inline="true" from the xml dialplan and everything works as expected.

<action inline="true" application="lua" data="ASKTestLua.lua $2"/>

Now:

<action application="lua" data="ASKTestLua.lua $2"/>

Not sure why I put it there in the first place.

Sorry about that.

Andrew

From: Andrew Keil
Sent: Monday, 8 December 2014 1:18 PM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Re- LUA audio playback on Windows 64bit not working?

To FreeSWITCH users,

I am just starting to use Lua inside FreeSWITCH and I am having difficulty with playing back audio to the caller.  It seems the Media is not ready or available even though I have answered the caller using session:answer().
It also seems that session:ready() always returns false after session:answer().  Very strange.

Version: FreeSWITCH 1.5.12b(64bit) on Windows 8.1 Pro  (installed from freeswitch.msi) - default installation except for conf\vars.xml (default_password changed)

conf\dialplan\default\00_ASKTestLua.xml:
<include>
    <!-- To test Lua script  -->
    <extension name="ivr_demo">
      <condition field="destination_number" expression="^(9101)$">
                <!-- <action application="answer"/> -->
                <!-- <action application="sleep" data="1000"/> -->
<!-- <action application="playback" data="ivr/ivr-welcome_to_freeswitch.wav"/> -->
                <action inline="true" application="lua" data="ASKTestLua.lua $2"/>
      </condition>
    </extension>
</include>

scripts\ASKTestLua.lua:
-- Test Lua application

-- answer the call
session:answer();

session:sleep(1000);

prompt = "ivr/ivr-welcome_to_freeswitch.wav";
freeswitch.consoleLog("INFO", string.format("prompt: %s\n", prompt));

session:execute("playback",prompt);
--session:streamFile(prompt);

freeswitch.consoleLog("INFO", "Prompt played\n")

-- hangup
session:hangup();


Logs inside FS_CLI: 2014-12-08 12:44:15.002800 [DEBUG] switch_core_session.c:2725 Application playback Requires media! pre_answering channel sofia/internal/1000 at 192.168.15.74<mailto:sofia/internal/1000 at 192.168.15.74>
EXECUTE sofia/internal/1000 at 192.168.15.74<mailto:sofia/internal/1000 at 192.168.15.74> playback(ivr/ivr-welcome_to_freeswitch.wav)

vs.

conf\dialplan\default\00_ASKTestLua.xml:
<include>
    <!-- To test Lua script  -->
    <extension name="ivr_demo">
      <condition field="destination_number" expression="^(9101)$">
                <action application="answer"/>
                <action application="sleep" data="1000"/>
<action application="playback" data="ivr/ivr-welcome_to_freeswitch.wav"/>
                <!-- <action inline="true" application="lua" data="ASKTestLua.lua $2"/> -->
      </condition>
    </extension>
</include>

Which works!!  However this is not using Lua.

I have tried using "session:streamFile(prompt);" instead of "session:execute("playback",prompt);" and this also does not work.

I would appreciate any insight into why this is happening.

Kind Regards,

Andrew Keil
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From bordmi at rarus.ru  Mon Dec  8 10:14:54 2014
From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=)
Date: Mon, 8 Dec 2014 11:14:54 +0400
Subject: [Freeswitch-users] Re- LUA audio playback on Windows 64bit not
 working? (FOUND & FIXED)
In-Reply-To: <HKXPR04MB04568FD4BE213278E491EAFFF3640@HKXPR04MB0456.apcprd04.prod.outlook.com>
References: <HKXPR04MB04568FD4BE213278E491EAFFF3640@HKXPR04MB0456.apcprd04.prod.outlook.com>
Message-ID: <CAL2h29gPbfxP-Pr+VuSVSRK7rnSXH7PpQApGN36NqCwopQkXQw@mail.gmail.com>

In the hunting state of dialplan treatment FS doesn`t permit to work with
media, only in execution state.

2014-12-08 9:12 GMT+03:00 Andrew Keil <andrew.keil at visytel.com>:

>  To all,
>
>
>
> Found the issue below:
>
>
>
> Removed the inline=?true? from the xml dialplan and everything works as
> expected.
>
>
>
> <action inline="true" application="lua" data="ASKTestLua.lua $2"/>
>
>
>
> Now:
>
>
>
> <action application="lua" data="ASKTestLua.lua $2"/>
>
>
>
> Not sure why I put it there in the first place.
>
>
>
> Sorry about that.
>
>
>
> Andrew
>
>
>
> *From:* Andrew Keil
> *Sent:* Monday, 8 December 2014 1:18 PM
> *To:* 'freeswitch-users at lists.freeswitch.org'
> *Subject:* Re- LUA audio playback on Windows 64bit not working?
>
>
>
> To FreeSWITCH users,
>
>
>
> I am just starting to use Lua inside FreeSWITCH and I am having difficulty
> with playing back audio to the caller.  It seems the Media is not ready or
> available even though I have answered the caller using session:answer().
>
> It also seems that session:ready() always returns false after
> session:answer().  Very strange.
>
>
>
> Version: FreeSWITCH 1.5.12b(64bit) on Windows 8.1 Pro  (installed from
> freeswitch.msi) ? default installation except for conf\vars.xml
> (default_password changed)
>
>
>
> *conf\dialplan\default\00_ASKTestLua.xml:*
>
> <include>
>
>     <!-- To test Lua script  -->
>
>     <extension name="ivr_demo">
>
>       <condition field="destination_number" expression="^(9101)$">
>
>                 <!-- <action application="answer"/> -->
>
>                 <!-- <action application="sleep" data="1000"/>
> -->
>
> <!-- <action application="playback"
> data="ivr/ivr-welcome_to_freeswitch.wav"/> -->
>
>                 <action inline="true" application="lua"
> data="ASKTestLua.lua $2"/>
>
>       </condition>
>
>     </extension>
>
> </include>
>
>
>
> *scripts\ASKTestLua.lua:*
>
> -- Test Lua application
>
>
>
> -- answer the call
>
> session:answer();
>
>
>
> session:sleep(1000);
>
>
>
> prompt = "ivr/ivr-welcome_to_freeswitch.wav";
>
> freeswitch.consoleLog("INFO", string.format("prompt: %s\n", prompt));
>
>
>
> session:execute("playback",prompt);
>
> --session:streamFile(prompt);
>
>
>
> freeswitch.consoleLog("INFO", "Prompt played\n")
>
>
>
> -- hangup
>
> session:hangup();
>
>
>
>
>
> Logs inside FS_CLI: 2014-12-08 12:44:15.002800 [DEBUG]
> switch_core_session.c:2725 Application playback Requires media!
> pre_answering channel sofia/internal/1000 at 192.168.15.74
>
> EXECUTE sofia/internal/1000 at 192.168.15.74
> playback(ivr/ivr-welcome_to_freeswitch.wav)
>
>
>
> vs.
>
>
>
> conf\dialplan\default\00_ASKTestLua.xml:
>
> <include>
>
>     <!-- To test Lua script  -->
>
>     <extension name="ivr_demo">
>
>       <condition field="destination_number" expression="^(9101)$">
>
>                 <action application="answer"/>
>
>                 <action application="sleep" data="1000"/>
>
> <action application="playback" data="ivr/ivr-welcome_to_freeswitch.wav"/>
>
>                 <!-- <action inline="true" application="lua"
> data="ASKTestLua.lua $2"/> -->
>
>       </condition>
>
>     </extension>
>
> </include>
>
>
>
> Which works!!  However this is not using Lua.
>
>
>
> I have tried using ?session:streamFile(prompt);? instead of
> ?session:execute("playback",prompt);? and this also does not work.
>
>
>
> I would appreciate any insight into why this is happening.
>
>
>
> Kind Regards,
>
>
>
> Andrew Keil
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
with best regrds,
Dmitriy Borisov
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From bordmi at rarus.ru  Mon Dec  8 10:17:12 2014
From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=)
Date: Mon, 8 Dec 2014 11:17:12 +0400
Subject: [Freeswitch-users] Re- LUA audio playback on Windows 64bit not
 working? (FOUND & FIXED)
In-Reply-To: <CAL2h29gPbfxP-Pr+VuSVSRK7rnSXH7PpQApGN36NqCwopQkXQw@mail.gmail.com>
References: <HKXPR04MB04568FD4BE213278E491EAFFF3640@HKXPR04MB0456.apcprd04.prod.outlook.com>
	<CAL2h29gPbfxP-Pr+VuSVSRK7rnSXH7PpQApGN36NqCwopQkXQw@mail.gmail.com>
Message-ID: <CAL2h29jofYbZe1S1V0Z=iB+05fXg6cRhP+KgF3vuw=pikeGvmA@mail.gmail.com>

But "inline" directive force FS to run script on hunting phase

My yesturday`s sunday have died with this issue... :(

2014-12-08 10:14 GMT+03:00 ???????, ??????? <bordmi at rarus.ru>:

> In the hunting state of dialplan treatment FS doesn`t permit to work with
> media, only in execution state.
>
> 2014-12-08 9:12 GMT+03:00 Andrew Keil <andrew.keil at visytel.com>:
>
>>  To all,
>>
>>
>>
>> Found the issue below:
>>
>>
>>
>> Removed the inline=?true? from the xml dialplan and everything works as
>> expected.
>>
>>
>>
>> <action inline="true" application="lua" data="ASKTestLua.lua $2"/>
>>
>>
>>
>> Now:
>>
>>
>>
>> <action application="lua" data="ASKTestLua.lua $2"/>
>>
>>
>>
>> Not sure why I put it there in the first place.
>>
>>
>>
>> Sorry about that.
>>
>>
>>
>> Andrew
>>
>>
>>
>> *From:* Andrew Keil
>> *Sent:* Monday, 8 December 2014 1:18 PM
>> *To:* 'freeswitch-users at lists.freeswitch.org'
>> *Subject:* Re- LUA audio playback on Windows 64bit not working?
>>
>>
>>
>> To FreeSWITCH users,
>>
>>
>>
>> I am just starting to use Lua inside FreeSWITCH and I am having
>> difficulty with playing back audio to the caller.  It seems the Media is
>> not ready or available even though I have answered the caller using
>> session:answer().
>>
>> It also seems that session:ready() always returns false after
>> session:answer().  Very strange.
>>
>>
>>
>> Version: FreeSWITCH 1.5.12b(64bit) on Windows 8.1 Pro  (installed from
>> freeswitch.msi) ? default installation except for conf\vars.xml
>> (default_password changed)
>>
>>
>>
>> *conf\dialplan\default\00_ASKTestLua.xml:*
>>
>> <include>
>>
>>     <!-- To test Lua script  -->
>>
>>     <extension name="ivr_demo">
>>
>>       <condition field="destination_number" expression="^(9101)$">
>>
>>                 <!-- <action application="answer"/> -->
>>
>>                 <!-- <action application="sleep" data="1000"/>
>> -->
>>
>> <!-- <action application="playback"
>> data="ivr/ivr-welcome_to_freeswitch.wav"/> -->
>>
>>                 <action inline="true" application="lua"
>> data="ASKTestLua.lua $2"/>
>>
>>       </condition>
>>
>>     </extension>
>>
>> </include>
>>
>>
>>
>> *scripts\ASKTestLua.lua:*
>>
>> -- Test Lua application
>>
>>
>>
>> -- answer the call
>>
>> session:answer();
>>
>>
>>
>> session:sleep(1000);
>>
>>
>>
>> prompt = "ivr/ivr-welcome_to_freeswitch.wav";
>>
>> freeswitch.consoleLog("INFO", string.format("prompt: %s\n", prompt));
>>
>>
>>
>> session:execute("playback",prompt);
>>
>> --session:streamFile(prompt);
>>
>>
>>
>> freeswitch.consoleLog("INFO", "Prompt played\n")
>>
>>
>>
>> -- hangup
>>
>> session:hangup();
>>
>>
>>
>>
>>
>> Logs inside FS_CLI: 2014-12-08 12:44:15.002800 [DEBUG]
>> switch_core_session.c:2725 Application playback Requires media!
>> pre_answering channel sofia/internal/1000 at 192.168.15.74
>>
>> EXECUTE sofia/internal/1000 at 192.168.15.74
>> playback(ivr/ivr-welcome_to_freeswitch.wav)
>>
>>
>>
>> vs.
>>
>>
>>
>> conf\dialplan\default\00_ASKTestLua.xml:
>>
>> <include>
>>
>>     <!-- To test Lua script  -->
>>
>>     <extension name="ivr_demo">
>>
>>       <condition field="destination_number" expression="^(9101)$">
>>
>>                 <action application="answer"/>
>>
>>                 <action application="sleep" data="1000"/>
>>
>> <action application="playback" data="ivr/ivr-welcome_to_freeswitch.wav"/>
>>
>>                 <!-- <action inline="true" application="lua"
>> data="ASKTestLua.lua $2"/> -->
>>
>>       </condition>
>>
>>     </extension>
>>
>> </include>
>>
>>
>>
>> Which works!!  However this is not using Lua.
>>
>>
>>
>> I have tried using ?session:streamFile(prompt);? instead of
>> ?session:execute("playback",prompt);? and this also does not work.
>>
>>
>>
>> I would appreciate any insight into why this is happening.
>>
>>
>>
>> Kind Regards,
>>
>>
>>
>> Andrew Keil
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> with best regrds,
> Dmitriy Borisov
>



-- 
with best regrds,
Dmitriy Borisov
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From mbike2000ru at yahoo.com  Mon Dec  8 10:28:41 2014
From: mbike2000ru at yahoo.com (Dmitry)
Date: Mon, 8 Dec 2014 07:28:41 +0000 (UTC)
Subject: [Freeswitch-users] outgoing speech path is broken sometimes in
	1.4.14
Message-ID: <1071887578.4011778.1418023721454.JavaMail.yahoo@jws100167.mail.ne1.yahoo.com>

Hi
I installed FS 1.4.14 from git
Now I test outgoing calls with Radius authorization.
The problem that I encountered is that sometimes there is no RTP from Freeswitch. I see it in tcpdump dump.
Iptables ?-no restrictions.
I also tested when there is no mod_rad_auth loaded (I commented it out in autoload_configs/modules.conf.xml). - the same result
And when I put the call on hold - then when I return - the call path is normal.
Does anyone encountered the same behaviour? The version?FreeSWITCH Version 1.4.14+git~20141119T221113Z~ca1d990cfc~64bit (git ca1d990 2014-11-19 22:11:13Z 64bit)
P.S. I am new to FS. I came from Asterisk world.
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From den.pavlovsky at gmail.com  Mon Dec  8 10:40:40 2014
From: den.pavlovsky at gmail.com (Den Pavlovsky)
Date: Mon, 8 Dec 2014 13:40:40 +0600
Subject: [Freeswitch-users] outgoing speech path is broken sometimes in
	1.4.14
Message-ID: <CAAfQe6vHGjCNpQSkxZjk+BORsjwsLNjwGbkMK9D5z3wPTEi+gA@mail.gmail.com>

Hi

I installed FS 1.4.14 from git

Now I test outgoing calls with Radius authorization.

The problem that I encountered is that sometimes there is no RTP from
Freeswitch. I see it in tcpdump dump.

Iptables  -no restrictions.

I also tested when there is no mod_rad_auth loaded (I commented it out in
autoload_configs/modules.conf.xml). - the same result

And when I put the call on hold - then when I return - the call path is
normal.

Any help on how to debug this issue is appreciated.

Does anyone encountered the same behaviour? The version FreeSWITCH Version
1.4.14+git~20141119T221113Z~ca1d990cfc~64bit (git ca1d990 2014-11-19
22:11:13Z 64bit)

P.S. I am new to FS. I came from Asterisk world.
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From danb.lists at gmail.com  Mon Dec  8 11:04:43 2014
From: danb.lists at gmail.com (DanB)
Date: Mon, 08 Dec 2014 09:04:43 +0100
Subject: [Freeswitch-users] Recommendation for Billing for Hosted
	PBX	Platform
In-Reply-To: <mailman.2053.1418023063.581.freeswitch-users@lists.freeswitch.org>
References: <mailman.2053.1418023063.581.freeswitch-users@lists.freeswitch.org>
Message-ID: <54855B9B.8010409@gmail.com>

Hi Deon,

If you are looking for a billing engine (which I find it more suitable 
than complete billing system for hosted environments, same approach as 
FreeSWITCH +/or Kazoo - it does not enforce any business rules), take a 
look on CGRateS.
It is opensource, very fast (12k+ requests per second), real-time, 
scalable (fire up more instances as you need them), feature-rich 
(multi-tenancy, supports advanced concepts as multiple calls out of same 
account, multiple balances inside one account, derived charging - eg: 
reseller chaining, grouped rating - charge using various billing 
increments within same call), API driven (full set of APIs available in 
both JSON and GOB) and production ready (runs already at MVNOs, ISPs, 
wholesale suppliers sites live).

Happy to help you more!
DanB



From notify.sina at gmail.com  Mon Dec  8 11:55:02 2014
From: notify.sina at gmail.com (Notify Me)
Date: Mon, 8 Dec 2014 09:55:02 +0100
Subject: [Freeswitch-users] Trouble Playing Audio to external dialler after
 bridged call to external SIP URI is hung up
Message-ID: <CACK8u8LL210JwftK3J-bg4K=uePeEFW7W3pcjKf7qKuUEHeVKA@mail.gmail.com>

Hi!

I am still a Freeswitch Newbie, and I am trying to learn by doing.
I have installed and using  version
1.5.15b+git~20141120T035109Z~79de78a0fb~64bit on a 64-bit CentOS 6.6
kvm node.

I have been able to successfully transfer calls for an external
dialler, through a SIP trunk, to call an external SIP URI. This works
and I can see it in the logs as very successful. The external SIP URI
is supposed to process the caller_id, and send the dialler an SMS.

The problem is the SIP URI being dialled ends the call very quickly
and the external dialler has a busy tone and the call is dropped, or
other signals that do not indicate that the call was successful and
may dial again several times, if the SMS does not get delivered in
time. The SIP URI sends back a 100 Trying and a 480 Temporarily
Unavailable response as you can see below from a dump of the sequence.
I have tried to introduce ivr sounds before and after the user
connects and the bridge is dropped but is also not working. I can see
that the wav files are being called from the logs but I dont hear
anything at all. Please can anyone guide me?


my dialplan:

public:

<include>
 <extension name="inbound">
  <condition field="destination_number"  expression="^012345(\d{2})$"/>
   <action application="set" data="ringback=${us-ring}"/>
  <condition field="caller_id_number"  expression="^0(\d{10})$"
require-nested="true">
     <action application="set" data="effective_caller_id_number=+234$1"/>
     <action application="transfer" data="1212 XML default"/>
  </condition>
 </extension>
</include>

default:
<extension name="1212">
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="transfer_after_bridge=1213:XML:default"/>
       <action application="set" data="exec_after_bridge_app=transfer"/>
       <action application="set" data="exec_after_bridge_arg="1213"/>
       <action application="bridge"
data="sofia/external/sip:user.name at sip.othersipgw.ip.address.com;${effective_caller_id_number}"/>
 </condition>
</extension>

<!-- What to do after othersipgw drops the call with -->
<extension name="1213">
  <condition field="destination_number" expression="1213">
    <action application="set" data="instant_ringback=true"/>
    <action application="set" data="transfer_ringback=$${us-ring}"/>
    <action application="playback" data="ivr/ivr-thank_you.wav"/>
    <action application="playback" data="ivr/ivr-thank_you.wav"/>
   </condition>
</extension>



TCP DUMP
09:31:58.808760 IP (tos 0x0, ttl 64, id 34090, offset 0, flags [none],
proto UDP (17), length 1146)
    my.natted.priv.ip.address.5080 > othersipgw.ip.address.5060: SIP,
length: 1118
        INVITE sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0
        Via: SIP/2.0/UDP
my.public.ip.address:5080;rport;branch=z9hG4bK8eDcDvZpBm4yr
        Max-Forwards: 5
        From: "0diallednumber"
<sip:+234diallednumber at my.public.ip.address>;tag=6pKBvpXc32yeD
        To: <sip:user.name at sip.othersipgw.com;+234diallednumber>
        Call-ID: 840abf24-f957-1232-2ca9-525400ecad09
        CSeq: 68677695 INVITE
        Contact: <sip:mod_sofia at my.public.ip.address:5080>
        User-Agent:
FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~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: 247
        X-FS-Support: update_display,send_info
        Remote-Party-ID: "0diallednumber"
<sip:+234diallednumber at my.public.ip.address>;party=calling;screen=yes;privacy=off

        v=0
        o=FreeSWITCH 1418006836 1418006837 IN IP4 my.public.ip.address
        s=FreeSWITCH
        c=IN IP4 my.public.ip.address
        t=0 0
        m=audio 20682 RTP/AVP 8 0 101 13
        a=rtpmap:8 PCMA/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20

09:31:59.080097 IP (tos 0x0, ttl 53, id 5734, offset 0, flags [none],
proto UDP (17), length 384)
    othersipgw.ip.address.5060 > my.natted.priv.ip.address.5080: SIP,
length: 356
        SIP/2.0 100 Trying
        Via: SIP/2.0/UDP
my.public.ip.address:5080;rport=5080;branch=z9hG4bK8eDcDvZpBm4yr
        From: "0diallednumber"
<sip:+234diallednumber at my.public.ip.address>;tag=6pKBvpXc32yeD
        To: <sip:user.name at sip.othersipgw.com;+234diallednumber>
        Call-ID: 840abf24-f957-1232-2ca9-525400ecad09
        CSeq: 68677695 INVITE
        User-Agent: FreeSWITCH-mod_sofia/1.2.14
        Content-Length: 0


09:31:59.145629 IP (tos 0x0, ttl 53, id 5735, offset 0, flags [none],
proto UDP (17), length 1015)
    othersipgw.ip.address.5060 > my.natted.priv.ip.address.5080: SIP,
length: 987
        SIP/2.0 480 Temporarily Unavailable
        Via: SIP/2.0/UDP
my.public.ip.address:5080;rport=5080;branch=z9hG4bK8eDcDvZpBm4yr
        Max-Forwards: 4
        From: "0diallednumber"
<sip:+234diallednumber at my.public.ip.address>;tag=6pKBvpXc32yeD
        To: <sip:user.name at sip.othersipgw.com;+234diallednumber>;tag=Ur840N1DvjvpH
        Call-ID: 840abf24-f957-1232-2ca9-525400ecad09
        CSeq: 68677695 INVITE
        User-Agent: FreeSWITCH-mod_sofia/1.2.14
        Accept: application/sdp
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
        Supported: timer, precondition, path, replaces
        Allow-Events: talk, hold, conference, presence, dialog,
line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer
        Reason: Q.850;cause=16;text="NORMAL_CLEARING"
        Content-Length: 0
        X-FS-Display-Name: user.name
        X-FS-Display-Number: sip:user.name at sip.othersipgw.com
        Remote-Party-ID: "user.name"
<sip:user.name at sip.othersipgw.com>;party=calling;privacy=off;screen=no


09:31:59.146980 IP (tos 0x0, ttl 64, id 34091, offset 0, flags [none],
proto UDP (17), length 412)
    my.natted.priv.ip.address.5080 > othersipgw.ip.address.5060: SIP,
length: 384
        ACK sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0
        Via: SIP/2.0/UDP
my.public.ip.address:5080;rport;branch=z9hG4bK8eDcDvZpBm4yr
        Max-Forwards: 5
        From: "0diallednumber"
<sip:+234diallednumber at my.public.ip.address>;tag=6pKBvpXc32yeD
        To: <sip:user.name at sip.othersipgw.com;+234diallednumber>;tag=Ur840N1DvjvpH
        Call-ID: 840abf24-f957-1232-2ca9-525400ecad09
        CSeq: 68677695 ACK
        Content-Length: 0


From iwada.bassey at gmail.com  Mon Dec  8 12:15:53 2014
From: iwada.bassey at gmail.com (Iwada Eja)
Date: Mon, 8 Dec 2014 04:15:53 -0500
Subject: [Freeswitch-users] Recommendation for Billing for Hosted PBX
	Platform
In-Reply-To: <54855B9B.8010409@gmail.com>
References: <mailman.2053.1418023063.581.freeswitch-users@lists.freeswitch.org>
	<54855B9B.8010409@gmail.com>
Message-ID: <CAHaUJVY2tTZQbpFGN00upjCCr=kBrcbXd=z05p19FhEN32g43g@mail.gmail.com>

+1 for CGRateS.
 Been using it in Production with no hassles

On Mon, Dec 8, 2014 at 3:04 AM, DanB <danb.lists at gmail.com> wrote:

> Hi Deon,
>
> If you are looking for a billing engine (which I find it more suitable
> than complete billing system for hosted environments, same approach as
> FreeSWITCH +/or Kazoo - it does not enforce any business rules), take a
> look on CGRateS.
> It is opensource, very fast (12k+ requests per second), real-time,
> scalable (fire up more instances as you need them), feature-rich
> (multi-tenancy, supports advanced concepts as multiple calls out of same
> account, multiple balances inside one account, derived charging - eg:
> reseller chaining, grouped rating - charge using various billing
> increments within same call), API driven (full set of APIs available in
> both JSON and GOB) and production ready (runs already at MVNOs, ISPs,
> wholesale suppliers sites live).
>
> Happy to help you more!
> DanB
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kind Regards
Iwada
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From mkvonarx at gmail.com  Mon Dec  8 13:29:33 2014
From: mkvonarx at gmail.com (Markus von Arx)
Date: Mon, 8 Dec 2014 11:29:33 +0100
Subject: [Freeswitch-users] Windows build system
In-Reply-To: <E706BBBE-829E-4A71-A299-5C61C5011CF0@jerris.com>
References: <E706BBBE-829E-4A71-A299-5C61C5011CF0@jerris.com>
Message-ID: <CAHA5ALf1A6JEg-OPcLc+w5v2-96r15YBk_ADkADwS=KwOVmGkg@mail.gmail.com>

We are fine with your Visual Studio plans. We only use VS2012 at the moment
and actually would like to migrate to VS2013.

Not so sure about the use of chocolatey though. Are you planning to use it
to download libraries required by the build or for installing
software/tools required for the build? Both look a bit problematic to me,
because chocolatey (as far as I understand it) always acts globally on the
target machine, meaning that it installs the libs/tools/software not
locally inside the build directory but globally on the machine. I wouldn't
like that at all. I don't want a build tool/process to install anything
outside the build directory. If you manage to use chocolatey to only work
in the build directory that's fine for me. But I'd strongly vote against
any use of chocolatey to install libraries, tools or any software globally
or outside the build directory. I wouldn't like a build tool/process/system
installing anything on my machine for me automatically. Kind of like
calling apt-get on a Linux machine from a build script. I think that is a
no-go. Also, chocolatey is not so good detecting installed software that
was not installed by chocolatey itself and would often try to re-install
software that is already there. I'd vote against using chocolatey in the
FreeSWITCH build if you ask me. Wouldn't nuget be the more natural choice
anyway to install modules/libraries in Visual Studio? Just my opinion.

Markus


2014-12-03 21:31 GMT+01:00 Michael Jerris <mike at jerris.com>:

> Given the recent announcements by Microsoft about the community edition
> 2013 being available, we are working to migrate the build system towards
> using that as our primary build.  As part of this process we will be very
> soon dropping support for any version of Visual Studio prior to 2012.  If
> you feel strongly about needing support for these older versions, please
> speak up now with an offer to maintain these legacy build systems.  We are
> also investigating moving to using chocolatey as a new system to manage
> dependencies on windows instead of maintaining the build for all our deps
> ourselves.   It is also possible we will drop support for the 2012 build
> system in the not so distant future.  Could the community chime in here as
> to what their needs are, and what they are willing to do to help support
> the windows builds so we can determine what we plan to support going
> forward.
>
> Thanks
> Mike
>
> https://chocolatey.org/
> http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From den.pavlovsky at gmail.com  Mon Dec  8 14:28:45 2014
From: den.pavlovsky at gmail.com (Den Pavlovsky)
Date: Mon, 8 Dec 2014 17:28:45 +0600
Subject: [Freeswitch-users] outgoing speech path is broken sometimes in
	1.4.14
In-Reply-To: <CAAfQe6vHGjCNpQSkxZjk+BORsjwsLNjwGbkMK9D5z3wPTEi+gA@mail.gmail.com>
References: <CAAfQe6vHGjCNpQSkxZjk+BORsjwsLNjwGbkMK9D5z3wPTEi+gA@mail.gmail.com>
Message-ID: <CAAfQe6tFMSEDkdsCBoS+p8Wy8kJUDV3AtiyfT9=oN1_FDpMMmQ@mail.gmail.com>

I found what caused such behaviour:

I have 2 VLANs coming to my box.

When I commented out the second VLAN - I have encountered no problems since
then.
I didn't quite get what xml files to change so as to work with 2 VLANs.

2014-12-08 12:40 GMT+05:00 Den Pavlovsky <den.pavlovsky at gmail.com>:

> Hi
>
> I installed FS 1.4.14 from git
>
> Now I test outgoing calls with Radius authorization.
>
> The problem that I encountered is that sometimes there is no RTP from
> Freeswitch. I see it in tcpdump dump.
>
> Iptables  -no restrictions.
>
> I also tested when there is no mod_rad_auth loaded (I commented it out in
> autoload_configs/modules.conf.xml). - the same result
>
> And when I put the call on hold - then when I return - the call path is
> normal.
>
> Any help on how to debug this issue is appreciated.
>
> Does anyone encountered the same behaviour? The version FreeSWITCH Version
> 1.4.14+git~20141119T221113Z~ca1d990cfc~64bit (git ca1d990 2014-11-19
> 22:11:13Z 64bit)
>
> P.S. I am new to FS. I came from Asterisk world.
>
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From avi at avimarcus.net  Mon Dec  8 16:21:10 2014
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 8 Dec 2014 13:21:10 +0000
Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound
 Caller ID to external SIP Gateway URI
In-Reply-To: <CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com>
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com>
	<0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com>
	<CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com>
	<0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com>
	<CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com>
Message-ID: <0000014a2a0f8f54-5ad8d656-a43c-424b-aa41-8a315705cea0-000000@email.amazonses.com>

Since there's only a 100 trying and no 18X, you won't get a ringing tone.
If it's short, just play a message after. If not, you can use lua/js and
run this second call as a bgapi call and try to capture the response.
-Avi


On Sun, Dec 7, 2014 at 1:47 PM, Notify Me <notify.sina at gmail.com> wrote:

> The delay is very short, they send a100 Trying  and then a 480 temporarily
> Unavailable back, as I can see in tcpdump. After they send the fail
> and its acknowledged, they end it and send an SMS back.
> This is a full dump
>
> 12:41:27.169928 IP (tos 0x0, ttl 64, id 34092, offset 0, flags [none],
> proto UDP (17), length 1146)
>     10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 1118
>         INVITE sip:sina.nowahala at sip.othersipgw.com;+234diallednumber
> SIP/2.0
>         Via: SIP/2.0/UDP
> my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q
>         Max-Forwards: 5
>         From: "234diallednumber"
> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>         To: <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>
>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>         CSeq: 68640179 INVITE
>         Contact: <sip:mod_sofia at my.public.ip.address:5080>
>         User-Agent:
> FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~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: 247
>         X-FS-Support: update_display,send_info
>         Remote-Party-ID: "234diallednumber"
> <sip:+234diallednumber at my.public.ip.address
> >;party=calling;screen=yes;privacy=off
>
>         v=0
>         o=FreeSWITCH 1417930197 1417930198 IN IP4 my.public.ip.address
>         s=FreeSWITCH
>         c=IN IP4 my.public.ip.address
>         t=0 0
>         m=audio 22290 RTP/AVP 8 0 101 13
>         a=rtpmap:8 PCMA/8000
>         a=rtpmap:0 PCMU/8000
>         a=rtpmap:101 telephone-event/8000
>         a=fmtp:101 0-16
>         a=ptime:20
>
> 12:41:27.385419 IP (tos 0x0, ttl 53, id 5736, offset 0, flags [none],
> proto UDP (17), length 384)
>     othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 356
>         SIP/2.0 100 Trying
>         Via: SIP/2.0/UDP
> my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q
>         From: "234diallednumber"
> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>         To: <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>
>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>         CSeq: 68640179 INVITE
>         User-Agent: FreeSWITCH-mod_sofia/1.2.14
>         Content-Length: 0
>
>
> 12:41:27.424422 IP (tos 0x0, ttl 53, id 5737, offset 0, flags [none],
> proto UDP (17), length 1015)
>     othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 987
>         SIP/2.0 480 Temporarily Unavailable
>         Via: SIP/2.0/UDP
> my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q
>         Max-Forwards: 4
>         From: "234diallednumber"
> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>         To: <sip:sina.nowahala at sip.othersipgw.com
> ;+234diallednumber>;tag=8c8DZK1aK9mKN
>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>         CSeq: 68640179 INVITE
>         User-Agent: FreeSWITCH-mod_sofia/1.2.14
>         Accept: application/sdp
>         Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
> UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>         Supported: timer, precondition, path, replaces
>         Allow-Events: talk, hold, conference, presence, dialog,
> line-seize, call-info, sla, include-session-description,
> presence.winfo, message-summary, refer
>         Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>         Content-Length: 0
>         X-FS-Display-Name: sina.nowahala
>         X-FS-Display-Number: sip:sina.nowahala at sip.othersipgw.com
>         Remote-Party-ID: "sina.nowahala"
> <sip:sina.nowahala at sip.othersipgw.com>;party=calling;privacy=off;screen=no
>
>
> 12:41:27.424789 IP (tos 0x0, ttl 64, id 34093, offset 0, flags [none],
> proto UDP (17), length 412)
>     10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 384
>         ACK sip:sina.nowahala at sip.othersipgw.com;+234diallednumber SIP/2.0
>         Via: SIP/2.0/UDP
> my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q
>         Max-Forwards: 5
>         From: "234diallednumber"
> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>         To: <sip:sina.nowahala at sip.othersipgw.com
> ;+234diallednumber>;tag=8c8DZK1aK9mKN
>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>         CSeq: 68640179 ACK
>         Content-Length: 0
>
>
> So you recommend a transfer on fail wav to be played? I have been
> tearing my hair out trying to play sounds before the call is
> transferred to them, so that the dialler hears something.
>
> On Sun, Dec 7, 2014 at 4:44 AM, Avi Marcus <avi at avimarcus.net> wrote:
> > How long is the delay? I assume they return a normal_clearing or fail
> 4XX?
> > If the delay is short and they have a "real" status return, then.. for
> > success, you can just queue a "success.wav" to play after.
> > And for fails, you can use
> > https://wiki.freeswitch.org/wiki/Variable_transfer_on_fail and transfer
> to
> > an extension that plays an error message.
> >
> > This SMS sending sounds like a normal API call.. which if it was, could
> be
> > done with curl or inside a lua/js script.
> >
> > -Avi
> >
> > On Fri, Dec 5, 2014 at 6:47 PM, Notify Me <notify.sina at gmail.com> wrote:
> >>
> >> Hi Avi!
> >>
> >> Thanks for the reply, I did figure it out, thanks! This is what I did,
> >>
> >> <include>
> >>  <extension name="firstsipgw-inbound">
> >>   <condition field="destination_number"  expression="^0123450(\d{2})$"/>
> >>   <condition field="caller_id_number"  expression="^0(\d{10})$"
> >> require-nested="true">
> >>      <action application="set"
> data="effective_caller_id_number=+234$1"/>
> >>      <action application="set"
> >> data="caller_id_number=+234${effective_caller_id_number}"/>
> >>      <action application="set" data="transfer_ringback=$${us-ring}"/>
> >>      <action application="set"
> >> data="transfer_ringback=file_string://$${hold_music}"/>
> >>      <action application="set"
> >> data="transfer_ringback=local_stream://connecting"/>
> >>        <action application="bridge"
> >>
> >> data="sofia/external/sip:user.name at sip.othersipgw.com
> ;${effective_caller_id_number}"/>
> >>   </condition>
> >>  </extension>
> >> </include>
> >>
> >> I can see from the logs that dials the othersipgw.com URI, and I can
> >> see tcpdumps that corresponds to the traffic being sent,  and the
> >> other side gets it. The other side immediately drops the call and does
> >> some processing that returns an SMS to the dialler.
> >>
> >> I dont know if I can trouble you for more help.. there is no ringing
> >> when the transaction happens, and the dialler is unsure what is
> >> happening, if the call actually connected. As you might have guessed
> >> from the above I am trying to make it ring for a few seconds before
> >> the call drops. Is this possible? I would like for the dialler to be
> >> able to hear it ring a few times before the call is cut.
> >>
> >> Any help gratefully accepted.
> >>
> >> On Tue, Dec 2, 2014 at 2:22 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >> > Hi - did you figure this out yet?
> >> >
> >> > One comment:
> >> >  <action application="bridge" data="sofia/gateway/othersipgw/+234$1"/>
> >> > -- I
> >> > don't know if $1 is available anymore. You might want to just set that
> >> > as
> >> > part of the actual number to route, e.g. add a 1212 prefix and match
> it
> >> > again or set it as a channel variable. You can see in the logs if the
> $1
> >> > is
> >> > resolving correctly.
> >> >
> >> > -Avi
> >> >
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://confluence.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >
> >
>
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From notify.sina at gmail.com  Mon Dec  8 17:07:57 2014
From: notify.sina at gmail.com (Sina Owolabi)
Date: Mon, 8 Dec 2014 15:07:57 +0100
Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound
 Caller ID to external SIP Gateway URI
In-Reply-To: <0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com>
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com>
	<0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com>
	<CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com>
	<0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com>
	<CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com>
	<0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com>
Message-ID: <CACK8u8J1b1UcEQ1R+M0puRT9vACtYGeuKBhHs_2ug9Q_qgJ6Wg@mail.gmail.com>

I've tried to play a message after but I cannot hear it. Please can
you tell me if this is correct?
Thanks again for all your help:


<extension name="1212">
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="set" data="transfer_after_bridge=1213:XML:default"/>
       <action application="set" data="exec_after_bridge_app=transfer"/>
       <action application="set" data="exec_after_bridge_arg="1213"/>
       <action application="bridge"
data="sofia/external/sip:user.name at sip.othersipgw.ip.address.com;${effective_caller_id_number}"/>
 </condition>
</extension>

<!-- What to do after othersipgw drops the connection-->
<extension name="1213">
  <condition field="destination_number" expression="1213">
    <action application="set" data="instant_ringback=true"/>
    <action application="set" data="transfer_ringback=$${us-ring}"/>
    <action application="playback" data="ivr/ivr-thank_you.wav"/>
    <action application="playback" data="ivr/ivr-thank_you.wav"/>
   </condition>
</extension>

On Mon, Dec 8, 2014 at 2:21 PM, Avi Marcus <avi at avimarcus.net> wrote:
> Since there's only a 100 trying and no 18X, you won't get a ringing tone.
> If it's short, just play a message after. If not, you can use lua/js and run
> this second call as a bgapi call and try to capture the response.
> -Avi
>
>
> On Sun, Dec 7, 2014 at 1:47 PM, Notify Me <notify.sina at gmail.com> wrote:
>>
>> The delay is very short, they send a100 Trying  and then a 480 temporarily
>> Unavailable back, as I can see in tcpdump. After they send the fail
>> and its acknowledged, they end it and send an SMS back.
>> This is a full dump
>>
>> 12:41:27.169928 IP (tos 0x0, ttl 64, id 34092, offset 0, flags [none],
>> proto UDP (17), length 1146)
>>     10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 1118
>>         INVITE sip:sina.nowahala at sip.othersipgw.com;+234diallednumber
>> SIP/2.0
>>         Via: SIP/2.0/UDP
>> my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q
>>         Max-Forwards: 5
>>         From: "234diallednumber"
>> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>>         To: <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>
>>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>>         CSeq: 68640179 INVITE
>>         Contact: <sip:mod_sofia at my.public.ip.address:5080>
>>         User-Agent:
>> FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~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: 247
>>         X-FS-Support: update_display,send_info
>>         Remote-Party-ID: "234diallednumber"
>>
>> <sip:+234diallednumber at my.public.ip.address>;party=calling;screen=yes;privacy=off
>>
>>         v=0
>>         o=FreeSWITCH 1417930197 1417930198 IN IP4 my.public.ip.address
>>         s=FreeSWITCH
>>         c=IN IP4 my.public.ip.address
>>         t=0 0
>>         m=audio 22290 RTP/AVP 8 0 101 13
>>         a=rtpmap:8 PCMA/8000
>>         a=rtpmap:0 PCMU/8000
>>         a=rtpmap:101 telephone-event/8000
>>         a=fmtp:101 0-16
>>         a=ptime:20
>>
>> 12:41:27.385419 IP (tos 0x0, ttl 53, id 5736, offset 0, flags [none],
>> proto UDP (17), length 384)
>>     othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 356
>>         SIP/2.0 100 Trying
>>         Via: SIP/2.0/UDP
>> my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q
>>         From: "234diallednumber"
>> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>>         To: <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>
>>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>>         CSeq: 68640179 INVITE
>>         User-Agent: FreeSWITCH-mod_sofia/1.2.14
>>         Content-Length: 0
>>
>>
>> 12:41:27.424422 IP (tos 0x0, ttl 53, id 5737, offset 0, flags [none],
>> proto UDP (17), length 1015)
>>     othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 987
>>         SIP/2.0 480 Temporarily Unavailable
>>         Via: SIP/2.0/UDP
>> my.public.ip.address:5080;rport=5080;branch=z9hG4bK2S6KjD1jXFj1Q
>>         Max-Forwards: 4
>>         From: "234diallednumber"
>> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>>         To:
>> <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>;tag=8c8DZK1aK9mKN
>>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>>         CSeq: 68640179 INVITE
>>         User-Agent: FreeSWITCH-mod_sofia/1.2.14
>>         Accept: application/sdp
>>         Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
>> UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>>         Supported: timer, precondition, path, replaces
>>         Allow-Events: talk, hold, conference, presence, dialog,
>> line-seize, call-info, sla, include-session-description,
>> presence.winfo, message-summary, refer
>>         Reason: Q.850;cause=16;text="NORMAL_CLEARING"
>>         Content-Length: 0
>>         X-FS-Display-Name: sina.nowahala
>>         X-FS-Display-Number: sip:sina.nowahala at sip.othersipgw.com
>>         Remote-Party-ID: "sina.nowahala"
>> <sip:sina.nowahala at sip.othersipgw.com>;party=calling;privacy=off;screen=no
>>
>>
>> 12:41:27.424789 IP (tos 0x0, ttl 64, id 34093, offset 0, flags [none],
>> proto UDP (17), length 412)
>>     10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 384
>>         ACK sip:sina.nowahala at sip.othersipgw.com;+234diallednumber SIP/2.0
>>         Via: SIP/2.0/UDP
>> my.public.ip.address:5080;rport;branch=z9hG4bK2S6KjD1jXFj1Q
>>         Max-Forwards: 5
>>         From: "234diallednumber"
>> <sip:+234diallednumber at my.public.ip.address>;tag=04tDF6pXQU4Ng
>>         To:
>> <sip:sina.nowahala at sip.othersipgw.com;+234diallednumber>;tag=8c8DZK1aK9mKN
>>         Call-ID: d1ef2fdb-f8a8-1232-80a6-525400ecad09
>>         CSeq: 68640179 ACK
>>         Content-Length: 0
>>
>>
>> So you recommend a transfer on fail wav to be played? I have been
>> tearing my hair out trying to play sounds before the call is
>> transferred to them, so that the dialler hears something.
>>
>> On Sun, Dec 7, 2014 at 4:44 AM, Avi Marcus <avi at avimarcus.net> wrote:
>> > How long is the delay? I assume they return a normal_clearing or fail
>> > 4XX?
>> > If the delay is short and they have a "real" status return, then.. for
>> > success, you can just queue a "success.wav" to play after.
>> > And for fails, you can use
>> > https://wiki.freeswitch.org/wiki/Variable_transfer_on_fail and transfer
>> > to
>> > an extension that plays an error message.
>> >
>> > This SMS sending sounds like a normal API call.. which if it was, could
>> > be
>> > done with curl or inside a lua/js script.
>> >
>> > -Avi
>> >
>> > On Fri, Dec 5, 2014 at 6:47 PM, Notify Me <notify.sina at gmail.com> wrote:
>> >>
>> >> Hi Avi!
>> >>
>> >> Thanks for the reply, I did figure it out, thanks! This is what I did,
>> >>
>> >> <include>
>> >>  <extension name="firstsipgw-inbound">
>> >>   <condition field="destination_number"
>> >> expression="^0123450(\d{2})$"/>
>> >>   <condition field="caller_id_number"  expression="^0(\d{10})$"
>> >> require-nested="true">
>> >>      <action application="set"
>> >> data="effective_caller_id_number=+234$1"/>
>> >>      <action application="set"
>> >> data="caller_id_number=+234${effective_caller_id_number}"/>
>> >>      <action application="set" data="transfer_ringback=$${us-ring}"/>
>> >>      <action application="set"
>> >> data="transfer_ringback=file_string://$${hold_music}"/>
>> >>      <action application="set"
>> >> data="transfer_ringback=local_stream://connecting"/>
>> >>        <action application="bridge"
>> >>
>> >>
>> >> data="sofia/external/sip:user.name at sip.othersipgw.com;${effective_caller_id_number}"/>
>> >>   </condition>
>> >>  </extension>
>> >> </include>
>> >>
>> >> I can see from the logs that dials the othersipgw.com URI, and I can
>> >> see tcpdumps that corresponds to the traffic being sent,  and the
>> >> other side gets it. The other side immediately drops the call and does
>> >> some processing that returns an SMS to the dialler.
>> >>
>> >> I dont know if I can trouble you for more help.. there is no ringing
>> >> when the transaction happens, and the dialler is unsure what is
>> >> happening, if the call actually connected. As you might have guessed
>> >> from the above I am trying to make it ring for a few seconds before
>> >> the call drops. Is this possible? I would like for the dialler to be
>> >> able to hear it ring a few times before the call is cut.
>> >>
>> >> Any help gratefully accepted.
>> >>
>> >> On Tue, Dec 2, 2014 at 2:22 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> >> > Hi - did you figure this out yet?
>> >> >
>> >> > One comment:
>> >> >  <action application="bridge"
>> >> > data="sofia/gateway/othersipgw/+234$1"/>
>> >> > -- I
>> >> > don't know if $1 is available anymore. You might want to just set
>> >> > that
>> >> > as
>> >> > part of the actual number to route, e.g. add a 1212 prefix and match
>> >> > it
>> >> > again or set it as a channel variable. You can see in the logs if the
>> >> > $1
>> >> > is
>> >> > resolving correctly.
>> >> >
>> >> > -Avi
>> >> >
>> >> >
>> >> >
>> >> > _________________________________________________________________________
>> >> > Professional FreeSWITCH Consulting Services:
>> >> > consulting at freeswitch.org
>> >> > http://www.freeswitchsolutions.com
>> >> >
>> >> > Official FreeSWITCH Sites
>> >> > http://www.freeswitch.org
>> >> > http://confluence.freeswitch.org
>> >> > http://www.cluecon.com
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at 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 vitaliy.davudov at vts24.ru  Mon Dec  8 17:20:01 2014
From: vitaliy.davudov at vts24.ru (Vitaly)
Date: Mon, 08 Dec 2014 17:20:01 +0300
Subject: [Freeswitch-users] Log-auth-failures question
Message-ID: <5485B391.5050506@vts24.ru>

Hi, list!

In sip-profiles: "log-auth-failures" - write log entries ( Warning ) on 
authentication failures ( Registration & Invite ) in 
/usr/local/freeswitch/log/freeswitch.log files. Is it possible to write 
these entries in other file of in other directory?

Thanks in advance.

-- 
Best regards,
Vitaly.



From flokrrr at gmail.com  Mon Dec  8 17:31:40 2014
From: flokrrr at gmail.com (Florent Krieg)
Date: Mon, 8 Dec 2014 15:31:40 +0100
Subject: [Freeswitch-users] Log-auth-failures question
In-Reply-To: <5485B391.5050506@vts24.ru>
References: <5485B391.5050506@vts24.ru>
Message-ID: <CAABKsuRkbTHU1bUAHF+HGsZmnP1k_p+g64dp5F69SAj66wY7tg@mail.gmail.com>

Hi Vitaly,

You can configure the logfile path in the config file (logfile.conf):
<!-- File to log to -->
<!--<param name="logfile" value="/var/log/freeswitch.log"/>-->
But it will change the path of the file for all the logs, not only auth
failures!

Florent

2014-12-08 15:20 GMT+01:00 Vitaly <vitaliy.davudov at vts24.ru>:

> Hi, list!
>
> In sip-profiles: "log-auth-failures" - write log entries ( Warning ) on
> authentication failures ( Registration & Invite ) in
> /usr/local/freeswitch/log/freeswitch.log files. Is it possible to write
> these entries in other file of in other directory?
>
> Thanks in advance.
>
> --
> Best regards,
> Vitaly.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Dec  8 18:37:36 2014
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 8 Dec 2014 15:37:36 +0000
Subject: [Freeswitch-users] Log-auth-failures question
In-Reply-To: <CAABKsuRkbTHU1bUAHF+HGsZmnP1k_p+g64dp5F69SAj66wY7tg@mail.gmail.com>
References: <5485B391.5050506@vts24.ru>
	<CAABKsuRkbTHU1bUAHF+HGsZmnP1k_p+g64dp5F69SAj66wY7tg@mail.gmail.com>
Message-ID: <CAFiqYunBeRZMeG1p-Xo554jx0qQ1+m3E7fCY=1u-o+KNVdrs6w@mail.gmail.com>

If you're wanting to do to implement fail2ban and don't want large log
files you can also look at https://wiki.freeswitch.org/wiki/Mod_fail2ban

On 8 December 2014 at 14:31, Florent Krieg <flokrrr at gmail.com> wrote:

> Hi Vitaly,
>
> You can configure the logfile path in the config file (logfile.conf):
> <!-- File to log to -->
> <!--<param name="logfile" value="/var/log/freeswitch.log"/>-->
> But it will change the path of the file for all the logs, not only auth
> failures!
>
> Florent
>
> 2014-12-08 15:20 GMT+01:00 Vitaly <vitaliy.davudov at vts24.ru>:
>
>> Hi, list!
>>
>> In sip-profiles: "log-auth-failures" - write log entries ( Warning ) on
>> authentication failures ( Registration & Invite ) in
>> /usr/local/freeswitch/log/freeswitch.log files. Is it possible to write
>> these entries in other file of in other directory?
>>
>> Thanks in advance.
>>
>> --
>> Best regards,
>> Vitaly.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aqsyounas at gmail.com  Mon Dec  8 20:06:05 2014
From: aqsyounas at gmail.com (aqs younas)
Date: Mon, 8 Dec 2014 22:06:05 +0500
Subject: [Freeswitch-users] Building FS with mod_java support
In-Reply-To: <CAEA-bfHo6KOrMresGtG8oDpJGEOwQUp2f5=8PmrNsBD=D8M=WQ@mail.gmail.com>
References: <CAEA-bfHo6KOrMresGtG8oDpJGEOwQUp2f5=8PmrNsBD=D8M=WQ@mail.gmail.com>
Message-ID: <5485da7c.475cc20a.5edd.6ff2@mx.google.com>

Uncomment mod_java from modules.conf  in source directory and do there compile and install instead making it compile from  jave directory.

-----Original Message-----
From: "Victor Medina" <vittico at gmail.com>
Sent: ?12/?5/?2014 5:47 PM
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Building FS with mod_java support

Hi all!

I?ve been trying to build Freeswitch 1.4 with Java support. It
compiles just fine, installation fails with:

making install mod_java
make[4]: Entering directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make  install-am
make[5]: Entering directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[6]: Entering directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[6]: Nothing to be done for `install-exec-am'.
cp freeswitch.jar /opt/CloudVoice-vPBX/fs/scripts
cp: cannot stat ?freeswitch.jar?: No such file or directory
make[6]: *** [install-data-local] Error 1
make[6]: Leaving directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[5]: *** [install-am] Error 2
make[5]: Leaving directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[4]: *** [install] Error 2
make[4]: Leaving directory
`/home/administrador/freeswitch/src/mod/languages/mod_java'
make[3]: *** [mod_java-install] Error 1
make[3]: Leaving directory `/home/administrador/freeswitch/src/mod'
make[2]: *** [install-recursive] Error 1
make[2]: Leaving directory `/home/administrador/freeswitch/src'

Can someone help?

Sin mas a que hacer referencia,

Victor Medina

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

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

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.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 Dec  8 20:15:47 2014
From: mike at jerris.com (Michael Jerris)
Date: Mon, 8 Dec 2014 12:15:47 -0500
Subject: [Freeswitch-users] Billing software
In-Reply-To: <CAFLo2Qyd1uq+TOhc_sX5hahDku_DTHrQXdH3hXC+H5QFRLi4mg@mail.gmail.com>
References: <CAFLo2QzXJxyEDQ5YeNB3Engu4_LcTbdFELfXGhzgVChS4UoVbw@mail.gmail.com>
	<E84CC2DA-E8F2-4B02-8F09-B60D4406407E@gmail.com>
	<CAFLo2QyynMEY=NzJCZYmVAGFjiwNg7MgSGFrWCTWJYpyRnyCRQ@mail.gmail.com>
	<518CA7DD.8020802@gmail.com>
	<CAFLo2Qyd1uq+TOhc_sX5hahDku_DTHrQXdH3hXC+H5QFRLi4mg@mail.gmail.com>
Message-ID: <3792C5C9-2B4D-4E9C-BA65-B857F7D9893A@jerris.com>

Is this something that would be useful to integrate directly into Fusion?

> On Dec 7, 2014, at 9:58 AM, Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com> wrote:
> 
> Maybe a little late,
> 
> Thanks everyone, but I am really against to install an extra freeswitch instance for the billing. It makes very complex the call flow, troubleshoting and in most cases FAX wont work.  
> 
> I did try the ASTPP before, horrible, hard to configure. I consider myself good on freeswitch, i know to code on perl, and a good sysadmin.  But after watching how complex is to get this to work I abort.
> 
> So I decide to code my own billing and gladly, I can tell you that if you have all information needed (such as carrier rates and about 15 mins to get an explanation on how it works, you can have it working in less than an hour).  I started to study FusionPBX code, and after some months I have a working nice FusionPBX LCR+Billing native application.  It fully integrates in the Fusion, not out side, so you are able to bill for anything, incoming, internal (ext to ext, ext to conference, ext to ivr), or outgoing calls and it is able to get money by Paypal, Stripe (a good gw for CreditCards) and offline.  Currently, I have just release version 1.0.2
> 
> If someone is interested, you can go to my website www.okay.com.mx/en <http://www.okay.com.mx/en>  or contact me (gtalk prefered)
> 
> LD
> 
> 2013-05-10 3:55 GMT-04:00 Deon Vermeulen <vermeulen.deon at gmail.com <mailto:vermeulen.deon at gmail.com>>:
> Hi Luis
> 
> We just bought Commercial services from ASTPP.
> When it comes to comparing price, features this is the BEST solution out there at the moment.
> 
> This is a very active project and you can contact Samir Doshi, Project Maintainer & Developer,  directly for more information wrt your technical questions.
> 
> samir at astpp.org <mailto:samir at astpp.org>
> 
> 
> Kind Regards
> Deon

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From mike at jerris.com  Mon Dec  8 20:22:02 2014
From: mike at jerris.com (Michael Jerris)
Date: Mon, 8 Dec 2014 12:22:02 -0500
Subject: [Freeswitch-users] Windows build system
In-Reply-To: <CAHA5ALf1A6JEg-OPcLc+w5v2-96r15YBk_ADkADwS=KwOVmGkg@mail.gmail.com>
References: <E706BBBE-829E-4A71-A299-5C61C5011CF0@jerris.com>
	<CAHA5ALf1A6JEg-OPcLc+w5v2-96r15YBk_ADkADwS=KwOVmGkg@mail.gmail.com>
Message-ID: <EFC5D572-AF50-42A8-A8CA-D15BB697DB7A@jerris.com>

I'm a bit out of date on what these tools provide and need someone to get me up to speed on what is available.  Can you help me understand what the options are for maintaining c libraries in a way that is cleaner than needing to stuff them all into our build.


> On Dec 8, 2014, at 5:29 AM, Markus von Arx <mkvonarx at gmail.com> wrote:
> 
> We are fine with your Visual Studio plans. We only use VS2012 at the moment and actually would like to migrate to VS2013.
> 
> Not so sure about the use of chocolatey though. Are you planning to use it to download libraries required by the build or for installing software/tools required for the build? Both look a bit problematic to me, because chocolatey (as far as I understand it) always acts globally on the target machine, meaning that it installs the libs/tools/software not locally inside the build directory but globally on the machine. I wouldn't like that at all. I don't want a build tool/process to install anything outside the build directory. If you manage to use chocolatey to only work in the build directory that's fine for me. But I'd strongly vote against any use of chocolatey to install libraries, tools or any software globally or outside the build directory. I wouldn't like a build tool/process/system installing anything on my machine for me automatically. Kind of like calling apt-get on a Linux machine from a build script. I think that is a no-go. Also, chocolatey is not so good detecting installed software that was not installed by chocolatey itself and would often try to re-install software that is already there. I'd vote against using chocolatey in the FreeSWITCH build if you ask me. Wouldn't nuget be the more natural choice anyway to install modules/libraries in Visual Studio? Just my opinion.
> 
> Markus
> 
> 
> 2014-12-03 21:31 GMT+01:00 Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>>:
> Given the recent announcements by Microsoft about the community edition 2013 being available, we are working to migrate the build system towards using that as our primary build.  As part of this process we will be very soon dropping support for any version of Visual Studio prior to 2012.  If you feel strongly about needing support for these older versions, please speak up now with an offer to maintain these legacy build systems.  We are also investigating moving to using chocolatey as a new system to manage dependencies on windows instead of maintaining the build for all our deps ourselves.   It is also possible we will drop support for the 2012 build system in the not so distant future.  Could the community chime in here as to what their needs are, and what they are willing to do to help support the windows builds so we can determine what we plan to support going forward.
> 
> Thanks
> Mike
> 
> https://chocolatey.org/ <https://chocolatey.org/>
> http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx <http://www.visualstudio.com/en-us/news/vs2013-community-vs.aspx>
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From luis.daniel.lucio at gmail.com  Mon Dec  8 20:28:41 2014
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Mon, 8 Dec 2014 12:28:41 -0500
Subject: [Freeswitch-users] Billing software
In-Reply-To: <3792C5C9-2B4D-4E9C-BA65-B857F7D9893A@jerris.com>
References: <CAFLo2QzXJxyEDQ5YeNB3Engu4_LcTbdFELfXGhzgVChS4UoVbw@mail.gmail.com>
	<E84CC2DA-E8F2-4B02-8F09-B60D4406407E@gmail.com>
	<CAFLo2QyynMEY=NzJCZYmVAGFjiwNg7MgSGFrWCTWJYpyRnyCRQ@mail.gmail.com>
	<518CA7DD.8020802@gmail.com>
	<CAFLo2Qyd1uq+TOhc_sX5hahDku_DTHrQXdH3hXC+H5QFRLi4mg@mail.gmail.com>
	<3792C5C9-2B4D-4E9C-BA65-B857F7D9893A@jerris.com>
Message-ID: <CAFLo2Qw8gjEUZmSYHro2nJ02Oif=JC6a8hJOcfkwBacAAJf3ug@mail.gmail.com>

Michael, indeed it is.  Its a native application that fully integrates
into the FusionPBX

2014-12-08 12:15 GMT-05:00 Michael Jerris <mike at jerris.com>:
> Is this something that would be useful to integrate directly into Fusion?
>
> On Dec 7, 2014, at 9:58 AM, Luis Daniel Lucio Quiroz
> <luis.daniel.lucio at gmail.com> wrote:
>
> Maybe a little late,
>
> Thanks everyone, but I am really against to install an extra freeswitch
> instance for the billing. It makes very complex the call flow,
> troubleshoting and in most cases FAX wont work.
>
> I did try the ASTPP before, horrible, hard to configure. I consider myself
> good on freeswitch, i know to code on perl, and a good sysadmin.  But after
> watching how complex is to get this to work I abort.
>
> So I decide to code my own billing and gladly, I can tell you that if you
> have all information needed (such as carrier rates and about 15 mins to get
> an explanation on how it works, you can have it working in less than an
> hour).  I started to study FusionPBX code, and after some months I have a
> working nice FusionPBX LCR+Billing native application.  It fully integrates
> in the Fusion, not out side, so you are able to bill for anything, incoming,
> internal (ext to ext, ext to conference, ext to ivr), or outgoing calls and
> it is able to get money by Paypal, Stripe (a good gw for CreditCards) and
> offline.  Currently, I have just release version 1.0.2
>
> If someone is interested, you can go to my website www.okay.com.mx/en  or
> contact me (gtalk prefered)
>
> LD
>
> 2013-05-10 3:55 GMT-04:00 Deon Vermeulen <vermeulen.deon at gmail.com>:
>>
>> Hi Luis
>>
>> We just bought Commercial services from ASTPP.
>> When it comes to comparing price, features this is the BEST solution out
>> there at the moment.
>>
>> This is a very active project and you can contact Samir Doshi, Project
>> Maintainer & Developer,  directly for more information wrt your technical
>> questions.
>>
>> samir at astpp.org
>>
>>
>> Kind Regards
>> Deon
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jim at homeqwik.com  Mon Dec  8 23:52:32 2014
From: jim at homeqwik.com (Jim Hughes)
Date: Mon, 8 Dec 2014 13:52:32 -0700
Subject: [Freeswitch-users] Logging Time After Transfer
Message-ID: <eaaf3d67a4286c5425094c97ce8351a3@mail.gmail.com>

We would like our Call Detail Records from FreeSwitch to log the time after
a call is transferred.



I read entries in the forums that if we ?enable legs? we might get that
info, but it didn?t seem to work. We got lots more information, but not the
time after transfer.



For example: An agent receives a call and talks to somebody for 1 minute,
then transfers the call to a 3rd party and that call continues for 5
minutes.



We?d like the Call Detail Record to contain both pieces of information:



Time Before Transfer: 1 Minute

Time After Transfer: 5 Minutes



Has somebody already done this so we don?t have to re-invent the wheel?



Thanks,



Jim.
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From t.mahe at b-and-c.net  Tue Dec  9 01:23:17 2014
From: t.mahe at b-and-c.net (=?windows-1252?Q?Tristan_Mah=E9?=)
Date: Mon, 08 Dec 2014 14:23:17 -0800
Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and
 1.4.14
In-Reply-To: <54827C1D.5000401@b-and-c.net>
References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail>
	<54827C1D.5000401@b-and-c.net>
Message-ID: <548624D5.5050902@b-and-c.net>

Hi everyone,

Just a little followup, this is caused by <param name="rtp-timer-name"
value="soft"/> in a sip_profile. Happens on baremetal servers as well as
Xen guest.

Still investigating on the exact root of this behaviour change, but at
least now we know what is causing the trouble.

Best,

Tristan.

Le 05/12/2014 19:46, Tristan Mah? a ?crit :
> Hi guys,
> 
> While trying to find out the root cause of this, I just want to know if
> someone is also seeing a huge cpu load with freeswitch 1.4.14, which was
> not present in 1.4.12.
> 
> OS is debian wheezy, packages are official ones.
> 
> Same config, same trafic pattern, just an upgrade from 1.4.12 ( no other
> packages updated ).
> 
> We were previously seeing load around 0.5, with pikes up to around 2.
> 
> We now are seeing a constant load of around 2, and pikes up to 20...
> 
> If someone is also seeing this, that would be interesting to exchange,
> helping find out what is causing this !
> 
> I'll file a jira once I find the root of this problem ( and no, this is
> production servers, we can't put master there, we need stable ).
> 
> Best,
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


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From kbdfck at gmail.com  Tue Dec  9 01:33:26 2014
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 9 Dec 2014 02:33:26 +0400
Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and
	1.4.14
In-Reply-To: <548624D5.5050902@b-and-c.net>
References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail>
	<54827C1D.5000401@b-and-c.net> <548624D5.5050902@b-and-c.net>
Message-ID: <CA+GmxbYtoHfAK+W0=5cF+FBjGvSLGFKBepciDb-hBxHY7m_GZA@mail.gmail.com>

So, have you solved your problem? Did you change rtp-timer-name?
After we moved to 64-bit system, 1.4 branch shows huge LA spikes up to 35
on 4-core (or 4 single core virtual processor) with only 200-300 concurrent
channels and very low CPS (1-5).  On the same call load LA of 1.2 server is
significantly lower.
Setting 1.4 64bit to normal priority with -np helps to lower LA, but not
completely solves a problem.
It shows very big values in "ni" top fields (20-30%), but changing nice
value of FS process to zero or positive value doesn't help.




2014-12-09 2:23 GMT+04:00 Tristan Mah? <t.mahe at b-and-c.net>:

> Hi everyone,
>
> Just a little followup, this is caused by <param name="rtp-timer-name"
> value="soft"/> in a sip_profile. Happens on baremetal servers as well as
> Xen guest.
>
> Still investigating on the exact root of this behaviour change, but at
> least now we know what is causing the trouble.
>
> Best,
>
> Tristan.
>
> Le 05/12/2014 19:46, Tristan Mah? a ?crit :
> > Hi guys,
> >
> > While trying to find out the root cause of this, I just want to know if
> > someone is also seeing a huge cpu load with freeswitch 1.4.14, which was
> > not present in 1.4.12.
> >
> > OS is debian wheezy, packages are official ones.
> >
> > Same config, same trafic pattern, just an upgrade from 1.4.12 ( no other
> > packages updated ).
> >
> > We were previously seeing load around 0.5, with pikes up to around 2.
> >
> > We now are seeing a constant load of around 2, and pikes up to 20...
> >
> > If someone is also seeing this, that would be interesting to exchange,
> > helping find out what is causing this !
> >
> > I'll file a jira once I find the root of this problem ( and no, this is
> > production servers, we can't put master there, we need stable ).
> >
> > Best,
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,

Dmitry Sytchev,
IT Engineer
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From t.mahe at b-and-c.net  Tue Dec  9 01:48:42 2014
From: t.mahe at b-and-c.net (=?windows-1252?Q?Tristan_Mah=E9?=)
Date: Mon, 08 Dec 2014 14:48:42 -0800
Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and
 1.4.14
In-Reply-To: <CA+GmxbYtoHfAK+W0=5cF+FBjGvSLGFKBepciDb-hBxHY7m_GZA@mail.gmail.com>
References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail>	<54827C1D.5000401@b-and-c.net>
	<548624D5.5050902@b-and-c.net>
	<CA+GmxbYtoHfAK+W0=5cF+FBjGvSLGFKBepciDb-hBxHY7m_GZA@mail.gmail.com>
Message-ID: <54862ACA.6000506@b-and-c.net>

Still on it, but I have some servers deployed without rtp-timer-name set
to soft ( blocking read is something I'd like to avoid, but needs to
find out what changed in FS code and also find out the impact on our
users ), and so far they have load avg divided by 10 with the same
amount of calls and the same trafic as those with rtp-timer-name to soft.

Had no issue with 1.4.12 as said previously, but 1.4.14 has some
interesting fixes that justify staying in production.

Le 08/12/2014 14:33, Dmitry Sytchev a ?crit :
> So, have you solved your problem? Did you change rtp-timer-name?
> After we moved to 64-bit system, 1.4 branch shows huge LA spikes up to
> 35 on 4-core (or 4 single core virtual processor) with only 200-300
> concurrent channels and very low CPS (1-5).  On the same call load LA of
> 1.2 server is significantly lower.
> Setting 1.4 64bit to normal priority with -np helps to lower LA, but not
> completely solves a problem.
> It shows very big values in "ni" top fields (20-30%), but changing nice
> value of FS process to zero or positive value doesn't help.
> 
> 
> 
> 
> 2014-12-09 2:23 GMT+04:00 Tristan Mah? <t.mahe at b-and-c.net
> <mailto:t.mahe at b-and-c.net>>:
> 
>     Hi everyone,
> 
>     Just a little followup, this is caused by <param name="rtp-timer-name"
>     value="soft"/> in a sip_profile. Happens on baremetal servers as well as
>     Xen guest.
> 
>     Still investigating on the exact root of this behaviour change, but at
>     least now we know what is causing the trouble.
> 
>     Best,
> 
>     Tristan.
> 
>     Le 05/12/2014 19:46, Tristan Mah? a ?crit :
>     > Hi guys,
>     >
>     > While trying to find out the root cause of this, I just want to
>     know if
>     > someone is also seeing a huge cpu load with freeswitch 1.4.14,
>     which was
>     > not present in 1.4.12.
>     >
>     > OS is debian wheezy, packages are official ones.
>     >
>     > Same config, same trafic pattern, just an upgrade from 1.4.12 ( no
>     other
>     > packages updated ).
>     >
>     > We were previously seeing load around 0.5, with pikes up to around 2.
>     >
>     > We now are seeing a constant load of around 2, and pikes up to 20...
>     >
>     > If someone is also seeing this, that would be interesting to exchange,
>     > helping find out what is causing this !
>     >
>     > I'll file a jira once I find the root of this problem ( and no,
>     this is
>     > production servers, we can't put master there, we need stable ).
>     >
>     > Best,
>     >
>     >
>     >
>     >
>     _________________________________________________________________________
>     > 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
> 
> 
> 
> 
> -- 
> Best regards,
> 
> Dmitry Sytchev,
> IT Engineer
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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.keil at visytel.com  Tue Dec  9 02:00:39 2014
From: andrew.keil at visytel.com (Andrew Keil)
Date: Mon, 8 Dec 2014 23:00:39 +0000
Subject: [Freeswitch-users] Re - FreeSWITCH Lua CUSTOM events interrupting
 session:streamFile playback - is this possible?
Message-ID: <HKXPR04MB0456E779E02E79F7E5AB30B3F3640@HKXPR04MB0456.apcprd04.prod.outlook.com>

To FreeSWITCH users,

After progressing with Lua inside FreeSWITCH I have a question that I cannot source an answer for within the current FreeSWITCH documentation.

I have a requirement for an external application to process an event (via the event socket layer) then return the results back to the application while an audio file is played back in a loop to the caller.

>From the current documentation this method allows for the event to be sent then the return event is "consumed" using a polling approach.  However the caller is listening to silence.

function poll()
    -- create event and listener
    local event = freeswitch.Event("CUSTOM", "ping::running?")
    local con = freeswitch.EventConsumer("CUSTOM", "ping::running!")

    -- add text ad libitum
    event:addHeader("hi", "there")
    -- fire event
    event:fire()
    -- and wait for reply but not very long
                local retevent = con:pop(1, 5000)
    if retevent then
        print("reply received")
                                freeswitch.consoleLog("DEBUG",  string.format("reply received: %s\n",retevent:getHeader("Result")))
        return true
    end
    print("no reply")
                freeswitch.consoleLog("DEBUG", "no reply\n")
    return false
end

Some questions:

1)      Is there a way to playback audio (eg. session:streamFile(...)) while this takes place, since currently session:streamFile(...) seems to be a blocking function (ie. Finishes when the audio file is played back completely)?

2)      Is there a way to use session:setInputCallback(...) to handle an external CUSTOM event being returned (since this would then be able to interrupt the session:streamFile(...) just like it does for DTMF or speech recognition)?

If you have a preferred approach to solving this then I am open to your suggestions.

I appreciate any assistance that you can give.

Kind Regards,

Andrew Keil

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From anthony.minessale at gmail.com  Tue Dec  9 02:02:41 2014
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 8 Dec 2014 17:02:41 -0600
Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and
	1.4.14
In-Reply-To: <54862ACA.6000506@b-and-c.net>
References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail>
	<54827C1D.5000401@b-and-c.net> <548624D5.5050902@b-and-c.net>
	<CA+GmxbYtoHfAK+W0=5cF+FBjGvSLGFKBepciDb-hBxHY7m_GZA@mail.gmail.com>
	<54862ACA.6000506@b-and-c.net>
Message-ID: <CAKbxfG9qAnSOS0wEOLefafszWsz1SB_pbjq7mJTFH2Z1JSO0QA@mail.gmail.com>

This should have been in JIRA... Drop the mentality that you have to make a
case first to open a JIRA.
The actual process should be, if its any question about how FS works, if
you are right or wrong about it, file it as a JIRA.  If its wrong we can
just close it........

If you want to send one email on the list to get some attention to your
jira for discussion its ok but really you want to have the entire thing
preserved there.


I pushed a patch (4bcf1d888a4629993923527c26d4dbc6dae5d470) that might be
related to your problem but now I have to wing it and use email where I get
200 emails a day and there is no tracking at all in email when it comes to
issues so you put more resources on me to try to remember to look for your
reply when we have a perfectly valid issue tracker system in place.....


If master works better, good, if not, try

git checkout 878a04715ae801ccd9587249240f5a6e4f16dd0e src/switch_time.c
 and make install_core

All of this should be transposed back to JIRA by one of you.....




On Mon, Dec 8, 2014 at 4:48 PM, Tristan Mah? <t.mahe at b-and-c.net> wrote:

> Still on it, but I have some servers deployed without rtp-timer-name set
> to soft ( blocking read is something I'd like to avoid, but needs to
> find out what changed in FS code and also find out the impact on our
> users ), and so far they have load avg divided by 10 with the same
> amount of calls and the same trafic as those with rtp-timer-name to soft.
>
> Had no issue with 1.4.12 as said previously, but 1.4.14 has some
> interesting fixes that justify staying in production.
>
> Le 08/12/2014 14:33, Dmitry Sytchev a ?crit :
> > So, have you solved your problem? Did you change rtp-timer-name?
> > After we moved to 64-bit system, 1.4 branch shows huge LA spikes up to
> > 35 on 4-core (or 4 single core virtual processor) with only 200-300
> > concurrent channels and very low CPS (1-5).  On the same call load LA of
> > 1.2 server is significantly lower.
> > Setting 1.4 64bit to normal priority with -np helps to lower LA, but not
> > completely solves a problem.
> > It shows very big values in "ni" top fields (20-30%), but changing nice
> > value of FS process to zero or positive value doesn't help.
> >
> >
> >
> >
> > 2014-12-09 2:23 GMT+04:00 Tristan Mah? <t.mahe at b-and-c.net
> > <mailto:t.mahe at b-and-c.net>>:
> >
> >     Hi everyone,
> >
> >     Just a little followup, this is caused by <param
> name="rtp-timer-name"
> >     value="soft"/> in a sip_profile. Happens on baremetal servers as
> well as
> >     Xen guest.
> >
> >     Still investigating on the exact root of this behaviour change, but
> at
> >     least now we know what is causing the trouble.
> >
> >     Best,
> >
> >     Tristan.
> >
> >     Le 05/12/2014 19:46, Tristan Mah? a ?crit :
> >     > Hi guys,
> >     >
> >     > While trying to find out the root cause of this, I just want to
> >     know if
> >     > someone is also seeing a huge cpu load with freeswitch 1.4.14,
> >     which was
> >     > not present in 1.4.12.
> >     >
> >     > OS is debian wheezy, packages are official ones.
> >     >
> >     > Same config, same trafic pattern, just an upgrade from 1.4.12 ( no
> >     other
> >     > packages updated ).
> >     >
> >     > We were previously seeing load around 0.5, with pikes up to around
> 2.
> >     >
> >     > We now are seeing a constant load of around 2, and pikes up to
> 20...
> >     >
> >     > If someone is also seeing this, that would be interesting to
> exchange,
> >     > helping find out what is causing this !
> >     >
> >     > I'll file a jira once I find the root of this problem ( and no,
> >     this is
> >     > production servers, we can't put master there, we need stable ).
> >     >
> >     > Best,
> >     >
> >     >
> >     >
> >     >
> >
>  _________________________________________________________________________
> >     > 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
> >
> >
> >
> >
> > --
> > Best regards,
> >
> > Dmitry Sytchev,
> > IT Engineer
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From dujinfang at gmail.com  Tue Dec  9 02:15:45 2014
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 9 Dec 2014 07:15:45 +0800
Subject: [Freeswitch-users] Re - FreeSWITCH Lua CUSTOM events
 interrupting session:streamFile playback - is this possible?
In-Reply-To: <HKXPR04MB0456E779E02E79F7E5AB30B3F3640@HKXPR04MB0456.apcprd04.prod.outlook.com>
References: <HKXPR04MB0456E779E02E79F7E5AB30B3F3640@HKXPR04MB0456.apcprd04.prod.outlook.com>
Message-ID: <FBDD49CEBFC34FB8AA95229D9EF76562@gmail.com>

I believe the playback_terminitors chan var works, or you could try set an input callback and return ?break? when dtmf detected.  

--  
Seven Du
http://about.me/dujinfang
http://www.dujinfang.com

http://www.freeswitch.org.cn


Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Tuesday, December 9, 2014 at 7:00 AM, Andrew Keil wrote:

> To FreeSWITCH users,
>   
> After progressing with Lua inside FreeSWITCH I have a question that I cannot source an answer for within the current FreeSWITCH documentation.
>   
> I have a requirement for an external application to process an event (via the event socket layer) then return the results back to the application while an audio file is played back in a loop to the caller.
>   
> From the current documentation this method allows for the event to be sent then the return event is ?consumed? using a polling approach.  However the caller is listening to silence.
>   
> function poll()
>     -- create event and listener
>     local event = freeswitch.Event("CUSTOM", "ping::running?")
>     local con = freeswitch.EventConsumer("CUSTOM", "ping::running!")
>   
>     -- add text ad libitum
>     event:addHeader("hi", "there")
>     -- fire event
>     event:fire()
>     -- and wait for reply but not very long
>                 local retevent = con:pop(1, 5000)
>     if retevent then
>         print("reply received")
>                                 freeswitch.consoleLog("DEBUG",  string.format("reply received: %s\n",retevent:getHeader("Result")))
>         return true
>     end
>     print("no reply")
>                 freeswitch.consoleLog("DEBUG", "no reply\n")  
>     return false
> end
>   
> Some questions:
> 1)      Is there a way to playback audio (eg. session:streamFile(?)) while this takes place, since currently session:streamFile(?) seems to be a blocking function (ie. Finishes when the audio file is played back completely)?
> 2)      Is there a way to use session:setInputCallback(?) to handle an external CUSTOM event being returned (since this would then be able to interrupt the session:streamFile(?) just like it does for DTMF or speech recognition)?
>   
> If you have a preferred approach to solving this then I am open to your suggestions.
>   
> I appreciate any assistance that you can give.
>   
> Kind Regards,
>   
> Andrew Keil
>   
>  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:  
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org)
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
>  


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From krice at freeswitch.org  Tue Dec  9 02:25:33 2014
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 08 Dec 2014 23:25:33 +0000
Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch)
	November 30th-December 6th
Message-ID: <5486336d44e1e_a5711c53302017d@ip-10-69-141-60.mail>

New Post on freeswitch.org from kathleen
check it out at http://ift.tt/1AWUUpw
FreeSWITCH Week in Review (Master Branch) November 30th-December 6th
Hello, again. This week in the FreeSWITCH master branch we had 27 commits. The features this week are: timezone support for mod_say_{de,es,ja,nl,th,zh},? and the addition of a drop_dtmf_masking_tone channel variable which, if set, would replace all DTMF tones by the single tone specified in the variable.
New features that were added:

e55aee1 FS-7025 Add drop_dtmf_masking_tone channel_variable [Jira: http://ift.tt/1yJc1Mn]

a8c5a0c FS-7048 Add timezone support to mod_say_{de,es,ja,nl,th,zh}

Improvements in cross platform build supports:

dc9e904 FS-7025 fix compiler warning introduced from e55aee14 [Jira: http://ift.tt/1yJc1Mn]

b69c93e FS-7030 More work toward fixing FS build on Windows Visual Studio 2012 [Jira: http://ift.tt/1AWUUFW]

db66cdb Fix mrcp libraries to build correctly

c327455 FS-7030 More work toward getting FS to build on Windows Visual Studio 2012 [Jira: http://ift.tt/1AWUUFW]

b341ff7 FS-7046: fix data types and casting on some vars to silence windows build warnings in mod_verto [Jira: http://ift.tt/1yJc4rg]

7ce5171 FS-7046 follow up on type change in mod_verto [Jira: http://ift.tt/1yJc4rg]

The following bugs were squashed:

46adbec FS-7030 #comment [unimrcp] restore visual studio 2010/2012 project files added by FS project [Jira: http://ift.tt/1AWUUFW]

bad5dc3 FS-7037 Fix for T38 fax break started by commit 5bbef7f1e50 [Jira: http://ift.tt/1AWUXBF]

72c3df5 FS-6891 FS-6713 #comment revert 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira: http://ift.tt/1yJc22F]

2a7b022 FS-6980 #resolve don?t crash when using native recording on recordstop the redo [Jira: http://ift.tt/11E7i0U]

35ba6a3 FS-6766 Fix verto caller ringback missing on conference bridge in mod_verto [Jira: http://ift.tt/1AWUUG2]

e8cf9c7 FS-7045 Guarantee that dialed call can be joined when answered event is sent in mod_rayo [Jira: http://ift.tt/1yJc4rj]

4be6290 FS-7052 Moving jb queue swap operation out of the debug block. [Jira: http://ift.tt/1AWUXS0]

843e495 FS-7051 Preserve the annexb=no/yes status in mod_sangoma_codec [Jira: http://ift.tt/1yJc4rl]

158c1f2 FS-7002 Fix for recorded audio being choppy when diferent ptimes present and record session starts on bleg [Jira: http://ift.tt/1AWUXS3]

The complete list of commits can be found here:2014_11_30-2014_12_7

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From t.mahe at b-and-c.net  Tue Dec  9 02:28:33 2014
From: t.mahe at b-and-c.net (=?UTF-8?B?VHJpc3RhbiBNYWjDqQ==?=)
Date: Mon, 08 Dec 2014 15:28:33 -0800
Subject: [Freeswitch-users] Huge CPU LOAD increase between 1.4.12 and
 1.4.14
In-Reply-To: <CAKbxfG9qAnSOS0wEOLefafszWsz1SB_pbjq7mJTFH2Z1JSO0QA@mail.gmail.com>
References: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail>	<54827C1D.5000401@b-and-c.net>
	<548624D5.5050902@b-and-c.net>	<CA+GmxbYtoHfAK+W0=5cF+FBjGvSLGFKBepciDb-hBxHY7m_GZA@mail.gmail.com>	<54862ACA.6000506@b-and-c.net>
	<CAKbxfG9qAnSOS0wEOLefafszWsz1SB_pbjq7mJTFH2Z1JSO0QA@mail.gmail.com>
Message-ID: <54863421.1080809@b-and-c.net>

thanks Anthony, FS-7066 created with the content of the email.

Will try this patch on our test server with a little flow of trafic as
soon as I have the chance !

Le 08/12/2014 15:02, Anthony Minessale a ?crit :
> This should have been in JIRA... Drop the mentality that you have to
> make a case first to open a JIRA.
> The actual process should be, if its any question about how FS works, if
> you are right or wrong about it, file it as a JIRA.  If its wrong we can
> just close it........
> 
> If you want to send one email on the list to get some attention to your
> jira for discussion its ok but really you want to have the entire thing
> preserved there.
> 
> 
> I pushed a patch (4bcf1d888a4629993923527c26d4dbc6dae5d470) that might
> be related to your problem but now I have to wing it and use email where
> I get 200 emails a day and there is no tracking at all in email when it
> comes to issues so you put more resources on me to try to remember to
> look for your reply when we have a perfectly valid issue tracker system
> in place.....
> 
> 
> If master works better, good, if not, try 
> 
> git checkout 878a04715ae801ccd9587249240f5a6e4f16dd0e src/switch_time.c
>  and make install_core
> 
> All of this should be transposed back to JIRA by one of you.....
> 
> 
> 
> 
> On Mon, Dec 8, 2014 at 4:48 PM, Tristan Mah? <t.mahe at b-and-c.net
> <mailto:t.mahe at b-and-c.net>> wrote:
> 
>     Still on it, but I have some servers deployed without rtp-timer-name set
>     to soft ( blocking read is something I'd like to avoid, but needs to
>     find out what changed in FS code and also find out the impact on our
>     users ), and so far they have load avg divided by 10 with the same
>     amount of calls and the same trafic as those with rtp-timer-name to
>     soft.
> 
>     Had no issue with 1.4.12 as said previously, but 1.4.14 has some
>     interesting fixes that justify staying in production.
> 
>     Le 08/12/2014 14:33, Dmitry Sytchev a ?crit :
>     > So, have you solved your problem? Did you change rtp-timer-name?
>     > After we moved to 64-bit system, 1.4 branch shows huge LA spikes up to
>     > 35 on 4-core (or 4 single core virtual processor) with only 200-300
>     > concurrent channels and very low CPS (1-5).  On the same call load LA of
>     > 1.2 server is significantly lower.
>     > Setting 1.4 64bit to normal priority with -np helps to lower LA, but not
>     > completely solves a problem.
>     > It shows very big values in "ni" top fields (20-30%), but changing nice
>     > value of FS process to zero or positive value doesn't help.
>     >
>     >
>     >
>     >
>     > 2014-12-09 2:23 GMT+04:00 Tristan Mah? <t.mahe at b-and-c.net <mailto:t.mahe at b-and-c.net>
>     > <mailto:t.mahe at b-and-c.net <mailto:t.mahe at b-and-c.net>>>:
>     >
>     >     Hi everyone,
>     >
>     >     Just a little followup, this is caused by <param
>     name="rtp-timer-name"
>     >     value="soft"/> in a sip_profile. Happens on baremetal servers
>     as well as
>     >     Xen guest.
>     >
>     >     Still investigating on the exact root of this behaviour
>     change, but at
>     >     least now we know what is causing the trouble.
>     >
>     >     Best,
>     >
>     >     Tristan.
>     >
>     >     Le 05/12/2014 19:46, Tristan Mah? a ?crit :
>     >     > Hi guys,
>     >     >
>     >     > While trying to find out the root cause of this, I just want to
>     >     know if
>     >     > someone is also seeing a huge cpu load with freeswitch 1.4.14,
>     >     which was
>     >     > not present in 1.4.12.
>     >     >
>     >     > OS is debian wheezy, packages are official ones.
>     >     >
>     >     > Same config, same trafic pattern, just an upgrade from
>     1.4.12 ( no
>     >     other
>     >     > packages updated ).
>     >     >
>     >     > We were previously seeing load around 0.5, with pikes up to
>     around 2.
>     >     >
>     >     > We now are seeing a constant load of around 2, and pikes up
>     to 20...
>     >     >
>     >     > If someone is also seeing this, that would be interesting to
>     exchange,
>     >     > helping find out what is causing this !
>     >     >
>     >     > I'll file a jira once I find the root of this problem ( and no,
>     >     this is
>     >     > production servers, we can't put master there, we need stable ).
>     >     >
>     >     > Best,
>     >     >
>     >     >
>     >     >
>     >     >
>     >   
>      _________________________________________________________________________
>     >     > Professional FreeSWITCH Consulting Services:
>     >     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto: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>
>     >     <mailto: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>
>     <mailto: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>
>     >     <mailto: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
>     >
>     >
>     >
>     >
>     > --
>     > Best regards,
>     >
>     > Dmitry Sytchev,
>     > IT Engineer
>     >
>     >
>     >
>     _________________________________________________________________________
>     > 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
> 
> 
> 
> 
> -- 
> 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 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


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From umairedu at gmail.com  Tue Dec  9 13:31:41 2014
From: umairedu at gmail.com (Umair Khan)
Date: Tue, 9 Dec 2014 15:31:41 +0500
Subject: [Freeswitch-users] Billing software
In-Reply-To: <CAFLo2Qyd1uq+TOhc_sX5hahDku_DTHrQXdH3hXC+H5QFRLi4mg@mail.gmail.com>
References: <CAFLo2QzXJxyEDQ5YeNB3Engu4_LcTbdFELfXGhzgVChS4UoVbw@mail.gmail.com>
	<E84CC2DA-E8F2-4B02-8F09-B60D4406407E@gmail.com>
	<CAFLo2QyynMEY=NzJCZYmVAGFjiwNg7MgSGFrWCTWJYpyRnyCRQ@mail.gmail.com>
	<518CA7DD.8020802@gmail.com>
	<CAFLo2Qyd1uq+TOhc_sX5hahDku_DTHrQXdH3hXC+H5QFRLi4mg@mail.gmail.com>
Message-ID: <CAG21K9xxiUsu1bJn8o7B_OASFvJ73iDHS7_U36h+mtkM4f_n7Q@mail.gmail.com>

Hey Guys,

Checkout Hyperbilling.com for freeswitch billing.

Thanks & Regards

On Sun, Dec 7, 2014 at 7:58 PM, Luis Daniel Lucio Quiroz <
luis.daniel.lucio at gmail.com> wrote:

> Maybe a little late,
>
> Thanks everyone, but I am really against to install an extra freeswitch
> instance for the billing. It makes very complex the call flow,
> troubleshoting and in most cases FAX wont work.
>
> I did try the ASTPP before, horrible, hard to configure. I consider myself
> good on freeswitch, i know to code on perl, and a good sysadmin.  But after
> watching how complex is to get this to work I abort.
>
> So I decide to code my own billing and gladly, I can tell you that if you
> have all information needed (such as carrier rates and about 15 mins to get
> an explanation on how it works, you can have it working in less than an
> hour).  I started to study FusionPBX code, and after some months I have a
> working nice FusionPBX LCR+Billing native application.  It fully integrates
> in the Fusion, not out side, so you are able to bill for anything,
> incoming, internal (ext to ext, ext to conference, ext to ivr), or outgoing
> calls and it is able to get money by Paypal, Stripe (a good gw for
> CreditCards) and offline.  Currently, I have just release version 1.0.2
>
> If someone is interested, you can go to my website www.okay.com.mx/en  or
> contact me (gtalk prefered)
>
> LD
>
> 2013-05-10 3:55 GMT-04:00 Deon Vermeulen <vermeulen.deon at gmail.com>:
>
>> Hi Luis
>>
>> We just bought Commercial services from ASTPP.
>> When it comes to comparing price, features this is the BEST solution out
>> there at the moment.
>>
>> This is a very active project and you can contact Samir Doshi, Project
>> Maintainer & Developer,  directly for more information wrt your technical
>> questions.
>>
>> samir at astpp.org
>>
>>
>> Kind Regards
>> Deon
>>
>>  Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com>
>>  May 9, 2013 11:46 PM
>>
>>
>> Deon, thank you. Can you talk me about your exerience and how fast they
>> are on fixing bugs. I have a really good or bad luck to find bugs.
>> _________________________________________________________________________
>> 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
>>  Deon Vermeulen <vermeulen.deon at gmail.com>
>>  May 5, 2013 4:52 PM
>> Check out ASTPP
>>
>>
>> Kind Regards
>> Deon Vermeulen
>>
>> Sent from my iPhone
>>
>>  Luis Daniel Lucio Quiroz <luis.daniel.lucio at gmail.com>
>>  May 5, 2013 3:36 PM
>> What other options for FS compatible software other than vBilling do you
>> recommend me?
>> _________________________________________________________________________
>> 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
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From akhilgarg7 at gmail.com  Tue Dec  9 16:28:19 2014
From: akhilgarg7 at gmail.com (akhil garg)
Date: Tue, 9 Dec 2014 18:58:19 +0530
Subject: [Freeswitch-users] fs_cli hangs
Message-ID: <CA+mgNr50KX3Z9wY0h2iBAK+dtA1eEBvvuFEwNpKh2eCrvvwGcQ@mail.gmail.com>

freeswitch is running on 192.168.1.1 by command freeswitch.

file "freeswitch/autoload_configs/event_socket.conf.xml" has been edited
  <param name="listen-ip" value="172.26.1.47"/>


running "fs_cli -H 192.168.1.1 -d 7" from other machine e.g. 192.168.1.2 on
lan is successfull.

running "fs_cli -H 192.168.1.1 -d 7" from same machine i.e. 192.168.1.1
hangs.


Regards,
Akhil
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From akhilgarg7 at gmail.com  Tue Dec  9 16:28:49 2014
From: akhilgarg7 at gmail.com (akhil garg)
Date: Tue, 9 Dec 2014 18:58:49 +0530
Subject: [Freeswitch-users] fs_cli hangs
In-Reply-To: <CA+mgNr50KX3Z9wY0h2iBAK+dtA1eEBvvuFEwNpKh2eCrvvwGcQ@mail.gmail.com>
References: <CA+mgNr50KX3Z9wY0h2iBAK+dtA1eEBvvuFEwNpKh2eCrvvwGcQ@mail.gmail.com>
Message-ID: <CA+mgNr77GAKxMkjMqnjYozo7C7-+mXM6pBKnfNRtgJ_QDv385Q@mail.gmail.com>

I am using freeswitch 1.5.6b

On Tue, Dec 9, 2014 at 6:58 PM, akhil garg <akhilgarg7 at gmail.com> wrote:

> freeswitch is running on 192.168.1.1 by command freeswitch.
>
> file "freeswitch/autoload_configs/event_socket.conf.xml" has been edited
>   <param name="listen-ip" value="172.26.1.47"/>
>
>
> running "fs_cli -H 192.168.1.1 -d 7" from other machine e.g. 192.168.1.2
> on lan is successfull.
>
> running "fs_cli -H 192.168.1.1 -d 7" from same machine i.e. 192.168.1.1
> hangs.
>
>
> Regards,
> Akhil
>



-- 
regards,
akhil
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From vitaliy.davudov at vts24.ru  Tue Dec  9 16:46:05 2014
From: vitaliy.davudov at vts24.ru (Vitaly)
Date: Tue, 09 Dec 2014 16:46:05 +0300
Subject: [Freeswitch-users] Log-auth-failures question
In-Reply-To: <CAFiqYunBeRZMeG1p-Xo554jx0qQ1+m3E7fCY=1u-o+KNVdrs6w@mail.gmail.com>
References: <5485B391.5050506@vts24.ru>	<CAABKsuRkbTHU1bUAHF+HGsZmnP1k_p+g64dp5F69SAj66wY7tg@mail.gmail.com>
	<CAFiqYunBeRZMeG1p-Xo554jx0qQ1+m3E7fCY=1u-o+KNVdrs6w@mail.gmail.com>
Message-ID: <5486FD1D.4080900@vts24.ru>

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From blefko5361 at gmail.com  Tue Dec  9 18:19:32 2014
From: blefko5361 at gmail.com (Bruce Lefko)
Date: Tue, 9 Dec 2014 09:19:32 -0600
Subject: [Freeswitch-users] packaging FS debs using existing source tree
Message-ID: <CAJNDQt5AxJrF8wkFdsoRmRQNmi1kvhEA40QqLmZn-CF0E0mvaQ@mail.gmail.com>

I'd like to try making some changes to FS and then use the debian/util.sh
to create deb packages, however it looks like that script re-checkouts the
source to create the packages.

Is there a way to make that script use the existing source tree that's
already been checked out?

Thanks!
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From leo.noordergraaf at deanconnect.nl  Tue Dec  9 18:34:42 2014
From: leo.noordergraaf at deanconnect.nl (Leo Noordergraaf [Dean Connect])
Date: Tue, 9 Dec 2014 15:34:42 +0000
Subject: [Freeswitch-users] packaging FS debs using existing source tree
In-Reply-To: <CAJNDQt5AxJrF8wkFdsoRmRQNmi1kvhEA40QqLmZn-CF0E0mvaQ@mail.gmail.com>
References: <CAJNDQt5AxJrF8wkFdsoRmRQNmi1kvhEA40QqLmZn-CF0E0mvaQ@mail.gmail.com>
Message-ID: <1418139238.27339.3.camel@debian>

I solved this by checking in my changes in a branch.
Once everything is checked in, debian/util.sh works like a charm.
Leo

On Tue, 2014-12-09 at 09:19 -0600, Bruce Lefko wrote:
> I'd like to try making some changes to FS and then use the
> debian/util.sh to create deb packages, however it looks like that
> script re-checkouts the source to create the packages.
> 
> 
> Is there a way to make that script use the existing source tree that's
> already been checked out?
> 
> 
> Thanks!


From blefko5361 at gmail.com  Tue Dec  9 18:37:32 2014
From: blefko5361 at gmail.com (Bruce Lefko)
Date: Tue, 9 Dec 2014 09:37:32 -0600
Subject: [Freeswitch-users] packaging FS debs using existing source tree
In-Reply-To: <1418139238.27339.3.camel@debian>
References: <CAJNDQt5AxJrF8wkFdsoRmRQNmi1kvhEA40QqLmZn-CF0E0mvaQ@mail.gmail.com>
	<1418139238.27339.3.camel@debian>
Message-ID: <CAJNDQt7_38Yabuqsnk9hQ4doWZV0+RG-JoU2Bv6ncXZc3GZeWA@mail.gmail.com>

So just to confirm, as long as I check in my changes locally (on the
machine where I'm building), I can just run debian/util.sh and it'll pick
up my changes?  I don't have to push my changes out to stash or anything
right?

Thanks!

On Tue, Dec 9, 2014 at 9:34 AM, Leo Noordergraaf [Dean Connect] <
leo.noordergraaf at deanconnect.nl> wrote:

> I solved this by checking in my changes in a branch.
> Once everything is checked in, debian/util.sh works like a charm.
> Leo
>
> On Tue, 2014-12-09 at 09:19 -0600, Bruce Lefko wrote:
> > I'd like to try making some changes to FS and then use the
> > debian/util.sh to create deb packages, however it looks like that
> > script re-checkouts the source to create the packages.
> >
> >
> > Is there a way to make that script use the existing source tree that's
> > already been checked out?
> >
> >
> > 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 leo.noordergraaf at deanconnect.nl  Tue Dec  9 18:54:43 2014
From: leo.noordergraaf at deanconnect.nl (Leo Noordergraaf [Dean Connect])
Date: Tue, 9 Dec 2014 15:54:43 +0000
Subject: [Freeswitch-users] packaging FS debs using existing source tree
In-Reply-To: <CAJNDQt7_38Yabuqsnk9hQ4doWZV0+RG-JoU2Bv6ncXZc3GZeWA@mail.gmail.com>
References: <CAJNDQt5AxJrF8wkFdsoRmRQNmi1kvhEA40QqLmZn-CF0E0mvaQ@mail.gmail.com>
	<1418139238.27339.3.camel@debian>
	<CAJNDQt7_38Yabuqsnk9hQ4doWZV0+RG-JoU2Bv6ncXZc3GZeWA@mail.gmail.com>
Message-ID: <1418140438.27339.12.camel@debian>

That's right, you don't have to push anything.
Later you can do 'git pull' and your changes are automatically merged.
Actually, I didn't even use a separate branch for the check-in.
Leo

On Tue, 2014-12-09 at 09:37 -0600, Bruce Lefko wrote:
> So just to confirm, as long as I check in my changes locally (on the machine where I'm building), I can just run debian/util.sh and it'll pick up my changes?  I don't have to push my changes out to stash or anything right?
> 
> Thanks!
> 
> On Tue, Dec 9, 2014 at 9:34 AM, Leo Noordergraaf [Dean Connect] <leo.noordergraaf at deanconnect.nl> wrote:
> 
> 
> 	I solved this by checking in my changes in a branch.
> 	Once everything is checked in, debian/util.sh works like a charm.
> 	Leo
> 	
> 
> 	On Tue, 2014-12-09 at 09:19 -0600, Bruce Lefko wrote:
> 	> I'd like to try making some changes to FS and then use the
> 	> debian/util.sh to create deb packages, however it looks like that
> 	> script re-checkouts the source to create the packages.
> 	>
> 	>
> 	> Is there a way to make that script use the existing source tree that's
> 	> already been checked out?
> 	>
> 	>
> 	> 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 luis.daniel.lucio at gmail.com  Tue Dec  9 19:15:12 2014
From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz)
Date: Tue, 9 Dec 2014 11:15:12 -0500
Subject: [Freeswitch-users] Billing software
In-Reply-To: <CAG21K9xxiUsu1bJn8o7B_OASFvJ73iDHS7_U36h+mtkM4f_n7Q@mail.gmail.com>
References: <CAFLo2QzXJxyEDQ5YeNB3Engu4_LcTbdFELfXGhzgVChS4UoVbw@mail.gmail.com>
	<E84CC2DA-E8F2-4B02-8F09-B60D4406407E@gmail.com>
	<CAFLo2QyynMEY=NzJCZYmVAGFjiwNg7MgSGFrWCTWJYpyRnyCRQ@mail.gmail.com>
	<518CA7DD.8020802@gmail.com>
	<CAFLo2Qyd1uq+TOhc_sX5hahDku_DTHrQXdH3hXC+H5QFRLi4mg@mail.gmail.com>
	<CAG21K9xxiUsu1bJn8o7B_OASFvJ73iDHS7_U36h+mtkM4f_n7Q@mail.gmail.com>
Message-ID: <CAFLo2Qx6F7qn2=3FLYR0JbGMxE7GKi2juWXBs6maswJ0fkdmCA@mail.gmail.com>

2014-12-09 5:31 GMT-05:00 Umair Khan <umairedu at gmail.com>:

> Hyperbilling.com


wow  http://www.hyperbilling.com/pricing/

that is expensive!!

Mine is 49.99 monthly, the current call limit is set by your hardware not a
license
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From brian at freeswitch.org  Tue Dec  9 21:09:01 2014
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2014 12:09:01 -0600
Subject: [Freeswitch-users] fs_cli hangs
In-Reply-To: <CA+mgNr50KX3Z9wY0h2iBAK+dtA1eEBvvuFEwNpKh2eCrvvwGcQ@mail.gmail.com>
References: <CA+mgNr50KX3Z9wY0h2iBAK+dtA1eEBvvuFEwNpKh2eCrvvwGcQ@mail.gmail.com>
Message-ID: <CAEJMVkAHSYWFzimvJ11Fw_K+XQvEt8FimdU+Ufj4NSEzg8L0rQ@mail.gmail.com>

on debian?

On Tue, Dec 9, 2014 at 7:28 AM, akhil garg <akhilgarg7 at gmail.com> wrote:

> freeswitch is running on 192.168.1.1 by command freeswitch.
>
> file "freeswitch/autoload_configs/event_socket.conf.xml" has been edited
>   <param name="listen-ip" value="172.26.1.47"/>
>
>
> running "fs_cli -H 192.168.1.1 -d 7" from other machine e.g. 192.168.1.2
> on lan is successfull.
>
> running "fs_cli -H 192.168.1.1 -d 7" from same machine i.e. 192.168.1.1
> hangs.
>
>
> Regards,
> Akhil
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From steveayre at gmail.com  Wed Dec 10 00:28:06 2014
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 9 Dec 2014 21:28:06 +0000
Subject: [Freeswitch-users] fs_cli hangs
In-Reply-To: <CA+mgNr50KX3Z9wY0h2iBAK+dtA1eEBvvuFEwNpKh2eCrvvwGcQ@mail.gmail.com>
References: <CA+mgNr50KX3Z9wY0h2iBAK+dtA1eEBvvuFEwNpKh2eCrvvwGcQ@mail.gmail.com>
Message-ID: <CAFiqYuk1L3H+3TZBnb3dOfQDroY=tXkzkSDEqSZ-xPBzzJC7jg@mail.gmail.com>

Do you have a firewall protecting the port? If so perhaps it's blocking the
local connection based on the IP you connect from.

Does it eventually time out, eg after 30-60 seconds? Or hang forever?



On 9 December 2014 at 13:28, akhil garg <akhilgarg7 at gmail.com> wrote:

> freeswitch is running on 192.168.1.1 by command freeswitch.
>
> file "freeswitch/autoload_configs/event_socket.conf.xml" has been edited
>   <param name="listen-ip" value="172.26.1.47"/>
>
>
> running "fs_cli -H 192.168.1.1 -d 7" from other machine e.g. 192.168.1.2
> on lan is successfull.
>
> running "fs_cli -H 192.168.1.1 -d 7" from same machine i.e. 192.168.1.1
> hangs.
>
>
> Regards,
> Akhil
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joelewhite at gmail.com  Wed Dec 10 00:47:21 2014
From: joelewhite at gmail.com (Joel White)
Date: Tue, 9 Dec 2014 16:47:21 -0500
Subject: [Freeswitch-users] Caller ID
In-Reply-To: <CACOYK=gFNp+Asi37N_FBUu_BVXYcreyLgGhv-5ozHVrK=0AHcQ@mail.gmail.com>
References: <CAK0YwNkxp7ZyKZRtTxROGmcNEXBUrj49hSX+tM2iQmFgZtBBxQ@mail.gmail.com>
	<CACOYK=gFNp+Asi37N_FBUu_BVXYcreyLgGhv-5ozHVrK=0AHcQ@mail.gmail.com>
Message-ID: <CAK0YwNnNNj9eAQCY368c4ABSdxM=A2De-O5ow6R8hq6X0PzD_Q@mail.gmail.com>

Sorry for the delay in response.  I have gone over the config with a fine
tooth comb, it matches another config of a server in which the caller id
works fine.  What I am seeing however is that in this system the variable
is not exported to the dialplan.  I may be missing something, and most
likely I am.  I do have a question though.  Is there a way to see what
variables are defined for a particular user in the FreeSWITCH console?



On Thu, Nov 13, 2014 at 1:56 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
wrote:

> Probably your ITSP does not allow you to set the caller ID? Did you run a
> SIP packet trace to see what caller ID is sent out?
> On Nov 13, 2014 4:45 PM, "Joel White" <joelewhite at gmail.com> wrote:
>
>> I have several installations of FreeSWITCH.  I have managed to get one to
>> dynamically create the user directory from PostgreSQL and it properly sets
>> the outbound caller id.  I have another system, very similar in how it was
>> configured.  The second installation also dynamically generates the user
>> directory from PostgreSQL as well.  I am however having an issue with
>> setting the caller id on this other system.  I have went through with a
>> fine tooth comb and am having a hard time locating the discrepancy between
>> the two systems.
>>
>> Has anyone had an issue similar to this?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joelewhite at gmail.com  Wed Dec 10 00:59:14 2014
From: joelewhite at gmail.com (Joel White)
Date: Tue, 9 Dec 2014 16:59:14 -0500
Subject: [Freeswitch-users] Caller ID
In-Reply-To: <CAK0YwNnNNj9eAQCY368c4ABSdxM=A2De-O5ow6R8hq6X0PzD_Q@mail.gmail.com>
References: <CAK0YwNkxp7ZyKZRtTxROGmcNEXBUrj49hSX+tM2iQmFgZtBBxQ@mail.gmail.com>
	<CACOYK=gFNp+Asi37N_FBUu_BVXYcreyLgGhv-5ozHVrK=0AHcQ@mail.gmail.com>
	<CAK0YwNnNNj9eAQCY368c4ABSdxM=A2De-O5ow6R8hq6X0PzD_Q@mail.gmail.com>
Message-ID: <CAK0YwNnofeJOi4UJ6k9Q8g+4ZaRYc7ssYNigD2-XYodhRn-Eog@mail.gmail.com>

Here is some output of the Lua script on the server not pushing caller id

2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from
gen_dir_user_xml.lua, generated XML:
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="voip.net">
      <user id="26342" mailbox="26342">
        <params>
          <param name="password" value="XXXXXX"/>
          <param name="vm-password" value="26342"/>
          <param name="vm-enabled" value="true"/>
          <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
    </params>
        <variables>
          <variable name="toll_allow" value="local,domestic"/>
          <variable name="user_context" value="default"/>
          <variable name="outbound_caller_id_name" value="Michelle"/>
          <variable name="outbound_caller_id_number" value="1321XXXXXXX"/>
      <variable name="accountcode" value="PA"/>
      <!--<variable name="callgroup" value="techsupport"/>-->
        </variables>
      </user>
    </domain>
  </section>
</document>


And some output from the Lua script on the server with CID functioning

2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from
gen_dir_user_xml.lua, generated XML:
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="sip.voice.com">
      <user id="53201" mailbox="53201">
        <params>
          <param name="password" value="XXXXXXX"/>
          <param name="vm-password" value="53201"/>
          <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
        </params>
        <variables>
          <variable name="toll_allow" value="local,domestic"/>
          <variable name="user_context" value="default"/>
          <variable name="outbound_caller_id_name" value="Voice.com"/>
          <variable name="outbound_caller_id_number" value="1303XXXXXX"/>
          <!--<variable name="callgroup" value="techsupport"/>-->
        </variables>
      </user>
    </domain>
  </section>
</document>


Of course I removed any identifiable information, but it looks like the CID
is being set.  What am I missing here that is not allowing for the variable
to be passed to the dialplan?

On Tue, Dec 9, 2014 at 4:47 PM, Joel White <joelewhite at gmail.com> wrote:

> Sorry for the delay in response.  I have gone over the config with a fine
> tooth comb, it matches another config of a server in which the caller id
> works fine.  What I am seeing however is that in this system the variable
> is not exported to the dialplan.  I may be missing something, and most
> likely I am.  I do have a question though.  Is there a way to see what
> variables are defined for a particular user in the FreeSWITCH console?
>
>
>
> On Thu, Nov 13, 2014 at 1:56 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> Probably your ITSP does not allow you to set the caller ID? Did you run a
>> SIP packet trace to see what caller ID is sent out?
>> On Nov 13, 2014 4:45 PM, "Joel White" <joelewhite at gmail.com> wrote:
>>
>>> I have several installations of FreeSWITCH.  I have managed to get one
>>> to dynamically create the user directory from PostgreSQL and it properly
>>> sets the outbound caller id.  I have another system, very similar in how it
>>> was configured.  The second installation also dynamically generates the
>>> user directory from PostgreSQL as well.  I am however having an issue with
>>> setting the caller id on this other system.  I have went through with a
>>> fine tooth comb and am having a hard time locating the discrepancy between
>>> the two systems.
>>>
>>> Has anyone had an issue similar to this?
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From joelewhite at gmail.com  Wed Dec 10 01:02:56 2014
From: joelewhite at gmail.com (Joel White)
Date: Tue, 9 Dec 2014 17:02:56 -0500
Subject: [Freeswitch-users] Caller ID
In-Reply-To: <CAK0YwNnofeJOi4UJ6k9Q8g+4ZaRYc7ssYNigD2-XYodhRn-Eog@mail.gmail.com>
References: <CAK0YwNkxp7ZyKZRtTxROGmcNEXBUrj49hSX+tM2iQmFgZtBBxQ@mail.gmail.com>
	<CACOYK=gFNp+Asi37N_FBUu_BVXYcreyLgGhv-5ozHVrK=0AHcQ@mail.gmail.com>
	<CAK0YwNnNNj9eAQCY368c4ABSdxM=A2De-O5ow6R8hq6X0PzD_Q@mail.gmail.com>
	<CAK0YwNnofeJOi4UJ6k9Q8g+4ZaRYc7ssYNigD2-XYodhRn-Eog@mail.gmail.com>
Message-ID: <CAK0YwNmTgU+T5BM_oysYDvFK9ZA_8aOnHmY=M+vjCXgpQ4kH=g@mail.gmail.com>

This is what I get when the dialplan executes

EXECUTE sofia/internal/26342 at voip.net
set(effective_caller_id_number=)
2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/
26342 at voip.net SET [effective_caller_id_number]=[UNDEF]


Kinda strange and I could not find a discrepancy between the dialplan
configuration of the working server vs the non-working server



On Tue, Dec 9, 2014 at 4:59 PM, Joel White <joelewhite at gmail.com> wrote:

> Here is some output of the Lua script on the server not pushing caller id
>
> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from
> gen_dir_user_xml.lua, generated XML:
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="voip.net">
>       <user id="26342" mailbox="26342">
>         <params>
>           <param name="password" value="XXXXXX"/>
>           <param name="vm-password" value="26342"/>
>           <param name="vm-enabled" value="true"/>
>           <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>     </params>
>         <variables>
>           <variable name="toll_allow" value="local,domestic"/>
>           <variable name="user_context" value="default"/>
>           <variable name="outbound_caller_id_name" value="Michelle"/>
>           <variable name="outbound_caller_id_number" value="1321XXXXXXX"/>
>       <variable name="accountcode" value="PA"/>
>       <!--<variable name="callgroup" value="techsupport"/>-->
>         </variables>
>       </user>
>     </domain>
>   </section>
> </document>
>
>
> And some output from the Lua script on the server with CID functioning
>
> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from
> gen_dir_user_xml.lua, generated XML:
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="sip.voice.com">
>       <user id="53201" mailbox="53201">
>         <params>
>           <param name="password" value="XXXXXXX"/>
>           <param name="vm-password" value="53201"/>
>           <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>         </params>
>         <variables>
>           <variable name="toll_allow" value="local,domestic"/>
>           <variable name="user_context" value="default"/>
>           <variable name="outbound_caller_id_name" value="Voice.com"/>
>           <variable name="outbound_caller_id_number" value="1303XXXXXX"/>
>           <!--<variable name="callgroup" value="techsupport"/>-->
>         </variables>
>       </user>
>     </domain>
>   </section>
> </document>
>
>
> Of course I removed any identifiable information, but it looks like the
> CID is being set.  What am I missing here that is not allowing for the
> variable to be passed to the dialplan?
>
> On Tue, Dec 9, 2014 at 4:47 PM, Joel White <joelewhite at gmail.com> wrote:
>
>> Sorry for the delay in response.  I have gone over the config with a fine
>> tooth comb, it matches another config of a server in which the caller id
>> works fine.  What I am seeing however is that in this system the variable
>> is not exported to the dialplan.  I may be missing something, and most
>> likely I am.  I do have a question though.  Is there a way to see what
>> variables are defined for a particular user in the FreeSWITCH console?
>>
>>
>>
>> On Thu, Nov 13, 2014 at 1:56 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
>> wrote:
>>
>>> Probably your ITSP does not allow you to set the caller ID? Did you run
>>> a SIP packet trace to see what caller ID is sent out?
>>> On Nov 13, 2014 4:45 PM, "Joel White" <joelewhite at gmail.com> wrote:
>>>
>>>> I have several installations of FreeSWITCH.  I have managed to get one
>>>> to dynamically create the user directory from PostgreSQL and it properly
>>>> sets the outbound caller id.  I have another system, very similar in how it
>>>> was configured.  The second installation also dynamically generates the
>>>> user directory from PostgreSQL as well.  I am however having an issue with
>>>> setting the caller id on this other system.  I have went through with a
>>>> fine tooth comb and am having a hard time locating the discrepancy between
>>>> the two systems.
>>>>
>>>> Has anyone had an issue similar to this?
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
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From joelewhite at gmail.com  Wed Dec 10 01:06:29 2014
From: joelewhite at gmail.com (Joel White)
Date: Tue, 9 Dec 2014 17:06:29 -0500
Subject: [Freeswitch-users] Caller ID
In-Reply-To: <CAK0YwNmTgU+T5BM_oysYDvFK9ZA_8aOnHmY=M+vjCXgpQ4kH=g@mail.gmail.com>
References: <CAK0YwNkxp7ZyKZRtTxROGmcNEXBUrj49hSX+tM2iQmFgZtBBxQ@mail.gmail.com>
	<CACOYK=gFNp+Asi37N_FBUu_BVXYcreyLgGhv-5ozHVrK=0AHcQ@mail.gmail.com>
	<CAK0YwNnNNj9eAQCY368c4ABSdxM=A2De-O5ow6R8hq6X0PzD_Q@mail.gmail.com>
	<CAK0YwNnofeJOi4UJ6k9Q8g+4ZaRYc7ssYNigD2-XYodhRn-Eog@mail.gmail.com>
	<CAK0YwNmTgU+T5BM_oysYDvFK9ZA_8aOnHmY=M+vjCXgpQ4kH=g@mail.gmail.com>
Message-ID: <CAK0YwNmNJHdEMLKihywttEO3PDe3MXHv7dF7x=-oj8XqmE7W9A@mail.gmail.com>

Here is the version running on the server that works properly

FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git
b942d0f 2014-11-03 19:53:00Z 64bit)

And the version of the server having issue with CID

FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git
b942d0f 2014-11-03 19:53:00Z 64bit)

On Tue, Dec 9, 2014 at 5:02 PM, Joel White <joelewhite at gmail.com> wrote:

> This is what I get when the dialplan executes
>
> EXECUTE sofia/internal/26342 at voip.net
> set(effective_caller_id_number=)
> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/
> 26342 at voip.net SET [effective_caller_id_number]=[UNDEF]
>
>
> Kinda strange and I could not find a discrepancy between the dialplan
> configuration of the working server vs the non-working server
>
>
>
> On Tue, Dec 9, 2014 at 4:59 PM, Joel White <joelewhite at gmail.com> wrote:
>
>> Here is some output of the Lua script on the server not pushing caller id
>>
>> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from
>> gen_dir_user_xml.lua, generated XML:
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>>   <section name="directory">
>>     <domain name="voip.net">
>>       <user id="26342" mailbox="26342">
>>         <params>
>>           <param name="password" value="XXXXXX"/>
>>           <param name="vm-password" value="26342"/>
>>           <param name="vm-enabled" value="true"/>
>>           <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>     </params>
>>         <variables>
>>           <variable name="toll_allow" value="local,domestic"/>
>>           <variable name="user_context" value="default"/>
>>           <variable name="outbound_caller_id_name" value="Michelle"/>
>>           <variable name="outbound_caller_id_number" value="1321XXXXXXX"/>
>>       <variable name="accountcode" value="PA"/>
>>       <!--<variable name="callgroup" value="techsupport"/>-->
>>         </variables>
>>       </user>
>>     </domain>
>>   </section>
>> </document>
>>
>>
>> And some output from the Lua script on the server with CID functioning
>>
>> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from
>> gen_dir_user_xml.lua, generated XML:
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>>   <section name="directory">
>>     <domain name="sip.voice.com">
>>       <user id="53201" mailbox="53201">
>>         <params>
>>           <param name="password" value="XXXXXXX"/>
>>           <param name="vm-password" value="53201"/>
>>           <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>         </params>
>>         <variables>
>>           <variable name="toll_allow" value="local,domestic"/>
>>           <variable name="user_context" value="default"/>
>>           <variable name="outbound_caller_id_name" value="Voice.com"/>
>>           <variable name="outbound_caller_id_number" value="1303XXXXXX"/>
>>           <!--<variable name="callgroup" value="techsupport"/>-->
>>         </variables>
>>       </user>
>>     </domain>
>>   </section>
>> </document>
>>
>>
>> Of course I removed any identifiable information, but it looks like the
>> CID is being set.  What am I missing here that is not allowing for the
>> variable to be passed to the dialplan?
>>
>> On Tue, Dec 9, 2014 at 4:47 PM, Joel White <joelewhite at gmail.com> wrote:
>>
>>> Sorry for the delay in response.  I have gone over the config with a
>>> fine tooth comb, it matches another config of a server in which the caller
>>> id works fine.  What I am seeing however is that in this system the
>>> variable is not exported to the dialplan.  I may be missing something, and
>>> most likely I am.  I do have a question though.  Is there a way to see what
>>> variables are defined for a particular user in the FreeSWITCH console?
>>>
>>>
>>>
>>> On Thu, Nov 13, 2014 at 1:56 PM, Stanislav Sinyagin <ssinyagin at gmail.com
>>> > wrote:
>>>
>>>> Probably your ITSP does not allow you to set the caller ID? Did you run
>>>> a SIP packet trace to see what caller ID is sent out?
>>>> On Nov 13, 2014 4:45 PM, "Joel White" <joelewhite at gmail.com> wrote:
>>>>
>>>>> I have several installations of FreeSWITCH.  I have managed to get one
>>>>> to dynamically create the user directory from PostgreSQL and it properly
>>>>> sets the outbound caller id.  I have another system, very similar in how it
>>>>> was configured.  The second installation also dynamically generates the
>>>>> user directory from PostgreSQL as well.  I am however having an issue with
>>>>> setting the caller id on this other system.  I have went through with a
>>>>> fine tooth comb and am having a hard time locating the discrepancy between
>>>>> the two systems.
>>>>>
>>>>> Has anyone had an issue similar to this?
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>
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From joseph at onsip.com  Wed Dec 10 01:30:49 2014
From: joseph at onsip.com (Joseph Frazier)
Date: Tue, 9 Dec 2014 17:30:49 -0500
Subject: [Freeswitch-users] Incomplete UPDATEs after Attended Transfer
In-Reply-To: <CAANWw7G0BkpbYrF3JSSkUgrtYyqZNtpiw0j548_PwMG1B4Faaw@mail.gmail.com>
References: <CAANWw7G0BkpbYrF3JSSkUgrtYyqZNtpiw0j548_PwMG1B4Faaw@mail.gmail.com>
Message-ID: <CAANWw7GVqRt6N-Eq+1oYxMhjAhA2mO6QK1wkErwjaMtexW=4pA@mail.gmail.com>

Hi everyone,

I'm trying to do attended transfers on my FS 1.5.14 box and have the
endpoints receive UPDATE requests indicating that the identity of the other
leg has changed. I installed the v1.5.14 tag from source and uncommented
the following line of sip_profiles/internal.xml:

   <param name="NDLB-force-rport" value="true"/>

Other than the above line, my configuration is completely vanilla.

I have three extensions registered: 1000, 1007, 1006. My scenario is:

   - 1000 calls 1007
   - 1000 calls 1006
   - 1000 attended transfers 1007 to 1006

Here is my fs_cli with the following debugging options:

sofia global siptrace on
> sofia loglevel all 9
> sofia tracelevel alert
> console loglevel debug
> fsctl debug_level 10


https://gist.github.com/joseph-onsip/de90a66afdf37994c334

After receiving the REFER, FS sends an UPDATE to the refer target, but not
to the referee. Is there a configuration option I can set in order to have
an UPDATE sent to the referee as well?

Thanks,
Joseph
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From krice at freeswitch.org  Wed Dec 10 02:40:20 2014
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 09 Dec 2014 23:40:20 +0000
Subject: [Freeswitch-users] Happy holidays!
Message-ID: <548788643371_3cb969133813668@ip-10-136-113-197.mail>

New Post on freeswitch.org from kathleen
check it out at http://ift.tt/1sgteqg
Happy holidays!
Hello, FreeSWITCH community! As I am sure you are all aware, the holiday season is upon us! The developers have been busy moving FreeSWITCH in some exciting new directions and I know some of you are very excited about some of the results. If you are feeling particularly appreciative, I?ve provided a few links to the wish-lists of the developers. These lists will make it easy to spread the holiday cheer and show just how much you are enjoying FreeSWITCH. Happy holidays!

Anthony Minessale (anthm) ? http://ift.tt/1IwrlQE

Mike Jerris (MikeJ) ? http://ift.tt/1Iwrkwa

Brian West (bkw_) ? http://ift.tt/1uje6sN

Ken Rice (SwK) ? http://ift.tt/1IwrlQI

William King (quentusrex) ? http://ift.tt/1IwrkMo

?

-Kathleen

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From brian at freeswitch.org  Wed Dec 10 02:54:22 2014
From: brian at freeswitch.org (Brian West)
Date: Tue, 9 Dec 2014 17:54:22 -0600
Subject: [Freeswitch-users] Incomplete UPDATEs after Attended Transfer
In-Reply-To: <CAANWw7GVqRt6N-Eq+1oYxMhjAhA2mO6QK1wkErwjaMtexW=4pA@mail.gmail.com>
References: <CAANWw7G0BkpbYrF3JSSkUgrtYyqZNtpiw0j548_PwMG1B4Faaw@mail.gmail.com>
	<CAANWw7GVqRt6N-Eq+1oYxMhjAhA2mO6QK1wkErwjaMtexW=4pA@mail.gmail.com>
Message-ID: <CAEJMVkBV=sAEvoxOFwWJV3PghopL_NECaJAxAy7e4XcsLG2Rnw@mail.gmail.com>

What device is the Referee?

On Tue, Dec 9, 2014 at 4:30 PM, Joseph Frazier <joseph at onsip.com> wrote:

> Hi everyone,
>
> I'm trying to do attended transfers on my FS 1.5.14 box and have the
> endpoints receive UPDATE requests indicating that the identity of the other
> leg has changed. I installed the v1.5.14 tag from source and uncommented
> the following line of sip_profiles/internal.xml:
>
>    <param name="NDLB-force-rport" value="true"/>
>
> Other than the above line, my configuration is completely vanilla.
>
> I have three extensions registered: 1000, 1007, 1006. My scenario is:
>
>    - 1000 calls 1007
>    - 1000 calls 1006
>    - 1000 attended transfers 1007 to 1006
>
> Here is my fs_cli with the following debugging options:
>
> sofia global siptrace on
>> sofia loglevel all 9
>> sofia tracelevel alert
>> console loglevel debug
>> fsctl debug_level 10
>
>
> https://gist.github.com/joseph-onsip/de90a66afdf37994c334
>
> After receiving the REFER, FS sends an UPDATE to the refer target, but not
> to the referee. Is there a configuration option I can set in order to have
> an UPDATE sent to the referee as well?
>
> Thanks,
> Joseph
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From joseph at onsip.com  Wed Dec 10 05:13:25 2014
From: joseph at onsip.com (Joseph Frazier)
Date: Tue, 9 Dec 2014 21:13:25 -0500
Subject: [Freeswitch-users] Incomplete UPDATEs after Attended Transfer
In-Reply-To: <CAEJMVkBV=sAEvoxOFwWJV3PghopL_NECaJAxAy7e4XcsLG2Rnw@mail.gmail.com>
References: <CAANWw7G0BkpbYrF3JSSkUgrtYyqZNtpiw0j548_PwMG1B4Faaw@mail.gmail.com>
	<CAANWw7GVqRt6N-Eq+1oYxMhjAhA2mO6QK1wkErwjaMtexW=4pA@mail.gmail.com>
	<CAEJMVkBV=sAEvoxOFwWJV3PghopL_NECaJAxAy7e4XcsLG2Rnw@mail.gmail.com>
Message-ID: <CAANWw7EoJW84KDzJewdds3oa9taC8uTBA76zfKoWfDKG7PQyVg@mail.gmail.com>

The Referee is a "Grandstream GXV3275 1.0.2.15". For context, here's where
it OKs the INVITE from the first call:

https://gist.github.com/joseph-onsip/de90a66afdf37994c334#file-gistfile1-txt-L1813-L1838

Thanks,
Joseph

On Tue, Dec 9, 2014 at 6:54 PM, Brian West <brian at freeswitch.org> wrote:

> What device is the Referee?
>
> On Tue, Dec 9, 2014 at 4:30 PM, Joseph Frazier <joseph at onsip.com> wrote:
>
>> Hi everyone,
>>
>> I'm trying to do attended transfers on my FS 1.5.14 box and have the
>> endpoints receive UPDATE requests indicating that the identity of the other
>> leg has changed. I installed the v1.5.14 tag from source and uncommented
>> the following line of sip_profiles/internal.xml:
>>
>>    <param name="NDLB-force-rport" value="true"/>
>>
>> Other than the above line, my configuration is completely vanilla.
>>
>> I have three extensions registered: 1000, 1007, 1006. My scenario is:
>>
>>    - 1000 calls 1007
>>    - 1000 calls 1006
>>    - 1000 attended transfers 1007 to 1006
>>
>> Here is my fs_cli with the following debugging options:
>>
>> sofia global siptrace on
>>> sofia loglevel all 9
>>> sofia tracelevel alert
>>> console loglevel debug
>>> fsctl debug_level 10
>>
>>
>> https://gist.github.com/joseph-onsip/de90a66afdf37994c334
>>
>> After receiving the REFER, FS sends an UPDATE to the refer target, but
>> not to the referee. Is there a configuration option I can set in order to
>> have an UPDATE sent to the referee as well?
>>
>> Thanks,
>> Joseph
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From avi at avimarcus.net  Wed Dec 10 08:50:53 2014
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 10 Dec 2014 05:50:53 +0000
Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound
 Caller ID to external SIP Gateway URI
In-Reply-To: <CACK8u8J1b1UcEQ1R+M0puRT9vACtYGeuKBhHs_2ug9Q_qgJ6Wg@mail.gmail.com>
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com>
	<0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com>
	<CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com>
	<0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com>
	<CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com>
	<0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com>
	<CACK8u8J1b1UcEQ1R+M0puRT9vACtYGeuKBhHs_2ug9Q_qgJ6Wg@mail.gmail.com>
Message-ID: <0000014a32c0097e-f9c55def-289f-48c9-bd1f-00e755c51f28-000000@email.amazonses.com>

I would try something "simpler". This will always play the file, though, no
matter the response from the othersipgw.


<extension name="1212">
       <action application="set" data="hangup_after_bridge=false"/>
       <action application="bridge"
data="sofia/external/sip:user.name at sip.othersipgw.ip.address.com
;${effective_caller_id_number}"/>
       <action application="transfer" data="1213"/>
 </condition>
</extension>

<!-- What to do after othersipgw drops the connection-->
<extension name="1213">
  <condition field="destination_number" expression="1213">
    <action application="playback" data="ivr/ivr-thank_you.wav"/>
   </condition>
</extension>
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From akhilgarg7 at gmail.com  Wed Dec 10 09:16:56 2014
From: akhilgarg7 at gmail.com (akhil garg)
Date: Wed, 10 Dec 2014 11:46:56 +0530
Subject: [Freeswitch-users] fs_cli hangs
Message-ID: <CA+mgNr6spPPvLM1YLu66=dxfMGRBce-o_7SqnCn=y53prgazQA@mail.gmail.com>

It hangs forever.

command "netstat -anlp | grep 8021" gives the following output.

tcp        0      0 127.0.0.1:34893         127.0.0.1:8021
 ESTABLISHED 2020/fs_cli
tcp        0     54 127.0.0.1:8021          127.0.0.1:34893
ESTABLISHED 1862/freeswitch


-- 
regards,
akhil
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From avi at avimarcus.net  Wed Dec 10 14:13:02 2014
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 10 Dec 2014 11:13:02 +0000
Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound
 Caller ID to external SIP Gateway URI
In-Reply-To: <CACK8u8K+DA7qGEHoYnpHms_tetv-wQaQpafEv+e5MPZcmvizOg@mail.gmail.com>
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com>
	<0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com>
	<CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com>
	<0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com>
	<CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com>
	<0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com>
	<CACK8u8J1b1UcEQ1R+M0puRT9vACtYGeuKBhHs_2ug9Q_qgJ6Wg@mail.gmail.com>
	<0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com>
	<CACK8u8K+DA7qGEHoYnpHms_tetv-wQaQpafEv+e5MPZcmvizOg@mail.gmail.com>
Message-ID: <0000014a33e6f7d3-05962ccb-d0e7-424d-aafc-a4dcd666d72f-000000@email.amazonses.com>

Your log still shows: Action set(exec_after_bridge_arg=1213)

... and if they set a 180, then you can set the ringback.
-Avi
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From italorossib at gmail.com  Wed Dec 10 14:47:30 2014
From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=)
Date: Wed, 10 Dec 2014 08:47:30 -0300
Subject: [Freeswitch-users] Doc-Sprint Friday 12 December 2014
In-Reply-To: <CA+fkitLK8w7h1a+kiQr=vQ-wt0oSGKaQ7q4HWhbep52FS65kMA@mail.gmail.com>
References: <CAMf7LBiAKzHdsML1mGoPPn-ykKd8Zu72m2JrwMfka_Aedx-BDA@mail.gmail.com>
	<CAHaUJVbNaOzCUiwYN-ka5OCY=OsY9M2Ri_zKwyoRKrGuoGVwgw@mail.gmail.com>
	<CA+fkitLK8w7h1a+kiQr=vQ-wt0oSGKaQ7q4HWhbep52FS65kMA@mail.gmail.com>
Message-ID: <CAJR7f2FgvT4qHGU5RBJs5Ti78Hgtf2hfChRy_hQb0ZpZFUF+7Q@mail.gmail.com>

Confirmed so far:

- Italo Rossi (+4)
- Areski Belaid
- Iwada Eja
- Adolphe Cher-Aime
- Bote

Who else?

On Fri, Dec 5, 2014 at 12:14 PM, Adolphe Cher-Aime <acheraime at gmail.com>
wrote:

> I'm in.
>
>
> --
> Adolphe
>
> On Thu, Dec 4, 2014 at 3:00 PM, Iwada Eja <iwada.bassey at gmail.com> wrote:
>
>> I'm In
>>
>> On Thu, Dec 4, 2014 at 1:20 PM, Areski <areski at gmail.com> wrote:
>>
>>> Hi everyone,
>>>
>>> We are planning to organize an other doc sprint on *Friday 12 December
>>> at 10am CT*.
>>> It will be 4 hours long but you can join for less time.
>>>
>>> The Doc-sprint will focus on migrating the remaining pages from
>>> MediaWiki (https://wiki.freeswitch.org) to Confluence Wiki (
>>> https://freeswitch.org/confluence).
>>>
>>> We will use an FS IRC channel during the sprint: *#freeswitch-docs*
>>> and will track our work on the spreadsheet:
>>> https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333
>>>
>>> During the sprint, please change the URL's "Status" you are working on
>>> to "Editing" with your name next to it so we don't duplicate work.
>>>
>>> Some extra information:
>>> - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration
>>> -
>>> https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation
>>>
>>> We hope to get a maximum number of people signed up!
>>>
>>> Peoples confirmed so far:
>>> - Italo Rossi (+4)
>>> - Areski Belaid
>>>
>>>
>>> So, who is in?
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Kind Regards
>> Iwada
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
?talo Rossi
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From liam at maxumdata.com  Wed Dec 10 12:20:29 2014
From: liam at maxumdata.com (Liam Farr)
Date: Wed, 10 Dec 2014 22:20:29 +1300
Subject: [Freeswitch-users] Mod perl using $session->setVariable inline
Message-ID: <CAJR0VfFO0P_zqWAWiKY1h1dubzivG+a8pX54KguWcsTjjgzB3g@mail.gmail.com>

Hi,

I need to set a variable using a perl script inline.

Normally I would use $session->setVariable('some_var',$my_value);

However I cant find any options to execute this inline / as an inline
action. (I need to evaluate the variable later in the hunting phase of the
dialplan hence need it to be an inline action).

Is there a way to do this?

Or is there another way to achieve the same result, e.g. hacking
$session->execute("set","some_var=$my_value") to run as an inline action?

Any help would be much appreciated, I've scoured the wiki, confluence and
git repo and come up empty.

-- 
Kind Regards


Liam Farr
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From notify.sina at gmail.com  Wed Dec 10 13:05:20 2014
From: notify.sina at gmail.com (Sina Owolabi)
Date: Wed, 10 Dec 2014 11:05:20 +0100
Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound
 Caller ID to external SIP Gateway URI
In-Reply-To: <0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com>
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com>
	<0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com>
	<CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com>
	<0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com>
	<CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com>
	<0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com>
	<CACK8u8J1b1UcEQ1R+M0puRT9vACtYGeuKBhHs_2ug9Q_qgJ6Wg@mail.gmail.com>
	<0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com>
Message-ID: <CACK8u8K+DA7qGEHoYnpHms_tetv-wQaQpafEv+e5MPZcmvizOg@mail.gmail.com>

It doesn't seem to, though I do hear some kind of a tiny pause between
when the othersipgw is dialled. I hope being NATed is not a problem!
They are now sending a "180 Ringing" before they send a "480
unavailable" in their reply and a single ring:
Here's a tcp dump of a call:

11:00:33.241307 IP (tos 0x0, ttl 64, id 34094, offset 0, flags [none],
proto UDP (17), length 1146)
    10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 1118
        INVITE sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0
        Via: SIP/2.0/UDP
my.public.addr.ess:5080;rport;branch=z9hG4bKZBN9c87mpSHDa
        Max-Forwards: 5
        From: "0diallednumber"
<sip:+234diallednumber at my.public.addr.ess>;tag=eF54Q5UjvXQZB
        To: <sip:user.name at sip.othersipgw.com;+234diallednumber>
        Call-ID: 38c0157b-faf6-1232-f39d-525400ecad09
        CSeq: 68766752 INVITE
        Contact: <sip:mod_sofia at my.public.addr.ess:5080>
        User-Agent:
FreeSWITCH-mod_sofia/1.5.15b+git~20141120T035109Z~79de78a0fb~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: 247
        X-FS-Support: update_display,send_info
        Remote-Party-ID: "0diallednumber"
<sip:+234diallednumber at my.public.addr.ess>;party=calling;screen=yes;privacy=off

        v=0
        o=FreeSWITCH 1418182171 1418182172 IN IP4 my.public.addr.ess
        s=FreeSWITCH
        c=IN IP4 my.public.addr.ess
        t=0 0
        m=audio 23462 RTP/AVP 8 0 101 13
        a=rtpmap:8 PCMA/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-16
        a=ptime:20

11:00:33.497529 IP (tos 0x0, ttl 53, id 5741, offset 0, flags [none],
proto UDP (17), length 1038)
    othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 1010
        SIP/2.0 180 Ringing
        Via: SIP/2.0/UDP
my.public.addr.ess:5080;rport=5080;branch=z9hG4bKZBN9c87mpSHDa
        From: "0diallednumber"
<sip:+234diallednumber at my.public.addr.ess>;tag=eF54Q5UjvXQZB
        To: <sip:user.name at sip.othersipgw.com;+234diallednumber>;tag=tt76SS77SZvjr
        Call-ID: 38c0157b-faf6-1232-f39d-525400ecad09
        CSeq: 68766752 INVITE
        Contact: <sip:user.name at othersipgw.ip.address:5060;transport=udp>
        User-Agent: FreeSWITCH-mod_sofia/1.2.14
        Accept: application/sdp
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
        Supported: timer, precondition, path, replaces
        Allow-Events: talk, hold, conference, presence, dialog,
line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer
        Content-Length: 0
        X-FS-Display-Name: user.name
        X-FS-Display-Number: sip:user.name at sip.othersipgw.com
        X-FS-Support: update_display,send_info
        Remote-Party-ID: "user.name"
<sip:user.name at sip.othersipgw.com>;party=calling;privacy=off;screen=no


11:00:36.552573 IP (tos 0x0, ttl 53, id 5742, offset 0, flags [none],
proto UDP (17), length 990)
    othersipgw.ip.address.5060 > 10.22.0.252.5080: SIP, length: 962
        SIP/2.0 480 Temporarily Unavailable
        Via: SIP/2.0/UDP
my.public.addr.ess:5080;rport=5080;branch=z9hG4bKZBN9c87mpSHDa
        Max-Forwards: 4
        From: "0diallednumber"
<sip:+234diallednumber at my.public.addr.ess>;tag=eF54Q5UjvXQZB
        To: <sip:user.name at sip.othersipgw.com;+234diallednumber>;tag=tt76SS77SZvjr
        Call-ID: 38c0157b-faf6-1232-f39d-525400ecad09
        CSeq: 68766752 INVITE
        User-Agent: FreeSWITCH-mod_sofia/1.2.14
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
        Supported: timer, precondition, path, replaces
        Allow-Events: talk, hold, conference, presence, dialog,
line-seize, call-info, sla, include-session-description,
presence.winfo, message-summary, refer
        Reason: Q.850;cause=16;text="NORMAL_CLEARING"
        Content-Length: 0
        X-FS-Display-Name: user.name
        X-FS-Display-Number: sip:user.name at sip.othersipgw.com
        Remote-Party-ID: "user.name"
<sip:user.name at sip.othersipgw.com>;party=calling;privacy=off;screen=no


11:00:36.552866 IP (tos 0x0, ttl 64, id 34095, offset 0, flags [none],
proto UDP (17), length 412)
    10.22.0.252.5080 > othersipgw.ip.address.5060: SIP, length: 384
        ACK sip:user.name at sip.othersipgw.com;+234diallednumber SIP/2.0
        Via: SIP/2.0/UDP
my.public.addr.ess:5080;rport;branch=z9hG4bKZBN9c87mpSHDa
        Max-Forwards: 5
        From: "0diallednumber"
<sip:+234diallednumber at my.public.addr.ess>;tag=eF54Q5UjvXQZB
        To: <sip:user.name at sip.othersipgw.com;+234diallednumber>;tag=tt76SS77SZvjr
        Call-ID: 38c0157b-faf6-1232-f39d-525400ecad09
        CSeq: 68766752 ACK
        Content-Length: 0


 Here's a log of a call after deleting and using your suggestions:

1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[NOTICE] switch_channel.c:1055 New Channel
sofia/external/0diallednumber at first.sipgw.net
[1517b746-8051-11e4-909f-2d01d6dac59f]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1053 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1053 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_NEW
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] sofia.c:8834 sofia/external/0diallednumber at first.sipgw.net
receiving invite from 62.173.32.89:5060 version: 1.5.15b git 79de78a
2014-11-20 03:51:09Z 64bit
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] sofia.c:6614 Channel
sofia/external/0diallednumber at first.sipgw.net entering state
[received][100]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] sofia.c:6624 Remote SDP:
1517b746-8051-11e4-909f-2d01d6dac59f v=0
1517b746-8051-11e4-909f-2d01d6dac59f o=CGPLeg006300 1254596129
627298065 IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f s=Cisco SDP 0
1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f t=0 0
1517b746-8051-11e4-909f-2d01d6dac59f o=CGPLeg006300 1254596129
627298065 IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f s=Cisco SDP 0
1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f t=0 0
1517b746-8051-11e4-909f-2d01d6dac59f
a=mediagateway:first.sipgw.net:2852669:10.51.2.3
1517b746-8051-11e4-909f-2d01d6dac59f m=audio 60008 RTP/AVP 18 8 0 101 100
1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:18 G729/8000
1517b746-8051-11e4-909f-2d01d6dac59f m=audio 60008 RTP/AVP 18 8 0 101 100
1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:18 G729/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:8 PCMA/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:0 PCMU/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:101 telephone-event/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:101 0-15
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:0 PCMU/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:101 telephone-event/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:101 0-15
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:100 X-NSE/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:100 200-202
1517b746-8051-11e4-909f-2d01d6dac59f a=rtcpping:T:181652:18165278
1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:100 200-202
1517b746-8051-11e4-909f-2d01d6dac59f a=rtcpping:T:181652:18165278
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 1 audio RTP/AVP 100
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 2 image udptl t38
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=rtpmap:100 X-NSE/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 2 image udptl t38
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=rtpmap:100 X-NSE/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=fmtp:100 200-202
1517b746-8051-11e4-909f-2d01d6dac59f a=X-sqn:0
1517b746-8051-11e4-909f-2d01d6dac59f
1517b746-8051-11e4-909f-2d01d6dac59f a=X-sqn:0
1517b746-8051-11e4-909f-2d01d6dac59f
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] sofia.c:6890 (sofia/external/0diallednumber at first.sipgw.net)
State Change CS_NEW -> CS_INIT
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] sofia.c:6890 (sofia/external/0diallednumber at first.sipgw.net)
State Change CS_NEW -> CS_INIT
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:491
(sofia/external/0diallednumber at first.sipgw.net) State NEW
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_INIT
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:512
(sofia/external/0diallednumber at first.sipgw.net) State INIT
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:87 sofia/external/0diallednumber at first.sipgw.net
SOFIA INIT
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:40
sofia/external/0diallednumber at first.sipgw.net Standard INIT
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:40
sofia/external/0diallednumber at first.sipgw.net Standard INIT
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:48
(sofia/external/0diallednumber at first.sipgw.net) State Change CS_INIT
-> CS_ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_sessio1517b746-8051-11e4-909f-2d01d6dac59f
2014-12-10 10:44:06.245285 [DEBUG] switch_core_session.c:1388 Send
signal sofia/external/0diallednumber at first.sipgw.net [BREAK]
n.c:1388 Send signal sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:512
(sofia/external/0diallednumber at first.sipgw.net) State INIT going to
sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:512
(sofia/external/0diallednumber at first.sipgw.net) State INIT going to
sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_channel.c:2184
(sofia/external/0diallednumber at first.sipgw.net) Callstate Change DOWN
-> RINGING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:528
(sofia/external/0diallednumber at first.sipgw.net) State ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:123 sofia/external/0diallednumber at first.sipgw.net
SOFIA ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:166
sofia/external/0diallednumber at first.sipgw.net Standard ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [INFO]
mod_dialplan_xml.c:635 Processing 0diallednumber
<0diallednumber>->016311084 in context public
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [public->unloop]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [public->unloop]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[public->outside_call] continue=true
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[public->outside_call] continue=true
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Absolute Condition
[outside_call]
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
set(outside_call=true)
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[public->call_debug] continue=true
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[call_debug] ${call_debug}(false) =~ /^true$/ break=never
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[public->public_extensions] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[public->public_extensions] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[public_extensions] destination_number(016311084) =~ /^(10[01][0-9])$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[public_extensions] destination_number(016311084) =~ /^(10[01][0-9])$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[public->public_did] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[public_did] destination_number(016311084) =~ /^(5551212)$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[public->ipnx-inbound] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS)
[ipnx-inbound] destination_number(016311084) =~ /^0163110(\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS)
[ipnx-inbound] caller_id_number(0diallednumber) =~ /^0(\d{10})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
set(effective_caller_id_number=+234diallednumber)
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action transfer(1212 XML
default)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:216
(sofia/external/0diallednumber at first.sipgw.net) State Change
CS_ROUTING -> CS_EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:528
(sofia/external/0diallednumber at first.sipgw.net) State ROUTING going to
sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:535
(sofia/external/0diallednumber at first.sipgw.net) State EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:178 sofia/external/0diallednumber at first.sipgw.net
SOFIA EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:258
sofia/external/0diallednumber at first.sipgw.net Standard EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net set(outside_call=true)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_dptools.c:1435
sofia/external/0diallednumber at first.sipgw.net SET
[outside_call]=[true]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net export(RFC2822_DATE=Wed,
10 Dec 2014 10:44:06 +0100)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_channel.c:1247 EXPORT (export_vars)
[RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
set(effective_caller_id_number=+234diallednumber)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_dptools.c:1435
sofia/external/0diallednumber at first.sipgw.net SET
[effective_caller_id_number]=[+234diallednumber]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net transfer(1212 XML
default)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_ivr.c:1847
(sofia/external/0diallednumber at first.sipgw.net) State Change
CS_EXECUTE -> CS_ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:908 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[NOTICE] switch_ivr.c:1854 Transfer
sofia/external/0diallednumber at first.sipgw.net to XML[1212 at default]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:535
(sofia/external/0diallednumber at first.sipgw.net) State EXECUTE going to
sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:528
(sofia/external/0diallednumber at first.sipgw.net) State ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:123 sofia/external/0diallednumber at first.sipgw.net
SOFIA ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:166
sofia/external/0diallednumber at first.sipgw.net Standard ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [INFO]
mod_dialplan_xml.c:635 Processing 0diallednumber
<0diallednumber>->1212 in context default
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->unloop] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->tod_example] continue=true
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Date/Time Match (PASS)
[tod_example] break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action set(open=true)
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->holiday_example] continue=true
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Date/TimeMatch (FAIL)
[holiday_example] break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->global-intercept] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[global-intercept] destination_number(1212) =~ /^886$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->group-intercept] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[group-intercept] destination_number(1212) =~ /^\*8$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->intercept-ext] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[intercept-ext] destination_number(1212) =~ /^\*\*(\d+)$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->redial] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [redial]
destination_number(1212) =~ /^(redial|870)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->global] continue=true
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global]
${call_debug}(false) =~ /^true$/ break=never
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global]
${default_password}(N0w4h4l4) =~ /^1234$/ break=never
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global]
${rtp_has_crypto}() =~
/^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/
break=never
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [global]
${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED
NEGOTIATION)/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global]
${switch_r_sdp}(v=0
1517b746-8051-11e4-909f-2d01d6dac59f o=CGPLeg006300 1254596129
627298065 IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f s=Cisco SDP 0
1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f t=0 0
1517b746-8051-11e4-909f-2d01d6dac59f
a=mediagateway:first.sipgw.net:2852669:10.51.2.3
1517b746-8051-11e4-909f-2d01d6dac59f m=audio 60008 RTP/AVP 18 8 0 101 100
1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:18 G729/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:8 PCMA/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:0 PCMU/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:101 telephone-event/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:101 0-15
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:100 X-NSE/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:100 200-202
1517b746-8051-11e4-909f-2d01d6dac59f a=rtcpping:T:181652:18165278
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 1 audio RTP/AVP 100
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 2 image udptl t38
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=rtpmap:100 X-NSE/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=fmtp:100 200-202
1517b746-8051-11e4-909f-2d01d6dac59f a=X-sqn:0
1517b746-8051-11e4-909f-2d01d6dac59f ) =~
/(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Absolute Condition
[global]
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
hash(insert/${domain_name}-last_dial/global/${uuid})
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->snom-demo-2] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[snom-demo-2] destination_number(1212) =~ /^9001$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->snom-demo-1] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[snom-demo-1] destination_number(1212) =~ /^9000$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->eavesdrop] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [eavesdrop]
destination_number(1212) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->eavesdrop] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [eavesdrop]
destination_number(1212) =~ /^779$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->call_return] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[call_return] destination_number(1212) =~ /^\*69$|^869$|^lcr$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->del-group] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [del-group]
destination_number(1212) =~ /^80(\d{2})$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->add-group] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [add-group]
destination_number(1212) =~ /^81(\d{2})$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->call-group-simo] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[call-group-simo] destination_number(1212) =~ /^82(\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->call-group-order] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[call-group-order] destination_number(1212) =~ /^83(\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->extension-intercom] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[extension-intercom] destination_number(1212) =~ /^8(10[01][0-9])$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->Local_Extension] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[Local_Extension] destination_number(1212) =~ /^(10[01][0-9])$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->Local_Extension_Skinny] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[Local_Extension_Skinny] destination_number(1212) =~ /^(11[01][0-9])$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->group_dial_sales] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[group_dial_sales] destination_number(1212) =~ /^2000$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->group_dial_support] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[group_dial_support] destination_number(1212) =~ /^2001$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->group_dial_billing] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[group_dial_billing] destination_number(1212) =~ /^2002$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->operator] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [operator]
destination_number(1212) =~ /^(operator|0)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->vmain]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [vmain]
destination_number(1212) =~ /^vmain$|^4000$|^\*98$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->sip_uri] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [sip_uri]
destination_number(1212) =~ /^sip:(.*)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->nb_conferences] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[nb_conferences] destination_number(1212) =~ /^(30\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->wb_conferences] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[wb_conferences] destination_number(1212) =~ /^(31\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->uwb_conferences] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[uwb_conferences] destination_number(1212) =~ /^(32\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->cdquality_conferences] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[cdquality_conferences] destination_number(1212) =~ /^(33\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->freeswitch_public_conf_via_sip] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[freeswitch_public_conf_via_sip] destination_number(1212) =~
/^9(888|8888|1616|3232)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->mad_boss_intercom] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[mad_boss_intercom] destination_number(1212) =~ /^0911$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->mad_boss_intercom] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[mad_boss_intercom] destination_number(1212) =~ /^0912$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->mad_boss] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [mad_boss]
destination_number(1212) =~ /^0913$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ivr_demo] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ivr_demo]
destination_number(1212) =~ /^5000$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->dynamic_conference] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[dynamic_conference] destination_number(1212) =~ /^5001$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->rtp_multicast_page] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[rtp_multicast_page] destination_number(1212) =~ /^pagegroup$|^7243$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->park]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park]
destination_number(1212) =~ /^5900$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->unpark] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark]
destination_number(1212) =~ /^5901$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->valet_park] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[valet_park] destination_number(1212) =~ /^(6000)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->valet_park] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[valet_park] destination_number(1212) =~ /^((?!6000)60\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->park]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park]
destination_number(1212) =~ /park\+(\d+)/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->unpark] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark]
destination_number(1212) =~ /^parking$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->park]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park]
destination_number(1212) =~ /callpark/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->unpark] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark]
destination_number(1212) =~ /pickup/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->wait]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [wait]
destination_number(1212) =~ /^wait$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->fax_receive] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[fax_receive] destination_number(1212) =~ /^9178$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->fax_transmit] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[fax_transmit] destination_number(1212) =~ /^9179$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ringback_180] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[ringback_180] destination_number(1212) =~ /^9180$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ringback_183_uk_ring] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[ringback_183_uk_ring] destination_number(1212) =~ /^9181$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ringback_183_music_ring] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[ringback_183_music_ring] destination_number(1212) =~ /^9182$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ringback_post_answer_uk_ring] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[ringback_post_answer_uk_ring] destination_number(1212) =~ /^9183$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ringback_post_answer_music] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[ringback_post_answer_music] destination_number(1212) =~ /^9184$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ClueCon] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ClueCon]
destination_number(1212) =~ /^9191$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->show_info] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [show_info]
destination_number(1212) =~ /^9192$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[public_did] destination_number(016311084) =~ /^(5551212)$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[public->ipnx-inbound] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS)
[ipnx-inbound] destination_number(016311084) =~ /^0163110(\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS)
[ipnx-inbound] caller_id_number(0diallednumber) =~ /^0(\d{10})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
set(effective_caller_id_number=+234diallednumber)
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action transfer(1212 XML
default)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:216
(sofia/external/0diallednumber at first.sipgw.net) State Change
CS_ROUTING -> CS_EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:528
(sofia/external/0diallednumber at first.sipgw.net) State ROUTING going to
sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:535
(sofia/external/0diallednumber at first.sipgw.net) State EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:178 sofia/external/0diallednumber at first.sipgw.net
SOFIA EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:258
sofia/external/0diallednumber at first.sipgw.net Standard EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net set(outside_call=true)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_dptools.c:1435
sofia/external/0diallednumber at first.sipgw.net SET
[outside_call]=[true]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net export(RFC2822_DATE=Wed,
10 Dec 2014 10:44:06 +0100)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_channel.c:1247 EXPORT (export_vars)
[RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
set(effective_caller_id_number=+234diallednumber)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_dptools.c:1435
sofia/external/0diallednumber at first.sipgw.net SET
[effective_caller_id_number]=[+234diallednumber]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net transfer(1212 XML
default)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_ivr.c:1847
(sofia/external/0diallednumber at first.sipgw.net) State Change
CS_EXECUTE -> CS_ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:908 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[NOTICE] switch_ivr.c:1854 Transfer
sofia/external/0diallednumber at first.sipgw.net to XML[1212 at default]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:535
(sofia/external/0diallednumber at first.sipgw.net) State EXECUTE going to
sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:528
(sofia/external/0diallednumber at first.sipgw.net) State ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:123 sofia/external/0diallednumber at first.sipgw.net
SOFIA ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:166
sofia/external/0diallednumber at first.sipgw.net Standard ROUTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285 [INFO]
mod_dialplan_xml.c:635 Processing 0diallednumber
<0diallednumber>->1212 in context default
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->unloop] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->tod_example] continue=true
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Date/Time Match (PASS)
[tod_example] break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action set(open=true)
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->holiday_example] continue=true
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Date/TimeMatch (FAIL)
[holiday_example] break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->global-intercept] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[global-intercept] destination_number(1212) =~ /^886$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->group-intercept] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[group-intercept] destination_number(1212) =~ /^\*8$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->intercept-ext] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[intercept-ext] destination_number(1212) =~ /^\*\*(\d+)$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->redial] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [redial]
destination_number(1212) =~ /^(redial|870)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->global] continue=true
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global]
${call_debug}(false) =~ /^true$/ break=never
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global]
${default_password}(N0w4h4l4) =~ /^1234$/ break=never
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global]
${rtp_has_crypto}() =~
/^(AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8|AES_CM_256_HMAC_SHA1_80|AES_CM_192_HMAC_SHA1_80|AES_CM_128_HMAC_SHA1_80|AES_CM_256_HMAC_SHA1_32|AES_CM_192_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_32|AES_CM_128_NULL_AUTH)$/
break=never
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [global]
${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED
NEGOTIATION)/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [global]
${switch_r_sdp}(v=0
1517b746-8051-11e4-909f-2d01d6dac59f o=CGPLeg006300 1254596129
627298065 IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f s=Cisco SDP 0
1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f t=0 0
1517b746-8051-11e4-909f-2d01d6dac59f
a=mediagateway:first.sipgw.net:2852669:10.51.2.3
1517b746-8051-11e4-909f-2d01d6dac59f m=audio 60008 RTP/AVP 18 8 0 101 100
1517b746-8051-11e4-909f-2d01d6dac59f c=IN IP4 first.sipgw.public.addresss
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:18 G729/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:8 PCMA/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:0 PCMU/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:101 telephone-event/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:101 0-15
1517b746-8051-11e4-909f-2d01d6dac59f a=rtpmap:100 X-NSE/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=fmtp:100 200-202
1517b746-8051-11e4-909f-2d01d6dac59f a=rtcpping:T:181652:18165278
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 1 audio RTP/AVP 100
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cap: 2 image udptl t38
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=rtpmap:100 X-NSE/8000
1517b746-8051-11e4-909f-2d01d6dac59f a=X-cpar: a=fmtp:100 200-202
1517b746-8051-11e4-909f-2d01d6dac59f a=X-sqn:0
1517b746-8051-11e4-909f-2d01d6dac59f ) =~
/(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Absolute Condition
[global]
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number})
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
hash(insert/${domain_name}-last_dial/global/${uuid})
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->snom-demo-2] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[snom-demo-2] destination_number(1212) =~ /^9001$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->snom-demo-1] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[snom-demo-1] destination_number(1212) =~ /^9000$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->eavesdrop] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [eavesdrop]
destination_number(1212) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->eavesdrop] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [eavesdrop]
destination_number(1212) =~ /^779$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->call_return] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[call_return] destination_number(1212) =~ /^\*69$|^869$|^lcr$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->del-group] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [del-group]
destination_number(1212) =~ /^80(\d{2})$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->add-group] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [add-group]
destination_number(1212) =~ /^81(\d{2})$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->call-group-simo] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[call-group-simo] destination_number(1212) =~ /^82(\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->call-group-order] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[call-group-order] destination_number(1212) =~ /^83(\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->extension-intercom] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[extension-intercom] destination_number(1212) =~ /^8(10[01][0-9])$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->Local_Extension] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[Local_Extension] destination_number(1212) =~ /^(10[01][0-9])$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->Local_Extension_Skinny] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[Local_Extension_Skinny] destination_number(1212) =~ /^(11[01][0-9])$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->group_dial_sales] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[group_dial_sales] destination_number(1212) =~ /^2000$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->group_dial_support] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[group_dial_support] destination_number(1212) =~ /^2001$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->group_dial_billing] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[group_dial_billing] destination_number(1212) =~ /^2002$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->operator] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [operator]
destination_number(1212) =~ /^(operator|0)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->vmain]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [vmain]
destination_number(1212) =~ /^vmain$|^4000$|^\*98$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->sip_uri] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [sip_uri]
destination_number(1212) =~ /^sip:(.*)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->nb_conferences] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[nb_conferences] destination_number(1212) =~ /^(30\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->wb_conferences] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[wb_conferences] destination_number(1212) =~ /^(31\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->uwb_conferences] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[uwb_conferences] destination_number(1212) =~ /^(32\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->cdquality_conferences] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[cdquality_conferences] destination_number(1212) =~ /^(33\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->freeswitch_public_conf_via_sip] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[freeswitch_public_conf_via_sip] destination_number(1212) =~
/^9(888|8888|1616|3232)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->mad_boss_intercom] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[mad_boss_intercom] destination_number(1212) =~ /^0911$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->mad_boss_intercom] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[mad_boss_intercom] destination_number(1212) =~ /^0912$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->mad_boss] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [mad_boss]
destination_number(1212) =~ /^0913$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ivr_demo] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ivr_demo]
destination_number(1212) =~ /^5000$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->dynamic_conference] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[dynamic_conference] destination_number(1212) =~ /^5001$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->rtp_multicast_page] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[rtp_multicast_page] destination_number(1212) =~ /^pagegroup$|^7243$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->park]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park]
destination_number(1212) =~ /^5900$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->unpark] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark]
destination_number(1212) =~ /^5901$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->valet_park] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[valet_park] destination_number(1212) =~ /^(6000)$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->valet_park] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[valet_park] destination_number(1212) =~ /^((?!6000)60\d{2})$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->park]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park]
destination_number(1212) =~ /park\+(\d+)/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->unpark] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark]
destination_number(1212) =~ /^parking$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->park]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [park]
source(mod_sofia) =~ /mod_sofia/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [park]
destination_number(1212) =~ /callpark/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->unpark] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [unpark]
source(mod_sofia) =~ /mod_sofia/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [unpark]
destination_number(1212) =~ /pickup/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->wait]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [wait]
destination_number(1212) =~ /^wait$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->fax_receive] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[fax_receive] destination_number(1212) =~ /^9178$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->fax_transmit] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[fax_transmit] destination_number(1212) =~ /^9179$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ringback_180] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[ringback_180] destination_number(1212) =~ /^9180$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ringback_183_uk_ring] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[ringback_183_uk_ring] destination_number(1212) =~ /^9181$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ringback_183_music_ring] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[ringback_183_music_ring] destination_number(1212) =~ /^9182$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ringback_post_answer_uk_ring] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[ringback_post_answer_uk_ring] destination_number(1212) =~ /^9183$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ringback_post_answer_music] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[ringback_post_answer_music] destination_number(1212) =~ /^9184$/
break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->ClueCon] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [ClueCon]
destination_number(1212) =~ /^9191$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->show_info] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [show_info]
destination_number(1212) =~ /^9192$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->video_record] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[video_record] destination_number(1212) =~ /^9193$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->video_playback] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[video_playback] destination_number(1212) =~ /^9194$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->delay_echo] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[delay_echo] destination_number(1212) =~ /^9195$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->echo]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [echo]
destination_number(1212) =~ /^9196$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->milliwatt] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [milliwatt]
destination_number(1212) =~ /^9197$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->tone_stream] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[tone_stream] destination_number(1212) =~ /^9198$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->zrtp_enrollement] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[zrtp_enrollement] destination_number(1212) =~ /^9787$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->hold_music] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[hold_music] destination_number(1212) =~ /^9664$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->laugh
break] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [laugh
break] destination_number(1212) =~ /^9386$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->1212]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [1212]
destination_number(1212) =~ /1212/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
set(hangup_after_bridge=false)
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
bridge(sofia/external/sip:user.name at sip.othersipgw.com;${effective_caller_id_number})
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
set(exec_after_bridge_arg=1213)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:216
(sofia/external/0diallednumber at first.sipgw.net) State Change
CS_ROUTING -> CS_EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:528
(sofia/external/0diallednumber at first.sipgw.net) State ROUTING going to
sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:535
(sofia/external/0diallednumber at first.sipgw.net) State EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:178 sofia/external/0diallednumber at first.sipgw.net
SOFIA EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:258
sofia/external/0diallednumber at first.sipgw.net Standard EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net set(open=true)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_dptools.c:1435
sofia/external/0diallednumber at first.sipgw.net SET [open]=[true]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
hash(insert/10.22.0.252-spymap/0diallednumber/1517b746-8051-11e4-909f-2d01d6dac59f)
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
hash(insert/10.22.0.252-last_dial/0diallednumber/1212)
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
hash(insert/10.22.0.252-last_dial/global/1517b746-8051-11e4-909f-2d01d6dac59f)
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net export(RFC2822_DATE=Wed,
10 Dec 2014 10:44:06 +0100)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_channel.c:1247 EXPORT (export_vars)
[RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
set(hangup_after_bridge=false)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_dptools.c:1435
sofia/external/0diallednumber at first.sipgw.net SET
[hangup_after_bridge]=[false]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
bridge(sofia/external/sip:user.name at sip.othersipgw.com;+234diallednumber)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_channel.c:1201
sofia/external/0diallednumber at first.sipgw.net EXPORTING[export_vars]
[RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] to event
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_channel.c:1201
sofia/external/0diallednumber at first.sipgw.net EXPORTING[export_vars]
[RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] to event
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_ivr_originate.c:2079 Parsing global variables
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[NOTICE] switch_channel.c:1055 New Channel
sofia/external/sip:user.name at sip.othersipgw.com
[15187302-8051-11e4-90b0-2d01d6dac59f]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:4615
(sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_NEW
-> CS_INIT
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:4685 [zrtp_passthru] Setting a-leg
inherit_codec=true
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:4688 [zrtp_passthru] Setting b-leg
absolute_codec_string=PCMA at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/sip:user.name at sip.othersipgw.com) Running State Change
CS_INIT
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:512
(sofia/external/sip:user.name at sip.othersipgw.com) State INIT
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:87 sofia/external/sip:user.name at sip.othersipgw.com
SOFIA INIT
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] sofia_glue.c:1232
sofia/external/sip:user.name at sip.othersipgw.com sending invite
version: 1.5.15b git 79de78a 2014-11-20 03:51:09Z 64bit
15187302-8051-11e4-90b0-2d01d6dac59f Local SDP:
15187302-8051-11e4-90b0-2d01d6dac59f v=0
15187302-8051-11e4-90b0-2d01d6dac59f o=FreeSWITCH 1418187966
1418187967 IN IP4 my.public.addr.ess
15187302-8051-11e4-90b0-2d01d6dac59f s=FreeSWITCH
15187302-8051-11e4-90b0-2d01d6dac59f c=IN IP4 my.public.addr.ess
15187302-8051-11e4-90b0-2d01d6dac59f t=0 0
15187302-8051-11e4-90b0-2d01d6dac59f m=audio 16680 RTP/AVP 8 0 101 13
15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:8 PCMA/8000
15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:0 PCMU/8000
15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:101 telephone-event/8000
15187302-8051-11e4-90b0-2d01d6dac59f a=fmtp:101 0-16
15187302-8051-11e4-90b0-2d01d6dac59f a=ptime:20
15187302-8051-11e4-90b0-2d01d6dac59f a=sendrecv
15187302-8051-11e4-90b0-2d01d6dac59f
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:40
sofia/external/sip:user.name at sip.othersipgw.com Standard INIT
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:48
(sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_INIT
-> CS_ROUTING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:512
(sofia/external/sip:user.name at sip.othersipgw.com) State INIT going to
sleep
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/sip:user.name at sip.othersipgw.com) Running State Change
CS_ROUTING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:528
(sofia/external/sip:user.name at sip.othersipgw.com) State ROUTING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:123
sofia/external/sip:user.name at sip.othersipgw.com SOFIA ROUTING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_ivr_originate.c:67
(sofia/external/sip:user.name at sip.othersipgw.com) State Change
CS_ROUTING -> CS_CONSUME_MEDIA
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:528
(sofia/external/sip:user.name at sip.othersipgw.com) State ROUTING going
to sleep
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/sip:user.name at sip.othersipgw.com) Running State Change
CS_CONSUME_MEDIA
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:547
(sofia/external/sip:user.name at sip.othersipgw.com) State CONSUME_MEDIA
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:547
(sofia/external/sip:user.name at sip.othersipgw.com) State CONSUME_MEDIA
going to sleep
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1053 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] sofia.c:6614 Channel
sofia/external/sip:user.name at sip.othersipgw.com entering state
[calling][0]
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->video_record] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[video_record] destination_number(1212) =~ /^9193$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->video_playback] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[video_playback] destination_number(1212) =~ /^9194$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->delay_echo] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[delay_echo] destination_number(1212) =~ /^9195$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->echo]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [echo]
destination_number(1212) =~ /^9196$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->milliwatt] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [milliwatt]
destination_number(1212) =~ /^9197$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->tone_stream] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[tone_stream] destination_number(1212) =~ /^9198$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->zrtp_enrollement] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[zrtp_enrollement] destination_number(1212) =~ /^9787$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing
[default->hold_music] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL)
[hold_music] destination_number(1212) =~ /^9664$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->laugh
break] continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (FAIL) [laugh
break] destination_number(1212) =~ /^9386$/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net parsing [default->1212]
continue=false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Regex (PASS) [1212]
destination_number(1212) =~ /1212/ break=on-false
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
set(hangup_after_bridge=false)
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
bridge(sofia/external/sip:user.name at sip.othersipgw.com;${effective_caller_id_number})
1517b746-8051-11e4-909f-2d01d6dac59f Dialplan:
sofia/external/0diallednumber at first.sipgw.net Action
set(exec_after_bridge_arg=1213)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:216
(sofia/external/0diallednumber at first.sipgw.net) State Change
CS_ROUTING -> CS_EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:528
(sofia/external/0diallednumber at first.sipgw.net) State ROUTING going to
sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:535
(sofia/external/0diallednumber at first.sipgw.net) State EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:178 sofia/external/0diallednumber at first.sipgw.net
SOFIA EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:258
sofia/external/0diallednumber at first.sipgw.net Standard EXECUTE
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net set(open=true)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_dptools.c:1435
sofia/external/0diallednumber at first.sipgw.net SET [open]=[true]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
hash(insert/10.22.0.252-spymap/0diallednumber/1517b746-8051-11e4-909f-2d01d6dac59f)
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
hash(insert/10.22.0.252-last_dial/0diallednumber/1212)
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
hash(insert/10.22.0.252-last_dial/global/1517b746-8051-11e4-909f-2d01d6dac59f)
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net export(RFC2822_DATE=Wed,
10 Dec 2014 10:44:06 +0100)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_channel.c:1247 EXPORT (export_vars)
[RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
set(hangup_after_bridge=false)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_dptools.c:1435
sofia/external/0diallednumber at first.sipgw.net SET
[hangup_after_bridge]=[false]
1517b746-8051-11e4-909f-2d01d6dac59f EXECUTE
sofia/external/0diallednumber at first.sipgw.net
bridge(sofia/external/sip:user.name at sip.othersipgw.com;+234diallednumber)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_channel.c:1201
sofia/external/0diallednumber at first.sipgw.net EXPORTING[export_vars]
[RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] to event
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_channel.c:1201
sofia/external/0diallednumber at first.sipgw.net EXPORTING[export_vars]
[RFC2822_DATE]=[Wed, 10 Dec 2014 10:44:06 +0100] to event
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_ivr_originate.c:2079 Parsing global variables
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[NOTICE] switch_channel.c:1055 New Channel
sofia/external/sip:user.name at sip.othersipgw.com
[15187302-8051-11e4-90b0-2d01d6dac59f]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:4615
(sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_NEW
-> CS_INIT
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:4685 [zrtp_passthru] Setting a-leg
inherit_codec=true
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:4688 [zrtp_passthru] Setting b-leg
absolute_codec_string=PCMA at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/sip:user.name at sip.othersipgw.com) Running State Change
CS_INIT
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:512
(sofia/external/sip:user.name at sip.othersipgw.com) State INIT
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:87 sofia/external/sip:user.name at sip.othersipgw.com
SOFIA INIT
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] sofia_glue.c:1232
sofia/external/sip:user.name at sip.othersipgw.com sending invite
version: 1.5.15b git 79de78a 2014-11-20 03:51:09Z 64bit
15187302-8051-11e4-90b0-2d01d6dac59f Local SDP:
15187302-8051-11e4-90b0-2d01d6dac59f v=0
15187302-8051-11e4-90b0-2d01d6dac59f o=FreeSWITCH 1418187966
1418187967 IN IP4 my.public.addr.ess
15187302-8051-11e4-90b0-2d01d6dac59f s=FreeSWITCH
15187302-8051-11e4-90b0-2d01d6dac59f c=IN IP4 my.public.addr.ess
15187302-8051-11e4-90b0-2d01d6dac59f t=0 0
15187302-8051-11e4-90b0-2d01d6dac59f m=audio 16680 RTP/AVP 8 0 101 13
15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:8 PCMA/8000
15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:0 PCMU/8000
15187302-8051-11e4-90b0-2d01d6dac59f a=rtpmap:101 telephone-event/8000
15187302-8051-11e4-90b0-2d01d6dac59f a=fmtp:101 0-16
15187302-8051-11e4-90b0-2d01d6dac59f a=ptime:20
15187302-8051-11e4-90b0-2d01d6dac59f a=sendrecv
15187302-8051-11e4-90b0-2d01d6dac59f
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:40
sofia/external/sip:user.name at sip.othersipgw.com Standard INIT
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:48
(sofia/external/sip:user.name at sip.othersipgw.com) State Change CS_INIT
-> CS_ROUTING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:512
(sofia/external/sip:user.name at sip.othersipgw.com) State INIT going to
sleep
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/sip:user.name at sip.othersipgw.com) Running State Change
CS_ROUTING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:528
(sofia/external/sip:user.name at sip.othersipgw.com) State ROUTING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] mod_sofia.c:123
sofia/external/sip:user.name at sip.othersipgw.com SOFIA ROUTING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_ivr_originate.c:67
(sofia/external/sip:user.name at sip.othersipgw.com) State Change
CS_ROUTING -> CS_CONSUME_MEDIA
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:528
(sofia/external/sip:user.name at sip.othersipgw.com) State ROUTING going
to sleep
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:472
(sofia/external/sip:user.name at sip.othersipgw.com) Running State Change
CS_CONSUME_MEDIA
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:547
(sofia/external/sip:user.name at sip.othersipgw.com) State CONSUME_MEDIA
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_state_machine.c:547
(sofia/external/sip:user.name at sip.othersipgw.com) State CONSUME_MEDIA
going to sleep
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] switch_core_session.c:1053 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.245285
[DEBUG] sofia.c:6614 Channel
sofia/external/sip:user.name at sip.othersipgw.com entering state
[calling][0]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.965294
[DEBUG] switch_core_session.c:1053 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.965294
[DEBUG] switch_core_session.c:1053 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
2014-12-10 10:44:06.965294 [INFO] sofia.c:1203
sofia/external/sip:user.name at sip.othersipgw.com Update Callee ID to
"user.name" <sip:user.name at sip.othersipgw.com>
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.965294
[DEBUG] sofia.c:6614 Channel
sofia/external/sip:user.name at sip.othersipgw.com entering state
[proceeding][180]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.965294
[NOTICE] sofia.c:6716 Ring-Ready
sofia/external/sip:user.name at sip.othersipgw.com!
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:06.965294
[DEBUG] switch_channel.c:3277
(sofia/external/sip:user.name at sip.othersipgw.com) Callstate Change
DOWN -> RINGING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.965294
[NOTICE] mod_sofia.c:2086 Ring-Ready
sofia/external/0diallednumber at first.sipgw.net!
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.965294
[DEBUG] switch_core_session.c:1053 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.965294
[DEBUG] sofia.c:6614 Channel
sofia/external/0diallednumber at first.sipgw.net entering state
[early][180]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.965294
[DEBUG] switch_core_session.c:908 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:06.965294
[NOTICE] switch_ivr_originate.c:527 Ring Ready
sofia/external/0diallednumber at first.sipgw.net!
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_session.c:1053 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_session.c:1053 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_session.c:1053 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] sofia.c:5840 Remote Reason: 16
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] sofia.c:6614 Channel
sofia/external/sip:user.name at sip.othersipgw.com entering state
[terminated][480]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[NOTICE] sofia.c:7530 Hangup
sofia/external/sip:user.name at sip.othersipgw.com [CS_CONSUME_MEDIA]
[NORMAL_CLEARING]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_channel.c:3222 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [KILL]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:472
(sofia/external/sip:user.name at sip.othersipgw.com) Running State Change
CS_HANGUP
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:735
(sofia/external/sip:user.name at sip.othersipgw.com) Callstate Change
RINGING -> HANGUP
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:737
(sofia/external/sip:user.name at sip.othersipgw.com) State HANGUP
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] mod_sofia.c:413 Channel
sofia/external/sip:user.name at sip.othersipgw.com hanging up, cause:
NORMAL_CLEARING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:60
sofia/external/sip:user.name at sip.othersipgw.com Standard HANGUP,
cause: NORMAL_CLEARING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:737
(sofia/external/sip:user.name at sip.othersipgw.com) State HANGUP going
to sleep
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:504
(sofia/external/sip:user.name at sip.othersipgw.com) State Change
CS_HANGUP -> CS_REPORTING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:472
(sofia/external/sip:user.name at sip.othersipgw.com) Running State Change
CS_REPORTING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:823
(sofia/external/sip:user.name at sip.othersipgw.com) State REPORTING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:104
sofia/external/sip:user.name at sip.othersipgw.com Standard REPORTING,
cause: NORMAL_CLEARING
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:823
(sofia/external/sip:user.name at sip.othersipgw.com) State REPORTING
going to sleep
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:498
(sofia/external/sip:user.name at sip.othersipgw.com) State Change
CS_REPORTING -> CS_DESTROY
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/sip:user.name at sip.othersipgw.com [BREAK]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_session.c:1615 Session 232
(sofia/external/sip:user.name at sip.othersipgw.com) Locked, Waiting on
external entities
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_ivr_originate.c:3698 Originate Resulted in Error Cause:
16 [NORMAL_CLEARING]
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[NOTICE] switch_core_session.c:1633 Session 232
(sofia/external/sip:user.name at sip.othersipgw.com) Ended
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[NOTICE] switch_core_session.c:1637 Close Channel
sofia/external/sip:user.name at sip.othersipgw.com [CS_DESTROY]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334 [INFO]
mod_dptools.c:3234 Originate Failed.  Cause: NORMAL_CLEARING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[NOTICE] switch_channel.c:4724 Hangup
sofia/external/0diallednumber at first.sipgw.net [CS_EXECUTE]
[NORMAL_CLEARING]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_channel.c:3222 Send signal
sofia/external/0diallednumber at first.sipgw.net [KILL]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_session.c:2893
sofia/external/0diallednumber at first.sipgw.net skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:535
(sofia/external/0diallednumber at first.sipgw.net) State EXECUTE going to
sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_HANGUP
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_HANGUP
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:626
(sofia/external/sip:user.name at sip.othersipgw.com) Running State Change
CS_DESTROY
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:636
(sofia/external/sip:user.name at sip.othersipgw.com) State DESTROY
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] mod_sofia.c:323
sofia/external/sip:user.name at sip.othersipgw.com SOFIA DESTROY
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:111
sofia/external/sip:user.name at sip.othersipgw.com Standard DESTROY
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:636
(sofia/external/sip:user.name at sip.othersipgw.com) State DESTROY going
to sleep
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:111
sofia/external/sip:user.name at sip.othersipgw.com Standard DESTROY
15187302-8051-11e4-90b0-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:636
(sofia/external/sip:user.name at sip.othersipgw.com) State DESTROY going
to sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:735
(sofia/external/0diallednumber at first.sipgw.net) Callstate Change
RINGING -> HANGUP
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] switch_core_state_machine.c:737
(sofia/external/0diallednumber at first.sipgw.net) State HANGUP
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] mod_sofia.c:407 sofia/external/0diallednumber at first.sipgw.net
Overriding SIP cause 480 with 480 from the other leg
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.965334
[DEBUG] mod_sofia.c:413 Channel
sofia/external/0diallednumber at first.sipgw.net hanging up, cause:
NORMAL_CLEARING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] mod_sofia.c:549 Responding to INVITE with: 480
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:60
sofia/external/0diallednumber at first.sipgw.net Standard HANGUP, cause:
NORMAL_CLEARING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:737
(sofia/external/0diallednumber at first.sipgw.net) State HANGUP going to
sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:504
(sofia/external/0diallednumber at first.sipgw.net) State Change CS_HANGUP
-> CS_REPORTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_REPORTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:823
(sofia/external/0diallednumber at first.sipgw.net) State REPORTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:104
sofia/external/0diallednumber at first.sipgw.net Standard REPORTING,
cause: NORMAL_CLEARING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:737
(sofia/external/0diallednumber at first.sipgw.net) State HANGUP going to
sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:504
(sofia/external/0diallednumber at first.sipgw.net) State Change CS_HANGUP
-> CS_REPORTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:472
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_REPORTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:823
(sofia/external/0diallednumber at first.sipgw.net) State REPORTING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:104
sofia/external/0diallednumber at first.sipgw.net Standard REPORTING,
cause: NORMAL_CLEARING
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:823
(sofia/external/0diallednumber at first.sipgw.net) State REPORTING going
to sleep
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:498
(sofia/external/0diallednumber at first.sipgw.net) State Change
CS_REPORTING -> CS_DESTROY
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_session.c:1615 Session 231
(sofia/external/0diallednumber at first.sipgw.net) Locked, Waiting on
external entities
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_session.c:1388 Send signal
sofia/external/0diallednumber at first.sipgw.net [BREAK]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_session.c:1615 Session 231
(sofia/external/0diallednumber at first.sipgw.net) Locked, Waiting on
external entities
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[NOTICE] switch_core_session.c:1633 Session 231
(sofia/external/0diallednumber at first.sipgw.net) Ended
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[NOTICE] switch_core_session.c:1637 Close Channel
sofia/external/0diallednumber at first.sipgw.net [CS_DESTROY]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[NOTICE] switch_core_session.c:1637 Close Channel
sofia/external/0diallednumber at first.sipgw.net [CS_DESTROY]
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:626
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_DESTROY
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:626
(sofia/external/0diallednumber at first.sipgw.net) Running State Change
CS_DESTROY
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:636
(sofia/external/0diallednumber at first.sipgw.net) State DESTROY
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] mod_sofia.c:323 sofia/external/0diallednumber at first.sipgw.net
SOFIA DESTROY
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:111
sofia/external/0diallednumber at first.sipgw.net Standard DESTROY
1517b746-8051-11e4-909f-2d01d6dac59f 2014-12-10 10:44:09.985315
[DEBUG] switch_core_state_machine.c:636
(sofia/external/0diallednumber at first.sipgw.net) State DESTROY going to
sleep

On Wed, Dec 10, 2014 at 6:50 AM, Avi Marcus <avi at avimarcus.net> wrote:
> I would try something "simpler". This will always play the file, though, no
> matter the response from the othersipgw.
>
>
> <extension name="1212">
>        <action application="set" data="hangup_after_bridge=false"/>
>        <action application="bridge"
> data="sofia/external/sip:user.name at sip.othersipgw.ip.address.com;${effective_caller_id_number}"/>
>        <action application="transfer" data="1213"/>
>  </condition>
> </extension>
>
> <!-- What to do after othersipgw drops the connection-->
> <extension name="1213">
>   <condition field="destination_number" expression="1213">
>     <action application="playback" data="ivr/ivr-thank_you.wav"/>
>    </condition>
> </extension>
>


From jobindcruz at gmail.com  Wed Dec 10 15:13:40 2014
From: jobindcruz at gmail.com (jobin dcruz)
Date: Wed, 10 Dec 2014 17:43:40 +0530
Subject: [Freeswitch-users] Call Conference Time out issue
Message-ID: <CAKcJ+11xnR-JbzgRdEwKUJkEiGW7f_ovL8GcZ70G4Zf7w7hSMA@mail.gmail.com>

Hi,

I had set call conference hang up time,but it is working after 4 seconds as
per attendees count.

This is dial plan example

<action application="set" data="execute_on_answer_2=sched_hangup +${time}
alloted_timeout"/>

Ex: i had set one min(60 sec) hangup time,but call conference ended with 64
sec
-- 
Jobin D'cruz (LAMP Developer)
Email: jobindcruz at gmail.com | lampdeveloper.jobindcruz at gmail.com



Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework
Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI
MooTools Smarty Firefox Extension
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From fs at voice2net.ca  Wed Dec 10 15:52:41 2014
From: fs at voice2net.ca (Darcy Primrose)
Date: Wed, 10 Dec 2014 07:52:41 -0500
Subject: [Freeswitch-users] cisco spa50xG and music on hold
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com><0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com><CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com><0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com><CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com><0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com><CACK8u8J1b1UcEQ1R+M0puRT9vACtYGeuKBhHs_2ug9Q_qgJ6Wg@mail.gmail.com><0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com>
	<CACK8u8K+DA7qGEHoYnpHms_tetv-wQaQpafEv+e5MPZcmvizOg@mail.gmail.com>
Message-ID: <45748814B5CE4A8C9CC11B597D06F498@DARCY>

We have a problem with music on hold on the 504g phones, wonder if anyone 
has this working.  When  an inbound call to a 504g is placed on hold, the 
stream is played towards to cisco504g instead of the calling party who is 
now on hold.  In the Cisco 504g we have the moh server set blank.  Also, if 
we park the call to the 504g, retrieve it, then place it on hold, then the 
music on hold works.

Any one have any thoughts or ideas on this.

Darcy Primrose
Voice2Net 



From vipkilla at gmail.com  Wed Dec 10 16:00:14 2014
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 10 Dec 2014 08:00:14 -0500
Subject: [Freeswitch-users] cisco spa50xG and music on hold
In-Reply-To: <45748814B5CE4A8C9CC11B597D06F498@DARCY>
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com>
	<0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com>
	<CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com>
	<0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com>
	<CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com>
	<0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com>
	<CACK8u8J1b1UcEQ1R+M0puRT9vACtYGeuKBhHs_2ug9Q_qgJ6Wg@mail.gmail.com>
	<0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com>
	<CACK8u8K+DA7qGEHoYnpHms_tetv-wQaQpafEv+e5MPZcmvizOg@mail.gmail.com>
	<45748814B5CE4A8C9CC11B597D06F498@DARCY>
Message-ID: <CAC-LwPPhofO+3_sMc4EP=7dgL3vhEmNdTMjzS+-a=71+0ZC+dA@mail.gmail.com>

Hi Darcy,
I know there are some issues with MOH right now.
Please see https://freeswitch.org/jira/browse/FS-7015
I had to revert to commit 39be8777608f3eed13738a50e03c0969b366414b which
doesn't seem to have the MOH issues.
Thanks
/V

On Wed, Dec 10, 2014 at 7:52 AM, Darcy Primrose <fs at voice2net.ca> wrote:

> We have a problem with music on hold on the 504g phones, wonder if anyone
> has this working.  When  an inbound call to a 504g is placed on hold, the
> stream is played towards to cisco504g instead of the calling party who is
> now on hold.  In the Cisco 504g we have the moh server set blank.  Also, if
> we park the call to the 504g, retrieve it, then place it on hold, then the
> music on hold works.
>
> Any one have any thoughts or ideas on this.
>
> Darcy Primrose
> Voice2Net
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fs at voice2net.ca  Wed Dec 10 16:23:29 2014
From: fs at voice2net.ca (Darcy Primrose)
Date: Wed, 10 Dec 2014 08:23:29 -0500
Subject: [Freeswitch-users] cisco spa50xG and music on hold
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com><0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com><CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com><0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com><CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com><0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com><CACK8u8J1b1UcEQ1R+M0puRT9vACtYGeuKBhHs_2ug9Q_qgJ6Wg@mail.gmail.com><0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com><CACK8u8K+DA7qGEHoYnpHms_tetv-wQaQpafEv+e5MPZcmvizOg@mail.gmail.com><45748814B5CE4A8C9CC11B597D06F498@DARCY>
	<CAC-LwPPhofO+3_sMc4EP=7dgL3vhEmNdTMjzS+-a=71+0ZC+dA@mail.gmail.com>
Message-ID: <98566ABE87D844A7BA99E0ED4C433571@DARCY>

This happens on all Versions of Freeswitch I have running with the Cisco spa50xG phones, I have tried it on a version that is relatively stable from 2012 and on the latest version I just download from GIT, All other phones work perfectly, that is Snom, Grandstream and Yealink.

Darcy
  ----- Original Message ----- 
  From: Vik Killa 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, December 10, 2014 8:00 AM
  Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold


  Hi Darcy,
  I know there are some issues with MOH right now. 
  Please see https://freeswitch.org/jira/browse/FS-7015
  I had to revert to commit 39be8777608f3eed13738a50e03c0969b366414b which doesn't seem to have the MOH issues.
  Thanks
  /V


  On Wed, Dec 10, 2014 at 7:52 AM, Darcy Primrose <fs at voice2net.ca> wrote:

    We have a problem with music on hold on the 504g phones, wonder if anyone
    has this working.  When  an inbound call to a 504g is placed on hold, the
    stream is played towards to cisco504g instead of the calling party who is
    now on hold.  In the Cisco 504g we have the moh server set blank.  Also, if
    we park the call to the 504g, retrieve it, then place it on hold, then the
    music on hold works.

    Any one have any thoughts or ideas on this.

    Darcy Primrose
    Voice2Net


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

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

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





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


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

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

  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.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  Wed Dec 10 16:52:58 2014
From: miha at softnet.si (Miha)
Date: Wed, 10 Dec 2014 14:52:58 +0100
Subject: [Freeswitch-users] Listening on two different ports
Message-ID: <5488503A.1080502@softnet.si>

Hi,

which config should be made that FS will listen on two different ports 
(for exp 5060 and 5070) and also sending on two different ports.

I have created two external profiles, so I must create  also to 
different internal profiles?

br
miha



From bote_radio at botecomm.com  Wed Dec 10 17:31:18 2014
From: bote_radio at botecomm.com (Bote Man)
Date: Wed, 10 Dec 2014 09:31:18 -0500
Subject: [Freeswitch-users] Listening on two different ports
In-Reply-To: <5488503A.1080502@softnet.si>
References: <5488503A.1080502@softnet.si>
Message-ID: <038301d01485$f70e74b0$e52b5e10$@com>

One profile listens on one port.

The names "external" and "internal" are arbitrary, just be careful as to
which dialplan you hand off inbound calls so that they are processed
with minimal permissions necessary.

Bote


> -----Original Message-----
> From: Miha
> Sent: Wednesday, 10 December, 2014 08:53
> Subject: [Freeswitch-users] Listening on two different ports
> 
> Hi,
> 
> which config should be made that FS will listen on two different ports
> (for exp 5060 and 5070) and also sending on two different ports.
> 
> I have created two external profiles, so I must create  also to
> different internal profiles?
> 
> br
> miha
> 
> 




From vipkilla at gmail.com  Wed Dec 10 17:33:06 2014
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 10 Dec 2014 09:33:06 -0500
Subject: [Freeswitch-users] cisco spa50xG and music on hold
In-Reply-To: <98566ABE87D844A7BA99E0ED4C433571@DARCY>
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com>
	<0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com>
	<CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com>
	<0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com>
	<CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com>
	<0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com>
	<CACK8u8J1b1UcEQ1R+M0puRT9vACtYGeuKBhHs_2ug9Q_qgJ6Wg@mail.gmail.com>
	<0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com>
	<CACK8u8K+DA7qGEHoYnpHms_tetv-wQaQpafEv+e5MPZcmvizOg@mail.gmail.com>
	<45748814B5CE4A8C9CC11B597D06F498@DARCY>
	<CAC-LwPPhofO+3_sMc4EP=7dgL3vhEmNdTMjzS+-a=71+0ZC+dA@mail.gmail.com>
	<98566ABE87D844A7BA99E0ED4C433571@DARCY>
Message-ID: <CAC-LwPNRMV6cc4GKkP4Z19YDbyquBrj13WpRuZ7Vf5ksPChWXQ@mail.gmail.com>

I have two SPA50X setup with FS and I'm not having this issue.

On Wed, Dec 10, 2014 at 8:23 AM, Darcy Primrose <fs at voice2net.ca> wrote:

>  This happens on all Versions of Freeswitch I have running with the Cisco
> spa50xG phones, I have tried it on a version that is relatively stable from
> 2012 and on the latest version I just download from GIT, All other phones
> work perfectly, that is Snom, Grandstream and Yealink.
>
> Darcy
>
> ----- Original Message -----
> *From:* Vik Killa <vipkilla at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, December 10, 2014 8:00 AM
> *Subject:* Re: [Freeswitch-users] cisco spa50xG and music on hold
>
> Hi Darcy,
> I know there are some issues with MOH right now.
> Please see https://freeswitch.org/jira/browse/FS-7015
> I had to revert to commit 39be8777608f3eed13738a50e03c0969b366414b which
> doesn't seem to have the MOH issues.
> Thanks
> /V
>
> On Wed, Dec 10, 2014 at 7:52 AM, Darcy Primrose <fs at voice2net.ca> wrote:
>
>> We have a problem with music on hold on the 504g phones, wonder if anyone
>> has this working.  When  an inbound call to a 504g is placed on hold, the
>> stream is played towards to cisco504g instead of the calling party who is
>> now on hold.  In the Cisco 504g we have the moh server set blank.  Also,
>> if
>> we park the call to the 504g, retrieve it, then place it on hold, then the
>> music on hold works.
>>
>> Any one have any thoughts or ideas on this.
>>
>> Darcy Primrose
>> Voice2Net
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From aqsyounas at gmail.com  Wed Dec 10 17:39:14 2014
From: aqsyounas at gmail.com (Aqs Younas)
Date: Wed, 10 Dec 2014 19:39:14 +0500
Subject: [Freeswitch-users] error: Frame size too big: 12121
Message-ID: <CAGVOjVXp+6p-Ad2bfQxtH2RCS12g_gXz78S1zu4KAF9LpO0_zA@mail.gmail.com>

Hi, All

I am new to free-switch. Sometimes i encounter this error while playing  a
stream with mod_shout.

These are the logs that i see while running freeswitch without background.

[parse.c:1026] error: Frame size too big: 12121
[parse.c:1026] error: Frame size too big: 12121
[parse.c:1026] error: Frame size too big: 12121
[parse.c:1026] error: Frame size too big: 12121
[parse.c:1026] error: Frame size too big: 12121
[parse.c:1026] error: Frame size too big: 12121

2014-12-10 14:30:38.149601 [ERR] mod_shout.c:692 Error opening
streaming205.radionomy.com/Aurora-de-Siempre?group=0&countrycode=SV&similarprocess=0&crashed=0
(data stream timeout)

Any Help would be much appreciated.
Thanks.
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From bote_radio at botecomm.com  Wed Dec 10 17:42:04 2014
From: bote_radio at botecomm.com (Bote Man)
Date: Wed, 10 Dec 2014 09:42:04 -0500
Subject: [Freeswitch-users] cisco spa50xG and music on hold
In-Reply-To: <98566ABE87D844A7BA99E0ED4C433571@DARCY>
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com><0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com><CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com><0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com><CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com><0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com><CACK8u8J1b1UcEQ1R+M0puRT9vACtYGeuKBhHs_2ug9Q_qgJ6Wg@mail.gmail.com><0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com><CACK8u8K+DA7qGEHoYnpHms_tetv-wQaQpafEv+e5MPZcmvizOg@mail.gmail.com><45748814B5CE4A8C9CC11B597D06F498@DARCY>	<CAC-LwPPhofO+3_sMc4EP=7dgL3vhEmNdTMjzS+-a=71+0ZC+dA@mail.gmail.com>
	<98566ABE87D844A7BA99E0ED4C433571@DARCY>
Message-ID: <038401d01487$7794b800$66be2800$@com>

Can you compare the signaling sent by the Cisco phones against the signaling sent by the others like the Grandstream? It sounds like the Cisco is sending a message that is confusing FS.

 

Although I have read comments here that indicate that Cisco phones do lots of strange things :- )

 

Bote

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Darcy Primrose
Sent: Wednesday, 10 December, 2014 08:23
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold

 

This happens on all Versions of Freeswitch I have running with the Cisco spa50xG phones, I have tried it on a version that is relatively stable from 2012 and on the latest version I just download from GIT, All other phones work perfectly, that is Snom, Grandstream and Yealink.

 

Darcy

----- Original Message ----- 

From: Vik Killa <mailto:vipkilla at gmail.com>  

To: FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org>  

Sent: Wednesday, December 10, 2014 8:00 AM

Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold

 

Hi Darcy, 

I know there are some issues with MOH right now. 

Please see https://freeswitch.org/jira/browse/FS-7015

I had to revert to commit 39be8777608f3eed13738a50e03c0969b366414b which doesn't seem to have the MOH issues.

Thanks

/V

 

On Wed, Dec 10, 2014 at 7:52 AM, Darcy Primrose <fs at voice2net.ca> wrote:

We have a problem with music on hold on the 504g phones, wonder if anyone
has this working.  When  an inbound call to a 504g is placed on hold, the
stream is played towards to cisco504g instead of the calling party who is
now on hold.  In the Cisco 504g we have the moh server set blank.  Also, if
we park the call to the 504g, retrieve it, then place it on hold, then the
music on hold works.

Any one have any thoughts or ideas on this.

Darcy Primrose
Voice2Net


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

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

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

 

  _____  

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

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

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

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From bilaln018 at gmail.com  Wed Dec 10 18:09:55 2014
From: bilaln018 at gmail.com (Bilal Abbasi)
Date: Wed, 10 Dec 2014 20:09:55 +0500
Subject: [Freeswitch-users] [Issue][CDR with mod_cdr_mongodb]
Message-ID: <CADSVerCY_8keFnoJAe90QM8xUSGMotmNY=e3VOHc57EEQxsJ_w@mail.gmail.com>

Hi All,
I am newbie in Freeswitch, i have successfully configured mongodb with
freeswitch.
The issue is that there is no schema provided for mongodb collection CDR.
So when ever i put a test call it logs too much information ,so what should
i do if i want limited variables to log in that collecction?
Please help

Regards
Bilal Abbasi
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From fs at voice2net.ca  Wed Dec 10 19:02:15 2014
From: fs at voice2net.ca (Darcy Primrose)
Date: Wed, 10 Dec 2014 11:02:15 -0500
Subject: [Freeswitch-users] cisco spa50xG and music on hold
References: <CACK8u8+XVL-woGCVBMWG6=i6YRxDXR_yaTQV2gpM9FfQ5PZY+w@mail.gmail.com><0000014a0b2a7f3a-e057e705-e811-4903-b2b9-bbd020abb4a7-000000@email.amazonses.com><CACK8u8JS9eO+ReCV1mPhUufZfcv3c4ummzz+1jYSBOWG=ri0UA@mail.gmail.com><0000014a22d99893-8fb2f862-1a14-4cb2-b363-8e924d56012d-000000@email.amazonses.com><CACK8u8K+A+0_9HiQv4g7qwAP8C56aQ6LE-Z38xy3-KQsgYo+Yg@mail.gmail.com><0000014a2a0f8f9d-1038c04d-e533-483b-8322-bd27ace4ba19-000000@email.amazonses.com><CACK8u8J1b1UcEQ1R+M0puRT9vACtYGeuKBhHs_2ug9Q_qgJ6Wg@mail.gmail.com><0000014a32c00982-5991e950-58e0-477c-bcab-ead784e5d4d6-000000@email.amazonses.com><CACK8u8K+DA7qGEHoYnpHms_tetv-wQaQpafEv+e5MPZcmvizOg@mail.gmail.com><45748814B5CE4A8C9CC11B597D06F498@DARCY>	<CAC-LwPPhofO+3_sMc4EP=7dgL3vhEmNdTMjzS+-a=71+0ZC+dA@mail.gmail.com><98566ABE87D844A7BA99E0ED4C433571@DARCY>
	<038401d01487$7794b800$66be2800$@com>
Message-ID: <BC5E1518E9A64CC0A9B67DAB99C60B03@DARCY>

The Cisco does the sendonly invite to mod_sofia at domain where as the other products send the sendonly invite to 1001 at domain, that seems to be the main difference.  I will poke at it to see why. I am also running the lastest version of cisco software, perhaps they did a recent patch that caused this.

Darcy
  ----- Original Message ----- 
  From: Bote Man 
  To: 'FreeSWITCH Users Help' 
  Sent: Wednesday, December 10, 2014 9:42 AM
  Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold


  Can you compare the signaling sent by the Cisco phones against the signaling sent by the others like the Grandstream? It sounds like the Cisco is sending a message that is confusing FS.

   

  Although I have read comments here that indicate that Cisco phones do lots of strange things :- )

   

  Bote

   

   

  From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Darcy Primrose
  Sent: Wednesday, 10 December, 2014 08:23
  To: FreeSWITCH Users Help
  Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold

   

  This happens on all Versions of Freeswitch I have running with the Cisco spa50xG phones, I have tried it on a version that is relatively stable from 2012 and on the latest version I just download from GIT, All other phones work perfectly, that is Snom, Grandstream and Yealink.

   

  Darcy

    ----- Original Message ----- 

    From: Vik Killa 

    To: FreeSWITCH Users Help 

    Sent: Wednesday, December 10, 2014 8:00 AM

    Subject: Re: [Freeswitch-users] cisco spa50xG and music on hold

     

    Hi Darcy, 

    I know there are some issues with MOH right now. 

    Please see https://freeswitch.org/jira/browse/FS-7015

    I had to revert to commit 39be8777608f3eed13738a50e03c0969b366414b which doesn't seem to have the MOH issues.

    Thanks

    /V

     

    On Wed, Dec 10, 2014 at 7:52 AM, Darcy Primrose <fs at voice2net.ca> wrote:

    We have a problem with music on hold on the 504g phones, wonder if anyone
    has this working.  When  an inbound call to a 504g is placed on hold, the
    stream is played towards to cisco504g instead of the calling party who is
    now on hold.  In the Cisco 504g we have the moh server set blank.  Also, if
    we park the call to the 504g, retrieve it, then place it on hold, then the
    music on hold works.

    Any one have any thoughts or ideas on this.

    Darcy Primrose
    Voice2Net


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

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

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

     


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

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

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

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



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


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

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

  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From telishisheer at gmail.com  Wed Dec 10 19:42:37 2014
From: telishisheer at gmail.com (Shisheer Teli)
Date: Wed, 10 Dec 2014 22:12:37 +0530
Subject: [Freeswitch-users] Ldap alise and freeswitch integration problem
In-Reply-To: <CACkRGLyAT_HwST0zGPfJX2pEsAJhYPZ9dqWMdkvnTZbyG8hbiA@mail.gmail.com>
References: <CACkRGLyAT_HwST0zGPfJX2pEsAJhYPZ9dqWMdkvnTZbyG8hbiA@mail.gmail.com>
Message-ID: <CACkRGLzfEzLuBuEdpkgc5b32-a6DW75Q56JV6yjAchbg+UdK7Q@mail.gmail.com>

 from my experiments and test i found that freeswitch sending password in
plain text.

Test 1:
when i set ldap password in md5 , in freeswitch cli i see hash password and
authentication failed.

Test 2:
when i set ldap password in plain text, in freeswitch cli i can see plain
text password and authentication success.

On Tue, Dec 2, 2014 at 9:55 AM, Shisheer Teli <telishisheer at gmail.com>
wrote:

> Hi,
>
> I am able to bind with any alise on ldap server except userPassword (MD5)
> alise.
>
> when i bind password with userPassword , authentication fails.
>
> any MD5 authentication configuration in freeswitch?
>
> Configuration file:
> <configuration name="xml_ldap.conf">
>
> <bindings>
> <binding name="directory">
>   <param name="basedn" value="dc=example,dc=com"/>
>   <param name="filter" value="(telephoneNumber=%d)" bindings="directory"/>
>   <param name="url" value="ldap://ldap.example.com"/>
>   <param name="binddn" value="cn=use,dc=example,dc=com"/>
>   <param name="bindpass" value="XXXX"/>
>    <trans>
>    <tran name="id" mapfrom="uid"/>
>    <tran name="password" mapfrom="userPassword"/>
>    </trans>
> </binding>
> </bindings>
>
> </configuration>
>
>
>
> --
> Regards,
> Shisheer T
> Phone: +91-022 2278 2763
>
>


-- 
Regards,
Shisheer Teli
Phone: +91-022 2278 2519 / 2121
shisheer at tifr.res.in
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From hexade at hotmail.com  Wed Dec 10 19:45:40 2014
From: hexade at hotmail.com (Adelia C.)
Date: Wed, 10 Dec 2014 11:45:40 -0500
Subject: [Freeswitch-users] Twilio SIP trunk config - outbound call rejected
In-Reply-To: <COL128-W64053C71E7D5A496970947DA620@phx.gbl>
References: <COL128-W64053C71E7D5A496970947DA620@phx.gbl>
Message-ID: <COL128-W35CBC229939AB458A49047DA620@phx.gbl>


Any idea why this INVITE is rejected with 400 Invalid Phone Number by Twilio?
The first INVITE asks for Proxy Authentication, that seems to pass and this is rejected.
Thank you!

INVITE sip:5101112222 at mycompany.pstn.twilio.com SIP/2.0
Via: SIP/2.0/UDP 10.50.1.116:5080;rport;branch=z9hG4bKa8QvU1B6QmeKc
Max-Forwards: 69
From: "unknown" <sip:4152221111 at mycompany.pstn.twilio.com>;tag=6UetZr6rDZgHF
To: <sip:5101112222 at mycompany.pstn.twilio.com>
Call-ID: 1ffda8b7-faaf-1232-8380-a9a65d1e96b1
CSeq: 68751485 INVITE
Contact: <sip:gw+twiliotrunk at 10.50.1.116:5080;transport=udp;gw=twiliotrunk>
Expires: 3600
User-Agent: FreeSWITCH-mod_sofia/1.5.12b~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Proxy-Authorization: Digest username="4152221111", realm="sip.twilio.com", nonce="f722ae99d0a818bc422baa5d8", cnonce="IA1gJvqvEg6mmXR6WsQ", opaque="e89d2328e07da020d75a", algorithm=MD5, uri="sip:5101112222 at mycompany.pstn.twilio.com", response="e65d3dea5b1e3b35306cfa121ec2ae4d", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 268
X-FS-Support: update_display,send_info
Remote-Party-ID: "unknown" <sip:1000 at mycompany.pstn.twilio.com>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1418153731 1418153732 IN IP4 10.50.1.116
s=FreeSWITCH
c=IN IP4 10.50.1.116
t=0 0
m=audio 21366 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--

SIP/2.0 400 Invalid phone number
To: <sip:5101112222 at ingenio.pstn.twilio.com>;tag=08602519_6772d868_5eef74bb-957c-486e-b261-2a2e63704d73
Via: SIP/2.0/UDP 10.50.1.116:5080;received=10.50.1.116;rport=5080;branch=z9hG4bKa8QvU1B6QmeKc
CSeq: 68751485 INVITE
Call-ID: 1ffda8b7-faaf-1232-8380-a9a65d1e96b1
From: "unknown" <sip:4152221111 at ingenio.pstn.twilio.com>;tag=6UetZr6rDZgHF
Contact: <sip:171.118.14.130:5060>
Content-Length: 0



--
 		 	   		   		 	   		  
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From mike at jerris.com  Wed Dec 10 20:28:07 2014
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Dec 2014 12:28:07 -0500
Subject: [Freeswitch-users] Mod perl using $session->setVariable inline
In-Reply-To: <CAJR0VfFO0P_zqWAWiKY1h1dubzivG+a8pX54KguWcsTjjgzB3g@mail.gmail.com>
References: <CAJR0VfFO0P_zqWAWiKY1h1dubzivG+a8pX54KguWcsTjjgzB3g@mail.gmail.com>
Message-ID: <38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com>

If you are in a perl script using session->setVariable, how are you still in the dialplan?  This doesn't make sense to me.  There should be no need for an inline if you are already in the script.

> On Dec 10, 2014, at 4:20 AM, Liam Farr <liam at maxumdata.com> wrote:
> 
> Hi,
> 
> I need to set a variable using a perl script inline.
> 
> Normally I would use $session->setVariable('some_var',$my_value);
> 
> However I cant find any options to execute this inline / as an inline action. (I need to evaluate the variable later in the hunting phase of the dialplan hence need it to be an inline action).
> 
> Is there a way to do this?
> 
> Or is there another way to achieve the same result, e.g. hacking $session->execute("set","some_var=$my_value") to run as an inline action?
> 
> Any help would be much appreciated, I've scoured the wiki, confluence and git repo and come up empty.
> 
> -- 
> Kind Regards
> 
> 
> Liam Farr



From wampir990 at gmail.com  Wed Dec 10 20:48:19 2014
From: wampir990 at gmail.com (Jacekalex)
Date: Wed, 10 Dec 2014 18:48:19 +0100
Subject: [Freeswitch-users] Mod_sofia build error.
Message-ID: <54888763.1020107@gmail.com>


Hi

OS  Gentoo Linux x86_64
gcc (Gentoo Hardened 4.8.3 p1.1, pie-0.5.9) 4.8.3

The previous system:
gcc (Gentoo Hardened 4.7.3-r1 p1.3, pie-0.5.5) 4.7.3

Freeswitch versions - all 1.4.x

Mod_sofia  - sofia-sip error:
...
Making all in tport
mke[9]: Entering directory
'/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
     LTake[8]: Entering directory
'/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
make  all-am
maCOMPILE tport_type_sctp.lo
tport_type_sctp.c: In function ?tport_sctp_init_socket?:
tport_type_sctp.c:206:10: error: variable ?initmsg? has initializer but
incomplete type
   struct sctp_initmsg initmsg = { 0 };
          ^
tport_type_sctp.c:206:10: error: excess elements in struct initializer
[-Werror]
tport_type_sctp.c:206:10: error: (near initialization for ?initmsg?)
[-Werror]
tport_type_sctp.c:206:23: error: storage size of ?initmsg? isn?t known
   struct sctp_initmsg initmsg = { 0 };
                       ^
tport_type_sctp.c:211:36: error: ?SCTP_INITMSG? undeclared (first use in
this function)
   if (setsockopt(socket, SOL_SCTP, SCTP_INITMSG, &initmsg, sizeof
initmsg) < 0)
                                    ^
tport_type_sctp.c:211:36: note: each undeclared identifier is reported
only once for each function it appears in
tport_type_sctp.c:206:23: error: unused variable ?initmsg?
[-Werror=unused-variable]
   struct sctp_initmsg initmsg = { 0 };
                       ^
cc1: all warnings being treated as errors
Makefile:1452: recipe for target 'tport_type_sctp.lo' failed
make[9]: *** [tport_type_sctp.lo] Error 1
make[9]: Leaving directory
'/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
Makefile:736: recipe for target 'all' failed
make[8]: *** [all] Error 2
make[8]: Leaving directory
'/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
Makefile:547: recipe for target 'all-recursive' failed
make[7]: *** [all-recursive] Error 1
make[7]: Leaving directory
'/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua'
Makefile:490: recipe for target 'all-recursive' failed
make[6]: *** [all-recursive] Error 1
make[6]: Leaving directory
'/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip'
Makefile:410: recipe for target 'all' failed
make[5]: *** [all] Error 2
make[5]: Leaving directory
'/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip'
Makefile:906: recipe for target
'/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la'
failed
make[4]: ***
[/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la]
Error 2
make[4]: Leaving directory
'/home/fabryka/local/src/freeswitch/freeswitch481/src/mod/endpoints/mod_sofia'
Makefile:577: recipe for target 'mod_sofia-all' failed
make[3]: *** [mod_sofia-all] Error 1
make[3]: Leaving directory
'/home/fabryka/local/src/freeswitch/freeswitch481/src/mod'
Makefile:491: recipe for target 'all-recursive' failed
make[2]: *** [all-recursive] Error 1
make[2]: Leaving directory
'/home/fabryka/local/src/freeswitch/freeswitch481/src'
Makefile:2487: recipe for target 'all-recursive' failed
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory
'/home/fabryka/local/src/freeswitch/freeswitch481'
Makefile:969: recipe for target 'all' failed
make: *** [all] Error 2


Error amazes me because sofia-sip library  from  portage ebuild
installed without a problem:
 qlist -UCvq sofia-sip
net-libs/sofia-sip-1.12.11 ssl static-libs
http://data.gpo.zugaina.org/gentoo/net-libs/sofia-sip/


The latest version of FreeSWITCH, I have been able to install without a
problem are:
FreeSWITCH version: 1.5.14b+git~20140725T212415Z~c411f8c7a9~64bit (git
c411f8c 2014-07-25 21:24:15Z 64bit)


Where can I find the source FreeSWITCH-1.5.x, or alternatively, how to
improve this error?


Cheers

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From liam at maxumdata.com  Wed Dec 10 20:56:33 2014
From: liam at maxumdata.com (Liam Farr)
Date: Thu, 11 Dec 2014 06:56:33 +1300
Subject: [Freeswitch-users] Mod perl using $session->setVariable inline
In-Reply-To: <38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com>
References: <CAJR0VfFO0P_zqWAWiKY1h1dubzivG+a8pX54KguWcsTjjgzB3g@mail.gmail.com>
	<38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com>
Message-ID: <CAJR0VfH2dnUJ1C3Wuvq4PEOugqpwU3QYOCwbvrTnEvaR48BvxQ@mail.gmail.com>

Heres an expanded example;



Extension runs the perl script as part of the public dialplan first;


<extension name="run-script" continue="true">

   <condition>

  <action application="perl" data="/etc/freeswitch/scripts/my-script.pl"/>

   </condition>

</extension>



Script sets some variables;


$my_value = "123";

$session->setVariable('some_var',$my_value);



Use that variable later in another extension in the public dialplan as a
condition selector;



<extension name="bridge-call" >

   <condition field="context" expression="public" />

   <condition field="${some_var}" expression="123" >

   <action application="set" data="call_direction=inbound"/>

   <action application="bridge" data="sofia/internal/${
destination_number}@1.2.3.4:5060" />

   </condition>

</extension>



However this will fail, because the perl script "$session->setVariable" run
earlier is not inline, so you cant make use of it in the condition
(<condition field="${some_var}" expression="123" >) in the second extension
that exists in a later part of the public dialplan. Hence I need a way to
set inline variables from my perl script.



Cheers


Liam Farr



On 11 December 2014 at 06:28, Michael Jerris <mike at jerris.com> wrote:

> If you are in a perl script using session->setVariable, how are you still
> in the dialplan?  This doesn't make sense to me.  There should be no need
> for an inline if you are already in the script.
>
> > On Dec 10, 2014, at 4:20 AM, Liam Farr <liam at maxumdata.com> wrote:
> >
> > Hi,
> >
> > I need to set a variable using a perl script inline.
> >
> > Normally I would use $session->setVariable('some_var',$my_value);
> >
> > However I cant find any options to execute this inline / as an inline
> action. (I need to evaluate the variable later in the hunting phase of the
> dialplan hence need it to be an inline action).
> >
> > Is there a way to do this?
> >
> > Or is there another way to achieve the same result, e.g. hacking
> $session->execute("set","some_var=$my_value") to run as an inline action?
> >
> > Any help would be much appreciated, I've scoured the wiki, confluence
> and git repo and come up empty.
> >
> > --
> > Kind Regards
> >
> >
> > Liam Farr
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 10 20:58:19 2014
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Dec 2014 12:58:19 -0500
Subject: [Freeswitch-users] Mod_sofia build error.
In-Reply-To: <54888763.1020107@gmail.com>
References: <54888763.1020107@gmail.com>
Message-ID: <422D5329-EA7F-4D72-8D2D-BD9EAF8D7F32@jerris.com>

This could be related to bugs in gentoo headers related to sctp or due to issues in the autoconf detection for these.  We would be happy to take patches to correct this issue that don't break other distro/os if you can provide a patch.


> On Dec 10, 2014, at 12:48 PM, Jacekalex <wampir990 at gmail.com> wrote:
> 
> 
> Hi
> 
> OS  Gentoo Linux x86_64
> gcc (Gentoo Hardened 4.8.3 p1.1, pie-0.5.9) 4.8.3
> 
> The previous system:
> gcc (Gentoo Hardened 4.7.3-r1 p1.3, pie-0.5.5) 4.7.3
> 
> Freeswitch versions - all 1.4.x
> 
> Mod_sofia  - sofia-sip error:
> ...
> Making all in tport
> mke[9]: Entering directory
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
>     LTake[8]: Entering directory
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
> make  all-am
> maCOMPILE tport_type_sctp.lo
> tport_type_sctp.c: In function ?tport_sctp_init_socket?:
> tport_type_sctp.c:206:10: error: variable ?initmsg? has initializer but
> incomplete type
>   struct sctp_initmsg initmsg = { 0 };
>          ^
> tport_type_sctp.c:206:10: error: excess elements in struct initializer
> [-Werror]
> tport_type_sctp.c:206:10: error: (near initialization for ?initmsg?)
> [-Werror]
> tport_type_sctp.c:206:23: error: storage size of ?initmsg? isn?t known
>   struct sctp_initmsg initmsg = { 0 };
>                       ^
> tport_type_sctp.c:211:36: error: ?SCTP_INITMSG? undeclared (first use in
> this function)
>   if (setsockopt(socket, SOL_SCTP, SCTP_INITMSG, &initmsg, sizeof
> initmsg) < 0)
>                                    ^
> tport_type_sctp.c:211:36: note: each undeclared identifier is reported
> only once for each function it appears in
> tport_type_sctp.c:206:23: error: unused variable ?initmsg?
> [-Werror=unused-variable]
>   struct sctp_initmsg initmsg = { 0 };
>                       ^
> cc1: all warnings being treated as errors
> Makefile:1452: recipe for target 'tport_type_sctp.lo' failed
> make[9]: *** [tport_type_sctp.lo] Error 1
> make[9]: Leaving directory
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
> Makefile:736: recipe for target 'all' failed
> make[8]: *** [all] Error 2
> make[8]: Leaving directory
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
> Makefile:547: recipe for target 'all-recursive' failed
> make[7]: *** [all-recursive] Error 1
> make[7]: Leaving directory
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua'
> Makefile:490: recipe for target 'all-recursive' failed
> make[6]: *** [all-recursive] Error 1
> make[6]: Leaving directory
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip'
> Makefile:410: recipe for target 'all' failed
> make[5]: *** [all] Error 2
> make[5]: Leaving directory
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip'
> Makefile:906: recipe for target
> '/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la'
> failed
> make[4]: ***
> [/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la]
> Error 2
> make[4]: Leaving directory
> '/home/fabryka/local/src/freeswitch/freeswitch481/src/mod/endpoints/mod_sofia'
> Makefile:577: recipe for target 'mod_sofia-all' failed
> make[3]: *** [mod_sofia-all] Error 1
> make[3]: Leaving directory
> '/home/fabryka/local/src/freeswitch/freeswitch481/src/mod'
> Makefile:491: recipe for target 'all-recursive' failed
> make[2]: *** [all-recursive] Error 1
> make[2]: Leaving directory
> '/home/fabryka/local/src/freeswitch/freeswitch481/src'
> Makefile:2487: recipe for target 'all-recursive' failed
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory
> '/home/fabryka/local/src/freeswitch/freeswitch481'
> Makefile:969: recipe for target 'all' failed
> make: *** [all] Error 2
> 
> 
> Error amazes me because sofia-sip library  from  portage ebuild
> installed without a problem:
> qlist -UCvq sofia-sip
> net-libs/sofia-sip-1.12.11 ssl static-libs
> http://data.gpo.zugaina.org/gentoo/net-libs/sofia-sip/
> 
> 
> The latest version of FreeSWITCH, I have been able to install without a
> problem are:
> FreeSWITCH version: 1.5.14b+git~20140725T212415Z~c411f8c7a9~64bit (git
> c411f8c 2014-07-25 21:24:15Z 64bit)
> 
> 
> Where can I find the source FreeSWITCH-1.5.x, or alternatively, how to
> improve this error?
> 
> 
> Cheers



From mike at jerris.com  Wed Dec 10 21:01:55 2014
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Dec 2014 13:01:55 -0500
Subject: [Freeswitch-users] Mod perl using $session->setVariable inline
In-Reply-To: <CAJR0VfH2dnUJ1C3Wuvq4PEOugqpwU3QYOCwbvrTnEvaR48BvxQ@mail.gmail.com>
References: <CAJR0VfFO0P_zqWAWiKY1h1dubzivG+a8pX54KguWcsTjjgzB3g@mail.gmail.com>
	<38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com>
	<CAJR0VfH2dnUJ1C3Wuvq4PEOugqpwU3QYOCwbvrTnEvaR48BvxQ@mail.gmail.com>
Message-ID: <E4D774C3-5C51-4144-8C7C-B9030FA64F2B@jerris.com>

That perl script will not run at all until the dialplan parsing is complete.  You would need to run the perl script inline (which I am not sure we allow) or run the perl script, then transfer to re-enter the dialplan after it is run.

> On Dec 10, 2014, at 12:56 PM, Liam Farr <liam at maxumdata.com> wrote:
> 
> Heres an expanded example;
> 
> 
> 
> 
> 
> Extension runs the perl script as part of the public dialplan first;
> 
> 
> 
> <extension name="run-script" continue="true">
> 
>    <condition>
> 
> 	  <action application="perl" data="/etc/freeswitch/scripts/my-script.pl <http://my-script.pl/>"/>
> 
>    </condition>
> 
> </extension>
> 
> 
> 
> 
> 
> Script sets some variables;
> 
> 
> 
> $my_value = "123";
> 
> $session->setVariable('some_var',$my_value);
> 
> 
> 
> 
> 
> Use that variable later in another extension in the public dialplan as a condition selector;
> 
> 
> 
> 
> 
> <extension name="bridge-call" >
> 
>    <condition field="context" expression="public" />
> 
>    <condition field="${some_var}" expression="123" >
> 
> 	   <action application="set" data="call_direction=inbound"/>
> 
> 	   <action application="bridge" data="sofia/internal/${destination_number}@1.2.3.4:5060 <http://destination_number}@1.2.3.4:5060>" />
> 
>    </condition>
> 
> </extension>
> 
> 
> 
> 
> 
> However this will fail, because the perl script "$session->setVariable" run earlier is not inline, so you cant make use of it in the condition (<condition field="${some_var}" expression="123" >) in the second extension that exists in a later part of the public dialplan. Hence I need a way to set inline variables from my perl script.
> 
> 
> 
> 
> 
> Cheers
> 
> 
> 
> Liam Farr
> 
> 
> 
> 
> On 11 December 2014 at 06:28, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
> If you are in a perl script using session->setVariable, how are you still in the dialplan?  This doesn't make sense to me.  There should be no need for an inline if you are already in the script.
> 
> > On Dec 10, 2014, at 4:20 AM, Liam Farr <liam at maxumdata.com <mailto:liam at maxumdata.com>> wrote:
> >
> > Hi,
> >
> > I need to set a variable using a perl script inline.
> >
> > Normally I would use $session->setVariable('some_var',$my_value);
> >
> > However I cant find any options to execute this inline / as an inline action. (I need to evaluate the variable later in the hunting phase of the dialplan hence need it to be an inline action).
> >
> > Is there a way to do this?
> >
> > Or is there another way to achieve the same result, e.g. hacking $session->execute("set","some_var=$my_value") to run as an inline action?
> >
> > Any help would be much appreciated, I've scoured the wiki, confluence and git repo and come up empty.
> >
> > --
> > Kind Regards
> >
> >
> > Liam Farr
> 

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From brian at freeswitch.org  Wed Dec 10 21:10:30 2014
From: brian at freeswitch.org (Brian West)
Date: Wed, 10 Dec 2014 12:10:30 -0600
Subject: [Freeswitch-users] Mod_sofia build error.
In-Reply-To: <422D5329-EA7F-4D72-8D2D-BD9EAF8D7F32@jerris.com>
References: <54888763.1020107@gmail.com>
	<422D5329-EA7F-4D72-8D2D-BD9EAF8D7F32@jerris.com>
Message-ID: <CAEJMVkB3vOTGWgkzuOp2xRA6Ei6pDxpm6787ZTCwQ3bsXEju8Q@mail.gmail.com>

Oh and file a JIRA! :P

On Wed, Dec 10, 2014 at 11:58 AM, Michael Jerris <mike at jerris.com> wrote:

> This could be related to bugs in gentoo headers related to sctp or due to
> issues in the autoconf detection for these.  We would be happy to take
> patches to correct this issue that don't break other distro/os if you can
> provide a patch.
>
>
> > On Dec 10, 2014, at 12:48 PM, Jacekalex <wampir990 at gmail.com> wrote:
> >
> >
> > Hi
> >
> > OS  Gentoo Linux x86_64
> > gcc (Gentoo Hardened 4.8.3 p1.1, pie-0.5.9) 4.8.3
> >
> > The previous system:
> > gcc (Gentoo Hardened 4.7.3-r1 p1.3, pie-0.5.5) 4.7.3
> >
> > Freeswitch versions - all 1.4.x
> >
> > Mod_sofia  - sofia-sip error:
> > ...
> > Making all in tport
> > mke[9]: Entering directory
> >
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
> >     LTake[8]: Entering directory
> >
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
> > make  all-am
> > maCOMPILE tport_type_sctp.lo
> > tport_type_sctp.c: In function ?tport_sctp_init_socket?:
> > tport_type_sctp.c:206:10: error: variable ?initmsg? has initializer but
> > incomplete type
> >   struct sctp_initmsg initmsg = { 0 };
> >          ^
> > tport_type_sctp.c:206:10: error: excess elements in struct initializer
> > [-Werror]
> > tport_type_sctp.c:206:10: error: (near initialization for ?initmsg?)
> > [-Werror]
> > tport_type_sctp.c:206:23: error: storage size of ?initmsg? isn?t known
> >   struct sctp_initmsg initmsg = { 0 };
> >                       ^
> > tport_type_sctp.c:211:36: error: ?SCTP_INITMSG? undeclared (first use in
> > this function)
> >   if (setsockopt(socket, SOL_SCTP, SCTP_INITMSG, &initmsg, sizeof
> > initmsg) < 0)
> >                                    ^
> > tport_type_sctp.c:211:36: note: each undeclared identifier is reported
> > only once for each function it appears in
> > tport_type_sctp.c:206:23: error: unused variable ?initmsg?
> > [-Werror=unused-variable]
> >   struct sctp_initmsg initmsg = { 0 };
> >                       ^
> > cc1: all warnings being treated as errors
> > Makefile:1452: recipe for target 'tport_type_sctp.lo' failed
> > make[9]: *** [tport_type_sctp.lo] Error 1
> > make[9]: Leaving directory
> >
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
> > Makefile:736: recipe for target 'all' failed
> > make[8]: *** [all] Error 2
> > make[8]: Leaving directory
> >
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/tport'
> > Makefile:547: recipe for target 'all-recursive' failed
> > make[7]: *** [all-recursive] Error 1
> > make[7]: Leaving directory
> >
> '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua'
> > Makefile:490: recipe for target 'all-recursive' failed
> > make[6]: *** [all-recursive] Error 1
> > make[6]: Leaving directory
> > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip'
> > Makefile:410: recipe for target 'all' failed
> > make[5]: *** [all] Error 2
> > make[5]: Leaving directory
> > '/home/fabryka/local/src/freeswitch/freeswitch481/libs/sofia-sip'
> > Makefile:906: recipe for target
> > '/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/
> libsofia-sip-ua.la'
> > failed
> > make[4]: ***
> > [/usr/local/src/freeswitch/freeswitch481/libs/sofia-sip/libsofia-sip-ua/
> libsofia-sip-ua.la]
> > Error 2
> > make[4]: Leaving directory
> >
> '/home/fabryka/local/src/freeswitch/freeswitch481/src/mod/endpoints/mod_sofia'
> > Makefile:577: recipe for target 'mod_sofia-all' failed
> > make[3]: *** [mod_sofia-all] Error 1
> > make[3]: Leaving directory
> > '/home/fabryka/local/src/freeswitch/freeswitch481/src/mod'
> > Makefile:491: recipe for target 'all-recursive' failed
> > make[2]: *** [all-recursive] Error 1
> > make[2]: Leaving directory
> > '/home/fabryka/local/src/freeswitch/freeswitch481/src'
> > Makefile:2487: recipe for target 'all-recursive' failed
> > make[1]: *** [all-recursive] Error 1
> > make[1]: Leaving directory
> > '/home/fabryka/local/src/freeswitch/freeswitch481'
> > Makefile:969: recipe for target 'all' failed
> > make: *** [all] Error 2
> >
> >
> > Error amazes me because sofia-sip library  from  portage ebuild
> > installed without a problem:
> > qlist -UCvq sofia-sip
> > net-libs/sofia-sip-1.12.11 ssl static-libs
> > http://data.gpo.zugaina.org/gentoo/net-libs/sofia-sip/
> >
> >
> > The latest version of FreeSWITCH, I have been able to install without a
> > problem are:
> > FreeSWITCH version: 1.5.14b+git~20140725T212415Z~c411f8c7a9~64bit (git
> > c411f8c 2014-07-25 21:24:15Z 64bit)
> >
> >
> > Where can I find the source FreeSWITCH-1.5.x, or alternatively, how to
> > improve this error?
> >
> >
> > 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




-- 

*Brian West*
brian at freeswitch.org


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

*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  Wed Dec 10 21:11:06 2014
From: aqsyounas at gmail.com (Aqs Younas)
Date: Wed, 10 Dec 2014 23:11:06 +0500
Subject: [Freeswitch-users] Core Dumped Assertion `channel != ((void *)0)'
	failed
Message-ID: <CAGVOjVXbL4mo_-u8YT01g901e=pCS97pEth+VeB4S=fzaA=XGA@mail.gmail.com>

Hi,All

I am new to freeswitch. Freeswitch crashed out after playing a url with
mod_shout.

In freeswitch logs i have this.

[parse.c:710] error: decode header failed before first valid one, going to
read again
freeswitch: src/switch_channel.c:909: switch_channel_get_variable_dup:
Assertion `channel != ((void *)0)' failed.

What can i do to prevent my freeswitch from crashing?

Any help would be much appreciated.

Thanks
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From mike at jerris.com  Wed Dec 10 21:14:14 2014
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Dec 2014 13:14:14 -0500
Subject: [Freeswitch-users] Core Dumped Assertion `channel != ((void
	*)0)' failed
In-Reply-To: <CAGVOjVXbL4mo_-u8YT01g901e=pCS97pEth+VeB4S=fzaA=XGA@mail.gmail.com>
References: <CAGVOjVXbL4mo_-u8YT01g901e=pCS97pEth+VeB4S=fzaA=XGA@mail.gmail.com>
Message-ID: <86D27DE1-128F-42CC-917E-02C02BD2A16B@jerris.com>

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

> On Dec 10, 2014, at 1:11 PM, Aqs Younas <aqsyounas at gmail.com> wrote:
> 
> Hi,All
> 
> I am new to freeswitch. Freeswitch crashed out after playing a url with mod_shout.
> 
> In freeswitch logs i have this.
> 
> [parse.c:710] error: decode header failed before first valid one, going to read again
> freeswitch: src/switch_channel.c:909: switch_channel_get_variable_dup: Assertion `channel != ((void *)0)' failed.
> 
> What can i do to prevent my freeswitch from crashing?
> 
> Any help would be much appreciated.

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From joelewhite at gmail.com  Wed Dec 10 22:27:50 2014
From: joelewhite at gmail.com (Joel White)
Date: Wed, 10 Dec 2014 14:27:50 -0500
Subject: [Freeswitch-users] Caller ID - Not Defined when executing dialplan,
 although Lua script is pulling CID from Database
Message-ID: <CAK0YwNkcb5HPjw_8mXezXDx3fEtnJcv8DLZL+Siv2kjq2a3zGw@mail.gmail.com>

I have gone over the config with a fine tooth comb, it matches another
config of a server in which the caller id works fine.  What I am seeing
however is that in this system the variable is not exported to the
dialplan.  I may be missing something, and most likely I am.  I do have a
question though.  Is there a way to see what variables are defined for a
particular user in the FreeSWITCH console?


Here is some output of the Lua script on the server not pushing caller id

2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from
gen_dir_user_xml.lua, generated XML:
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="voip.net">
      <user id="26342" mailbox="26342">
        <params>
          <param name="password" value="XXXXXX"/>
          <param name="vm-password" value="26342"/>
          <param name="vm-enabled" value="true"/>
          <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
    </params>
        <variables>
          <variable name="toll_allow" value="local,domestic"/>
          <variable name="user_context" value="default"/>
          <variable name="outbound_caller_id_name" value="Michelle"/>
          <variable name="outbound_caller_id_number" value="1321XXXXXXX"/>
      <variable name="accountcode" value="PA"/>
      <!--<variable name="callgroup" value="techsupport"/>-->
        </variables>
      </user>
    </domain>
  </section>
</document>


And some output from the Lua script on the server with CID functioning

2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from
gen_dir_user_xml.lua, generated XML:
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="sip.voice.com">
      <user id="53201" mailbox="53201">
        <params>
          <param name="password" value="XXXXXXX"/>
          <param name="vm-password" value="53201"/>
          <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
        </params>
        <variables>
          <variable name="toll_allow" value="local,domestic"/>
          <variable name="user_context" value="default"/>
          <variable name="outbound_caller_id_name" value="Voice.com"/>
          <variable name="outbound_caller_id_number" value="1303XXXXXX"/>
          <!--<variable name="callgroup" value="techsupport"/>-->
        </variables>
      </user>
    </domain>
  </section>
</document>


Of course I removed any identifiable information, but it looks like the CID
is being set.  What am I missing here that is not allowing for the variable
to be passed to the dialplan?


This is what I get when the dialplan executes

EXECUTE sofia/internal/26342 at voip.net
set(effective_caller_id_number=)
2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/
26342 at voip.net SET [effective_caller_id_number]=[UNDEF]


Kinda strange and I could not find a discrepancy between the dialplan
configuration of the working server vs the non-working server



Here is the version running on the server that works properly

FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git
b942d0f 2014-11-03 19:53:00Z 64bit)

And the version of the server having issue with CID

FreeSWITCH Version 1.4.13+git~20141103T195300Z~
b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)



I used diff and compared both servers conf directory recursively.  I could
not find a discrepancy in the files aside from Switch name, etc.


What am I missing?

Could there be anything that that would overwrite the CID variable after it
is set by Lua (Generating the user profile)?
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From ahabiba at gmail.com  Wed Dec 10 22:58:55 2014
From: ahabiba at gmail.com (Ahmed Habiba)
Date: Wed, 10 Dec 2014 22:58:55 +0300
Subject: [Freeswitch-users] SIP Header Manipulation
Message-ID: <29A45828-2427-492A-B18E-9260B4880745@gmail.com>

Dears,

kindly i have a case as below, now I need in case of B number is not registered to  modify the sip header ?TO? parameter and add prefix to the A Number, do you have any idea how can I do

[A Number] <???>[SIP GW]<????>[FreeSWitch]<???>[B Number]


Your help will be appreciated.


Thanks,

Ahmed Habiba.

From ahabiba at gmail.com  Wed Dec 10 23:02:38 2014
From: ahabiba at gmail.com (Ahmed Habiba)
Date: Wed, 10 Dec 2014 23:02:38 +0300
Subject: [Freeswitch-users] Twilio SIP trunk config - outbound call
	rejected
Message-ID: <46F54EC6-E3D0-42C0-9AC0-71B48CDE3FA7@gmail.com>

Mainly in Twilio if you are using sip trunk you should always call with international code prefix like you should put +1 before number for USA.

it looks like the number you are using in local format ? 5101112222 <sip:5101112222 at mycompany.pstn.twilio.com>?  it should "+xxx 510111   "



From: Adelia C. <hexade at hotmail.com <mailto:hexade at hotmail.com>>
To: "freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>" <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
Date: December 10, 2014 at 7:45:40 PM GMT+3
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org <mailto:freeswitch-users at lists.freeswitch.org>>
Subject: [Freeswitch-users] Twilio SIP trunk config - outbound call rejected



Any idea why this INVITE is rejected with 400 Invalid Phone Number by Twilio?
The first INVITE asks for Proxy Authentication, that seems to pass and this is rejected.
Thank you!

INVITE sip:5101112222 at mycompany.pstn.twilio.com <sip:5101112222 at mycompany.pstn.twilio.com> SIP/2.0
Via: SIP/2.0/UDP 10.50.1.116:5080;rport;branch=z9hG4bKa8QvU1B6QmeKc
Max-Forwards: 69
From: "unknown" <sip:4152221111 at mycompany.pstn.twilio.com <sip:4152221111 at mycompany.pstn.twilio.com>>;tag=6UetZr6rDZgHF
To: <sip:5101112222 at mycompany.pstn.twilio.com <sip:5101112222 at mycompany.pstn.twilio.com>>
Call-ID: 1ffda8b7-faaf-1232-8380-a9a65d1e96b1
CSeq: 68751485 INVITE
Contact: <sip:gw+twiliotrunk at 10.50.1.116:5080;transport=udp;gw=twiliotrunk <sip:gw+twiliotrunk at 10.50.1.116:5080;transport=udp;gw=twiliotrunk>>
Expires: 3600
User-Agent: FreeSWITCH-mod_sofia/1.5.12b~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Proxy-Authorization: Digest username="4152221111", realm="sip.twilio.com <http://sip.twilio.com/>", nonce="f722ae99d0a818bc422baa5d8", cnonce="IA1gJvqvEg6mmXR6WsQ", opaque="e89d2328e07da020d75a", algorithm=MD5, uri="sip:5101112222 at mycompany.pstn.twilio.com <sip:5101112222 at mycompany.pstn.twilio.com>", response="e65d3dea5b1e3b35306cfa121ec2ae4d", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 268
X-FS-Support: update_display,send_info
Remote-Party-ID: "unknown" <sip:1000 at mycompany.pstn.twilio.com <sip:1000 at mycompany.pstn.twilio.com>>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1418153731 1418153732 IN IP4 10.50.1.116
s=FreeSWITCH
c=IN IP4 10.50.1.116
t=0 0
m=audio 21366 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
--

SIP/2.0 400 Invalid phone number
To: <sip:5101112222 at ingenio.pstn.twilio.com <sip:5101112222 at ingenio.pstn.twilio.com>>;tag=08602519_6772d868_5eef74bb-957c-486e-b261-2a2e63704d73
Via: SIP/2.0/UDP 10.50.1.116:5080;received=10.50.1.116;rport=5080;branch=z9hG4bKa8QvU1B6QmeKc
CSeq: 68751485 INVITE
Call-ID: 1ffda8b7-faaf-1232-8380-a9a65d1e96b1
From: "unknown" <sip:4152221111 at ingenio.pstn.twilio.com <sip:4152221111 at ingenio.pstn.twilio.com>>;tag=6UetZr6rDZgHF
Contact: <sip:171.118.14.130:5060 <sip:171.118.14.130:5060>>
Content-Length: 0



--

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From ssinyagin at gmail.com  Thu Dec 11 03:47:10 2014
From: ssinyagin at gmail.com (Stanislav Sinyagin)
Date: Thu, 11 Dec 2014 01:47:10 +0100
Subject: [Freeswitch-users] Mod perl using $session->setVariable inline
In-Reply-To: <CAJR0VfH2dnUJ1C3Wuvq4PEOugqpwU3QYOCwbvrTnEvaR48BvxQ@mail.gmail.com>
References: <CAJR0VfFO0P_zqWAWiKY1h1dubzivG+a8pX54KguWcsTjjgzB3g@mail.gmail.com>
	<38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com>
	<CAJR0VfH2dnUJ1C3Wuvq4PEOugqpwU3QYOCwbvrTnEvaR48BvxQ@mail.gmail.com>
Message-ID: <CACOYK=iJBoJ+6RpbfdT-8pgZ9o2Y0q8ed6A43keOeK+3NnUrSQ@mail.gmail.com>

I'm using mod_perl quite intensively, and I do all the logic and
bridging from within the Perl script. It works fine, and you don't
really need to return the control back to the dialplan.

The only concern is, that if you do "bridge" from the script, your
perl script stays in memory and finishes only when the call ends. It
should be harmless for a modestly loaded server, and for a massive
service you would anyway do much better with an external script
talking via ESL.





On Wed, Dec 10, 2014 at 6:56 PM, Liam Farr <liam at maxumdata.com> wrote:
> Heres an expanded example;
>
>
>
> Extension runs the perl script as part of the public dialplan first;
>
>
> <extension name="run-script" continue="true">
>
>    <condition>
>
>  <action application="perl" data="/etc/freeswitch/scripts/my-script.pl"/>
>
>    </condition>
>
> </extension>
>
>
>
> Script sets some variables;
>
>
> $my_value = "123";
>
> $session->setVariable('some_var',$my_value);
>
>
>
> Use that variable later in another extension in the public dialplan as a
> condition selector;
>
>
>
> <extension name="bridge-call" >
>
>    <condition field="context" expression="public" />
>
>    <condition field="${some_var}" expression="123" >
>
>   <action application="set" data="call_direction=inbound"/>
>
>   <action application="bridge"
> data="sofia/internal/${destination_number}@1.2.3.4:5060" />
>
>    </condition>
>
> </extension>
>
>
>
> However this will fail, because the perl script "$session->setVariable" run
> earlier is not inline, so you cant make use of it in the condition
> (<condition field="${some_var}" expression="123" >) in the second extension
> that exists in a later part of the public dialplan. Hence I need a way to
> set inline variables from my perl script.
>
>
>
> Cheers
>
>
> Liam Farr
>
>
>
> On 11 December 2014 at 06:28, Michael Jerris <mike at jerris.com> wrote:
>>
>> If you are in a perl script using session->setVariable, how are you still
>> in the dialplan?  This doesn't make sense to me.  There should be no need
>> for an inline if you are already in the script.
>>
>> > On Dec 10, 2014, at 4:20 AM, Liam Farr <liam at maxumdata.com> wrote:
>> >
>> > Hi,
>> >
>> > I need to set a variable using a perl script inline.
>> >
>> > Normally I would use $session->setVariable('some_var',$my_value);
>> >
>> > However I cant find any options to execute this inline / as an inline
>> > action. (I need to evaluate the variable later in the hunting phase of the
>> > dialplan hence need it to be an inline action).
>> >
>> > Is there a way to do this?
>> >
>> > Or is there another way to achieve the same result, e.g. hacking
>> > $session->execute("set","some_var=$my_value") to run as an inline action?
>> >
>> > Any help would be much appreciated, I've scoured the wiki, confluence
>> > and git repo and come up empty.
>> >
>> > --
>> > Kind Regards
>> >
>> >
>> > Liam Farr
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Dec 11 04:27:57 2014
From: mike at jerris.com (Michael Jerris)
Date: Wed, 10 Dec 2014 20:27:57 -0500
Subject: [Freeswitch-users] Mod perl using $session->setVariable inline
In-Reply-To: <CACOYK=iJBoJ+6RpbfdT-8pgZ9o2Y0q8ed6A43keOeK+3NnUrSQ@mail.gmail.com>
References: <CAJR0VfFO0P_zqWAWiKY1h1dubzivG+a8pX54KguWcsTjjgzB3g@mail.gmail.com>
	<38660852-0B8E-43DD-929A-0D3C8CB2DD5B@jerris.com>
	<CAJR0VfH2dnUJ1C3Wuvq4PEOugqpwU3QYOCwbvrTnEvaR48BvxQ@mail.gmail.com>
	<CACOYK=iJBoJ+6RpbfdT-8pgZ9o2Y0q8ed6A43keOeK+3NnUrSQ@mail.gmail.com>
Message-ID: <2A1755A5-FD0D-45A7-B67E-C2C5F1AC5E29@jerris.com>

The issue here is he is expecting the value to be set later in the dialplan. That won't work as he's doing.  Another option for both of you is to handle the logic in perl, then set vars from perl for what to bridge to, then have a bridge line in the dp fter perl that uses those vars (not a condition).

> On Dec 10, 2014, at 7:47 PM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> 
> I'm using mod_perl quite intensively, and I do all the logic and
> bridging from within the Perl script. It works fine, and you don't
> really need to return the control back to the dialplan.
> 
> The only concern is, that if you do "bridge" from the script, your
> perl script stays in memory and finishes only when the call ends. It
> should be harmless for a modestly loaded server, and for a massive
> service you would anyway do much better with an external script
> talking via ESL.
> 
> 
> 
> 
> 
> On Wed, Dec 10, 2014 at 6:56 PM, Liam Farr <liam at maxumdata.com> wrote:
>> Heres an expanded example;
>> 
>> 
>> 
>> Extension runs the perl script as part of the public dialplan first;
>> 
>> 
>> <extension name="run-script" continue="true">
>> 
>>   <condition>
>> 
>> <action application="perl" data="/etc/freeswitch/scripts/my-script.pl"/>
>> 
>>   </condition>
>> 
>> </extension>
>> 
>> 
>> 
>> Script sets some variables;
>> 
>> 
>> $my_value = "123";
>> 
>> $session->setVariable('some_var',$my_value);
>> 
>> 
>> 
>> Use that variable later in another extension in the public dialplan as a
>> condition selector;
>> 
>> 
>> 
>> <extension name="bridge-call" >
>> 
>>   <condition field="context" expression="public" />
>> 
>>   <condition field="${some_var}" expression="123" >
>> 
>>  <action application="set" data="call_direction=inbound"/>
>> 
>>  <action application="bridge"
>> data="sofia/internal/${destination_number}@1.2.3.4:5060" />
>> 
>>   </condition>
>> 
>> </extension>
>> 
>> 
>> 
>> However this will fail, because the perl script "$session->setVariable" run
>> earlier is not inline, so you cant make use of it in the condition
>> (<condition field="${some_var}" expression="123" >) in the second extension
>> that exists in a later part of the public dialplan. Hence I need a way to
>> set inline variables from my perl script.
>> 
>> 
>> 
>> Cheers
>> 
>> 
>> Liam Farr
>> 
>> 
>> 
>> On 11 December 2014 at 06:28, Michael Jerris <mike at jerris.com> wrote:
>>> 
>>> If you are in a perl script using session->setVariable, how are you still
>>> in the dialplan?  This doesn't make sense to me.  There should be no need
>>> for an inline if you are already in the script.
>>> 
>>>> On Dec 10, 2014, at 4:20 AM, Liam Farr <liam at maxumdata.com> wrote:
>>>> 
>>>> Hi,
>>>> 
>>>> I need to set a variable using a perl script inline.
>>>> 
>>>> Normally I would use $session->setVariable('some_var',$my_value);
>>>> 
>>>> However I cant find any options to execute this inline / as an inline
>>>> action. (I need to evaluate the variable later in the hunting phase of the
>>>> dialplan hence need it to be an inline action).
>>>> 
>>>> Is there a way to do this?
>>>> 
>>>> Or is there another way to achieve the same result, e.g. hacking
>>>> $session->execute("set","some_var=$my_value") to run as an inline action?
>>>> 
>>>> Any help would be much appreciated, I've scoured the wiki, confluence
>>>> and git repo and come up empty.
>>>> 
>>>> --
>>>> Kind Regards
>>>> 
>>>> 
>>>> Liam Farr
>>> 



From andrew.keil at visytel.com  Thu Dec 11 07:11:15 2014
From: andrew.keil at visytel.com (Andrew Keil)
Date: Thu, 11 Dec 2014 04:11:15 +0000
Subject: [Freeswitch-users] Re - FreeSWITCH Lua CUSTOM events
 interrupting session:streamFile playback - is this possible?
In-Reply-To: <FBDD49CEBFC34FB8AA95229D9EF76562@gmail.com>
References: <HKXPR04MB0456E779E02E79F7E5AB30B3F3640@HKXPR04MB0456.apcprd04.prod.outlook.com>
	<FBDD49CEBFC34FB8AA95229D9EF76562@gmail.com>
Message-ID: <HKNPR04MB04494893C9A325706DE3367CF3630@HKNPR04MB0449.apcprd04.prod.outlook.com>

Seven Du,

Thanks for your response.  I eventually found a way by sending via the Event Socket Layer:

sendmsg <channel uuid>
call-command: execute
execute-app-name: break

Then sending the CUSTOM message and using the con:pop(1,5000) approach to pop the CUSTOM message of the event queue.

This approach solves the problem, although I wish I could send an event via ESL that will trigger the setInputCallback(?) function.  However I have not found a way to do that as yet.

Andrew

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du
Sent: Tuesday, 9 December 2014 10:16 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Re - FreeSWITCH Lua CUSTOM events interrupting session:streamFile playback - is this possible?

I believe the playback_terminitors chan var works, or you could try set an input callback and return ?break? when dtmf detected.

--
Seven Du
http://about.me/dujinfang
http://www.dujinfang.com
http://www.freeswitch.org.cn

Sent with Sparrow<http://www.sparrowmailapp.com/?sig>


On Tuesday, December 9, 2014 at 7:00 AM, Andrew Keil wrote:

To FreeSWITCH users,



After progressing with Lua inside FreeSWITCH I have a question that I cannot source an answer for within the current FreeSWITCH documentation.



I have a requirement for an external application to process an event (via the event socket layer) then return the results back to the application while an audio file is played back in a loop to the caller.



From the current documentation this method allows for the event to be sent then the return event is ?consumed? using a polling approach.  However the caller is listening to silence.



function poll()

    -- create event and listener

    local event = freeswitch.Event("CUSTOM", "ping::running?")

    local con = freeswitch.EventConsumer("CUSTOM", "ping::running!")



    -- add text ad libitum

    event:addHeader("hi", "there")

    -- fire event

    event:fire()

    -- and wait for reply but not very long

                local retevent = con:pop(1, 5000)

    if retevent then

        print("reply received")

                                freeswitch.consoleLog("DEBUG",  string.format("reply received: %s\n",retevent:getHeader("Result")))

        return true

    end

    print("no reply")

                freeswitch.consoleLog("DEBUG", "no reply\n")

    return false

end



Some questions:

Is there a way to playback audio (eg. session:streamFile(?)) while this takes place, since currently session:streamFile(?) seems to be a blocking function (ie. Finishes when the audio file is played back completely)?

Is there a way to use session:setInputCallback(?) to handle an external CUSTOM event being returned (since this would then be able to interrupt the session:streamFile(?) just like it does for DTMF or speech recognition)?



If you have a preferred approach to solving this then I am open to your suggestions.



I appreciate any assistance that you can give.



Kind Regards,



Andrew Keil


_________________________________________________________________________
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 akhilgarg7 at gmail.com  Thu Dec 11 12:00:09 2014
From: akhilgarg7 at gmail.com (akhil garg)
Date: Thu, 11 Dec 2014 14:30:09 +0530
Subject: [Freeswitch-users] fs_cli hangs
Message-ID: <CA+mgNr5S9_L6LcfqUwYhMfFJOpRxn3uDCitM_cwNQ7h7LuODhw@mail.gmail.com>

running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but no
success.


------------------------------------------------------------------------------------------------------------------------------------------------
OUTPUT 1:
------------------------------------------------------------------------------------------------------------------------------------------------
[DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
/root/.fs_cli_conf.
[DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
/etc/fs_cli.conf.
[DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
profile
[DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
------------------------------------------------------------------------------------------------------------------------------------------------




------------------------------------------------------------------------------------------------------------------------------------------------
OUTPUT 2:
------------------------------------------------------------------------------------------------------------------------------------------------
[DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
/root/.fs_cli_conf.
[DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
/etc/fs_cli.conf.
[DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
profile
[DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
[DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
[auth/request]
[DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: auth/request


[DEBUG] esl.c:1465 esl_send() SEND
auth ClueCon
------------------------------------------------------------------------------------------------------------------------------------------------




------------------------------------------------------------------------------------------------------------------------------------------------
OUTPUT 3:
------------------------------------------------------------------------------------------------------------------------------------------------
[DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
/root/.fs_cli_conf.
[DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
/etc/fs_cli.conf.
[DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
profile
[DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
[DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
[auth/request]
[DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: auth/request


[DEBUG] esl.c:1465 esl_send() SEND
auth ClueCon


[DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
[command/reply]
[DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK
accepted]
[DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: +OK accepted


[DEBUG] esl.c:1465 esl_send() SEND
log
------------------------------------------------------------------------------------------------------------------------------------------------



-- 
regards,
akhil
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From mbsip at gazeta.pl  Thu Dec 11 16:45:17 2014
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Thu, 11 Dec 2014 14:45:17 +0100
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call request
Message-ID: <CACynCY7wFKvxkBOaQ7xyvHyCreMGOA1ixn1k=cRWt_rQjBKwRg@mail.gmail.com>

Hello,

I am running FreeSWITCH Version
1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
problem with some call requests setup.
Here is one of them:

2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
sofia/outside_1/20049112223344 []
2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
(sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send signal
sofia/outside_1/20049112223344 [BREAK]
2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
(sofia/outside_1/20049112223344) Running State Change CS_INIT
2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
(sofia/outside_1/20049112223344) State INIT
2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
sofia/outside_1/20049112223344 SOFIA INIT
2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
sofia/outside_1/20049112223344 Patched SDP
 ---
 v=0
 o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
 s=SIP Call
 c=IN IP4 10.10.10.12
 t=0 0
 m=audio 24782 RTP/AVP 18 19
 c=IN IP4 10.10.10.12
 a=rtpmap:18 G729/8000
 a=fmtp:18 annexb=yes
 a=rtpmap:19 CN/8000

 +++
 v=0
 o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
 s=FreeSWITCH
 c=IN IP4 10.10.10.166
 t=0 0
 m=audio 19690 RTP/AVP 18 19
 c=IN IP4 10.10.10.166
 a=rtpmap:18 G729/8000
 a=fmtp:18 annexb=yes
 a=rtpmap:19 CN/8000

 2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
2014-09-17 23:11:20Z 64bit
 Local SDP:
 v=0
 o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
 s=FreeSWITCH
 c=IN IP4 10.10.10.166
 t=0 0
 m=audio 19690 RTP/AVP 18 19
 c=IN IP4 10.10.10.166
 a=rtpmap:18 G729/8000
 a=fmtp:18 annexb=yes
 a=rtpmap:19 CN/8000

sending out SDP exactly as shown above.
The other side is responsing with 100, then 180 Ringing wo/SDP then 183
w/SDP

 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
sofia/outside_1/20049112223344 entering state [proceeding][183]
 2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
 v=0
 o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
 s=Dialogic-SIP
 c=IN IP4 10.10.10.198
 t=0 0
 m=audio 10024 RTP/AVP 19 18
 a=rtpmap:19 CN/8000
 a=rtpmap:18 G729/8000
 a=fmtp:18 annexb=yes
 a=silenceSupp:off - - - -
 a=ptime:20

 2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
sofia/outside_1/20049112223344 Patched SDP
 ---
 v=0
 o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
 s=FreeSWITCH
 c=IN IP4 10.10.10.166
 t=0 0
 m=audio 29492 RTP/AVP 18 19
 c=IN IP4 10.10.10.166
 a=rtpmap:18 G729/8000
 a=fmtp:18 annexb=yes
 a=rtpmap:19 CN/8000

 +++
 v=0
 o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
 s=FreeSWITCH
 c=IN IP4 10.10.10.166
 t=0 0
 m=audio 29492 RTP/AVP 18 19
 c=IN IP4 10.10.10.166
 a=rtpmap:18 G729/8000
 a=fmtp:18 annexb=yes
 a=rtpmap:19 CN/8000

2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 *Invalid codec CN!*
2014-12-11 14:29:12.214608 [ERR] switch_core_media.c:2294 *Can't load
codec?*
2014-12-11 14:29:12.214608 [NOTICE] switch_core_media.c:2295 Hangup
sofia/outside_1/20049112223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
sofia/outside_1/20049112223344 [KILL]

and Freeswitch is terminating the call by using CANCEL.

My modules.conf.xml config part looks like following:
    <!-- Codec Interfaces -->
    <load module="mod_spandsp"/>
    <load module="mod_g723_1"/>
    <load module="mod_g729"/>
    <!--load module="mod_amr"/-->
    <!--<load module="mod_ilbc"/>-->
    <load module="mod_h26x"/>
    <!--load module="mod_vp8"/-->
    <!--load module="mod_b64"/-->
    <!--<load module="mod_siren"/>-->
    <!--<load module="mod_isac"/>-->
    <!--<load module="mod_celt"/>-->
    <!--load module="mod_opus"/-->

Is it a problem with CN definition?
Could you please lead me where the problem is located?

Thanks in advance.
Mac.
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From blasterjr at gmail.com  Thu Dec 11 18:26:02 2014
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Thu, 11 Dec 2014 08:26:02 -0700
Subject: [Freeswitch-users] Caller ID - Not Defined when executing
 dialplan, although Lua script is pulling CID from Database
In-Reply-To: <CAK0YwNkcb5HPjw_8mXezXDx3fEtnJcv8DLZL+Siv2kjq2a3zGw@mail.gmail.com>
References: <CAK0YwNkcb5HPjw_8mXezXDx3fEtnJcv8DLZL+Siv2kjq2a3zGw@mail.gmail.com>
Message-ID: <CAO3hyzGa6CzOkyH=Fthdh2RB-4cYwoMnE83mWQJ-97NGHLf+6A@mail.gmail.com>

The variable outbound_caller_id_number and outbound_caller_id_name are not
related to the caller id on outbound calls.

On your outbound dial plan you need to set something like the following

      <action application="export"
data="effective_caller_id_name=${user_data(${username}@${domain_name} var
outbound_caller_id_name)}"/>
      <action application="export"
data="effective_caller_id_number=${user_data(${username}@${domain_name} var
outbound_caller_id_number)}"/>

This will cause the system to pull the settings from the users profile and
use it for the outgoing call.

On Wed, Dec 10, 2014 at 12:27 PM, Joel White <joelewhite at gmail.com> wrote:

> I have gone over the config with a fine tooth comb, it matches another
> config of a server in which the caller id works fine.  What I am seeing
> however is that in this system the variable is not exported to the
> dialplan.  I may be missing something, and most likely I am.  I do have a
> question though.  Is there a way to see what variables are defined for a
> particular user in the FreeSWITCH console?
>
>
> Here is some output of the Lua script on the server not pushing caller id
>
> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from
> gen_dir_user_xml.lua, generated XML:
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="voip.net">
>       <user id="26342" mailbox="26342">
>         <params>
>           <param name="password" value="XXXXXX"/>
>           <param name="vm-password" value="26342"/>
>           <param name="vm-enabled" value="true"/>
>           <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>     </params>
>         <variables>
>           <variable name="toll_allow" value="local,domestic"/>
>           <variable name="user_context" value="default"/>
>           <variable name="outbound_caller_id_name" value="Michelle"/>
>           <variable name="outbound_caller_id_number" value="1321XXXXXXX"/>
>       <variable name="accountcode" value="PA"/>
>       <!--<variable name="callgroup" value="techsupport"/>-->
>         </variables>
>       </user>
>     </domain>
>   </section>
> </document>
>
>
> And some output from the Lua script on the server with CID functioning
>
> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from
> gen_dir_user_xml.lua, generated XML:
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="sip.voice.com">
>       <user id="53201" mailbox="53201">
>         <params>
>           <param name="password" value="XXXXXXX"/>
>           <param name="vm-password" value="53201"/>
>           <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>         </params>
>         <variables>
>           <variable name="toll_allow" value="local,domestic"/>
>           <variable name="user_context" value="default"/>
>           <variable name="outbound_caller_id_name" value="Voice.com"/>
>           <variable name="outbound_caller_id_number" value="1303XXXXXX"/>
>           <!--<variable name="callgroup" value="techsupport"/>-->
>         </variables>
>       </user>
>     </domain>
>   </section>
> </document>
>
>
> Of course I removed any identifiable information, but it looks like the
> CID is being set.  What am I missing here that is not allowing for the
> variable to be passed to the dialplan?
>
>
> This is what I get when the dialplan executes
>
> EXECUTE sofia/internal/26342 at voip.net
> set(effective_caller_id_number=)
> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/
> 26342 at voip.net SET [effective_caller_id_number]=[UNDEF]
>
>
> Kinda strange and I could not find a discrepancy between the dialplan
> configuration of the working server vs the non-working server
>
>
>
> Here is the version running on the server that works properly
>
> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git
> b942d0f 2014-11-03 19:53:00Z 64bit)
>
> And the version of the server having issue with CID
>
> FreeSWITCH Version 1.4.13+git~20141103T195300Z~
> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)
>
>
>
> I used diff and compared both servers conf directory recursively.  I could
> not find a discrepancy in the files aside from Switch name, etc.
>
>
> What am I missing?
>
> Could there be anything that that would overwrite the CID variable after
> it is set by Lua (Generating the user profile)?
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 11 18:32:52 2014
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Dec 2014 10:32:52 -0500
Subject: [Freeswitch-users] Caller ID - Not Defined when executing
	dialplan, although Lua script is pulling CID from Database
In-Reply-To: <CAO3hyzGa6CzOkyH=Fthdh2RB-4cYwoMnE83mWQJ-97NGHLf+6A@mail.gmail.com>
References: <CAK0YwNkcb5HPjw_8mXezXDx3fEtnJcv8DLZL+Siv2kjq2a3zGw@mail.gmail.com>
	<CAO3hyzGa6CzOkyH=Fthdh2RB-4cYwoMnE83mWQJ-97NGHLf+6A@mail.gmail.com>
Message-ID: <C09C7BA3-86EC-467B-A993-1BC4653A278F@jerris.com>

you should be looking at the origination_caller_id_* vars, not effective

> On Dec 11, 2014, at 10:26 AM, Chris Tunbridge <blasterjr at gmail.com> wrote:
> 
> The variable outbound_caller_id_number and outbound_caller_id_name are not related to the caller id on outbound calls.  
> 
> On your outbound dial plan you need to set something like the following
> 
>       <action application="export" data="effective_caller_id_name=${user_data(${username}@${domain_name} var outbound_caller_id_name)}"/>
>       <action application="export" data="effective_caller_id_number=${user_data(${username}@${domain_name} var outbound_caller_id_number)}"/>
> 
> This will cause the system to pull the settings from the users profile and use it for the outgoing call.
> 
> On Wed, Dec 10, 2014 at 12:27 PM, Joel White <joelewhite at gmail.com <mailto:joelewhite at gmail.com>> wrote:
> I have gone over the config with a fine tooth comb, it matches another config of a server in which the caller id works fine.  What I am seeing however is that in this system the variable is not exported to the dialplan.  I may be missing something, and most likely I am.  I do have a question though.  Is there a way to see what variables are defined for a particular user in the FreeSWITCH console?
> 
> 
> Here is some output of the Lua script on the server not pushing caller id
> 
> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML:
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="voip.net <http://voip.net/>">
>       <user id="26342" mailbox="26342">
>         <params>
>           <param name="password" value="XXXXXX"/>
>           <param name="vm-password" value="26342"/>
>           <param name="vm-enabled" value="true"/>
>           <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>     </params>
>         <variables>
>           <variable name="toll_allow" value="local,domestic"/>
>           <variable name="user_context" value="default"/>
>           <variable name="outbound_caller_id_name" value="Michelle"/>
>           <variable name="outbound_caller_id_number" value="1321XXXXXXX"/>
>       <variable name="accountcode" value="PA"/>
>       <!--<variable name="callgroup" value="techsupport"/>-->
>         </variables>
>       </user>
>     </domain>
>   </section>
> </document>
> 
> 
> And some output from the Lua script on the server with CID functioning
> 
> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML:
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="sip.voice.com <http://sip.voice.com/>">
>       <user id="53201" mailbox="53201">
>         <params>
>           <param name="password" value="XXXXXXX"/>
>           <param name="vm-password" value="53201"/>
>           <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>         </params>
>         <variables>
>           <variable name="toll_allow" value="local,domestic"/>
>           <variable name="user_context" value="default"/>
>           <variable name="outbound_caller_id_name" value="Voice.com"/>
>           <variable name="outbound_caller_id_number" value="1303XXXXXX"/>
>           <!--<variable name="callgroup" value="techsupport"/>-->
>         </variables>
>       </user>
>     </domain>
>   </section>
> </document>
> 
> 
> Of course I removed any identifiable information, but it looks like the CID is being set.  What am I missing here that is not allowing for the variable to be passed to the dialplan?
> 
> 
> This is what I get when the dialplan executes
> 
> EXECUTE sofia/internal/26342 at voip.net <mailto:26342 at voip.net> 
> set(effective_caller_id_number=)
> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/26342 at voip.net <mailto:26342 at voip.net> SET [effective_caller_id_number]=[UNDEF]
> 
> 
> Kinda strange and I could not find a discrepancy between the dialplan configuration of the working server vs the non-working server
> 
> 
> 
> Here is the version running on the server that works properly
> 
> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)
> 
> And the version of the server having issue with CID
> 
> FreeSWITCH Version 1.4.13+git~20141103T195300Z~
> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)
> 
> 
> 
> I used diff and compared both servers conf directory recursively.  I could not find a discrepancy in the files aside from Switch name, etc.
> 
> 
> What am I missing?
> 
> Could there be anything that that would overwrite the CID variable after it is set by Lua (Generating the user profile)?
> 

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From mike at jerris.com  Thu Dec 11 18:34:48 2014
From: mike at jerris.com (Michael Jerris)
Date: Thu, 11 Dec 2014 10:34:48 -0500
Subject: [Freeswitch-users] Caller ID - Not Defined when executing
	dialplan, although Lua script is pulling CID from Database
In-Reply-To: <CAO3hyzGa6CzOkyH=Fthdh2RB-4cYwoMnE83mWQJ-97NGHLf+6A@mail.gmail.com>
References: <CAK0YwNkcb5HPjw_8mXezXDx3fEtnJcv8DLZL+Siv2kjq2a3zGw@mail.gmail.com>
	<CAO3hyzGa6CzOkyH=Fthdh2RB-4cYwoMnE83mWQJ-97NGHLf+6A@mail.gmail.com>
Message-ID: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com>

you should be looking at the origination_caller_id_* vars, not effective

> On Dec 11, 2014, at 10:26 AM, Chris Tunbridge <blasterjr at gmail.com <mailto:blasterjr at gmail.com>> wrote:
> 
> The variable outbound_caller_id_number and outbound_caller_id_name are not related to the caller id on outbound calls.  
> 
> On your outbound dial plan you need to set something like the following
> 
>       <action application="export" data="effective_caller_id_name=${user_data(${username}@${domain_name} var outbound_caller_id_name)}"/>
>       <action application="export" data="effective_caller_id_number=${user_data(${username}@${domain_name} var outbound_caller_id_number)}"/>
> 
> This will cause the system to pull the settings from the users profile and use it for the outgoing call.
> 
> On Wed, Dec 10, 2014 at 12:27 PM, Joel White <joelewhite at gmail.com <mailto:joelewhite at gmail.com>> wrote:
> I have gone over the config with a fine tooth comb, it matches another config of a server in which the caller id works fine.  What I am seeing however is that in this system the variable is not exported to the dialplan.  I may be missing something, and most likely I am.  I do have a question though.  Is there a way to see what variables are defined for a particular user in the FreeSWITCH console?
> 
> 
> Here is some output of the Lua script on the server not pushing caller id
> 
> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML:
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="voip.net <http://voip.net/>">
>       <user id="26342" mailbox="26342">
>         <params>
>           <param name="password" value="XXXXXX"/>
>           <param name="vm-password" value="26342"/>
>           <param name="vm-enabled" value="true"/>
>           <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>     </params>
>         <variables>
>           <variable name="toll_allow" value="local,domestic"/>
>           <variable name="user_context" value="default"/>
>           <variable name="outbound_caller_id_name" value="Michelle"/>
>           <variable name="outbound_caller_id_number" value="1321XXXXXXX"/>
>       <variable name="accountcode" value="PA"/>
>       <!--<variable name="callgroup" value="techsupport"/>-->
>         </variables>
>       </user>
>     </domain>
>   </section>
> </document>
> 
> 
> And some output from the Lua script on the server with CID functioning
> 
> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from gen_dir_user_xml.lua, generated XML:
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="sip.voice.com <http://sip.voice.com/>">
>       <user id="53201" mailbox="53201">
>         <params>
>           <param name="password" value="XXXXXXX"/>
>           <param name="vm-password" value="53201"/>
>           <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>         </params>
>         <variables>
>           <variable name="toll_allow" value="local,domestic"/>
>           <variable name="user_context" value="default"/>
>           <variable name="outbound_caller_id_name" value="Voice.com <http://voice.com/>"/>
>           <variable name="outbound_caller_id_number" value="1303XXXXXX"/>
>           <!--<variable name="callgroup" value="techsupport"/>-->
>         </variables>
>       </user>
>     </domain>
>   </section>
> </document>
> 
> 
> Of course I removed any identifiable information, but it looks like the CID is being set.  What am I missing here that is not allowing for the variable to be passed to the dialplan?
> 
> 
> This is what I get when the dialplan executes
> 
> EXECUTE sofia/internal/26342 at voip.net <mailto:26342 at voip.net> 
> set(effective_caller_id_number=)
> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/26342 at voip.net <mailto:26342 at voip.net> SET [effective_caller_id_number]=[UNDEF]
> 
> 
> Kinda strange and I could not find a discrepancy between the dialplan configuration of the working server vs the non-working server
> 
> 
> 
> Here is the version running on the server that works properly
> 
> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)
> 
> And the version of the server having issue with CID
> 
> FreeSWITCH Version 1.4.13+git~20141103T195300Z~
> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)
> 
> 
> 
> I used diff and compared both servers conf directory recursively.  I could not find a discrepancy in the files aside from Switch name, etc.
> 
> 
> What am I missing?
> 
> Could there be anything that that would overwrite the CID variable after it is set by Lua (Generating the user profile)?
> 

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From steveayre at gmail.com  Thu Dec 11 18:53:36 2014
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 11 Dec 2014 15:53:36 +0000
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call
	request
In-Reply-To: <CACynCY7wFKvxkBOaQ7xyvHyCreMGOA1ixn1k=cRWt_rQjBKwRg@mail.gmail.com>
References: <CACynCY7wFKvxkBOaQ7xyvHyCreMGOA1ixn1k=cRWt_rQjBKwRg@mail.gmail.com>
Message-ID: <CAFiqYum79cR0Tfa-Guo+R+3YruNUTm_jeZAsf4WsL7SZMB6E=w@mail.gmail.com>

Your version is 3 months old. Can you reproduce this on the latest git head?

On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>
> Hello,
>
> I am running FreeSWITCH Version
> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
> problem with some call requests setup.
> Here is one of them:
>
> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
> sofia/outside_1/20049112223344 []
> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send signal
> sofia/outside_1/20049112223344 [BREAK]
> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
> (sofia/outside_1/20049112223344) Running State Change CS_INIT
> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
> (sofia/outside_1/20049112223344) State INIT
> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
> sofia/outside_1/20049112223344 SOFIA INIT
> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
> sofia/outside_1/20049112223344 Patched SDP
>  ---
>  v=0
>  o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>  s=SIP Call
>  c=IN IP4 10.10.10.12
>  t=0 0
>  m=audio 24782 RTP/AVP 18 19
>  c=IN IP4 10.10.10.12
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=rtpmap:19 CN/8000
>
>  +++
>  v=0
>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>  s=FreeSWITCH
>  c=IN IP4 10.10.10.166
>  t=0 0
>  m=audio 19690 RTP/AVP 18 19
>  c=IN IP4 10.10.10.166
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=rtpmap:19 CN/8000
>
>  2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
> 2014-09-17 23:11:20Z 64bit
>  Local SDP:
>  v=0
>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>  s=FreeSWITCH
>  c=IN IP4 10.10.10.166
>  t=0 0
>  m=audio 19690 RTP/AVP 18 19
>  c=IN IP4 10.10.10.166
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=rtpmap:19 CN/8000
>
> sending out SDP exactly as shown above.
> The other side is responsing with 100, then 180 Ringing wo/SDP then 183
> w/SDP
>
>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
> sofia/outside_1/20049112223344 entering state [proceeding][183]
>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>  v=0
>  o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>  s=Dialogic-SIP
>  c=IN IP4 10.10.10.198
>  t=0 0
>  m=audio 10024 RTP/AVP 19 18
>  a=rtpmap:19 CN/8000
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=silenceSupp:off - - - -
>  a=ptime:20
>
>  2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
> sofia/outside_1/20049112223344 Patched SDP
>  ---
>  v=0
>  o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>  s=FreeSWITCH
>  c=IN IP4 10.10.10.166
>  t=0 0
>  m=audio 29492 RTP/AVP 18 19
>  c=IN IP4 10.10.10.166
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=rtpmap:19 CN/8000
>
>  +++
>  v=0
>  o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>  s=FreeSWITCH
>  c=IN IP4 10.10.10.166
>  t=0 0
>  m=audio 29492 RTP/AVP 18 19
>  c=IN IP4 10.10.10.166
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=rtpmap:19 CN/8000
>
> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 *Invalid codec
> CN!*
> 2014-12-11 14:29:12.214608 [ERR] switch_core_media.c:2294 *Can't load
> codec?*
> 2014-12-11 14:29:12.214608 [NOTICE] switch_core_media.c:2295 Hangup
> sofia/outside_1/20049112223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
> sofia/outside_1/20049112223344 [KILL]
>
> and Freeswitch is terminating the call by using CANCEL.
>
> My modules.conf.xml config part looks like following:
>     <!-- Codec Interfaces -->
>     <load module="mod_spandsp"/>
>     <load module="mod_g723_1"/>
>     <load module="mod_g729"/>
>     <!--load module="mod_amr"/-->
>     <!--<load module="mod_ilbc"/>-->
>     <load module="mod_h26x"/>
>     <!--load module="mod_vp8"/-->
>     <!--load module="mod_b64"/-->
>     <!--<load module="mod_siren"/>-->
>     <!--<load module="mod_isac"/>-->
>     <!--<load module="mod_celt"/>-->
>     <!--load module="mod_opus"/-->
>
> Is it a problem with CN definition?
> Could you please lead me where the problem is located?
>
> Thanks in advance.
> Mac.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 11 19:12:00 2014
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 11 Dec 2014 10:12:00 -0600
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call
 request
In-Reply-To: <CAFiqYum79cR0Tfa-Guo+R+3YruNUTm_jeZAsf4WsL7SZMB6E=w@mail.gmail.com>
Message-ID: <D0AF1E70.8F750%krice@freeswitch.org>

CN is comfort noise... Your cisco is sending it, and freeswitch is passing
it thru because you have told it to use proxy media mode...


On 12/11/14 9:53 AM, "Steven Ayre" <steveayre at gmail.com> wrote:

> Your version is 3 months old. Can you reproduce this on the latest git head?
> 
> On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>> Hello,
>> 
>> I am running?FreeSWITCH Version
>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
>> problem with some call requests setup.
>> Here is one of them:
>> 
>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
>> sofia/outside_1/20049112223344 []
>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send signal
>> sofia/outside_1/20049112223344 [BREAK]
>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
>> (sofia/outside_1/20049112223344) Running State Change CS_INIT
>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
>> (sofia/outside_1/20049112223344) State INIT
>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
>> sofia/outside_1/20049112223344 SOFIA INIT
>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
>> sofia/outside_1/20049112223344 Patched SDP
>> ?---
>> ?v=0
>> ?o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>> ?s=SIP Call
>> ?c=IN IP4 10.10.10.12
>> ?t=0 0
>> ?m=audio 24782 RTP/AVP 18 19
>> ?c=IN IP4 10.10.10.12
>> ?a=rtpmap:18 G729/8000
>> ?a=fmtp:18 annexb=yes
>> ?a=rtpmap:19 CN/8000
>> 
>> ?+++
>> ?v=0
>> ?o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>> ?s=FreeSWITCH
>> ?c=IN IP4 10.10.10.166
>> ?t=0 0
>> ?m=audio 19690 RTP/AVP 18 19
>> ?c=IN IP4 10.10.10.166
>> ?a=rtpmap:18 G729/8000
>> ?a=fmtp:18 annexb=yes
>> ?a=rtpmap:19 CN/8000
>> 
>> ?2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
>> 2014-09-17 23:11:20Z 64bit
>> ?Local SDP:
>> ?v=0
>> ?o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>> ?s=FreeSWITCH
>> ?c=IN IP4 10.10.10.166
>> ?t=0 0
>> ?m=audio 19690 RTP/AVP 18 19
>> ?c=IN IP4 10.10.10.166
>> ?a=rtpmap:18 G729/8000
>> ?a=fmtp:18 annexb=yes
>> ?a=rtpmap:19 CN/8000
>> 
>> sending out SDP exactly as shown above.
>> The other side is responsing with 100, then 180 Ringing wo/SDP then 183 w/SDP
>> ?
>> ?2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
>> sofia/outside_1/20049112223344 entering state [proceeding][183]
>> ?2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>> ?v=0
>> ?o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>> ?s=Dialogic-SIP
>> ?c=IN IP4 10.10.10.198
>> ?t=0 0
>> ?m=audio 10024 RTP/AVP 19 18
>> ?a=rtpmap:19 CN/8000
>> ?a=rtpmap:18 G729/8000
>> ?a=fmtp:18 annexb=yes
>> ?a=silenceSupp:off - - - -
>> ?a=ptime:20
>> 
>> ?2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
>> sofia/outside_1/20049112223344 Patched SDP
>> ?---
>> ?v=0
>> ?o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>> ?s=FreeSWITCH
>> ?c=IN IP4 10.10.10.166
>> ?t=0 0
>> ?m=audio 29492 RTP/AVP 18 19
>> ?c=IN IP4 10.10.10.166
>> ?a=rtpmap:18 G729/8000
>> ?a=fmtp:18 annexb=yes
>> ?a=rtpmap:19 CN/8000
>> 
>> ?+++
>> ?v=0
>> ?o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>> ?s=FreeSWITCH
>> ?c=IN IP4 10.10.10.166
>> ?t=0 0
>> ?m=audio 29492 RTP/AVP 18 19
>> ?c=IN IP4 10.10.10.166
>> ?a=rtpmap:18 G729/8000
>> ?a=fmtp:18 annexb=yes
>> ?a=rtpmap:19 CN/8000
>> 
>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651 Invalid codec CN!
>> 2014-12-11 14:29:12.214608 [ERR] switch_core_media.c:2294 Can't load codec?
>> 2014-12-11 14:29:12.214608 [NOTICE] switch_core_media.c:2295 Hangup
>> sofia/outside_1/20049112223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
>> sofia/outside_1/20049112223344 [KILL]
>> 
>> and Freeswitch is terminating the call by using CANCEL.
>> 
>> My modules.conf.xml config part looks like following:
>> ? ? <!-- Codec Interfaces -->
>> ? ? <load module="mod_spandsp"/>
>> ? ? <load module="mod_g723_1"/>
>> ? ? <load module="mod_g729"/>
>> ? ? <!--load module="mod_amr"/-->
>> ? ? <!--<load module="mod_ilbc"/>-->
>> ? ? <load module="mod_h26x"/>
>> ? ? <!--load module="mod_vp8"/-->
>> ? ? <!--load module="mod_b64"/-->
>> ? ? <!--<load module="mod_siren"/>-->
>> ? ? <!--<load module="mod_isac"/>-->
>> ? ? <!--<load module="mod_celt"/>-->
>> ? ? <!--load module="mod_opus"/-->
>> 
>> Is it a problem with CN definition?
>> Could you please lead me where the problem is located?
>> 
>> Thanks in advance.
>> Mac.
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From mbsip at gazeta.pl  Thu Dec 11 19:37:06 2014
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Thu, 11 Dec 2014 17:37:06 +0100
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call
	request
In-Reply-To: <D0AF1E70.8F750%krice@freeswitch.org>
References: <CAFiqYum79cR0Tfa-Guo+R+3YruNUTm_jeZAsf4WsL7SZMB6E=w@mail.gmail.com>
	<D0AF1E70.8F750%krice@freeswitch.org>
Message-ID: <CACynCY473-rhBSU4ftwN=HzK1KJai8R-nAwwKnWhCLzb4c6ZyQ@mail.gmail.com>

Thanks for prompt reply.

Steven, later today i will try to update FS to the latest git and check
this out.

Ken, i am fine with CN, passing thru is what i really need, but the
question is why FS generates such ERRors and how to get out of this.

Thanks
Mac.

2014-12-11 17:12 GMT+01:00 Ken Rice <krice at freeswitch.org>:

>  CN is comfort noise... Your cisco is sending it, and freeswitch is
> passing it thru because you have told it to use proxy media mode...
>
>
>
> On 12/11/14 9:53 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
> Your version is 3 months old. Can you reproduce this on the latest git
> head?
>
> On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>
> Hello,
>
> I am running FreeSWITCH Version
> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
> problem with some call requests setup.
> Here is one of them:
>
> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
> sofia/outside_1/20049112223344 []
> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send signal
> sofia/outside_1/20049112223344 [BREAK]
> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
> (sofia/outside_1/20049112223344) Running State Change CS_INIT
> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
> (sofia/outside_1/20049112223344) State INIT
> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
> sofia/outside_1/20049112223344 SOFIA INIT
> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
> sofia/outside_1/20049112223344 Patched SDP
>  ---
>  v=0
>  o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>  s=SIP Call
>  c=IN IP4 10.10.10.12
>  t=0 0
>  m=audio 24782 RTP/AVP 18 19
>  c=IN IP4 10.10.10.12
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=rtpmap:19 CN/8000
>
>  +++
>  v=0
>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>  s=FreeSWITCH
>  c=IN IP4 10.10.10.166
>  t=0 0
>  m=audio 19690 RTP/AVP 18 19
>  c=IN IP4 10.10.10.166
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=rtpmap:19 CN/8000
>
>  2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
> 2014-09-17 23:11:20Z 64bit
>  Local SDP:
>  v=0
>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>  s=FreeSWITCH
>  c=IN IP4 10.10.10.166
>  t=0 0
>  m=audio 19690 RTP/AVP 18 19
>  c=IN IP4 10.10.10.166
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=rtpmap:19 CN/8000
>
> sending out SDP exactly as shown above.
> The other side is responsing with 100, then 180 Ringing wo/SDP then 183
> w/SDP
>
>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
> sofia/outside_1/20049112223344 entering state [proceeding][183]
>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>  v=0
>  o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>  s=Dialogic-SIP
>  c=IN IP4 10.10.10.198
>  t=0 0
>  m=audio 10024 RTP/AVP 19 18
>  a=rtpmap:19 CN/8000
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=silenceSupp:off - - - -
>  a=ptime:20
>
>  2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
> sofia/outside_1/20049112223344 Patched SDP
>  ---
>  v=0
>  o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>  s=FreeSWITCH
>  c=IN IP4 10.10.10.166
>  t=0 0
>  m=audio 29492 RTP/AVP 18 19
>  c=IN IP4 10.10.10.166
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=rtpmap:19 CN/8000
>
>  +++
>  v=0
>  o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>  s=FreeSWITCH
>  c=IN IP4 10.10.10.166
>  t=0 0
>  m=audio 29492 RTP/AVP 18 19
>  c=IN IP4 10.10.10.166
>  a=rtpmap:18 G729/8000
>  a=fmtp:18 annexb=yes
>  a=rtpmap:19 CN/8000
>
> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651
> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR]
> switch_core_media.c:2294
> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE]
> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344
> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
> sofia/outside_1/20049112223344 [KILL]
>
> and Freeswitch is terminating the call by using CANCEL.
>
> My modules.conf.xml config part looks like following:
>     <!-- Codec Interfaces -->
>     <load module="mod_spandsp"/>
>     <load module="mod_g723_1"/>
>     <load module="mod_g729"/>
>     <!--load module="mod_amr"/-->
>     <!--<load module="mod_ilbc"/>-->
>     <load module="mod_h26x"/>
>     <!--load module="mod_vp8"/-->
>     <!--load module="mod_b64"/-->
>     <!--<load module="mod_siren"/>-->
>     <!--<load module="mod_isac"/>-->
>     <!--<load module="mod_celt"/>-->
>     <!--load module="mod_opus"/-->
>
> Is it a problem with CN definition?
> Could you please lead me where the problem is located?
>
> Thanks in advance.
> Mac.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Ken
>
>
>
> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Thu Dec 11 19:41:47 2014
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Dec 2014 10:41:47 -0600
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call
	request
In-Reply-To: <CACynCY473-rhBSU4ftwN=HzK1KJai8R-nAwwKnWhCLzb4c6ZyQ@mail.gmail.com>
References: <CAFiqYum79cR0Tfa-Guo+R+3YruNUTm_jeZAsf4WsL7SZMB6E=w@mail.gmail.com>
	<D0AF1E70.8F750%krice@freeswitch.org>
	<CACynCY473-rhBSU4ftwN=HzK1KJai8R-nAwwKnWhCLzb4c6ZyQ@mail.gmail.com>
Message-ID: <CAKbxfG9HOnFwhz3F_54kKVC4xWZfWfwyOtpQb8pjSk3y0_Gosg@mail.gmail.com>

For years I beg ppl to post issues to jira not the mailing list.
if you start typing and keywords like "problem" "not working" "issue"
appear in your text, you probably should be filing it to JIRA not here.


On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica <mbsip at gazeta.pl> wrote:

> Thanks for prompt reply.
>
> Steven, later today i will try to update FS to the latest git and check
> this out.
>
> Ken, i am fine with CN, passing thru is what i really need, but the
> question is why FS generates such ERRors and how to get out of this.
>
> Thanks
> Mac.
>
> 2014-12-11 17:12 GMT+01:00 Ken Rice <krice at freeswitch.org>:
>
>>  CN is comfort noise... Your cisco is sending it, and freeswitch is
>> passing it thru because you have told it to use proxy media mode...
>>
>>
>>
>> On 12/11/14 9:53 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>
>> Your version is 3 months old. Can you reproduce this on the latest git
>> head?
>>
>> On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>
>> Hello,
>>
>> I am running FreeSWITCH Version
>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
>> problem with some call requests setup.
>> Here is one of them:
>>
>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
>> sofia/outside_1/20049112223344 []
>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send signal
>> sofia/outside_1/20049112223344 [BREAK]
>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
>> (sofia/outside_1/20049112223344) Running State Change CS_INIT
>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
>> (sofia/outside_1/20049112223344) State INIT
>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
>> sofia/outside_1/20049112223344 SOFIA INIT
>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
>> sofia/outside_1/20049112223344 Patched SDP
>>  ---
>>  v=0
>>  o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>>  s=SIP Call
>>  c=IN IP4 10.10.10.12
>>  t=0 0
>>  m=audio 24782 RTP/AVP 18 19
>>  c=IN IP4 10.10.10.12
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=rtpmap:19 CN/8000
>>
>>  +++
>>  v=0
>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>  s=FreeSWITCH
>>  c=IN IP4 10.10.10.166
>>  t=0 0
>>  m=audio 19690 RTP/AVP 18 19
>>  c=IN IP4 10.10.10.166
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=rtpmap:19 CN/8000
>>
>>  2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
>> 2014-09-17 23:11:20Z 64bit
>>  Local SDP:
>>  v=0
>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>  s=FreeSWITCH
>>  c=IN IP4 10.10.10.166
>>  t=0 0
>>  m=audio 19690 RTP/AVP 18 19
>>  c=IN IP4 10.10.10.166
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=rtpmap:19 CN/8000
>>
>> sending out SDP exactly as shown above.
>> The other side is responsing with 100, then 180 Ringing wo/SDP then 183
>> w/SDP
>>
>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
>> sofia/outside_1/20049112223344 entering state [proceeding][183]
>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>>  v=0
>>  o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>>  s=Dialogic-SIP
>>  c=IN IP4 10.10.10.198
>>  t=0 0
>>  m=audio 10024 RTP/AVP 19 18
>>  a=rtpmap:19 CN/8000
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=silenceSupp:off - - - -
>>  a=ptime:20
>>
>>  2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
>> sofia/outside_1/20049112223344 Patched SDP
>>  ---
>>  v=0
>>  o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>>  s=FreeSWITCH
>>  c=IN IP4 10.10.10.166
>>  t=0 0
>>  m=audio 29492 RTP/AVP 18 19
>>  c=IN IP4 10.10.10.166
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=rtpmap:19 CN/8000
>>
>>  +++
>>  v=0
>>  o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>>  s=FreeSWITCH
>>  c=IN IP4 10.10.10.166
>>  t=0 0
>>  m=audio 29492 RTP/AVP 18 19
>>  c=IN IP4 10.10.10.166
>>  a=rtpmap:18 G729/8000
>>  a=fmtp:18 annexb=yes
>>  a=rtpmap:19 CN/8000
>>
>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651
>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR]
>> switch_core_media.c:2294
>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE]
>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344
>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
>> sofia/outside_1/20049112223344 [KILL]
>>
>> and Freeswitch is terminating the call by using CANCEL.
>>
>> My modules.conf.xml config part looks like following:
>>     <!-- Codec Interfaces -->
>>     <load module="mod_spandsp"/>
>>     <load module="mod_g723_1"/>
>>     <load module="mod_g729"/>
>>     <!--load module="mod_amr"/-->
>>     <!--<load module="mod_ilbc"/>-->
>>     <load module="mod_h26x"/>
>>     <!--load module="mod_vp8"/-->
>>     <!--load module="mod_b64"/-->
>>     <!--<load module="mod_siren"/>-->
>>     <!--<load module="mod_isac"/>-->
>>     <!--<load module="mod_celt"/>-->
>>     <!--load module="mod_opus"/-->
>>
>> Is it a problem with CN definition?
>> Could you please lead me where the problem is located?
>>
>> Thanks in advance.
>> Mac.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> Ken
>>
>>
>>
>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>> Twitter: @FreeSWITCH
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From sudhansh at gmail.com  Thu Dec 11 20:05:55 2014
From: sudhansh at gmail.com (Sudhanshu)
Date: Thu, 11 Dec 2014 22:35:55 +0530
Subject: [Freeswitch-users] Immediate disconnect while setting up incoming
Message-ID: <CALR8eWsrWe9vrJRxN+DWqa1BKpcA68L2Jf49fZcLmVs7=UdzEg@mail.gmail.com>

The setup is a GSM gateway, where I am trying to bridge all incoming calls
to the included sims to an external number.

The logs are: https://pastebin.freeswitch.org/23714
<https://www.google.com/url?q=https%3A%2F%2Fpastebin.freeswitch.org%2F23714&sa=D&sntz=1&usg=AFQjCNFn6olU7lpG9-4ifMIAfngLrkpvrA>

While trying a call to one of the sims, a ring or two can be heard followed
by an immediate busy tone, signifying disconnect, which is corroborated by
the logs. Why is this sudden disconnect originating? Can someone kindly
help?
--
Sudhanshu
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From sudhansh at gmail.com  Thu Dec 11 20:10:35 2014
From: sudhansh at gmail.com (Sudhanshu)
Date: Thu, 11 Dec 2014 22:40:35 +0530
Subject: [Freeswitch-users] fs_cli hangs
In-Reply-To: <CA+mgNr5S9_L6LcfqUwYhMfFJOpRxn3uDCitM_cwNQ7h7LuODhw@mail.gmail.com>
References: <CA+mgNr5S9_L6LcfqUwYhMfFJOpRxn3uDCitM_cwNQ7h7LuODhw@mail.gmail.com>
Message-ID: <CALR8eWsO6HK2ghd0cyXuVLMHF5BOBCcqP5uLX_Ey1UArS3YSrg@mail.gmail.com>

Enable debug logging and then check freeswitch.log.
What happens when you start freeswitch as anormal process (and not as a
daemon)?

--
Sudhanshu

On Thu, Dec 11, 2014 at 2:30 PM, akhil garg <akhilgarg7 at gmail.com> wrote:

> running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but no
> success.
>
>
>
> ------------------------------------------------------------------------------------------------------------------------------------------------
> OUTPUT 1:
>
> ------------------------------------------------------------------------------------------------------------------------------------------------
> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
> /root/.fs_cli_conf.
> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
> /etc/fs_cli.conf.
> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
> profile
> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>
> ------------------------------------------------------------------------------------------------------------------------------------------------
>
>
>
>
>
> ------------------------------------------------------------------------------------------------------------------------------------------------
> OUTPUT 2:
>
> ------------------------------------------------------------------------------------------------------------------------------------------------
> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
> /root/.fs_cli_conf.
> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
> /etc/fs_cli.conf.
> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
> profile
> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
> [auth/request]
> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: auth/request
>
>
> [DEBUG] esl.c:1465 esl_send() SEND
> auth ClueCon
>
> ------------------------------------------------------------------------------------------------------------------------------------------------
>
>
>
>
>
> ------------------------------------------------------------------------------------------------------------------------------------------------
> OUTPUT 3:
>
> ------------------------------------------------------------------------------------------------------------------------------------------------
> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
> /root/.fs_cli_conf.
> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
> /etc/fs_cli.conf.
> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
> profile
> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
> [auth/request]
> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: auth/request
>
>
> [DEBUG] esl.c:1465 esl_send() SEND
> auth ClueCon
>
>
> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
> [command/reply]
> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK
> accepted]
> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
> Event-Name: SOCKET_DATA
> Content-Type: command/reply
> Reply-Text: +OK accepted
>
>
> [DEBUG] esl.c:1465 esl_send() SEND
> log
>
> ------------------------------------------------------------------------------------------------------------------------------------------------
>
>
>
> --
> regards,
> akhil
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From idokan at gmail.com  Thu Dec 11 20:32:30 2014
From: idokan at gmail.com (ik)
Date: Thu, 11 Dec 2014 19:32:30 +0200
Subject: [Freeswitch-users] Controling playback recording position
Message-ID: <CAOccsKDYbd4NmE5CKzZEoeEz++-5Qmi6T_t33pMprY+yQKYJMQ@mail.gmail.com>

Hello,

How can I control the position of the recording that is played ?
For example, go 10 seconds forward or backward, like with Asterisk's
ControlPlayback cmd.

Thanks,
Ido
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From mbsip at gazeta.pl  Thu Dec 11 20:47:18 2014
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Thu, 11 Dec 2014 18:47:18 +0100
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call
	request
In-Reply-To: <CAKbxfG9HOnFwhz3F_54kKVC4xWZfWfwyOtpQb8pjSk3y0_Gosg@mail.gmail.com>
References: <CAFiqYum79cR0Tfa-Guo+R+3YruNUTm_jeZAsf4WsL7SZMB6E=w@mail.gmail.com>
	<D0AF1E70.8F750%krice@freeswitch.org>
	<CACynCY473-rhBSU4ftwN=HzK1KJai8R-nAwwKnWhCLzb4c6ZyQ@mail.gmail.com>
	<CAKbxfG9HOnFwhz3F_54kKVC4xWZfWfwyOtpQb8pjSk3y0_Gosg@mail.gmail.com>
Message-ID: <CACynCY7=0R3PZ7LTRWtf+nPRhRT7YAXJePemjn01Jb-B+-EOzQ@mail.gmail.com>

Sure i'll try with jira

Mac.

2014-12-11 17:41 GMT+01:00 Anthony Minessale <anthony.minessale at gmail.com>:

> For years I beg ppl to post issues to jira not the mailing list.
> if you start typing and keywords like "problem" "not working" "issue"
> appear in your text, you probably should be filing it to JIRA not here.
>
>
> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica <mbsip at gazeta.pl> wrote:
>
>> Thanks for prompt reply.
>>
>> Steven, later today i will try to update FS to the latest git and check
>> this out.
>>
>> Ken, i am fine with CN, passing thru is what i really need, but the
>> question is why FS generates such ERRors and how to get out of this.
>>
>> Thanks
>> Mac.
>>
>> 2014-12-11 17:12 GMT+01:00 Ken Rice <krice at freeswitch.org>:
>>
>>>  CN is comfort noise... Your cisco is sending it, and freeswitch is
>>> passing it thru because you have told it to use proxy media mode...
>>>
>>>
>>>
>>> On 12/11/14 9:53 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>>
>>> Your version is 3 months old. Can you reproduce this on the latest git
>>> head?
>>>
>>> On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>>
>>> Hello,
>>>
>>> I am running FreeSWITCH Version
>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
>>> problem with some call requests setup.
>>> Here is one of them:
>>>
>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
>>> sofia/outside_1/20049112223344 []
>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send
>>> signal sofia/outside_1/20049112223344 [BREAK]
>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT
>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
>>> (sofia/outside_1/20049112223344) State INIT
>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
>>> sofia/outside_1/20049112223344 SOFIA INIT
>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
>>> sofia/outside_1/20049112223344 Patched SDP
>>>  ---
>>>  v=0
>>>  o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>>>  s=SIP Call
>>>  c=IN IP4 10.10.10.12
>>>  t=0 0
>>>  m=audio 24782 RTP/AVP 18 19
>>>  c=IN IP4 10.10.10.12
>>>  a=rtpmap:18 G729/8000
>>>  a=fmtp:18 annexb=yes
>>>  a=rtpmap:19 CN/8000
>>>
>>>  +++
>>>  v=0
>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>  s=FreeSWITCH
>>>  c=IN IP4 10.10.10.166
>>>  t=0 0
>>>  m=audio 19690 RTP/AVP 18 19
>>>  c=IN IP4 10.10.10.166
>>>  a=rtpmap:18 G729/8000
>>>  a=fmtp:18 annexb=yes
>>>  a=rtpmap:19 CN/8000
>>>
>>>  2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
>>> 2014-09-17 23:11:20Z 64bit
>>>  Local SDP:
>>>  v=0
>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>  s=FreeSWITCH
>>>  c=IN IP4 10.10.10.166
>>>  t=0 0
>>>  m=audio 19690 RTP/AVP 18 19
>>>  c=IN IP4 10.10.10.166
>>>  a=rtpmap:18 G729/8000
>>>  a=fmtp:18 annexb=yes
>>>  a=rtpmap:19 CN/8000
>>>
>>> sending out SDP exactly as shown above.
>>> The other side is responsing with 100, then 180 Ringing wo/SDP then 183
>>> w/SDP
>>>
>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
>>> sofia/outside_1/20049112223344 entering state [proceeding][183]
>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>>>  v=0
>>>  o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>>>  s=Dialogic-SIP
>>>  c=IN IP4 10.10.10.198
>>>  t=0 0
>>>  m=audio 10024 RTP/AVP 19 18
>>>  a=rtpmap:19 CN/8000
>>>  a=rtpmap:18 G729/8000
>>>  a=fmtp:18 annexb=yes
>>>  a=silenceSupp:off - - - -
>>>  a=ptime:20
>>>
>>>  2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
>>> sofia/outside_1/20049112223344 Patched SDP
>>>  ---
>>>  v=0
>>>  o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>>>  s=FreeSWITCH
>>>  c=IN IP4 10.10.10.166
>>>  t=0 0
>>>  m=audio 29492 RTP/AVP 18 19
>>>  c=IN IP4 10.10.10.166
>>>  a=rtpmap:18 G729/8000
>>>  a=fmtp:18 annexb=yes
>>>  a=rtpmap:19 CN/8000
>>>
>>>  +++
>>>  v=0
>>>  o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>>>  s=FreeSWITCH
>>>  c=IN IP4 10.10.10.166
>>>  t=0 0
>>>  m=audio 29492 RTP/AVP 18 19
>>>  c=IN IP4 10.10.10.166
>>>  a=rtpmap:18 G729/8000
>>>  a=fmtp:18 annexb=yes
>>>  a=rtpmap:19 CN/8000
>>>
>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651
>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR]
>>> switch_core_media.c:2294
>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE]
>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344
>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
>>> sofia/outside_1/20049112223344 [KILL]
>>>
>>> and Freeswitch is terminating the call by using CANCEL.
>>>
>>> My modules.conf.xml config part looks like following:
>>>     <!-- Codec Interfaces -->
>>>     <load module="mod_spandsp"/>
>>>     <load module="mod_g723_1"/>
>>>     <load module="mod_g729"/>
>>>     <!--load module="mod_amr"/-->
>>>     <!--<load module="mod_ilbc"/>-->
>>>     <load module="mod_h26x"/>
>>>     <!--load module="mod_vp8"/-->
>>>     <!--load module="mod_b64"/-->
>>>     <!--<load module="mod_siren"/>-->
>>>     <!--<load module="mod_isac"/>-->
>>>     <!--<load module="mod_celt"/>-->
>>>     <!--load module="mod_opus"/-->
>>>
>>> Is it a problem with CN definition?
>>> Could you please lead me where the problem is located?
>>>
>>> Thanks in advance.
>>> Mac.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> ------------------------------
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> Ken
>>>
>>>
>>>
>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>> Twitter: @FreeSWITCH
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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 Dec 11 21:07:23 2014
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Dec 2014 12:07:23 -0600
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call
	request
In-Reply-To: <CACynCY7=0R3PZ7LTRWtf+nPRhRT7YAXJePemjn01Jb-B+-EOzQ@mail.gmail.com>
References: <CAFiqYum79cR0Tfa-Guo+R+3YruNUTm_jeZAsf4WsL7SZMB6E=w@mail.gmail.com>
	<D0AF1E70.8F750%krice@freeswitch.org>
	<CACynCY473-rhBSU4ftwN=HzK1KJai8R-nAwwKnWhCLzb4c6ZyQ@mail.gmail.com>
	<CAKbxfG9HOnFwhz3F_54kKVC4xWZfWfwyOtpQb8pjSk3y0_Gosg@mail.gmail.com>
	<CACynCY7=0R3PZ7LTRWtf+nPRhRT7YAXJePemjn01Jb-B+-EOzQ@mail.gmail.com>
Message-ID: <CAEJMVkBjD-vvVCDuKEyYj9OuNSa5P3BorXkQUGu8PTBYZ5q2Qg@mail.gmail.com>

Funny, payload 13 is CN, which is 0x13 in base16 or 19 in base10, Please
file a JIRA, I would need to see a packet capture for this one.


On Thu, Dec 11, 2014 at 11:47 AM, Maciej Bylica <mbsip at gazeta.pl> wrote:

> Sure i'll try with jira
>
> Mac.
>
> 2014-12-11 17:41 GMT+01:00 Anthony Minessale <anthony.minessale at gmail.com>
> :
>
>> For years I beg ppl to post issues to jira not the mailing list.
>> if you start typing and keywords like "problem" "not working" "issue"
>> appear in your text, you probably should be filing it to JIRA not here.
>>
>>
>> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>
>>> Thanks for prompt reply.
>>>
>>> Steven, later today i will try to update FS to the latest git and check
>>> this out.
>>>
>>> Ken, i am fine with CN, passing thru is what i really need, but the
>>> question is why FS generates such ERRors and how to get out of this.
>>>
>>> Thanks
>>> Mac.
>>>
>>> 2014-12-11 17:12 GMT+01:00 Ken Rice <krice at freeswitch.org>:
>>>
>>>>  CN is comfort noise... Your cisco is sending it, and freeswitch is
>>>> passing it thru because you have told it to use proxy media mode...
>>>>
>>>>
>>>>
>>>> On 12/11/14 9:53 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>>>
>>>> Your version is 3 months old. Can you reproduce this on the latest git
>>>> head?
>>>>
>>>> On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>>>
>>>> Hello,
>>>>
>>>> I am running FreeSWITCH Version
>>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
>>>> problem with some call requests setup.
>>>> Here is one of them:
>>>>
>>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
>>>> sofia/outside_1/20049112223344 []
>>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
>>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
>>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send
>>>> signal sofia/outside_1/20049112223344 [BREAK]
>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
>>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT
>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
>>>> (sofia/outside_1/20049112223344) State INIT
>>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
>>>> sofia/outside_1/20049112223344 SOFIA INIT
>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>  ---
>>>>  v=0
>>>>  o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>>>>  s=SIP Call
>>>>  c=IN IP4 10.10.10.12
>>>>  t=0 0
>>>>  m=audio 24782 RTP/AVP 18 19
>>>>  c=IN IP4 10.10.10.12
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=rtpmap:19 CN/8000
>>>>
>>>>  +++
>>>>  v=0
>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>  s=FreeSWITCH
>>>>  c=IN IP4 10.10.10.166
>>>>  t=0 0
>>>>  m=audio 19690 RTP/AVP 18 19
>>>>  c=IN IP4 10.10.10.166
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=rtpmap:19 CN/8000
>>>>
>>>>  2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
>>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
>>>> 2014-09-17 23:11:20Z 64bit
>>>>  Local SDP:
>>>>  v=0
>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>  s=FreeSWITCH
>>>>  c=IN IP4 10.10.10.166
>>>>  t=0 0
>>>>  m=audio 19690 RTP/AVP 18 19
>>>>  c=IN IP4 10.10.10.166
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=rtpmap:19 CN/8000
>>>>
>>>> sending out SDP exactly as shown above.
>>>> The other side is responsing with 100, then 180 Ringing wo/SDP then 183
>>>> w/SDP
>>>>
>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
>>>> sofia/outside_1/20049112223344 entering state [proceeding][183]
>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>>>>  v=0
>>>>  o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>>>>  s=Dialogic-SIP
>>>>  c=IN IP4 10.10.10.198
>>>>  t=0 0
>>>>  m=audio 10024 RTP/AVP 19 18
>>>>  a=rtpmap:19 CN/8000
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=silenceSupp:off - - - -
>>>>  a=ptime:20
>>>>
>>>>  2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>  ---
>>>>  v=0
>>>>  o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>>>>  s=FreeSWITCH
>>>>  c=IN IP4 10.10.10.166
>>>>  t=0 0
>>>>  m=audio 29492 RTP/AVP 18 19
>>>>  c=IN IP4 10.10.10.166
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=rtpmap:19 CN/8000
>>>>
>>>>  +++
>>>>  v=0
>>>>  o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>>>>  s=FreeSWITCH
>>>>  c=IN IP4 10.10.10.166
>>>>  t=0 0
>>>>  m=audio 29492 RTP/AVP 18 19
>>>>  c=IN IP4 10.10.10.166
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=rtpmap:19 CN/8000
>>>>
>>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651
>>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR]
>>>> switch_core_media.c:2294
>>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE]
>>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344
>>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
>>>> sofia/outside_1/20049112223344 [KILL]
>>>>
>>>> and Freeswitch is terminating the call by using CANCEL.
>>>>
>>>> My modules.conf.xml config part looks like following:
>>>>     <!-- Codec Interfaces -->
>>>>     <load module="mod_spandsp"/>
>>>>     <load module="mod_g723_1"/>
>>>>     <load module="mod_g729"/>
>>>>     <!--load module="mod_amr"/-->
>>>>     <!--<load module="mod_ilbc"/>-->
>>>>     <load module="mod_h26x"/>
>>>>     <!--load module="mod_vp8"/-->
>>>>     <!--load module="mod_b64"/-->
>>>>     <!--<load module="mod_siren"/>-->
>>>>     <!--<load module="mod_isac"/>-->
>>>>     <!--<load module="mod_celt"/>-->
>>>>     <!--load module="mod_opus"/-->
>>>>
>>>> Is it a problem with CN definition?
>>>> Could you please lead me where the problem is located?
>>>>
>>>> Thanks in advance.
>>>> Mac.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> ------------------------------
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> --
>>>> Ken
>>>>
>>>>
>>>>
>>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>> Twitter: @FreeSWITCH
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From brian at freeswitch.org  Thu Dec 11 21:08:13 2014
From: brian at freeswitch.org (Brian West)
Date: Thu, 11 Dec 2014 12:08:13 -0600
Subject: [Freeswitch-users] Controling playback recording position
In-Reply-To: <CAOccsKDYbd4NmE5CKzZEoeEz++-5Qmi6T_t33pMprY+yQKYJMQ@mail.gmail.com>
References: <CAOccsKDYbd4NmE5CKzZEoeEz++-5Qmi6T_t33pMprY+yQKYJMQ@mail.gmail.com>
Message-ID: <CAEJMVkDxgrB9DbCvMAH4ZJTTpS7kJ7qoBwc9_kFJDprCtNHtTA@mail.gmail.com>

See scripts/lua/callback.lua, javascript has a similar method too.



On Thu, Dec 11, 2014 at 11:32 AM, ik <idokan at gmail.com> wrote:

> Hello,
>
> How can I control the position of the recording that is played ?
> For example, go 10 seconds forward or backward, like with Asterisk's
> ControlPlayback cmd.
>
> Thanks,
> Ido
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From arsen.semionov at gmail.com  Thu Dec 11 21:44:13 2014
From: arsen.semionov at gmail.com (Arsen)
Date: Thu, 11 Dec 2014 20:44:13 +0200
Subject: [Freeswitch-users] Upgrade 1.4.14 TLS issue
Message-ID: <CABoraY5LzxRaGAeVGzvU0C93qP880cJT-SY=+zm1sxVaPmzoQw@mail.gmail.com>

Hi guys,

I've upgraded my FS from 1.2 to 1.4.14 without changes in the config files
and TLS stopped working. I can't register any more (503 service unavailable
error).
I don't see any logs in the console, but with ngrep I can see that the
traffic hits the server on port 5061.
tls_version =  sslv23
if I change to tlsv1 I can register but outgoing calls don't work
Any ideas?

-- 
Regards,
Arsen.
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From blasterjr at gmail.com  Thu Dec 11 21:51:59 2014
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Thu, 11 Dec 2014 11:51:59 -0700
Subject: [Freeswitch-users] Caller ID - Not Defined when executing
 dialplan, although Lua script is pulling CID from Database
In-Reply-To: <9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com>
References: <CAK0YwNkcb5HPjw_8mXezXDx3fEtnJcv8DLZL+Siv2kjq2a3zGw@mail.gmail.com>
	<CAO3hyzGa6CzOkyH=Fthdh2RB-4cYwoMnE83mWQJ-97NGHLf+6A@mail.gmail.com>
	<9312DD37-1999-4B29-A2C0-BC8429C6B866@jerris.com>
Message-ID: <CAO3hyzG6+ZrEgT9YGfh4jmB6UOp+rjNosEiHLf+cMEepEfdsFA@mail.gmail.com>

Errr yeah, sorry Michael is right they're origination not effective :P

On Thu, Dec 11, 2014 at 8:34 AM, Michael Jerris <mike at jerris.com> wrote:

> you should be looking at the origination_caller_id_* vars, not effective
>
> On Dec 11, 2014, at 10:26 AM, Chris Tunbridge <blasterjr at gmail.com> wrote:
>
> The variable outbound_caller_id_number and outbound_caller_id_name are not
> related to the caller id on outbound calls.
>
> On your outbound dial plan you need to set something like the following
>
>       <action application="export"
> data="effective_caller_id_name=${user_data(${username}@${domain_name} var
> outbound_caller_id_name)}"/>
>       <action application="export"
> data="effective_caller_id_number=${user_data(${username}@${domain_name}
> var outbound_caller_id_number)}"/>
>
> This will cause the system to pull the settings from the users profile and
> use it for the outgoing call.
>
> On Wed, Dec 10, 2014 at 12:27 PM, Joel White <joelewhite at gmail.com> wrote:
>
>> I have gone over the config with a fine tooth comb, it matches another
>> config of a server in which the caller id works fine.  What I am seeing
>> however is that in this system the variable is not exported to the
>> dialplan.  I may be missing something, and most likely I am.  I do have a
>> question though.  Is there a way to see what variables are defined for a
>> particular user in the FreeSWITCH console?
>>
>>
>> Here is some output of the Lua script on the server not pushing caller id
>>
>> 2014-12-09 16:54:22.819251 [NOTICE] switch_cpp.cpp:1328 Debug from
>> gen_dir_user_xml.lua, generated XML:
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>>   <section name="directory">
>>     <domain name="voip.net">
>>       <user id="26342" mailbox="26342">
>>         <params>
>>           <param name="password" value="XXXXXX"/>
>>           <param name="vm-password" value="26342"/>
>>           <param name="vm-enabled" value="true"/>
>>           <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>     </params>
>>         <variables>
>>           <variable name="toll_allow" value="local,domestic"/>
>>           <variable name="user_context" value="default"/>
>>           <variable name="outbound_caller_id_name" value="Michelle"/>
>>           <variable name="outbound_caller_id_number" value="1321XXXXXXX"/>
>>       <variable name="accountcode" value="PA"/>
>>       <!--<variable name="callgroup" value="techsupport"/>-->
>>         </variables>
>>       </user>
>>     </domain>
>>   </section>
>> </document>
>>
>>
>> And some output from the Lua script on the server with CID functioning
>>
>> 2014-12-09 21:50:55.458996 [NOTICE] switch_cpp.cpp:1328 Debug from
>> gen_dir_user_xml.lua, generated XML:
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>>   <section name="directory">
>>     <domain name="sip.voice.com">
>>       <user id="53201" mailbox="53201">
>>         <params>
>>           <param name="password" value="XXXXXXX"/>
>>           <param name="vm-password" value="53201"/>
>>           <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>         </params>
>>         <variables>
>>           <variable name="toll_allow" value="local,domestic"/>
>>           <variable name="user_context" value="default"/>
>>           <variable name="outbound_caller_id_name" value="Voice.com"/>
>>           <variable name="outbound_caller_id_number" value="1303XXXXXX"/>
>>           <!--<variable name="callgroup" value="techsupport"/>-->
>>         </variables>
>>       </user>
>>     </domain>
>>   </section>
>> </document>
>>
>>
>> Of course I removed any identifiable information, but it looks like the
>> CID is being set.  What am I missing here that is not allowing for the
>> variable to be passed to the dialplan?
>>
>>
>> This is what I get when the dialplan executes
>>
>> EXECUTE sofia/internal/26342 at voip.net
>> set(effective_caller_id_number=)
>> 2014-12-09 17:00:24.839237 [DEBUG] mod_dptools.c:1435 sofia/internal/
>> 26342 at voip.net SET [effective_caller_id_number]=[UNDEF]
>>
>>
>> Kinda strange and I could not find a discrepancy between the dialplan
>> configuration of the working server vs the non-working server
>>
>>
>>
>> Here is the version running on the server that works properly
>>
>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git
>> b942d0f 2014-11-03 19:53:00Z 64bit)
>>
>> And the version of the server having issue with CID
>>
>> FreeSWITCH Version 1.4.13+git~20141103T195300Z~
>> b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)
>>
>>
>>
>> I used diff and compared both servers conf directory recursively.  I
>> could not find a discrepancy in the files aside from Switch name, etc.
>>
>>
>> What am I missing?
>>
>> Could there be anything that that would overwrite the CID variable after
>> it is set by Lua (Generating the user profile)?
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.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  Thu Dec 11 21:54:03 2014
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Thu, 11 Dec 2014 11:54:03 -0700
Subject: [Freeswitch-users] fs_cli hangs
In-Reply-To: <CALR8eWsO6HK2ghd0cyXuVLMHF5BOBCcqP5uLX_Ey1UArS3YSrg@mail.gmail.com>
References: <CA+mgNr5S9_L6LcfqUwYhMfFJOpRxn3uDCitM_cwNQ7h7LuODhw@mail.gmail.com>
	<CALR8eWsO6HK2ghd0cyXuVLMHF5BOBCcqP5uLX_Ey1UArS3YSrg@mail.gmail.com>
Message-ID: <CAO3hyzGv7xbjDBQ+ZXkGAnau-YMkFW7oKBVGPXg1-iqZxBqDCQ@mail.gmail.com>

Can you do a pastebin of the contents of your
conf/autoload_configs/event_socket.conf.xml


On Thu, Dec 11, 2014 at 10:10 AM, Sudhanshu <sudhansh at gmail.com> wrote:

> Enable debug logging and then check freeswitch.log.
> What happens when you start freeswitch as anormal process (and not as a
> daemon)?
>
> --
> Sudhanshu
>
> On Thu, Dec 11, 2014 at 2:30 PM, akhil garg <akhilgarg7 at gmail.com> wrote:
>
>> running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but no
>> success.
>>
>>
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> OUTPUT 1:
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /root/.fs_cli_conf.
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /etc/fs_cli.conf.
>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
>> profile
>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> OUTPUT 2:
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /root/.fs_cli_conf.
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /etc/fs_cli.conf.
>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
>> profile
>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>> [auth/request]
>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>> Event-Name: SOCKET_DATA
>> Content-Type: auth/request
>>
>>
>> [DEBUG] esl.c:1465 esl_send() SEND
>> auth ClueCon
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> OUTPUT 3:
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /root/.fs_cli_conf.
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /etc/fs_cli.conf.
>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
>> profile
>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>> [auth/request]
>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>> Event-Name: SOCKET_DATA
>> Content-Type: auth/request
>>
>>
>> [DEBUG] esl.c:1465 esl_send() SEND
>> auth ClueCon
>>
>>
>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>> [command/reply]
>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK
>> accepted]
>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>> Event-Name: SOCKET_DATA
>> Content-Type: command/reply
>> Reply-Text: +OK accepted
>>
>>
>> [DEBUG] esl.c:1465 esl_send() SEND
>> log
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>
>>
>>
>> --
>> regards,
>> akhil
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From my.post at hotmail.com  Thu Dec 11 22:25:42 2014
From: my.post at hotmail.com (Post)
Date: Thu, 11 Dec 2014 22:25:42 +0300
Subject: [Freeswitch-users] How to play custom ringback using outbound ES.
Message-ID: <BLU436-SMTP296D64462B0A39D375B43CED630@phx.gbl>

Hi everyone,

     This is probably very simple, but I am stuck. I am using simple 
perl script to listen for incoming connection from freeswitch. I am 
willing to learn how to send a custom ringback to legA user. So far I 
managed to play some wav file in the following way:

   $con->execute("pre_answer", "", $uuid);
   $con->execute("playback", "somefile.wav", $uuid);

So this works ok.

Now I am trying to send a custom ringback (here are my guesses):

1)
   $con->execute("set", "ringback=\$\${ru-ring}", $uuid);
   $con->execute("pre_answer", "", $uuid);

doesn't work (hear silence on legA user, see 183 with SDP in fs_cli 
sofia trace)



2)
   $con->execute("set", "ringback=\$\${ru-ring}", $uuid);
   $con->execute("ring_ready", "", $uuid);

doesn't work either (hear RBT, pre-programmed in endpoint, see 180 w/o 
SDP in fs_cli sofia trace).

Please give some advice.

Regards,

Pavel.




From krice at freeswitch.org  Thu Dec 11 22:37:10 2014
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 11 Dec 2014 13:37:10 -0600
Subject: [Freeswitch-users] Upgrade 1.4.14 TLS issue
In-Reply-To: <CABoraY5LzxRaGAeVGzvU0C93qP880cJT-SY=+zm1sxVaPmzoQw@mail.gmail.com>
Message-ID: <D0AF4E86.8F763%krice@freeswitch.org>

This could be because anything below TLSv1 is no longer support due to the
various attacks and downgrade attacks (like POODLE etc)

If you are using something below TLSv1 you might as well just turn it off


On 12/11/14 12:44 PM, "Arsen" <arsen.semionov at gmail.com> wrote:

> Hi guys,
> 
> I've upgraded my FS from 1.2 to 1.4.14 without changes in the config files and
> TLS stopped working. I can't register any more (503 service unavailable
> error).
> I don't see any logs in the console, but with ngrep I can see that the traffic
> hits the server on port 5061.
> tls_version = ?sslv23
> if I change to tlsv1 I can register but outgoing calls don't work
> Any ideas?

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
Twitter: @FreeSWITCH


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From arsen.semionov at gmail.com  Thu Dec 11 22:47:54 2014
From: arsen.semionov at gmail.com (Arsen)
Date: Thu, 11 Dec 2014 21:47:54 +0200
Subject: [Freeswitch-users] Upgrade 1.4.14 TLS issue
In-Reply-To: <D0AF4E86.8F763%krice@freeswitch.org>
References: <CABoraY5LzxRaGAeVGzvU0C93qP880cJT-SY=+zm1sxVaPmzoQw@mail.gmail.com>
	<D0AF4E86.8F763%krice@freeswitch.org>
Message-ID: <CABoraY5Ve+gPqbuPYMBz1PbuaVeeHbHph8Gxokdb2_C_XO-40w@mail.gmail.com>

Thanks for responce.
Sip profile is configured with SSLv23, I've just recreated ssl certificates
still getting 503 Service unavailable while registering...

On Thursday, December 11, 2014, Ken Rice <krice at freeswitch.org> wrote:

>  This could be because anything below TLSv1 is no longer support due to
> the various attacks and downgrade attacks (like POODLE etc)
>
> If you are using something below TLSv1 you might as well just turn it off
>
>
> On 12/11/14 12:44 PM, "Arsen" <arsen.semionov at gmail.com> wrote:
>
> Hi guys,
>
> I've upgraded my FS from 1.2 to 1.4.14 without changes in the config files
> and TLS stopped working. I can't register any more (503 service unavailable
> error).
> I don't see any logs in the console, but with ngrep I can see that the
> traffic hits the server on port 5061.
> tls_version =  sslv23
> if I change to tlsv1 I can register but outgoing calls don't work
> Any ideas?
>
>
> --
> Ken
>
>
>
> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
> Twitter: @FreeSWITCH
>
>

-- 
Regards,
Arsen.
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From pjintheusa at gmail.com  Thu Dec 11 23:35:15 2014
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 11 Dec 2014 15:35:15 -0500
Subject: [Freeswitch-users] Dialplan - Lua in-line expansion
Message-ID: <CAGK4cYxy4DTZVEGWprdkBB8JD9RY1fC2TD81HKB9pHYaDBTJGQ@mail.gmail.com>

I am going crazy - please tell me what I am missing:

I have a dialplan:

<action application="set" data="VUM_SAY_CID=${lua(util_get10digits.lua
${VUM_Outgoing_CallerID_Number})}" inline="true"/>


I have a lua script:


freeswitch.consoleLog("NOTICE","util_get10digits.lua \n")
freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n")
local result = "none"
*result = string.sub(argv[1], -10)*
freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n")
*return result*


I run the dialplan:

Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 Action
set(VUM_SAY_CID=${lua(util_get10digits.lua
${VUM_Outgoing_CallerID_Number})}) I
2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 util_get10digits.lua
2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed
[+12154791697]
*2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result:
[2154791697]*
2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/
2154791697 at 64.158.162.74 destroy/unlink session from object
EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 set(VUM_SAY_CID=)
*2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435
sofia/Carrier/2154791697 at 64.158.162.74 <2154791697 at 64.158.162.74> SET
[VUM_SAY_CID]=[UNDEF]*


The lua script runs correctly - how ever the result is not returned the
VUM_SAY_CID variable in the dialplan


The work around is to set the session variable in the script. But I
wondering what I am missing. Is "return result" not how you return a result
from Lua?


Thanks for any help.


Phil
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From chad at apartmentlines.com  Thu Dec 11 23:46:56 2014
From: chad at apartmentlines.com (Chad Phillips)
Date: Thu, 11 Dec 2014 13:46:56 -0700
Subject: [Freeswitch-users] Dialplan - Lua in-line expansion
In-Reply-To: <CAGK4cYxy4DTZVEGWprdkBB8JD9RY1fC2TD81HKB9pHYaDBTJGQ@mail.gmail.com>
References: <CAGK4cYxy4DTZVEGWprdkBB8JD9RY1fC2TD81HKB9pHYaDBTJGQ@mail.gmail.com>
Message-ID: <D133FCCA6BEB4C2B9A4383C689309413@gmail.com>

how about setting the channel var directly in the script?  

session:setVariable("VUM_SAY_CID", result);

although for what you?re doing, i really think there?s a cleaner way to do it straight from the dialplan XML...  


On Thursday, December 11, 2014 at 1:35 PM, Phillip Jones wrote:

> I am going crazy - please tell me what I am missing:
>  
> I have a dialplan:
>  
> <action application="set" data="VUM_SAY_CID=${lua(util_get10digits.lua ${VUM_Outgoing_CallerID_Number})}" inline="true"/>
>  
>  
> I have a lua script:
>  
>  
> freeswitch.consoleLog("NOTICE","util_get10digits.lua \n")
> freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n")
> local result = "none"
> result = string.sub(argv[1], -10)
> freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n")
> return result
>  
>  
> I run the dialplan:
>  
> Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 (mailto:2154791697 at 64.158.162.74) Action set(VUM_SAY_CID=${lua(util_get10digits.lua ${VUM_Outgoing_CallerID_Number})}) I
> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 util_get10digits.lua
> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed [+12154791697]
> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result: [2154791697]
> 2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/2154791697 at 64.158.162.74 (mailto:2154791697 at 64.158.162.74) destroy/unlink session from object
> EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 (mailto:2154791697 at 64.158.162.74) set(VUM_SAY_CID=)
> 2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435 sofia/Carrier/2154791697 at 64.158.162.74 (mailto:2154791697 at 64.158.162.74) SET [VUM_SAY_CID]=[UNDEF]
>  
>  
>  
> The lua script runs correctly - how ever the result is not returned the VUM_SAY_CID variable in the dialplan
>  
>  
> The work around is to set the session variable in the script. But I wondering what I am missing. Is "return result" not how you return a result from Lua?
>  
>  
> Thanks for any help.
>  
>  
> Phil  
> _________________________________________________________________________
> 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 joseph at onsip.com  Thu Dec 11 23:43:08 2014
From: joseph at onsip.com (Joseph Frazier)
Date: Thu, 11 Dec 2014 15:43:08 -0500
Subject: [Freeswitch-users] Incomplete UPDATEs after Attended Transfer
In-Reply-To: <CAANWw7EoJW84KDzJewdds3oa9taC8uTBA76zfKoWfDKG7PQyVg@mail.gmail.com>
References: <CAANWw7G0BkpbYrF3JSSkUgrtYyqZNtpiw0j548_PwMG1B4Faaw@mail.gmail.com>
	<CAANWw7GVqRt6N-Eq+1oYxMhjAhA2mO6QK1wkErwjaMtexW=4pA@mail.gmail.com>
	<CAEJMVkBV=sAEvoxOFwWJV3PghopL_NECaJAxAy7e4XcsLG2Rnw@mail.gmail.com>
	<CAANWw7EoJW84KDzJewdds3oa9taC8uTBA76zfKoWfDKG7PQyVg@mail.gmail.com>
Message-ID: <CAANWw7G1aqKBs4n_q-4=eQcZzZfyNDU1N3itbTNiNJQTjwJ5mw@mail.gmail.com>

I tried the same scenario with proxy-refer, and found that it's still the
case that no UPDATE is sent to the Referee, but also that the UPDATE sent
to the refer target is incorrect. I reported the latter bug to JIRA as
FS-7080: https://freeswitch.org/jira/browse/FS-7080

I'm still unclear on whether I should expect the Referee to receive an
UPDATE. I did see in mod_sofia.c that the nua_update calls are guarded by
User-Agent checks, none of which appear to match the Grandstream:
https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/endpoints/mod_sofia/mod_sofia.c#1630

Perhaps that is the problem?

Thanks,
Joseph

On Tue, Dec 9, 2014 at 9:13 PM, Joseph Frazier <joseph at onsip.com> wrote:

> The Referee is a "Grandstream GXV3275 1.0.2.15". For context, here's where
> it OKs the INVITE from the first call:
>
>
> https://gist.github.com/joseph-onsip/de90a66afdf37994c334#file-gistfile1-txt-L1813-L1838
>
> Thanks,
> Joseph
>
> On Tue, Dec 9, 2014 at 6:54 PM, Brian West <brian at freeswitch.org> wrote:
>
>> What device is the Referee?
>>
>> On Tue, Dec 9, 2014 at 4:30 PM, Joseph Frazier <joseph at onsip.com> wrote:
>>
>>> Hi everyone,
>>>
>>> I'm trying to do attended transfers on my FS 1.5.14 box and have the
>>> endpoints receive UPDATE requests indicating that the identity of the other
>>> leg has changed. I installed the v1.5.14 tag from source and uncommented
>>> the following line of sip_profiles/internal.xml:
>>>
>>>    <param name="NDLB-force-rport" value="true"/>
>>>
>>> Other than the above line, my configuration is completely vanilla.
>>>
>>> I have three extensions registered: 1000, 1007, 1006. My scenario is:
>>>
>>>    - 1000 calls 1007
>>>    - 1000 calls 1006
>>>    - 1000 attended transfers 1007 to 1006
>>>
>>> Here is my fs_cli with the following debugging options:
>>>
>>> sofia global siptrace on
>>>> sofia loglevel all 9
>>>> sofia tracelevel alert
>>>> console loglevel debug
>>>> fsctl debug_level 10
>>>
>>>
>>> https://gist.github.com/joseph-onsip/de90a66afdf37994c334
>>>
>>> After receiving the REFER, FS sends an UPDATE to the refer target, but
>>> not to the referee. Is there a configuration option I can set in order to
>>> have an UPDATE sent to the referee as well?
>>>
>>> Thanks,
>>> Joseph
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From pjintheusa at gmail.com  Fri Dec 12 00:32:05 2014
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 11 Dec 2014 16:32:05 -0500
Subject: [Freeswitch-users] Dialplan - Lua in-line expansion
In-Reply-To: <D133FCCA6BEB4C2B9A4383C689309413@gmail.com>
References: <CAGK4cYxy4DTZVEGWprdkBB8JD9RY1fC2TD81HKB9pHYaDBTJGQ@mail.gmail.com>
	<D133FCCA6BEB4C2B9A4383C689309413@gmail.com>
Message-ID: <CAGK4cYxP8hepPN0Pi+Um2KGJvjRmQkQqOW5wQ3=qK=oTE6dQ6A@mail.gmail.com>

Yeah - thanks - I realize I can do that - but it just drives me crazy I can
not get in-line expansion to work. I can't see what I am missing.

On Thu, Dec 11, 2014 at 3:46 PM, Chad Phillips <chad at apartmentlines.com>
wrote:

>  how about setting the channel var directly in the script?
>
> session:setVariable("VUM_SAY_CID", result);
>
> although for what you?re doing, i really think there?s a cleaner way to do
> it straight from the dialplan XML...
>
> On Thursday, December 11, 2014 at 1:35 PM, Phillip Jones wrote:
>
> I am going crazy - please tell me what I am missing:
>
> I have a dialplan:
>
> <action application="set" data="VUM_SAY_CID=${lua(util_get10digits.lua
> ${VUM_Outgoing_CallerID_Number})}" inline="true"/>
>
>
> I have a lua script:
>
>
> freeswitch.consoleLog("NOTICE","util_get10digits.lua \n")
> freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n")
> local result = "none"
> *result = string.sub(argv[1], -10)*
> freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n")
> *return result*
>
>
> I run the dialplan:
>
> Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 Action
> set(VUM_SAY_CID=${lua(util_get10digits.lua
> ${VUM_Outgoing_CallerID_Number})}) I
> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328
> util_get10digits.lua
> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed
> [+12154791697]
> *2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result:
> [2154791697 <%5B2154791697>]*
> 2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/
> 2154791697 at 64.158.162.74 destroy/unlink session from object
> EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 set(VUM_SAY_CID=)
> *2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435
> sofia/Carrier/2154791697 at 64.158.162.74 <2154791697 at 64.158.162.74> SET
> [VUM_SAY_CID]=[UNDEF]*
>
>
> The lua script runs correctly - how ever the result is not returned the
> VUM_SAY_CID variable in the dialplan
>
>
> The work around is to set the session variable in the script. But I
> wondering what I am missing. Is "return result" not how you return a result
> from Lua?
>
>
> Thanks for any help.
>
>
> Phil
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mbsip at gazeta.pl  Fri Dec 12 02:03:58 2014
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Fri, 12 Dec 2014 00:03:58 +0100
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call
	request
In-Reply-To: <CACynCY7=0R3PZ7LTRWtf+nPRhRT7YAXJePemjn01Jb-B+-EOzQ@mail.gmail.com>
References: <CAFiqYum79cR0Tfa-Guo+R+3YruNUTm_jeZAsf4WsL7SZMB6E=w@mail.gmail.com>
	<D0AF1E70.8F750%krice@freeswitch.org>
	<CACynCY473-rhBSU4ftwN=HzK1KJai8R-nAwwKnWhCLzb4c6ZyQ@mail.gmail.com>
	<CAKbxfG9HOnFwhz3F_54kKVC4xWZfWfwyOtpQb8pjSk3y0_Gosg@mail.gmail.com>
	<CACynCY7=0R3PZ7LTRWtf+nPRhRT7YAXJePemjn01Jb-B+-EOzQ@mail.gmail.com>
Message-ID: <CACynCY6JDc3X5_O1-VRMCCa2DkNYf4YRtObe17qsP9apCOU_Ow@mail.gmail.com>

Steven, i have the same issue on FreeSWITCH Version
1.5.15b+git~20141211T210353Z~be0e09bd7f
I am about to rise this ticket on jira.
Thanks anyway.

BR
Mac.

2014-12-11 18:47 GMT+01:00 Maciej Bylica <mbsip at gazeta.pl>:

> Sure i'll try with jira
>
> Mac.
>
> 2014-12-11 17:41 GMT+01:00 Anthony Minessale <anthony.minessale at gmail.com>
> :
>
>> For years I beg ppl to post issues to jira not the mailing list.
>> if you start typing and keywords like "problem" "not working" "issue"
>> appear in your text, you probably should be filing it to JIRA not here.
>>
>>
>> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>
>>> Thanks for prompt reply.
>>>
>>> Steven, later today i will try to update FS to the latest git and check
>>> this out.
>>>
>>> Ken, i am fine with CN, passing thru is what i really need, but the
>>> question is why FS generates such ERRors and how to get out of this.
>>>
>>> Thanks
>>> Mac.
>>>
>>> 2014-12-11 17:12 GMT+01:00 Ken Rice <krice at freeswitch.org>:
>>>
>>>>  CN is comfort noise... Your cisco is sending it, and freeswitch is
>>>> passing it thru because you have told it to use proxy media mode...
>>>>
>>>>
>>>>
>>>> On 12/11/14 9:53 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>>>
>>>> Your version is 3 months old. Can you reproduce this on the latest git
>>>> head?
>>>>
>>>> On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>>>
>>>> Hello,
>>>>
>>>> I am running FreeSWITCH Version
>>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
>>>> problem with some call requests setup.
>>>> Here is one of them:
>>>>
>>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
>>>> sofia/outside_1/20049112223344 []
>>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
>>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
>>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send
>>>> signal sofia/outside_1/20049112223344 [BREAK]
>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
>>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT
>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
>>>> (sofia/outside_1/20049112223344) State INIT
>>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
>>>> sofia/outside_1/20049112223344 SOFIA INIT
>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>  ---
>>>>  v=0
>>>>  o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>>>>  s=SIP Call
>>>>  c=IN IP4 10.10.10.12
>>>>  t=0 0
>>>>  m=audio 24782 RTP/AVP 18 19
>>>>  c=IN IP4 10.10.10.12
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=rtpmap:19 CN/8000
>>>>
>>>>  +++
>>>>  v=0
>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>  s=FreeSWITCH
>>>>  c=IN IP4 10.10.10.166
>>>>  t=0 0
>>>>  m=audio 19690 RTP/AVP 18 19
>>>>  c=IN IP4 10.10.10.166
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=rtpmap:19 CN/8000
>>>>
>>>>  2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
>>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
>>>> 2014-09-17 23:11:20Z 64bit
>>>>  Local SDP:
>>>>  v=0
>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>  s=FreeSWITCH
>>>>  c=IN IP4 10.10.10.166
>>>>  t=0 0
>>>>  m=audio 19690 RTP/AVP 18 19
>>>>  c=IN IP4 10.10.10.166
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=rtpmap:19 CN/8000
>>>>
>>>> sending out SDP exactly as shown above.
>>>> The other side is responsing with 100, then 180 Ringing wo/SDP then 183
>>>> w/SDP
>>>>
>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
>>>> sofia/outside_1/20049112223344 entering state [proceeding][183]
>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>>>>  v=0
>>>>  o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>>>>  s=Dialogic-SIP
>>>>  c=IN IP4 10.10.10.198
>>>>  t=0 0
>>>>  m=audio 10024 RTP/AVP 19 18
>>>>  a=rtpmap:19 CN/8000
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=silenceSupp:off - - - -
>>>>  a=ptime:20
>>>>
>>>>  2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>  ---
>>>>  v=0
>>>>  o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>>>>  s=FreeSWITCH
>>>>  c=IN IP4 10.10.10.166
>>>>  t=0 0
>>>>  m=audio 29492 RTP/AVP 18 19
>>>>  c=IN IP4 10.10.10.166
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=rtpmap:19 CN/8000
>>>>
>>>>  +++
>>>>  v=0
>>>>  o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>>>>  s=FreeSWITCH
>>>>  c=IN IP4 10.10.10.166
>>>>  t=0 0
>>>>  m=audio 29492 RTP/AVP 18 19
>>>>  c=IN IP4 10.10.10.166
>>>>  a=rtpmap:18 G729/8000
>>>>  a=fmtp:18 annexb=yes
>>>>  a=rtpmap:19 CN/8000
>>>>
>>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651
>>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR]
>>>> switch_core_media.c:2294
>>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE]
>>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344
>>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
>>>> sofia/outside_1/20049112223344 [KILL]
>>>>
>>>> and Freeswitch is terminating the call by using CANCEL.
>>>>
>>>> My modules.conf.xml config part looks like following:
>>>>     <!-- Codec Interfaces -->
>>>>     <load module="mod_spandsp"/>
>>>>     <load module="mod_g723_1"/>
>>>>     <load module="mod_g729"/>
>>>>     <!--load module="mod_amr"/-->
>>>>     <!--<load module="mod_ilbc"/>-->
>>>>     <load module="mod_h26x"/>
>>>>     <!--load module="mod_vp8"/-->
>>>>     <!--load module="mod_b64"/-->
>>>>     <!--<load module="mod_siren"/>-->
>>>>     <!--<load module="mod_isac"/>-->
>>>>     <!--<load module="mod_celt"/>-->
>>>>     <!--load module="mod_opus"/-->
>>>>
>>>> Is it a problem with CN definition?
>>>> Could you please lead me where the problem is located?
>>>>
>>>> Thanks in advance.
>>>> Mac.
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> ------------------------------
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> --
>>>> Ken
>>>>
>>>>
>>>>
>>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>> Twitter: @FreeSWITCH
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From mbsip at gazeta.pl  Fri Dec 12 03:00:32 2014
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Fri, 12 Dec 2014 01:00:32 +0100
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call
	request
In-Reply-To: <CAEJMVkBjD-vvVCDuKEyYj9OuNSa5P3BorXkQUGu8PTBYZ5q2Qg@mail.gmail.com>
References: <CAFiqYum79cR0Tfa-Guo+R+3YruNUTm_jeZAsf4WsL7SZMB6E=w@mail.gmail.com>
	<D0AF1E70.8F750%krice@freeswitch.org>
	<CACynCY473-rhBSU4ftwN=HzK1KJai8R-nAwwKnWhCLzb4c6ZyQ@mail.gmail.com>
	<CAKbxfG9HOnFwhz3F_54kKVC4xWZfWfwyOtpQb8pjSk3y0_Gosg@mail.gmail.com>
	<CACynCY7=0R3PZ7LTRWtf+nPRhRT7YAXJePemjn01Jb-B+-EOzQ@mail.gmail.com>
	<CAEJMVkBjD-vvVCDuKEyYj9OuNSa5P3BorXkQUGu8PTBYZ5q2Qg@mail.gmail.com>
Message-ID: <CACynCY6gSx3v6c--Pocss1do=5tRF-fpwVxsg5NC5Pxu87gyFw@mail.gmail.com>

I will gather packet capture and file JIRA.

Thank You
Maciej.


Funny, payload 13 is CN, which is 0x13 in base16 or 19 in base10, Please
> file a JIRA, I would need to see a packet capture for this one.
>
>
> On Thu, Dec 11, 2014 at 11:47 AM, Maciej Bylica <mbsip at gazeta.pl
> <javascript:_e(%7B%7D,'cvml','mbsip at gazeta.pl');>> wrote:
>
>> Sure i'll try with jira
>>
>> Mac.
>>
>> 2014-12-11 17:41 GMT+01:00 Anthony Minessale <anthony.minessale at gmail.com
>> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>>:
>>
>>> For years I beg ppl to post issues to jira not the mailing list.
>>> if you start typing and keywords like "problem" "not working" "issue"
>>> appear in your text, you probably should be filing it to JIRA not here.
>>>
>>>
>>> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica <mbsip at gazeta.pl
>>> <javascript:_e(%7B%7D,'cvml','mbsip at gazeta.pl');>> wrote:
>>>
>>>> Thanks for prompt reply.
>>>>
>>>> Steven, later today i will try to update FS to the latest git and check
>>>> this out.
>>>>
>>>> Ken, i am fine with CN, passing thru is what i really need, but the
>>>> question is why FS generates such ERRors and how to get out of this.
>>>>
>>>> Thanks
>>>> Mac.
>>>>
>>>> 2014-12-11 17:12 GMT+01:00 Ken Rice <krice at freeswitch.org
>>>> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>>:
>>>>
>>>>>  CN is comfort noise... Your cisco is sending it, and freeswitch is
>>>>> passing it thru because you have told it to use proxy media mode...
>>>>>
>>>>>
>>>>>
>>>>> On 12/11/14 9:53 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>>>>
>>>>> Your version is 3 months old. Can you reproduce this on the latest git
>>>>> head?
>>>>>
>>>>> On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>>>>
>>>>> Hello,
>>>>>
>>>>> I am running FreeSWITCH Version
>>>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
>>>>> problem with some call requests setup.
>>>>> Here is one of them:
>>>>>
>>>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
>>>>> sofia/outside_1/20049112223344 []
>>>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
>>>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
>>>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send
>>>>> signal sofia/outside_1/20049112223344 [BREAK]
>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
>>>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT
>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
>>>>> (sofia/outside_1/20049112223344) State INIT
>>>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
>>>>> sofia/outside_1/20049112223344 SOFIA INIT
>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
>>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>>  ---
>>>>>  v=0
>>>>>  o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>>>>>  s=SIP Call
>>>>>  c=IN IP4 10.10.10.12
>>>>>  t=0 0
>>>>>  m=audio 24782 RTP/AVP 18 19
>>>>>  c=IN IP4 10.10.10.12
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=rtpmap:19 CN/8000
>>>>>
>>>>>  +++
>>>>>  v=0
>>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>>  s=FreeSWITCH
>>>>>  c=IN IP4 10.10.10.166
>>>>>  t=0 0
>>>>>  m=audio 19690 RTP/AVP 18 19
>>>>>  c=IN IP4 10.10.10.166
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=rtpmap:19 CN/8000
>>>>>
>>>>>  2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
>>>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
>>>>> 2014-09-17 23:11:20Z 64bit
>>>>>  Local SDP:
>>>>>  v=0
>>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>>  s=FreeSWITCH
>>>>>  c=IN IP4 10.10.10.166
>>>>>  t=0 0
>>>>>  m=audio 19690 RTP/AVP 18 19
>>>>>  c=IN IP4 10.10.10.166
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=rtpmap:19 CN/8000
>>>>>
>>>>> sending out SDP exactly as shown above.
>>>>> The other side is responsing with 100, then 180 Ringing wo/SDP then
>>>>> 183 w/SDP
>>>>>
>>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
>>>>> sofia/outside_1/20049112223344 entering state [proceeding][183]
>>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>>>>>  v=0
>>>>>  o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>>>>>  s=Dialogic-SIP
>>>>>  c=IN IP4 10.10.10.198
>>>>>  t=0 0
>>>>>  m=audio 10024 RTP/AVP 19 18
>>>>>  a=rtpmap:19 CN/8000
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=silenceSupp:off - - - -
>>>>>  a=ptime:20
>>>>>
>>>>>  2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
>>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>>  ---
>>>>>  v=0
>>>>>  o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>>>>>  s=FreeSWITCH
>>>>>  c=IN IP4 10.10.10.166
>>>>>  t=0 0
>>>>>  m=audio 29492 RTP/AVP 18 19
>>>>>  c=IN IP4 10.10.10.166
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=rtpmap:19 CN/8000
>>>>>
>>>>>  +++
>>>>>  v=0
>>>>>  o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>>>>>  s=FreeSWITCH
>>>>>  c=IN IP4 10.10.10.166
>>>>>  t=0 0
>>>>>  m=audio 29492 RTP/AVP 18 19
>>>>>  c=IN IP4 10.10.10.166
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=rtpmap:19 CN/8000
>>>>>
>>>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651
>>>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR]
>>>>> switch_core_media.c:2294
>>>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE]
>>>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344
>>>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
>>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
>>>>> sofia/outside_1/20049112223344 [KILL]
>>>>>
>>>>> and Freeswitch is terminating the call by using CANCEL.
>>>>>
>>>>> My modules.conf.xml config part looks like following:
>>>>>     <!-- Codec Interfaces -->
>>>>>     <load module="mod_spandsp"/>
>>>>>     <load module="mod_g723_1"/>
>>>>>     <load module="mod_g729"/>
>>>>>     <!--load module="mod_amr"/-->
>>>>>     <!--<load module="mod_ilbc"/>-->
>>>>>     <load module="mod_h26x"/>
>>>>>     <!--load module="mod_vp8"/-->
>>>>>     <!--load module="mod_b64"/-->
>>>>>     <!--<load module="mod_siren"/>-->
>>>>>     <!--<load module="mod_isac"/>-->
>>>>>     <!--<load module="mod_celt"/>-->
>>>>>     <!--load module="mod_opus"/-->
>>>>>
>>>>> Is it a problem with CN definition?
>>>>> Could you please lead me where the problem is located?
>>>>>
>>>>> Thanks in advance.
>>>>> Mac.
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> ------------------------------
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> --
>>>>> Ken
>>>>>
>>>>>
>>>>>
>>>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>>> Twitter: @FreeSWITCH
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>
>>>
>>> _________________________________________________________________________
>>> 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
> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>
>
>
> *Twitter: @FreeSWITCH , @briankwest*
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
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From jungleboogie0 at gmail.com  Fri Dec 12 03:33:19 2014
From: jungleboogie0 at gmail.com (jungle Boogie)
Date: Thu, 11 Dec 2014 16:33:19 -0800
Subject: [Freeswitch-users] TLS versions and PFS settings
In-Reply-To: <CAMeZp61VcqxJoHZArqw1UF0yZr2ocy0dJaWKuuKrFVZxGVR5GQ@mail.gmail.com>
References: <CAMeZp6139f7SAAAxwP9i3K5VRpLwWSGRHK4pYoOaEfEvbE9_oQ@mail.gmail.com>
	<CAEJMVkBHAgp-YXF_d99StVmQ5xnncm-zDUa9uWXsyV=g27YxVw@mail.gmail.com>
	<CAMeZp61VcqxJoHZArqw1UF0yZr2ocy0dJaWKuuKrFVZxGVR5GQ@mail.gmail.com>
Message-ID: <CAKE2PDvMrg0uva15hBeaObDCJ8Sopc7LVagERAvnPziyNMetrA@mail.gmail.com>

Hi Daniel,
On 13 November 2014 at 12:15, Daniel Ivanov <sertys at gmail.com> wrote:
>
> Thanks, Brian. Precise as always. And yet how do i enable perfect forward
> secrecy?
>

Do you get PFS to work for you?


>  13 ????. 2014 ?. 18:20 ???????????? "Brian West" <brian at freeswitch.org>
> ???????:
>
> in a sofia profile:
>>
>>     <!-- TLS version default: tlsv1,tlsv1.1,tlsv1.2 -->
>>
>>
>>
>>     <param name="tls-version" value="$${sip_tls_version}"/>
>>
>>
>>
>>
>>
>>
>>
>>     <!-- TLS ciphers default: ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH  -->
>>
>>
>>
>>     <param name="tls-ciphers" value="$${sip_tls_ciphers}"/>
>>
>>
>>
>>
>>
>>
>>
>> On Thu, Nov 13, 2014 at 8:55 AM, Daniel Ivanov <sertys at gmail.com> wrote:
>>
>>> What are the options to enable PFS on TLS support, i found the ability
>>> added in a changelog from feb,2014, but can't find the corresponding config
>>> file params.
>>> Also can i specify a list of allowed ciphers?
>>>
>>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>> --
-------
inum: 883510009027723
sip: jungleboogie at sip2sip.info
xmpp: jungle-boogie at jit.si
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From akhilgarg7 at gmail.com  Fri Dec 12 09:29:39 2014
From: akhilgarg7 at gmail.com (akhil garg)
Date: Fri, 12 Dec 2014 11:59:39 +0530
Subject: [Freeswitch-users] fs_cli hangs
In-Reply-To: <CAO3hyzGv7xbjDBQ+ZXkGAnau-YMkFW7oKBVGPXg1-iqZxBqDCQ@mail.gmail.com>
References: <CA+mgNr5S9_L6LcfqUwYhMfFJOpRxn3uDCitM_cwNQ7h7LuODhw@mail.gmail.com>
	<CALR8eWsO6HK2ghd0cyXuVLMHF5BOBCcqP5uLX_Ey1UArS3YSrg@mail.gmail.com>
	<CAO3hyzGv7xbjDBQ+ZXkGAnau-YMkFW7oKBVGPXg1-iqZxBqDCQ@mail.gmail.com>
Message-ID: <CA+mgNr5YZzc=HmjEY148NywzVF8utwaeVVyqarvHdBa11ypjuA@mail.gmail.com>

pastebin: 23716


regards,
akhil

On Fri, Dec 12, 2014 at 12:24 AM, Chris Tunbridge <blasterjr at gmail.com>
wrote:
>
> Can you do a pastebin of the contents of your
> conf/autoload_configs/event_socket.conf.xml
>
>
> On Thu, Dec 11, 2014 at 10:10 AM, Sudhanshu <sudhansh at gmail.com> wrote:
>
>> Enable debug logging and then check freeswitch.log.
>> What happens when you start freeswitch as anormal process (and not as a
>> daemon)?
>>
>> --
>> Sudhanshu
>>
>> On Thu, Dec 11, 2014 at 2:30 PM, akhil garg <akhilgarg7 at gmail.com> wrote:
>>
>>> running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but
>>> no success.
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> OUTPUT 1:
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /root/.fs_cli_conf.
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /etc/fs_cli.conf.
>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using
>>> builtin profile
>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>>
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> OUTPUT 2:
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /root/.fs_cli_conf.
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /etc/fs_cli.conf.
>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using
>>> builtin profile
>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>>> [auth/request]
>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>>> Event-Name: SOCKET_DATA
>>> Content-Type: auth/request
>>>
>>>
>>> [DEBUG] esl.c:1465 esl_send() SEND
>>> auth ClueCon
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>>
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> OUTPUT 3:
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /root/.fs_cli_conf.
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /etc/fs_cli.conf.
>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using
>>> builtin profile
>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>>> [auth/request]
>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>>> Event-Name: SOCKET_DATA
>>> Content-Type: auth/request
>>>
>>>
>>> [DEBUG] esl.c:1465 esl_send() SEND
>>> auth ClueCon
>>>
>>>
>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>>> [command/reply]
>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK
>>> accepted]
>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>>> Event-Name: SOCKET_DATA
>>> Content-Type: command/reply
>>> Reply-Text: +OK accepted
>>>
>>>
>>> [DEBUG] esl.c:1465 esl_send() SEND
>>> log
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>>
>>>
>>>
>>> --
>>> regards,
>>> akhil
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>

-- 
regards,
akhil
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From akhilgarg7 at gmail.com  Fri Dec 12 09:36:30 2014
From: akhilgarg7 at gmail.com (akhil garg)
Date: Fri, 12 Dec 2014 12:06:30 +0530
Subject: [Freeswitch-users] fs_cli hangs
In-Reply-To: <CALR8eWsO6HK2ghd0cyXuVLMHF5BOBCcqP5uLX_Ey1UArS3YSrg@mail.gmail.com>
References: <CA+mgNr5S9_L6LcfqUwYhMfFJOpRxn3uDCitM_cwNQ7h7LuODhw@mail.gmail.com>
	<CALR8eWsO6HK2ghd0cyXuVLMHF5BOBCcqP5uLX_Ey1UArS3YSrg@mail.gmail.com>
Message-ID: <CA+mgNr6VK4F5QoOoqgVAGX=OmY6iMOAj69X_QacCnhj5dMiH9A@mail.gmail.com>

Please see the logs: pastebin : 23717



regards,
akhil

On Thu, Dec 11, 2014 at 10:40 PM, Sudhanshu <sudhansh at gmail.com> wrote:
>
> Enable debug logging and then check freeswitch.log.
> What happens when you start freeswitch as anormal process (and not as a
> daemon)?
>
> --
> Sudhanshu
>
> On Thu, Dec 11, 2014 at 2:30 PM, akhil garg <akhilgarg7 at gmail.com> wrote:
>
>> running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but no
>> success.
>>
>>
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> OUTPUT 1:
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /root/.fs_cli_conf.
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /etc/fs_cli.conf.
>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
>> profile
>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> OUTPUT 2:
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /root/.fs_cli_conf.
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /etc/fs_cli.conf.
>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
>> profile
>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>> [auth/request]
>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>> Event-Name: SOCKET_DATA
>> Content-Type: auth/request
>>
>>
>> [DEBUG] esl.c:1465 esl_send() SEND
>> auth ClueCon
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>
>>
>>
>>
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> OUTPUT 3:
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /root/.fs_cli_conf.
>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>> /etc/fs_cli.conf.
>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin
>> profile
>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>> [auth/request]
>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>> Event-Name: SOCKET_DATA
>> Content-Type: auth/request
>>
>>
>> [DEBUG] esl.c:1465 esl_send() SEND
>> auth ClueCon
>>
>>
>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>> [command/reply]
>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK
>> accepted]
>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>> Event-Name: SOCKET_DATA
>> Content-Type: command/reply
>> Reply-Text: +OK accepted
>>
>>
>> [DEBUG] esl.c:1465 esl_send() SEND
>> log
>>
>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>
>>
>>
>> --
>> regards,
>> akhil
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>

-- 
regards,
akhil
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From bordmi at rarus.ru  Fri Dec 12 09:59:01 2014
From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=)
Date: Fri, 12 Dec 2014 10:59:01 +0400
Subject: [Freeswitch-users] Dialplan - Lua in-line expansion
In-Reply-To: <CAGK4cYxP8hepPN0Pi+Um2KGJvjRmQkQqOW5wQ3=qK=oTE6dQ6A@mail.gmail.com>
References: <CAGK4cYxy4DTZVEGWprdkBB8JD9RY1fC2TD81HKB9pHYaDBTJGQ@mail.gmail.com>
	<D133FCCA6BEB4C2B9A4383C689309413@gmail.com>
	<CAGK4cYxP8hepPN0Pi+Um2KGJvjRmQkQqOW5wQ3=qK=oTE6dQ6A@mail.gmail.com>
Message-ID: <CAL2h29jGCY6_93EDwSDa3YDBZnt1Oo=0PnRb45aXcosh7qe+Og@mail.gmail.com>

change
return result
to
io.write(result)

2014-12-12 0:32 GMT+03:00 Phillip Jones <pjintheusa at gmail.com>:

> Yeah - thanks - I realize I can do that - but it just drives me crazy I
> can not get in-line expansion to work. I can't see what I am missing.
>
> On Thu, Dec 11, 2014 at 3:46 PM, Chad Phillips <chad at apartmentlines.com>
> wrote:
>
>>  how about setting the channel var directly in the script?
>>
>> session:setVariable("VUM_SAY_CID", result);
>>
>> although for what you?re doing, i really think there?s a cleaner way to
>> do it straight from the dialplan XML...
>>
>> On Thursday, December 11, 2014 at 1:35 PM, Phillip Jones wrote:
>>
>> I am going crazy - please tell me what I am missing:
>>
>> I have a dialplan:
>>
>> <action application="set" data="VUM_SAY_CID=${lua(util_get10digits.lua
>> ${VUM_Outgoing_CallerID_Number})}" inline="true"/>
>>
>>
>> I have a lua script:
>>
>>
>> freeswitch.consoleLog("NOTICE","util_get10digits.lua \n")
>> freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n")
>> local result = "none"
>> *result = string.sub(argv[1], -10)*
>> freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n")
>> *return result*
>>
>>
>> I run the dialplan:
>>
>> Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 Action
>> set(VUM_SAY_CID=${lua(util_get10digits.lua
>> ${VUM_Outgoing_CallerID_Number})}) I
>> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328
>> util_get10digits.lua
>> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed
>> [+12154791697]
>> *2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result:
>> [2154791697 <%5B2154791697>]*
>> 2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/
>> 2154791697 at 64.158.162.74 destroy/unlink session from object
>> EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 set(VUM_SAY_CID=)
>> *2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435
>> sofia/Carrier/2154791697 at 64.158.162.74 <2154791697 at 64.158.162.74> SET
>> [VUM_SAY_CID]=[UNDEF]*
>>
>>
>> The lua script runs correctly - how ever the result is not returned the
>> VUM_SAY_CID variable in the dialplan
>>
>>
>> The work around is to set the session variable in the script. But I
>> wondering what I am missing. Is "return result" not how you return a result
>> from Lua?
>>
>>
>> Thanks for any help.
>>
>>
>> Phil
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 best regards,
Dmitriy Borisov
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From bordmi at rarus.ru  Fri Dec 12 11:52:38 2014
From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=)
Date: Fri, 12 Dec 2014 12:52:38 +0400
Subject: [Freeswitch-users] Dialplan - Lua in-line expansion
In-Reply-To: <CAL2h29jGCY6_93EDwSDa3YDBZnt1Oo=0PnRb45aXcosh7qe+Og@mail.gmail.com>
References: <CAGK4cYxy4DTZVEGWprdkBB8JD9RY1fC2TD81HKB9pHYaDBTJGQ@mail.gmail.com>
	<D133FCCA6BEB4C2B9A4383C689309413@gmail.com>
	<CAGK4cYxP8hepPN0Pi+Um2KGJvjRmQkQqOW5wQ3=qK=oTE6dQ6A@mail.gmail.com>
	<CAL2h29jGCY6_93EDwSDa3YDBZnt1Oo=0PnRb45aXcosh7qe+Og@mail.gmail.com>
Message-ID: <CAL2h29iFfcKG3wB5Mm58C-fyUJs9q+iMQJkrh+GuZDhn2haL8g@mail.gmail.com>

Oh, sorry, not *io.write(result)* but *stream:write(result)*

2014-12-12 9:59 GMT+03:00 ???????, ??????? <bordmi at rarus.ru>:

> change
> return result
> to
> io.write(result)
>
> 2014-12-12 0:32 GMT+03:00 Phillip Jones <pjintheusa at gmail.com>:
>
>> Yeah - thanks - I realize I can do that - but it just drives me crazy I
>> can not get in-line expansion to work. I can't see what I am missing.
>>
>> On Thu, Dec 11, 2014 at 3:46 PM, Chad Phillips <chad at apartmentlines.com>
>> wrote:
>>
>>>  how about setting the channel var directly in the script?
>>>
>>> session:setVariable("VUM_SAY_CID", result);
>>>
>>> although for what you?re doing, i really think there?s a cleaner way to
>>> do it straight from the dialplan XML...
>>>
>>> On Thursday, December 11, 2014 at 1:35 PM, Phillip Jones wrote:
>>>
>>> I am going crazy - please tell me what I am missing:
>>>
>>> I have a dialplan:
>>>
>>> <action application="set" data="VUM_SAY_CID=${lua(util_get10digits.lua
>>> ${VUM_Outgoing_CallerID_Number})}" inline="true"/>
>>>
>>>
>>> I have a lua script:
>>>
>>>
>>> freeswitch.consoleLog("NOTICE","util_get10digits.lua \n")
>>> freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n")
>>> local result = "none"
>>> *result = string.sub(argv[1], -10)*
>>> freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n")
>>> *return result*
>>>
>>>
>>> I run the dialplan:
>>>
>>> Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 Action
>>> set(VUM_SAY_CID=${lua(util_get10digits.lua
>>> ${VUM_Outgoing_CallerID_Number})}) I
>>> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328
>>> util_get10digits.lua
>>> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed
>>> [+12154791697]
>>> *2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result:
>>> [2154791697 <%5B2154791697>]*
>>> 2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/
>>> 2154791697 at 64.158.162.74 destroy/unlink session from object
>>> EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 set(VUM_SAY_CID=)
>>> *2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435
>>> sofia/Carrier/2154791697 at 64.158.162.74 <2154791697 at 64.158.162.74> SET
>>> [VUM_SAY_CID]=[UNDEF]*
>>>
>>>
>>> The lua script runs correctly - how ever the result is not returned the
>>> VUM_SAY_CID variable in the dialplan
>>>
>>>
>>> The work around is to set the session variable in the script. But I
>>> wondering what I am missing. Is "return result" not how you return a result
>>> from Lua?
>>>
>>>
>>> Thanks for any help.
>>>
>>>
>>> Phil
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 best regards,
> Dmitriy Borisov
>



-- 
with best regards,
Dmitriy Borisov
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From idokan at gmail.com  Fri Dec 12 13:22:59 2014
From: idokan at gmail.com (ik)
Date: Fri, 12 Dec 2014 12:22:59 +0200
Subject: [Freeswitch-users] Controling playback recording position
In-Reply-To: <CAEJMVkDxgrB9DbCvMAH4ZJTTpS7kJ7qoBwc9_kFJDprCtNHtTA@mail.gmail.com>
References: <CAOccsKDYbd4NmE5CKzZEoeEz++-5Qmi6T_t33pMprY+yQKYJMQ@mail.gmail.com>
	<CAEJMVkDxgrB9DbCvMAH4ZJTTpS7kJ7qoBwc9_kFJDprCtNHtTA@mail.gmail.com>
Message-ID: <CAOccsKBq8xGW+NWUh3EQzKtK7YHvBM+i8fcmVamUeMCjqdRysA@mail.gmail.com>

That's very simple and easy
Thank you

On Thu, Dec 11, 2014 at 8:08 PM, Brian West <brian at freeswitch.org> wrote:
>
> See scripts/lua/callback.lua, javascript has a similar method too.
>
>
>
> On Thu, Dec 11, 2014 at 11:32 AM, ik <idokan at gmail.com> wrote:
>
>> Hello,
>>
>> How can I control the position of the recording that is played ?
>> For example, go 10 seconds forward or backward, like with Asterisk's
>> ControlPlayback cmd.
>>
>> Thanks,
>> Ido
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sertys at gmail.com  Fri Dec 12 15:33:58 2014
From: sertys at gmail.com (Daniel Ivanov)
Date: Fri, 12 Dec 2014 13:33:58 +0100
Subject: [Freeswitch-users] TLS versions and PFS settings
In-Reply-To: <CAKE2PDvMrg0uva15hBeaObDCJ8Sopc7LVagERAvnPziyNMetrA@mail.gmail.com>
References: <CAMeZp6139f7SAAAxwP9i3K5VRpLwWSGRHK4pYoOaEfEvbE9_oQ@mail.gmail.com>
	<CAEJMVkBHAgp-YXF_d99StVmQ5xnncm-zDUa9uWXsyV=g27YxVw@mail.gmail.com>
	<CAMeZp61VcqxJoHZArqw1UF0yZr2ocy0dJaWKuuKrFVZxGVR5GQ@mail.gmail.com>
	<CAKE2PDvMrg0uva15hBeaObDCJ8Sopc7LVagERAvnPziyNMetrA@mail.gmail.com>
Message-ID: <CAMeZp63b+F-YhY6kiVh4UpqE4iNapKejG+8Q=25ysE9N25DhOA@mail.gmail.com>

Unfortunately not. Still looking for a way to do it.

On Fri, Dec 12, 2014 at 1:33 AM, jungle Boogie <jungleboogie0 at gmail.com>
wrote:

> Hi Daniel,
> On 13 November 2014 at 12:15, Daniel Ivanov <sertys at gmail.com> wrote:
>>
>> Thanks, Brian. Precise as always. And yet how do i enable perfect forward
>> secrecy?
>>
>
> Do you get PFS to work for you?
>
>
>>  13 ????. 2014 ?. 18:20 ???????????? "Brian West" <brian at freeswitch.org>
>> ???????:
>>
>> in a sofia profile:
>>>
>>>     <!-- TLS version default: tlsv1,tlsv1.1,tlsv1.2 -->
>>>
>>>
>>>
>>>     <param name="tls-version" value="$${sip_tls_version}"/>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>     <!-- TLS ciphers default: ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH  -->
>>>
>>>
>>>
>>>     <param name="tls-ciphers" value="$${sip_tls_ciphers}"/>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Thu, Nov 13, 2014 at 8:55 AM, Daniel Ivanov <sertys at gmail.com> wrote:
>>>
>>>> What are the options to enable PFS on TLS support, i found the ability
>>>> added in a changelog from feb,2014, but can't find the corresponding config
>>>> file params.
>>>> Also can i specify a list of allowed ciphers?
>>>>
>>>>
>>> --
>>>
>>> *Brian West*
>>> brian at freeswitch.org
>>>
>>> --
> -------
> 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 aqsyounas at gmail.com  Fri Dec 12 15:39:30 2014
From: aqsyounas at gmail.com (Aqs Younas)
Date: Fri, 12 Dec 2014 17:39:30 +0500
Subject: [Freeswitch-users] Giving up resync after 1024 bytes
Message-ID: <CAGVOjVWTS508DVuToCQUAVPurx44PeTfaw4Buc7O1jSdbJ25Qw@mail.gmail.com>

Hi, All

I am newbie to freeswitch, while playing multiple stream with mod_shout my
freeswitch logs flood with this type of errors.

[layer3.c:454] error: big_values too large!
[layer3.c:454] error: big_values too large!
Note: Illegal Audio-MPEG-Header 0xbbb7df0d at offset 19830.
Note: Trying to resync...
Note: Hit end of (available) data during resync.
2014-12-12 12:35:42.564619 [WARNING] switch_core_file.c:230 File has 2
channels, muxing to 1 channel will occur.
Note: Illegal Audio-MPEG-Header 0xbbb7df0d at offset 19830.
Note: Trying to resync...
Note: Skipped 1024 bytes in input.
[parse.c:682] error: Giving up resync after 1024 bytes - your stream is not
nice... (maybe increasing resync limit could help).
Note: Illegal Audio-MPEG-Header 0xef0000a8 at offset 20857.
Note: Trying to resync...
Note: Skipped 64 bytes in input.

Could someone please help me in understanding or solving this problem.

Thanks
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From thomas.granvej6 at gmail.com  Fri Dec 12 16:04:24 2014
From: thomas.granvej6 at gmail.com (=?UTF-8?Q?Thomas_L=C3=B8cke?=)
Date: Fri, 12 Dec 2014 14:04:24 +0100
Subject: [Freeswitch-users] FreeBSD or Debian
Message-ID: <CAHJ5xPmRn4v_tt+VTbJpKgV-JXJojY8Ftbgaff=mCumA7MdezQ@mail.gmail.com>

Hey all,

I'm involved in a project where we'll soon need to spin up a couple of
FreeSWITCH instances, and we're a bit torn between Debian and FreeBSD.

Anybody out there with experience with both? Are there any differences?
Will Debian be "better" due to FreeSWITCH primarily targeting Linux?

We're very slightly leaning towards FreeBSD due to jails, ZFS and other
nice FreeBSD tools, but we're not averse to Debian in any way. We're
probably at 52% FreeBSD and 48% Debian.

The instances will run on hardware with decent specs. No less than 16 cores
(probably 32) and no less than 32GB RAM and SSD in every single bay. We're
not targeting low spec.

Any and all advice is welcomed with open arms.

:o)
Thomas L?cke
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From karl-theo_hofer at inteli-sim.com  Fri Dec 12 16:42:00 2014
From: karl-theo_hofer at inteli-sim.com (kthofer)
Date: Fri, 12 Dec 2014 14:42:00 +0100
Subject: [Freeswitch-users] mkdir not working
Message-ID: <548AF0A8.90909@inteli-sim.com>

Hi there
we try to create a new directory
  $con->execute("mkdir", "$vmail_base");
to store some Voice mail files

But for some reason FS is not even trying to create the directory
we checked the user rights
the spelling
but still nothing not even in the linux logs we can not see that FS is 
trying it atall.

Any suggestions??
what is going wrong?
Please let me know if you have a solution or workaround.

-- 
With best regards

Karl Theo Hofer



From alipey at gmail.com  Fri Dec 12 17:54:07 2014
From: alipey at gmail.com (Ali Pey)
Date: Fri, 12 Dec 2014 09:54:07 -0500
Subject: [Freeswitch-users] Record sound quality not good but pcap is quite
	good
Message-ID: <CA+q4kR+wJ=CBQ97FThSU6BuQPM0cSZmrntkyic8s8KPtyXo2iA@mail.gmail.com>

Hello,

I use record to record part of a call and sometimes I get some static
noises and distortion; however when I listen to the call on pcap capture,
it's quite clear and no noise.

I record to a wav file and then use sox to convert it to ulaw.

Any suggestions? Any help would be greatly appreciated.


Thanks,
Ali Pey
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From bordmi at rarus.ru  Fri Dec 12 17:55:48 2014
From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5IC8gRG1pdHJpeSBCb3Jpc292?=)
Date: Fri, 12 Dec 2014 18:55:48 +0400
Subject: [Freeswitch-users] FreeBSD or Debian
In-Reply-To: <CAHJ5xPmRn4v_tt+VTbJpKgV-JXJojY8Ftbgaff=mCumA7MdezQ@mail.gmail.com>
References: <CAHJ5xPmRn4v_tt+VTbJpKgV-JXJojY8Ftbgaff=mCumA7MdezQ@mail.gmail.com>
Message-ID: <CAL2h29jPvZKAmFYG3WMLJ1YMU+Pk6rQQ=hS5mdDK=cyWG-t05Q@mail.gmail.com>

We have test platform on FreeBSD and working platform on Debian. I found
only one pricipial difference between platforms: on Debian we have native
Corosync/Pacemaker, which are the best tool for clustering until
mod_ha_cluster will released.

But our platform is not working under huge load, we are not released our
prudct still.

2014-12-12 16:04 GMT+03:00 Thomas L?cke <thomas.granvej6 at gmail.com>:

> Hey all,
>
> I'm involved in a project where we'll soon need to spin up a couple of
> FreeSWITCH instances, and we're a bit torn between Debian and FreeBSD.
>
> Anybody out there with experience with both? Are there any differences?
> Will Debian be "better" due to FreeSWITCH primarily targeting Linux?
>
> We're very slightly leaning towards FreeBSD due to jails, ZFS and other
> nice FreeBSD tools, but we're not averse to Debian in any way. We're
> probably at 52% FreeBSD and 48% Debian.
>
> The instances will run on hardware with decent specs. No less than 16
> cores (probably 32) and no less than 32GB RAM and SSD in every single bay.
> We're not targeting low spec.
>
> Any and all advice is welcomed with open arms.
>
> :o)
> Thomas L?cke
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 best regards,
Dmitriy Borisov
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From krice at freeswitch.org  Fri Dec 12 18:02:24 2014
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 12 Dec 2014 15:02:24 +0000
Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder!
Message-ID: <548b0380440a6_363a80b33046726@ip-10-155-242-219.mail>

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All
Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info!
-- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon
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From mbsip at gazeta.pl  Fri Dec 12 18:07:14 2014
From: mbsip at gazeta.pl (Maciej Bylica)
Date: Fri, 12 Dec 2014 16:07:14 +0100
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call
	request
In-Reply-To: <CACynCY6gSx3v6c--Pocss1do=5tRF-fpwVxsg5NC5Pxu87gyFw@mail.gmail.com>
References: <CAFiqYum79cR0Tfa-Guo+R+3YruNUTm_jeZAsf4WsL7SZMB6E=w@mail.gmail.com>
	<D0AF1E70.8F750%krice@freeswitch.org>
	<CACynCY473-rhBSU4ftwN=HzK1KJai8R-nAwwKnWhCLzb4c6ZyQ@mail.gmail.com>
	<CAKbxfG9HOnFwhz3F_54kKVC4xWZfWfwyOtpQb8pjSk3y0_Gosg@mail.gmail.com>
	<CACynCY7=0R3PZ7LTRWtf+nPRhRT7YAXJePemjn01Jb-B+-EOzQ@mail.gmail.com>
	<CAEJMVkBjD-vvVCDuKEyYj9OuNSa5P3BorXkQUGu8PTBYZ5q2Qg@mail.gmail.com>
	<CACynCY6gSx3v6c--Pocss1do=5tRF-fpwVxsg5NC5Pxu87gyFw@mail.gmail.com>
Message-ID: <CACynCY7Wbi6V-9XXc-c+Dz4xV5jh8B-ZPC8K7eydioJtZWhkXA@mail.gmail.com>

   1. FS-7086 <https://freeswitch.org/jira/browse/FS-7086>


2014-12-12 1:00 GMT+01:00 Maciej Bylica <mbsip at gazeta.pl>:

> I will gather packet capture and file JIRA.
>
> Thank You
> Maciej.
>
>
> Funny, payload 13 is CN, which is 0x13 in base16 or 19 in base10, Please
>> file a JIRA, I would need to see a packet capture for this one.
>>
>>
>> On Thu, Dec 11, 2014 at 11:47 AM, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>
>>> Sure i'll try with jira
>>>
>>> Mac.
>>>
>>> 2014-12-11 17:41 GMT+01:00 Anthony Minessale <
>>> anthony.minessale at gmail.com>:
>>>
>>>> For years I beg ppl to post issues to jira not the mailing list.
>>>> if you start typing and keywords like "problem" "not working" "issue"
>>>> appear in your text, you probably should be filing it to JIRA not here.
>>>>
>>>>
>>>> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica <mbsip at gazeta.pl>
>>>> wrote:
>>>>
>>>>> Thanks for prompt reply.
>>>>>
>>>>> Steven, later today i will try to update FS to the latest git and
>>>>> check this out.
>>>>>
>>>>> Ken, i am fine with CN, passing thru is what i really need, but the
>>>>> question is why FS generates such ERRors and how to get out of this.
>>>>>
>>>>> Thanks
>>>>> Mac.
>>>>>
>>>>> 2014-12-11 17:12 GMT+01:00 Ken Rice <krice at freeswitch.org>:
>>>>>
>>>>>>  CN is comfort noise... Your cisco is sending it, and freeswitch is
>>>>>> passing it thru because you have told it to use proxy media mode...
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 12/11/14 9:53 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>>>>>
>>>>>> Your version is 3 months old. Can you reproduce this on the latest
>>>>>> git head?
>>>>>>
>>>>>> On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> I am running FreeSWITCH Version
>>>>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
>>>>>> problem with some call requests setup.
>>>>>> Here is one of them:
>>>>>>
>>>>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
>>>>>> sofia/outside_1/20049112223344 []
>>>>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
>>>>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
>>>>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send
>>>>>> signal sofia/outside_1/20049112223344 [BREAK]
>>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
>>>>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT
>>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
>>>>>> (sofia/outside_1/20049112223344) State INIT
>>>>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
>>>>>> sofia/outside_1/20049112223344 SOFIA INIT
>>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
>>>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>>>  ---
>>>>>>  v=0
>>>>>>  o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>>>>>>  s=SIP Call
>>>>>>  c=IN IP4 10.10.10.12
>>>>>>  t=0 0
>>>>>>  m=audio 24782 RTP/AVP 18 19
>>>>>>  c=IN IP4 10.10.10.12
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>
>>>>>>  +++
>>>>>>  v=0
>>>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>>>  s=FreeSWITCH
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  t=0 0
>>>>>>  m=audio 19690 RTP/AVP 18 19
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>
>>>>>>  2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
>>>>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
>>>>>> 2014-09-17 23:11:20Z 64bit
>>>>>>  Local SDP:
>>>>>>  v=0
>>>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>>>  s=FreeSWITCH
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  t=0 0
>>>>>>  m=audio 19690 RTP/AVP 18 19
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>
>>>>>> sending out SDP exactly as shown above.
>>>>>> The other side is responsing with 100, then 180 Ringing wo/SDP then
>>>>>> 183 w/SDP
>>>>>>
>>>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
>>>>>> sofia/outside_1/20049112223344 entering state [proceeding][183]
>>>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>>>>>>  v=0
>>>>>>  o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>>>>>>  s=Dialogic-SIP
>>>>>>  c=IN IP4 10.10.10.198
>>>>>>  t=0 0
>>>>>>  m=audio 10024 RTP/AVP 19 18
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=silenceSupp:off - - - -
>>>>>>  a=ptime:20
>>>>>>
>>>>>>  2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
>>>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>>>  ---
>>>>>>  v=0
>>>>>>  o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>>>>>>  s=FreeSWITCH
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  t=0 0
>>>>>>  m=audio 29492 RTP/AVP 18 19
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>
>>>>>>  +++
>>>>>>  v=0
>>>>>>  o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>>>>>>  s=FreeSWITCH
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  t=0 0
>>>>>>  m=audio 29492 RTP/AVP 18 19
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>
>>>>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651
>>>>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR]
>>>>>> switch_core_media.c:2294
>>>>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE]
>>>>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344
>>>>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
>>>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
>>>>>> sofia/outside_1/20049112223344 [KILL]
>>>>>>
>>>>>> and Freeswitch is terminating the call by using CANCEL.
>>>>>>
>>>>>> My modules.conf.xml config part looks like following:
>>>>>>     <!-- Codec Interfaces -->
>>>>>>     <load module="mod_spandsp"/>
>>>>>>     <load module="mod_g723_1"/>
>>>>>>     <load module="mod_g729"/>
>>>>>>     <!--load module="mod_amr"/-->
>>>>>>     <!--<load module="mod_ilbc"/>-->
>>>>>>     <load module="mod_h26x"/>
>>>>>>     <!--load module="mod_vp8"/-->
>>>>>>     <!--load module="mod_b64"/-->
>>>>>>     <!--<load module="mod_siren"/>-->
>>>>>>     <!--<load module="mod_isac"/>-->
>>>>>>     <!--<load module="mod_celt"/>-->
>>>>>>     <!--load module="mod_opus"/-->
>>>>>>
>>>>>> Is it a problem with CN definition?
>>>>>> Could you please lead me where the problem is located?
>>>>>>
>>>>>> Thanks in advance.
>>>>>> Mac.
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> ------------------------------
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Ken
>>>>>>
>>>>>>
>>>>>>
>>>>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>>>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>>>> Twitter: @FreeSWITCH
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>
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From norm at goes.com  Fri Dec 12 17:07:59 2014
From: norm at goes.com (Norm)
Date: Fri, 12 Dec 2014 09:07:59 -0500
Subject: [Freeswitch-users] FreeBSD or Debian
In-Reply-To: <CAHJ5xPmRn4v_tt+VTbJpKgV-JXJojY8Ftbgaff=mCumA7MdezQ@mail.gmail.com>
References: <CAHJ5xPmRn4v_tt+VTbJpKgV-JXJojY8Ftbgaff=mCumA7MdezQ@mail.gmail.com>
Message-ID: <70910985-9657-41DF-93DA-58AD7D4B862D@goes.com>

Debian

> On Dec 12, 2014, at 8:04 AM, Thomas L?cke <thomas.granvej6 at gmail.com> wrote:
> 
> Hey all,
> 
> I'm involved in a project where we'll soon need to spin up a couple of FreeSWITCH instances, and we're a bit torn between Debian and FreeBSD.
> 
> Anybody out there with experience with both? Are there any differences? Will Debian be "better" due to FreeSWITCH primarily targeting Linux?
> 
> We're very slightly leaning towards FreeBSD due to jails, ZFS and other nice FreeBSD tools, but we're not averse to Debian in any way. We're probably at 52% FreeBSD and 48% Debian.
> 
> The instances will run on hardware with decent specs. No less than 16 cores (probably 32) and no less than 32GB RAM and SSD in every single bay. We're not targeting low spec.
> 
> Any and all advice is welcomed with open arms.
> 
> :o)
> Thomas L?cke
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 michel.brabants at gmail.com  Fri Dec 12 18:23:56 2014
From: michel.brabants at gmail.com (Michel Brabants)
Date: Fri, 12 Dec 2014 16:23:56 +0100
Subject: [Freeswitch-users] freeswitch 1.4 and encryption/rtcp-mux
Message-ID: <CAHt+dNE8JNjbCRFMpJ0q1mwsFWUJ5-oMr7eFqNAohPq7kce7MA@mail.gmail.com>

Hello,

I recently started upgrading to FS 1.4, but I encountered 2 difficulties of
which I'm still looking into one:

 1) DTLS-configuration seems to be required, although we don't use it
currently. We use normal sip-profiles (no webrtc). The option to disable
it, is "webrtc_enable_dtls=false", which can b set in the dialplan. But why
is it trying to enable it by default? Can you disable it also in a profile?

2) Also a change because of webrtc seemingly. When receiving an invite
(without sdp - 3pcc-request), freeswitch in the end response in its 200 OK
with a rtcp-mux-line in its sdp. We don't want rtcp-mux, just rtp-port+1
for rtcp. When looking at the code, I don't currently know why FS sends
back the myx-parameter as it seems only enabled when the other ends
proposes it or am I missing something?

Thanks,

Michel
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From grcamauer at gmail.com  Fri Dec 12 18:34:05 2014
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Fri, 12 Dec 2014 12:34:05 -0300
Subject: [Freeswitch-users] FreeBSD or Debian
In-Reply-To: <70910985-9657-41DF-93DA-58AD7D4B862D@goes.com>
References: <CAHJ5xPmRn4v_tt+VTbJpKgV-JXJojY8Ftbgaff=mCumA7MdezQ@mail.gmail.com>
	<70910985-9657-41DF-93DA-58AD7D4B862D@goes.com>
Message-ID: <CAPNTnnr3NAw6Wk0-yEJ-8M3w_BuqpqCNzRj7GZjy12sZu4E6CQ@mail.gmail.com>

I tend to use machines with ZFS only as dedicated file servers.  ZFS is a
great file system, but it takes over the machine and uses as much memory
and resources as it can get it's hands on.  I would not use it for
FreeSwitch if there will be a lot of file-io going on.

Guillermo

On Fri, Dec 12, 2014 at 11:07 AM, Norm <norm at goes.com> wrote:

> Debian
>
> > On Dec 12, 2014, at 8:04 AM, Thomas L?cke <thomas.granvej6 at gmail.com>
> wrote:
> >
> > Hey all,
> >
> > I'm involved in a project where we'll soon need to spin up a couple of
> FreeSWITCH instances, and we're a bit torn between Debian and FreeBSD.
> >
> > Anybody out there with experience with both? Are there any differences?
> Will Debian be "better" due to FreeSWITCH primarily targeting Linux?
> >
> > We're very slightly leaning towards FreeBSD due to jails, ZFS and other
> nice FreeBSD tools, but we're not averse to Debian in any way. We're
> probably at 52% FreeBSD and 48% Debian.
> >
> > The instances will run on hardware with decent specs. No less than 16
> cores (probably 32) and no less than 32GB RAM and SSD in every single bay.
> We're not targeting low spec.
> >
> > Any and all advice is welcomed with open arms.
> >
> > :o)
> > Thomas L?cke
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://confluence.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 pjintheusa at gmail.com  Fri Dec 12 18:44:09 2014
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 12 Dec 2014 10:44:09 -0500
Subject: [Freeswitch-users] Dialplan - Lua in-line expansion
In-Reply-To: <CAL2h29iFfcKG3wB5Mm58C-fyUJs9q+iMQJkrh+GuZDhn2haL8g@mail.gmail.com>
References: <CAGK4cYxy4DTZVEGWprdkBB8JD9RY1fC2TD81HKB9pHYaDBTJGQ@mail.gmail.com>
	<D133FCCA6BEB4C2B9A4383C689309413@gmail.com>
	<CAGK4cYxP8hepPN0Pi+Um2KGJvjRmQkQqOW5wQ3=qK=oTE6dQ6A@mail.gmail.com>
	<CAL2h29jGCY6_93EDwSDa3YDBZnt1Oo=0PnRb45aXcosh7qe+Og@mail.gmail.com>
	<CAL2h29iFfcKG3wB5Mm58C-fyUJs9q+iMQJkrh+GuZDhn2haL8g@mail.gmail.com>
Message-ID: <CAGK4cYz3NqLk107XBDVPjMnRSEskP3pR5=5fT-2qRvzjXjBx2g@mail.gmail.com>

Perfect - thanks very much.

On Fri, Dec 12, 2014 at 3:52 AM, ???????, ??????? <bordmi at rarus.ru> wrote:
>
> Oh, sorry, not *io.write(result)* but *stream:write(result)*
>
> 2014-12-12 9:59 GMT+03:00 ???????, ??????? <bordmi at rarus.ru>:
>
>> change
>> return result
>> to
>> io.write(result)
>>
>> 2014-12-12 0:32 GMT+03:00 Phillip Jones <pjintheusa at gmail.com>:
>>
>>> Yeah - thanks - I realize I can do that - but it just drives me crazy I
>>> can not get in-line expansion to work. I can't see what I am missing.
>>>
>>> On Thu, Dec 11, 2014 at 3:46 PM, Chad Phillips <chad at apartmentlines.com>
>>> wrote:
>>>
>>>>  how about setting the channel var directly in the script?
>>>>
>>>> session:setVariable("VUM_SAY_CID", result);
>>>>
>>>> although for what you?re doing, i really think there?s a cleaner way to
>>>> do it straight from the dialplan XML...
>>>>
>>>> On Thursday, December 11, 2014 at 1:35 PM, Phillip Jones wrote:
>>>>
>>>> I am going crazy - please tell me what I am missing:
>>>>
>>>> I have a dialplan:
>>>>
>>>> <action application="set" data="VUM_SAY_CID=${lua(util_get10digits.lua
>>>> ${VUM_Outgoing_CallerID_Number})}" inline="true"/>
>>>>
>>>>
>>>> I have a lua script:
>>>>
>>>>
>>>> freeswitch.consoleLog("NOTICE","util_get10digits.lua \n")
>>>> freeswitch.consoleLog("NOTICE","Argument passed [" .. argv[1] .."]\n")
>>>> local result = "none"
>>>> *result = string.sub(argv[1], -10)*
>>>> freeswitch.consoleLog("NOTICE", "Result: [" .. result .."]\n")
>>>> *return result*
>>>>
>>>>
>>>> I run the dialplan:
>>>>
>>>> Dialplan: sofia/Carrier/2154791697 at 64.158.162.74 Action
>>>> set(VUM_SAY_CID=${lua(util_get10digits.lua
>>>> ${VUM_Outgoing_CallerID_Number})}) I
>>>> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328
>>>> util_get10digits.lua
>>>> 2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Argument passed
>>>> [+12154791697]
>>>> *2014-12-11 15:20:21.795274 [NOTICE] switch_cpp.cpp:1328 Result:
>>>> [2154791697 <%5B2154791697>]*
>>>> 2014-12-11 15:20:21.795274 [DEBUG] switch_cpp.cpp:1075 sofia/Carrier/
>>>> 2154791697 at 64.158.162.74 destroy/unlink session from object
>>>> EXECUTE sofia/Carrier/2154791697 at 64.158.162.74 set(VUM_SAY_CID=)
>>>> *2014-12-11 15:20:21.795274 [DEBUG] mod_dptools.c:1435
>>>> sofia/Carrier/2154791697 at 64.158.162.74 <2154791697 at 64.158.162.74> SET
>>>> [VUM_SAY_CID]=[UNDEF]*
>>>>
>>>>
>>>> The lua script runs correctly - how ever the result is not returned the
>>>> VUM_SAY_CID variable in the dialplan
>>>>
>>>>
>>>> The work around is to set the session variable in the script. But I
>>>> wondering what I am missing. Is "return result" not how you return a result
>>>> from Lua?
>>>>
>>>>
>>>> Thanks for any help.
>>>>
>>>>
>>>> Phil
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 best regards,
>> Dmitriy Borisov
>>
>
>
>
> --
> with best regards,
> Dmitriy Borisov
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From areski at gmail.com  Fri Dec 12 18:56:23 2014
From: areski at gmail.com (Areski)
Date: Fri, 12 Dec 2014 16:56:23 +0100
Subject: [Freeswitch-users] [Freeswitch-docs] Doc-Sprint Friday 12
 December 2014
In-Reply-To: <CAJR7f2FgvT4qHGU5RBJs5Ti78Hgtf2hfChRy_hQb0ZpZFUF+7Q@mail.gmail.com>
References: <CAMf7LBiAKzHdsML1mGoPPn-ykKd8Zu72m2JrwMfka_Aedx-BDA@mail.gmail.com>
	<CAHaUJVbNaOzCUiwYN-ka5OCY=OsY9M2Ri_zKwyoRKrGuoGVwgw@mail.gmail.com>
	<CA+fkitLK8w7h1a+kiQr=vQ-wt0oSGKaQ7q4HWhbep52FS65kMA@mail.gmail.com>
	<CAJR7f2FgvT4qHGU5RBJs5Ti78Hgtf2hfChRy_hQb0ZpZFUF+7Q@mail.gmail.com>
Message-ID: <CAMf7LBj-LKTGWcTox=AzJsYTJHW45JLXD23rsNFNQS2iOyT7bA@mail.gmail.com>

We will be starting in 5 minutes, come and join us.
IRC: #freeswitch-docs (freenode)

On Wed, Dec 10, 2014 at 12:47 PM, ?talo Rossi <italorossib at gmail.com> wrote:
>
> Confirmed so far:
>
> - Italo Rossi (+4)
> - Areski Belaid
> - Iwada Eja
> - Adolphe Cher-Aime
> - Bote
>
> Who else?
>
> On Fri, Dec 5, 2014 at 12:14 PM, Adolphe Cher-Aime <acheraime at gmail.com>
> wrote:
>
>> I'm in.
>>
>>
>> --
>> Adolphe
>>
>> On Thu, Dec 4, 2014 at 3:00 PM, Iwada Eja <iwada.bassey at gmail.com> wrote:
>>
>>> I'm In
>>>
>>> On Thu, Dec 4, 2014 at 1:20 PM, Areski <areski at gmail.com> wrote:
>>>
>>>> Hi everyone,
>>>>
>>>> We are planning to organize an other doc sprint on *Friday 12 December
>>>> at 10am CT*.
>>>> It will be 4 hours long but you can join for less time.
>>>>
>>>> The Doc-sprint will focus on migrating the remaining pages from
>>>> MediaWiki (https://wiki.freeswitch.org) to Confluence Wiki (
>>>> https://freeswitch.org/confluence).
>>>>
>>>> We will use an FS IRC channel during the sprint: *#freeswitch-docs*
>>>> and will track our work on the spreadsheet:
>>>> https://docs.google.com/spreadsheets/d/1qsG-kRymvKlNBapnBLw86W130VdbnK6naYapbR_UNds/edit?pli=1#gid=1187898333
>>>>
>>>> During the sprint, please change the URL's "Status" you are working on
>>>> to "Editing" with your name next to it so we don't duplicate work.
>>>>
>>>> Some extra information:
>>>> - https://freeswitch.org/confluence/display/FREESWITCH/Wiki+Migration
>>>> -
>>>> https://freeswitch.org/confluence/display/FREESWITCH/Contributing+Documentation
>>>>
>>>> We hope to get a maximum number of people signed up!
>>>>
>>>> Peoples confirmed so far:
>>>> - Italo Rossi (+4)
>>>> - Areski Belaid
>>>>
>>>>
>>>> So, who is in?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Kind Regards
>>> Iwada
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> _______________________________________________
> Freeswitch-docs mailing list
> Freeswitch-docs at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-docs
>
>

-- 
Kind regards,
/Areski

----
Arezqui Belaid, <areski at gmail.com>
Founder at Star2Billing (www.star2billing.com)

Tel: +34650784355
Twitter: http://twitter.com/areskib
LinkedIn: http://www.linkedin.com/in/areski
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From anuragrana31189 at gmail.com  Fri Dec 12 20:25:10 2014
From: anuragrana31189 at gmail.com (Anurag Rana)
Date: Fri, 12 Dec 2014 22:55:10 +0530
Subject: [Freeswitch-users] Error while making gsmopen module in FS.
Message-ID: <CAFjUrj5+EHSgBfcokt031rAgtyYjhdJQ6Afxejn7C8eUfHvkyg@mail.gmail.com>

Hi All,

I installed FreeSwitch and now I am trying to add gsmopen module.
I edited the modules.conf file and uncommented the line
"endpoints/mod_gsmopen"

Now While running the command "make install all" , I am getting this error -
==================================================
making install mod_gsmopen
make[4]: Entering directory
`/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
make  install-am
make[5]: Entering directory
`/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
  CXX      mod_gsmopen_la-mod_gsmopen.lo
In file included from mod_gsmopen.cpp:34:0:
gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or directory
 #include "ctb-0.16/ctb.h"
                          ^
compilation terminated.
make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1
make[5]: Leaving directory
`/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
make[4]: *** [install] Error 2
make[4]: Leaving directory
`/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
make[3]: *** [mod_gsmopen-install] Error 1
make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
make[2]: *** [install-recursive] Error 1
make[2]: Leaving directory `/usr/local/src/freeswitch/src'
make[1]: *** [install-recursive] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch'
make: *** [install] Error 2
========================================================

?[image: Inline image 1]?



?Please help  me resolving this error.?


Anurag Rana <https://sites.google.com/site/homepagerana/>
M.Tech CSE, IIIT-Delhi,
https://sites.google.com/site/homepagerana/
??
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From brian at freeswitch.org  Fri Dec 12 20:44:37 2014
From: brian at freeswitch.org (Brian West)
Date: Fri, 12 Dec 2014 11:44:37 -0600
Subject: [Freeswitch-users] Error while making gsmopen module in FS.
In-Reply-To: <CAFjUrj5+EHSgBfcokt031rAgtyYjhdJQ6Afxejn7C8eUfHvkyg@mail.gmail.com>
References: <CAFjUrj5+EHSgBfcokt031rAgtyYjhdJQ6Afxejn7C8eUfHvkyg@mail.gmail.com>
Message-ID: <CAEJMVkBBbYgkQGqMNtAAoMQWrNK0NzFe5-f4WC7zGjSg3ozPLQ@mail.gmail.com>

I would review the confluence or wiki pages on setting this up, you're
missing the ilbs/files to compile the module properly.

On Fri, Dec 12, 2014 at 11:25 AM, Anurag Rana <anuragrana31189 at gmail.com>
wrote:
>
> Hi All,
>
> I installed FreeSwitch and now I am trying to add gsmopen module.
> I edited the modules.conf file and uncommented the line
> "endpoints/mod_gsmopen"
>
> Now While running the command "make install all" , I am getting this error
> -
> ==================================================
> making install mod_gsmopen
> make[4]: Entering directory
> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
> make  install-am
> make[5]: Entering directory
> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>   CXX      mod_gsmopen_la-mod_gsmopen.lo
> In file included from mod_gsmopen.cpp:34:0:
> gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or directory
>  #include "ctb-0.16/ctb.h"
>                           ^
> compilation terminated.
> make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1
> make[5]: Leaving directory
> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
> make[4]: *** [install] Error 2
> make[4]: Leaving directory
> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
> make[3]: *** [mod_gsmopen-install] Error 1
> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
> make[2]: *** [install-recursive] Error 1
> make[2]: Leaving directory `/usr/local/src/freeswitch/src'
> make[1]: *** [install-recursive] Error 1
> make[1]: Leaving directory `/usr/local/src/freeswitch'
> make: *** [install] Error 2
> ========================================================
>
> ?[image: Inline image 1]?
>
>
>
> ?Please help  me resolving this error.?
>
>
> Anurag Rana <https://sites.google.com/site/homepagerana/>
> M.Tech CSE, IIIT-Delhi,
> https://sites.google.com/site/homepagerana/
> ??
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
*iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
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From gmaruzz at gmail.com  Fri Dec 12 20:48:07 2014
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 12 Dec 2014 18:48:07 +0100
Subject: [Freeswitch-users] Error while making gsmopen module in FS.
In-Reply-To: <CAEJMVkBBbYgkQGqMNtAAoMQWrNK0NzFe5-f4WC7zGjSg3ozPLQ@mail.gmail.com>
References: <CAFjUrj5+EHSgBfcokt031rAgtyYjhdJQ6Afxejn7C8eUfHvkyg@mail.gmail.com>
	<CAEJMVkBBbYgkQGqMNtAAoMQWrNK0NzFe5-f4WC7zGjSg3ozPLQ@mail.gmail.com>
Message-ID: <CALXCt0piQ02dCvnrDLEoaioyC=FQRhOQMURrXegM7FgGAXy9_A@mail.gmail.com>

The wiki page describes the prerequisites and the way to install.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Dec 12, 2014 6:45 PM, "Brian West" <brian at freeswitch.org> wrote:

> I would review the confluence or wiki pages on setting this up, you're
> missing the ilbs/files to compile the module properly.
>
> On Fri, Dec 12, 2014 at 11:25 AM, Anurag Rana <anuragrana31189 at gmail.com>
> wrote:
>>
>> Hi All,
>>
>> I installed FreeSwitch and now I am trying to add gsmopen module.
>> I edited the modules.conf file and uncommented the line
>> "endpoints/mod_gsmopen"
>>
>> Now While running the command "make install all" , I am getting this
>> error -
>> ==================================================
>> making install mod_gsmopen
>> make[4]: Entering directory
>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>> make  install-am
>> make[5]: Entering directory
>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>   CXX      mod_gsmopen_la-mod_gsmopen.lo
>> In file included from mod_gsmopen.cpp:34:0:
>> gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or directory
>>  #include "ctb-0.16/ctb.h"
>>                           ^
>> compilation terminated.
>> make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1
>> make[5]: Leaving directory
>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>> make[4]: *** [install] Error 2
>> make[4]: Leaving directory
>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>> make[3]: *** [mod_gsmopen-install] Error 1
>> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>> make[2]: *** [install-recursive] Error 1
>> make[2]: Leaving directory `/usr/local/src/freeswitch/src'
>> make[1]: *** [install-recursive] Error 1
>> make[1]: Leaving directory `/usr/local/src/freeswitch'
>> make: *** [install] Error 2
>> ========================================================
>>
>> ?[image: Inline image 1]?
>>
>>
>>
>> ?Please help  me resolving this error.?
>>
>>
>> Anurag Rana <https://sites.google.com/site/homepagerana/>
>> M.Tech CSE, IIIT-Delhi,
>> https://sites.google.com/site/homepagerana/
>> ??
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri Dec 12 21:06:34 2014
From: mike at jerris.com (Michael Jerris)
Date: Fri, 12 Dec 2014 13:06:34 -0500
Subject: [Freeswitch-users] freeswitch 1.4 and encryption/rtcp-mux
In-Reply-To: <CAHt+dNE8JNjbCRFMpJ0q1mwsFWUJ5-oMr7eFqNAohPq7kce7MA@mail.gmail.com>
References: <CAHt+dNE8JNjbCRFMpJ0q1mwsFWUJ5-oMr7eFqNAohPq7kce7MA@mail.gmail.com>
Message-ID: <8B568748-EEE1-4E26-A3BB-302345FB0A01@jerris.com>


> On Dec 12, 2014, at 10:23 AM, Michel Brabants <michel.brabants at gmail.com> wrote:
> 
> Hello,
> 
> I recently started upgrading to FS 1.4, but I encountered 2 difficulties of which I'm still looking into one:
> 
>  1) DTLS-configuration seems to be required, although we don't use it currently. We use normal sip-profiles (no webrtc). The option to disable it, is "webrtc_enable_dtls=false", which can b set in the dialplan. But why is it trying to enable it by default? Can you disable it also in a profile?
> 

In what way do you think that some configuration is required?

> 2) Also a change because of webrtc seemingly. When receiving an invite (without sdp - 3pcc-request), freeswitch in the end response in its 200 OK with a rtcp-mux-line in its sdp. We don't want rtcp-mux, just rtp-port+1 for rtcp. When looking at the code, I don't currently know why FS sends back the myx-parameter as it seems only enabled when the other ends proposes it or am I missing something?
> 

Please report a bug on this.



From anuragrana31189 at gmail.com  Fri Dec 12 21:24:35 2014
From: anuragrana31189 at gmail.com (Anurag Rana)
Date: Fri, 12 Dec 2014 23:54:35 +0530
Subject: [Freeswitch-users] Error while making gsmopen module in FS.
In-Reply-To: <CALXCt0piQ02dCvnrDLEoaioyC=FQRhOQMURrXegM7FgGAXy9_A@mail.gmail.com>
References: <CAFjUrj5+EHSgBfcokt031rAgtyYjhdJQ6Afxejn7C8eUfHvkyg@mail.gmail.com>
	<CAEJMVkBBbYgkQGqMNtAAoMQWrNK0NzFe5-f4WC7zGjSg3ozPLQ@mail.gmail.com>
	<CALXCt0piQ02dCvnrDLEoaioyC=FQRhOQMURrXegM7FgGAXy9_A@mail.gmail.com>
Message-ID: <CAFjUrj7G6DUcekM1uM8aMW2S-7vsSgp8_CNdab6+JJfAkeyN6w@mail.gmail.com>

According to Wiki page I need to compile and install gsmlib and libctb.

These both are available in folder.
*/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen*

Earlier make command was complaining about not finding "ctb-0.16/ctb.h", So
I *copied* the folder "ctb-0.16" from
*/endpoints/mod_gsmopen/libctb-0.16/include* to */endpoints/mod_gsmopen/ . *now
this error is gone but a new error is coming up.

/usr/bin/ld: cannot find -lctb-0.16

this is because of below line in MAKEFILE

*mod_gsmopen_la_LDFLAGS = -avoid-version -module -no-undefined -lctb-0.16
-lgsmme*


any suggestion please?



Anurag Rana <https://sites.google.com/site/homepagerana/>
M.Tech CSE, IIIT-Delhi,
https://sites.google.com/site/homepagerana/
<http://home.iiitd.edu.in/%7Eanurag1392/>




On Fri, Dec 12, 2014 at 11:18 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:
>
> The wiki page describes the prerequisites and the way to install.
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Dec 12, 2014 6:45 PM, "Brian West" <brian at freeswitch.org> wrote:
>
>> I would review the confluence or wiki pages on setting this up, you're
>> missing the ilbs/files to compile the module properly.
>>
>> On Fri, Dec 12, 2014 at 11:25 AM, Anurag Rana <anuragrana31189 at gmail.com>
>> wrote:
>>>
>>> Hi All,
>>>
>>> I installed FreeSwitch and now I am trying to add gsmopen module.
>>> I edited the modules.conf file and uncommented the line
>>> "endpoints/mod_gsmopen"
>>>
>>> Now While running the command "make install all" , I am getting this
>>> error -
>>> ==================================================
>>> making install mod_gsmopen
>>> make[4]: Entering directory
>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>> make  install-am
>>> make[5]: Entering directory
>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>>   CXX      mod_gsmopen_la-mod_gsmopen.lo
>>> In file included from mod_gsmopen.cpp:34:0:
>>> gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or directory
>>>  #include "ctb-0.16/ctb.h"
>>>                           ^
>>> compilation terminated.
>>> make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1
>>> make[5]: Leaving directory
>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>> make[4]: *** [install] Error 2
>>> make[4]: Leaving directory
>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>> make[3]: *** [mod_gsmopen-install] Error 1
>>> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>>> make[2]: *** [install-recursive] Error 1
>>> make[2]: Leaving directory `/usr/local/src/freeswitch/src'
>>> make[1]: *** [install-recursive] Error 1
>>> make[1]: Leaving directory `/usr/local/src/freeswitch'
>>> make: *** [install] Error 2
>>> ========================================================
>>>
>>> ?[image: Inline image 1]?
>>>
>>>
>>>
>>> ?Please help  me resolving this error.?
>>>
>>>
>>> Anurag Rana <https://sites.google.com/site/homepagerana/>
>>> M.Tech CSE, IIIT-Delhi,
>>> https://sites.google.com/site/homepagerana/
>>> ??
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at gmail.com  Fri Dec 12 21:31:21 2014
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 12 Dec 2014 19:31:21 +0100
Subject: [Freeswitch-users] Error while making gsmopen module in FS.
In-Reply-To: <CAFjUrj7G6DUcekM1uM8aMW2S-7vsSgp8_CNdab6+JJfAkeyN6w@mail.gmail.com>
References: <CAFjUrj5+EHSgBfcokt031rAgtyYjhdJQ6Afxejn7C8eUfHvkyg@mail.gmail.com>
	<CAEJMVkBBbYgkQGqMNtAAoMQWrNK0NzFe5-f4WC7zGjSg3ozPLQ@mail.gmail.com>
	<CALXCt0piQ02dCvnrDLEoaioyC=FQRhOQMURrXegM7FgGAXy9_A@mail.gmail.com>
	<CAFjUrj7G6DUcekM1uM8aMW2S-7vsSgp8_CNdab6+JJfAkeyN6w@mail.gmail.com>
Message-ID: <CALXCt0rRzCse=Jh+izNBMOp64Ti-4d9n_ADO3hCUF0TC=ES7Lw@mail.gmail.com>

Follow strictly, step by step and from the beginning the instructions in
the wiki page.

Everyone succeeds, you'll eventually succeed too.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On Dec 12, 2014 7:27 PM, "Anurag Rana" <anuragrana31189 at gmail.com> wrote:

> According to Wiki page I need to compile and install gsmlib and libctb.
>
> These both are available in folder.
> */usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen*
>
> Earlier make command was complaining about not finding "ctb-0.16/ctb.h",
> So I *copied* the folder "ctb-0.16" from
> */endpoints/mod_gsmopen/libctb-0.16/include* to */endpoints/mod_gsmopen/
> . *now this error is gone but a new error is coming up.
>
> /usr/bin/ld: cannot find -lctb-0.16
>
> this is because of below line in MAKEFILE
>
> *mod_gsmopen_la_LDFLAGS = -avoid-version -module -no-undefined -lctb-0.16
> -lgsmme*
>
>
> any suggestion please?
>
>
>
> Anurag Rana <https://sites.google.com/site/homepagerana/>
> M.Tech CSE, IIIT-Delhi,
> https://sites.google.com/site/homepagerana/
> <http://home.iiitd.edu.in/%7Eanurag1392/>
>
>
>
>
> On Fri, Dec 12, 2014 at 11:18 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>>
>> The wiki page describes the prerequisites and the way to install.
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On Dec 12, 2014 6:45 PM, "Brian West" <brian at freeswitch.org> wrote:
>>
>>> I would review the confluence or wiki pages on setting this up, you're
>>> missing the ilbs/files to compile the module properly.
>>>
>>> On Fri, Dec 12, 2014 at 11:25 AM, Anurag Rana <anuragrana31189 at gmail.com
>>> > wrote:
>>>>
>>>> Hi All,
>>>>
>>>> I installed FreeSwitch and now I am trying to add gsmopen module.
>>>> I edited the modules.conf file and uncommented the line
>>>> "endpoints/mod_gsmopen"
>>>>
>>>> Now While running the command "make install all" , I am getting this
>>>> error -
>>>> ==================================================
>>>> making install mod_gsmopen
>>>> make[4]: Entering directory
>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>>> make  install-am
>>>> make[5]: Entering directory
>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>>>   CXX      mod_gsmopen_la-mod_gsmopen.lo
>>>> In file included from mod_gsmopen.cpp:34:0:
>>>> gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or directory
>>>>  #include "ctb-0.16/ctb.h"
>>>>                           ^
>>>> compilation terminated.
>>>> make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1
>>>> make[5]: Leaving directory
>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>>> make[4]: *** [install] Error 2
>>>> make[4]: Leaving directory
>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>>> make[3]: *** [mod_gsmopen-install] Error 1
>>>> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>>>> make[2]: *** [install-recursive] Error 1
>>>> make[2]: Leaving directory `/usr/local/src/freeswitch/src'
>>>> make[1]: *** [install-recursive] Error 1
>>>> make[1]: Leaving directory `/usr/local/src/freeswitch'
>>>> make: *** [install] Error 2
>>>> ========================================================
>>>>
>>>> ?[image: Inline image 1]?
>>>>
>>>>
>>>>
>>>> ?Please help  me resolving this error.?
>>>>
>>>>
>>>> Anurag Rana <https://sites.google.com/site/homepagerana/>
>>>> M.Tech CSE, IIIT-Delhi,
>>>> https://sites.google.com/site/homepagerana/
>>>> ??
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anuragrana31189 at gmail.com  Fri Dec 12 21:45:30 2014
From: anuragrana31189 at gmail.com (Anurag Rana)
Date: Sat, 13 Dec 2014 00:15:30 +0530
Subject: [Freeswitch-users] Error while making gsmopen module in FS.
In-Reply-To: <CALXCt0rRzCse=Jh+izNBMOp64Ti-4d9n_ADO3hCUF0TC=ES7Lw@mail.gmail.com>
References: <CAFjUrj5+EHSgBfcokt031rAgtyYjhdJQ6Afxejn7C8eUfHvkyg@mail.gmail.com>
	<CAEJMVkBBbYgkQGqMNtAAoMQWrNK0NzFe5-f4WC7zGjSg3ozPLQ@mail.gmail.com>
	<CALXCt0piQ02dCvnrDLEoaioyC=FQRhOQMURrXegM7FgGAXy9_A@mail.gmail.com>
	<CAFjUrj7G6DUcekM1uM8aMW2S-7vsSgp8_CNdab6+JJfAkeyN6w@mail.gmail.com>
	<CALXCt0rRzCse=Jh+izNBMOp64Ti-4d9n_ADO3hCUF0TC=ES7Lw@mail.gmail.com>
Message-ID: <CAFjUrj7AQTZ925kFkAZJGfnFh5NfO0af7aVQ=_4Xsy0YDvczqA@mail.gmail.com>

Done. Thanks.



Anurag Rana <https://sites.google.com/site/homepagerana/>
M.Tech CSE, IIIT-Delhi,
https://sites.google.com/site/homepagerana/
<http://home.iiitd.edu.in/%7Eanurag1392/>




On Sat, Dec 13, 2014 at 12:01 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:
>
> Follow strictly, step by step and from the beginning the instructions in
> the wiki page.
>
> Everyone succeeds, you'll eventually succeed too.
>
> sent from my mobile,
> Giovanni Maruzzelli
> cell: +39 347 266 56 18
> On Dec 12, 2014 7:27 PM, "Anurag Rana" <anuragrana31189 at gmail.com> wrote:
>
>> According to Wiki page I need to compile and install gsmlib and libctb.
>>
>> These both are available in folder.
>> */usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen*
>>
>> Earlier make command was complaining about not finding "ctb-0.16/ctb.h",
>> So I *copied* the folder "ctb-0.16" from
>> */endpoints/mod_gsmopen/libctb-0.16/include* to */endpoints/mod_gsmopen/
>> . *now this error is gone but a new error is coming up.
>>
>> /usr/bin/ld: cannot find -lctb-0.16
>>
>> this is because of below line in MAKEFILE
>>
>> *mod_gsmopen_la_LDFLAGS = -avoid-version -module -no-undefined -lctb-0.16
>> -lgsmme*
>>
>>
>> any suggestion please?
>>
>>
>>
>> Anurag Rana <https://sites.google.com/site/homepagerana/>
>> M.Tech CSE, IIIT-Delhi,
>> https://sites.google.com/site/homepagerana/
>> <http://home.iiitd.edu.in/%7Eanurag1392/>
>>
>>
>>
>>
>> On Fri, Dec 12, 2014 at 11:18 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>>
>>> The wiki page describes the prerequisites and the way to install.
>>>
>>> sent from my mobile,
>>> Giovanni Maruzzelli
>>> cell: +39 347 266 56 18
>>> On Dec 12, 2014 6:45 PM, "Brian West" <brian at freeswitch.org> wrote:
>>>
>>>> I would review the confluence or wiki pages on setting this up, you're
>>>> missing the ilbs/files to compile the module properly.
>>>>
>>>> On Fri, Dec 12, 2014 at 11:25 AM, Anurag Rana <
>>>> anuragrana31189 at gmail.com> wrote:
>>>>>
>>>>> Hi All,
>>>>>
>>>>> I installed FreeSwitch and now I am trying to add gsmopen module.
>>>>> I edited the modules.conf file and uncommented the line
>>>>> "endpoints/mod_gsmopen"
>>>>>
>>>>> Now While running the command "make install all" , I am getting this
>>>>> error -
>>>>> ==================================================
>>>>> making install mod_gsmopen
>>>>> make[4]: Entering directory
>>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>>>> make  install-am
>>>>> make[5]: Entering directory
>>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>>>>   CXX      mod_gsmopen_la-mod_gsmopen.lo
>>>>> In file included from mod_gsmopen.cpp:34:0:
>>>>> gsmopen.h:103:26: fatal error: ctb-0.16/ctb.h: No such file or
>>>>> directory
>>>>>  #include "ctb-0.16/ctb.h"
>>>>>                           ^
>>>>> compilation terminated.
>>>>> make[5]: *** [mod_gsmopen_la-mod_gsmopen.lo] Error 1
>>>>> make[5]: Leaving directory
>>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>>>> make[4]: *** [install] Error 2
>>>>> make[4]: Leaving directory
>>>>> `/usr/local/src/freeswitch/src/mod/endpoints/mod_gsmopen'
>>>>> make[3]: *** [mod_gsmopen-install] Error 1
>>>>> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod'
>>>>> make[2]: *** [install-recursive] Error 1
>>>>> make[2]: Leaving directory `/usr/local/src/freeswitch/src'
>>>>> make[1]: *** [install-recursive] Error 1
>>>>> make[1]: Leaving directory `/usr/local/src/freeswitch'
>>>>> make: *** [install] Error 2
>>>>> ========================================================
>>>>>
>>>>> ?[image: Inline image 1]?
>>>>>
>>>>>
>>>>>
>>>>> ?Please help  me resolving this error.?
>>>>>
>>>>>
>>>>> Anurag Rana <https://sites.google.com/site/homepagerana/>
>>>>> M.Tech CSE, IIIT-Delhi,
>>>>> https://sites.google.com/site/homepagerana/
>>>>> ??
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at 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
>>>>
>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From sphotos1982 at gmail.com  Fri Dec 12 22:20:40 2014
From: sphotos1982 at gmail.com (spros p)
Date: Fri, 12 Dec 2014 21:20:40 +0200
Subject: [Freeswitch-users] LUA ESL Help
Message-ID: <CAJ0dgAzm1wsHg4GfNi+jaiaHSx8EZfjsY3f4Dhvc3Nm7NRi-NQ@mail.gmail.com>

Hello,

I am trying to run a simple ESL script using Lua. I have installed Lua 5.1
and the respective dev packages on my linux machine. Make luamod in
/libs/esl/ directory works just fine.

However when executing: freeswitch/libs/esl/lua/single_command.lua, i
receive the following:

lua: lua/single_command.lua:2: module 'ESL' not found:
        no field package.preload['ESL']
        no file './ESL.lua'
        no file '/usr/share/lua/5.1/ESL.lua'
        no file '/usr/share/lua/5.1/ESL/init.lua'
        no file '/usr/lib64/lua/5.1/ESL.lua'
        no file '/usr/lib64/lua/5.1/ESL/init.lua'
        no file './ESL.so'
        no file '/usr/lib64/lua/5.1/ESL.so'
        no file '/usr/lib64/lua/5.1/loadall.so'
stack traceback:
        [C]: in function 'require'
        lua/single_command.lua:2: in main chunk
        [C]: ?

Any help would be greatly appreciated

Best,
SP
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From aqsyounas at gmail.com  Fri Dec 12 22:37:33 2014
From: aqsyounas at gmail.com (Aqs Younas)
Date: Sat, 13 Dec 2014 00:37:33 +0500
Subject: [Freeswitch-users] How to play a stream other than mp3 with
	mod_shout.
Message-ID: <CAGVOjVUjyzgWWZ4+N1FBM-NeVe3Xx9VUDWJ8xkBBB4rs78Tyig@mail.gmail.com>

Hi, All

How can i play a live stream other than mp3 with mod_shout or any module.?
Is there any way to buffer the stream before playing it with mod_shout.


Currently i have a list of streams and when i play them with mod_shout some
work fine but others give (time out) error.

How can i play mostly stream in freeswitch?

Thanks
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From ahabiba at gmail.com  Fri Dec 12 22:39:16 2014
From: ahabiba at gmail.com (Ahmed Habiba)
Date: Fri, 12 Dec 2014 22:39:16 +0300
Subject: [Freeswitch-users] SIP Header Modification / Manipulation
In-Reply-To: <29A45828-2427-492A-B18E-9260B4880745@gmail.com>
References: <29A45828-2427-492A-B18E-9260B4880745@gmail.com>
Message-ID: <0C78DA91-56C6-4901-B1BA-676257133525@gmail.com>

Dears,

Kindly help me on how can I modify the SIP header in the re-invite in the below scenario:


[A Number] <???>[FreeSWitch]<???>[B Number]


1-A Number Call B Number (Sip Invite sent from A number)
2-B number is not registered.
3-Call logic let FreeSwitch answer the call( Freeswitch send re-invite to A number)

in the part number 3 I like to do modification on the SIP headers, by for example changing the From or To.
 

Your help will be appreciated.


Thanks,

Ahmed Habiba.



From jason.holden at start.ca  Fri Dec 12 22:40:12 2014
From: jason.holden at start.ca (Jason Holden)
Date: Fri, 12 Dec 2014 14:40:12 -0500
Subject: [Freeswitch-users] intigration of sms using an ip provider
Message-ID: <D7A9C71CF36A45559B5644E92B8FEB44@bob>

Anyone have any information on this?

 

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From andrew at cassidywebservices.co.uk  Sat Dec 13 00:16:48 2014
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 12 Dec 2014 21:16:48 +0000
Subject: [Freeswitch-users] FreeBSD or Debian
In-Reply-To: <CAPNTnnr3NAw6Wk0-yEJ-8M3w_BuqpqCNzRj7GZjy12sZu4E6CQ@mail.gmail.com>
References: <CAHJ5xPmRn4v_tt+VTbJpKgV-JXJojY8Ftbgaff=mCumA7MdezQ@mail.gmail.com>
	<70910985-9657-41DF-93DA-58AD7D4B862D@goes.com>
	<CAPNTnnr3NAw6Wk0-yEJ-8M3w_BuqpqCNzRj7GZjy12sZu4E6CQ@mail.gmail.com>
Message-ID: <CAMbW9o+7wh0dpiuEbTbgKgedB=fiPn5sfYa-gb7Qq5qnfLB6+A@mail.gmail.com>

debian-kfreebsd? :P

Debian on the freebsd kernel. Sounds rather sadomasochistic to me...

On 12 December 2014 at 15:34, Guillermo Ruiz Camauer <grcamauer at gmail.com>
wrote:
>
> I tend to use machines with ZFS only as dedicated file servers.  ZFS is a
> great file system, but it takes over the machine and uses as much memory
> and resources as it can get it's hands on.  I would not use it for
> FreeSwitch if there will be a lot of file-io going on.
>
> Guillermo
>
> On Fri, Dec 12, 2014 at 11:07 AM, Norm <norm at goes.com> wrote:
>
>> Debian
>>
>> > On Dec 12, 2014, at 8:04 AM, Thomas L?cke <thomas.granvej6 at gmail.com>
>> wrote:
>> >
>> > Hey all,
>> >
>> > I'm involved in a project where we'll soon need to spin up a couple of
>> FreeSWITCH instances, and we're a bit torn between Debian and FreeBSD.
>> >
>> > Anybody out there with experience with both? Are there any differences?
>> Will Debian be "better" due to FreeSWITCH primarily targeting Linux?
>> >
>> > We're very slightly leaning towards FreeBSD due to jails, ZFS and other
>> nice FreeBSD tools, but we're not averse to Debian in any way. We're
>> probably at 52% FreeBSD and 48% Debian.
>> >
>> > The instances will run on hardware with decent specs. No less than 16
>> cores (probably 32) and no less than 32GB RAM and SSD in every single bay.
>> We're not targeting low spec.
>> >
>> > Any and all advice is welcomed with open arms.
>> >
>> > :o)
>> > Thomas L?cke
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://confluence.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>


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


*T <info at cassidywebservices.co.uk> *03300 100 960  *F
<info at cassidywebservices.co.uk> *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From mike at jerris.com  Sat Dec 13 00:37:42 2014
From: mike at jerris.com (Michael Jerris)
Date: Fri, 12 Dec 2014 16:37:42 -0500
Subject: [Freeswitch-users] SIP Header Modification / Manipulation
In-Reply-To: <0C78DA91-56C6-4901-B1BA-676257133525@gmail.com>
References: <29A45828-2427-492A-B18E-9260B4880745@gmail.com>
	<0C78DA91-56C6-4901-B1BA-676257133525@gmail.com>
Message-ID: <6F5A84DB-2678-4BA7-80C3-05E62F19B271@jerris.com>

if you are looking to update the display of the number that is being called on the caller phone, you want to be looking at display udpates, and callee_id_name and callee_id_number params.

> On Dec 12, 2014, at 2:39 PM, Ahmed Habiba <ahabiba at gmail.com> wrote:
> 
> Dears,
> 
> Kindly help me on how can I modify the SIP header in the re-invite in the below scenario:
> 
> 
> [A Number] <???>[FreeSWitch]<???>[B Number]
> 
> 
> 1-A Number Call B Number (Sip Invite sent from A number)
> 2-B number is not registered.
> 3-Call logic let FreeSwitch answer the call( Freeswitch send re-invite to A number)
> 
> in the part number 3 I like to do modification on the SIP headers, by for example changing the From or To.
> 
> 
> Your help will be appreciated.
> 



From anthony.minessale at gmail.com  Sat Dec 13 00:43:49 2014
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 12 Dec 2014 15:43:49 -0600
Subject: [Freeswitch-users] Record sound quality not good but pcap is
 quite good
In-Reply-To: <CA+q4kR+wJ=CBQ97FThSU6BuQPM0cSZmrntkyic8s8KPtyXo2iA@mail.gmail.com>
References: <CA+q4kR+wJ=CBQ97FThSU6BuQPM0cSZmrntkyic8s8KPtyXo2iA@mail.gmail.com>
Message-ID: <CAKbxfG9CDBX=ravcmxdMA+BYpAy0f62K1SN6i7NjvXrN01z7sA@mail.gmail.com>

Try latest master.

On Fri, Dec 12, 2014 at 8:54 AM, Ali Pey <alipey at gmail.com> wrote:
>
> Hello,
>
> I use record to record part of a call and sometimes I get some static
> noises and distortion; however when I listen to the call on pcap capture,
> it's quite clear and no noise.
>
> I record to a wav file and then use sox to convert it to ulaw.
>
> Any suggestions? Any help would be greatly appreciated.
>
>
> Thanks,
> Ali Pey
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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From blasterjr at gmail.com  Sat Dec 13 03:48:22 2014
From: blasterjr at gmail.com (Chris Tunbridge)
Date: Fri, 12 Dec 2014 17:48:22 -0700
Subject: [Freeswitch-users] mkdir not working
In-Reply-To: <548AF0A8.90909@inteli-sim.com>
References: <548AF0A8.90909@inteli-sim.com>
Message-ID: <CAO3hyzG-scDKOgOZmzMXUG1sH7OtajVo8CE9ROhbzYUzWO77nw@mail.gmail.com>

You could try using something like the following instead.

$con->execute("system", "mkdir -p $vmail_base");

but i'm not certain as to where this is being called.

On Fri, Dec 12, 2014 at 6:42 AM, kthofer <karl-theo_hofer at inteli-sim.com>
wrote:
>
> Hi there
> we try to create a new directory
>   $con->execute("mkdir", "$vmail_base");
> to store some Voice mail files
>
> But for some reason FS is not even trying to create the directory
> we checked the user rights
> the spelling
> but still nothing not even in the linux logs we can not see that FS is
> trying it atall.
>
> Any suggestions??
> what is going wrong?
> Please let me know if you have a solution or workaround.
>
> --
> With best regards
>
> Karl Theo Hofer
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From chad at apartmentlines.com  Sat Dec 13 08:13:41 2014
From: chad at apartmentlines.com (Chad Phillips)
Date: Fri, 12 Dec 2014 22:13:41 -0700
Subject: [Freeswitch-users] LUA ESL Help
In-Reply-To: <CAJ0dgAzm1wsHg4GfNi+jaiaHSx8EZfjsY3f4Dhvc3Nm7NRi-NQ@mail.gmail.com>
References: <CAJ0dgAzm1wsHg4GfNi+jaiaHSx8EZfjsY3f4Dhvc3Nm7NRi-NQ@mail.gmail.com>
Message-ID: <B335E1470D244A69BB82E563843B4E7F@gmail.com>

The paths listed in the error message are the paths that Lua looks in for the module. So you either need to put your ESL.so in one of those locations, or add its location to path list. Check http://www.lua.org/pil/8.1.html for detailed info on how Lua does path handling.


On Friday, December 12, 2014 at 12:20 PM, spros p wrote:

> Hello, 
> 
> I am trying to run a simple ESL script using Lua. I have installed Lua 5.1 and the respective dev packages on my linux machine. Make luamod in /libs/esl/ directory works just fine. 
> 
> However when executing: freeswitch/libs/esl/lua/single_command.lua, i receive the following: 
> 
> lua: lua/single_command.lua:2: module 'ESL' not found: 
>         no field package.preload['ESL'] 
>         no file './ESL.lua' 
>         no file '/usr/share/lua/5.1/ESL.lua' 
>         no file '/usr/share/lua/5.1/ESL/init.lua' 
>         no file '/usr/lib64/lua/5.1/ESL.lua' 
>         no file '/usr/lib64/lua/5.1/ESL/init.lua' 
>         no file './ESL.so' 
>         no file '/usr/lib64/lua/5.1/ESL.so' 
>         no file '/usr/lib64/lua/5.1/loadall.so' 
> stack traceback: 
>         [C]: in function 'require' 
>         lua/single_command.lua:2: in main chunk 
>         [C]: ? 
> 
> Any help would be greatly appreciated
> 
> Best,
> SP
> 
> 
> _________________________________________________________________________
> 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 omortimer at gmail.com  Sat Dec 13 11:42:29 2014
From: omortimer at gmail.com (Oz Mortimer)
Date: Sat, 13 Dec 2014 08:42:29 +0000
Subject: [Freeswitch-users] mkdir not working
In-Reply-To: <CAO3hyzG-scDKOgOZmzMXUG1sH7OtajVo8CE9ROhbzYUzWO77nw@mail.gmail.com>
References: <548AF0A8.90909@inteli-sim.com>
	<CAO3hyzG-scDKOgOZmzMXUG1sH7OtajVo8CE9ROhbzYUzWO77nw@mail.gmail.com>
Message-ID: <4D344D60-43F9-47EE-92D6-246697C07ADB@gmail.com>

Assuming your using php, you don't need to use freeswitch.
system("mkdir -p $vmail_base");

Making sure that the script owner has permissions to create this directory.



> On 13 Dec 2014, at 00:48, Chris Tunbridge <blasterjr at gmail.com> wrote:
> 
> You could try using something like the following instead.
> 
> $con->execute("system", "mkdir -p $vmail_base");
> 
> but i'm not certain as to where this is being called.
> 
>> On Fri, Dec 12, 2014 at 6:42 AM, kthofer <karl-theo_hofer at inteli-sim.com> wrote:
>> Hi there
>> we try to create a new directory
>>   $con->execute("mkdir", "$vmail_base");
>> to store some Voice mail files
>> 
>> But for some reason FS is not even trying to create the directory
>> we checked the user rights
>> the spelling
>> but still nothing not even in the linux logs we can not see that FS is
>> trying it atall.
>> 
>> Any suggestions??
>> what is going wrong?
>> Please let me know if you have a solution or workaround.
>> 
>> --
>> With best regards
>> 
>> Karl Theo Hofer
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From sudhansh at gmail.com  Sat Dec 13 15:08:16 2014
From: sudhansh at gmail.com (Sudhanshu)
Date: Sat, 13 Dec 2014 17:38:16 +0530
Subject: [Freeswitch-users] fs_cli hangs
In-Reply-To: <CA+mgNr6VK4F5QoOoqgVAGX=OmY6iMOAj69X_QacCnhj5dMiH9A@mail.gmail.com>
References: <CA+mgNr5S9_L6LcfqUwYhMfFJOpRxn3uDCitM_cwNQ7h7LuODhw@mail.gmail.com>
	<CALR8eWsO6HK2ghd0cyXuVLMHF5BOBCcqP5uLX_Ey1UArS3YSrg@mail.gmail.com>
	<CA+mgNr6VK4F5QoOoqgVAGX=OmY6iMOAj69X_QacCnhj5dMiH9A@mail.gmail.com>
Message-ID: <CALR8eWssOsrKtt-mycZqWHuiQRwz+Ow3vvnS4CbhcrZvd6QS0A@mail.gmail.com>

Always use syntax highlighting whenever possible.

Have you tried disabling the modules which fail to load in your
modules.conf.xml? Ideally freeswitch should handle it, but still, it is
worth a shot.

Also, What happens when you start freeswitch as a normal process (and not
as a daemon)? Do you see a freeswitch command prompt?

--
Sudhanshu

On Fri, Dec 12, 2014 at 12:06 PM, akhil garg <akhilgarg7 at gmail.com> wrote:
>
> Please see the logs: pastebin : 23717
>
>
>
> regards,
> akhil
>
> On Thu, Dec 11, 2014 at 10:40 PM, Sudhanshu <sudhansh at gmail.com> wrote:
>>
>> Enable debug logging and then check freeswitch.log.
>> What happens when you start freeswitch as anormal process (and not as a
>> daemon)?
>>
>> --
>> Sudhanshu
>>
>> On Thu, Dec 11, 2014 at 2:30 PM, akhil garg <akhilgarg7 at gmail.com> wrote:
>>
>>> running "fs_cli -H 127.0.0.1 -P 8021 -d 7" gives different outputs but
>>> no success.
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> OUTPUT 1:
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /root/.fs_cli_conf.
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /etc/fs_cli.conf.
>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using
>>> builtin profile
>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>>
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> OUTPUT 2:
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /root/.fs_cli_conf.
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /etc/fs_cli.conf.
>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using
>>> builtin profile
>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>>> [auth/request]
>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>>> Event-Name: SOCKET_DATA
>>> Content-Type: auth/request
>>>
>>>
>>> [DEBUG] esl.c:1465 esl_send() SEND
>>> auth ClueCon
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>>
>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> OUTPUT 3:
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /root/.fs_cli_conf.
>>> [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is
>>> /etc/fs_cli.conf.
>>> [DEBUG] fs_cli.c:1438 main() profile default does not exist using
>>> builtin profile
>>> [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1]
>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>>> [auth/request]
>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>>> Event-Name: SOCKET_DATA
>>> Content-Type: auth/request
>>>
>>>
>>> [DEBUG] esl.c:1465 esl_send() SEND
>>> auth ClueCon
>>>
>>>
>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] =
>>> [command/reply]
>>> [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK
>>> accepted]
>>> [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE
>>> Event-Name: SOCKET_DATA
>>> Content-Type: command/reply
>>> Reply-Text: +OK accepted
>>>
>>>
>>> [DEBUG] esl.c:1465 esl_send() SEND
>>> log
>>>
>>> ------------------------------------------------------------------------------------------------------------------------------------------------
>>>
>>>
>>>
>>> --
>>> regards,
>>> akhil
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> --
> regards,
> akhil
>
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From karl-theo_hofer at inteli-sim.com  Sat Dec 13 16:57:46 2014
From: karl-theo_hofer at inteli-sim.com (kthofer)
Date: Sat, 13 Dec 2014 14:57:46 +0100
Subject: [Freeswitch-users] mkdir not working
In-Reply-To: <4D344D60-43F9-47EE-92D6-246697C07ADB@gmail.com>
References: <548AF0A8.90909@inteli-sim.com>
	<CAO3hyzG-scDKOgOZmzMXUG1sH7OtajVo8CE9ROhbzYUzWO77nw@mail.gmail.com>
	<4D344D60-43F9-47EE-92D6-246697C07ADB@gmail.com>
Message-ID: <BB7F0CC0-A48D-42DC-A974-3B3E9235A0D2@inteli-sim.com>


Hi there i am using perl 
Will try the suggestions 
But still strange that FS is not even trying to create a directory


With best regards

Karl Theo Hofer 

> Am 13 dec 2014 um 09:42 schrieb Oz Mortimer <omortimer at gmail.com>:
> 
> Assuming your using php, you don't need to use freeswitch.
> system("mkdir -p $vmail_base");
> 
> Making sure that the script owner has permissions to create this directory.
> 
> 
> 
>> On 13 Dec 2014, at 00:48, Chris Tunbridge <blasterjr at gmail.com> wrote:
>> 
>> You could try using something like the following instead.
>> 
>> $con->execute("system", "mkdir -p $vmail_base");
>> 
>> but i'm not certain as to where this is being called.
>> 
>>> On Fri, Dec 12, 2014 at 6:42 AM, kthofer <karl-theo_hofer at inteli-sim.com> wrote:
>>> Hi there
>>> we try to create a new directory
>>>   $con->execute("mkdir", "$vmail_base");
>>> to store some Voice mail files
>>> 
>>> But for some reason FS is not even trying to create the directory
>>> we checked the user rights
>>> the spelling
>>> but still nothing not even in the linux logs we can not see that FS is
>>> trying it atall.
>>> 
>>> Any suggestions??
>>> what is going wrong?
>>> Please let me know if you have a solution or workaround.
>>> 
>>> --
>>> With best regards
>>> 
>>> Karl Theo Hofer
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: 
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: 
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From max at nysolutions.com  Sun Dec 14 03:04:40 2014
From: max at nysolutions.com (Moishe Grunstein)
Date: Sun, 14 Dec 2014 00:04:40 +0000
Subject: [Freeswitch-users] How to play a stream other than mp3
	with	mod_shout.
In-Reply-To: <CAGVOjVUjyzgWWZ4+N1FBM-NeVe3Xx9VUDWJ8xkBBB4rs78Tyig@mail.gmail.com>
References: <CAGVOjVUjyzgWWZ4+N1FBM-NeVe3Xx9VUDWJ8xkBBB4rs78Tyig@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10315B56728@ex2.connex.local>

You can try mod_vlc

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>
Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified
[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 Aqs Younas
Sent: Friday, December 12, 2014 2:38 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout.

Hi, All
How can i play a live stream other than mp3 with mod_shout or any module.? Is there any way to buffer the stream before playing it with mod_shout.

Currently i have a list of streams and when i play them with mod_shout some work fine but others give (time out) error.
How can i play mostly stream in freeswitch?

Thanks
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From aqsyounas at gmail.com  Sun Dec 14 17:46:35 2014
From: aqsyounas at gmail.com (aqs younas)
Date: Sun, 14 Dec 2014 19:46:35 +0500
Subject: [Freeswitch-users] How to play a stream other than mp3
 withmod_shout.
In-Reply-To: <BA6A05D48E282C4C9989496521266BA10315B56728@ex2.connex.local>
References: <CAGVOjVUjyzgWWZ4+N1FBM-NeVe3Xx9VUDWJ8xkBBB4rs78Tyig@mail.gmail.com>
	<BA6A05D48E282C4C9989496521266BA10315B56728@ex2.connex.local>
Message-ID: <548da2cb.85f0c20a.2718.1cb2@mx.google.com>

Many thanks for your reply.I will give it a try.

-----Original Message-----
From: "Moishe Grunstein" <max at nysolutions.com>
Sent: ?12/?14/?2014 5:05 AM
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] How to play a stream other than mp3 withmod_shout.

You can try mod_vlc
 
Thanks,
 
Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com 
Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified

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 Aqs Younas
Sent: Friday, December 12, 2014 2:38 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] How to play a stream other than mp3 with mod_shout.
 
Hi, All
How can i play a live stream other than mp3 with mod_shout or any module.? Is there any way to buffer the stream before playing it with mod_shout.


Currently i have a list of streams and when i play them with mod_shout some work fine but others give (time out) error.
How can i play mostly stream in freeswitch?
 
Thanks
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From danny.gershman at gmail.com  Sun Dec 14 22:37:56 2014
From: danny.gershman at gmail.com (Danny Gershman)
Date: Sun, 14 Dec 2014 14:37:56 -0500
Subject: [Freeswitch-users] How to play a stream other than mp3 with
	mod_shout.
In-Reply-To: <CAGVOjVUjyzgWWZ4+N1FBM-NeVe3Xx9VUDWJ8xkBBB4rs78Tyig@mail.gmail.com>
References: <CAGVOjVUjyzgWWZ4+N1FBM-NeVe3Xx9VUDWJ8xkBBB4rs78Tyig@mail.gmail.com>
Message-ID: <CAEorZynkKUxMrS5H4G7v=BTYFa_ooySz9jivkrucRu5XgFn1dg@mail.gmail.com>

Also mod_rtmp lets you play from an FMS server.

On Friday, December 12, 2014, Aqs Younas <aqsyounas at gmail.com> wrote:

> Hi, All
>
> How can i play a live stream other than mp3 with mod_shout or any module.?
> Is there any way to buffer the stream before playing it with mod_shout.
>
>
> Currently i have a list of streams and when i play them with mod_shout
> some work fine but others give (time out) error.
>
> How can i play mostly stream in freeswitch?
>
> Thanks
>
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From steveayre at gmail.com  Mon Dec 15 03:32:03 2014
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 15 Dec 2014 00:32:03 +0000
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call
	request
In-Reply-To: <CAEJMVkBjD-vvVCDuKEyYj9OuNSa5P3BorXkQUGu8PTBYZ5q2Qg@mail.gmail.com>
References: <CAFiqYum79cR0Tfa-Guo+R+3YruNUTm_jeZAsf4WsL7SZMB6E=w@mail.gmail.com>
	<D0AF1E70.8F750%krice@freeswitch.org>
	<CACynCY473-rhBSU4ftwN=HzK1KJai8R-nAwwKnWhCLzb4c6ZyQ@mail.gmail.com>
	<CAKbxfG9HOnFwhz3F_54kKVC4xWZfWfwyOtpQb8pjSk3y0_Gosg@mail.gmail.com>
	<CACynCY7=0R3PZ7LTRWtf+nPRhRT7YAXJePemjn01Jb-B+-EOzQ@mail.gmail.com>
	<CAEJMVkBjD-vvVCDuKEyYj9OuNSa5P3BorXkQUGu8PTBYZ5q2Qg@mail.gmail.com>
Message-ID: <CAFiqYukCD2PmgpBwHBPEQbH3FYHq9BMx08LzHXcri22wz7kd0A@mail.gmail.com>

That shouldn't really matter though should it? IIRC static payload types
can be used for codecs other than the one statically assigned by IANA if
a=rtpmap is given... so "a=rtpmap:19 CN/8000" should allow 19 to be CN,
overriding the default 19 assignment (which is 'Reserved' apparently)


On 11 December 2014 at 18:07, Brian West <brian at freeswitch.org> wrote:
>
> Funny, payload 13 is CN, which is 0x13 in base16 or 19 in base10, Please
> file a JIRA, I would need to see a packet capture for this one.
>
>
> On Thu, Dec 11, 2014 at 11:47 AM, Maciej Bylica <mbsip at gazeta.pl> wrote:
>
>> Sure i'll try with jira
>>
>> Mac.
>>
>> 2014-12-11 17:41 GMT+01:00 Anthony Minessale <anthony.minessale at gmail.com
>> >:
>>
>>> For years I beg ppl to post issues to jira not the mailing list.
>>> if you start typing and keywords like "problem" "not working" "issue"
>>> appear in your text, you probably should be filing it to JIRA not here.
>>>
>>>
>>> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>>
>>>> Thanks for prompt reply.
>>>>
>>>> Steven, later today i will try to update FS to the latest git and check
>>>> this out.
>>>>
>>>> Ken, i am fine with CN, passing thru is what i really need, but the
>>>> question is why FS generates such ERRors and how to get out of this.
>>>>
>>>> Thanks
>>>> Mac.
>>>>
>>>> 2014-12-11 17:12 GMT+01:00 Ken Rice <krice at freeswitch.org>:
>>>>
>>>>>  CN is comfort noise... Your cisco is sending it, and freeswitch is
>>>>> passing it thru because you have told it to use proxy media mode...
>>>>>
>>>>>
>>>>>
>>>>> On 12/11/14 9:53 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>>>>
>>>>> Your version is 3 months old. Can you reproduce this on the latest git
>>>>> head?
>>>>>
>>>>> On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>>>>
>>>>> Hello,
>>>>>
>>>>> I am running FreeSWITCH Version
>>>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
>>>>> problem with some call requests setup.
>>>>> Here is one of them:
>>>>>
>>>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
>>>>> sofia/outside_1/20049112223344 []
>>>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
>>>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
>>>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send
>>>>> signal sofia/outside_1/20049112223344 [BREAK]
>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
>>>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT
>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
>>>>> (sofia/outside_1/20049112223344) State INIT
>>>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
>>>>> sofia/outside_1/20049112223344 SOFIA INIT
>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
>>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>>  ---
>>>>>  v=0
>>>>>  o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>>>>>  s=SIP Call
>>>>>  c=IN IP4 10.10.10.12
>>>>>  t=0 0
>>>>>  m=audio 24782 RTP/AVP 18 19
>>>>>  c=IN IP4 10.10.10.12
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=rtpmap:19 CN/8000
>>>>>
>>>>>  +++
>>>>>  v=0
>>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>>  s=FreeSWITCH
>>>>>  c=IN IP4 10.10.10.166
>>>>>  t=0 0
>>>>>  m=audio 19690 RTP/AVP 18 19
>>>>>  c=IN IP4 10.10.10.166
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=rtpmap:19 CN/8000
>>>>>
>>>>>  2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
>>>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
>>>>> 2014-09-17 23:11:20Z 64bit
>>>>>  Local SDP:
>>>>>  v=0
>>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>>  s=FreeSWITCH
>>>>>  c=IN IP4 10.10.10.166
>>>>>  t=0 0
>>>>>  m=audio 19690 RTP/AVP 18 19
>>>>>  c=IN IP4 10.10.10.166
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=rtpmap:19 CN/8000
>>>>>
>>>>> sending out SDP exactly as shown above.
>>>>> The other side is responsing with 100, then 180 Ringing wo/SDP then
>>>>> 183 w/SDP
>>>>>
>>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
>>>>> sofia/outside_1/20049112223344 entering state [proceeding][183]
>>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>>>>>  v=0
>>>>>  o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>>>>>  s=Dialogic-SIP
>>>>>  c=IN IP4 10.10.10.198
>>>>>  t=0 0
>>>>>  m=audio 10024 RTP/AVP 19 18
>>>>>  a=rtpmap:19 CN/8000
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=silenceSupp:off - - - -
>>>>>  a=ptime:20
>>>>>
>>>>>  2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
>>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>>  ---
>>>>>  v=0
>>>>>  o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>>>>>  s=FreeSWITCH
>>>>>  c=IN IP4 10.10.10.166
>>>>>  t=0 0
>>>>>  m=audio 29492 RTP/AVP 18 19
>>>>>  c=IN IP4 10.10.10.166
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=rtpmap:19 CN/8000
>>>>>
>>>>>  +++
>>>>>  v=0
>>>>>  o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>>>>>  s=FreeSWITCH
>>>>>  c=IN IP4 10.10.10.166
>>>>>  t=0 0
>>>>>  m=audio 29492 RTP/AVP 18 19
>>>>>  c=IN IP4 10.10.10.166
>>>>>  a=rtpmap:18 G729/8000
>>>>>  a=fmtp:18 annexb=yes
>>>>>  a=rtpmap:19 CN/8000
>>>>>
>>>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651
>>>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR]
>>>>> switch_core_media.c:2294
>>>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE]
>>>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344
>>>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
>>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
>>>>> sofia/outside_1/20049112223344 [KILL]
>>>>>
>>>>> and Freeswitch is terminating the call by using CANCEL.
>>>>>
>>>>> My modules.conf.xml config part looks like following:
>>>>>     <!-- Codec Interfaces -->
>>>>>     <load module="mod_spandsp"/>
>>>>>     <load module="mod_g723_1"/>
>>>>>     <load module="mod_g729"/>
>>>>>     <!--load module="mod_amr"/-->
>>>>>     <!--<load module="mod_ilbc"/>-->
>>>>>     <load module="mod_h26x"/>
>>>>>     <!--load module="mod_vp8"/-->
>>>>>     <!--load module="mod_b64"/-->
>>>>>     <!--<load module="mod_siren"/>-->
>>>>>     <!--<load module="mod_isac"/>-->
>>>>>     <!--<load module="mod_celt"/>-->
>>>>>     <!--load module="mod_opus"/-->
>>>>>
>>>>> Is it a problem with CN definition?
>>>>> Could you please lead me where the problem is located?
>>>>>
>>>>> Thanks in advance.
>>>>> Mac.
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> ------------------------------
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> --
>>>>> Ken
>>>>>
>>>>>
>>>>>
>>>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>>> Twitter: @FreeSWITCH
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://confluence.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Mon Dec 15 04:18:19 2014
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 14 Dec 2014 19:18:19 -0600
Subject: [Freeswitch-users] Invalid codec CN is tearing down the call
	request
In-Reply-To: <CAFiqYukCD2PmgpBwHBPEQbH3FYHq9BMx08LzHXcri22wz7kd0A@mail.gmail.com>
References: <CAFiqYum79cR0Tfa-Guo+R+3YruNUTm_jeZAsf4WsL7SZMB6E=w@mail.gmail.com>
	<D0AF1E70.8F750%krice@freeswitch.org>
	<CACynCY473-rhBSU4ftwN=HzK1KJai8R-nAwwKnWhCLzb4c6ZyQ@mail.gmail.com>
	<CAKbxfG9HOnFwhz3F_54kKVC4xWZfWfwyOtpQb8pjSk3y0_Gosg@mail.gmail.com>
	<CACynCY7=0R3PZ7LTRWtf+nPRhRT7YAXJePemjn01Jb-B+-EOzQ@mail.gmail.com>
	<CAEJMVkBjD-vvVCDuKEyYj9OuNSa5P3BorXkQUGu8PTBYZ5q2Qg@mail.gmail.com>
	<CAFiqYukCD2PmgpBwHBPEQbH3FYHq9BMx08LzHXcri22wz7kd0A@mail.gmail.com>
Message-ID: <CAKbxfG_n6fFNeH8qJ_F2sa6Tc9t5rZeKMmW-Q3_u+Yvc5p96sw@mail.gmail.com>

It doesn't matter.  Also bug should be fixed, it was not related to cn.

On Sunday, December 14, 2014, Steven Ayre <steveayre at gmail.com> wrote:

> That shouldn't really matter though should it? IIRC static payload types
> can be used for codecs other than the one statically assigned by IANA if
> a=rtpmap is given... so "a=rtpmap:19 CN/8000" should allow 19 to be CN,
> overriding the default 19 assignment (which is 'Reserved' apparently)
>
>
> On 11 December 2014 at 18:07, Brian West <brian at freeswitch.org
> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>> wrote:
>>
>> Funny, payload 13 is CN, which is 0x13 in base16 or 19 in base10, Please
>> file a JIRA, I would need to see a packet capture for this one.
>>
>>
>> On Thu, Dec 11, 2014 at 11:47 AM, Maciej Bylica <mbsip at gazeta.pl
>> <javascript:_e(%7B%7D,'cvml','mbsip at gazeta.pl');>> wrote:
>>
>>> Sure i'll try with jira
>>>
>>> Mac.
>>>
>>> 2014-12-11 17:41 GMT+01:00 Anthony Minessale <
>>> anthony.minessale at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','anthony.minessale at gmail.com');>>:
>>>
>>>> For years I beg ppl to post issues to jira not the mailing list.
>>>> if you start typing and keywords like "problem" "not working" "issue"
>>>> appear in your text, you probably should be filing it to JIRA not here.
>>>>
>>>>
>>>> On Thu, Dec 11, 2014 at 10:37 AM, Maciej Bylica <mbsip at gazeta.pl
>>>> <javascript:_e(%7B%7D,'cvml','mbsip at gazeta.pl');>> wrote:
>>>>
>>>>> Thanks for prompt reply.
>>>>>
>>>>> Steven, later today i will try to update FS to the latest git and
>>>>> check this out.
>>>>>
>>>>> Ken, i am fine with CN, passing thru is what i really need, but the
>>>>> question is why FS generates such ERRors and how to get out of this.
>>>>>
>>>>> Thanks
>>>>> Mac.
>>>>>
>>>>> 2014-12-11 17:12 GMT+01:00 Ken Rice <krice at freeswitch.org
>>>>> <javascript:_e(%7B%7D,'cvml','krice at freeswitch.org');>>:
>>>>>
>>>>>>  CN is comfort noise... Your cisco is sending it, and freeswitch is
>>>>>> passing it thru because you have told it to use proxy media mode...
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 12/11/14 9:53 AM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>>>>>
>>>>>> Your version is 3 months old. Can you reproduce this on the latest
>>>>>> git head?
>>>>>>
>>>>>> On 11 December 2014 at 13:45, Maciej Bylica <mbsip at gazeta.pl> wrote:
>>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> I am running FreeSWITCH Version
>>>>>> 1.5.14b+git~20140917T231120Z~8f85b5204c~64bit, proxy media mode and heaving
>>>>>> problem with some call requests setup.
>>>>>> Here is one of them:
>>>>>>
>>>>>> 2014-12-11 13:38:15.034612 [NOTICE] switch_channel.c:1055 New Channel
>>>>>> sofia/outside_1/20049112223344 []
>>>>>> 2014-12-11 13:38:15.034612 [DEBUG] mod_sofia.c:4579
>>>>>> (sofia/outside_1/20049112223344) State Change CS_NEW -> CS_INIT
>>>>>> 2014-12-11 13:38:15.034612 [DEBUG] switch_core_session.c:1388 Send
>>>>>> signal sofia/outside_1/20049112223344 [BREAK]
>>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:472
>>>>>> (sofia/outside_1/20049112223344) Running State Change CS_INIT
>>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_state_machine.c:512
>>>>>> (sofia/outside_1/20049112223344) State INIT
>>>>>> 2014-12-11 13:38:15.054610 [DEBUG] mod_sofia.c:87
>>>>>> sofia/outside_1/20049112223344 SOFIA INIT
>>>>>> 2014-12-11 13:38:15.054610 [DEBUG] switch_core_media.c:7510
>>>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>>>  ---
>>>>>>  v=0
>>>>>>  o=CiscoSystemsSIP-GW-UserAgent 6016 7716 IN IP4 10.10.10.226
>>>>>>  s=SIP Call
>>>>>>  c=IN IP4 10.10.10.12
>>>>>>  t=0 0
>>>>>>  m=audio 24782 RTP/AVP 18 19
>>>>>>  c=IN IP4 10.10.10.12
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>
>>>>>>  +++
>>>>>>  v=0
>>>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>>>  s=FreeSWITCH
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  t=0 0
>>>>>>  m=audio 19690 RTP/AVP 18 19
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>
>>>>>>  2014-12-11 13:38:15.054610 [DEBUG] sofia_glue.c:1228
>>>>>> sofia/outside_1/20049112223344 sending invite version: 1.5.14b git 8f85b52
>>>>>> 2014-09-17 23:11:20Z 64bit
>>>>>>  Local SDP:
>>>>>>  v=0
>>>>>>  o=FreeSWITCH 0255539876 0255539877 IN IP4 10.10.10.166
>>>>>>  s=FreeSWITCH
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  t=0 0
>>>>>>  m=audio 19690 RTP/AVP 18 19
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>
>>>>>> sending out SDP exactly as shown above.
>>>>>> The other side is responsing with 100, then 180 Ringing wo/SDP then
>>>>>> 183 w/SDP
>>>>>>
>>>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6423 Channel
>>>>>> sofia/outside_1/20049112223344 entering state [proceeding][183]
>>>>>>  2014-12-11 14:29:12.214608 [DEBUG] sofia.c:6433 Remote SDP:
>>>>>>  v=0
>>>>>>  o=Dialogic_SDP 11471558 0 IN IP4 10.10.10.218
>>>>>>  s=Dialogic-SIP
>>>>>>  c=IN IP4 10.10.10.198
>>>>>>  t=0 0
>>>>>>  m=audio 10024 RTP/AVP 19 18
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=silenceSupp:off - - - -
>>>>>>  a=ptime:20
>>>>>>
>>>>>>  2014-12-11 14:29:12.214608 [DEBUG] switch_core_media.c:7510
>>>>>> sofia/outside_1/20049112223344 Patched SDP
>>>>>>  ---
>>>>>>  v=0
>>>>>>  o=FreeSWITCH 0351957506 0351957507 IN IP4 10.10.10.166
>>>>>>  s=FreeSWITCH
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  t=0 0
>>>>>>  m=audio 29492 RTP/AVP 18 19
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>
>>>>>>  +++
>>>>>>  v=0
>>>>>>  o=FreeSWITCH 0351957506 0351957508 IN IP4 10.10.10.166
>>>>>>  s=FreeSWITCH
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  t=0 0
>>>>>>  m=audio 29492 RTP/AVP 18 19
>>>>>>  c=IN IP4 10.10.10.166
>>>>>>  a=rtpmap:18 G729/8000
>>>>>>  a=fmtp:18 annexb=yes
>>>>>>  a=rtpmap:19 CN/8000
>>>>>>
>>>>>> 2014-12-11 14:29:12.214608 [ERR] switch_core_codec.c:651
>>>>>> *Invalid codec CN! *2014-12-11 14:29:12.214608 [ERR]
>>>>>> switch_core_media.c:2294
>>>>>> *Can't load codec? *2014-12-11 14:29:12.214608 [NOTICE]
>>>>>> switch_core_media.c:2295 Hangup sofia/outside_1/20049112223344
>>>>>> [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
>>>>>> 2014-12-11 14:29:12.214608 [DEBUG] switch_channel.c:3222 Send signal
>>>>>> sofia/outside_1/20049112223344 [KILL]
>>>>>>
>>>>>> and Freeswitch is terminating the call by using CANCEL.
>>>>>>
>>>>>> My modules.conf.xml config part looks like following:
>>>>>>     <!-- Codec Interfaces -->
>>>>>>     <load module="mod_spandsp"/>
>>>>>>     <load module="mod_g723_1"/>
>>>>>>     <load module="mod_g729"/>
>>>>>>     <!--load module="mod_amr"/-->
>>>>>>     <!--<load module="mod_ilbc"/>-->
>>>>>>     <load module="mod_h26x"/>
>>>>>>     <!--load module="mod_vp8"/-->
>>>>>>     <!--load module="mod_b64"/-->
>>>>>>     <!--<load module="mod_siren"/>-->
>>>>>>     <!--<load module="mod_isac"/>-->
>>>>>>     <!--<load module="mod_celt"/>-->
>>>>>>     <!--load module="mod_opus"/-->
>>>>>>
>>>>>> Is it a problem with CN definition?
>>>>>> Could you please lead me where the problem is located?
>>>>>>
>>>>>> Thanks in advance.
>>>>>> Mac.
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> ------------------------------
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Ken
>>>>>>
>>>>>>
>>>>>>
>>>>>> *http://www.FreeSWITCH.org <http://www.FreeSWITCH.org>
>>>>>> http://www.ClueCon.com <http://www.ClueCon.com> http://www.OSTAG.org
>>>>>> <http://www.OSTAG.org> *irc.freenode.net #freeswitch
>>>>>> Twitter: @FreeSWITCH
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://confluence.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>> <javascript:_e(%7B%7D,'cvml','brian at freeswitch.org');>
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
>> http://www.freeswitchbook.com
>> http://www.freeswitchcookbook.com
>>
>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378)
>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','consulting at freeswitch.org');>
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users at lists.freeswitch.org');>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>

-- 
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
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From sphotos1982 at gmail.com  Sat Dec 13 15:46:36 2014
From: sphotos1982 at gmail.com (spros p)
Date: Sat, 13 Dec 2014 14:46:36 +0200
Subject: [Freeswitch-users] LUA ESL Help
In-Reply-To: <B335E1470D244A69BB82E563843B4E7F@gmail.com>
References: <CAJ0dgAzm1wsHg4GfNi+jaiaHSx8EZfjsY3f4Dhvc3Nm7NRi-NQ@mail.gmail.com>
	<B335E1470D244A69BB82E563843B4E7F@gmail.com>
Message-ID: <CAJ0dgAwxh7+jCSNZS3b4dpTmpk8Sz+GDMhvRHqumde0JctvX7A@mail.gmail.com>

Thanks Chad - I don't think copying the ESL.so in the paths is the correct
approach for resolving. I have in any case made a symbolic link of the
ESL.so file to both /usr/share/lua/5.1 and /usr/lib64/lua/5.1 with no avail.

My suspicion is that LUA_PATH and LUA_CPATH need to be defined with ESL
details. This is fine for the CPATH, as i know where ESL.so resides.

However, I cannot define PATH, as there doesn't seem to be a ESL.lua file.

Best,
Spyros


On Sat, Dec 13, 2014 at 7:13 AM,