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[mod_fsv] How the video recording really looks like?
by K K 26 Feb '13

26 Feb '13

Hi, I would like to record the both sides video on FreeSWITCH during the video call between two UAC. I would like to record each video call separately. I have tried the mod_fsv by adding following configuration line: <action application="record_fsv" data="$${base_dir}/recordings/video/${strftime(%Y-%m-%d-%H-%M-%S)}_$1_${caller_id_number}.fsv"/> under the <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> The problem is that after adding that now when I make a call to any extension I am automatically connected and the *.fsv file is being created. The party which I have called did not receive any incoming call. It seems that I am getting automatically the connection to FreeSWITCH which is recording my video and not moving my call to the destination. So i guess that is the idea of "video recording" whit mod_fsv? I am not able to achieve my goal?

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Best current ATA and Setup for Faxing
by Chris Ferreira 26 Feb '13

26 Feb '13

Hi Folks, We had previously been working with the Cisco SPA112 for T.38 Faxing through Freeswitch and out to Flowroute. At the time we didn't get it working 100% perfectly. I now have one of these SPA112's that I need to replace because someone fried it by plugging in a much larger power supply by accident. Is my scenario above still a good starting point for Faxing or should I be looking at another ATA or provider other than Flowroute? Also, one more question, and this may be better suited to the business list (let me know if I should move it there), but we are looking for a part time as needed remote or local Freeswitch Admin / Engineer / Tech. Our previous person is MIA. Thanks Everyone! -Chris

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Favorite CNAM provider
by Steven Schoch 25 Feb '13

25 Feb '13

Now that I got Caller-ID-Name working, I need to choose a provider. I have it setup now to use opencnam.com, but many numbers are "UNKNOWN" (which I'm not complaining about, because that service is free). There is also VoIPCNAM.com, which costs $2 a month, no big deal and well worth it, but I want to make sure it has good data. Feedback anyone? -- Steve

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How do I execute shell command from Lua?
by JP 25 Feb '13

25 Feb '13

I entered this into my Lua script local f = io.popen("pwd") -- runs command local o = f:read("*a") -- read output of command and executed the script from fs_cli. This only caused the fs_cli to become unresponsive. Is there something I am doing wrong? Thanks JP

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Re: [Freeswitch-users] Door intercom/gate controller
by Mr Nathan Downes 25 Feb '13

25 Feb '13

Hi Michael, Thanks for the response, it worked in the opposite direction, the gate controller (OOB) calls phones (Inband) kind of like the door thing on an apartment block. Start_dtmf worked if I called the gate controller from the phones, but not in the other direction which is how it will be used. It has been resolved as it was discovered the ATA devices inbuilt to the fibre ONT have an option for OOB DTMF, enabling this on the POTS profile allowed the device to function. I am still interested on how I could of got it to work though, is start_dtmf a one directional app? Thanks, Nathan From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, 16 February 2013 6:35 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Door intercom/gate controller Can you show us the dialplan for sending a call to the gate controller? Also, pastebin the console debug output for a call to the gate controller where you press digits but the controller doesn't respond. Thanks, MC On Wed, Feb 13, 2013 at 2:42 AM, Mr Nathan Downes <nathandownes(a)hotmail.com> wrote: Hi list, I have a gate controller that only understands RFC 2833 but it calls endpoints that can only provide inband DTMF, I can't seem to get the digits recognised by the gate controller to trigger the relay using start_dtmf_generate, as they are coming back to the a-leg? Rather than out the b-leg.. It works fine when I call a SPA502g as it will provide RFC 2833.. is there some trickery I can achieve this? Or am I just doing it wrong?? J _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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"Outgoing Call" on phone display
by Steven Schoch 25 Feb '13

25 Feb '13

When I place a call from my Polycom phone and the remote party answers, the display changes to say "Outgoing Call(<number>)" when the remote party answers. This is annoying because it makes the important bit of information (the actual number) scroll off the right of the screen. I'm trying to figure out where this "Outgoing Call" string is coming from. I ran an SIP trace and found some messages that get sent to the phone when the remote party answers. One of these messages includes this line: Remote-Party-ID: "Outbound Call" <1xxxxxxxxxx>;party=calling;privacy=off;screen=no A later message includes this line: P-Asserted-Identity: "Outbound Call" <1xxxxxxxxxx> Where are these coming from, and can I change them? -- Steve

4 9

Configure
by Venkateshwaran Thirugnanam 25 Feb '13

25 Feb '13

Hi All, How to configure 3 freeswitch server to Centralized mysql DB? Please guild Regards, Kumaran T

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Bounty for FS-5118 - Tel URI parameters (npdi, rn) appearing in SIP URI on outbound calls
by Victor Chukalovskiy 25 Feb '13

25 Feb '13

Hello, I'm offering bounty to have npdi and rn= issue sorted out. I'll be also posting this in the bounty section of the WiKi shortly. Amount depends on the scope of work. If I get it right, FS-5118 needs to be first reviewed by developers to move any further with this. I'm ready to offer my feedback and detailed description of what is needed. At this point, I see two different path: (1) Attack FS-5118 directly. That is, provide a way to pass tel URI parameters correctly when bridging call to a SIP gateway OR (2) Come-up with channel variables for npdi or npdi= and for rn= that are: -Automatically inserted into tel URI in the INVITE on outbound SIP calls -Automatically populated with parsed values on incoming SIP call (to avoid doing this in the dial-plan http://wiki.freeswitch.org/wiki/Regular_Expression#LNP_.28Local_Number_Port…) I believe FS project will benefit from the ability to easily send and receive npdi or npdi= and rn= (ideally, by means of dedicated channel variables). -Victor

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Can local_stream MOH have a time limit/max duration?
by Mario G 25 Feb '13

25 Feb '13

Could not find anything in the wiki and this may not be possible but... I have early media with instructions, is has voice, ring, voice, music. I want to change the ring to music but I could not find a way to limit the time the music plays. See the second example, I would like to limit the first moh to a maximum time. If not possible, is this a "bounty" enhancement? Thanks for any help. Mario G Currently: <action application="set" data="ringback=file_string://${lua_ring1}!tone_stream://${us-ring};loops=3!file_string://${lua_ring2}!local_stream://moh"/> Would Like (add something to limit the first moh): <action application="set" data="ringback=file_string://${lua_ring1}!local_stream://moh!file_string://${lua_ring2}!local_stream://moh"/>

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Polycom specific problem
by Sean Devoy 25 Feb '13

25 Feb '13

This was initially reported to me as a call being dropped issue. I have just had an update that says if the user stays on the line for "a few seconds or so" the call returned to its previous state. Now I am skeptical if it is a FS issue at all. Alas, you all are my best source of information so I appeal to you with my question and undying gratitude. My first question for you Polycom users is "Does Polycom have a different call waiting tone when the users is dialing out and not connected yet?" The current issue, reported by several users, is intermittent. If the Polycom user is dialing out (and the call is not answered yet) and a new call comes in (incoming calls ring all 6 extensions) the Polycom user gets a "busy signal" for 1 to 2 seconds and then it returns to their ringing outgoing call. They have been absolutely adamant that it is not the "nice little call waiting tone" they get when a call is connected and another call comes in. It is only in this "pre-connected state" that they get the "busy signal" and they swear it is intermittent. I will add that the phone configuration is Line ringing on both extensions on the phone. I have turned on global siptrace to capture everything I can the next time it happens. Any ideas? Sean

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