January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

need help on call recording
by haloha 16 Jan '13

16 Jan '13

hi all i try to record all calls to/from local extension so my dialplan <extension name="Local_Extension"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 2011"/> <action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/> <action application="set" data="RECORD_ARTIST=FreeSWITCH"/> <action application="set" data="RECORD_COMMENT=FreeSWITCH"/> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="/etc/freeswitch/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${caller_id_number}_${destination_number}.wav"/>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/>  <action application="set" data="call_timeout=30"/>  <action application="set" data="hangup_after_bridge=true"/>  <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>  <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> FS only records the caller voicem FS doesnt record the called voice How do i record the entire calls(caller and called) Thank you Ha`

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ZRTP error on call setup when dialling client uses ZRTP
by Ben 16 Jan '13

16 Jan '13

Hi all, My server is running the latest git stable build of Freeswitch, all config is default and server/clients are on simple LAN setup. I'm having an issue when using ZRTP on one client only, specifically the dialling client. The calling device has ZRTP enabled, and the called device does not. In this scenario, the FS server should act as a man in the middle and exchange keys with the calling device. This results in the A leg being encrypted and the B leg un-encrypted. However as soon as the key exchange initiates, the A client errors and tears down the call. What's strange is that when I initiate the call from the non-zrtp client to the zrtp client, the keys exchange fine between server and client A, and 2-way audio is present. So it's only when it's the dialling handset do I have the problem. I've then gone on and tested the same thing on the latest git, and the problem goes away. So my question is, can anyone advise on what changes/patches have been applied to the latest git that has resolved this issue? I've tried to search the git tree, but I'll be honest I'm a bit unfamiliar with how it works and I'm not sure that I'm searching properly. The ZRTP device is an iPhone running Groundwire, and it doesn't seem to matter what codec I use. Thanks for the help!

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Issue - playing mp4 video files
by Muqtheear S 16 Jan '13

16 Jan '13

1. Please tell me how to convert mpeg4-generic encoded audio data to PCMU encoded audio data in .mp4 file. I tried with ffmpeg, but couldn't succeed. 2. If you have any mp4 file with PCMU sound track, please share with me. 3. Does freeswitch comes with mpeg4-generic codec support? Thanks & Regards, Muqtheear.S

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configuring Acoustic Model for german language problem
by Shahzad Bhatti 16 Jan '13

16 Jan '13

Hello Everybody, i am trying to configure Acoustic Model for german language on localhost as wiki link http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Acoustic_Model_for_german_… but when i use ./etc/voxforge_de_sphinx_train.transcription i got error ./etc/voxforge_de_sphinx_train.transcription: line 1: syntax error near unexpected token `(' ./etc/voxforge_de_sphinx_train.transcription: line 1: `<s> MAN UNTERSCHEIDET ABSCHLUSSFREIHEIT </s> (anonymous-20080117-ihl_de5-040)' any positive reply will be highly appreciated. regards shahzad bhatti

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Is Asterisk snappier than FreeSWITCH to set up calls?
by Emrah 15 Jan '13

15 Jan '13

Ever since I converted to FS, I consistently felt a little delay in setting up my calls. I thought it was just me being too picky or that the roundtrip required for the digest auth was the cause… Since I now learned that Digest is also required by Asterisk, I am out of ideas… I tested both SIP servers in parallel and captured a trace. On my Asterisk, the extension 114 dials an IAX peer. The kind of thing that should add more delay… On my FS box, it's milliwatt, executed immediately. Asterisk: http://pastebin.freeswitch.org/20445 FS: http://pastebin.freeswitch.org/20444 Why is Asterisk snappier? I set all log levels to 0 on my FS box… Thanks for any ideas. Best, Emrah

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Temporarily stop/start Freeswitch 200OKs on OPTIONS Ping?
by Adelia C. 15 Jan '13

15 Jan '13

Is there a way to do this? I found some old Q&As on this forum that reject this as non-spec compliant (I agree) BUT ... I need to disable 200OKs short period of time. This is my use case: I want to stop new traffic to specific Freeswitch servers (drain them) previous to a new deploy. There are reasons for deploying this way (as opposed to just dropping the new code in) which I will not explain here. For now, I need to do it this way. I don't control the SBC so the easiest way is to stop replying to OPTIONS on the FS servers to go out of commission. The SBC will stop forwarding new calls while handling those in progress. Can I do this? Thank you, Adelia

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Can't restablish media
by Javier Gallart 15 Jan '13

15 Jan '13

Hello I was wondering if someone has experienced this situation. I have a call with bypass_media=true; and this call fails with cause 488. I catch that faliure in a lua script where i set bypass_media=false and change the codec via the variable absolute_codec_string, expecting the Sangoma D500 takes care of the trasncoding . This second call is successful, however when a 183 is received at the b-leg, I see this debug message at the console: 2013-01-15 17:24:27.489568 [ERR] switch_ivr.c:1487 Can't re-establsh media on sofia/internal/xxxxxx(a)y.y.y.y The 183 message is absorbed by FS, and there is no media transmitted back to a-leg (therefore no ringback heared). However the 200 OK is correctly transmitted, and the calls works apparently works fine. Any idea? This happens at least with Freeswitch versions 1.2.1, 1.2.3 and 1.2.5.3 Thanks in advance Javi

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Re: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension
by Raimund Sacherer 15 Jan '13

15 Jan '13

Thank you Dmitry, are you saying you only use the opensource capture part from voip-monitor on all your 10 FS servers and you do not use their webinterface at all? If I understood this correctly, this opensource does all the capturing and it calculates MOS and Loss scores and stores it in a central MySQL database? That sound's great, I thought the MOS and Loss calculations are in their proprietery part, but if it's included in the opensource part it could be interesting to create your own web-visualization. On the other hand their licence seems to be fairly cheap, some 480 euros ... but I am not sure if this is per server you monitor, or per server you deploy the web-interface to. best, Rai ----- Original Message ----- From: "Dmitry Sytchev" <kbdfck(a)gmail.com> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Viernes, 11 de Enero 2013 14:35:33 Subject: Re: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension We use VoipMonitor open-source part, with homebrew custom WEB-UI. This is a really cool software, it continuously evolves and captures pretty well even large amounts of VoiP traffic. I highly recommend this software, since it doesn't only captures signaling traffic, but also can capture media/rtcp, automatically convert captured calls to wav files or store raw pcaps for further processing. it also can analyze MOS/Loss scores and many many other different parameters and store them in database per call. We use single instance to store data for investigation of calls with problems on more than 10 servers. 2013/1/11 Raimund Sacherer < raimund.sacherer(a)logitravel.com > Does anyone have any experience with voip monitor? http://www.voipmonitor.org/ We will start evaluating it in our environment as we are switching more and more to SIP providers, therefore I would love to have better Monitoring. What voipmonitor says is that they are able to filter the calls in realtime with low overhead an if call quality drops, they can alert you, and they seem to have a rather nice web interface and they also are used by others to store the RTP's ... What seems to be especially nice is that their capture daemon (as they state it) adds very little overhead to a system, so depending on call quantity they say it's neglect-able. Would love to hear some comparisions between this and HOMER if some can provide it. Thank you, best Ray From: "Cal Leeming [Simplicity Media Ltd]" < cal.leeming(a)simplicitymedialtd.co.uk > To: "FreeSWITCH Users Help" < freeswitch-users(a)lists.freeswitch.org > Sent: Domingo, 6 de Enero 2013 21:01:24 Subject: [Freeswitch-users] Bounty for - FS-4268 Sip trace with filter for IP or extension Could a core dev speculate on how much the expected bounty would be for this? Thanks Cal _________________________________________________________________________ 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 -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 Síguenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para móvil Logitravel.com Este correo electrónico y, en su caso, cualquier fichero anexo, contiene información de carácter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgación, copia o distribución a terceros sin la previa autorización escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electrónico por error, se ruega notifíquese inmediatamente esta circunstancia mediante reenvío a la dirección electrónica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formarán parte de un fichero registrado como CLIENTES con número de inscripción 2070610043 en la Agencia General de Protección de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposición, rectificación y cancelación a estos datos que deberá ejercer mediante escrito a la dirección anteriormente citada. _________________________________________________________________________ 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 -- Best regards, Dmitry Sytchev, IT Engineer _________________________________________________________________________ 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 --

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Re: [Freeswitch-users] How to pause and resume play an audio file in conference?
by 王迪 15 Jan '13

15 Jan '13

I will try uuid_fileman on a channel. How to file the wihsklist Jira? Create a improvement issue or new feature issue on Jira? 1. Re: How to pause and resume play an audio file in conference? (Steven Ayre) ------------------------------------------------------------ From: Steven Ayre <steveayre(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] How to pause and resume play an audio filein conference? Date: Mon14 Jan 2013 12:51:14 +0000 I suspect it needs a new feature to be added to mod_conference. You could file a wishlist Jira. Or as a workaround, originate a channel that calls into the conference and play the audio on that channel. You'd then be able to use uuid_fileman on that channel. -Steve On 14 January 2013 12:50, Steven Ayre <steveayre(a)gmail.com> wrote:

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Chatplan: is msg received success?
by Pip Live 15 Jan '13

15 Jan '13

Hi, I use Freeswitch to handle SIP message between SIP extentions, how can server know does the message be received by client or not? I noticed even if a SIP extention is offline/unregister, SIP message will still be sent by server but just the receiver will never get it (even if later the receiver online/registered again). What I want to achieve is: send SIP message to an extention, if received OK, nothing happen; if not, forward the message to user's email address. brgds, Alex

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