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FREESWITCH CONFERENCE BLF
by alberto Villa 25 May '16

25 May '16

Hello, I have 4 phones, 3 will join a conference room called 'myConf' and I'm trying to get BLF status for this conference on the fourth phone, which will eventually join the conference by pulling the related BLF button. On the phone I want BLF I set to monitor the status using 'sip:myConf@192.168.0.252' (which works when monitorning a normal sip user), then I tried the followings: Set the <advertise> field like this <advertise> <room name="myConf@$${subdomain}"/> </advertise> in conf/autoload_configs/conference.conf.xml but this doesn't work. Set: <action application="set" data="presence_id=myConf@${domain_name}"/> in my dialplan before setting up the conference, still doesn't work. Issuing: presence in myConf(a)192.168.0.252 on-the-phone "on the phone" from fs_cli, but still doesn't work. The only thing I got working is to manually send the event using ESL like this: event = freeswitch.Event('PRESENCE_IN'); event:addHeader('proto', 'sip'); event:addHeader('from', 'myConf'..'@'..'192.168.0.252'); event:addHeader('login', 'myConf'..'@'..'192.168.0.252'); event:addHeader('event_type', 'presence'); event:addHeader('alt_event_type', 'dialog'); event:addHeader('Presence-Call-Direction', 'outbound'); event:addHeader('state', 'active'); event:addHeader('unique-id', uuid); event:addHeader('event_count', '1') event:addHeader('answer-state', 'confirmed'); event:fire(); In this case the BLF correctly lights up, but when the phone issues a new "SUBSCRIBE" request the BLF is cleared. Wrapping up: is there a way to use BLF to monitor the status of a conference call from a phone not invited when the conference is set up? Thank you -- Dr. Villa Alberto Sw Engineer SeleTech srl via Volturno, 37 20861 Brugherio (MB) tel: +39 039 884053 email: a.villa(a)seletech.com web: www.seletech.com www.seletech.eu --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

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How to communicate between mod_v8 and mod_java
by Николай 25 May '16

25 May '16

I have javascript running on mod_v8 that controls my calls redirecting. And now I need execute java program using mod_java to get some data. What is the simplest way to call a method in java class (session.execute?) and return result back to javascript? Kind regards, Nick.

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Call forwarding
by Nduwayezu, Joselyne 25 May '16

25 May '16

Hello, I would like to forward did call to an extension. I'm sure that what i did is correct because it generates an erroo like: cannot create outgoing channel of type USER cause User is not registered. 1) I've created as user related to the did in /directory/default.xml 2) made chenge in Local_extension, contion field= ${sip_to_user} expression=my_did 3)added an extension related to my_did in public.xml Is that way i have to do to forward my call to an extension? What is the difference between call forward and call transfer? Thank you NDUWAYEZU Joselyne -- Haute École de Bruxelles

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Fax....How I hate you.
by Blake Priddy 24 May '16

24 May '16

Is anyone out there having success with sending/receiving faxes with Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I have referred myself to this document https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I was just wanting to see what everyone is doing out there with flowroute for the people who still must use faxing.. -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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ClueCon 2016 Sponsorship opportunities!
by Kathleen King 24 May '16

24 May '16

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No audio if call answered after 30 seconds
by Stefano Favaro 24 May '16

24 May '16

Hello, I have a problem (probably with nat) with calls that are answered after 30 seconds. In these cases I have no audio. It seems that this happens only if users are connected to my FreeSwitch using some specific italian internet providers. In all other cases rtp is ok. I have made several tests, and with Italian carrier "Wind" I have always found this problem. Is there any tips to solve this problem? Thanks. Stefano Favaro Sviluppo Servizi ed Applicazioni _____________________ Edistar Srl a socio unico soggetta a direzione e coordinamento di YourVoice SpA Via Artigianato 1 – I – 31050 Vedelago (TV) Italy Phone +39 0423 7331 – Fax +39 0423 733133 mobile: +39 3456636386 skype: stefanofavaro www.edistar.com Le informazioni trasmesse attraverso la presente e-mail ed i suoi allegati sono dirette esclusivamente al destinatario e devono ritenersi riservate con divieto di diffusione e di uso nei giudizi salva espressa autorizzazione; nel caso di utilizzo senza espressa autorizzazione, potrà essere effettuata denuncia alla competente Autorità. La diffusione e la comunicazione da parte di soggetto diverso dal destinatario è vietata dall’art. 616 e ss. c.p. e dal d. l.vo n. 196/03. Se la presente e-mail e i suoi allegati fossero stati ricevuti per errore da persona diversa dal destinatario preghiamo di distruggere quanto ricevuto e di rinviare al mittente con lo stesso mezzo. Grazie per la collaborazione. This message may contain confidential and/or privileged information. If you are not the addressee or authorized to receive this for the addressee, you must not use, copy, disclose or take any action based on this message or any information herein. If you have received this message in error, please advise the sender immediately by reply e-mail and delete this message. Thank you for your cooperation.

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Lua freeswitch.email question
by Abaci B 23 May '16

23 May '16

Is there a way to send an email from Lua using freeswitch.email <https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaA…> that has multiple files attached?

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Sangoma Card on Windows
by edrperes 23 May '16

23 May '16

Is possible to use Sangoma USBfxo with FreeSwitch on Windows operating system? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Sangoma-Card-on-Windows-tp759… Sent from the freeswitch-users mailing list archive at Nabble.com.

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How to set/chanage DTMF Mode triples: Suppress, Detect, Regenerate?
by Andrew Newlands 23 May '16

23 May '16

I would like to dynamically manage the "DTMF Mode" during a call, by controlling the options described here: https://freeswitch.org/confluence/display/FREESWITCH/DTMF As follows.... DTMF behavior for a single audio path can be captured by a triple of booleans as follows: Mode Suppress Detect Regenerate passthrough . . . passthrough . X . Intercept X . . Intercept X X . Intercept X X X This looks ideal for my scenario as I need to use different settings for each channel on the call, and will also need to change the settings "on the fly". Can anyone please tell me how/where I can set/change the different options? FYI: I need to support inband and rfc2833 as I have no control over what providers will send). Many thanks. Andy

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Can't get rtcp to work...
by Allan Kristensen 23 May '16

23 May '16

Hello, I've recently moved from Asterisk to Freeswitch and I'm trying to setup a FS on a cloud solution. I have mixed results from previous projects with Asterisk on cloud servers, but this one is just going to proxy media, so I'll give it a shot ;-) I'm going to log the rtcp stats into the cdr, so I can be sure things are ok, but most of the rtcp related stats all contain "0" and mos score 4.5 (even if I randomly drop rtp packets for a couple of seconds to simulate packet loss... not good). The client uses an asterisk, which is sending rtcp reciever reports to FS, however FS is not listening on the rtcp port at all (rtp port +1) and server just sends back icmp unreach. (it's not muxed into the rtp) Also FS is not sending rtcp packets, from what I can see, so I suspect rtcp is somehow not enabled for the channel / session at all. I've added the "rtcp-audio-interval-msec" to the sip profile, so it should work, but it's not.. I've checked the source and even enabled the RTCP_DEBUG in the source to get a clue, but no help there. Asterisk is not actually adding anything about rtcp to the sdp, but is that really needed to get the rtcp working? I'm not using ICE or anything, just a simple setup for this. Is there any way to see if FS is disabling the rtcp feature and if so, why? Any ideas on what can be done to get things working? ;-) Thank you and have a excellent day! / Allan

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