August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

conference video cordinates position
by Mirko Brankovic 05 Aug '16

05 Aug '16

Hi, Was anyone trying to get events like video position in the vid-layout? It would be cool to broadcast each video cordinates in event (maybe on each change) so that we can get something like [ALERT] mod_verto.c:5207 EVENT BROADCAST FSevent.custom::conference::maintenance { "eventChannel": "FSevent.custom::conference::maintenance", "data": { "Event-Name": "CUSTOM", "Core-UUID": "c4c53039-e6fa-4515-adb4-ce14a8674687", ... "Event-Subclass": "conference::maintenance", "Conference-Name": "mirko", "Conference-Size": "1", "Conference-Ghosts": "0", "Conference-Profile-Name": "test", "Conference-Unique-ID": "2db98498-8d54-4c3c-be4a-df03ddbd92e9", } } globals.enable_presence = SWITCH_TRUE; globals.enable_fs_events = SWITCH_TRUE; -- Regards, Mirko

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EXECUTE_ON_ANSWER with Bridge
by Don Hawkins 05 Aug '16

05 Aug '16

Simple yet complicated issue I've been working on for about 48 hours. I could really use some assistance.... 1. I'm bridging calls via httapi and that is working fine. <execute action="$action" application="bridge" data="sofia/gateway/voipms/2145551212"></execute> 2. I want to be notified at my URL via CURL when, A: The call starts ringing and B: The call is answered. I'm using this code for that:    <execute application=\"export\" data=\"nolocal:execute_on_answer=curl http://myurl.com/test3.php post event=RingRing_Answer&\${uuid}&state=\${state}&direction=\${direction}\"></execute> The problem comes because it's a bridge, as far as FreeSWITCH is concerned once it's connected to the gateway and the gateway answers, the "execute_on_answer" should be triggered, but i need it to execute when the party actually answers the phone, not when the gateway "answers". There must be a simple way around this, open to suggestions, thanks!

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spandsp.conf.xml for Brazil
by Gonzalo Gasca Meza 05 Aug '16

05 Aug '16

Hi all, Looking to see if someone can provide a sample configuration for Brazil for spandsp.conf.xml Im interested in SIT, REORDER and BUSY or guidance in configuring this file. Here are the Brazil tones: http://www.itu.int/pub/T-SP-OB.781-2003 Frequency in HZ value is understandable, seems to be min and max are tone duration. Where can I find these reference? This is the US sample: <descriptor name="1"> <tone name="CED_TONE"> <element freq1="2100" freq2="0" min="700" max="0"/> </tone> <tone name="SIT"> <element freq1="950" freq2="0" min="256" max="400"/> <element freq1="1400" freq2="0" min="256" max="400"/> <element freq1="1800" freq2="0" min="256" max="400"/> </tone> <tone name="RING_TONE" description="North America ring"> <element freq1="440" freq2="480" min="1200" max="0"/> </tone> <tone name="REORDER_TONE"> <element freq1="480" freq2="620" min="224" max="316"/> <element freq1="0" freq2="0" min="168" max="352"/> <element freq1="480" freq2="620" min="224" max="316"/> </tone> <tone name="BUSY_TONE"> <element freq1="480" freq2="620" min="464" max="536"/> <element freq1="0" freq2="0" min="464" max="572"/> <element freq1="480" freq2="620" min="464" max="536"/> </tone> </descriptor> Thank you

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Skype
by Rick Jarvis 04 Aug '16

04 Aug '16

Skype are telling me that it is no longer possible to receive calls to Skype names / addresses (as appose to Skype numbers) via SIP with Skype Connect. Does anyone know if this is the case? Their support is awful, but this would seem a bit of a backward step, and I don’t know whether to trust what I’m being told. The alternative is Skypopen, of course. But I notice that it has a maximum of 6 concurrent calls. Is this for resource reasons? I’m thinking of building a server to handle all Skype to SIP calls for our customers, so would be interested to know if it’s possible to bypass the limit with a beefy spec…? Thanks R

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Jitsi + Freeswitch (and general questions about SFU)
by Colin Morelli 04 Aug '16

04 Aug '16

Hey there, Has anyone had success pairing up Freeswitch + Jitsi? Specifically I'm interested in the SFU capabilities of Jitsi to replace just the video conferencing portions of Freeswitch. I'm doing a lot of other things in FS that will continue to be important on their own (DTMF, voicemail, SIP + PSTN integration, etc), but the SFU support would be an awesome addition. I saw that there's a SIP <-> Jitsi bridge which looks like it might be promising - but I feel like the complexity would fairly quickly become overwhelming. Alternatively, I wonder how much work it would be to add video-mode="sfu" to mod_conference in FS. It's something that (in the future, not quite yet), I might be interested in taking a stab at. Has anyone come up with a list of things that would need to happen for SFU support? I might be a bit naive here, but it seems FS has most of the building blocks in place already, and mostly it would come down a) having a video stream selection process, and b) generating appropriate SDPs when participants are added/removed. I'm sure I must be missing something, though. Thoughts? Best, Colin

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g729 patent expiration
by Max Clark 04 Aug '16

04 Aug '16

Hi all, So I fell down the Google rabbit hole this weekend - the subject of my curiosity: US Patent No. 5,920,8531 This patent has a priority/filing date of August 23, 1996, which would imply that the g729 is in the open next month. Is this accurate? Am I missing another patent that keeps this behind the commercial pay wall? I'm asking b/c I'd obviously rather not license channels and find out it was unnecessary a month later. Max

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Faxing transcoding issues.
by Andrew Cassidy 04 Aug '16

04 Aug '16

Good morning all, I've been playing with faxing and t38 over the last few days. I use a wholesale provider who detects and reinvites for t38, and have implemented the same in my systems. Passing t38 through unmodified is not an issue. The issue I do have, is in the event that the sender refuses the reinvite, I'd like freeswitch to transcode. In theory, this is set up correctly, but see this error message repeated over and over and ultimately the fax fails: [DEBUG] mod_spandsp_fax.c:1756 sofia/external/xxx(a)provider.com Error decoding UDPTL (172 bytes) Every other scenario I have worked through is working fine. For reference, the fax-specific dialplan entries: <action application="export" data="fax_enable_t38=true"/> <action application="export" data="fax_enable_t38_request=true"/> <action application="export" data="t38_passthru=true"/> <action application="export" data="nolocal:sip_execute_on_image=t38_gateway self" /> <action application="set" data="execute_on_answer=t38_gateway peer" /> Thanks in advance. -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director 03303 880 960 andrew(a)cassidyweb.co.uk <andrew(a)cassidywebservices.co.uk> www.cassidyweb.co.uk <http://www.cassidywebservices.co.uk>

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origination_audio_mode in Freeswitch Dialplan
by Saurabh Verma 04 Aug '16

04 Aug '16

Hi I want to implement one way audio stream through Freeswitch(Default Media Mode). SoftClient1 -----------> Freeswitch --------------->SofClient2 (will speak only) (will listen only) For this i have set *origination_audio_mode=sendonly* in the dialplan at the time of bridge.After setting this parameter in freeswitch forward the invite to b-leg with sendonly media attribute and hence SoftClient2 replies with recvonly in 200OK.The packets flow like below: SoftClient1 --------------(sendrecv)----------------> Freeswitch -------------(sendonly)--------------> SoftClien2 <-------------(sendrecv)---------------- (origination_audio_mode=sendonly) <------------(recvonly)-------------- *Now the problem is I am not not able to hear any sound at SoftClient2.* When i checked the wireshark capture at Freeswitch i found that Freeswitch is sending RTP packets on Private IP of SoftClient2 and not the Public IP. If i remove this parameter,Freeswitch starts sending RTP to the public IP and i am able to hear sound both way. Can i get some help on this? *Why the Freeswitch server is sending the RTP packets on Private IP with origination_audio_mode=sendonly?* -- *With Warm Regards:* *Saurabh Kumar Verma*

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conference events
by Mirko Brankovic 04 Aug '16

04 Aug '16

HI, What should be the proper way to collect events that are triggered in conference? I see mod_verto ALERTS like: 2016-08-03 11:53:24.248436 [ALERT] mod_verto.c:5207 EVENT BROADCAST conference-liveArray.test(a)vw232.voiceworks.nl { "data": { "action": "modify", "name": "test", .... .... but when I connect to wss on 8082 port of mod_verto we don't get those messages. Should I only connect to event_socket (8021) but then how to subscribe to them? Thanks -- Regards, Mirko

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Debian repo update schedule
by Ciprian Dosoftei 04 Aug '16

04 Aug '16

Hi all! Just wondering what it the typical update schedule for the Debian repo (with the obvious exception of major version releases). Going forward, I would prefer to stick to the official repo rather than manually compiling it, however there is a certain bug fix I am targeting ( *FS-9303*). Thanks!

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