July 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Verto/videoconference using React.js?
by Chad Phillips 15 Jul '16

15 Jul '16

Curious if anyone has implemented a Verto/vidoconference front end using a React.js stack? Looking for examples/starter templates.

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Extension not visible
by uzair iltifat 15 Jul '16

15 Jul '16

Hi, I want to register two SIP accounts so that they can call each other, for this purpose I have added two extension 1100.xml and 1200 by coping 1000.xml and then edit my dialplan so that it can find my added extension like this <condition field="destination_number" expression="^(10[01][0-9]|1100|1200)$"> and finally i reloaded xml's through fs_cli, I am using Zoiper(soft-phone) for testing, Now the problem is when i register an account in Zoiper with 1100 extension it in is not registering it. After digging out this problem i found out that my user is not registered except 1000.xml, by executing the list_users command and sofia contact 1100 (there is an attached Image), how can i register one. Kindly guide me is there any thing i am missing OR i have to do some thing else for "registering two SIP accounts so that they can call each other". thanks in advance, p://uzair.ahmed.netcv.com/http://uzair.ahmed.netcv.com <http://uzair.ahmed.netcv.com/http://uzair.ahmed.netcv.com/> Uzair Ahmed [image: http://]about.me/uzairIltifat <http://about.me/uzairIltifat?promo=email_sig> http://uzair.ahmed.netcv.com

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mod_amqp on OSX El Captain
by Andrea Batazzi 15 Jul '16

15 Jul '16

Hello, trying to compile mod_amqp on OSX El Captain but am getting this error: You must install librabbitmq1 and librabbitmq-dev to build this module. Stop. No brew formulas available for librabbitmq. Anyone has managed to and can point me to the right direction ? Thank you, Andrea Batazzi

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Error receiving second call Verto
by Azlain Saavedra 14 Jul '16

14 Jul '16

Hi im user verto.js library and i can execute first call verto-client to verto-client but when i execute second call to same verto-client so failed bucause this is redirect to destroy event first receive call into to "ringing" event --> answering --> active --> hangout --> destroy second receive call into to "destroy" event this is my code (function() { var vertoHandle, vertoCallbacks, destinationNumber, currentCall, ext, passExt; function getParameterByName(name) { name = name.replace(/[\[]/, "\\[").replace(/[\]]/, "\\]"); var regex = new RegExp("[\\?&]" + name + "=([^&#]*)"), results = regex.exec(location.search); return results === null ? "" : decodeURIComponent(results[1].replace(/\+/g, " ")); } ext = getParameterByName("ext"); passExt = getParameterByName("pass"); $.verto.init({}, bootstrap); function bootstrap(status) { console.log("extension: " + ext) vertoHandle = new jQuery.verto({ login: ext + '@conference2.cibersys.com', passwd: passExt, socketUrl: 'wss://conference2.cibersys.com:9061', ringFile: 'sounds/bell_ring2.wav', tag: 'video-container', // specifying video tag in our html deviceParams: { useCamera: true, // asking for camera permissions and devices useMic: true, useSpeak: true }, iceServers: true }, vertoCallbacks); document.getElementById("make-call").addEventListener("click", makeCall); document.getElementById("answer-call").addEventListener("click", answerCall); document.getElementById("hang-up-call").addEventListener("click", hangupCall); document.getElementById("mute-call").addEventListener("click", muteCall); document.getElementById("unmute-call").addEventListener("click", unmuteCall); document.getElementById("mute-unmute-call").addEventListener("click", muteUnmuteCall); document.getElementById("hold-call").addEventListener("click", holdCall); document.getElementById("unhold-call").addEventListener("click", unholdCall); document.getElementById("transfer-call").addEventListener("click", transferCall); document.getElementById("muteVideo-call").addEventListener("click", muteVideoCall); document.getElementById("unmuteVideo-call").addEventListener("click", unmuteVideoCall); }; function makeCall() { destinationNumber = prompt("Insert destination number"); if(destinationNumber) { currentCall = vertoHandle.newCall({ destination_number: destinationNumber, caller_id_name: "Test Guy", caller_id_number: destinationNumber, outgoingBandwidth: "default", incomingBandwidth: "default", useVideo: true, // telling verto to make a call with video support mirrorInput: true, // telling verto to mirror user's webcam useStereo: true, useMic: true, useSpeak: true, dedEnc: false, userVariables: { avatar: "", email: "test(a)test.com" } }); } }; function answerCall() { currentCall.answer({ useStereo: true, useCamera: true, useVideo: true, useMic: true, callee_id_name: "answer ponit", callee_id_number: ext }); }; function hangupCall() { currentCall.hangup(); }; function muteCall() { currentCall.setMute("off"); }; function unmuteCall() { currentCall.setMute("on"); }; function muteVideoCall() { currentCall.setVideoMute("off"); }; function unmuteVideoCall() { currentCall.setVideoMute("on"); }; function muteUnmuteCall() { currentCall.mute("toggle"); }; function holdCall() { currentCall.hold(); }; function unholdCall() { currentCall.unhold(); }; function transferCall() { destinationNumber = prompt("Insert transfer destination number"); if(destinationNumber) { currentCall.transfer(destinationNumber); } }; vertoCallbacks = { onWSLogin: onWSLogin, onWSClose: onWSClose, onDialogState: onDialogState }; function onWSLogin(verto, success) { console.log('onWSLogin', success, verto); }; function onWSClose(verto, success) { console.log('onWSClose', success); }; function onDialogState(dialog) { console.debug('onDialogState', dialog); if(!currentCall) { currentCall = dialog; } if (currentCall.state.name == 'ringing') { console.warn('onDialogState ringing', currentCall); } if (currentCall.state.name == 'new') { console.warn('Someone is calling you, new!'); } if (currentCall.state.name == 'requesting') { console.warn('Someone is calling you, requesting!'); } if (currentCall.state.name == 'recovering') { console.warn('Someone is calling you, recovering!'); } if (currentCall.state.name == 'active') { //$scope.offLocalVideo(); console.warn('Someone is calling you, active!'); } if (currentCall.state.name == 'held') { console.warn('Someone is calling you, held!'); } if (currentCall.state.name == 'purge') { console.warn('Someone is calling you, purge!'); } if (currentCall.state.name == 'early') { console.warn('Someone is calling you, early!'); } if (currentCall.state.name == 'answering') { console.warn('Someone is calling you, answering!'); } if (currentCall.state.name == 'hangup') { //$scope.resetCall(); console.warn('Someone is calling you, hangup!'); //$scope.resetCall(); //console.log("sreamLocalVideo.getVideoTracks()", sreamLocalVideo.getVideoTracks()[0].stop()); //sreamLocalVideo.src=null; } if (currentCall.state.name == 'destroy') { console.warn('Someone is calling you, destroy!'); } } })(); -- T.S.U Azlain Saavedra Movil: +58 04142528099 Skype: azlain.saavedra

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WebRTC and mod_verto integration
by acedj 14 Jul '16

14 Jul '16

Hi guys, sorry for my english but i have a terrible problem with my installation of freeswitch with fusionPBX. I have a main server with fusion pbx and here i'd like to configure a WebRTC server where clients can directly call the extensions that are already set. The result should be like this: https://webrtc.freeswitch.org/verto/index.html#page-main I have tried to follow the official guide but i can't find the main installation dir in /usr/local/freeswitch... I have also tried to create a new installation of freeswitch but i don't understand how to do this operation. Thank's in advice. Davide -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/WebRTC-and-mod-verto-integrat… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Problem with 'sips' prefix
by Miroslav Levanic 14 Jul '16

14 Jul '16

Hi, I'm using FS 1.4.18. I'm using TLS and SRTP. I'm trying to initiate outbound call with destination set to sips:something using "originate {return_ring_ready=true}<destination> &park xml default" command. Checking outgoing trace, I can see that destination in INVITE SIP Request URI becomes sip:something. Is there a way to force or enable 'sips' instead of 'sip'? Regards, Miro

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change or alter member_id starting value at freeswitch startup
by Jarad Morgan 14 Jul '16

14 Jul '16

Hello all, We run a farm of freeswitch boxes as conferencing servers and manage various call controls through external applications via the event socket. Every once in a while we run into a rare condition where we end up with two calls (on different servers) having the same member_id which causes some confusion for our external apps. Does anyone know if its possible to alter the member_id at freeswitch startup? Basically trying to avoid calls having the same member_id across all our freeswitch servers. Thanks!!

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Calling WebRTC + IP Phones Through Proxy
by Colin Morelli 14 Jul '16

14 Jul '16

Hey all, I'm looking to be able to dial out to both WebRTC and IP phones through a cluster of proxy servers. Normally, FS is smart enough (I believe) to send separate INVITEs based on the type of clients registered. However, in this case FS has no idea who is registered since that happens at another layer. I'm able to make WebRTC calls by setting media_webrtc=true. But this breaks the ability for some IP phones to understand the SDP. Now, I should pretty easily be able to fork two separate dials to the proxy, one with a WebRTC SDP and one with a standard RTP/SRTP SDP. Then, in the proxy, filter the recipients of each call appropriately. Is this a reasonable strategy? Any other suggestions? Best, Colin

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Load test mod_conference with video
by Mirko Brankovic 14 Jul '16

14 Jul '16

Hi, Can anyone suggest how to "stress"/load test mod_conference with video calls. I have tried with sipp custom scenario and after sending ACK to Fs I used play_pcap_video to stream pcap file. But it failed to generate network RX load on server even thou sipp users joind conference and TX load was increasing. Is there any other tool I could use ? Any suggestion from anyone that tried it can be helpful. -- Regards, Mirko

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SIP Profile go Down
by Ahmed habiba 14 Jul '16

14 Jul '16

Hi, I face some issue, I have one FS instance that has been running for long time, today I was not able to connect and when I run “sofia status “ I get below Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) Which means that my lan IPV4 profile was not running I’m not sure about the reason but when I reload mod_sofia and run “sofia status” I get the below: Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) internal_lan profile sip:mod_sofia@192.168.1.7:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) I’m not sure what could let a profile go down? your kind help will be appreciated. Thanks, Ahmed Habiba,

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