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Hello We have seen DTMF error WCS: 08:22:38,400 INFO RestApiRouter - HTTP-pool-3-thread-193 Use controller class com.flashphoner.rest.server.rest_v2.RestCallController with path /rest-api/call/send_dtmf 08:22:38,400 INFO RestCallController - HTTP-pool-3-thread-193 handleRequest...

We checked again with the following test: 1. Call from softphone1 to the extension number provided (7002). 2. Stop the call on WCS side by /rest-api/call/terminate. The hold music plays in softphone 3. Call from softphone2 with credentials defined in add_wcs_to_conference.sh script to the...

Good day. We fixed the issue in WCS build 5.2.1047. Web SDK should also be updated to 2.0.199.

Good day. We fixed the issue in WCS build 5.2.1047. Web SDK should also be updated to 2.0.199.

Good day. About the ticket WCS-3271. We fixed the issue in build 5.2.1045, RTSP stream from dump you've provided is playing successfully.

We checked the flow. Seems like Asterisk sends silence when WCS established call to the conference. If we establish SIP call via our test Asterisk/OpenSIPS PBX, all seems working. So this is may be Asterisk configuration issue. One issue is in SIP call parameters: sipDomain parameter ends with...

Good day. Yes, you can: 1. Publish WebRTC streams From a web camera in a browser via WebRTC 2. Add those streams to mixer automatically Automatic mixer creation configuration or using REST API REST queries 3. Republish mixer output stream as RTMP using REST API REST queries

remoteVideo item cannot be hidden because video item to play a stream is added as child to it If you know on client side what to play (audio-only or video+audio), set iframe size dynamically as needed: - for video+audio, set <iframe ... width=480 height=270> - for audio only, set <iframe ...

Please provide SSH access to the instance using this form, we will check.

Please try player.js from archive attached. The player width and height can be set by iframe attributes, for example <iframe ... width=480 height=50>

We made a test: published video+audio stream to the server from which Embed Player should play, and played from the page you've provided. Seems like Embed Player is working: audio only is playing, and only start/stop and volume controls are shown: You can change iframe size if this seems too...

Seems you mistyped in folder name: FlashphonerWebCallServer with big S.

We responded you in this topic. It is not necessary to update the whole instance. Please note that you must configure WCS instance from the scratch when updating to a new image. You can update WebSDK only: 1. Download and unpack WebSDK installation package wget...

Good day. We will schedule AMI update. You can update an instance to the latest build using commands sudo systemctl stop webcallserver sudo /usr/local/FlashphonerWebCallServer/bin/webcallserver update sudo /usr/local/FlashphonerWebCallServer/bin/webcallserver set-permissions sudo systemctl...

Please create your own topic next time. We tested SIP as RTMP call on your server via our test PBX (OpenSIPS, Asterisk) from WCS to Bria softphone, with republishing RTMP stream to our demo server. Audio is playing normally from your server in Player example and from demo server in Media Devices...

In startup.log file we see that WCS is starting in WCS_NON_ROOT=true mode: [2021-09-27 01:58:21] INFO startWithSudo - Starting FlashphonerWebCallServer as user flashphoner Please make sure that you've set WCS_NON_ROOT=false in /usr/local/FlashphonerWebCallServer/bin/setenv.sh file (this is not...

Please update Web SDK to the latest build 2.0.198. If this does not help, please provide SSH access to the server and to the publishing/playing pages using this form.

Для тестирования Вы можете использовать наш демо сервер (логин и пароль demo). Также Вы можете развернуть сервер в AWS EC2 из готового образа. Если у Вас возникают трудности с развертыванием сервера, Вы можете запросить услугу первоначальной настройки. На величину задержки при трансляции WebRTC...

Please provide a minimal Vue JS project ready to build and test in which the issue is reproducing. Standalone vue file is not enough. Please use this form to send.

Good day. You can adjust microphone gain as shown in this example: $("#micGainControl").slider({ range: "min", min: 0, max: 100, value: currentGainValue, step: 10, animate: true, slide: function (event, ui) {...