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Hi Max! Unfortunatelly I couldn't hear anything. Is also seems that the player app's volume slider doesn't get the in the modified player.js function. The slider is set in the default position at about 50%. May be this is an indicatior for you for anything. And even if I replace I cannot...

Hi Max! Also that didn't help. After adjusting the player.js and using "rtmp_stream80" as the stream instead of "stream80" the wav is again hearable in the browser but again with unacceptable slow playback with gaps. Also it seems that the (bad quality) playback does not mix the stream from the...

Hi Max! OFF -> ON is working, thank you! And the other issue: It was my mistake, very sorry for that! When I tried it yesterday for the very first time, it worked like a charme. I could hear my own voice (phone -> web app) in very good quality and with a smallest latency I ever realized! Only...

Deleting the speex16 codec didn't help. Also increasing the RAM for Java not. (followed by WCS restart) I will try to check the tweaking hints later today. May be I find something there. I also recognized that the cpu usage for Java is rather high with about 25% only for my test. Okay, 25%...

DAMN, IT'S WORKING!!! :):) I only have some voice quality issues now. My voice, and also the played wav fles, are about 30% too slow. One other point: When I have played a wav file with the sound_on command I am not able to play any further wav files. Even if I wait until the other payback...

Ahaaa! :) Okay, all looked nice now excepted that no sound could be heard. From the logs: published=false and bitrate/quality = 0 tell me: There was really nothing given out. But why not? I had connected the "player" app to the matching stream ("stream78"). Additionally to the default values in...

Hi Max! I updated the WCS but now every POST request via REST api returns a 308 error: Any ideas? And another question to the hint to add the above listed parameters to the file WCS_HOME/conf/media_transponder.sdp : That file wasn't existing per default, is that correct? I created that file and...

Wow, that was fast! :) I'll check it out later today. Thank you in advance! Marc

Okay. Good to know that this time it wasn't basing on my weak skill. ;)

Hi Max! If disabling the generate_av_for_ua=all setting and restarting WCS there is no more any error message. But I cannot hear the published stream. Best regards Marc

Hello! First of all that example only works with a Flash supporting browser. Or am I wrong? Finally it must work on a browser without Flash. So a test with another browser supporting Flash would not make sense. Then you wrote to find out the mediaSessionId I use /push/find_all when I do...

Thank you, Max, but to be honest I would appreciate if you could give me a working REST-API code example. I tried, I think, 100 hundred of combinations in the way you pointed to but nothing of it worked. I still have problems to inject a wav file into a stream to the app user. Just let's ignore...

Hi Max! Can you give me a final hint so that I can make a decision which of the two ways of implementation I will follow? Best Marc

Or do you think it is possible to initialize two streams simultanously to the user? One stream for receiving the sip audio from the callee and a second where I can send the wav audio stream back to the user? So I would play/inject the wav file from the server two times in the same moment: One...

Hi Max! Unfortunatelly that is not a makable way for me. There are too many audio files available (1000 first names combined with dozens of city names and so on). So I will switch back to the original plan. Best regards Marc

Hi Max! May be any news to this issue? Best Marc

Max, that was a big step forward! :) I can hear now what I am talking into the called mobile in the client player! Also I can (like before) hear the injected wav file in the mobile. But I cannot hear it in the client player. I tried it among others with...

Please keep in mind that I sent you two different sip account data. The first ones are for my Asterisk server, the second ones for my VOIP provider access. You should use the second ones. Obviously I will use them later in the final project directly and not like in the past via the Asterisk server.

Yes, that should work. Feel free to do so. Thank you!

It is already set to true. May be you can have a look on my server directly if you have the time? I had sent you the login data recently.