I have two questions related to poor sound quality, and have not been able to find a posting that deals with my particular situation. Currently I am strictly using a VoIP configuration with softphones only (until I get everything working to my satisfaction). I am also connected to my ITSP by an IAX ‘trunk’. Also, I am almost exclusively using open source software at this point, in order to understand any associated limitations.
I am using the X-Lite softphone and when I call people they are hearing a good deal of noice on the line, as well as the occaisional ‘clipping’ of my voice. I used the Audio Turning Wizard as part of the X-Lite softphone in order to ‘train’ the system and remove the background noice, but that does not appear to have had any affect. What am I missing?
I am trying to upload WAV files to my PBX, via the AMP (for convenience). They are my own recordings that were made with Audacity, creating an MP3 file. I then converted them to WAV using LAME. This was all done on my Windows PC - from which I am accessing the AMP. Asterisk doesn’t seem to recognize the WAV file since I am not able to here anything when I try to play the recording back from within Asterisk. I can’t believe that the WAV files generated from within Windows, would not be compatible with a Linux system. Is that the case, or am I missing something else?