Hello all! I'm experiencing some problems with clipping and the rate of playback of wav files from Freeswitch. I've not raised this before as I thought the problems were due to hosting over an ADSL line. However I've now seen the same issues running from a dedicated host. My test setup is: Client <-> NAT ADSL <-> Freeswitch I've replicated using both the Gizmo and FWD clients from Windows. If I substitute Asterisk with Freeswitch then the audio comes over fine. The two test samples are one of myself and the fpm-sunshine.wav that is bundled with asterisk. I've tried playing around with the codecs but see the same problems regardless. Also I've performed a test with a conference hosted on the Freeswitch instance with myself and friend dialling in using Gizmo and the audio is fine. Which implied to me that the link between Gizmo to Freeswitch is ok. So I'm guessing that there is something wrong with the samples that is causing the problem? They're both in this format: $ file fpm-sunshine.wav RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz My deployment platform is Debian stable and the Freeswitch SVN revision is 7343. I've tried looking in Jira but keep getting a system error page (for search terms 'audio', 'wav' and others). Is anyone else seeing the same issues or has a solution? thanks, - Dale