Greetings, I'm using a 3rd party application (ndev dragonmobile) to get the transcription of some audio recorded by FreeSWITCH. Think about it as a voicemail transcription service. The problem I'm facing is that, when I record a session using uuid_record, the output file is encoded in PCM 16-bit @ 8khz. Correct me if I'm wrong, but my understanding is that if I want to capture audio from calls coming from the PSTN (analog/landlines), the best I can do is to record it in 8-bits (using G.711). I don't want to use sox (or any other tool) to resample the output file, and what I've tried so far is setting the sample_rate variable on the diaplan as recommended on the wiki: http://wiki.freeswitch.org/wiki/Variable_record_rate <action application="set" data="record_sample_rate=8000"/> This doesn't have any effect on the generated wav file, which is still encoded in 16-bits. So my question is: does this variable affects the behavior of the uuid_record command? Or, do I really need to encode the audio output in 8-bits when the origin of the call comes from the PSTN? How is FreeSWITCH encoding the audio in 16-bits if, in theory, the best rate we can get from an analog line is 8-bits? Sorry if I'm misunderstanding something, but I'm not a telephony/voip guy, more like a java developer. Thanks for your help.