Hi Andy
Yes we are talking about a "mic" - the "mic" in the handset of a PSTN telephone; the "mic" in the bottom part of a mobile phone, the "mic" in the handset of a SIP DeskPhone, the "mic" when using a PC with microphone attached to use a SIP Softphone.
Please keep in mind that when recording the Voicemail, the PBX receives this data as a waveform, or an audio signal (obviously after decoding it according to the codec with which it was received). The PBX stores this waveform WITHOUT ANY VOLUME LEVEL ADJUSTMENT AT ALL. The only processing is to convert the data format from whatever codec it is received in to WAV format.
This behaviour is by design - in any case it wouldn't be predictable. Please understand that when 2 internal extensions talk to each other (for example) the PBX doesn't even receive the media at all, since the devices send the audio to each other directly. When a gateway device delivers a call to an extension, again the gateway and the phone exchange media directly without involving the PBX at all.
I agree that its worth looking at - but the PBX Server is the wrong place to be looking in this case. And unfortunately there are many points where investigation is necessary. And most (if not all) of these points are not in our control. So it would likely be an academic exercise in any case.
If the playback volume you experience is acceptable, it seems likely that one of the entities between the PBX and your ear (both non-inclusive) is boosting the volume itself.
So if you are retrieving this using a SIP phone such as a Snom 320, it would probably be the Snom itself boosting the volume.
If you are retrieving this using a PSTN analog phone (by dialing in from the outside to some gateway that forwards to 999), it could be either the phone itself, or the switching equipment at the telecomms provider (telephone company), or the gateway device connecting the PSTN network to the PBX Server.
If you are retrieving this using a PSTN analog phone (by dialing in from the outside to some voip provider line that forwards to 999), it could be either the phone itself, or the switching equipment at the telecomms provider (telephone company), or the gateway equipment that interfaces the PSTN call to the VoIP Providers VoIP media proxy, or the VoIP Provider's media proxy itself.
One workaround could be to include functionality within the 3CX IVR to automatically adjust volume levels before saving the WAV files. But its definitely not on the current roadmap. And I haven't seen many requests about this issue, so we understandably concentrate on issues where more people benefit, quicker.
Hope this clears things up a bit.
Regards
Kevin