Can someone kindly point me to the area within the FreePBX config where I can increase the volume of recorded calls, specifically the audio of people we’re calling and people who call into my PBX.
I can hear my users just fine, and we can hear callers just fine while we’re on the phone with them, but the recorded audio is so low on their end that it’s like we’re hearing only half the call.
Make a test call to 1 415 421 0020. Open the recording in an audio editor and report the level, relative to full scale sine wave. Also, what kind of trunks do you have?
DC offset 0.000078
Min level -0.349121
Max level 0.349335
Pk lev dB -9.14
RMS lev dB -12.88
RMS Pk dB -12.28
RMS Tr dB -77.25
Crest factor 1.54
Flat factor 0.00
Pk count 3
Bit-depth 15/16
Num samples 80.8k
Length s 10.100
Scale max 1.000000
Window s 0.050
Not personally familiar with Vega’s but if they use the DAHDI channel driver then try adjusting the rxgain/txgain settlings in /etc/asterisk/chan_dahdi.conf (or whatever is ‘included’) if you do that you need to stop asterisk, restart dahdi, start asterisk to see what happened.
I’m hesitant to do that as the calls themselves are fine. It’s only the recordings that are one-sided.
Live calls are perfectly fine and don’t need to be adjusted, but when I play back the recordings, I can only hear my staff and I struggle to hear the other party.
Well, its not an irreversible change . The human ear has a quite remarkable automatic gain control built in, not so much chan_dahdi and the consistent discrepancy between rx and tx in the recordings strongly suggest an imbalance of several db’s which the human agc can easily normalize.
If you have it avaiable on the vega, ‘dahdi_monitor’ is very useful for watching/recording/stereo-izing your calls
dahdi_monitor -vv <channel_active>
gives you an approximation of an old fashioned stereo pair of vu-meters, (there is a man page also)