Is there a way to increase the volume of one side of a recording? In all of my recordings, the local person is very easy to hear (quite loud at times, actually) while the remote person is very difficult to hear. The actual calls themselves are fine and there are typically no issues hearing the other person while on the phone (though I suppose this could be a volume setting on the phone) – it’s only the recordings that are seemingly one-sided.
I have a VEGA60 and a VEGA100 and I believe the issue lies either with them or a related setting in PBXAct as internal calls have a normalized volume for both parties.
If anyone has any ideas on where to look, I’d appreciate it. Thanks in advance!
The MixMonitor command has tx/rx gain settings, that only affect the volume of the recording, not the call itself.
asterisk*CLI> core show application MixMonitor
v(x): Adjust the *heard* volume by a factor of <x> (range '-4' to '4')
V(x): Adjust the *spoken* volume by a factor of <x> (range '-4' to '4')
W(x): Adjust both, *heard and spoken* volumes by a factor of <x> (range '-4' to '4')
I think you are trying to address symptoms, not causes. The only time you might get low levels on remote parties is if they are on very long analogue lines, and that would only be a very few of your remote parties.
Is it possible you are only recording one side of the call and the remote party recording is the echo from the local phone?
Is it possibly that you have G.729 pass through inbound, but a codec that can be handled outbound?
Most business calls are becoming digital end to end, and there should be no degradation in audio level.
On every single one of my calls, the remote party is extremely difficult to hear when I listen to the recording. I can hear the local person just fine. Local to local calls are also just fine. It’s only when calls go through my two VEGAs (inbound and outbound) that I have issues hearing the other party, and only on the recording. The actual calls themselves are just fine.
The Sangoma Vega 100 is an ISDN gateway, and ISDN should have audio already normalised by the carrier. The Vega 60 is an analogue, or BRI, gateway. For BRI, levels should also be normalised in the network. For analogue, they should be corrected in the gateway. before they reach VoIP. I’m assuming your Vega 60 is configured with FXO cards.
If local calls are recorded at correct levels and reach the phones at correct levels, I’d have to assume the phones are compensating for the low levels, but that will increase noise and reduce dynamic range.
I’m assuming the local phones are all VoIP.
It looks as though there is only a -18 dB + 6 dB gain adjustment range in the Vega, and it sounds like you are more than 6 dB down. Is there any possibility that someone turned the gain right down, possibly because of getting echo problems?
I do note they do G.729, or as typo-ed, in one place, G.279. However, I have a feeling that PBXact comes with the codec and licences.
As an aside, the manual only gives uk.pool.ntp.org, as an example, or maybe the default, for the NTP server. This server domain name should only be used from the UK, and should be replaced by a more local one, elsewhere.