[asterisk-users] Monitor, MixMonitor and volume levels

[Steve Davies]

Hi,

I have started using the call recording facilities in Asterisk 1.2
recently, and having worked out some of the foibles regarding call
forwarding etc etc, I think I have a mostly working system.

I do still seem to have a problem with recording volume though. It
seems that all SIP call legs are recorded at "normal" volume, but all
my Zap (ISDN) and IAX (via Provider -> ISDN) calls are recorded at a
massively reduced volume.

-> It does not matter whether the call originates inside or outside the box
-> It does not matter which channel is Monitored (Zap, IAX or SIP)
-> The caller/callee can hear each other fine regardless of the call
source and destination.
-> I also tried both Monitor and MixMonitor with the same results.
-> The recording of the ISDN or IAX leg is so quiet that it is often
impossible to hear.
-> SIP to SIP records 100% okay
-> Recording using different codecs makes no difference
-> Voicemail recording volume is fine, regardless of call source.

I considered using MixMonitor's volume settings, but cannot always
identify which channel needs a volume boost (Local channels can
obscure the call source or destination)

I can use 'sox' to modify the levels to a usable point, but this
amplifies background noise to a ridiculous degree so is not
particularly satisfactory.

Given that the call proceeds "normally" where is all of the volume
being lost? We generally use aLaw end-to-end (which is the codec used
on UK ISDN lines) so there should be almost no modification of the
voice packets required at-all. Why does the recording differ from the
audio being heard? I looked at the source and could see no obvious
reason!

Thanks for any pointers. I am happy to try experiments on our
development system if it helps...

Regards,
Steve