[asterisk-users] audio gain in SIP channel

[lore]

hello all,
i'm trying to do what in object with an asterisk box 11.11 on centos6.5,
using functions
AGC and VOLUME, but seems that does not work at all.
There is a way to check this values during setup/call?
Maybe is it not possible realize what i'd like to do?

Could anyone can help me on this?

thanks a lot in advance

regards

Lorenzo
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