Integrating AGC+DENOISE+JITTER with FreePBX (crackly tx audio)

[ghost]

No documentation is provided to integrate AGC+DENOISE and JITTERBUFFER in Asterisk >12 (ex. 18).

I'm trying to figure out how to best integrate them as my TX audio is relatively poor quality.

This is what my extensions_custom.conf looks like, roughly:

[dongle-incoming]
exten => sms,1,Verbose(Incoming SMS from ${CALLERID(num)} ${BASE64_DECODE(${SMS_BASE64})})
...
exten => sms,n,Hangup()
exten => _.,1,Set(CALLERID(name)=${CALLERID(num)})
exten => _.,n,Goto(from-trunk,${EXTEN},1)

[macro-dial-one-custom]
; denoise
exten => s,n,Set(DENOISE(rx)=on)
exten => s,n,Set(DENOISE(tx)=on)
; agc
exten => s,n,Set(AGC(rx)=4000)
exten => s,n,Set(AGC(tx)=4000)

The dongle.conf context is set to dongle-incoming.

How are we supposed to integrate the AGC and co in a custom extension with FreePBX? from-sip is not used anywhere.

[Sarmingsteiner]

Maybe not the best solution, but for us the following manual configuration in dongle.conf works best:
rxgain=4 ; increase the incoming volume; may be negative
txgain=-5 ; increase the outgoing volume; may be negative