Issue with sound quality

hello all,
I installed an asterisk 1.6.1.11 with a Digium B410p connected to 2 BRI channels. I use SIP protocol for internal phone.
I created a small IVR and if I call the IVR from internal to internal all works fine and the sound quality is verry good, instead, if I call from PSTN netword (through Digium Board) the sound quality is very low.
I tried with different audio format (gsm, alaw, ulaw, wav, etc.) but the result is always the same.
Could you help me to solve this issue?
thanks,
Ivan

hello all,
nobody have any idea how to solve this issue?
thanks,
Ivan

You can play with parameters like:

echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0

I tried to set these value, but nothing is changed. Have you any other idea?
thanks for your answer,
Ivan

try changing
span=,,<line build out (LBO)>,,
in /etc/dahdi/

I’m using mISDN and not DAHDI to manage my B410P card. Do you think that I sould use DAHDI drivers? and what values I should insert in your suggested parameters?
thanks,
Ivan

thanks all for your help,
I replaced mISDN driver with DAHDI and now the sound quality is much better.
Ivan