Audio quality problem with a800p and 5 fxs ports

When making or receiving calls the quality is sometimes really poor on the outgoing sound.
I have narrowed the problem down to somewhere between the phone and the asterisk system.
Attached is a recording of the bad sound.

This is running on an Intel D525MW board with an Intel atom processor D525

Details:
Asterisk Now install with FreePBX 2.9.0.9
Asterisk 1.6.2.21

Dahdi status:
Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO
OpenVox A1200P/A800P Board 13 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1)

Connected phone to the fxs port:
RCA 25414RE3-A (it's a 4 line business phone).

If someone could help me diagnose this problem that would be great.

I have checked the record, according to the wave, you need to adapt the rxgain and txgain, especially rxgain.
if the default value is like this:

rxgain=0.0
txgain=0.0

pls set it like this:

rxgain=-3.0
txgain=0.0

then restart asterisk and have a try!
Let me know the result, thanks!

I doubt the fxs port can drive your 4 line business phone directly, if you connect the phone to pstn line, it can work?

Yes the 4 line business phone works on the pstn, and it also works using vonage PAP2T adapters.

The rxgain change seems to have made a difference, I will monitor it and let you know.

I was reading about tuning and wondered if there was a procedure to tune fxs ports in a similar manner to the directions here:

www.mattgwatson.ca/2008/05/howto-tune-za...ces-on-asterisk-pbx/

or with a tool like fxotune.

Thanks.

Well the rxgain=-3.0 has made a big difference. However I am still having trouble with echo.

the generated system.conf has:

fxoks=1
echocanceller=mg2,1

When the line echocanceller=mg2,1 is in the system.conf there is no echo from the caller on the other end when calling out on a voip trunk.
But this causes the asterisk system to not recognize touch tones after a call is placed. For example dialing *97 for the voicemail system works, but when you enter your password asterisk doesn't recognize any of the tones the asterisk log reports the password entered as an empty string ''.

commenting out the echocanceller=mg2,1 in the system.conf restart asterisk and the above voicemail scenario works.

Is there another way that I can reduce echo? Or can I turn off echo cancellation for interal routes?

Zaptel/DAHDI DTMF Detection Problems
DTMF detection problems can be caused by a number of different factors. The most common is running the X Windows System. Another cause of DTMF detection problems is the relaxdtmf option in Zapata.conf/chan_dahdi.conf. It may need to be turned on or off.

SIP DTMF Detection Problems
If you are having problems sending DTMF digits amd are using a SIP phone, make sure the dtmfmode they have set is the same on the phone and in Asterisk. Also make sure you are not sending both inband and out-of-band (rfc2833) tones.

Echo issues
Pls set such values like this:

echocancel=yes
echocancelwhenbridged=no
faxdetect=incoming
;echotraining=800 //disable echotraining

After that, pls let me know the result!