Misdn : Intermitted call quality errors outgoing calls

Hi ,

I am receiving bad call quality issues when it comes to outgoing calls. We are using a digium BRI-410P with 4 lines (8 channels) . Is there anyway to tweak up the quality ?

Location : Italy ( Fastweb)
Client phones : GXP 2000
Asterisk version: 1.4.18.1

This is my sip.conf


[general]
videosupport=yes
pedantic=yes
context=default                 ; Default context for incoming calls
allowguest=no                   ; Allow or reject guest calls (default is yes)
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)

tos_sip=cs3                    ; Sets TOS for SIP packets.
tos_audio=ef                   ; Sets TOS for RTP audio packets.

maxexpiry=120                   ; Maximum allowed time of incoming registrations
                                ; and subscriptions (seconds)
disallow=all                    ; First disallow all codecs
allow=alaw                      ; Allow codecs in order of preference
allow=ulaw
allow=ilbc
allow=h261
allow=h263
allow=h263p

useragent=Asterisk PBX          ; Allows you to change the user agent string
matchexterniplocally = yes     ; Only substitute the externip or externhost setting if it matches
                                ; your localnet setting. Unless you have some sort of strange network
                                ; setup you will not need to enable this.

rtpholdtimeout=300              ; Terminate call if 300 seconds of no RTP or RTCP activity
                                ; on the audio channel
                                ; when we're on hold (must be > rtptimeout)
rtpkeepalive=20                 ; Send keepalives in the RTP stream to keep NAT open
                                ; (default is off - zero)
;--------------------------- SIP DEBUGGING ---------------------------------------------------
sipdebug = no                   ; Turn on SIP debugging by default, from
                                ; the moment the channel loads this configuration
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0     ; Also RFC1918
nat=no                  ; Global NAT settings  (Affects all peers and users)
canreinvite=no                  ; Asterisk by default tries to redirect the
                                ; RTP media stream (audio) to go directly from
                                ; the caller to the callee.  Some devices do not
                                ; support this (especially if one of them is behind a NAT).
                                ; The default setting is YES. If you have all clients
                                ; behind a NAT, or for some other reason wants Asterisk to
                                ; stay in the audio path, you may want to turn this off.

; Insert server specific include files below
#include "server_specific_sip.conf" ; Specific file for each country

And this is my misdn.conf



[general]

misdn_init=/etc/misdn-init.conf

debug=1

ntdebugflags=0
ntdebugfile=/var/log/misdn-nt.log
ntkeepcalls=no
tracefile=/var/log/asterisk/misdn.log
bridging=no
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh


[default]

context=misdn
language=en
musicclass=default
senddtmf=yes
far_alerting=no
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=no
pmp_l1_check=no
reject_cause=16
need_more_infos=no
nttimeout=no
method=standard
overlapdial=yes=5
dialplan=2
localdialplan=0
cpndialplan=2
early_bconnect=yes
incoming_early_audio=no
nodialtone=no
presentation=-1
screen=-1
echocancel=yes
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no
max_incoming=-1
max_outgoing=-1

[intern]
ports=1,2,3,4
context=misdn
msns=*

And my misdn-init.conf


card=1,0x4


te_ptp=1,2,3,4

poll=128
dsp_poll=128
dsp_options=0
dtmfthreshold=100
debug=0

Thanks !