Loss of outbound audio during a call (v1.2.3)

I?ve run into a problem where during a call (VOIP), all of a sudden the person I called can?t hear me, but I can hear them no problem. It seems to do it randomly. Any ideas? Where should I start looking ? I sure could use the help . . .

Trixbox v1.2.3 (Asterisk 1.2.12.1)

[quote]Sip.conf
[general]
vmexten=*97
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
disallow=all
allow=ulaw
allow=alaw
;context = from-sip-external ; Send unknown SIP callers to this context
context = from-trunk ; allows anonymous sip callers
dtmfmode=rfc2833
callerid = Unknown
externip=69.XX.XX.XXX
localnet=192.168.0.0/255.255.255.0

Sip_additional.conf
;incoming trunk settings
[69.XX.XX.XXX]
type=user
host=69.XX.XX.XXX
fromdomain=69.XX.XX.XXX
dtmfmode=rfc2833
context=from-pstn

;outgoing trunk settings
[out trunk name]
username=69.XX.XX.XX
type=peer
host=69.XX.XXX.XXX
fromdomain=XX.XX.XX.XXX
dtmfmode=inband ;rfc2833 and info wasnt working with my provider so set to inband temporarily
disallow=all
allow=ulaw

;Extension used for test calls
[2000]
username=2000
type=friend
secret=2000
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
pickupgroup=1
nat=never
mailbox=2000@device
host=dynamic
dtmfmode=rfc2833
disallow=all
context=from-internal
canreinvite=no
callgroup=1
callerid=Ext 2000<2000>
allow=ulaw[/quote]