7960 is registed, but not receiving calls (NAT=yes)

Hi,

I have been setting up a new trixbox server in a remote location (behind NAT). And all trunks are working fine.

I have struggled a bit with setting up a 7960 phone, and thanks to this forum, then it is now regisering, and I can make calls to my mobile, and listen to my voicemail (*97).

When I call from my mobile to my SIP phone number, then it goes direct to voice mail. And when I switch off voice mail for that Ext, then I get a busy tone.
---------------------
Server ext IP: 83.88.66.69
Port forwarding: 5060-TCP 5060-5082-UDP 10000-20000-UDP 69-UDP 8080-TCP
Home ext IP: 212.242.195.202
----------------------
# Cisco SIP Configuration
phone_label: "Ext 101"
nat_enable: "1"
nat_address: "83.88.66.69"
line1_name: "101"
line1_shortname: "101"
line1_displayname: "101"
line1_password: "101"
------------------------
# Image Version
image_version: "P0S3-08-9-00"

# Proxy Server
proxy1_address: "83.88.66.69"

# Proxy Server Port (default - 5060)
proxy1_port:"5060"

# Emergency Proxy info
proxy_emergency: "83.88.66.69"
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: "83.88.66.69"
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: "5060"

# NAT/Firewall Traversal
nat_enable: "1"
nat_address: "83.88.66.69"
voip_control_port: "5061"
start_media_port: "16384"
end_media_port: "32766"
nat_received_processing: "0"

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: "1"

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: "3600"

# Codec for media stream (g711ulaw (default), g711alaw, g729)
preferred_codec: "none"

# TOS bits in media stream [0-5] (Default - 5)
tos_media: "5"

# Enable VAD (0-disable (default), 1-enable)
enable_vad: "0"

# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default)

# Allow Transfer to be completed while target phone is still ringing
semi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (default)

# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged

# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: "1"

# Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt )
dtmf_outofband: "avt"

# DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: "3"

# SIP Timers
timer_t1: "500" ; Default 500 msec
timer_t2: "4000" ; Default 4 sec
sip_retx: "10" ; Default 11
sip_invite_retx: "6" ; Default 7
timer_invite_expires: "180" ; Default 180 sec

# Setting for Message speeddial to UOne box
messages_uri: "*97"

# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"

# Time Server
sntp_mode: "unicast"
sntp_server: "83.88.66.69"
time_zone: "EST"
dst_offset: "1"
dst_start_month: "Mar"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "2"
dst_start_time: "02"
dst_stop_month: "Nov"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "1"
dst_stop_time: "2"
dst_auto_adjust: "1"

# Do Not Disturb Control (0-off, 1-on, 2-off with no user control, 3-on with no user control)
dnd_control: "0" ; Default 0 (Do Not Disturb feature is off)

# Caller ID Blocking (0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
callerid_blocking: "0" ; Default 0 (Disable sending all calls as anonymous)

# Anonymous Call Blocking (0-disbaled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: "0" ; Default 0 (Disable blocking of anonymous calls)

# Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control)
call_waiting: "1" ; Default 1 (Call Waiting enabled)

# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: "101" ; Default 100

# XML file that specifies the dialplan desired
dial_template: "dialplan"

# Network Media Type (auto, full100, full10, half100, half10)
network_media_type: "auto"

#Autocompletion During Dial (0-off, 1-on [default])
autocomplete: "1"

#Time Format (0-12hr, 1-24hr [default])
time_format_24hr: "1"

# URL for external Phone Services
services_url: "http://83.88.66.69:8080/xmlservices/index.php"

# URL for external Directory location
directory_url: "http://83.88.66.69:8080/xmlservices/PhoneDirectory.php"

# URL for branding logo
logo_url: "http://83.88.66.69:8080/cisco/bmp/trixbox.bmp"

# Remote Party ID
remote_party_id: 1 ; 0-Disabled (default), 1-Enabled
----------------------------------------------------------------------
#sip_nat.conf
externip = 83.88.66.69
localnet = 10.0.0.0/255.255.255.0
----------------------------------------------------------------------
[101]
type=friend
secret=101
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=101@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/101
context=from-internal
canreinvite=no
callgroup=
callerid=device <101>
accountcode=
call-limit=50
-----------------------------------------------------------------

I hope you may have some idear how I can fix this issue.