[nOon]

Hi guys,
I try to connect cisco 7940 to my asterisk but i have this error:
Jan 19 13:14:10 NOTICE[7750] chan_sip.c: Registration from '<sip:299@172.16.1.3>' failed for '172.16.9.188' - ACL error (permit/deny)

And it's really because i can make call but not receive.

This my conf files:
Sipdefault.cnf ->
# Image Version
image_version: "P0S3-8-12-00"

# Proxy Server
proxy1_address: "172.16.1.3"

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

# Emergency Proxy info
proxy_emergency: "172.16.1.3"
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: "172.16.1.3"
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: "172.16.1.3"
outbound_proxy_port: "5060"

# NAT/Firewall Traversal
nat_enable: "0"
nat_address: ""
voip_control_port: "5060"
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: "g711ulaw"

# 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: "1" ; 0-Disabled, 1-Enabled (default)

# Telnet Level (enable or disable the ability to telnet into this phone
telnet_level: "0" ; 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: "11" ; Default 11
sip_invite_retx: "7" ; Default 7
timer_invite_expires: "180" ; Default 180 sec

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

#********* Release 2 new config parameters **********

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

# Time Server
sntp_mode: "unicast"
sntp_server: "193.67.79.202"
time_zone: "CET"
dst_offset: "1"
dst_start_month: "March"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "4"
dst_start_time: "02"
dst_stop_month: "Oct"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "8"
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: "100" ; 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: "0"

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

Next my SIPxxxxxx.cnf ->
image_version: "P0S3-8-12-00";

phone_label: "Test"

user_info: "none";

line1_name: "299";
line1_displayname: "299";
line1_shortname: "299";
line1_authname: "299";
line1_password: "53b9ee";

And my sip.conf:
[ciscotest]
type=friend
context=internal
secret=53b9ee
callerid="Cisco Test" <299>
laguage=fr
host=172.16.9.118
dtmfmode=rfc2833
dissallw=all
allow=ulaw
nat=no
qualify=no

If anyone have a solution that's to be great.

[nOon]

new information, when i use the debug of the phone i have:
[20:23:08:30396] LINE 6/1: --0x00071dad-- : SIP_REG_STATE_IDLE <- E_SIP_REG_TMR_EXPIRE
[20:23:08:30396] ccsip_register_send_msg: cmd=0x161c01
[20:23:08:30396] LINE 6/1: --0x00071c4d-- : SIP_REG_STATE_IDLE <- E_SIP_REG_REG_REQ
[20:23:09:30401] sipTransportSendMessage: ccb <6>: config <172.16.1.3>:<5060> - remote <0.0.0.0>:<5060>
[20:23:09:30401] sipTransportSendMessage: Opened a one-time UDP send channel to server <172.16.1.3>:<5060>, handle = 8 local port= 0
[20:23:09:30402] sipTransportSendMessage:Sent SIP message to <172.16.1.3>:<5060>, handle=<8>, length=<532>, message=
[20:23:09:30402] REGISTER sip:172.16.1.3 SIP/2.0
Via: SIP/2.0/UDP 172.16.9.188:5060;branch=z9hG4bK6d4ad7fe
From: <sip:299@172.16.1.3>;tag=000dbc3515b200097c0f5097-361f20d0
To: <sip:299@172.16.1.3>
Call-ID: 000dbc35-15b20003-1f961f4a-4c3d87bf@172.16.9.188
Max-Forwards: 70
Date: Tue, 19 Jan 2010 19:23:08 GMT
CSeq: 107 REGISTER
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:299@172.16.9.188:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-000dbc3515b2>";+u.sip!model.ccm.cisco.com="8"
Content-Length: 0
Expires: 3600

[20:23:09:30404] sipTransportSendMessage: Closed a one-time UDP send channel handle = 8
[20:23:09:30404] LINE 6/1: Registration state change: SIP_REG_STATE_IDLE ---> SIP_REG_STATE_REGISTERING
[20:23:09:30405] SIPProcessUDPMessage: recv UDP message from <172.16.1.3>:<50195>, length=<405>, message=
[20:23:09:30405] SIP/2.0 404 Not found
Via: SIP/2.0/UDP 172.16.9.188:5060;branch=z9hG4bK6d4ad7fe;received=172.16.9.188
From: <sip:299@172.16.1.3>;tag=000dbc3515b200097c0f5097-361f20d0
To: <sip:299@172.16.1.3>;tag=as1b422a97
Call-ID: 000dbc35-15b20003-1f961f4a-4c3d87bf@172.16.9.188
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0