Original Template found at http://www.voip-info.org/wiki/view/cisco+mass+deployment If you are using a NAT proxy please make sure to forward UDP ports 5060/5060 and 16384/32768  (Start port/End Port) to your internal IP number for your phone you might want to setup a static IP or have your router assign one for the Phone.  Along with other config files mentioned at " http://www.voip-info.org/wiki/view/cisco+mass+deployment" include these two files in your tftp directory for your phone, "SIPDefault.cnf" and "SIPXXXXXXXXX.cnf" XX represent your device's Mac Address.  I am also attaching some of the configuration files FILENAME: SIPDefault.cnf CONTENTS: # Image Version image_version: "P0S3-08-3-00" # Input your specific Firmware version here # Proxy Server proxy1_address: " sip.callwithus.com" # IP address here alternatively # Proxy Server Port (default - 5060) proxy1_port:"5060" # Emergency Proxy info proxy_emergency: "" # IP address here alternatively proxy_emergency_port: "5060" # Backup Proxy info proxy_backup: "" proxy_backup_port: "5060" # Outbound Proxy info outbound_proxy: "" outbound_proxy_port: "5060" # NAT/Firewall Traversal nat_enable: "1" nat_address: "XX.XXX.XXX.XX" # Your external IP adress voip_control_port: "5060" start_media_port: "16384" end_media_port: "32768" 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" # 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" ~np~# 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: "sip.callwithus.com" # IP address here alternatively time_zone: " EST" # Set to your timezone dst_offset: "1" dst_start_month: "April" dst_start_day: "" dst_start_day_of_week: "Sun" dst_start_week_of_month: "1" 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: "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: "0" # URL for branding logo logo_url: " http://pbx.mycompany.com/cisco/logo.bmp" # Remote Party ID remote_party_id: 1 ; 0-Disabled (default), 1-Enabled FILENAME: SIPXXXXXXXXX.cnf Where the XXs are the MAC address of your Cisco 7960 CONTENTS: # SIP Configuration Generic File # Image Version image_version: P0S3-08-3-00 phone_label: " " # Line 1 appearance line1_displayname: "username" line1_shortname:"CW" line1_name: username line1_authname: "username" line1_password: "password" # Line 2 appearance line2_displayname: "" line2_shortname: "" line2_name: UNPROVISIONED line2_authname: "UNPROVISIONED" line2_password: "UNPROVISIONED" # Line 3 appearance line3_displayname: "" line3_shortname: "" line3_name: UNPROVISIONED line3_authname: "UNPROVISIONED" line3_password: "UNPROVISIONED" # Line 4 appearance line4_displayname: "" line4_shortname: "" line4_name: UNPROVISIONED line4_authname: "UNPROVISIONED" line4_password: "UNPROVISIONED" # Line 5 appearance line5_displayname: "" line5_shortname: "" line5_name: UNPROVISIONED line5_authname: "UNPROVISIONED" line5_password: "UNPROVISIONED" # Line 6 appearance line6_displayname: "" line6_shortname: "" line6_name: UNPROVISIONED line6_authname: "UNPROVISIONED" line6_password: "UNPROVISIONED" # Phone Prompt (The prompt that will be displayed on console and telnet) phone_prompt: "SIP Phone" ; Limited to 15 characters (Default - SIP Phone) # Phone Password (Password to be used for console or telnet login) phone_password: "cisco" ; Limited to 31 characters (Default - cisco) # User classifcation used when Registering [ none(default), phone, ip ] user_info: none dialplan.xml: