Where can I get TFTP config files for Cisco 7921 to use with an Asterisk box?

# SIP Default Generic Configuration File 
 
# Image Version
image_version: P0S3-08-6-00
 
# Proxy Server
proxy1_address: "192.168.5.10"		; Can be dotted IP or FQDN
proxy2_address: ""		; Can be dotted IP or FQDN
proxy3_address: ""		; Can be dotted IP or FQDN
proxy4_address: ""		; Can be dotted IP or FQDN
proxy5_address: ""		; Can be dotted IP or FQDN
proxy6_address: ""		; Can be dotted IP or FQDN
 
# Proxy Server Port (default - 5060)
proxy1_port: 5060 
proxy2_port: 5060 
proxy3_port: 5060 
proxy4_port: 5060 
proxy5_port: 5060 
proxy6_port: 5060 
 
# 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, g729a)
preferred_codec: g711ulaw
 
# TOS bits in media stream [0-5] (Default - 5)
#tos_media: 5
dscpForAudio: 184
 
# 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 10
sip_invite_retx: 6 		; Default 6
timer_invite_expires: 180 	; Default 180 sec
 
####### New Parameters added in Release 2.0 #######
 
# Dialplan template (.xml format file relative to the TFTP root directory)
dial_template: dialplan
 
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: ""		; Example:  ./sip_phone/
 
# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "192.168.5.10"			; SNTP Server IP Address
sntp_mode: anycast	; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: EAST			; Time Zone Phone is in
dst_offset: 1			; Offset from Phone's time when DST is in effect 
dst_start_month: April		; Month in which DST starts
dst_start_day: ""		; Day of month in which DST starts
dst_start_day_of_week: Sun	; Day of week in which DST starts
dst_start_week_of_month: 1	; Week of month in which DST starts
dst_start_time: 02		; Time of day in which DST starts
dst_stop_month: Oct		; Month in which DST stops
dst_stop_day: ""		; Day of month in which DST stops
dst_stop_day_of_week: Sunday	; Day of week in which DST stops
dst_stop_week_of_month: 8	; Week of month in which DST stops 8=last week of month
dst_stop_time: 2		; Time of day in which DST stops
dst_auto_adjust: 0		; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1		; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
date_format : D/M/Y
 
# 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-disbaled, 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-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)
anonymous_call_block: 0		; Default 0 (Disable blocking of anonymous calls)
 
# DTMF AVT Payload (Dynamic payload range for AVT tones - 96-127)
dtmf_avt_payload: 101		; Default 101
 
# Sync value of the phone used for remote reset 
sync: 1				; Default 1
 
####### New Parameters added in Release 2.1 #######
 
# Backup Proxy Support
proxy_backup: ""		; Dotted IP of Backup Proxy
proxy_backup_port: 5060		; Backup Proxy port (default is 5060)
 
# Emergency Proxy Support
proxy_emergency: "" 		; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060	; Emergency Proxy port (default is 5060)
 
# Configurable VAD option
enable_vad: 0			; VAD setting 0-disable (Default), 1-enable
 
####### New Parameters added in Release 2.2 ######
 
# NAT/Firewall Traversal
nat_enable: 0                   ; 0-Disabled (default), 1-Enabled
nat_address: ""		        ; WAN IP address of NAT box (dotted IP or DNS A record only)
voip_control_port: 5060      	; UDP port used for SIP messages (default - 5060)
start_media_port: 16384 	; Start RTP range for media (default - 16384)
end_media_port: 32766   	; End RTP range for media (default - 32766)
nat_received_processing: 0	; 0-Disabled (default), 1-Enabled
 
# Outbound Proxy Support
outbound_proxy: "192.168.5.10"	 	; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060       ; default is 5060
 
####### New Parameter added in Release 3.0 #######
 
# Allow for the bridge on a 3way call to join remaining parties upon hangup
cnf_join_enable : 1		; 0-Disabled, 1-Enabled (default)
 
####### New Parameters added in Release 3.1 #######
 
# 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 the phone) 
telnet_level: 2			; 0-Disabled (default), 1-Enabled, 2-Privileged
 
####### New Parameters added in Release 4.0 #######
			; 0-Disabled (default), 1-Enabled
 
# XML URLs
services_url: "http://192.168.5.10/services.xml"		; URL for external Phone Services
directory_url: "http://192.168.5.10/pcg_dir.xml"		; URL for external Directory location
logo_url: "http://192.168.5.10/pcg.bmp"			; URL for branding logo to be used on phone display
 
# HTTP Proxy Support
http_proxy_addr: ""		; Address of HTTP Proxy server
http_proxy_port: 80		; Port of HTTP Proxy Server (80-default)
 
# Dynamic DNS/TFTP Support
dyn_dns_addr_1: ""              ; restricted to dotted IP
dyn_dns_addr_2: ""              ; restricted to dotted IP
dyn_tftp_addr: ""               ; restricted to dotted IP
 
# Remote Party ID
remote_party_id: 0		; 0-Disabled (default), 1-Enabled
 
####### New Parameters added in Release 4.4 #######
 
# Call Hold Ringback (0-off, 1-on, 2-off with no user control, 3-on with no user control)
call_hold_ringback: 0		; Default 0 (Call Hold Ringback feature is off)
 
####### New Parameters added in Release 6.0 #######
 
# Dialtone Stutter for MWI
stutter_msg_waiting: 0		; 0-Disabled (default), 1-Enabled
 
#Voice Mail extention
messages_uri: 8500
 
# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 0
 
#Relase DHCP
dhcp_address_released: yes
#Transfer by hanging up the phone
transfer_onhook_enabled:1
 
#Dissable Call Forward
local_cfwd_enable: 0
 
############# SIPMACADDRESS####################
# SIP Configuration Generic File 
 
# Line 1 appearance
line1_name:303
 
# Line 1 Registration Authentication 
line1_authname: "303"
 
# Line 1 Registration Password
line1_password: "11223344"
 
# Line 2 appearance
line2_name:
 
# Line 2 Registration Authentication
line2_authname: ""
 
# Line 2 Registration Password
line2_password: ""
 
 
####### New Parameters added in Release 2.0 #######
 
# All user_parameters have been removed
 
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: ""	; Has no effect on SIP messaging
 
# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "303"
 
# Line 2 Display Name (Display name to use for SIP messaging)
line2_displayname: ""
 
 
####### New Parameters added in Release 3.0 ######
 
# 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: "123" ; Limited to 31 characters (Default - cisco)
 
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none 
                                              

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