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

I am testing a Cisco 7921 phone with Asterisk.  The phone is on loan from Cisco, but no software was provided.  I am trying to get any config files that I need to send registration info to the phone.  Does anyone know where I can get them?  I may be able to get them from Cisco, but thought that EE might be faster and more informative since I will be communicating with asterisk and not Call Manager.
DanRaposoAsked:
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MacNuttinCommented:
You know if you used Trixbox2.5 or higher End Point Configuration includes CISCO phones.

try running setup-cisco from the console or a terminal session
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grbladesCommented:
There is a lot of info about the Cisco phones on the Asterisk wiki at http://www.voip-info.org/wiki-cisco+79xx
Hopefully you can find what you are looking for there.
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bwilks99Commented:
Hi,

We use the Cisco 7940s I know it is not the same but it may give you some pointers.

SIPDefault.cnf
and SIPMACADDRESS.cnf
# 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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cscorbetCommented:
Your issue may be that your phones are not running SIP firmware?. By default I believe phones are supplied with Cisco SCCP, Skinny Client Control Protocol firmware and you may need to chang the firwamre to sip in order for the Cisco phones to work on an Asterisk box. If you have a maintenance agreement with Cisco youll be able to download the SIP firmware from them, cisco.com.
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DanRaposoAuthor Commented:
We abandoned the project becasue we  don't want to del with Skinny and Asterisk.
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