[Asterisk-Users] Howto Setup TFTP server on Linux for Cisco 7 960

Wed Jan 26 07:17:04 MST 2005

Yes, this is frustrating I know. In fact the wiki could be updated to
provide this info. Basically if you have the phones out of the box
(brand spankin new) then you probly have the SCCP image installed on it
by default. Your tftp server root will need a number of files to start
if this is the case. Ok with that said, most of this I had to figure out
on my own. Cisco's website as we all know is a pain in the a$$ to find
any useful info on how to do anything. 

Be sure to remove #comments before experimenting. ALSO, DO THIS WITH ONE
PHONE AT A TIME. If you have other phones plugged in they WILL
automatically try to upgrade :) ..

[OS79XX.txt]
P0S3-07-3-00

# This is the version we used, S stands for Sip, 7-3 stands for 7.3 ..
If you need firmware
you'll have to get them off Cisco's site, there was a posting recently
stating where to obtain these without a cisco login.

# once you have that file in place your SEP (yes, SEP) device will start
looking for the file with that extension. It cuts off the file
extension, for example in your tftp root you will need:

P0S3-07-3-00.sb2
P0S3-07-3-00.loads

#Once you have those 3 files your phones should start upgrading, be
careful though. It's been known that older versions that come on the
phones have bugs and can blow up (crash) if you try to put too large an
image on them.

# Moving on, once you get that completed your phone should boot and
start looking for the following files. Before I post them below, take
note on how this all works. First you have a general config file,
SIPDefault.cnf .. This contains such things as your proxy address, logo,
services, directories, ntp, that kind of stuff. The second is your
SIP<MAC ADDRESS>.cnf .. This is per phone, that contains your phone line
info, names, etc.. 

[sipdefault.cnf]

# Image Version
image_version: "P0S3-07-3-00"

# Proxy Server
proxy1_address: "192.168.1.17"

# Proxy Server Port (default - 5061)
#proxy1_port:"5060"

# NAT/Firewall Traversal
nat_enable: "0"
nat_address: ""
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: "120"

# 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: "1"      ; 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: "8500"

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

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

# Time Server
sntp_mode: "directedbroadcast"
sntp_server: "17.254.0.49"
time_zone: "CST"
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: "1"

####### New Parameters added in Release 4.0 #######

# XML URLs
#services_url: "http://192.168.1.65/menu.pl" ; URL for external Phone
Services
directory_url: "http://192.168.1.17/directories.xml"               
# URL for external Directory location
logo_url: "http://192.168.1.17/netlogic.bmp"                    ; URL
for branding logo to be used on phone display

# put your own logo in the logo_url location; I include the 10-20.com
one for reference in building your own

# 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: "192.168.1.2"              ; restricted to dotted IP
dyn_dns_addr_2: ""              ; restricted to dotted IP
dyn_tftp_addr: "192.168.1.2"   ; restricted to dotted IP

# The dynamic tftp server should be set to whatever your TFTP server is.
This way, it
# keeps the tftp server setting even though you might be using DHCP
(default behavior
# is to use the DHCP server as a tftp server, which is rarely correct.)

# Remote Party ID
remote_party_id: 1              ; 0-Disabled (default), 1-Enabled

# EOF

---------------
#WHEW THAT WAS A LOT
---------------

# Luckily the next file is a lot shorter, it should be self explanatory

[SIP00059BB47680.cnf]

image_version: P0S3-07-3-00

line1_name: 107 

# Line 1 Registration Authentication 
line1_authname: "107"

# Line 1 Registration Password
line1_password: "LADEDA"

## See the pattern? For another line you would put line2_authname: "107"
etc..

####### 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: "Matt S 107"	; Has no effect on SIP messaging

# Line 1 Display Name (Display name to use for SIP messaging)
line1_displayname: "Matt S"

####### 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: "BLAHBLAH" ; Limited to 31 characters (Default - cisco)

# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none 

-----Original Message-----
From: Jose Cruz (Branders IT) [mailto:JCruz at branders.com] 

But how about the config files (SIP...) that needs to be inside the tftp
server, where can I get a sample of that? That's where the images for
the firmwares of the ip phones come from, on boot right?