cisco mass deployment

Cisco Mass Deployment


A work in progress................NOT complete so help instead of complain......I work with Media Wiki all day and this TIKI thing is much different...

See also Cisco 7940 7960 Single Step Upgrade for some other configuration ideas

Cisco phones like to operate in a client server setup. The phone being the client looks to the
server for information on how to run. This is great for mass deployment but not so cool for single
phone installs. Well the good news is that everyone can use these methods fo seting up
the phones.

Things you need to have
  • DHCP server
  • TFTP server
  • Config Files

DHCP


Dynamic Host Configuration Protocol is the fun little tool that gives you an IP address down at the coffee
house. It is very easy to work with but has some more advanced tools to aid us in our work.

Setup


DHCP must be setup to give the phones a TFTP server and proper networking information. DNS is prefered
but it can be hard coded,

Options


Example



incomplete......

ddns-update-style none;

subnet 192.168.200.0 netmask 255.255.255.0 {
option routers 192.168.200.1;
option subnet-mask 255.255.255.0;
option domain-name "internal.lan";
option domain-name-servers ns.internal.lan;

range dynamic-bootp 192.168.200.100 192.168.200.200;
default-lease-time 21600;
max-lease-time 43200;

host Phone01 {
hardware ethernet 12:34:56:78:AB:CD;
fixed-address 192.168.200.101;
}
}



TFTP


Setup


Files


Config files


About


Naming


Examples

SIPDefault.cnf

# Image Version
image_version: "P0S3-07-4-00"

# Proxy Server
proxy1_address: "pbx.mycompany.com" # IP address here alternatively

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

# Emergency Proxy info
proxy_emergency: "pbx.mycompany.com" # IP address here alternatively
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: "pbx.mycompany.com"
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_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: "3600"

# 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: "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: "pbx.mycompany.com" # IP address here alternatively
time_zone: "PST"
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 external Phone Services
services_url: "http://pbx.mycompany.com/cisco/services.php" # IP address here alternatively

# URL for external Directory location
directory_url: "http://pbx.mycompany.com/cisco/directory.php" # IP address here alternatively

# URL for branding logo
logo_url: "http://pbx.mycompany.com/cisco/logo.bmp" # IP address here alternatively

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


SIPXXXXXXXXX.cnf Where the XXs are the MAC address.

# SIP Configuration Generic File

# Image Version
image_version: P0S3-07-4-00
phone_label: " "

# Line 1 appearance
line1_displayname: "x100"
line1_shortname:"x100"
line1_name: 100
line1_authname: "myid"
line1_password: "mypassword"

# 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 (filename case sensitive and must match entry in SIPDefault.cnf above)

<DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>


OS79XX.txt (read by bootloader & contains your firmware version to use)

P003-07-4-00


xmlDefault.CNF.XML & XMLDefault.cnf.xml (Both to cover most firmware versions typo's)

<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName></processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation8 model="IP Phone 7940">P003-07-4-00</loadInformation8>
<loadInformation7 model="IP Phone 7960">P003-07-4-00</loadInformation7>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>


Andrew Latham - lathama at lathama dot com



Related Links:

Cisco Mass Deployment


A work in progress................NOT complete so help instead of complain......I work with Media Wiki all day and this TIKI thing is much different...

See also Cisco 7940 7960 Single Step Upgrade for some other configuration ideas

Cisco phones like to operate in a client server setup. The phone being the client looks to the
server for information on how to run. This is great for mass deployment but not so cool for single
phone installs. Well the good news is that everyone can use these methods fo seting up
the phones.

Things you need to have
  • DHCP server
  • TFTP server
  • Config Files

DHCP


Dynamic Host Configuration Protocol is the fun little tool that gives you an IP address down at the coffee
house. It is very easy to work with but has some more advanced tools to aid us in our work.

Setup


DHCP must be setup to give the phones a TFTP server and proper networking information. DNS is prefered
but it can be hard coded,

Options


Example



incomplete......

ddns-update-style none;

subnet 192.168.200.0 netmask 255.255.255.0 {
option routers 192.168.200.1;
option subnet-mask 255.255.255.0;
option domain-name "internal.lan";
option domain-name-servers ns.internal.lan;

range dynamic-bootp 192.168.200.100 192.168.200.200;
default-lease-time 21600;
max-lease-time 43200;

host Phone01 {
hardware ethernet 12:34:56:78:AB:CD;
fixed-address 192.168.200.101;
}
}



TFTP


Setup


Files


Config files


About


Naming


Examples

SIPDefault.cnf

# Image Version
image_version: "P0S3-07-4-00"

# Proxy Server
proxy1_address: "pbx.mycompany.com" # IP address here alternatively

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

# Emergency Proxy info
proxy_emergency: "pbx.mycompany.com" # IP address here alternatively
proxy_emergency_port: "5060"

# Backup Proxy info
proxy_backup: "pbx.mycompany.com"
proxy_backup_port: "5060"

# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_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: "3600"

# 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: "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: "pbx.mycompany.com" # IP address here alternatively
time_zone: "PST"
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 external Phone Services
services_url: "http://pbx.mycompany.com/cisco/services.php" # IP address here alternatively

# URL for external Directory location
directory_url: "http://pbx.mycompany.com/cisco/directory.php" # IP address here alternatively

# URL for branding logo
logo_url: "http://pbx.mycompany.com/cisco/logo.bmp" # IP address here alternatively

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


SIPXXXXXXXXX.cnf Where the XXs are the MAC address.

# SIP Configuration Generic File

# Image Version
image_version: P0S3-07-4-00
phone_label: " "

# Line 1 appearance
line1_displayname: "x100"
line1_shortname:"x100"
line1_name: 100
line1_authname: "myid"
line1_password: "mypassword"

# 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 (filename case sensitive and must match entry in SIPDefault.cnf above)

<DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="5"/> <!-- Anything else -->
</DIALTEMPLATE>


OS79XX.txt (read by bootloader & contains your firmware version to use)

P003-07-4-00


xmlDefault.CNF.XML & XMLDefault.cnf.xml (Both to cover most firmware versions typo's)

<Default>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<mgcpPorts>
<listen>2427</listen>
<keepAlive>2428</keepAlive>
</mgcpPorts>
</ports>
<processNodeName></processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<loadInformation8 model="IP Phone 7940">P003-07-4-00</loadInformation8>
<loadInformation7 model="IP Phone 7960">P003-07-4-00</loadInformation7>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<servicesURL></servicesURL>
</Default>


Andrew Latham - lathama at lathama dot com



Related Links:

Created by: lathama, Last modification: Wed 07 of Sep, 2005 (20:01 UTC) by devguy
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