Cisco-7940

Configuring Cisco 7940/7960 to work with SIP requires several steps. First, download the latest version of software update here: http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960Next, create a configuration based on the pattern described below. When the phone has downloaded this information from TFTP-server and is on, it's ready for work. Let's describe step by step. When the phone has downloaded this information from the TFTP-server and is turned on, it is ready for use. When the phone is on, it requests the following information from the TFTP-server:

  • The last file of software update
  • Dual-boot file (OS79XX.TXT)
  • A configuration file created specifically for this phone (MAC-address included in name)
  • Configuration file by default
  • Ring-list file
  • Dial-plan file
DHCP-server should provide the following options (or set it manually):

  • dhcp option #1 (subnet mask)
  • dhcp option #3 (gateway by default)
  • dhcp option #6 (DNS-server address)
  • dhcp option #15 (domain name)
  • dhcp option #50 (IP-address)
  • dhcp option #66 (TFTP-server address)
Initialization process of a Cisco IP-phone

1. Phone downloads software update file.

2. Phone gets its VLAN number – in order to get parameters from DHCP-server, in case of connection to gateway Cisco Catalyst, the phone should get number of the Voice-VLAN, which is set on gateway

3. Phone gets IP-address – from DHCP-server or from own settings.

4. Phone downloads the following files from TFTP-server or from memory:

  • SEP.cnf.xml – Created on TFTP-server file SEP.cnf.xml which has the following data (software update version): <device><loadInformation>P0S3-08-2-00</loadInformation></device> Phone checks the version of its software and, if it mismatches with specified version, updates it.
  • <Software update version>.loads – if the version matches, the phone uses the file which already exists in memory, specified in file SEP.cnf.xml.
  • OS79XX.TXT – this file provides compatibility of phone switching between SIP, MGCP or SCCP using the same TFTP-server.
  • SIPDefault.cnf – default parameters all phones, settings are described further.
  • SIP<MAC-address>.cnf – parameters of specific phone, structure is described further.
  • RINGLIST.DAT – lists files with ringtones and their location.
  • dialplan.xml - consists dial plan. It can be sent to phone with the help of Notify (NTFY) in Event-header.
5. Phone checks the software version.

Cisco IP-phone process of initialization

File name format should be SIPXXXXYYYYZZZZ.cnf, where XXXXYYYYZZZZ – MAC-address of phone in upper case letters.

Example of file's name: SIP00503EFFD842.cnf.

Each file line should be set as following:

variable_name : definition ; optional commentary

Key for the parameters which appear in the following example:

  • line1_name – number or Email address, used in registration. Number should be without hyphen, e-mail – without host name.
  • proxy1_address – IP-address of SIP proxy server, used by phone.
  • proxy1_port – Port number of SIP proxy server, used by phone.
Other parameters are changed in case of need. Example of configuration file SIPDefault.cnf:

# SIP Default Configuration File

# Image Version

image_version: P0S3-08-3-00

# Proxy Server

proxy1_address: 172.16.255.255

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

# 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: ""; SNTP Server IP Address

sntp_mode: anycast (default); unicast, multicast, or directedbroadcast

time_zone: EST; 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: 1; Enable(1-Default)/Disable(0) DST automatic adjustment

time_format_24hr: 1; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)

# Do Not Disturb Control

#(0-off (default), 1-on, 2-off with no user control, 3-on with no user control)

dnd_control: 0;

# Caller ID Blocking

#(0-disabled, 1-enabled, 2-disabled no user control, 3-enabled no user control)

callerid_blocking: 0; (Default is 0 - disabled and 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 is 0 - disabled and 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: 1; 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: 1; 0-Disabled (default), 1-Enabled

# Outbound Proxy Support

outbound_proxy: ""; 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: 1; 0-Disabled (default), 1-Enabled, 2-Privileged

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

# XML URLs

services_url: ""; URL for external Phone Services

directory_url: ""; URL for external Directory location

logo_url: ""; 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

Configuration file setup for separate IP-phone

The following parameters can be changed: anonymous_call_block, autocomplete, callerid_blocking, call_hold_ringback, call_waiting, dnd_control – other in case of need. File should be named as SIP.cnf. Example of configuration file for separate phone:

# SIP Configuration Generic File

# Line 1 appearance

line1_name: 1234567

# Line 1 Registration Authentication

line1_authname: "UNPROVISIONED"

# Line 1 Registration Password

line1_password: "UNPROVISIONED"

# Line 2 appearance

line2_name: football

# Line 2 Registration Authentication

line2_authname: "UNPROVISIONED"

# Line 2 Registration Password

line2_password: "UNPROVISIONED"

####### New Parameters added in Release 2.0 #######

# 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: "User ID"

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

# User classification used when Registering [ none (default), phone, ip ]

user_info: none

Parameters setup from Cisco IP phone menu

Certain parameters can be set manually using the Cisco IP phone menu. By default, the settings in Cisco IP phones 7940/7960 are blocked. To unblock, you need to enter a password, which is set in the configuration file. Do so by clicking Settings > Unlock Config. To block click Lock Config or Exit. Then save the changes made to parameters and the phone will be rebooted. Besides main settings, such as IP-address TFTP-server address, SIP parameters should be set in manual settings. After unblocking the phone's settings, select Settings > SIP Configuration. In the next menu set line1_name, proxy1_address, proxy1_port according to the information detailed above. If the phone should be authorized on a SIP proxy server, enter line1_authname and line1_password. By default, it is set as UNPROVISIONED.

Time settings shold be set from common configuration file, example for Moscow:

time_zone : BT

dst_offset : 01/00

dst_start_month : April

dst_start_day : 1

dst_start_time : 02/00

dst_stop_month : October

dst_stop_day : 1

dst_stop_time : 02/00

dst_stop_autoadjust : 1

DST – dates of day-light saving time correspondingly.

Note

When connecting to Zadarma IP PBX you need to use the internal IP PBX information that appears under "My PBX - Internal numbers".