By default, Cisco 7960G comes with SCCP image, so we need to convert it to SIP which is supported by most VOIP service provider on the market. The Cisco document describes how to do the conversion:

http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080094584.shtml , The following chapters give step-by-step ‘working-proved’ instruction with the real sample.

1) Setup TFTP server

a) Install TFTP32

Download from http://tftpd32.jounin.net/tftpd32_download.html

Install on a PC, if the PC ip address is 192.168.1.102, Setup as below

b) Configure TFTP server

Tftp32 Setting

DHCP Setting

2) Prepare SIP configuration files

c) Get 7960G SIP image from Cisco web site.(Registed user only), extract it to C:\tftproot

http://tools.cisco.com/support/downloads/go/Redirect.x?mdfid=278875240

The file RINGLIST.DAT, dailplan.xml, XMLDefault.cnf.xml may not be found from the site, you can create them by yourself, Below are the working samples.

d) XMLDefault.cnf.xml

<ports>

</ports>

</callManager>

</member>

</members>

</callManagerGroup>

<loadInformation7 model="Cisco 7960">P0S3-08-11-00</loadInformation7>

</Default>

e) RINGLIST.DAT

ringer1.pcm

ringer2.pcm

f) dialplan.xml (it can be different, if you don’t use voipdiscount)

<TEMPLATE MATCH="911" Timeout="0" Rewrite="0015107914200 " User="Phone"/> <!-- Fremont, CA, Emergency 911 -->

<TEMPLATE MATCH="*" Timeout="15" User="Phone"/> <!-- Anything else -->

</DIALTEMPLATE>

g) OS79XX.TXT (must contain the version of SIP image)

P003-08-11-00

h) SIPDefault.cnf (Important parameters are marked bold)

# SIP Default Generic Configuration File

# Image Version

image_version: P0S3-08-11-00

# Proxy Server

proxy1_address: "sip.voipdiscount.com" ; Can be dotted IP or FQDN

proxy2_address: "eu.voxalot.com" ; For a voice mail account

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: 60 ; although default is 3600, I set it as 60, otherwise, has an engaged line issue when other phone calls in.

# Codec for media stream (g711ulaw (default), g711alaw, g729a)

preferred_codec: g711ulaw

# TOS bits in media stream [0-5] (Default - 5)

tos_media: 5 ; it is obsolete, remove this line, otherwise phone initialization will show error

# 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: "0.north-america.pool.ntp.org" ; SNTP Server IP Address

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

time_zone: PST ; 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, 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: 1 ; 0-Disabled (default), 1-Enabled

nat_address: "192.168.1.1" ; WAN IP address of NAT box (dotted IP or DNS A record only), this is your home router/gateway ip

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

####### 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

# RTP Call Statistics (SIP BYE/200 OK message exchange)

call_stats: 0 ; 0-Disabled (default), 1-Enabled

i) SIP<YourPhoneMacAddress>.cnf

# SIP Configuration Generic File

# Line 1 appearance

line1_name: <your verified phone number> ; please refer to 9) Caller ID setting

# Line 1 short name

line1_shortname: "Daniel"

# Line 1 Registration Authentication

line1_authname: "<Your voipdiscount account name>"

# Line 1 Registration Password

line1_password: "<Your voipdiscount account password>"

# Line 2 appearance, it must be your voxalot account name, otherwise, can not register to voxalot

line2_name: <Your voxalot account name>

# Line 2 Registration Authentication

line2_authname: "<Your voxalot account name>"

# Line 2 Registration Password

line2_password: "<Your voxalot account 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: "Daniel"

# Line 1 Display Name (Display name to use for SIP messaging)

line1_displayname: "Daniel"

# Line 2 Display Name (Display name to use for SIP messaging)

line2_displayname: "Angela"

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

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

user_info: none

2. Phone Setup

3) Reset TFTP server IP

Now, we need to change the current TFTP server IP to our own TFTP32 ip address, in this example, it is 192.168.1.102, follow these steps on the phone:

j) Press settings->Network Configuration-># (# is to unlock the network configuration setting)

Note: After the SIP image is loaded, the unlock operation is different as settings->unlock Config, then type the password which is set in SIPDefault.cnf, In the above sample, the password is “a”, by default is “cisco”

k) Go to item 32 Alternate TFTP, change NO to Yes

l) Go to item 8 TFTP Server1, change IP to 192.168.1.102 (press * for . input)

m) Press Save softkey, then exit

3. Home router setup

You need to open UDP port forwarding for the IP phone

Below is the sample of Linksys WRT54G configuration

4. Setup incoming call

From www.ipkall.com , apply an account, you can get a free PSTN number with WA area code, in the application, give your <viopdiscount user id> as SIP phone number, sip.discount.com as SIP proxy server. In about four days, you will receive the free phone number through your registered email. Now, call this number, your voip phone will ring.

Note: the ipkall account can be expired if the number is not active (called and answered) in any 30 continuous days.

5. Setup voice mail

It is my favorite part, configure a voice mail system on the IP phone, whenever no one can answer the phone or the line is engaged, the voice mail system is invoked, the calling-party can leave a message on the internet mail server. After few seconds, the IP phone will receive a MWI (message waiting indicator) signal, and the LED light on the headset will be turned on. In the mean time, an email with the voice message (.wav file) has also been delivered to your email box. It is convenient and I enjoy it so much, below are the detail configuration.

4) Register an online voice mail account

I use a free service from www.voxalot.com, register a free account, in the account configuration, add a voice service provider

5) Add the voice mail account to IP phone line 2

Please refer to h) SIPDefault.cnf

6) Forward ipkall to the voice mail account

Now, change the ipkall configuration in Chapter 5 to the following:

SIP phone number: <the voxalot account number>

SIP proxy: eu.voxalot.com

7) Check the voice mail

Use line 2 to check the voice mail, dial *500, provide pin number (you set it from voxalot.com), then follow the voice prompt.

Or you can set the following line in SIP<macAddress>.cnf, so that you directly press the messages button on the phone to access your voice mail:

messages_uri: *500

line1_name: <your verified phone number> ; please refer to 9) Caller ID setting

n) Call any phone number over the world. (remember dial 00 as prefix, or create a dialplan to customize your dial out number)

o) Receive a call from the PSTN number you registered from IPKall.com

p) Access the voice mail from voxalot account.

8. Back up

12) Debug the phone

C:>telnet 192.168.1.103

Password :*

Cisco IP phone MAC:

Loadid: SW: P0S3-08-11-00 ARM: PAS3ARM1 Boot: PC030301 DSP: 4.0(5.0)[A0]

SIP Phone> sh status

Current Phone Status

--------------------

W350 unprovisioned proxy_backup <<< It is ok>>>

W351 unprovisioned proxy_emergency <<< it is ok>>>

SIP Phone> sh network

------ Network *FLASH* Configuration ------

Platform : Cisco Systems, Inc. IP Phone CP-7960G

Elapsed Time: 00:04:03

dhcp_server : 192.168.1.102

my_ip_addr : 192.168.1.103

subnet_mask : 255.255.255.0

defaultgw : 192.168.1.1

dyn_dns_addr_1 : 0.0.0.0

dyn_dns_addr_2 : 0.0.0.0

dns_addr : 192.168.1.1

primary_tftp_addr : 192.168.1.102

dyn_tftp_addr : 0.0.0.0

my_mac_addr :

domain_name :

my_name : SIP

Status Flags : 12310000

SIP Phone> sh regi

LINE REGISTRATION TABLE

Proxy Registration: ENABLED, state: REGISTERING

line APR state timer expires proxy:port

---- --- ------------- ---------- ---------- ----------------------------

1 111 REGISTERED 3595 3374 sip.voipdiscount.com:5060

2 111 REGISTERED 3600 2 eu.voxalot.com:5060

3 ... NONE 0 0 undefined:0

4 ... NONE 0 0 undefined:0

5 ... NONE 0 0 undefined:0

6 ... NONE 0 0 undefined:0

1-BU .1x IDLE 0 0 undefined:0

Note: APR is Authenticated, Provisioned, Registered

SIP Phone>dns –s 192.168.1.1 <<< set your home router ip>>>

SIP Phone> debug sip-messages