Settings to connect a Cisco 7960 handset running SIP firmware to SIPGATE UK
Moderators: I actually came on here to post these settings on the blog I used to have on here but which seems to have disappeared, so I'm dropping this in here in the hope these settings will be useful to someone else, or more likely me in 6 months time when I've forgotten how I did it! If it's not appropriate here please let me know where to put it!
The Cisco 7960 is one of the nicest looking phones, supporting 6 lines when configured with SIP firmware. This is available from Cisco or from selected mirrors.
I found jtech.net and useful very useful. ;)
I am using SIP firmware P0S3-07-4-00 as you can see from the default file.
Please note all the settings are configured to the 2nd button :) switch it to the first if you wish. I did find a slight lag in the conversation but a little tweaking or a faster internet speed might help with this.
The SIPDefault.cnf file
# SIP Default Generic Configuration File
# Image Version
image_version: P0S3-07-4-00
# Proxy Server
proxy1_address: "" ; Can be dotted IP or FQDN
proxy2_address: "sipgate.co.uk" ; 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: 8060
proxy2_port: 5060
proxy3_port: 8060
proxy4_port: 8060
proxy5_port: 8060
proxy6_port: 8060
# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1
# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 300 ; 55000
# 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: 300 ; 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: "ntp.sipgate.net" ; SNTP Server IP Address
sntp_mode: unicast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: GMT ; 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: 8060 ; Backup Proxy port (default is 5060)
# Emergency Proxy Support
proxy_emergency: "" ; Dotted IP of Emergency Proxy
proxy_emergency_port: 8060 ; 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: 5061 ; UDP port used for SIP messages (default - 5060)
start_media_port: 16384 ; Start RTP range for media (default - 16384) 10000
end_media_port: 32766 ; End RTP range for media (default - 32766) 56500And the SIP[your phone mac address here].cnf
phone_label: "cisco"
phone_password: "cisco"
phone_prompt: "SIP Phone"
line1_name: "UNPROVISIONED"
line1_shortname: "UNPROVISIONED"
line1_displayname: "UNPROVISIONED"
line1_authname: "UNPROVISIONED"
line1_password: "UNPROVISIONED"
line2_name: "SIPUSERID"
line2_shortname: "Sipgate"
line2_displayname: "xxxxxxx"
line2_authname: "SIPUSERID"
line2_password: "PASSWORD"
line3_name: "UNPROVISIONED"
line3_shortname: "UNPROVISIONED"
line3_displayname: "UNPROVISIONED"
line3_authname: "UNPROVISIONED"
line3_password: "UNPROVISIONED"
line4_name: "UNPROVISIONED"
line4_shortname: "UNPROVISIONED"
line4_displayname: "UNPROVISIONED"
line4_authname: "UNPROVISIONED"
line4_password: "UNPROVISIONED"
line5_name: "UNPROVISIONED"
line5_shortname: "UNPROVISIONED"
line5_displayname: "UNPROVISIONED"
line5_authname: "UNPROVISIONED"
line5_password: "UNPROVISIONED"
line6_name: "UNPROVISIONED"
line6_shortname: "UNPROVISIONED"
line6_displayname: "UNPROVISIONED"
line6_authname: "UNPROVISIONED"
line6_password: "UNPROVISIONED"Good luck!
Dave