I am using Trixbox with a Cisco 7960, here are the steps I used to install my server and phone. My instructions are very brief, you should find that the context sensitive help in the Trixbox interface quite useful for helping with your own customisations.
Install Trixbox CE 2.6.2 from Disk/ISO image downloaded from the Fonality website, this part is straightforward enough not to require instructions.
Once installation is complete and you have logged on as root then run the following commands:
system-config-network - to switch from dynamic (DHCP) to static IP
passwd-maint - specify the maint users (gui) password
setup-cisco - generate cisco script
setup-samba - enable CIFS (\\server\share)
vi /etc/resolv.conf and add nameserver [your nameserver ip here]
reboot
Now connect to the Trixbox web interface in your browser
http://[ip address of your Trixbox server]
In the top right "switch" to Admin mode by logging on as the maint user
Select PBX/PBX Settings menu
Extensions:
Add Extension, Generic SIP Device, Submit
User Extension: 203
Display Name: Office
Device Options
Secret: 203
Voicemail & Directory
Status: Enabled
Voicemail password: 203
Submit & Apply
Select Extension 203
Device Options
Set nat to never
General Settings
Allow anonymous Inbound SIP Calls? Yes
Submit & apply
Trunks
Add new SIP trunk
Outbound caller ID: [your_sip_tel_no]
Outgoing Settings
Trunk name: SipgateTrunk
PEER Details:
fromdomain=sipgate.co.uk
fromuser=[your_sipgate_id]
host=sipgate.co.uk
username=[your_sipgate_id]
secret=[your_sipgate_password]
type=peer
insecure=very
nat=no
qualify=yes
Registration: [your_sipgate_id]:[your_sipgate_password]@sipgate.co.uk/[your_sipgate_id]
Submit & apply
Outbound Routes
Route name: Default
Dial Patterns:
10000
0.
9|.
Trunk Sequence 0 SipgateTrunk
Submit & apply
Ring Groups
600
Group Description: Office
Extension list: 203
Destination if no answer: Voicemail [203]
Submit & apply
Inbound Routes
DID Number: [your_sipgate_id]
Ring Groups: Office [600]
Submit & apply
Configure the Cisco phone with the Endpoint manager
PBX/Endpoint Manager
Cisco Phone
Device ID (extension): 203
Phone Type: 7960
MAC address: 800613800613 (your phones mac here)
My SIPDefault file, located in:
\\192.168.1.10\share\tftpboot\SIPDefault
# Image Version
image_version: "P0S3-08-8-00"
# Proxy Server
proxy1_address: "192.168.1.10"
proxy2_address: ""
proxy3_address: ""
proxy4_address: ""
proxy5_address: ""
proxy6_address: ""
# Proxy Server Port (default - 5060)
proxy1_port:"5060"
proxy2_port:""
proxy3_port:""
proxy4_port:""
proxy5_port:""
proxy6_port:""
# Emergency Proxy info
proxy_emergency: "192.168.1.10"
proxy_emergency_port: "5060"
# Backup Proxy info
proxy_backup: ""
proxy_backup_port: "5060"
# Outbound Proxy info
outbound_proxy: ""
outbound_proxy_port: "5060"
# NAT/Firewall Traversal
nat_enable: "false"
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: "0"
# 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"
#********* Release 2 new config parameters **********
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: "./"
# Time Server
sntp_mode: "unicast"
sntp_server: "192.168.1.10"
time_zone: "GMT"
dst_offset: "1"
dst_start_month: "Mar"
dst_start_day: ""
dst_start_day_of_week: "Sun"
dst_start_week_of_month: "2"
dst_start_time: "02"
dst_stop_month: "Nov"
dst_stop_day: ""
dst_stop_day_of_week: "Sunday"
dst_stop_week_of_month: "1"
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"
# URL for external Phone Services
services_url: "http://192.168.1.10/xmlservices/index.php"
# URL for external Directory location
directory_url: "http://192.168.1.10/xmlservices/PhoneDirectory.php"
# URL for branding logo
logo_url: "http://192.168.1.10/cisco/bmp/asterisk.bmp"
My SIP file, located in:
\\192.168.1.10\share\tftpboot\SIP800613800613
# Cisco SIP Configuration
phone_label: "Office"
line1_name: "203"
line1_shortname: "203"
line1_displayname: "203"
line1_password: "203"
line2_name: "UNPROVISIONED"
line2_shortname: "UNPROVISIONED"
line2_displayname: "UNPROVISIONED"
line2_password: "UNPROVISIONED"
line3_name: "UNPROVISIONED"
line3_shortname: "UNPROVISIONED"
line3_displayname: "UNPROVISIONED"
line3_password: "UNPROVISIONED"
line4_name: "UNPROVISIONED"
line4_shortname: "UNPROVISIONED"
line4_displayname: "UNPROVISIONED"
line4_password: "UNPROVISIONED"
line5_name: "UNPROVISIONED"
line5_shortname: "UNPROVISIONED"
line5_displayname: "UNPROVISIONED"
line5_password: "UNPROVISIONED"
line6_name: "UNPROVISIONED"
line6_shortname: "UNPROVISIONED"
line6_displayname: "UNPROVISIONED"
line6_password: "UNPROVISIONED"
line1_authname: "203"
line2_authname: "UNPROVISIONED"
line3_authname: "UNPROVISIONED"
line4_authname: "UNPROVISIONED"
line5_authname: "UNPROVISIONED"
line6_authname: "UNPROVISIONED"