Trixbox CE 2.6.2 (Sipgate) and Cisco 7960 installation walkthrough

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"