1. How To Setup Asterisk On the MCF54451
DRAFT – VERSION 0.5
January 6/08

2. Starting Asterisk 1. Log into board: freescale login: user
Password: user
2. Change to super user
$su root
Password: root
Autostart Note:
For convenience a script example has been created:
/etc/rc.d/init.d/asteriskd
3. Run following (manual)
# modprobe mcf_wcfxs
# ztcfg
Asterisk debug verbosity
4. Start Asterisk
#asterisk -vvvvvvv

3. Asterisk Initialization
Asterisk , Copyright (C) Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= <verbose startup output> Asterisk Ready. #

4. Logging Into Web UI
IP address of board
Username: admin
Password: admin

5. GUI Initialization
Asterisk GUI will run some autoconfig

6. GUI Home Page APPLY CHANGES – VERY IMPORTANT System Status
Trunk Status
Extension
Status
Setup
Menus

7. Trunks and Extensions Trunks Extensions
Connect with external telephony elements
FXO port (Si3050) is an analog trunk that connects with PSTN
Extensions
Act like users on the phone system
May be digital or analog
Digital may be IAX / SIP or Analog
FXS (Si3210) is an analog extension that connect to a POTS (plain old telephone service) phone

8. Configuring an FXO Trunk
1. Select
Trunk Menu
2. Click: New Analog Trunk
3. Check FXO hardware channel (2)
4. Name trunk: FXO
5. Click add
6. APPLY CHANGES
APPLY CHANGES – VERY IMPORTANT

9. Confirming the FXO Trunk
1. Select
Status Menu

10. Configuring an Extension
To configure an extension you need to:
Create a dial plan
Create a User
Setup number
Define the type of extension (SIP/IAX/Analog)
Select the capabilities (3WayCalling/in-directory/call wait…)

11. About the Dialplan
The dialplan defines the rules that are assigned to each user. These rules define how outgoing calls will be handled. It can be used to restrict calling or route calls in different ways based on the number dialed (least cost routing).
The dialplan also defines the number of digits and starting number for extensions, parked calls, conferences, voice mail …..

12. Creating a Dial Plan 2. Click New Dial Plan 1. Select Dial Plan Menu
3. Use default
Settings
4. Click Save
5. APPLY CHANGES
APPLY CHANGES – VERY IMPORTANT

13. Making the Dial Plan Default
1. Select
Dial Plan
Menu
3. Check Default Box
APPLY CHANGES – VERY IMPORTANT

14. Creating a SIP user 2. Define Name 3. Select dial plan 1. Select User
Menu + click create new user
4. Enable voic
5. set password
6. Select SIP
Only
7. Confirm
u-law
9. Define capabilities
8. Set Password to extension number
10. Click update
11. APPLY CHANGES
APPLY CHANGES – VERY IMPORTANT

15. Configuring Zoiper PC SIP Client
1. Click tools icon
2. Add new SIP account
3. Enter Server IP
4. Enter User Account and password
User Account Information
Username: 6000
Password: 6000
5. Click OK

16. Confirming Success 1. Green Light! 2. Registered! 1. Select
Status Menu
1. Green Light!
2. Registered!

17. Setting Up an Analog Port (FXS)
Set up user as described before
Configure for an Analog Station on Port-1

18. Creating an Analog User (FXS)
1. Select User
Menu
2. Define Name
3. Select dial plan
4. Enable voic
5. set password
6. Select
Analog Station: port-1
7. Confirm
u-law
9. Click update
10. APPLY CHANGES
8. Define Capabilities
APPLY CHANGES – VERY IMPORTANT

19. Confirming Success
1. Green Light!

20. Disabling Echo Canceller
1. Select Options
Menu
2. Select “Show advanced options”
ADVANCED OPTIONS ARE DISPLAYED HERE
3. Select “File Editor”
APPLY CHANGES – VERY IMPORTANT

21. Disabling Echo Canceller
1. Select zapata.conf file
2. Expand Channels menu
3. Click on List to open editor
FILE EDITOR IS DISPLAYED HERE

22. Disabling Echo Canceller
3. Save & Apply changes
1. Edit echocanel=no
APPLY CHANGES – VERY IMPORTANT

23. Notes
After you have configured hardware you will need to restart Asterisk for the hardware to init properly
# killall asterisk
After you have reset Asterisk your SIP client will need to register with Asterisk to work properly (unreg/reg the zoiper client or wait for re-registration)