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

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

Asterisk Initialization Asterisk 1.4.21.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster@digium.com> 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. #

Logging Into Web UI http://192.168.1.82:8088/asterisk/static/config/index.html IP address of 54451 board Username: admin Password: admin

GUI Initialization Asterisk GUI will run some autoconfig

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

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

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

Confirming the FXO Trunk 1. Select Status Menu

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…)

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 …..

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

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

Creating a SIP user 2. Define Name 3. Select dial plan 1. Select User Menu + click create new user 4. Enable voicemail 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

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

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

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

Creating an Analog User (FXS) 1. Select User Menu 2. Define Name 3. Select dial plan 4. Enable voicemail 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

Confirming Success 1. Green Light!

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

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

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

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)