Transitioning of Exchange server 2003 to exchange server 2007 with Unified Messaging

 

Installed exchange server 2007 x64 on a 64bits server IBM with 6GB Memory

Updated service pack 2 for exchange 2007 x64

Transferred roles to the new server
- Public folders and offline address book for Free/Busy Schedule
- Replicated and check replications completed
- Move RUS to new server (although 2007 does not use RUS as 2003, but have to move to uninstall old 2003 server)
- Moved mailboxes to new server

Making Exchange to "Talk"

Installed Unified Communication Role on Exchange 2007

-Required windows Desktop feature

Integrating Trixbox CE

I have been using Asterisk from more than two years as my voice communication between my organization sites with no tears, it is running very smoothly.

After Exchange 2007 deployment I planned to use the Unified messages of MS Exchange with my existing Asterisks, initially it seems little clue less but after a small research I got some clues about it CAN work with Exchange and Asterisk Server.

Asterisk to allow SIP over TCP, go to PBX and config file editor.
Edit the sip_general_custom.conf file located in /etc/asterisk.

tcpenable=yes
tcpbindaddr=0.0.0.0

PBX Settings, Trunks, Add SIP Trunk. Enter the following details in

 “Outgoing Settings”:
Trunk Name: Exchange
PEER Details:
host=[IP Address of Exchange 2010 UM Server]
type=friend
insecure=very
transport=tcp
port=5065
context=from-internal

Associated outbound route.

Click “Outbound Routes” and add:
Route Name: Exchange
Intra Company Route: Checked
Dial Patterns:
6666
8800
8888
Trunk Sequence:
SIP/Exchange

For initial testing I used my existing two SIP extensions.

Extension can be configured simply in this way:

type: peer (without this Asterisk will not permit Exchange “play on phone”) Update, this field is not available until you add the extension and go back later and edit the details
Voicemail & Directory:
Status: “Enabled”
 

Configure Exchange 2010 UM:
Exchange Unified Messaging settings.
In the Exchange Management Console (EMC) go to Organization Configuration, Unified Messaging , New UM Dial Plan.

Name: UM Dial Plan
Number of digits in extension numbers: 4
URI type: Telephone Extension
VoIP security: Unsecured
Country/Region code: 92 [92 is for PK]

After creating this plan need to change some settings, go to properties, Subscriber access and add extension 8800. Then in the Settings tab change the Audio codec to G711.

Created a new UM IP Gateway:
Name: Trixbox
IP address: 10.0.0.10 (my Trixbox IP)
Dial plan: UM Dial Plan (this is the plan just created)
Please note upon submitting UM IP Gateway settings a Default Hunt Group will be automatically created – do not need to touch this.
Next a UM Mailbox Policy is created:
Name: Trixbox
Associated dial plan: UM Dial Plan (this is the plan just created)
 

Create the Auto Attendant.
Name: Trixbox AA
Associated dial plan: UM Dial Plan (this is the plan just created)
Pilot identifiers: 6666 click add, then 8888
Check both, create auto attendant as enabled and create auto attendant as speech-enabled.

Within the EMC go to Server Configuration, Unified Messaging, double click your server and go to the UM Settings tab. Add your Dial Plan “UM Dial Plan” and click ok.
 

Finally need to enable mailboxes for Unified Messaging.

Go to Recipient Configuration within the EMC and enable Unified Messaging for your intended mailbox. Browse to your Mailbox Policy “Trixbox” and enter the extension number – mine is 5348.

Now be able to dial your subscriber access number 8800 from the X-Lite client and get automatically forwarded to your Exchange voicemail box.

Likewise dial your Auto Attendant on either 6666 or 8888 you should be greeted by “Thank you for calling the Microsoft Exchange Auto Attendant” –
 

All you need to do to complete the integration by ensuring Trixbox routes unanswered calls to Exchange and not to its own voicemail system.

Head back into your Trixbox web GUI, PBX, Config File Editor, you need to edit the extensions.conf file located in /etc/asterisk. Specifically the section [macro-exten-vm].

You need to change:
exten => s,n,Macro(vm,${VMBOX},${DIALSTATUS},${IVR_RETVM})
to:
;exten => s,n,Macro(vm,${VMBOX},${DIALSTATUS},${IVR_RETVM})
exten => s,n,SIPAddHeader(Diversion: <tel:${EXTTOCALL}>\;reason=no-answer\;screen=no\;privacy=off)
exten => s,n,Dial(SIP/Exchange/8800)
exten => s,n,Hangup

This tells Trixbox to no longer route unanswered calls to its own voicemail but instead send them down the SIP trunk “Exchange” extension “8800” aka the subscriber access number.
 

Configuring the Inbound Route
Asterisk by default can't forward an incoming call to any arbitrary number. It must exist as a registered extension on the system. We want our calls coming in from the PSTN to be routed to the Exchange Server's extension, which Asterisk can't do on its own. However, there is a module we can install to do this for us. We will create Miscellaneous Destinations for both the AutoAttendant and Subscriber Access Number, and configure the inbound calls to be forwarded to one of those destinations.

Click Tools on the top menu of FreePBX, then on the left hand side, click Module Admin. Scroll down to the Inbound Call Control section, and click on Misc Destinations. Select Install as the action, and press the Process button at the bottom of the screen. When the module has installed, click Setup at the top of the FreePBX menu to return to the main configuration screen. Click the Misc Destinations option that has appeared on the left hand menu. Enter the following information for our destination.

Description: ExchangeAutoAttendant
Dial: 299

Click Submit Changes, and add a second destination
Description: ExchangeSubscriberAccess
Dial: 222

Click Submit Changes, and then Inbound Routes on the left hand menu. Enter the following information.

DID Number: blank
Leave Caller ID Number blank
Leave Zaptel channel blank
Leave the Fax Handling, Privacy, and Options sections at their defaults.
 

Under Set Destination, select Misc Destinations, and choose either ExchangeAutoAttendant or ExchangeSubscriberAccess, depending on where you want the incoming calls to go.

 

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