Configure the Mitel 3300 MCD for use with Microsoft Exchange 2007

Network Topology
This diagram shows how the testing network is configured for reference.

Configuration Notes
This section is a description of how the SIP Interop was configured. These notes should give a guideline how a device can be configured in a customer environment and how Microsoft Exchange 2007 3300 programming was configured in a test environment.

3300 ICP Configuration Notes
The following steps show how to program a 3300 ICP to interconnect with Microsoft Exchange 2007.

Network Requirements
• There must be adequate bandwidth to support the voice over IP. As a guide, the Ethernet bandwidth is approx 85 Kb/s per G.711 voice session and 29 Kb/s per G.729 voice session (assumes 20ms packetisation). As an example, for 20 simultaneous SIP sessions, the Ethernet bandwidth consumption will be approx 1.7 Mb/s for G.711 and 0.6Mb/s. Almost all Enterprise LAN networks can support this level of traffic without any special engineering. Please refer to the 3300 Engineering guidelines for further information. For high quality voice, the network connectivity must support a voice-quality grade of service (packet loss <1%, jitter < 30ms, one-way delay < 80ms).

Assumptions for the 3300 ICP Programming
• The SIP signalling connection uses UDP on Port 5060.

Licensing and Option Selection – SIP Licensing
Ensure that the 3300 ICP is equipped with enough SIP trunking licenses for the connection to Microsoft Exchange 2007. This can be verified within the License and Option Selection form. Enter the total number of licenses in the SIP Trunk Licences field. This is the maximum number of SIP trunk sessions that can be configured in the 3300 to be used with all service providers, applications and SIP trunking devices.

Figure 1 – License and Option Selection

Class of Service Assignment
The Class of Service Options Assignment form is used to create or edit a Class of Service and specify its options. Classes of Service, identified by Class of Service numbers, are referenced in the Trunk Service Assignment form for SIP trunks. Many different options may be required for your site deployment, but ensure that “Public Network Access via DPNSS” Class of Service Option is configured for all devices that make outgoing calls through the SIP trunks in the 3300.
• • •

Public Network Access via DPNSS set to Yes Campon Tone Security/FAX Machine set to Yes Busy Override Security set to Yes

Figure 2 – Class of Service

Network Element Assignment
Create a network element for Microsoft Exchange 2007. In this example, the Microsoft Exchange Server is reachable by FQDN and is defined as “Exchange” in the network element assignment form. Set the transport to TCP and port to 5060.

Figure 3 – Network Element Assignment

Trunk Service Assignment
This is configured in the Trunk Service Assignment form. In this example the Trunk Service Assignment is defined for Trunk Service Number 9 which will be used to direct incoming calls to an answer point in the 3300. Program the Non-dial In or Dial In Trunks (DID) according to the site requirements and what type of service was ordered from your service provider. The example below shows configuration for incoming DID calls. The 3300 will absorb zero digits of the DID number from Microsoft Exchange 2007 leaving 4 digits for the 3300 to translate and ring the remaining 4 digit extension. For example, Microsoft Exchange 2007 delivers 6513 through the SIP trunk to the 3300. The 3300 will ring extension 6513. Extension 6513 must be programmed as a valid dialable number in the 3300. Please refer to the 3300 System Administration documentation for further programming information.

Figure 4 – Trunk Service Assignment

SIP Peer Profile
The recommended connectivity via SIP Trunking does not require additional physical interfaces. IP/Ethernet connectivity is part of the base 3300 ICP Platform. The SIP Peer Profile should be configured with the following options: Network Element: The selected SIP Peer Profile needs to be associated with previously created “Exchange” Network Element. Address Type: Enter the Use IP Address in SIP messages. Trunk Service Assignment: Enter the trunk service assignment previously configured. SMDR: If Call Detail Records are required for SIP Trunking, the SMDR Tag should be configured (by default there is no SMDR and this field is left blank). Maximum Simultaneous Calls: This entry should be configured to maximum number of SIP trunks provided by Microsoft Exchange 2007. NOTE: Ensure the remaining SIP Peer profile policy options are similar the screen capture below.

Figure 5 – SIP Peer Profile Assignment

Digit Modification Number
Ensure that Digit Modification for outgoing calls on the SIP trunk to Microsoft Exchange 2007 absorbs or inject additional digits according to your dialling plan. In this example, we will be absorbing 0 digits (in this case will be 6595 to dial out).

Figure 6 – Digit Modification Assignment

Route Assignment
Create a route for SIP Trunks connecting a trunk to Microsoft Exchange 2007. In this example, the SIP trunk is assigned to Route Number 9. Choose SIP Trunk as a routing medium and choose the SIP Peer Profile and Digit Modification entry created earlier.

Figure 7 – SIP Trunk Route Assignment

ARS Digits Dialled Assignment
ARS initiates the routing of trunk calls when certain digits are dialled from a station. In this example, when a user dials 6595, the call will be routed to Microsoft Exchange 2007 (ie. Route 9).

Figure 8 – ARS Digit Dialed Assignment

Programming Call Forwarding Options at User Phones
Users’ phones can be configured to setup call forwarding always, no answer and busy situations to the Exchange UM Pilot Number. In addition, separate handling for internal and external calls can be configured. This configuration is performed at the set using the phone settings application. The settings application at the user’s phone allows programming of the forwarding destinations.
The call forwarding application (on the 5330/5340 phones) is used as follows: Program call forwarding To program call forwarding: 1. Launch Applications. 2. 3. 4. 5. 6. Press Call Forwarding. Press New Profile. Press Edit Profile Name (An on-screen keyboard is displayed). Press the appropriate keys in the on-screen keyboard to enter the profile name and press Save. This profile name identifies where your phone calls will be forwarded to. Select the check boxes opposite Call Forward categories as follows: • • • • • Always: forwards all your phone calls. Busy Internal: forward internal phone calls after several rings if your line is busy. Busy External: forward external phone calls after several rings if your line is busy. No Answer Int: forward internal phone calls and redirect your calls after several rings if you don't answer. No Answer Ext: forward external phone calls and redirects your calls after several rings if you don't answer.

7.

Note: You can select one or more settings, however, the Always setting takes priority over all other settings. For each of the Call Forward categories: • • • Press Edit Number to display the on-screen keyboard. In the on-screen keyboard, enter the appropriate number. 1300 in this case Press Save. The edit window closes. This profile is saved but it is not activated. To activate this profile, see Activate Call Forwarding below.

Activate call forwarding To turn Call Forward on once it has been programmed: 1. Press Applications. 2. Press Call Forwarding. 3. Press the appropriate Profile setting. 4. Press Activate. Cancel Call Forwarding To cancel Call Forwarding: 1. Press Applications. 2. Press Call Forwarding. 3. Press None setting. 4. Press Activate.

Microsoft Exchange 2007 Configuration Notes
Multiple Interfaces to Exchange UM Server(s)
The Mitel 3300 ICP supports a number of additional capabilities to support enhanced connectivity to Exchange UM Server: 1. Multiple interfaces to Exchange UM Server (s). The 3300 ICP Automatic Route Selection can be used to setup multiple routes facilitating failure scenarios. The pilot number for trunk group can be configured as a Route List with up to six individual Routes providing alternate paths for connectivity. The paths in the overall solution can be SIP Trunks connected to different 3300 controllers. 2. Load balancing is automatic – all of the routes will have traffic shared equally.

Resiliency Configuration
The 3300 ICP (acting as a PBX or a Gateway) can be configured to support multiple UM Servers in the failover scenario. Alternate Route Selection will automatically take place upon detection of loss of connectivity to the Exchange 2007 UM Server. Additionally, it is possible to configure multiple 3300 to support single or multiple UM Servers. In normal operation, traffic will be load-shared among the 3300(s). If one of them fails, traffic can be automatically rerouted over the other 3300(s). This capability is enabled using the 3300 ICP distributed network architecture.

Integration Testing and Confirmation
The following steps can be performed to ensure integration is complete and successful.

Dial Pilot Number and Mailbox Login
• • • • Dial the pilot number of the UM server from an extension that is NOT enabled for UM. Confirm hearing the greeting prompt: “Welcome, you are connected to Microsoft Exchange. To access your mailbox, enter your extension...” Enter the extension, followed by the mailbox PIN of an UM-enabled user. Confirm successful logon to the user’s mailbox.

Navigate Mailbox using Voice User Interface (VUI)
• • • • Logon to a user’s UM mailbox. If the user preference has been set to DTMF tones, activate the Voice User Interface (VUI) under personal options. Navigate through the mailbox and try out various voice commands to confirm that the VUI is working properly. This test confirms that the RTP is flowing in both directions and speech recognition is working properly.

Navigate Mailbox using Telephony User Interface (TUI)
• • • • Logon to a user’s UM mailbox. If the user preference has been set to voice, press “#0” to activate the Telephony User Interface (TUI). Navigate through the mailbox and try out the various key commands to confirm that the TUI is working properly. This test confirms that both the voice RTP and DTMF RTP (RFC 2833) are flowing in both directions.

Dial User Extension and Leave Voicemail
• Note: If you are having difficulty reaching the user’s UM voicemail, verify that the coverage path for the UM-enabled user’s phone is set to the pilot number of the UM server.

From an Internal Extension • • • From an internal extension, dial the extension for a UM-enabled user and leave a voicemail message. Confirm the voicemail message arrives in the called user’s inbox. Confirm this message displays a valid Active Directory name as the sender of this voicemail.

From an External Extension • • • From an external phone, dial the extension for a UM-enabled user and leave a voicemail message. Confirm the voicemail message arrives in the called user’s inbox. Confirm this message displays the phone number as the sender of this voicemail.

Dial Auto Attendant(AA)
• Create an Auto Attendant using the Exchange Management Console: • • • • • • • • Under the Exchange Management Console, expand “Organizational Configuration” and then click on “Unified Messaging”. Go to the Auto Attendant tab under the results pane. Click on the “New Auto Attendant…” under the action pane to invoke the AA wizard. Associate the AA with the appropriate dial plan and assign an extension for the AA. Create PBX dialing rules to always forward calls for the AA extension to the UM server. Confirm the AA extension is displayed in the diversion information of the SIP Invite.

Dial the extension of Auto Attendant. Confirm the AA answers the call.

Call Transfer by Directory Search
• Method one: Pilot Number Access • • • Dial the pilot number for the UM server from a phone that is NOT enabled for UM. To search for a user by name: Press # to be transferred to name Directory Search. • • Call Transfer by Directory Search by entering the name of a user in the same Dial Plan using the telephone keypad, last name first.

To search for a user by email alias: • • • Press “#” to be transferred to name Directory Search Press “# #” to be transferred to email alias Directory Search Call Transfer by Directory Search by entering the email alias of a user in the same Dial Plan using the telephone keypad, last name first.

Method two: Auto Attendant • • Follow the instructions in appendix section 5 to setup the AA. Call Transfer by Directory Search by speaking the name of a user in the same Dial Plan. If the AA is not speech enabled, type in the name using the telephone keypad.

Note: Even though some keys are associated with three or four numbers, for each letter, each key only needs to be pressed once regardless of the letter you want. Ignore spaces and symbols when spelling the name or email alias.

Called Party Answers • • Call Transfer by Directory Search to a user in the same dial plan and have the called party answer. Confirm the call is transferred successfully.

Called Party is Busy • • Call Transfer by Directory Search to a user in the same dial plan when the called party is busy. Confirm the calling user is routed to the correct voicemail.

Called Party does not Answer • • Call Transfer by Directory Search to a user in the same dial plan and have the called party not answer the call. Confirm the calling user is routed to the correct voicemail.

The Extension is Invalid • Assign an invalid extension to a user in the same dial plan. An invalid extension has the same number of digits as the user’s dial plan and has not been mapped on the PBX to any user or device. • • • • • UM Enable a user by invoking the “Enable-UMMailbox” wizard. Assign an unused extension to the user. Do not map the extension on the PBX to any user or device. Call Transfer by Directory Search to this user. Confirm the call fails and the caller is prompted with appropriate messages.

Play-On-Phone
• To access play-on-phone: • • • • Logon to Outlook Web Access (OWA) by going to URL https://<server name>/owa. After receiving a voicemail in the OWA inbox, open this voicemail message. At the top of this message, look for the Play-On-Phone field ( Play on Phone...). Click this field to access the Play-On-Phone feature.

To an Internal Extension • • • • Dial the extension for a UM-enabled user and leave a voicemail message. Logon to this called user’s mailbox in OWA. Once it is received in the user’s inbox, use OWA’s Play-On-Phone to dial an internal extension. Confirm the voicemail is delivered to the correct internal extension.

To an External Phone number • • • •

Dial the extension for a UM-enabled user and leave a voicemail message. Logon to the UM-enabled user’s mailbox in OWA. Confirm the voicemail is received in the user’s mailbox. Use OWA’s Play-On-Phone to dial an external phone number. Confirm the voicemail is delivered to the correct external phone number.

Troubleshooting: • Make sure the appropriate UMMailboxPolicy dialing rule is configured to make this call. As an example, open an Exchange Management Shell and type in the following commands: $dp = get-umdialplan -id <dial plan ID> $dp.ConfiguredInCountryOrRegionGroups.Clear() $dp.ConfiguredInCountryOrRegionGroups.Add("anywhere,*,*,") $dp.AllowedInCountryOrRegionGroups.Clear() $dp.AllowedInCountryOrRegionGroups.Add(“anywhere") $dp|set-umdialplan $mp = get-ummailboxpolicy -id <mailbox policy ID> $mp.AllowedInCountryGroups.Clear() $mp.AllowedInCountryGroups.Add("anywhere") $mp|set-ummailboxpolicy The user must be enabled for external dialing on the PBX. Depending on how the PBX is configured, you may need to prepend the trunk access code (e.g. 9) to the external phone number.

• • • • • • • • • • • •

Voicemail Button
• • • Configure a button on the phone of a UM-enabled user to route the user to the pilot number of the UM server. Press this voicemail button on the phone of an UM-enabled user. Confirm you are sent to the prompt: “Welcome, you are connected to Microsoft Exchange. <User Name>. Please enter your pin and press the pound key.” Note: If you are not hearing this prompt, verify that the button configured on the phone passes the user’s extension as the redirect number. This means that the user extension should appear in the diversion information of the SIP invite.

Message Waiting Indicator (MWI)
• Although Exchange 2007 UM does not natively support MWI, Geomant has created a 3rd party solution - MWI2007. This product also supports SMS message notification. Installation files and product documentation can be found on Geomant’s MWI 2007 website.

Test-UMConnectivity
• • • Run the Test-UMConnectivity diagnostic cmdlet by executing the following command in Exchange Management Shell: Test-UMConnectivity –UMIPGateway:<Gateway> -Phone:<Phone> |fl <Gateway> is the name (or IP address) of the gateway which is connected to UM, and through which you want to check the connectivity to the UM server. Make sure the gateway is configured to route calls to UM. <Phone> is a valid UM extension. First, try using the UM pilot number for the hunt-group linked to the gateway. Next, try using a CFNA number configured for the gateway. Please ensure that a user or an AA is present on the UM server with that number. The output shows the latency and reports if it was successful or there were any errors.

Test Fail-Over Configuration on IP-PBX with Two UM Servers
• This is only required for direct SIP integration with IP-PBX. If the IP-PBX supports fail-over configuration (e.g., round-robin calls between two or more UM servers): • • • • Provide the configuration steps in Section 5. Configure the IP-PBX to work with two UM servers. Simulate a failure in one UM server. Confirm the IP-PBX transfers new calls to the other UM server successfully.