Siptrunking with Microsoft Exchange Server 2007 UM

How-To-Guide

SIPTrunking with Microsoft Exchange Server 2007 UM

May 2007

About this document

This How-To-Guide is intended for Customer Service and Installation Personnel involved in the installation and maintenance of Mitel SIP Phones.

NOTICE

The information contained in this document is believed to be accurate in all respects but is not warranted by Mitel Networks Corporation. The information is subjected to change without notice and should not be construed in any way as a commitment by Mitel or any of its affiliates or subsidiaries. Mitel and its affiliates and subsidiaries assume no responsibility for any errors or omissions in this document. Revisions of this document or new editions of it may be issued to incorporate changes.

How-To-Guide for SIP Trunking with MS Exchange Server 2007 UM

May 2007

,Trademark of MITEL Networks Corporation

Table of Contents

Overview

Software & Hardware Setup

Technical Support

Limitations

3300 Setup Notes

Network Requirements

Assumptions for the 3300 ICP programming

Licensing and Option Selection - SIP Licensing

Class of Service Options Assignment

Network Element Assignment

SIP Peer Profile

Trunk Service Assignment

Digit Modification Assignment

Route Assignment

ARS Digits Dialed Assignment

Call Rerouting Assignment for Users’ Phones

Setting up Unified Messaging capabilities for user extensions

Programming Call Forwarding Options at User Phones

Multiple Interfaces to Exchange UM Server(s)

Resiliency Configuration

Integration Testing and Confirmation

Dial Pilot Number and Mailbox Login

Navigate Mailbox using Voice User Interface (VUI)

Navigate Mailbox using Telephony User Interface (TUI)

Dial User Extension and Leave Voicemail

Dial Auto Attendant(AA)

Call Transfer by Directory Search

Play-On-Phone

Voicemail Button

FAX

Message Waiting Indicator (MWI)

Test-UMConnectivity

Test Fail-Over Configuration on IP-PBX with Two UM Servers

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Overview

This document describes the configuration required to setup Mitel 3300 ICP with Exchange 2007 Unified Messaging using direct SIP connection. It also contains integration testing and confirmation testing procedures to ensure complete integration setup.

Software & Hardware Setup

This was the test setup to generate a basic SIP Trunking call between the Exchange 2007 Unified Messaging and the 3300 ICP using SIP Trunking.

Manufacturer / Variant / Software Version
Mitel / 3300 ICP – CXi, CX / MXe platforms / 7.1 UR2
Sets / Tested with 5212, 5224, 5340
Manufacturer / Variant / Software Version
Microsoft / Microsoft Exchange Server 2007 UM / RTM (685.25)

Technical Support

The information contained within this document has been provided AS IS. This document contains information about how to modify theconfiguration of your PBX or VoIP gateway. Improper configuration may result in the loss of service ofthe PBX or gateway. Microsoft is unable to provide support or assistance with the configuration ortroubleshooting of components described within. Microsoft recommends readers to engage the serviceof an Microsoft Exchange 2007 Unified Messaging Specialist to assist with the planning and deployment of Exchange Unified Messaging.

Microsoft Exchange 2007 Unified Messaging (UM) Specialists

These are Systems Integrators who have attended technical training on Exchange 2007 UnifiedMessaging conducted by Microsoft Exchange Engineering Team.

Limitations

The following limitations are related to the integration of 3300 ICP with Exchange 2007 UM server:

When using the message playback via Play to Phone feature, calls placed from Exchange 2007 can only be directed to a subset of Mitel devices. These include all dual-mode sets (5215, 5220, 5212, 5224) Navigator, 5235, 5330, 5340.
TLS is not supported by the 3300 ICP R7.1 UR2.
T.38 FAX is not directly supported by the 3300 ICP.
G.723.1 codec is not supported by the 3300 ICP.

3300 Setup Notes

The following steps show how to program a 3300 ICP to interconnect with Microsoft Exchange Server 2007 Unified Messaging using SIP trunking.

Network Requirements

There should be no Firewall or NAT devices between the Exchange UM Server and the 3300 ICP. If present, the Firewall / NAT must be capable of SIP traversal.
There must be adequate bandwidth to support the voice over IP connections between the 3300 ICP and the Exchange UM Server. 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 30ms packetization). As an example, for 20 simultaneous UM 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 Exchange UM Server has an FQDN of um.mitel.com and is reachable on an IP network from the 3300 ICP.
The pilot number for the Exchange UM Server is defined to be 1300 (the pilot number is fully configurable). This is the number that users dial to reach Unified Messaging and is the number that Unanswered and Busy Calls are forwarded to for Unified Messaging treatment.
The SIP signaling connection uses TCP/IP on Port 5060.

Licensing and Option Selection - SIP Licensing

Ensure that the 3300 ICP is equipped with enough SIP trunking licences for the connection to the Exchange 2007 Unified Messaging Server. 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.

Figure 1: Example of SIP Licensing

Class of ServiceOptions Assignment

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.

Figure 2: Example of Class of Service Options Assignment

Network Element Assignment

Create a network element for the Exchange UM Server. In this example, the Exchange Server is reachable at FQDN = um.mitel.com and is defined as “Exchange” in the network element assignment form.

Use the same FQDN for the SIP Peer, Outbound Proxy and Registrar. Set the transport to TCP and port to 5060.

Figure 3: Example of Network Element 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.
SMDR: If Call Detail Records are required for Exchange UM Server, the SMDR Tag should be configured (by default there is no SMDR and this field is left blank).
Fake Answer – Send SDP in Initial invite option must be set to “Yes”.
Session Timer: Session Timer should be disabled by entering zero

Figure 4: Example of SIP Peer Profile

Trunk Service Assignment

The Trunk Service Assignment for the SIP trunk used to connect to the Exchange UM Server must be configured to ensure:

No dialed digits are inserted
No dialed digits are absorbed.

This is configured in the Trunk Service Assignment form. In this example the Trunk Service Assignment is defined for Trunk Service Number 1.

Ensure that the Incoming Digit Modification – Absorb is set to absorb 0 digits and that no additional digits are inserted.

Figure 5: Example of Trunk Service Assignment

Digit Modification Assignment

Ensure that Digit Modification for outgoing calls on the SIP trunk to the Exchange UM Server does not absorb or inject additional digits.

Figure 6: Ensure UM Server does not absorb or inject additional digits

Route Assignment

Create a route for SIP Trunks connecting to the Exchange UM Server. In this example, the SIP trunk to the Exchange UM Server is assigned to Route Number 3. Choose SIP Trunk as a routing medium and choose the SIP Peer Profile and Digit Modification entry created earlier.

Figure 7: Example SIP Trunk Route Creation

ARS Digits Dialed Assignment

The pilot number for the Trunk Group is created in the ARS Digits Dialled Assignment Form. The digits point to the Route (or Route List) which identifies the SIP Trunk used to connect to the Exchange UM Server. In this case, the dialled digits 1300 route the call to Route Number 3, the SIP Trunk we have defined for connection to um.mitel.com, the Exchange UM Server.

Figure 8: Create Trunk Group Pilot Number for UM Server

Call Rerouting Assignment for Users’ Phones

There are two general approaches in managing the coverage for user’s phones. Call Rerouting is primarily intended for centralized setup of the coverage, where users are grouped together and their calls are rerouted to the preconfigured destinations based on time (day/night service), condition (e.g. busy, no answer) and call type (e.g. DID, ,CO, internal). Call Forwarding is typically used directly by users and controlled by them via the phone). Call Forwarding takes precedence over the Call Rerouting.

In Call Rerouting Assignment form each extension is associated with Rerouting index. This index is used to identify specific call coverage handling.

Figure 9: Configure Call Rerouting Assignment for extensions

Call Rerouting Day / Night1 / Night 2 reflect behavior related to immediate call rerouting. In this example calls are not rerouted immediately, and as a result will follow the Call Rerouting First Alternative.

Figure 10: Configure Call Rerouting Always Alternative

When Call Rerouting Always is not in effect the Busy/DND and No Answer conditions will define call coverage handling. In this example calls are rerouted to the pilot number for UM Server.

Figure 11: Configure Call Rerouting First Alternative

Setting up Unified Messaging capabilities for user extensions

Using the Group Administration tool in EMS, user extensions can be set up for default call forwarding to the Exchange UM Pilot Number. This will forward calls on Busy or No Answer conditions to the Unified Messaging Pilot Number

Figure 12: Setting up UM Pilot Number for User Extensions

In this example, User Joan Smith at extension 4321 has her Voice mail / Unified Messaging Number set to 1300, the Pilot Number for the Exchange UM.

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:

Launch Applications.
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.

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:

Press Applications.
Press Call Forwarding.
Press the appropriate Profile setting.
Press Activate.

Cancel Call Forwarding

To cancel Call Forwarding:

Press Applications.
Press Call Forwarding.
Press None setting.
Press Activate.

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:

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.
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.

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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.

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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.

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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.

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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.

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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.

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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.

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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.

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