Understanding Unified Messaging in Exchange Server 2013
What Is a PBX
A PBX usually is the core device or system that provides telephony and telephone features for residential or organizational use, such as for homes with multiple phones or businesses with thousands of phones. Every PBX connects externally to the public voice network.
Types of PBX:
Analog PBX: Classic voice transmission with analog wave forms
Digital PBX: Modern PBX handling voice in digital form
IP PBX: Digital PBX, with voice transmission via TCP/IP only. IP PBXs carry voice over data networks. The IP phone contains a Network Interface Card (NIC), and it is part of the network. The phone converts voice into digitized packets, which it then places on the data network. The network sends the voice packets through packet switching, a technique that enables a single network channel to handle multiple calls.
The IP PBX also acts as a gateway between the internal packet-switched network and the external circuit switched networks that telephone companies use. In this situation, external phone calls arrive at the IP PBX on the normal public phone lines, and the IP PBX converts the phone call to packets sent on the internal IP-based network.
Hybrid PBX: A combination of classic analog and digital voice
Phone extension: used by organizations.
Direct inward dialing: A Direct Inward Dialing (DID) phone number is a unique number that an organization assigns to a person. A user with a DID number can receive calls directly from an external phone without having to transfer the call. The DID is a combination of a company-specific phone number and the user’s extension. If the organization implements a PBX, the PBX maps DID numbers to internal extensions, so that it can route calls to the correct phone
Direct Inward Dialing
A Direct Inward Dialing (DID) phone number is a unique number that an organization assigns to a person.
A user with a DID number can receive calls directly from an external phone without having to transfer the call. The DID is a combination of a company-specific phone number and the user’s extension. If the
organization implements a PBX, the PBX maps DID numbers to internal extensions, so that it can route calls to the correct phone.
A dial plan consists of the rules that a PBX uses to determine what action to take when it receives a set of dialed numbers. For example, a “9” often triggers call setup to an outside line, so that users can call external phone numbers. When “9” is not the first number, the PBX needs to know how many numbers to collect before taking action. If internal extension numbers are four digits long, it waits for just four numbers before taking action.
A hunt group is a collection of extensions. In most cases, a hunt group represents a set of identical resources that an application or a group shares. This grouping provides more-efficient access to applications, such as voice mail, an auto attendant, or even a call center. This ensures that callers do not experience a busy signal. Instead, the PBX hunts for an open line to which to connect them.
A pilot number is the address or label that the PBX uses to identify a hunt group. It is an unused extension, meaning it is not associated with a person or phone.
For example, there may be a specific extension number 3900 for the telesales team, which may be the pilot number for the hunt group of telesales-extension numbers. When a call comes into the 3900 sales
number, the PBX recognizes it as a pilot number, and searches for an available line within the sales hunt group. The PBX then delivers the call to an available sales-extension number.
A PBX uses a set of directions that you configure for each extension, and it tells the PBX where to route unanswered calls and calls that receive busy signals. The set of directions is a coverage path. If a DID call arrives at the Unified Messaging server through a user’s desktop phone, and the line is busy or not answered within a certain number of rings, the PBX knows to send the call to the pilot number for the hunt group that attaches to the VoIP gateway. The PBX routes the call through the VoIP gateway to the
Unified Messaging server, where the caller can record a voice message. The Unified Messaging server sends the voice message to the Unified Messaging user’s mailbox.
Switched and Packet-Switched Networks
Telephony systems and computer systems usually use different networks to communicate. A telephony system typically uses a circuit-switching network, while a computer system uses a packetswitching network.
A circuit-switched network uses a dedicated connection between two network devices. For example, you pick up the telephone receiver and dial a phone number. By answering the call, the recipient completes the circuit. After the two nodes establish a call between them, only these two nodes can use the connection. When one of the nodes ends the call, the connection is removed.
Packet switching is a technique that divides a data message into smaller units, or packets. The network sends the packets to their destination by the best route available, and then reassembles them at the receiving end.
What Is VoIP
VoIP is a technology that enables an IP-based network to act as the transmission medium for telephone calls. It sends voice data in IP packets rather than by circuit-switched telephone lines.
There are a number of voice-related, IP-based protocols, and a Unified Messaging environment with
Exchange Server 2013 uses the following:
Session Initiation Protocol (SIP): SIP is a real-time signaling protocol that creates, manipulates, and
tears down interactive communication sessions on an IP network. You can use SIP in conjunction with
Transport Layer Security (TLS) to provide security. Exchange Server Unified Messaging uses SIP mapped over Transmission Control Protocol (TCP), and supports TLS for secured SIP environments. SIP clients, such as IP/VoIP gateways and IP/PBXs, can use TCP port 5060 or port 5061 (for Secure SIP) to connect to SIP servers.
Real-Time Transport Protocol (RTP): RTP is for voice transport between the IP gateway and the Unified
Messaging server. RTP provides high-quality, real-time, streaming voice delivery. One of the issues with sending voice messages over an IP network is that voice requires real-time transport, with specific quality requirements, to ensure that the voice sounds normal. If the protocol uses large packets, listeners must wait for the entire packet to arrive before they can respond. Any delay in packet delivery can produce undesirable periods of midstream silence, and packet loss can cause voice garbling.
What Is a VoIP Gateway
A VoIP gateway is a hardware device or product that converts traditional phone-system or circuitswitching protocols into data-networking or packet-switched protocols. Exchange 2013 servers that are running the Unified Messaging components can connect only to packet-switched data networks. This requirement means that organizations with a traditional PBX must deploy a VoIP gateway to communicate between the PBX and the Exchange 2013 servers. The PBX also provides access to the PSTN for internal phone users and for unified Messaging.
Options for Implementing VoIP Gateways
- Deploying Unified Messaging with an analog or digital PBX.
- Deploying Unified Messaging with an IP or hybrid PBX.
Deploying Unified Messaging with Microsoft Lync® Server: A Lync Server also can operate as a VoIP
gateway for the Exchange 2013 servers that are running Unified Messaging. Like Exchange servers,
Lync servers can communicate only on packet-switched networks, so no other VoIP gateways are
necessary for the Exchange Server.
Unified Messaging Features
Access to voice mail in user mailboxes. UMenabled users can access their voice mail from mobile phones, clients, and through Outlook Web App.
Play on Phone: UM-enabled users also can play their messages by using any normal phone to dial into Exchange 2013 or by using Microsoft Lync 2013. This arrangement also prevents others from listening to confidential voice mails if the computer only has external speakers.
Call answering: This feature supports playing personal greetings, recording messages, and answering incoming calls on behalf of other users.
Call Answering Rules: UM-enabled users can organize how the phone system handles their incoming calls. This feature is similar to Inbox rules, which users can apply to normal email messages. No call answering rules are activated by default.
Outlook Voice Access: UM-enabled users have two options for Outlook Voice Access: the Telephone
User Interface (TUI) and the Voice User Interface (VUI). This feature facilitates internal and external
access by using phone systems, and enable users to :
- Access voice mail.
- Listen to, forward, or reply to email messages.
- Listen to calendar information.
- Access or dial contacts who are stored in the global address list (GAL) or a group in their Contacts folder.
- Accept or cancel meeting requests.
- Set a voice-mail message to let callers know the called party is away.
- Set user-security preferences and personal options.
Voice Mail Preview: In Exchange 2013, the Unified Messaging feature uses Automatic Speech
Recognition (ASR) on new voice-mail messages. When users receive voice messages, the messages
contain both a recording and voice-mail preview text, which the system creates from the voice recording.
Message Waiting Indicator: The Message Waiting Indicator is any mechanism that indicates the existence of new Unified Messaging messages.
Missed call and voice-mail notifications by using SMS: If users are members of a hosted or consumer dial plan, and they configure their voice-mail settings, including their mobile phone number, with call
forwarding, they can receive notification about missed calls and newly arrived voice mail on their cell phones as short message service (SMS) text messages.
Protected Voice Mail: This extended feature is provided in conjunction with Active Directory® Rights
Management Service (AD RMS), and it enables the secure storage of voice-mail messages. This restricts the forwarding, copying, or extracting of voice file from email.
Voice mail form: The Outlook 2010, Outlook 2013, and Outlook Web App form for voice mail resembles the default email form.
IP-PBX: If your organization is using an IP-PBX, you must configure the IP-PBX to enable call routing to the Exchange 2013 servers.
Unified Messaging Mailbox Policies
UM mailbox policies enable you to configure the user experience or security settings to UMenabled mailboxes.
When you create a Unified Messaging mailbox policy, you can configure a wide variety of settings, including the following:
- Dial plan (required)
- Maximum greeting length
- Number of unsuccessful login attempts before it resets the password
- Minimum number of digits that a PIN requires
- Number of days until users must create a new PIN
- Number of previous passwords that it does not allow
- Restrictions on in-country/region or international calling
- Protected voice-mail settings
Note: Each Unified Messaging-enabled user’s mailbox must link only to one Unified Messaging mailbox policy.
- Unified Messaging Clients
- Microsoft Outlook client.
- Outlook Web App.
- Mobile devices and Microsoft Exchange ActiveSync clients.
- Outlook Voice Access.
Unified Messaging Auto Attendants
A Unified Messaging auto attendant is an optional component in a Unified Messaging deployment. It creates a voice-menu system that enables external and internal callers to navigate through voice menus to locate and place, or transfer, calls to company users or organizational departments.
Business Requirements for Unified Messaging
- Consolidated Access to Voice Mail and Email
- Voice Mail Protection
- Auto Attendant Service
- Reduction of Administrative Overhead
- Availability Requirements
Designing Infrastructure Requirements for Unified Messaging
The Mailbox server role provides most of the Unified Messaging services, including call answer, voice-mail recording, and auto-attendant services. When planning the Mailbox server role for Unified Messaging, ensure that you have 500 megabytes (MB) of additional disk space per Unified Messaging language pack on the operating system drive and approximately 250 kilobytes (KB) per voice message stored in the user’s mailbox.
The Mailbox server role also is responsible for transcribing voice mail messages if you enable the Voice Mail Preview feature. The capability for voice mail speech recognition that this processor requires is processor intensive. Therefore, we recommend at least 12 central processing unit (CPU) cores on the Mailbox server for an average installation of 1,000 users, and a minimum of 8 gigabytes (GB) RAM.
Client Access Server
The Client Access server role accepts Unified Messaging connections from different sources, such as IPPBX, IP Gateway, or Lync Server 2013. The incoming Session Initiation Protocol (SIP) traffic is redirected to the user’s associated Mailbox server. You then configure a SIP or Real-Time Protocol (RTP) connection between the Mailbox server and the call source, without any additional involvement from the Client Access server. Because the Client Access server only accepts and redirects the SIP connections, implementing Unified Messaging will not change the hardware requirements significantly for the Client Access server.
Active Directory Domain Services: also required as we know
PBX: Exchange Server 2013 Unified Messaging does not provide a telephony system, so you still must deploy some type of telephone system in the organization.
VoIP Gateway: If the PBX does not support IP networking, you will need to deploy a VoIP gateway between the Exchange 2013 servers and the PBX.
VoIP Phone: Organizations that have deployed a VoIP telephone system also have deployed VoIP phones. There are two types of VoIP phones available: software-based and hardware-based. A software-based phone, such as the Microsoft Lync system, is a communications program that runs from a computer. A hardware-based phone is similar to the phones found currently on desktops, except that they have added functionality.
The Lync Phone Edition is one such phone, but there are many other phones available.
Call Number Block Switching by Telephony Providers
In several countries, telephony providers offer a highly available phone line, known as line failover. In this scenario, each provided phone line always has a unique phone-number block assigned to it, which makes
it possible by using the second, redundant line as an individual PSTN connection. In case of a failure or outage of the primary phone line, the telephony provider will switch the primary call-number block to the second PSTN line. This requires a special design, and you need to consider connecting theses phone lines to the same gateways or to two different gateway.
Hard Drive Requirements for Storage
To install Exchange Server 2013 Unified Messaging, you need an additional 500 MB for each Unified Messaging language pack that you install. Another consideration is the duration of recorded messages in Unified Messaging. By default, there is a 20-minute maximum. You can modify this setting to between 5 and 100 minutes. Using the Windows® Media Audio (WMA) codec, a five-minute voice mail is approximately 250 KB in size.
High Availability for Unified Messaging
Exchange 2013: Implementing redundancy for the Unified Messaging components in Exchange Server 2013 is straightforward. You only need to deploy multiple Client Access servers and Mailbox servers, and then use the normal Exchange Server 2013 options to ensure high availability. It is helpful to ensure that all Exchange 2013 servers within the same location and dial plan have the same configuration.
Designing for Office 365 Unified Messaging
You can integrate Exchange Server 2013 Unified Messaging with Microsoft Office 365™ by deploying a hybrid solution. In this deployment, the organization has a PBX or VoIP phone system deployed on-premises, but the Exchange 2013 Client Access server and Mailbox server are located on Office 365. Since the Unified messaging services are located in a Microsoft data center, the VoIP traffic must cross the public Internet to reach the Exchange servers. To implement this, you must place a Session Border Controller (SBC) and a Microsoft Lync Edge Server at the edge of your internal network. All traffic from the VoIP Gateway and IP PBX to and from Office 365 passes through the SBC. All traffic from the Lync passes through the Lync Edge Server.
The purpose of the SBC is to protect the customer’s private network against attack and intrusion. It is for use at a network’s edge, and controls the flow of VoIP traffic to and from the private network to the public network (Internet). The SBC rewrites addressing information in headers when SIP messages pass from one network interface to the other. It secures the signaling and media data between itself and Office 365.
Designing for Unified Messaging Security
You can configure the VoIP security mode either when you are creating a new dial plan or after you have created a dial plan, by using the Exchange Administration Center (EAC) or Exchange Management Shell.
You have three options when configuring the VoIP security mode:
SIP secured: The SIP Secured setting means that only SIP traffic is encrypted by using TLS while RTP traffic is transmitted over TCP.
Secured: The Secured traffic means that both SIP traffic and streaming media sent by RTP traffic are encrypted by using TLS. If you are using a Lync Server as the VoIP gateway, this is the option that you must select.
Unsecured: All traffic is sent unencrypted. This is the default selection when you create a dial plan in Exchange Server 2013.
Note: When you configure the Unified Messaging dial plan to use SIP secured or Secured mode, Client Access and Mailbox servers will try to encrypt the SIP signaling traffic or the RTP media channels, or both. However, to send encrypted data to and from Client Access and Mailbox servers, you must configure the Unified Messaging dial plan correctly, and VoIP devices, such as VoIP gateways, IP PBXs, and SBCs, must support mutual TLS.
If you want to use mutual TLS to encrypt the VoIP traffic, you must have a certificate installed on the Client Access and Mailbox servers, and the other VoIP devices must trust the certificate. If you deploy an internal certification authority (CA) in the organization, you can use certificates from this CA if you can configure the VoIP devices to trust it. For example, if you are using Lync Server 2013 as the VoIP gateway, you should obtain certificates from the internal CA for both the Exchange 2013 servers and for the Lync 2013 servers. You also must configure the certificate for use by the Unified Messaging service on Mailbox servers and by Unified Messaging Call Router service on Client Access servers.
Consideration of Codecs and File Formats
A codec is a software program that transforms or codes: digital data into an audio file format or audio streaming format. It then converts, or decodes, the audio file back to the digital format. Codecs can vary in sound quality, the amount of bandwidth required to use them, and the system requirements necessary to do the encoding. Exchange Server 2013 Unified Messaging uses codecs for encoding media streams between the IP/VoIP gateways and the Exchange servers, and for encoding and storing voice messages.
Choosing a Codec for Encoding Media Streams
Exchange Server 2013 can use the G.711 (Pulse Code Modulation A-law (PCMA), which is used in Europe and other countries, and Pulse Code Modulation μ-law (PCMU), which is used in North America and Japan) and the G.723.1 codecs to encode media streams. By default, the Exchange Server 2013 servers use the G.723.1 codec. This codec is widely supported on VoIP gateways.
Note: If you use a Lync Server as the VoIP gateway, you have an additional option for providing higher quality voice recordings. When you configure a dial plan with a Lync Server as the Unified Messaging IP gateway, you have to configure the dial plan as a SIP uniform resource indicator (URI) dial plan. When you do this, the exchange servers will use RTAudio wideband or high-fidelity audio for recording voice messages.
RTAudio provides a higher sampling rate, so the quality of the voice recording will be better. When the RTAudio codec is used, the voice message will be recorded in high fidelity and stored as an audio file that has a .wma extension. When the voice message is played back to the user in Office Outlook or Outlook Web access, they will hear the voice message in high-fidelity audio. If users connect to their mailboxes by phone, the outbound media stream will be negotiated by using either the G.711 or G.723.1 codec. This means that callers will always hear lower fidelity audio over the telephone.
Choosing a Codec for Encoding Voice Messages
Exchange Server 2013 supports four codecs for encoding voice messages:
- MP3. This is the default format.
- Group System Mobile (GSM) 06.10
- G.711 PCM Linear
To choose the right codec for encoding voice messages, you need to consider the types of clients that will be used to access the voice messages, the storage requirements for each codec and the network bandwidth available for replaying voice messages. The codec options provide the following benefits:
MP3: The MP3 codec stores files in the .mp3 format, which means that it is compatible with the broadest range of mobile phones and devices and different computer operating systems. MP3 also provides very good compression of voice messages. A 30-second message recorded in an RTAudio codec will use about 120 KBs of storage, while a 30-second message recorded from a call using the G.723.1 codec will use about 60 KBs of storage.
WMA: WMA provides the highest level of compression of any of the codecs. Since the .wma file format has a much larger header section than the .wav file format, the file size difference is most noticeable for messages longer than 15 seconds in length. A 30-second message recorded in an RTAudio codec will use about 70 KBs of storage, while a 30-second message recorded from a call using the G.723.1 codec will use about 40 KBs of storage. Therefore, for the smallest, but highest quality, audio files, use the WMA audio codec.
G.711 PCM Linear: The G.711 PCM Linear audio codec creates uncompressed .wav audio files. Therefore, the voice-message recordings will require the most storage space. A 30-second message will consume about 240 KBs of storage. Because the files are not compressed, G.711 PCM Linear .wav audio files have the highest audio quality of the audio codecs that Unified Messaging uses. In most cases, the codecs that provide compression also provide acceptable sound quality, so we do not recommend the use of the G.711 PCM Linear audio codec in most cases.
GSM: The GSM audio codec creates .wav audio files that are compressed. A 30-second message will consume about 50 KBs, which is slightly larger than the audio file that the WMA audio codec creates.
Managing Codec and Voice Recording Settings
To configure the codec and voice recording settings, you will use the Set-UMDialPlan cmdlet. The following parameters apply the codec settings:
AudioCodec. Used to set the codec used in Exchange Server 2013 to record voice messages. The default is MP3.
MaxRecordingDuration. Used to set the maximum length of time that messages can be recorded.
The default is 20 minutes, but you can change the value to a number from 1 through 100. You may need to modify this number to balance storage requirements with the time necessary to leave meaningful messages.
Configure a UM IP Gateway Object
In Unified Messaging, the UM IP gateway object defines the connection point between the Exchange 2013 servers that are running the Unified Messaging services and the telephone network. Exchange Server 2013 uses the UM IP gateway object to accept calls from the telephone network and to route calls to the telephone network. The UM IP gateway object references a physical VoIP gateway, IP-PBX, SPC or a Lync Server. A UM IP gateway has organization-wide scope, and each UM IP gateway can reference only a single physical IP gateway. When you configure the UM IP gateway object in Exchange 2013, you must configure the target IP address and an object name.
The UM IP gateway contains one or more UM hunt groups and configuration settings. UM hunt groups link a UM IP gateway to a UM dial plan. By creating multiple UM hunt groups, you can associate a single UM IP gateway with multiple UM dial plans.
Configuring UM Dial Plan Objects
A dial plan object is a container object in AD DS that represents a set or grouping of PBXs logically that share common user-extension numbers. In practical terms, users’ extensions hosted on PBXs share a common extension number. Users can dial one another’s telephone extensions without appending a special number to the extension or dialing a full telephone number. A UM dial plan is a logical representation of a telephony dial plan. All users within a dial plan have a unique extension number, and the combination of dial plan and the user extension uniquely identifies each UM-enabled user. When you create the UM dial plan, you need to associate it with a Unified Messaging server.
In Unified Messaging, the following UM dial-plan topologies can exist:
A single dial plan that represents a subset of extensions or all extensions for an organization with one PBX or IP PBX. Use this configuration in small customer environments.
A single dial plan that represents a subset of extensions or all extensions for an organization with multiple PBXs or IP PBXs. Use this option in organizations that have deployed multiple PBXs, but a single set of extensions. Multiple dial plans that represent a subset of extensions or all extensions for an organization with one PBX or IP PBX. Use this in complex PBX environments for larger organizations.
Multiple dial plans that represent a subset of extensions or all extensions for an organization with multiple PBXs or IP PBXs. Use this topology is your organization has many geographically disparate locations.
A dial plan can exist in three different configurations:
Telephone Extension: This is the most common type of UM dial plan, and you use it with PBXs and IP gateways that support the telephone extension (TelExtn) URI type.
SIP URI: This is the dial plan that you use when integrating Exchange Server 2013 and Lync Server The SIP URI resembles an email address, and is sip:<user name>@<domain or IP address>
E.164: E.164 is the standard numbering format that you use for the international publictelecommunication numbering plan on the PSTN and some data networks. E.164 numbers can have a maximum of 15 digits, and typically are written with a plus sign before the telephone number. Use anE.164 dial plan type when the IP-PBX or VoIP gateway only support this type.
Configuring Hunt-Group Objects
A hunt group is an extension that is defined as a group of telephone numbers that are treated as one in some situations. Hunt groups often are used to identify a group of telephone extensions, such as help-desk or call-center personnel. When users call the phone number associated with the hunt group, the call is forwarded to any extension available in the hunt group. In most cases, a hunt group represents a set of identical resources that an application or a group shares. This provides more efficient access to applications such as voice mail or an auto attendant, so callers will not experience a busy signal.
A pilot number is the way in which the PBX identifies a hunt group. In other words, a pilot number is the address or label for the hunt group. It is a dummy extension, and does not have a person or phone associated with it. It is the number to which a coverage path routes a call.
Implementing UM Hunt Groups
The UM hunt group object is a logical representation of an existing PBX or Lync hunt group. The UM hunt group object locates the PBX or IP PBX hunt group from which an incoming call is received. The pilot number that you define for a hunt group in the PBX or IP PBX must be the same as the pilot number assigned to the UM hunt group. The pilot number matches the information about the incoming calls through the SIP signaling message information on the message. The pilot number enables the Client Access server to interpret the call together with the correct dial plan, so that the call can be routed correctly.
Implementing UM Auto Attendants
In telephony or Unified Messaging environments, an auto attendant is used to answer telephone calls and help callers search the internal phone system for an intended recipient, or it transfers callers to the extension of a u user or department without a receptionist or operator having to intervene. The auto attendant can provide a simple service, such as enabling callers to search for and connect to simple extensions, or complex services, such as speech recognition and a vast menu system that enables callers to sort and search the internal telephone system.
Exchange Server 2013 Unified Messaging enables you to create one or more UM auto attendants. An auto attendant provides the menu system that lets internal and external users navigate through configured options and place calls to desired recipients. You can present announcements through a .wav file or speech-to-text, so that the caller can navigate through the menu options quickly and easily, enabling them to locate and call the person with whom the user wants to speak.. For navigation, the caller can use dual tone multi-frequency (DTMF) or voice inputs.
Multiple Language Support with UM Auto Attendants
Many organizations need the capability to provide services in multiple languages. In these scenarios, you can configure the UM auto attendants to support more than one language. Each UM auto attendant is configured with a default prompt language. This setting defines the default language that the caller will hear when the auto attendant answers the incoming call.
Configuring Protected Voice Mail
To enable Protected Voice Mail, you need to:
Configure Unified Messaging by configuring the UM dial plan and UM IP gateway:
Configure UM Mailbox Policy that requires Protected Voice Mail. When configuring your UM Mailbox Policy to require Protected Voice Mail, configure the following parameters:
ProtectAuthenticatedVoiceMail. This parameter specifies whether the Exchange 2013 servers create protected voice mail messages for UM-enabled users. If the value is set to Private, only messages marked as private are protected. If the value is set to All, every voice mail message is protected. The default is none, which means that no protection is applied to voice-mail messages.
ProtectUnAuthenticatedVoiceMail. This parameter is the same as the previous parameter, but also applies to scenarios where automated messages are sent from the Unified Messaging system to the user mailbox.
RequireProtectedPlayOnPhone. This parameter specifies whether users can utilize Play on Phone to listen to protected voice-mail messages or whether they can use multimedia software to play protected messages. The default is $false, which means that users can use both means to listen to protected messages.
- Install and configure AD RMS and configure the integration of AD RMS and Exchange Server 2013.
You will use the Set-IRMConfiguration cmdlet to configure the integration.
Lync Server integration with UM
What Is Lync Server 2013
Exchange Server 2013 and Lync Server 2013 are designed to integrate and work together to provide a complete email and voice system. Exchange 2013 provides an email-messaging system, while Lync 2013 provides a telephony system when you configure it for Enterprise Voice. Unified Messaging can use Lync 2013 to provide the telephony component it needs, while Lync 2013 can use Unified Messaging to provide voicemail functionality. When you configure the integration of Exchange Server 2013 and Lync Server 2013, Exchange Server 2013 will use the Lync Server as its IP PBX. On Exchange Server 2013, you will configure an IP Gateway that references the Lync 2013 server.
Lync 2013 also provides other features that integrate with Unified Messaging, such as instant messaging, presence information, Web conferencing, and VoIP telephony:
Instant messaging: The Lync 2013 client provides instant messaging (IM) functionality that the Lync hosts. The solution provides IM features, such as group IM, and extends the internal IM infrastructure to external IM providers.
Presence information: Lync 2013 tracks presence information for all Lync users, and it provides this information to the Lync 2013 client and other applications, such as Outlook 2013.
Web conferencing: Lync 2013 can host on-premise conferences, which you can schedule or reschedule, and they can include IM, audio, video, application sharing, slide presentations, and other forms of data collaboration.
Audio conferencing: Users can join Lync 2013-based audio conferences by using any desk or mobile phone. When connecting to an audio conference by using a Web browser, users can provide a telephone number that the audio-conferencing services calls.
Integration with Office applications: When you implement Lync Server 2013, Exchange Server 2013, Microsoft SharePoint Server® 2013, and Microsoft Office 2013, you can provide a seamless user experience between all of the applications. For example, if you receive an email from another user, you can see the user presence information when you read the email. When a user sets an out-ofoffice response in Outlook, you will see that same response in your Lync client when viewing the user’s presence information.
Unified Contact Store: The Unified Contact Store feature enables users to store all contact information in there Exchange Server 2013 mailbox, so that the contact list is available in Lync, Outlook, and Outlook Web Access. The Unified Contact Store is enabled by default in Lync.
VoIP telephony: Enterprise Voice enables Lync 2013 users to place calls from their computers by clicking an Outlook or Lync contact. Users receive calls simultaneously on all of their registered user endpoints, which may be a VoIP phone, mobile phone, or Lync 2013 client. The Lync 2013 Attendant is an integrated call-management client application that enables a user, such as a receptionist, to manage many conversations simultaneously.
Support for remote users: Lync Server 2013 has an Edge Server role that enables remote users to use all Lync Server features without a virtual private network (VPN) connection.
Support for federation: You can configure federation with other organizations that are running Lync Server or Microsoft Office Communications Server, and provide full Lync functionality for users
between the two organizations.
With Lync, users can keep track of their contacts’ availability (Presence); conduct an Instant Messaging (IM) session; make calls via VoIP; initiate or join an audio, video, or web conference; or make a phone call within the Lync organization, with federated partners or to phones on the PSTN. The Microsoft Lync 2013 desktop client is available for Windows and for the Macintosh operating system, and mobile versions are available for Windows® Phone, iPhone iPad, and Android devices.
Enterprise Voice Components in Lync Server 2013
The Enterprise Voice component in Lync Server 2013 provides a full featured VoIP solution that you can use to enhance or replace traditional PBX telephone systems. The Enterprise Voice component provides the functionality that the following sections describe.
Placing and Receiving Voice Calls
Enterprise Voice enable users can initiate calls from a Lync client by typing a name or phone number on their keyboard, or using a dial pad displayed on their screen. Users also can utilize VoIP Phone Editions or mobile devices to make voice calls via the Lync Server infrastructure. These devices can be active simultaneously.
Users are alerted to incoming calls on all of their devices simultaneously, with customizable ringtones on IP phone devices and a notification similar to an instant message on their computers.
A Lync Server 2013 Enterprise Voice deployment supports calls to and from the PSTN. Connecting Enterprise Voice to the PSTN requires one or more of the following:
- A SIP trunk to an Internet Telephony Service Provider (ITSP)
- An IP-PBX connected to the PSTN
- A PSTN gateway connected to the PSTN
- A Survivable Branch Appliance (SBA) or Survivable Branch Server connected to the PSTN
Basic Call Features
Enterprise Voice provides all of the basic features that a traditional PBX provides. For example, while Lync users are on a call, they can answer additional incoming calls or initiate outgoing calls, and the existing active call is put on hold automatically. Users can transfer calls from one user to another, either directly or after the first user speaks privately with the second user. Users also can transfer calls to another device. For example, they could transfer an active call to their mobile phone.
Advanced Calling Features
Enterprise Voice includes several advanced calling features as well, such as:
- Call Parking, which enables users to put a call on hold, and then retrieve it from another phone. When a user parks a call, the original answering phone becomes free for another call.
- Delegation, which enables users to assign call handling to one or more assistants, such as a Personal
Assistant or a Colleague. The delegate can perform multiple calling tasks on behalf of the user who initiated the delegation, including screening calls, placing calls, and initiating conferences.
- Team calling, which enables a user to have incoming calls simultaneously ring the phones of teammates, for functions such as group-call pickup and department calling.
- Response Groups, which you can configure for queuing and routing calls intelligently to designated agents. You typically would use this for groups such as your information technology (IT) helpdesks, an accounting hotline, and other internal contact centers.
Lync Server 2013 supports enhanced 9-1-1 (E9-1-1) for North America. This feature provides additional location information to dispatchers of emergency services.
The new voice resiliency capability allows a site with an SBA or Survivable Branch Server to continue to provide users with the ability to make and receive Enterprise Voice calls if the wide area network (WAN) that connects the branch and central sites is down. You also can configure it to provide resiliency between central sites.
Enterprise Voice Options for Connecting Lync Servers to the PSTN
When you deploy Enterprise Voice on Lync Server, you have several options for connecting the Lync Server to the PSTN. Like the Exchange 2013 servers that provide Unified Messaging services, Lync servers can communicate only on packetswitched networks. Therefore, some type of gateway is required between the Lync server and the PSTN.
At a high level, there are three options for connecting a Lync Server deployment to the PSTN, including:
- Connecting through a VoIP gateway and traditional PBX. This scenario is common in organizations that have deployed an analog or digital PBX, and which want to retain the PBX for their telephone systems. In this deployment, the VoIP gateway provides protocol conversion between the packetbased network where Lync Server is deployed, and the PBX, which is connected to the PSTN.
- Connecting through an IP-PBX. This scenario is common for organizations that have deployed an IP-PBX, and which want to retain PBX for all or part of their telephone system. In this deployment, the IP-PBX provides protocol conversion between the PSTN and the packet-based network where Lync Server is deployed.
Overview of an Exchange Server 2013 and Lync Server 2013 Integration
Before configuring the integration of Exchange Server 2013 and Lync Server 2013, you need to understand how the integration works. When you configure the integration, you are configuring Exchange Server 2013 Unified Messaging components that will enable Exchange Server 2013 to communicate with Lync Server 2013.
Furthermore, you are configuring Lync Server 2013 components that enable Lync to communicate with exchange. Two tools are provided to configure the required objects.
To configure the integration of Lync and Exchange Unified Messaging, you must first run the ExchUCUtil script (ExchUCUtil.ps1) to configure the Exchange Server environment. The script does three things:
- It grants the Lync server accounts permission to read Exchange Unified Messaging AD DS objects, so that it can create contact objects for each auto attendant and subscriber access.
- It creates a UM IP gateway object for each Lync Server 2013 pool, and then associates the gateways to the UM SIP dial plans that you define for Lync Server 2013.
- It creates an UM hunt group for each UM IP gateway. The hunt group pilot identifier will be the name of the dial plan associated with the UM IP gateway.
You will run the ExchUCUtil script when you configure Exchange Unified Messaging integration initially with Lync Server 2013. You should run the script again whenever you create Exchange UM SIP dial plans that you will use to integrate with Lync Server, and whenever you add a new Lync 2013 server to the environment.
Exchange UM Integration Utility
The Exchange UM Integration Utility (OcsUMUtil.exe) is a tool that you will run on the Lync 2013 server. When you run the tool, you will create contact objects in Active Directory that Lync Server 2013 uses to link to the Exchange UM Auto Attendant and Subscriber Access numbers. When you run the tool, it will read the Exchange Unified Messaging configuration. You then can create the contact items in an existing organizational unit or you can create a new organizational unit. The contact items are assigned a SIP address and a phone number. By default, the subscriber access contact is assigned the phone number that you configured for outlook Voice Access on the UM dial plan, and the auto attendant contact is assigned the phone number assigned to the UM auto attendant.
The SIP dial plan that you configure on the Exchange Servers must use mutual TLS encryption for all traffic. This means that you must install a certificate on all Exchange 2013 servers that will communicate with the Lync 2013 servers, as well as on the Lync 2013 servers. The certificates that you deploy on both sets of servers must be trusted by the other set of servers. You can configure certificates in several ways:
- Obtain certificates from a trusted public CA for both sets of servers. This will eliminate any trust issues.
- If you deploy an internal CA, you can obtain certificates for both sets of servers from the internal CA.
- If you are using self-signed certificates, you must import the certificates to the trusted root certification authority node on all other servers. We do not recommend this approach.
Implementing Exchange Server UM Integration with Lync Server
To configure the integration of Exchange Server 2013 and Lync Server 2013, complete the following steps:
- Install Lync Server in the same location as the Exchange 2013 Client Access servers and Mailbox servers. A fast LAN connection should connect the servers.
- Configure the Enterprise Voice components on the Lync servers, including:
- PSTN connectivity. To provide full telephone functionality, the Lync servers must be able to send and receive calls from PSTN telephones.
- Dial plans. You will need to create dial plans for all internal users.
- Call routing rules. These rules define how calls are routed within the organization or to the PSTN.
- Normalization rules. These rules define how Lync will handle specific types of calls. For example, if
you want users to be able to dial a five-digit extension to reach other internal users, you will need to create a normalization rule that translates the five-digit extension into the full phone number.
- Verify that the infrastructure’s servers trust the certificates installed on the Exchange and Lync servers.
- Create and configure a SIP URI dial plan in Exchange 2013. You must configure the dial plan to use
the SIP Secured or Secured setting to enforce mutual TLS.
- Add all Client Access and Mailbox servers to the SIP dial plan. This will enable all Exchange servers to
answer incoming calls from Lync Server.
- Set the startup mode for the Unified Messaging services to Dual, and then restart the Microsoft
Exchange Unified Messaging service on each Mailbox server, and the Microsoft Exchange Unified
Messaging Call Router service on each Client Access server.
- Run the ExchUCUtil.ps1 script from the <Exchange Installation folder>\Exchange Server\Script folder on any Exchange Server.
- Run OcsUMUtil.exe from the %CommonProgramFiles%\Microsoft Lync Server 2013\Support folder on a Lync Server.
- Enable your users for UM and Enterprise Voice. When you enable users for voice mail, create a SIP address for the users who will use Enterprise Voice. In most cases, this SIP address will be the same SIP
address that will be used when a user is enabled for Enterprise Voice.
If you like this post, kindly leave your valuable feedback and comments.