A Communications Server Enterprise Voice deployment supports calls to and from the public switched telephone network (PSTN). PSTN calls require that you configure a SIP Trunk that connects Communications Server to an Internet telephony service provider (ITSP), to an IP-PBX on your local network, or to an IP-PSTN gateway by way of the Mediation Server or a supported hardware Survivable Branch Appliance.

For details about the PSTN connectivity options supported by Communications Server, see PSTN Connectivity. For details about the outbound call routes that need to be configured between Communications Server and ITSPs, IP-PBXes, or IP-PSTN gateways, see Planning Outbound Call Routing.

If you have deployed or plan to deploy Microsoft Exchange Server in your organization, you can use Exchange Unified Messaging (UM) features to provide voice mail to Enterprise Voice users. For details about integrating Exchange UM, see Exchange Unified Messaging Integration.

Media bypass is a new feature of Communications Server that enables media to bypass the Mediation Server in order to be processed by a SIP trunk’s site-consolidated media termination points (MTP) instead. Media bypass is only available for certain types of calls. For details about configuring support for media bypass in your Enterprise Voice deployment, see Media Bypass.

In Communications Server 2010, a single Mediation Server can now control multiple IP-PSTN gateways. In previous releases, there was a 1:1 ratio of Mediation Server to Gateways. In this release, when defining a call route, you specify the gateways associated with that route, but you do not specify which Mediation Servers are associated with the route. Instead, you use Topology Builder to associate gateways, and therefore routes, with one or more Mediation Servers. For details, see Multiple Gateway Support.

Communications Server 2010 provides you with the ability to manipulate the caller ID for outbound calls. As you plan outbound call routes, consider whether to manipulate the caller ID for calls placed by certain users, groups, sites, or all users.

In Communications Server 2010, you can help streamline outbound routing by creating one or more translation rules that are associated with the gateway during trunk configuration and which manipulate the Request URI of a call prior to routing it to the gateway. By performing outbound route translation on the server, you can reduce the configuration required on the gateway to manipulate phone numbers into a local dialing format.

Communications Server 2010 introduces call admission control as a way to manage whether calls can be established based on available network bandwidth. For details about assessing your network sites, your network bandwidth, and configuring call admission control policies to manage bandwidth, see Call Admission Control.

Also new to Communications Server is support for enhanced 9-1-1 (E9-1-1), a feature that provides location information to dispatchers of emergency services. For details about E9-1-1 and associating Enterprise Voice users’ phone numbers with their physical locations, see Emergency Services (E9-1-1).

Another new feature of Communications Server is the ability for Enterprise Voice users to have a second, unlisted telephone number for receiving incoming calls. For details about private telephone lines, see Private Telephone Lines.

Communications Server introduces support for common area phones which make it possible, for the first time, to use Communications Server to provide phone service in and to enable unified communications functionality from common areas, such as building lobbies. For details about planning for common area phones in your Communications Server environment, see Preparing for Devices.

Communications Server now supports analog devices in the Enterprise Voice environment. Analog devices include analog phones or analog fax machines connected to an analog port of a gateway or a PBX, ATA gateways with 2 to 4 analog ports into which analog devices can connect and which are connected to a SIP-PSTN gateway, and SIP-PSTN analog gateways (IP-PSTN gateways with native analog ports).

Note:
In the Beta release, no documentation is available to describe how to plan for integrating analog devices into your Enterprise Voice deployment.

Communications Server now includes support for Enterprise Voice users to continue making and receiving calls in the event a central site becomes unavailable by connecting to secondary central sites. Branch site resilience is the ability of a branch office to provide continuous Enterprise Voice service to its employees in the event the WAN link to its central site becomes unavailable. For details about planning for a resilient Enterprise Voice deployment, see Providing High Availability for Enterprise Voice, Central Site Voice Resilience, andBranch-Site Resiliency.

Communications Server includes call management features that affect how incoming calls are routed and answered. For example, you can enable call parking and specify what happens to incoming calls to unassigned phone numbers.

You can continue to use the feature from Microsoft Office Communications Server 2007 R2 in which you configure users to act as delegates for their manager’s incoming calls. You can also continue to configure routing and queuing of incoming calls to groups of designated users (called response groups). New functionality for response groups includes the ability for agents to handle incoming and outgoing calls anonymously, for agents using Microsoft Communications Server 2010 Attendant to answer waiting calls in any order, integrated manageability, more flexible IVR configurations and prompts, and a Web service that supports customized agent consoles.

For details about planning for these call management features, see Planning for Call Management Features.