Enterprise Voice White Book

PSTN Integration 6-1
Connecting to the PSTN

There are a number of options available for integrating a customers existing
telephony infrastructure with Skype for Business Server 2015. Before you begin
designing voice infrastructure, you should choose a telephony interoperability
option that best suits your current network and organization. A simple decision
model can be used to determine the best approach for upgrading, replacing, and
coexisting with their existing PBX.
Skype for Business Server 2015 provides various options for implementing
telephony interoperability. You should be familiar with the different telephony
interoperability optionsIP-PSTN gateways, Session Initiation Protocol (SIP)
trunking, and Direct SIP with IP-private branch exchange (PBX).
Background Definitions
Public Switched Telephone Network (PSTN)
In reality, the world is connected as one Public Switched Telephone Network (PSTN).
Phones, including land-lines, mobile phones, and Internet phones that are
addressable by using the <Country Code><Area Code><Number> format are
connected to the PSTN. Because different countries have different tariff and legal
requirements, each countrys phone system is typically referred to as a separate
PSTN. A multinational organization will typically have multiple connections to the
PSTN (one or more per country), based on concentration of internal users or
customers.
Private Branch eXchange

Private Branch eXchange (PBX) is a telephony device that acts as a switch for
switching calls for a particular company, branch office, or division. A PBX more
efficiently uses trunk lines connecting to the PSTN by routing private network calls
within the PBX, and it dynamically assigns trunk lines to extensions when a
connection to the PSTN is made by either an incoming or outgoing call. Almost
every large enterprise in the world uses PBX. With the advent of compact digital
PBX and telephone deregulation PBX, even small companies are connecting through
PBX.
Note: The alternative PBX for businesses has been centralized carrier-based
exchanges or CENTREX. About 10 million CENTREX managed extensions still exist
in the United States, and most of these will be replaced in the near future
(Centrex: It's alive (for now)!, Brad Reed, Network World February 07, 2012
11:14 AM ET., http://go.microsoft.com/fwlink/?LinkID=313692 CENTREX
replacement is an opportunity for Skype for Business.
PBX can be grouped in many different ways based on features. Most importantly for
Microsoft Skype for Business2015 is determining IP support.
An IP PBX is a PBX that supports the IP protocol to connect phones by using an

Ethernet or packet-switched local area network (LAN), and it sends its voice
conversations in IP packets.
A legacy PBX is a PBX that does not pass IP packets.
Voice over Internet Protocol (VoIP)

VoIP is hardware/software technology that enables the use of an IP-based network
as the transmission medium for telephone calls.
Session Initiation Protocol (SIP)

SIP is an industry standard protocol for controlling communication sessions such as
voice and video calls over IP.
Internet Telephony Service Provider (ITSP)

An Internet telephony service provider (ITSP) offers digital telecommunications
services by using VOIP technology. A companys telephone network connects by
using IP to the ITSP and the ITSP connects the calls to PSTN. The ITSP makes more
efficient use of the connections to the PSTN by sharing trunks with multiple
customers and routing long distance and international calls to local PSTN
connections.
UCOIP
Unified Communications Open Interoperability Program
(UCOIP)
Infrastructure qualified for Microsoft Skype for Business
http://go.microsoft.com/fwlink/?LinkID=223942
UCOIP includes a product qualification program for SIP and PSTN gateways and it
ensures that customers have seamless experiences with setup, support, and use of
qualified telephony infrastructure and services with the Microsoft Unified
Communications software, including Skype for Business Server and Exchange
Server. Only products that meet rigorous and extensive testing requirements and
conform to the specifications and test plans will receive qualification. While the
specifications are based on industry standards, this program also defines:
Specific requirements for interoperability with Skype for Business Server and
Exchange Server Voice Mail.
Testing requirements for qualifying interoperability with Skype for Business Server and
Exchange Server Voice Mail.
Installation, setup, and configuration requirements through a Quick Start Guide.
Release Notes with any known issues.
Documented support process between Microsoft and the vendor.
Enterprise-class standards for audio quality, reliability, and scalability.

Components
UCOIP covers the devices, components, infrastructure, and services to integrate
Skype for Business with the customers communication infrastructure. The following
components impact voice integration:
IP-PSTN Gateways independently qualified to work with Skype for Business Server,
including documentation and support.IP-PSTN gateways translate signaling and media
between the Enterprise Voice infrastructure and PSTN. These third-party hardware
components translate signals either directly or through a connection to SIP trunks. The
gateway terminates PSTN and is usually isolated in its own subnet. The Mediation
Server connects the IP-PSTN gateway to the enterprise network.All qualified gateways
must support TLS, but can enable TCP also. TCP is supported for gateways that do not
support TLS.
IP-PBX independently qualified to work with Skype for Business Server, including
documentation and support. The IP-PBX provides the connection to the PSTN.
IP-PBXs supported by Microsoft but have not gone through the formal UCOIP
qualification process. Sufficient internal testing has been performed by Microsoft such
that specific configurations are supported by Microsoft (where applicable with known
limitations). These configurations utilize the commercially available production SIP
trunk interface of the IP-PBX vendor, but may not be supported by the IP-PBX vendor.
In addition, IP-PBX vendor-provided complete documentation for installation and
setup, release notes, or documented support processes may not be available.
Wherever possible, Microsoft will endeavor to provide documentation for installation
and setup.
Service Providers
UCOIP also covers ITSP. A qualified ITSP uses qualified components and is
independently certified also.
Typical Legacy Enterprise PBX

A typical starting point at many enterprises is a strong PBX presence not only in
infrastructure, but also experience of the employees, managers, and support staff.
To coexist with or replace the existing PBX with Skype for Business, a thorough
understanding of the existing capabilities is needed. In many cases, the Skype for
Business environment needs to closely emulate those capabilities.
The existing PBX provides:
Connection points to the PSTN for voice phones, but it may also include fax, data, and
WATTS services.
Existing number plan for mapping trunk lines to extensions.
Functionality, which ultimately defines dialing habits or patterns of the users for
calling internal, local, and long distance numbers.
Limits to the class of service available at each extension, such as this phone can
outbound dial international numbers or this phone can place a call on hold.
Decision 1: Legacy PBX integration

Connecting to the Legacy PBX
Because most enterprise will be starting with one or more PBX systems, typically,
the first decision when planning the connection to the PSTN is; what role should the
PBX have in the new infrastructure, both in the short term and long term? The PBX
systems currently connect the enterprise phone system to the PSTN.
Advantages
No need to make changes to PSTN connection
PBX owns number plan
User can keep existing number, transparent to inbound calls
Internal calling at no cost

Benefit from the existing PBX infrastructure
Benefit from the existing trunk capacity
Disadvantages
PBX dependencies
Additional PBX cost
Requires PBX configuration
What happens when migration is done?
Migrating a user requires changes in the PBX
Note: Options for connecting through a PBX are covered in lesson 2.

Connecting directly to the PSTN
Advantages
Easy and fast
No additional PBX investments and configuration
Disadvantages
New numbers for the end-user
Internal calls between Skype for Business and legacy PBX use PSTN
o Need additional trunk capacity to call between Skype for Business and legacy PBX
o Costs may increase
Migrating a user requires changes at the Provider
Note: For customers without an existing PBX (rare), options for connecting directly to the
PSTN can be used without the PBX.
Decision 2: POTS/TDM or SIP Trunking

There are two options for connecting an enterprises telephone system to the PSTN.
Connecting Through a Gateway

A phone system can connect directly to the PSTN using a Plain Old Telephone
System/Time Division Multiplexing (POTS/TDM) trunk. In the case of Skype for
Business, an IP-PBX gateway is required to convert between SIP and TDM, which the
PSTN accepts. Connecting through a gateway provides the following benefits:
It is more broadly understood.
There is no wide area network (WAN) dependency.
Users have a local carrier choice.
Branch resiliency is available through branch connection to the PSTN.
Connecting through a SIP Trunk

A SIP trunk is an IP connection that establishes a SIP communications link between
your organization and an Internet telephony service provider (ITSP) beyond your
firewall. Typically, a SIP trunk is used to connect your organizations central site to
an ITSP. In some cases, you may also opt to use SIP trunking to connect your branch
site to an ITSP.Skype for Business and IP-PX can connect to an ITSP by using the SIP.
SIP trunking provides the following benefits:
It offers consolidation and numbering flexibility.
Disaster recovery is done through redundant connections to the ITSP and ITSP shared
excess capacity.
It provides end-to-end SIP call flow to enable features and supplementary services.
It can deploy central trunking for management or routing purpose.
It eliminates per-channel model to provide more flexibility in trunk provisioning.
You do not need an intermediary gateway (ITSP manages PSTN gateway).
It offers the potential to share ISP/ITSP.

Once it has been decided what role, if any, the existing PBX systems will play in the new
infrastructure and how the connection to the PSTN will be made, it is possible to determine
the best option for connecting. The next topics will examine each model in more
detailand describe how to connect Skype for Business to the PSTN.
Direct Connection Through a Gateway

PSTN Gateway
Public switched telephone network (PSTN) gateways are third-party hardware
components that translate signaling and media between the Enterprise Voice
infrastructure and the PSTN, either directly or through connection to SIP trunks. In
either topology, the gateway terminates the PSTN. The following are the basic
functions of the SIP/PSTN gateway:
Translates signaling and media between Mediation Server and the PSTN
Functions as a termination point for SIP trunks
Functions as an intermediary between Skype for Business Server 2015 and any
unsupported IP-PBX or Time Division Multiplexing (TDM) PBX
Connects analog phones and fax machines to the Skype for Business Server 2015
infrastructure
The gateway is isolated in its own subnet and is connected to the enterprise
network through the Skype for Business Server 2015, Mediation Server. Based on
call volume, geography, and redundancy, one or more gateways can be installed.
Each PSTN gateway that you deploy must have at least one corresponding
Mediation Server. Placing the gateways on a separate subnet makes adding
additional capacity in the future easier.
Supported Gateways
Gateway location may also determine the types of gateways that you choose and
how they are configured. There are dozens of PSTN protocols, none of which is a
worldwide standard. If all your gateways are located in a single country/region, this
is not an issue, but if you locate gateways in several countries/regions, each must
be configured according to the PSTN standards of that country/region. Moreover,

gateways that are certified for operation in, for example, Canada, may not be
certified in India, Brazil, or the European Union.
Gateway Sizing
The PSTN gateways that most organizations will consider deploying range in size
from 2 to as many as 960 ports. Larger gateways are typically cheaper to purchase
and operate if fully utilized, but smaller capacity gateways may be preferred
because the incremental cost to provide redundancy can be lower. When estimating
the number of ports your organization requires, use the following guidelines:
Organizations with light telephony usage (one PSTN call per user per hour) should
allocate one port for every 15 users.
Organizations with moderate telephony usage (two PSTN calls per user per hour)
should allocate one port for every 10 users.
Organizations with heavy telephony usage (three or more PSTN calls per user per
hour) should allocate one port for every five users.
Additional ports can be acquired as the number of users or amount of traffic in your
organization increases.
Mediation Server
You must deploy Skype for Business Server 2015, Mediation Server if you deploy the
Enterprise Voice workload and connect to another phone system, including in-house
IP-PBX, PSTN or Skype for Business Online.
The Mediation Server translates signaling, and in some configurations, media
between your internal Skype for Business Server 2015, Enterprise Voice
infrastructure, and a PSTN gateway or SIP trunk. The main functions of the
Mediation Server are as follows:
Encrypting and decrypting Secure Real-time Transport Protocol (SRTP) on the Skype
for Business Server side. All communications between Skype for Business components
are encrypted while communications with the PSTN or ITSP may not be encrypted.
Translating SIP over TCP (for gateways that do not support TLS) to SIP over mutual TLS
Translating media streams between Skype for Business Server and the gateway peer
of the Mediation Server. For external calls the media must go through the mediation
server.
Connecting clients that are outside the network to internal Interactive Connectivity
Establishment (ICE) components, which enable media traversal of network address
translation (NAT) devices and firewalls.
Acting as an intermediary for call flows that a gateway does not support, such as calls
from remote workers on an Enterprise Voice client.
In deployments that include SIP trunking, working with the SIP trunking service
provider to provide PSTN support, which eliminates the need for a PSTN gateway
The Mediation Server is collocated with the Front End Server by default. The
Mediation Server can also be deployed in a stand-alone pool for performance
reasons, or if you deploy SIP trunking, in which case, the stand-alone pool is
strongly recommended.
If you deploy Direct SIP connections to a qualified PSTN gateway that supports
media bypass and Domain Name System (DNS) load balancing, a stand-alone
Mediation Server pool is not necessary. A stand-alone Mediation Server pool is not
necessary because qualified gateways are capable of DNS load balancing to a pool
of Mediation Servers and they can receive traffic from any Mediation Server in a
pool.
Direct Connection Through SIP Trunking

SIP trunking is widely used in Skype for Business Server 2015 deployments for
providing PSTN connectivity. Using SIP trunking, you can connect the Mediation
Server to an ITSP by using SIP-over-TCP for signaling and G.711 for media
transmission.
For example, a virtual private network (VPN) is connected between the enterprise
network and the service provider network. However, a VPN connection is not
required for establishing a SIP trunk in Skype for Business Server 2015. To connect
the Mediation Server and the Session Border Controller (SBC) at the service
provider, you can use a VPN connection, a private connection, or a non-encrypted
connection over the Internet.
VPN Connection
In a VPN connection, the Mediation Server connects to the SIP trunk by using a VPN
connection to the service provider. The VPN can be used as an Internet connection
or as an existing Multiprotocol Label Switching (MPLS) WAN connection. The benefit
of the VPN connection is to create an isolated, secure channel to carry the SIP-
trunked calls.
Session Border Controller (SBC)

A session can be thought of as a call, but can include non-voice sessions for media
and data services also. A session uses a sequence of signals to establish, conduct,
and disconnect a session. The SBC handles the signals between the ITSP and
customer network. The SBC can be thought of as the logical separation between the
customer and the ITSP. Depending on the ITSP requirements, the SBC maybe
located at the customer or ITSP.
Private Connection
In a private connection, the Mediation Server connects to the SIP trunk by using a
private network connection to the service provider. This setup can be a dedicated
MPLS T1 connection installed for the purpose of SIP trunking.
Non-Encrypted Internet Connection

The Mediation Server uses the direct Internet connection to connect to the SIP
trunk, if the Internet connection is non-encrypted. This topology does not use an
isolated link to the SIP trunk, as in the case of private and VPN options.
Note: The connection to the ITSP will vary depending on the ITSP.
For example, some ITSPs locate the SBC at the customer site. Contact your ITSP for exact
requirements for connecting by using SIP trunking.
User Configuration
Skype for Business and PBX phone numbers can be the same.
Connecting to the Existing PBX

PBX systems provide one or more connections to the PSTN, so companies may
retain their PBX system to provide a trunk to the PSTN. In this case, when
configuring Skype for Business Server 2015, the PBX can be treated like any other
PSTN gateway.
Companies may also continue home users on their PBX, either short term during the
transition to Skype for Business Server 2015 or long term for some users who do not
require the functionality provided by Skype for Business. The PBX as a gateway and
PBX users should be treated as two separate workloads when defining dial plans.
This way it is possible to create rules that allow a Skype for Business Server 2015
caller to call a PBX user, but still block them from making external calls through the
PSTN gateway.
A PBX system implements routing rules similar to Skype for Business Server 2015,
using its own proprietary format. Therefore,Skype for Business Server 2015 dial
plans need to be coordinated with the PBX. The routes defined by Skype for
Business are responsible for routing to the PBX, then the PBX decides if the call will
be routed to the PSTN or to a PBX user.
Connecting Through PBX by Using SIP

When using a qualified or supported IP-PBX, Skype for Business Mediation Server
communicates directly with the IP-PBX by using SIP over TCP (SIP/TCP) or SIP over
TLS (SIP/TLS). Skype for Business Server 2015 already uses SIP/TLS for
communications between Skype for Business Front End pool and the Mediation
Server. The Mediation Server:
Works only for qualified or supported PBX systems (UCOIP)
Provides interconnection between IP-PBX and Skype for Business Server 2015
Provides voice capabilities between endpoints on either call control server
Enables endpoints on both sides to utilize features on the other call control server
Provides simplest method of interoperability, relying on standard SIP protocols
Connecting Through PBX by Using a Gateway

If connecting to a TDM or unsupported PBX, then Skype for Business Server 2015
can be connected to the PBX through a qualified gateway. It uses qualified gateways
(UCOIP):
As an intermediary in scenarios, such as SIP to TDM or H.323, or to other nonqualified

call controls. H.323 defines the protocols to provide audio-visual communication
sessions on any packet network.
As a gateway for back-to-back user agent (B2BUA) performs SIP-to-SIP transcoding for
calls between Skype for Business Server 2015 and other resources which also use SIP.
PSTN Sizing
The PSTN trunk capacity requirements can be calculated by using industry standard
methods.
First, start with the existing capacity. This capacity represents the current calling
patterns. Ensure that you take into consideration any changes to calling rules. For
example, the default setting for PBX phones may be inbound calls only, while Skype
for Business users can make local calls.
Note: When Skype for Business is connected directly to the PSTN, if there are
any legacy PBX users, PSTN capacity requirements may increase to handle calls
between Skype for Business clients and PBX users, until all users are migrated off
from the PBX.
Skype for Business includes additional features which may impact call volumes:
Simultaneous ringing. Simultaneous ringing enables a user to configure so that

incoming calls will ring at a primary phone number and a second number, including
PSTN phones, such as their mobile phone. The ringing or call invite has minimal
impact on PSTN capacity. On the other hand, the call session, if answered on the
secondary device, uses two PSTN connectionsone inbound connection for the
original call, and one outbound connection for the connection to the secondary device
on the PSTN.
PSTN conferencing. PSTN conferencing enables users to easily create multi-party

conferencing calls without the assistance of an operator. Each external user in the call
requires a separate PSTN connection.
Audio conferencing. Audio conferencing provides dial-in conferencing capabilities for

external users who cannot connect to web conferences or cannot receive audio
through the web. For public events, audio conferencing can require a large PSTN
trunk.
Mobile Users. Mobile users will connect through

In some cases, if Skype for Business is replacing an outside service for small and ad
hoc web and audio conferencing, the PSTN traffic may be reduced because
employees and internal users no longer have to call out through the PSTN to
connect to the conference.
Note: For larger public conferences a third party service may still be used.
Erlang B is a formula for estimating the number of PSTN trunks required to support
the peak call volume (Busy Hour Traffic) with an acceptable percentage of failed
calls, busy signal cause by no available trunk. Call volume is based on the number
and duration of calls. The result is expressed as an Erlang number, where an Erlang
represents one continuous call for 1 hour duration.
An Erlang B calculator is included in the Lync server 2010 and 2013
Bandwidth Calculator.
Inter-Trunk Routing
Skype for Business Server 2015 provides basic session management through the
support of inter-trunk routing. This new capability enables Skype for Business Server
to provide call control functionalities to downstream telephony systems. Inter-trunk
routing can interconnect an IP-PBX to a PSTN gateway so that calls from a PBX
phone can be routed to the PSTN, and incoming PSTN calls can be routed to a PBX
phone. Similarly, Skype for Business Server can interconnect two or more IP-PBX
systems so that calls can be placed and received between PBX phones from the
different IP-PBX systems.
Skype for Business supports the association of a set of PSTN usages on an incoming
trunk to determine a call route to an outgoing trunk.
Inter-trunk configuration remains familiar for the administrator with the use of existing
routing configuration concepts.
Media bypass in inter-trunk routing calls is supported.
Inter-trunk routing call authorization scope is at the trunk level.
The same call authorization applies to all calling endpoints connected through the
trunk.
The next slides examine the three inter-trunk routing options in more detail:
Incoming PSTN calls to an IP-PBX system through Skype for Business
Outgoing IP-PBX calls to a PSTN network through Skype for Business
Outgoing IP-PBX calls to another IP-PBX system through Skype for Business
Routing of IP-PBX Calls to PSTN by Using Skype for Business

Inter-trunk routing can handle inbound calls from the PSTN to a PBX phone user or route
an outbound call from a PBX phone user to the PSTN.
Incoming call from the PSTN to the PBX trunk

In this case Skype for Business Server 2015 is used route the call from the PSTN or
originating PBX user to the receiving PBX user. The call signaling goes through the
following:
1 The incoming call is routed from the originating receiving gateway to Skype for
Business Server 2015.
2 Skype for Business Server 2015 validates incoming trunk associated PSTN usages.
3 Analyzing the phone number Skype for Business Server 2015 determines a route, in
this case to the PBX.
4 If necessary, applies outbound translation rules, so the PBX can understand the phone
number and further route the call.
5 Mediation Server routes the call to an outgoing trunk to PBX.
6 PBX determines route to PBX user.
Outgoing call from the PBX trunk

In this case Skype for Business Server 2015 is used route the call from the
originating PBX user to an external receiver through the PSTN. The call signaling
goes through the following:
7 PBX determines the route for the call:
a. If the receiver is on the same PBX routes to receiving user,
b. If it is an external number accessible through this gateway, route to the PSTN.
c. All other numbers route to Skype for Business Mediation Server to determine
route.
8 Skype for Business Server determines route to PSTN through a PBX or gateway.
9 If necessary, applies outbound translation rules, so the PBX or gateway can

understand the phone number and further route the call.
10 Mediation Server routes the call to the outgoing trunk to PBX or gateway.
11 If routed to a PBX, determine if the call is destined to a number managed by that PBX,
otherwise the PBX or gateway connects the call to PSTN.
Translation Rules
Skype for Business Server 2015 Enterprise Voice requires that all dial strings be
normalized to E.164 format for the purpose of performing reverse number lookup
(RNL). In Microsoft Lync Server 2010, translation rules are supported only for called
numbers. New in Microsoft Skype for Business Server 2015, translation rules are
also supported for calling numbers. The trunk peer (that is, the associated gateway,
PBX, or SIP trunk) may require that numbers be in a local dialing format. To translate
numbers from E.164 format to a local dialing format, you can define one or more
translation rules to manipulate the request URI before you route it to the trunk peer.
For example, you can write a translation rule to remove +44 from the beginning of a
dial string and replace it with 0144.
By performing outbound route translation on the server, you can reduce the
configuration requirements on each individual trunk peer to translate phone
numbers into a local dialing format. When you plan which gateways and how many
gateways to associate with a specific Mediation Server cluster, it may be useful to
group trunk peers with similar local dialing requirements. This can reduce the
number of required translation rules and the time it takes to write them.
Note: Routing a call deals with passing the messages during the signaling
phase to connect the caller to the receiver. Once a call is connected media, in this
case a voice conversation, is passed between the caller and receiver. Typically,
inter-trunk routing uses Media Bypass to handle the call media to passed the
conversation over a more direct path between caller and receiver. Media Bypass is
covered in Module 7.
Routing of IP-PBX Calls to Another IP-PBX by Using Skype for

Business
If Skype for Business Server 2015 is integrated with multiple PBXs, inter-trunk routing can
be used to route calls between the PBX, without using the PSTN.
Call from one PBX to another PBX

In this case Skype for Business Server 2015 is used route the call from the
originating PBX user to a receiver on another PBX. The call signaling goes through
the following:
1 PBX determines the route for the call:
a If the receiver is on the same PBX routes to receiving user,
d. If it is an external number accessible through this gateway, route to the PSTN.
e. All other numbers route to Skype for Business Mediation Server to determine
route.
12 Skype for Business Server determines route to the receivers PBX.
13 If necessary, applies outbound translation rules, so the PBX can understand the phone
number and further route the call.
14 Mediation Server routes the call to receivers PBX.
15 The PBX determines if the call is destined to a number managed by that PBX and
routes the call to the receiver.
Configuring Inter-Trunk Routing

Skype for Business Server 2015 can interconnect an IP-PBX to a PSTN gateway so
that calls from a PBX phone can be routed to the PSTN, and incoming PSTN calls can
be routed to a PBX phone. Similarly, Skype for Business Server 2015 can
interconnect two or more IP-PBX systems so that calls can be placed and received
between PBX phones from different IP-PBX systems.
This inter-trunk routing feature can be configured by using the Skype for Business
Server Management Shell cmdlet, Set-CsTrunkConfiguration, with the new
parameter, PstnUsages. This parameter specifies the set of PSTN usage records to
use. A trunk uses this PSTN usage to determine a path and to route all incoming
calls accordingly.
If route translation is needed, then use the Set-
CsOutboundTranslationRulecmdlet.
The syntax for the cmdlet used for inter-trunk routing is:
Skype for BusinessMamagement Console Command Format

Set-CsTrunkConfiguration -Identity <TrunkId> -PstnUsages @{add="<UsageString>"}
Set-CsOutboundTranslationRule -Identity site:Redmond/OTR1 -Translation '$1'
Location Based Routing

Using Inter-trunk routing to route calls between PBXs and PSTN access points may
violate local laws and tariffs. To block calls which may violate laws for a given
location, Location based Routing rules can be configured to apply any call
originating from that location. Location based routing should be used in conjunction
with inter-trunk routing to block any trunk to trunk routes that are illegal.
Location based routing can also be used for Skype for Business users when your
company has a highly mobile workforce. Normally, a dial plan is associated with
users based on their E.164 number and does not change based on how where they
connect to Skype for Business. If the users frequently connect from different offices,
particularly in a other countries, the normal dial plan may not be the best, or legal,
for their current, so location based routing would apply.
Mediation Server
The Mediation Server translates signaling, and in some configurations, media
between your internal Skype for Business Server 2015 Enterprise Voice
infrastructure and a PSTN gateway or a SIP trunk. On the Skype for Business Server
2015 side, Mediation Server listens on a single mutual TLS (MTLS) transport
address. On the gateway side, Mediation Server listens on all listening ports
associated with trunks defined in the Topology document. All qualified gateways
must support TLS, but they can enable TCP also. TCP is supported for gateways that
do not support TLS.
If you also have an existing PBX in your environment, Mediation Server handles calls
between Enterprise Voice users and the PBX. If your PBX is an IP-PBX, you can
create a direct SIP connection between the PBX and Mediation Server. If your PBX is
a TDM PBX, you must also deploy a PSTN gateway between Mediation Server and
the PBX.
Collocation vs. Standalone

Mediation Servers can be collocated with front-end servers. The advantage is the
potential reduction in server count. The trade-off is the Mediation Server will
support fewer calls and media bypass must be enabled. In many cases, it is also
preferable to locate the Mediation Server closer to the gateway or PBX, so a
separate Mediation Server will be required.
The Mediation Server is collocated with the front-end server by default. The
Mediation Server can also be deployed in a stand-alone pool for performance
reasons, or if you deploy SIP trunking, in which case, the stand-alone pool is
strongly recommended.
Maximum Number of Calls

Role Placement
(with media)
Standalone 1100 1,500
Collocated 150
Media Bypass and Scalability

Media bypass refers to removing the Mediation Server from the media path
whenever possible for calls whose signaling traverses the Mediation Server. In other
words, after a call has started, the Mediation Server is no longer needed, so the
Mediation Server is removed from the path and a connection is established directly
between the Skype for Business client and the IP-PBX or gateway.
Media bypass can improve voice quality by reducing latency, needless translation,
possible packet loss, and the number of points of potential failure. Scalability can be
improved, because elimination of media processing for bypassed calls reduces the
load on the Mediation Server. This reduction in load complements the ability of the
Mediation Server to control multiple gateways.
Where a branch site without a Mediation Server is connected to a central site by one
or more WAN links with constrained bandwidth, media bypass lowers the bandwidth
requirement. It does this by allowing media from a client at a branch site to flow
directly to its local gateway without first having to flow across the WAN link to a
Mediation Server at the central site and back.
By relieving the Mediation Server from media processing, media bypass may also
reduce the number of Mediation Servers that an Enterprise Voice infrastructure
requires. For planning, do not count calls with media bypass.
Pool vs. Single Server

media bypass and DNS load balancing, a stand-alone Mediation Server pool is not
necessary. A stand-alone Mediation Server pool is not necessary because qualified
gateways are capable of DNS load balancing to a pool of Mediation Servers and they
can receive traffic from any Mediation Server in a pool.
However, using a pool of commonly configured Mediation Servers will greatly
improve throughput. The determining factor on pool versus single server is the IP-
PBX or gateways ability to support DNS load balancing. If it cannot support load
balancing, then separate servers with separately configured routing rules will be
required.
We also recommend that you collocate the Mediation Server on a front-end pool
when you have deployed IP-PBXs or connect to an ITSPs Session Border Controller
(SBC), as long as any of the following conditions are met:
The IP-PBX or SBC is configured to receive traffic from any Mediation Server in the
pool and can route traffic uniformly to all Mediation Servers in the pool.
The IP-PBX does not support media bypass, but the front-end pool, that is, hosting the
Mediation Server can handle voice transcoding for calls to which media bypass does
not apply.
You can use the Microsoft Skype for Business Server 2015, Planning Tool to evaluate
whether the front-end pool where you want to collocate the Mediation Server can
handle the load. If your environment cannot meet these requirements, then you
must deploy a stand-alone Mediation Server pool.
The main functions of the Mediation Server are as follows:
Encrypting and decrypting SRTP on the Skype for Business Server side
Translating SIP over TCP (for gateways that do not support TLS) to SIP over mutual TLS
Translating media streams between Skype for Business Server and the gateway peer
of the Mediation Server
Connecting clients that are outside the network to internal ICE components, which
enable media traversal of NAT and firewalls
Acting as an intermediary for call flows that a gateway does not support, such as calls
from remote workers on an Enterprise Voice client
In deployments that include SIP trunking, working with the SIP trunking service
provider to provide PSTN support, which eliminates the need for a PSTN gateway
Dependencies
The Mediation Server has the following dependencies:
Registrar required. The Registrar is the next hop for signaling in the Mediation Server
interactions with the Skype for Business Server 2015 network. Note that Mediation
Server can be collocated on a front-end server along with the Registrar, in addition to
being installed in a stand-alone pool consisting only of Mediation Servers. The
Registrar is collocated with a Mediation Server and PSTN gateway on a Survivable

Branch Appliance.
Monitoring Server. Optional but highly recommended. The Monitoring Server enables
the Mediation Server to record quality metrics associated with its media sessions.
Edge Server. Required for external user support. The Edge Server enables the
Mediation Server to interact with users who are located behind a NAT or firewall.
Topologies
The Skype for Business Server 2015 Mediation Server is, by default, collocated with
an instance of the Registrar on a Standard Edition server, a front-end pool, or
Survivable Branch Appliance. All Mediation Servers in a front-end pool must be
configured identically. Where performance is an issue, it may be preferable to
deploy one or more Mediation Servers in a dedicated stand-alone pool.
Alternatively, if you are deploying SIP trunking, we recommend that you deploy a
stand-alone Mediation Server pool.
media bypass and DNS load balancing, a stand-alone Mediation Server pool is not
necessary. A stand-alone Mediation Server pool is not necessary because qualified
gateways are capable of DNS load balancing to a pool of Mediation Servers and they
can receive traffic from any Mediation Server in a pool.
Media Bypass
Media bypass refers to removing the Mediation Server from the media path
whenever possible for calls whose signaling traverses the Mediation Server. Media
bypass can improve voice quality by reducing latency, needless translation, possible
packet loss, and the number of points of potential failure. Scalability can be
improved, because elimination of media processing for bypassed calls reduces the
load on the Mediation Server. This reduction in load complements the ability of the
Mediation Server to control multiple gateways.
Where a branch site without a Mediation Server is connected to a central site by one
or more WAN links with constrained bandwidth, media bypass lowers the bandwidth
requirement by allowing media from a client at a branch site to flow directly to its
local gateway without first having to flow across the WAN link to a Mediation Server
at the central site and back.
By relieving the Mediation Server from media processing, media bypass may also
reduce the number of Mediation Servers that an Enterprise Voice infrastructure
requires.
Media bypass is useful when you want to minimize the number of Mediation Servers
deployed. Typically, a Mediation Server pool will be deployed at a central site, and it
will control gateways at branch sites. Enabling media bypass allows media for PSTN
calls from clients at branch sites to flow directly through the gateways at those
sites. Skype for Business Server 2015 outbound call routes and Enterprise Voice
policies must be properly configured so that PSTN calls from clients at a branch site
are routed to the appropriate gateway.
Planning
After your Enterprise Voice structure is in place, planning for media bypass is
straightforward.
If you have a centralized topology without WAN links to branch sites, you can enable
global media bypass, because fine-tuned control is unnecessary.
If you have a distributed topology that consists of one or more network regions and
their affiliated branch sites, determine the following:
o Whether your Mediation Server peers are able to support the capabilities required
for media bypass
o Which sites in each network region are well-connected?
o Which combinations of media bypass and call admission control is appropriate for
your network?
Media Bypass Process

When you enable media bypass, a unique bypass ID is automatically generated for
a network region, and for all network sites without bandwidth constraints within that
region. Sites with bandwidth constraints within the region and sites connected to
the region over WAN links with bandwidth constraints are each assigned their own
unique bypass ID.
When a user makes a call to the PSTN, the Mediation Server compares the bypass
ID of the client subnet with the bypass ID of the gateway subnet. If the two bypass
IDs match, media bypass is used for the call. If the bypass IDs do not match, media
for the call must flow through the Mediation Server.
When a user receives a call from the PSTN, the users client compares its bypass ID
to that of the PSTN gateway. If the two bypass IDs match, media flows directly from
the gateway to the client, bypassing the Mediation Server.
Only Lync Server 2010 or above clients and devices support media bypass
interactions with a Mediation Server.
Media Bypass Modes

You must configure media bypass both globally and for each individual PSTN trunk.
When enabling media bypass globally, you have two choices: Always Bypass and
Use Site and Region Information.
Always Bypass. As the name suggests, Always Bypass means that bypass will be
attempted for all PSTN calls. Always Bypass is used for deployments where there is no
need to enable call admission control, nor is there a need to specify detailed
configuration information regarding when to attempt media bypass. Furthermore,
Always Bypass is used when there is full connectivity between clients and PSTN
gateways. In this configuration, all subnets are mapped to only one bypass ID, which
is computed by the system.
Use Site and Region Information.The bypass ID associated with site and region
configuration is used to make the bypass decision. This configuration provides the
flexibility to configure bypass for most common topologies, because it gives you fine-
grained control over when bypass happens, in addition to supporting interactions with
call admission control (CAC). The system tries to ease your task by automatically
assigning bypass IDs as follows.
o The system automatically assigns a single unique bypass ID to each region.
o Any site connected to a region over a WAN link without bandwidth constraints
inherits the same bypass ID as the region.
o A site associated with the region over a WAN link with constrained bandwidth is
assigned a different bypass ID from that of the region.
o Subnets associated with each site inherit the bypass ID for that site.
Location Based Routing

Location based routing is a new feature in Skype for Business Server 2015
Cumulative Update 1 (CU 1). While most routing logic is based on users SIP URL and
group membership, independent of the users current location, location based
routing overrides other routing and limits routing based the users current location.
If configured, location based routing supersedes user based routing policies.
Location is based on the subnet where the user is connected.
Some situations where location based routing may be used:
In India to protect local telephone company operators, a call within India can be
routed either through Internet or PSTN, but not both. Therefore, a caller using Skype
for Business to call a telephone that cannot be reach through the Internet or Intranet,
must enter the PSTN through a local gateway.
Some countries through regulations require all international calls enter the PSTN
within the country and therefore, the corporate WAN cannot be used for routing
international calls to avoid paying local tariffs. Location based routing can also be
used to limit the load placed on the corporate WAN by long distance traffic. Even
though routing a call to a different PSTN gateway may be cheaper. The additional load
on the WAN may interfere with more important traffic. In this case location based
routing can be used to force all external traffic to enter the PSTN at the closest
gateway.
Outbound example of Location Based Routing

Rajesh works in the Bangalore office of Adatum. He is visiting the Hyderabad office
for a few days to attend the Skype for Business User Group meeting. He connects to
the Adatum corporate intranet using wireless.
Upon arriving at the office, Rajesh calls his wife at home to let her know he arrived
safely. Since the home phone is reachable only through the PSTN, under Indian law
this call must enter the PSTN through the Hyderabad gateway, even though Rajesh is
based in Bangalore.
Rajesh calls his boss in Bangalore. This call can be routed over the corporate intranet,
because the call can be completed without entering the PSTN.
Location based routing policy would route all calls which do not match the internal
dial pattern or international dial pattern to be routed through the local gateway.
Inbound example of Location Based Routing

Adatum is holding a conference for customers all across India. The conference is
hosted on the Hyderabad Skype for Business pool. For customers calling into the
conference, Adatum must provide a dial-in number through the Hyderabad gateway.
A customer in Bangalore cannot call into a Bangalore number and be connected to a
conference using the Hyderabad Skype for Business Pool. A location based policy
can be implemented to block inter trunk routing of calls entering a gateway in India
from being routed to other locations in India, either IP or PSTN.
Configure Location Based Routing

To configure location based routing:
1 Create voice routing policy to associate the network site with the appropriate PSTN
usages. Make sure policy includes internal PBX users and does not include any
gateways which cannot be used from the site. In this example, users in Bangalore can
call PBX users in Bangalore and Hyderabad, and use only the Bangalore gateway to
connect to the PSTN. Similarly, Hyderabad users are limited to use the Hyderabad
gateway.
New-CsVoiceRoutingPolicy -Identity "BangaloreVoiceRoutingPolicy" -Name "Bangalore voice routing

policy" -PstnUsages@{add="Bangalore usage", "PBX BLR usage", "PBX HYD usage"}
New-CsVoiceRoutingPolicy -Identity "HyderabadVoiceRoutingPolicy" -Name " Hyderabad voice
routing policy" -PstnUsages@{add="Hyderabad usage", "PBX BLR usage", "PBX HYD usage"}
16 Configure location based routing for the applicable network sites and associate your
voice routing policies to them.
Set-CsNetworkSite -Identity "Bangalore" -EnableLocationBasedRouting $true -VoiceRoutingPolicy

"BangaloreVoiceRoutingPolicy"
Set-CsNetworkSite -Identity "Hyderabad" -EnableLocationBasedRouting $true -VoiceRoutingPolicy
"HyderabadVoiceRoutingPolicy"
17 Enable location based routing for trunks which you want to apply location based
routing restrictions to apply. Usually, this is done for gateways and not PBX users (The
trunk still needs to be defined for the PBX users.).
Set-CsTrunkConfiguration -Identity Service:PstnGateway:Trunk 1 BLR-GW -EnableLocationRestriction

$true -NetworkSiteID "Bangalore"
Set-CsTrunkConfiguration -Identity Service:PstnGateway:Trunk 2 HYD-GW
-EnableLocationRestriction $true -NetworkSiteID "Hyderabad"
18 Enable location based routing for users by enabling it in the voice policies.
Set-CsVoicePolicy -Identity "India voice policy" -PreventPSTNTollBypass $true
19 Globally enable location based routing to your routing configuration.
Set-CsRoutingConfiguration -EnableLocationBasedRouting $true

For more information on Location Based Routing see:

M:N Interworking Routing

A call connecting from Skype for Business Server 2015 must traverse a Mediation
Server and a gateway to get to the PSTN. PSTN connectivity can be achieved by
deploying multiple Mediation Servers connected to multiple gateways and
configuring more than one voice route for each user to access the PSTN.
Interworking RoutingHistory
Skype for Business Server 2015 greatly improved the mapping of Mediation Servers
(connection to the Skype for Business Pool) to Gateways (connection to the PSTN).
Lync Server 2010

Lync Server 2010 introduced the ability for multiple gateways to be connected to
the same mediation service.
Routes point to a gateway; the topology document is used to find an appropriate
mediation service, inserted into the routing path. Combined with media bypass, this
allows the mediation service to be collocated on the front-end server
Lync server 2013

In Lync server 2013, a trunk is defined as a combination of fully qualified domain
name (FQDN), Mediation Server SIP listening port, Gateway FQDN, and Gateway SIP
listening port. (Contrast this with Lync server 2010, where a route consisted of a list
of gateways.)
Skype for Business Server 2015

In Skype for Business Server 2015, a trunk is defined as a combination of fully
qualified domain name (FQDN), Mediation Server SIP listening port, Gateway FQDN,
and Gateway SIP listening port. (Contrast this with Lync server 2010, where a route
consisted of a list of gateways.)
This provides for:
Better resiliencyboth service and on-premises scenarios.
Better interworking with IP-PBXs for bypass.
TLS plus Secure Real-Time Transport Protocol (SRTP) for multiple SIP trunks to the
same SBC FQDN.When Outbound Routing matches a dialed PSTN number to a route,
the route will consist of a list of trunks.
Trunk and IP-PBX Interworking

Skype for Business Server 2015 introduces the concept of trunks between the
Mediation Server and PSTN gateways.
A trunk is a logical association between a Mediation Server and a TCP or TSL port on
the gateway. Multiple trunks between one Mediation Server and a gateway can be
used to configure multiple voice routes through the same gateway with different
characteristics (e.g Representative Media IP is a per-trunk parameter.). Multiple
trunks between a Mediation Server and PSTN gateway can be defined to represent
IP-PBX SIP termination. In other words, separate trunks can be used for calls
destined for phones on the PBX system and the PSTN when the PBX is used as a
PSTN gateway.
Separate trunks are defined for inbound and outbound calls. Each trunk will be
associated with the appropriate route for outbound calls from Mediation Server to
gateway. For inbound calls, per-trunk policy will be applied.
Trunk configuration will be scoped globally or per trunk; similarly, dial plan can be
scoped per trunk.
Trunk and IP-PBX InterworkingReal Life

Typical PBX deployment:
Central Call Control (PBX01)
Decentralized Media Termination Points (MTPs/Gateways)

Technical requirements:
Use MTPs on same site as Skype for Business Client; keep media local.
Enable media bypass.

Deployment and configuration:
2 Define PBX01 as the PSTN gateway:
o Use MTP1 as an alternate media IP address.
o The first trunk is created automatically.
20 Add additional trunks for remaining MTPs:
o Use different PBX listening ports.
21 Use PowerShell to define RepresentativeMediaIP for each MTP/trunk.

Notice when more than one trunk is defined between a Mediation Server and a PBX,
different ports must be used on the PBX, while the same port can be used on the
Mediation Server. Multiple ports can be configured on a Mediation Server for
interoperability with different PBX systems.
Configuration Details
Topology Builder
After you have decided on your deployment plan, you use Topology Builder to begin
deploying. When finished, you use Topology Builder to validate the topology, and
then, if it passes, you can publish the topology. When you publish the topology,
Skype for Business Server puts the topology into the Central Management Store,
which is created at this time if it does not already exist. When you install Skype for
Business Server on each server in your deployment, the server uses the topology
from the Central Management Store and gets installed based on its role in your
deployment.
Alternatively, if you are very familiar with Skype for Business Server and need less
prescriptive guidance, you can skip the Planning Tool and use the wizards in
Topology Builder for the initial design of your deployment, and for the validation and
publishing steps.
Using Topology Builder to plan and publish a topology is a necessary step. You
cannot bypass Topology Builder and install Skype for Business Server individually on
the servers in your deployment. Each server must use the topology from a
validated, published topology in the Central Management Store.
Windows PowerShell:
Identify the Trunk IDs, for example, Get-CsTrunk | FT Identity.
Use Windows PowerShell to configure Media IP Addresses for the remaining trunks.
o For example, Set-CsPstnGateway -Identity PstnGateway:PBX-10-MTP2.tailspin.local

-RepresentativeMediaIP "10.10.10.11
Verify the Media IP address for the trunks.
o For example, Get-CsTrunk | FT Identity, RepresentativeMediaIP.
Trunks and Resiliency

For PSTN routing resiliency Skype for Business Server 2015 should include multiple
Mediation Servers connected to one or more gateways. The gateways could be in
the same location to provide fault tolerance and increased capacity, or in different
locations to provide alternate voice routes to the PSTN. A single Mediation server
connected to a qualified gateway supports about 1000 sessions.
The example on the slide shows two Mediation Servers connected to two gateways.
Because there is no trunk from Mediation Server MS1 to gateway GW2, Mediation
Server MS1 does not provide back for Mediation Server. Any call with a voice route
using gateway GW2 will not connect if Mediation Server MS2.
Multiple Sites to the Same Service Provider

Skype for Business Server 2015:
Virtual gateways must be defined to allow connectivity from multiple Mediation Server
pools to the same SBC FQDN.
Virtual gateway FQDNs all resolve to the same IP address.
TLS cannot be used because the SBC certificate does not contain the virtual
gateways name.
Gateway-specific inbound policies cannot be applied when virtual gateways are used
(RNL of the IP-address does not resolve to virtual gateway).
Separates PSTN gateways and trunks.
Enables you to connect multiple trunks to one gateway.
Enables the use of TLS.
Allows for gateway-specific inbound policies.
M:N Interworking InterworkingTrunk Definition

A trunk is a combination of:
A Mediation Server FQDN
A Mediation Server SIP listening port
A Gateway FQDN
A Gateway SIP listening port

This provides for:
Better interworking with IP-PBXs for bypass.
TLS plus SRTP capability for multiple SIP trunks to the same SBC FQDN.
When Outbound Routing matches a dialed PSTN number to a route, the route will
consist of a list of trunks.
Contrast this with Lync Server 2010, where a route consisted of a list of gateways.
Auxiliary Calling Information

Mediation sends the following to SIP Peer:
P-Asserted-Identity (PAI)
History-Info
Referred-By (always on, cannot be disabled)

Relevant to billing:
Parameters show who forwarded calls to

PSTN for appropriate chargeback
Some carriers refuse calls that do not show clear egress from their network
Trunk configuration allows enabling and disabling forwarding.
Simultaneous Ringing Scenario:

3 Bob has enabled simultaneous ring to a PSTN phone.
22 An incoming call is forwarded to the external destination.
23 The from: field presents the original incoming number.
24 This enables Bob to see who is calling.
25 The History-Info and P-Asserted-Identity fields provide the target system information
for billing to the user who initiated the call (Bob).
Call Routing Reliability

Call routing reliability addresses maintain outward and inward voice traffic when a
Mediation Server or gateway is unavailable. Voice reliability needs to address
signaling, so calls can continue to be made and received, as well as media, existing
calls will not be dropped if a mediation server is down.
Fast Failover and Options Polling

Challenges with Lync Server 2010:
Rerouting of calls around failed infrastructure should be improved.
Resiliency from the Mediation Server to its next-hop proxy is necessary to improve
overall resiliency.
Outbound Routing (OBR) may take up to 20 minutes to realize that a Mediation Server
pool that was previously marked as down is back in service.
TCP socket timeout involved when the Mediation Server determines gateway failure
can lead to 40+ seconds delay time.
Solution with Skype for Business Server 2015:
Once every minute an OPTIONS poll is sent from the Mediation Server on each trunk.
If five consecutive OPTIONS polls have failed for a trunk, and an INVITE is received
from the front-end server destined for that trunk, the Mediation Server will not
attempt to send the INVITE to the gateway (as was the case in Lync Server 2010).
Call-Routing ReliabilityLost Connection

Voice Policy has only GW1 and GW3 (in that exact order) as options for the route:
The Mediation Servers poll the associated gateways every minute.
GW1 and GW2 do not respond to polls from MS-01.
GW1 and GW3 respond correctly to MS-02.
Response code 503informs the front-end server that no gateways are reachable from
MS-01.
SIPStack now knows that the Mediation Server has lost connectivity to all gateways
and triggers load balancing.
Front-end server tries the next Mediation Server to GW1.
Response code 503instructs the Front-End Server to retry other Mediation Servers in
the pool for the call.
After OBR has marked a Mediation Server pool as down, it starts sending options polls
to the Mediation Server pool so that it can quickly determine when the pool can be
used again for service calls.
The Mediation Server only responds with 200 OK if it can communicate with at least
one gateway.
Otherwise, it responds with response code 50x, which will not result in OBR marking
the Mediation Server pool as back in service.
Call-Routing ReliabilityGateway Down

Voice Policy has only GW1 and GW3(in that exact order) as options for the route:
The Mediation Servers poll the associated gateways every minute.
GW1 does not respond to polls from MS-01.
GW2 and GW3 respond correctly.
Response code 504 informs the Front-End Server that GW1 is down.
The gateway is marked as down.
The Front-End Server tries the next gateway available(enforced by policy).
Response code 504 instructsthe Front-End Server that the trunk is down and that a
different trunk should be tried.
Call-Routing Reliability and Retries

Lync Server 2010:
For non-bypass calls, the Mediation Server automatically sends a 183 response to OBR
in Lync Server 2010.
This results in the timer being immediately stopped, even though the gateway never
answers with an 18x; the net result is that misconfigured and/or nonfunctioning
gateways are masked, and OBR recovery to alternate routes is thwarted.

Outbound Routing (OBR) in the Front-End Server has a 10-second timer that it keeps
for every invite it sends to the Mediation Server.
If an 18x response is not received for the call within this 10-second interval, OBR tries
the next element in its routing algorithm.
Auto-generated response code 183 from the Mediation Server to OBR will not cause
the OBR timer to stop.
Timer can be disabled through Windows PowerShell; duration is fixed to 10 seconds.

Call-Routing ReliabilityNext-Hop Proxy

The Mediation Server tracks its next-hop proxy and backup next-hop proxy by
sending out periodic options polls:
Backup next-hop proxy is defined by pool pairing.
If the primary next-hop proxy is found to be down (failure to answer to five options
polls in a row), new invites from gateways are sent to the backup next-hop proxy.
Additionally, a 10-second timer is used for incoming calls, so if the primary next-hop
proxy is used for a call and no SIP response is received within this time, the call is
rerouted to the backup next-hop proxy.
Voice Routing Coexistence

The Mediation Server is the connection between the gateways and your Skype for
Business pools. As your company moves from earlier versions of Skype for Business
Server and Office Communicator, you need to be aware of the coexistence
limitations. The home server is the Front End Pool which hosts a user, while the
next-hop server is the next Skype for Business Server pool which the mediation
server connects. The next-hop server receives the incoming call and determines
which front end pool or interworking route to route the call to.
Outbound Calls
For outbound calls all pairings of Skype for Business Server 2015, Lync Server 2010
and Lync server 2010 are allowed, except hosting users on a Lync server 2010 Front
End Pool with a Skype for Business Server 2015 Mediation Server. Under no
circumstances is an environment with all three releases supported. The next-hop
server concept does not apply to outbound call, because the callers home pool
handles the routing.
Inbound Calls
For inbound calls the limitation is the next-hop server must be the same as the
Mediation Server. After that the recipients home pool can be Skype for Business
Server 2015, Lync Server 2013 and Lync Server 2010.
Survivable Branch Appliances (SBA)

Because bundle the Front End Server and Mediation Server in the same package
special limitations apply:
A Lync Server 2013 Survivable Branch Appliance (SBA) can be used with either a Lync
Server 2013 pool or a Skype for Business2015 pool.
When used with a Skype for Business2015 pool, the Lync Server 2013 SBA will still
write monitoring and archiving content to the Lync Server 2013monitoring store.
A Skype for Business2015 SBA only works with a Skype for Business2015 pool.
Call Via Work

Call Via Work is a new feature in Skype for Business Server which enables you to
integrate your Skype for Business solution with your existing PBX phone systems. A
user enabled for Call Via Work can click in Skype for Business to call another user,
either within your deployment or an external user. The call is completed using the
user's PBX phone. This enables a user with a PBX phone to include audio in their
rich Skype for Business conversations.
Call Via Work enables the following for PBX phone users:
Click-to-call experience, with the audio provided through the PBX phone.
Presence, user search, and IM integration-- for example, two Call Via Work users in an
IM session can add audio to their session, with the audio provided through the PBX
phones.
The ability to add IM, application sharing, and file transfer to a Call Via Work call.
One-click meeting join capability
Components and Call Flow

This shows the steps during a Call Via Work call and highlights the involved
components in a call via work scenario.First the caller clicks to call someone in the
Skype for Business client; then the UCWA rings the caller's phone. When the caller
picks up the phone, the recipient is called:
26 A user uses his Skype for Business to dial a number or uses the click-to-dial feature to
start a PSTN call, the Skype for Business client talks to the UCWA component on the
Skype for Business server to setup this call.
27 The Skype for Business Server sets up a call to the PBX and dials the users PBX
station number
28 The PBX rings the end user on this PBX station, the user picks up the call
29 Once the call is connected, Skype for Business will initiate a second call to the dialed
number
30 The PBX routes the call to the PSTN
31 The destination picks up the call, the PBX station is connected to the destination.
User experience
This is the user experience if a Call via Work is used to place a call. The screenshots
show the sequence for a user dialing a PSTN number.
4 The user dials a number from his Skype for Business client
5 The conversation window opens in the calling user interface
6 The outbound call to the PBX station is placed
7 The end user answers the call on his PBX station
8 A ring-back tone is played to the PBX station, Skype for Business Server places a call
to the destination number
9 The destination picks up the call, the call is connected and the users can talk.
10 The conversation window updates to connected
Mid-call control
The call is being performed using the PBX station as the audio endpoint, this
transfers most of the mid-call control options to the PBX station as well. E.g. if you
want to place a call on hold, you need to perform this using the capabilities that
your PBX station provides you with to accomplish this.
The Conversation Windows enables the user to end the call, this is really the only
call control option that Skype for Business can perform in this scenario.The call
timer feature is not available in a call-via-work scenario.
Adding Modalities
Call via Work supports adding modalities to a call if the called person is a Lync or Skype
for Business user. Pick a Skype for Business user from your contact list and place a call to
them. Call via Work will call you on your PBX Station to establish the call. Once the Skype
for Business user accepts the call you can add additional modalities to this conversation.
Supported modalities are
Instant Messaging (IM)
Application sharing
Desktop Sharing
File Transfer
Please note that adding Video is not supported in a call via work scenario.
You also cannot escalate an existing Call via Work call to a conference or add
conferencing workloads like:
PowerPoint
Whiteboard
Q&A
Poll
In order to escalate to a conference, you need to start the conference first and have
Call via Work join you in that conference.
Multiple Calls and remote participant activities

Once the 2nd call is established he will lose control over his 1 st Call via Work
conversation in thy Skype for Business client. He needs to perform all call control
functions, like hang up, on his PBX station.
We will warn the user before he accepts or places a 2 nd call while he is already in a
Call via Work conversation.
If the remote participant performs any call control functions on his end Skype for
Business has no knowledge about that and cannot inform the user about this. The
remote user can place the call on hold or transfer it to a different person but the
Skype for Business client will not show this change in the conversation window.
End a call
The user can decide to hang up the call on his PBX station, press the hang up
button in the conversation window or close the conversation window by pressing the
x.
Conversation history
The conversation history works as expected, the user can see all his calls in the
conversation history. Call via Work will only show the second part of the call, the
first call leg to his PBX station will not be shown in the conversation history window.
Avoid misleading missed call notification for inbound calls by configuring your PBX
to send a call completed elsewhere reason header in the CANCEL message. This
allows Skype for Business that the call has been answered on a different endpoint
and has not been missed.
Meetings
Call via Work supports fully supports all meeting scenarios.
Users that are configured for Call via Work will be asked which endpoint should be
used when they enter a meeting. Call via Work will auto-populate the Call me at:
section with the number that has been configured to be the audio endpoint in a Call
via Work scenario.
This works for all conferencing scenarios:
Scheduled conferences
Ad-hoc meetings
Ad-hoc incoming group calls
The conference will call the user and he will be added to the conference using his PBX
station as an audio device. All remaining conferencing modalities will stay in the Skype for
Business client.
Inbound Calls
Remember that Call via Work is a feature to support outbound calls only, Call via Work
cannot handle inbound calls.
Setup simultaneous ring in your Skype for Business client to handle inbound calls and
point this to you PBX handset station number. Any inbound calls will be forked to this
number and you can answer the call on this station.
Consider the placement of your PBX and the Skype for Business PSTN gateway. If your
PBX is the first component in the line you might not receive any inbound calls on Skype
for Business, thus all calls will land on the PBX station only.
Presence
Skype for Business will update the presence to in a call or in a conference call
whenever Skype for Business has knowledge of the call. This is the case if any
Skype for Business component is part of the call setup which may not be the case in
any scenario. Review these scenarios for more details.
Outbound call placed through Call via Work

Skype for Business has initiated this call and has knowledge of this call, the presence
will be updated accordingly
Outbound Meet Now / Group Call

will be updated accordingly
Inbound call answered on the PBX

This call does not touch any Skype for Business component but is visible to the PBX
only. Skype for Business has no knowledge of this call and cannot update your
presence, your presence is unchanged.
Inbound call answered on Skype

Skype for Business has knowledge of this call, the presence will be updated
accordingly.
Inbound Meet Now / Group Call

will be updated accordingly.
Prerequisites for Call Via Work

To enable any users for Call Via Work, you must have some pre-requisites in place.
For more information on these prerequisites, and for steps on how to enable users
for Call Via Work, see Deploy Call Via Work in Skype for Business Server 2015.
Call Via Work uses Unified Communications Web API (UCWA), which is automatically
installed on all Skype for Business Server Front End Servers. To enable users for Call
Via Work, you must also have the following prerequisites in place:
You must have a Mediation Server deployed, either as part of a Front End Server or
as a standalone role. You must also deploy an IP-PBX gateway.
All users who will be enabled for Call Via Work must have a Direct Inward Dialing
(DID) on the PBX phone system.
You must enable all Call Via Work users for Enterprise Voice. When you do this, you
must configure the Skype for Business DID number for each user to the
corresponding DID number for the corresponding PBX phone system.
All users who will be using Call Via Work must have Automatic Configuration
selected in their Advanced Connections option in their Skype for Business client.
This enables the client to discover the UCWA URLs. Automatic Configuration is the
default selection.
For each Call Via Work user, enable call forwarding and simultaneous ringing.
For each Call Via Work user, ensure that dial-in conferencing and conferencing
dial-out are enabled. This enables these users to get into and out of Skype for
Business conferences.
Ensure that delegation, team call, and response group are disabled for every Call
Via Work user.
Configuration
Call via Work is enabled through a Call via Work policy. To create a new policy, use
the new-csCallviaWorkPolicy cmdlet:
New-CsCallViaWorkPolicy Identity EnableCallViaWork Enabled $true
This policy enabled the user for Call via work and allows the user to select his PBX
station number. You may also create a policy that specifies the PBX station number
for the user. Use the UseAdminCallBackNumber option with the
AdminCallbackNumber option to define the number for the user.
New-CsCallViaWorkPolicy Identity EnableCallViaWork Enabled $true

UseAdminCallbackNumber $true AdminCallbackNumber"+483000"
Please be aware that you may need to specify a new policy for every user if
you want to use a unique AdminCallbackNumber for every user.
Apply the policy to a user using the grant-csCallviaWorkPolicy cmdlet:
Grant-CsCallViaWorkpolicy Identity EnableCallViawork Identity Amy@contoso.com
Applying the policy to a user allows the user to use the feature. The user now can
go into the Call handling section in the settings and enable Call via Work, it is not
turned on by applying the policy.
Limitations of Call Via Work

Call Via Work is a voice solution that requires little hardware setup, but has
limitations compared to the features available in full Enterprise Voice or remote call
control. Call Via Work has the following limitations:
If a Call Via Work user has set up call forwarding to the Call Via Work callback
number, and someone tries to invite this user to a meeting by the user's phone
number, the invitation will not reach the user. You should educate your users to
invite participants to meetings by clicking the name, not the phone number.
Enhanced 911 capability and malicious call tracing are not available during Call Via
Work calls.
Users enabled for Call Via Work cannot use the delegation, team call, or response
group features.
Users of Call Via Work cannot use Skype for Business to record a meeting, mute or
unmute the call, hold or transfer the call, or use call park.
Users cannot use Call Via Work to access their PBX voicemail messages.
Users of Call Via Work cannot escalate a session that started as a voice call to a
collaborative meeting that includes communications such as video, Powerpoint,
whiteboard, or One Note.
Users of Call Via Work cannot add more users to a 2-person call.
No support for deskphone pairing or VDI plugin pairing.
If a user answers a makes or answers a call using the PBX phone (and not using the
Skype for Business window), there will be no log of the call.
If your PBX system does not support REFER with Replaces, the following behavior
will happen. While on a Call Via Work call, if the user transfers the ongoing call from
the PBX Phone, the call window will not disappear from their Skype for Business
window. If the user then closes the call window, the call between the transfer target
and the transferee will end.
Voice Routing
When a Skype for Business user places a call, Skype for Business Server 2015 goes
through a sequence of steps to normalize the number, route the call and authorize
the call, as follows:
32 The user initiates a call from the client. This can be done by:
f. Entering a SIP URI that will bypass most of the logic and go directly to Inbound
Routing.
g. Using a Phone Number to invoke processes such as normalization and reverse

number lookup.
33 First, a check for an emergency number is performed. The check is done before
normalization to minimize delays.
b If the number qualifies as an emergency number, the Location Policy for

emergency numbers is applied and routing continues.
h. If the number does not qualify as an emergency number, the process continues.
34 Next, the process checks to determine whether the number is a global number,
according to RFC 3966 Global numbers starts with +. However, this does not mean
the number is in the E.164 format. Number from the Skype for Business contact list
should be as Global numbers.
c A Global number does not need further normalization.
i. A Non-Global number needs to be normalized.
35 A Non-Global number is normalized to an E.164 format as recommended.

Normalization is done by applying normalization rules defined as part of a Dial Plan.
Dial plans are assigned to users or groups of users based on the site or pool.
36 Has the user entered a valid number?At least one normalization rule must match.
37 Therefore, the process must also check to determine if a number is within a call park
orbit, in addition to the normalization rules.
38 Is there a Skype for Business user with this number?After the number is translated
to a global unique format (normalized to E.164), a Reverse Number Lookup (RNL) is
performed by comparing the number against the msRTCSIP-Line and msRTCSIP-
PrivateLine attributes in Active Directory.
39 If the RNL matches, outbound routing has completed processing for numbers that
match Skype for Business users. The call is transferred to inbound routing for further
processing:
d If there is a network outage or Call Access Control (CAC) network usage limits are
reached, Fallback PSTN usage may return continue processing at the next step.
j. Apply Called Party preferences, such as multiple endpoints, forwarding and

simultaneous ring which will dial additional numbers.
40 If there is no match for RNL, the process checks to determine if the number is in the
unassigned number range or Call Park Orbit range.
41 The call is then transferred to the Announcement or Call Park service. Announcement
and Call Park services are covered in the Voice Applications module.
42 Then, the Voice Policyfor the user is checked for PSTN usages to find a route for the
call. Every user is assigned a Voice Policy.The Voice Policy defines the PSTN usages for
the user and the voice features that are available for the user.
43 If possible, a route is selected which matches the selection criteria for the number and
matches the PSTN usage for the user.
44 Based on the selected route a gateway to Public Switched Telephone Network (PSTN)
or route to a Private Branch Exchange (PBX) is determined. Based on the gateway a
MediationServer and trunkare selected then used to complete the routing to the PSTN
or PBX.
45 The PSTN further routes the call to the intended number along with a tell-tale ring
being heard in the speaker of the original client.
Number Normalization and E.164

E.164 is an ITU-T recommendation that defines the international public
telecommunication numbering plan used in the Public Switched Telephone Network.
E.164 numbers are globally unique.
E.164 is the recommended numbering standard to use for Skype for Business.
Skype for Business fully supports E.164; most PBXs do not.
There is no reason to not use E.164.
E.164 simplifies call routing.
Numbers can be translated between Skype for Business and the PBX.
An E.164 number can be up to 15 digits. The first 1 to 3 digits are the country code,
where the first digit is the region code. For example, Europe is covered by regions 3
and 4. The length of the country code was set based on anticipated call volume
(shorter numbers reduce wear on mechanical switches) and number of subscribers
(larger number of subscribers means longer numbers required for the subscriber
number). North America was assigned country code, 1. The countries in North
America and the Caribbean already had an existing agreement to jointly handle
direct dial international calls within the region. 1 was already used as the standard
access number for long distance. In other regions, country codes are 2 or 3 digits.
The format of the rest of the number is up to each country. This includes the length
of the phone number. Most countries implement National Destination Codes (NDC)
as the first level of routing. Examples of NDC include the North American three-digit
area code, and Australian digit region or state code.
Note: The way people traditional write phone numbers has to deal with how
calls are routed. For example, in the United States a telephone branch circuit could
originally handle 10,000 subscribers and phone numbers were allocated in blocks
of 10,000, so phone numbers are as 1-555-555-5555 (<long distance access
code>-<3 digit area code>-<3 digit branch code>-<1 of 10,000 numbers>).
Most PBX do not use E164 formatted to call out to the PSTN or within the PBX. The
PBX needs to dial in a format recognized by the PSTN. The telephone companies
implemented shortened formats to reduce wear on equipment. For example, in
most countries to make a call within the country the caller dials a 1 digit access
code instead of a 3 digit country code, or when dialing a number within the same
branch, only the 4 digit subscriber number needs to be dialed.
Configuring Enterprise Voice
Scoping Configuration Items and Policies

Voice Policies are configured at four levels:
Global policy.This is the default voice policy that is installed with the product. You can
edit the global voice policy to meet the specific needs of your organization, but you
cannot rename or delete it. This policy applies to all Enterprise Voice users, groups,
and contact objects in your deployment unless you configure and assign a voice policy
with more specific scope. If you want to disable this policy entirely, ensure that all
sites and users have custom policies assigned to them.
Site policy.This policy applies to an entire site, except for any users, groups, or
contact objects that are assigned a specific user policy. To define a site voice policy,
you must specify the site to which the policy applies. If a user voice policy is not
assigned, the site policy is used.
Pool policy. This policy applies to users assigned to a particular Registrar pool. A site
can have one or more Registrar pools to support Enterprise Voice. Not all policy types
can be assigned at the pool level.
User policy.This policy can be assigned to individual users, groups, or contact objects.
This is the lowest level of policy. User policies can be deployed to enable features for
certain users or groups at a site, but not for others in the same site. For example, you
may want to disable long distance dialing for some employees. For the purpose of
assigning a policy, a contact object is treated as an individual user.
When applying policies to users, policies are not merged, and only the most specific
policy available is applied. Users can be assigned to a specific policy, or set to
<Automatic>. When set to automatic, the site policy, if available, is applied;

otherwise, the global applies.
Policies are configured and assigned by using either Skype for Business Server
Control Panel or Skype for Business Server PowerShell cmdlets.
FIGURE 2.1: SETTING USER POLICY TO AUTOMATIC
Note: Not every level is available for every type of policy. For example, voice
policies cannot be set at the pool level.
Dial Plans
A dial plan is a named set of normalization rules that translates phone numbers for
a named location, individual user, or contact object into a single standard (E.164)
format for purposes of phone authorization and call routing.
Normalization rules define how phone numbers, expressed in various formats, are to
be routed for each specified location, user, or contact object. The same dial string
may be interpreted and translated differently, depending on the location from which
it is dialed, and the person or contact object making the call.
Dial Plan Scope

A dial plan scope determines the hierarchical level at which the dial plan can be
applied. In Skype for Business Server, a user can be assigned a specific per-user dial
plan. If a user dial plan is not assigned, the Registrar pool dial plan is applied. If
there is no Registrar pool dial plan, the site dial plan is applied. Finally, if there is no
other dial plan applicable to the user, the global dial plan is applied. Dial plan scope
levels are defined as follows:
User dial plan. Can be assigned to individual users, groups, or contact objects. Voice
applications can look up a per-user dial plan when a call is received with the phone-
context set to user-default. For the purpose of assigning a dial plan, a contact object is
treated as an individual user.
Pool dial plan. Can be created at the service level for any PSTN gateway or Registrar
in your topology. To define a pool dial plan, you must specify the particular service
(PSTN gateway or Registrar pool) to which the dial plan applies.
Site dial plan. Can be created for an entire site, except for any users, groups, or
contact objects that are assigned a pool dial plan or user dial plan. To define a site dial
plan, you must specify the site to which the dial plan applies.
Global dial plan. The default dial plan installed with the product. You can edit the
global dial plan, but you cannot delete it. This dial plan applies to all Enterprise Voice
users, groups, and contact objects in your deployment, unless you configure and
assign a dial plan with a more specific scope.
As the administrator, you can manage and assign dial plan scope levels by using
Skype for Business Server Control Panel.
Normalization and Regular Expressions

Normalization Rules
Normalization rules define how phone numbers expressed in various formats are to
be routed for the named location. The same number string may be interpreted and
translated differently, depending on the location from where it is dialed.
Normalization rules are necessary for call routing because users can, and do, use
various formats when entering phone numbers in their Contacts lists.
Normalizing user-supplied phone numbers provides a consistent format that
facilitates the following tasks:
Match a dialed number to the intended recipients SIP-URI.
Apply dialing authorization rules to the calling party.

The following number groups are among those that your normalization rules may
need to account for:
International access code
Long distance access code
Local access code
Country code
Area code
Length of extension
Site prefix
Creating Normalization Rules

Normalization rules use .NET Framework regular expressions to specify numeric
match patterns that the server uses to translate dial strings to E.164 format for the
purpose of performing reverse number lookup. You create normalization rules in the
Skype for Business Server Control Panel, either by entering the expressions
manually, or by entering the starting digits and the length of the dial strings to be
matched, and then letting the Skype for Business Server Control Panel generate the
corresponding regular expression for you. Either way, when you finish, you can
enter a test number to verify that the normalization rule works as expected.
Note: For details about using .NET Framework regular expressions, see ".NET
Framework Regular Expressions" at http://go.microsoft.com/fwlink/p/?
linkId=140927.
Address Book Normalization

Address Book Server
Skype for Business Server requires standardized RFC 3966/E.164 phone numbers,
but in most cases, when synchronizing with external address books such as Active
Directory, the external address book does not use Address Book normalization to
format incoming phone numbers into E.164 format. It does not use Dial Plans.
To use phone numbers that are unstructured or inconsistently formatted, Skype for
Business Server relies on the Address Book Server to preprocess phone numbers
before they are handed off to the normalization rules. In Lync Server 2010, the
normalization rules are applied after the number is received from the Address Book
Service. The Address Book Service removes any white space and non-mandatory
characters from the phone number before applying the normalization rule. When a
phone number is used from the address book and the normalization rule is applied,
clients such as Skype for Business Phone Edition and Skype for Business Mobile can
use these normalized numbers.
Note: Examples of non-mandatory characters include: !, @, ., -, and

*. Parentheses are not removed because they are used to designate local access
code for E.123 formatted numbers.
Skype for Business 2015 brings a new set of Powershell commands for controlling
Address Book Normalization rules. In previous versions of Lync, these rules were
configured in the Company_Phone_Number_Normalization_Rules.txt file that was
stored in the address book storage on the Lync share. This previous method was not
particularly intuitive and prone to issues because of the text file format used.
The new Skype for Business address book normalizationPowershell commands offer
a way to import AddressbookNormalization Rule files. The command is listed below
and will import the Pattern and Translation Rule directly into the new Skype for
Business commands. In doing so, the import process will create a random GUID to
be used as a unique name for each normalization rule.
Import-CSCompanyPhoneNormalizationRules FileName
Company_Phone_Number_Normalization_Rules.txt Identity Global
To see the result use the following command

Get_CsAddressBookNormalizationRule.
Voice Policies
Overview of Routing and Authorization

When a call is placed Skype for Business normalizes the phone number to E.164
format and attempts to match to a Skype for Business user using the SIP URI. If the
number does not match, outbound call routing logic is applied. Outbound call
routing applies to calls that are destined for the PSTN through a gateway or trunk,
or to an internal user on a PBX. Outbound routing configuration includes:
Dial Planis a named set of normalization rules that translates phone numbers for a
named location, individual user, or contact object into a single standard (E.164)
format for purposes of phone authorization and call routing. A set of normalization
rules associated with a particular location constitutes a dial plan.
Normalization ruledefine how phone numbers expressed in various formats are to be

routed for each specified location, user, or contact object. The same dial string may
be interpreted and translated differently, depending on the location from which it is
dialed, the person that makes the call and the order of the rules.
Voice policy associates one or more PSTN usage records with users, group or contacts.
A voice policy also provides a list of calling features that you can enable or disable.
PSTN usage recordspecifies a class of call (such as internal, local, or long distance)
that can be made by various users, or groups of users, in an organization.
Voice routeassociates destination phone numbers with particular trunks and PSTN
usage records. A PSTN gateway is considered a trunk.
In order to make a call, a call route must exist from the caller to the recipient, the
caller must be associated with the route by a PSTN usage record and the caller must
be authorized by for the level of service necessary to complete the call.
There may be multiple potential routes for a call, but the first matching available
route will be used. For example, Mary is an Adatum Corporation employee in
Redmond who needs to call a customer in New York. She is not authorized to make
long distance calls, but the call can still be completed as a local call through the
New York office by Skype for Business Server.
Voice Policies
Creating Voice Policies:
11 Log on to the computer as a member of the RTCUniversalServerAdmins group, or as a

member of the CsVoiceAdministrator, CsServerAdministrator, or CsAdministrator role.
For details, see Delegate Setup Permissions.
46 Open a browser window, and then enter the Admin URL to open the Skype for
Business Server Control Panel. For details about the different methods you can use to
start Skype for Business Server Control Panel, see Open Skype for Business Server
Administrative Tools.
47 On the left navigation bar, click Voice Routing, and then click Voice Policy.
48 On the Voice Policy page, click New, and then select a scope for the new policy:
e Site policy applies to an entire site, except any users or groups that are assigned
to a user policy. If you select Site for a policy scope, choose the site from the
Select a Site dialog box. If a voice policy has already been created for a site, the
site does not appear in the Select a Site dialog box.
k. User policy can be applied to specified users or groups.
49 If the voice policy scope is User, enter a descriptive name for the policy in the Name
field.
50 (Optional) Enter additional descriptive information for the voice policy.
51 Select or clear the following check boxes to enable or disable each of the Calling
features for this voice policy.
f Voice mail escape prevents calls from being immediately routed to the users
mobile phone voice mail system when phone is off or out of range.
l. Call forwarding enables users to forward calls to other phones and client devices.
Skype for Business Server 2015 provides a significantly wider range of
configuration options for call forwarding by applying special voice policies.
m. Delegation enables users to specify other users to send and receive calls on their
behalf using simultaneous ring. This feature is enabled by default.
n. Call transfer enables users to transfer calls to other users. This feature is enabled
by default.
o. Call park enables users to park calls on hold and then pick up the call from a
different phone or client. This feature is disabled by default.
p. Simultaneous ringing enables incoming calls to ring on additional phones, such as

a mobile phone.Skype for Business Server 2015 provides a significantly wider
range of configuration options for simultaneous ringing. This feature is enabled by
default.
q. Team call enables users on a defined team to answer calls for other members of
the team. This feature is enabled by default.
r. PSTN re-route allows calls that would normally be routed over the enterprise Wide
Area Network (WAN) to be routed through the PSTN when the WAN is congested.
This feature is enabled by default.
s. Bandwidth policy override enables administrators to override call admission

control policy decisions for a particular user. Disabled by default.
t. Malicious call tracing enables users to report malicious calls (such as bomb
threats), which in turn flags the calls in the call detail records (CDRs). This feature
is disabled by default.
52 Associate PSTN usage records with this voice policy. Arrange the PSTN usage records
for optimum performance. A call will route using the first policy in the list that pattern
matches the normalized number. To change a records position in the list, highlight the
record name, and then click the up or down arrow.
53 Associate PSTN usage records for call forwarding and simultaneous ringing with this
voice policy. Different PSTN usage policies can be applied to call forwarding and
simultaneous ringing. For example, you may want to allow users to make international
calls, but you do not want them to be able to simultaneously ring or forward a call to
an international number.
54 (Optional) Enter a number to test the voice policy and click Go. The test results are
displayed under Translated number to test.
55 Click OK.
56 On the Voice Policy page, click Commit, and then click Commit all.
Managed Features
Call forwarding enables users to forward calls to other phones and client devices.
Skype for Business Server 2015 provides a significantly wider range of configuration
options for call forwarding. For example, if an organization does not want to allow
incoming calls to be forwarded externally to the PSTN, an administrator can apply a
special voice policy to deploy this restriction. It is enabled by default.
Delegation enables users to specify other users to send and receive calls on their
behalf. In Skype for Business Server 2015, a delegate can configure simultaneous
ringing that enables incoming calls to his or her manager to ring all of the delegates
simultaneous ringing targets. This provides the delegate with greater flexibility in
responding to calls directed to the manager. It is enabled by default.
Call transfer enables users to transfer calls to other users. It is enabled by default.
Call park enables users to park calls on hold and then pick up the call from a different
phone or client. It is disabled by default.
Simultaneous ringing enables incoming calls to ring on additional phones (for

example, a mobile phone) or other endpoint devices. Skype for Business Server 2015
provides a significantly wider range of configuration options for simultaneous ringing.
It is enabled by default.
Team call enables users on a defined team to answer calls for other members of the
team. It is enabled by default.
PSTN re-route enables users (who are assigned this policy) to re-route calls to other
enterprise users on the public switched telephone network (PSTN), if the wide area
network (WAN) is congested or unavailable. It is enabled by default.
Bandwidth policy override enables administrators to override call admission control

policy decisions for a particular user. It is disabled by default.
Malicious call tracing enables users to report malicious calls (such as bomb threats) by
using the client UI, which in turn, flags the calls in the call detail records (CDRs). It is
disabled by default.
PSTN Usage
A PSTN usage record specifies a class of call, such as outbound local, inbound local,
outbound international, and so on.
A Voice Policy associates a user with a PSTN usage record, granting the user that
class of call.
For outbound calls, a route determines which PSTN gateway to use for a usage
record and the dialed number, and for inbound calls, if a particular user can receive
calls from a specified PSTN gateway.
Configuring PSTN Usage

To add PSTN usage records with a voice policy, do any of the following:
To choose one or more records from a list of all PSTN usage records available in your
Enterprise Voice deployment, click Select. Highlight the records that you want to add
with this voice policy, and then click OK.
To remove a PSTN usage record from the voice policy, highlight the record, and then
click Remove.
To define a new PSTN usage record and associate it with this voice policy, do the
following:
12 In the voice policy click New.
57 In the Name field, enter a unique descriptive name for the record. For example, you
may want to create a PSTN usage record named Redmond for full-time employees
located in Redmond, and another named RedmondTemps for temporary employees.
Note: The PSTN usage record name must be unique within the Enterprise Voice
deployment. After the record is saved, the Name field cannot be edited.
58 Use any of the following methods to add and configure routes for this PSTN usage
record:
o To choose one or more routes from the list of all available routes in your Enterprise
Voice deployment, click Select, highlight the routes that you want to associate
with this PSTN usage record, and then click OK.
o To remove a route from the PSTN usage record, highlight the route, and then click
Remove.
o To define a new route and associate it with this PSTN usage record, click New.
Voice routes are covered later in this lesson.
o To edit a route that is already associated with this PSTN usage record, highlight
the route and click Show details.
59 Click OK.
o To edit a PSTN usage record that is already associated with this voice policy, do
the following:
13 Highlight the PSTN usage record that you want to edit, and then click Show
details.
60 Use any of the following methods to associate and configure routes for this
PSTN usage record:
o To choose one or more routes from the list of all available routes in your
Enterprise Voice deployment, click Select, highlight the routes you want to
associate with this PSTN usage record, and then click OK.
o To remove a route from this PSTN usage record, highlight the route, and then
click Remove.
o To define a new route and associate it with this PSTN usage record, click New.
o To edit a route that is already associated with this PSTN usage record, highlight
the route and click Show details.
61 Click OK.
Call Forwarding and Simultaneous Ring

In Skype for Business 2010, administrators can only turn on or off simultaneous ring
or call forwarding. There are no fine controls on these functions.
Skype for Business Server 2015 introduces the ability to assign policies to call
forwarding and simultaneous ring, to limit where and how users can set up this
feature.
In a real world scenario, you should avoid costly calls because users may set up call
forwarding to international destinations and call their own number at local cost.
Call Forwarding and Simultaneous Ring

An administrator can associate a set of PSTN usages to specify this call
authorization to a voice policy
Call authorization types:
Voice policy PSTN usages
Restrict to Skype for Business users only
Custom set of PSTN usages

The same call authorization applies to call forwarding and simultaneous ring.
Voice Routes
Voice route configuration includes:
Name
Description(optional)
Matching patternusing regular expressions that identifies the destination phone

numbers to which the route is applied. Matching patterns may also include
exclusionary expressions.
Trunks that you want to assign to be used by this route.
PSTN usage records that users must have in order use this voice routes.
Note: Trunk availability is covered in the Voice Resilience module.
Routes are also regular expressions that match a normalized number to select a
gateway with the exception of internal routes. Internal route are for internal
numbers and should include both within Skype for Business and on PBX.
Internal routes are used for routing calls through the corporate network and also
used for PSTN routing of internal calls for call admission control (CAC) or network
failure.
Other routes should be based on similar service and can include multiple gateways
in route if in the same location and has the same cost for calls. Otherwise create
multiple routes and use PSTN usage to group and prioritize.
Trunk Configuration
Trunk configuration is applied to trunks defined in Topology Builder. Trunk
configuration settings define the configuration and capabilities between a Mediation
server and:
A PSTN gateway,
An IP-PBX
A Session Border Controller (SBC)

Settings include:
Whether media bypass should be enabled on the trunks.
Whether-time transport control protocol (RTCP) packets are used.
Whether secure real-time protocol (SRTP) encryption is required on each trunk.
Note: Trunks are defined in Topology Builder.

Trunks are configured in the Skype for Business Control Panel or PowerShell.
Truck Configuration Example

Trunk configuration allows for centrally managing the number formatting prior to
routing to PBX/PSTN for both the calling and called number. The first rule recognizes
the number as a London office extension so only dials the extension number, while
the second number is for international users which need to translate the + to the
international access code, typically 011 in the United States.
Called number
Pattern to match Translation pattern
translation rule
To 4 digit PBX ^\+4433445(\d{4})$ $1

Replace + with 011 ^\+(44\d{1}\d+)$ 011$1
The above rules will create the following example dialing patterns:
Alice calls +44221234567.Based on the route translation pattern, the called number is
formatted to 01144221234567 when using the gateway in Redmond.
Bob calls Alices number, +44334455667,from his mobile phone.Based on the route
translation pattern, the calling number is formatted to 5667 when using the gateway
in Redmond.
The order rules are called matters. If the Replace + with 011 rule was first, then
calls to London extensions would always try to connect to the London office through
an international toll call. By using calling rules, the address book only needs to store
one number for a called party that can be used with any Skype for Business client
from any location, assuming the rules are written correctly.
The Skype for Business Control Panel can also be used to test the matching
patterns.
Assigning DID Numbers

Direct inward dialing (DID) enables calling users directly from outside lines without
going through a switchboard or automated attendant (AA). Direct inward dialing
presents special concerns with phone book entries, because the number used for
the internal call can be different then the number used for calling from outside.
Assigning DIDs to a User

Direct Inward Dialing means a user has been assigned an E.164 compliant number,
which guarantees that the number is globally unique and can be dialed directly
through the PSTN. The telecom provider associates a range of numbers with one or
more trunk lines connected to the PSTN gateway or PBX. Skype for Business can
receive inbound calls on any of the trunks and route the call to the appropriate user.
The alternatives to direct inward dialing are switchboards and Exchange Auto
Attendant (Exchange AA or AA), which require calling a central number and then
entering the users extension or talking with a receptionist.
Specifying a Line URI

When an extension is specified that matches the last digits of the number, that
number is interpreted as being a direct dial number. The extension is provided for
internal dialing. Using regular expressions, you can convert this number into an
extension, a local format, or an international dialing format for caller ID.
Best Practice: Always include the country code in the Line URI. One of the
most common mistakes is forgetting the 1 (North American Country Code) when
adding numbers from the United States.
If ext= is not specified, the user has to enter entire phone number in E.164 format
to log on to conferences.
FIGURE 2.2: WARNING MESSAGE WHEN EXTENSION IS NOT

CONFIGURED
In most cases, the phone number stored in Active Directory for a user will not be in
E.164 or RFC 3966 format, so regular expressions can be used to generate a Line
URI.
Internal-OnlyUsers Without DID

The alternative to direct inward dialing is internal numbers or extensions. In this
case, the ext part of the URI does not match the last digits of the phone number.
The phone number should be the number dialing in to the Exchange automated
attendant.
Skype for Business Server 2015 provides you with the ability to manipulate the
caller ID information displayed on outbound calls. As you plan outbound call routes,
consider whether to manipulate the caller ID for calls placed by certain users,
groups, sites, routes, or all users. For example, the caller ID can be manipulated to
show the main switchboard number, even for DID numbers. Also, the default caller
ID maybe the number for an outbound only trunk, which should be replaced by the
main inbound number.
Design a Dial Plan

A Skype for Business Server 2015 dial plan is a named set of normalization rules
that translate phone numbers for a named location, individual user, or contact
object for purposes of phone authorization and call routing. Dial plans determine the
route a call takes from the caller to the recipient for internal, outbound and inbound
calls.
Dial Plan Design Approach

Record all existing dialing habits:
o What are users dialing for internal, local, national or international numbers?
o Evaluate each dialing habit. Do we really need PBX-specific dialing habits?
Consider the current dial plan:
o Unique DID companywide versus overlapping number ranges
o 4-digit dialing, 5-digit dialing,and so on
Understand the Gateway and Mediation server locations.
Understand the customer requirements:
o Use least cost routing or not?
o Is call admission control implemented and are alternative routes required?Call

admission control (CAC) prevents oversubscription of VoIP networks. It is used in
the call setup phase and applies to real-time media traffic.
Migration Strategy
In most cases customers will be migrating from existing PBX systems. There are two
questions that need to be asked about keeping the existing PBX:
14 Is there a compelling reason to keep any existing users on the existing PBX system?
62 Can and should the existing PBX be used as a PSTN gateway?
Note: The advantages and disadvantages of PBX coexistence are examined in

the PSTN Integration module.
Along with the PBX a decision has to be made on what to do with the current dial-in
numbers and trunks, particularly direct inward dial (DID) numbers. From a technical
standpoint getting new DID numbers is simplest approach, but retaining existing
numbers is usually less disruptive to a business. There are cost advantages and
disadvantages to each approach as well.
Using the existing dialing patterns is usually accepted more readily by the users. As
we saw earlier many dialing rules for PBX were based on a locally focused static
workforce using mechanical systems with an expensive, sometimes monopolistic,
public phone system. Are these dialing rules appropriate now with globalized,
mobile workforce in the digital age?
Define Routes
The following steps should be taken to determine the routes which need to be
defined in Skype for Business:
15 Identify all PSTN gateways the capacity of inbound and outbound traffic each can
support.
63 Identify calling patterns and volume, inbound and outbound, between internal and
external parties.
64 Identify a cost for each gateway for each call pattern in step 2.
65 Concentrate on the highest volume routes and define least cost route and secondary
route, in case the least cost route is full.
66 Define all remaining routes to provide complete coverage. These can fall into general
categories, such as International calls or use the routes already defined.
67 It is a good idea to recheck the usage patterns against the defined routes to make
sure there are no bottlenecks
Use the above information to define Voice routes, PSTN usage and Voice policies.
If PBX users still exist also include a route from Skype for Business Server to the
range of numbers for the PBX users.
Least Cost Route

There are typically many possible routes between a caller and receiver. The first
matching and available voice route will be used to complete the call. Best practice
is to list the least costly trunks first. If the trunk is down, full or not authorized for
the user, then the next available trunk is used.
Real World Scenarios and Recommendations

Most dial plans start the dialing habits inherited from the legacy phone system. In
many cases the legacy system was less a system, but more often a mix of PBXs
acquired over many years with different functionality depending on when acquired
and where installed. Here are a few common dialing patterns inherited from legacy
phone systems and recommendations for implementing in Skype for Business.
Dial 9 for an outside line

In early PBX systems dialing 9 was necessary to connect directly to a trunk to the
PSTN. After dialing 9 the user had to wait for a second dial tone, then dial the
phone number of a normal directly connected public or private phone. With the
advent of advanced analog and digital PBX, users no longer connected directly to
the PSTN, so this rule was no longer necessary. Many PBX systems and companies
continued to require dialing 9 for an outside line as a simple rule to determine
which phones were authorized to make outside calls.
Dialing 9 is not required with Skype for Business. Skype for Business will
determine the voice policy to apply based on who the user is and where the user is
located. Then based on the voice policy analyze the number and determine the best
route for the call.
Dial 00 for international calls

In legacy phone systems this rule was necessary because companies wanted to
limit which phones could make international calls and in some cases different long
distance providers and trunks.
Again dialing 00 is not required with Skype for Business. Skype for Business will
determine the voice policy to apply based on who the user is and where the user is
located. Then based on the voice policy analyze the number and determine the best
route for the call. In fact, Skype for Business can route calls to different gateways in
different geographic regions to minimize the cost of a call. In large multinational
companies an international call could be routed to a different site and connected as
a local call.
Note: Before create rules for routing international calls, check all applicable
laws and tariffs.
Prefix an internal number with a PBX-specific trunk number

Companies grew and added offices with new PBX systems resulting in overlapping
number ranges. For example, the London office has 10000 phones with extensions
0000 to 9999. The Redmond office was then built with 2000 phones with extension
0000 to 1999. So the PBX or Site prefix was created as a way to overcome
overlapping number ranges. So to call a user in London dial 1 + extension, or to
call a user in Redmond dial 2 + extension. This worked with a few number of site,
but over 100 sites users cannot remember the PBX prefix.
Since Skype for Business converts numbers to E.164 format for routing the location
doesnt matter. In fact, Skype for Business can normalize calls to legacy PBX users
as well.
Always dial the full number, not extensions for internal numbers
With a highly mobile work force, it is important to keep dialing rules consistent from
office to office. The reaction by most planners is to always require users to dial the
full phone number. There are many problems with this approach. First, this
frustrates users, because they need to remember or store more digits. Second, the
specification of a full number varies from country to country and even within a
country. Third, it does not simplify the routing rules that much, because rules need
to insure that internal calls are not unnecessarily routed to the PSTN.
With Skype for Business users since numbers are associated with users, not
locations the number dialed is always the same and should be a short as possible.
The number dialed is the same from a phone, Skype for Business client or Skype for
Business mobile App. In fact with the contacts synchronized with Active Directory,
once users are familiar with link they dial by name not by number.
Route PlanningA Real World Example

A simple model for defining routes is shown on the slide. Instead of micro-analyzing
each gateway, five to seven route categories are defined, and then gateway specific
routes and matching patterns are defined. Typical categories categories from most
expensive to least expensive
International numbers that typically require dialing PSTN international access

number and country code.
Regional (not shown) international calls to neighboring countries at lower rates.
National Premium specialty or information services billed at a higher rate, 900

numbers.
National non-local numbers within the same country.
Local Premium (not shown) numbers dial like a local number but billed at a higher
rate.
Local typical the lowest cost numbers to dial or require a separate trunk.
When defining routes give then meaningful name such as site plus category. I the
example the site is Germany (DE).
Starting with least cost and working up to the costliest, define the routes and
matching patterns.
Note: Be care with simple country code rules. While many people think 1 is
the country code for the United States, it is in reality the code for the North
American and Caribbean region. So a call from within the United States matching
1 plus a 10 digit number could in reality be a very expensive long distance
number
Number Blocking
Sometimes it is desired to block calls to particular ranges of numbers, because of cost of
the call or the content. There are two ways to block a call the supported, also known as
traditional, method and the simplified alternative method.
The traditional method is to add an excluded range to the existing rules and
optionally create a separate range to allow access to limited number of users. Care
must be taken to exclude the range in all routes which may include that range. This
approach can be difficult to manage if the list of excluded phone number is
constantly changing.For example, if you wish to block the number to the local
footballhotline, except for the sales and support team that services that account,
create a national premium route which and use a voice policy to only assign it to the
account team.
The alternative method is to create an unassigned number range that matches
the desired blocked number range. All outbound calls to this number range will
receive a recorded message. Since the unassigned number check is performed
before call routing, all calls to those numbers are block without regard to caller.
Note: The unassigned number feature is covered in more detail in the Voice
Application module.
Call Park Service

Call Park is a call-management feature that enables you to place a call on hold and
retrieve it later from another phone. This feature is useful for continuing a call from
a different location and for transferring a call when the final recipient is unknown.
Calls from any IP, private branch exchange (PBX), PSTN, or mobile phone can be
parked.
Call Park provides an effective solution for organizations where departments share
calling responsibilities, and it enables the organization to better distribute calls.
Rather than transferring calls to a specific individual who may or may not be
available, Call Park enables anyone available in the organization to take the call.
You, therefore, need to understand the Call Park service and how to configure it for
your use.
Call ParkFeatures
Call park is a feature of some telephone systems that allows a user to put a call on
hold at one telephone set and continue the conversation from any other telephone
set. Call park is a functionality commonly seen in Legacy PBX systems. Skype for
Business implements call park functionality, enabling users to park and retrieve
calls, although Skype for Business relies on a 3rd party paging solution for the
announcement of parked calls. The Call Park application is automatically installed
when you deploy Enterprise Voice.
When a user parks a call, Skype for Business Server 2015 transfers the call to a
temporary number, known as an orbit, where the call is held until someone retrieves
it or it times out. After a call is parked, any Skype for Business2015 user can dial the
orbit number and retrieve the parked call.
Supported Clients
You can use any client to retrieve calls that are parked on Call park. This includes
typical IP common area phones and non-IP phones that are connected to the Skype
for Business Server 2015 infrastructure, including common area phones and PBX
phones. The following table lists the supported clients you can use to both park and
retrieve calls. In the table, Parker refers to the user parking the call, Parkee to the
type of call by originator, and Retriever to the user retrieving the call.
Client Parker Parkee Retriever
Skype for Business2015, Yes Yes Yes

Skype for Business
Common Area Phones,
Tanjay, Aries
Skype for Business 2010, Yes Yes Yes
Attendant Console
Legacy (before 2010) No Yes Yes, if registered with

clients Skype for Business Server
PBX/PSTN No Yes Yes (recommended within
the enterprise)
Note: Only audio calls can be parked. Instant messages and conferences
cannot be parked.
Safe-retrieve
The safe-retrieve feature, while not apparent to a user, and not intended to be a
secure retrieve, provides a way to avoid accidental retrieval of the wrong call. To
do a plain retrieve of a parked call, you just dial the orbit. This is a requirement to
make the user experience as simple as possible.
Because the orbit pool is finite, calls eventually get parked on the same orbit on
which old calls were parked (after a period of time). If the orbits get logged to the
Conversation History, distributed in email, or recorded in any other way, they can
become stale over time. Worse, if a new call is parked on the old orbit, there is a
slight chance of accidentally retrieving the wrong call. Safe-retrieve embeds a
parameter in the Tel: URI link that uniquely identifies the parked call in question and
it refuses to retrieve the call if the parameter doesnt match.
As a general rule, if retrieval happens as a result of a user explicitly dialing the orbit,
it is considered a normal retrieve. If retrieval happens because of special user
interface treatment, or by clicking a link, safe-retrieve then occurs. On the wire, the
only difference is that the outgoing INVITE for safe-retrieve has an additional SIP
header, ms-parked-call.
Skype for Business Call Retrieval

After a call has been parked, there are several options available for retrieving it. A
call can be retrieved by:
Dialing the orbit like any other extension.
Clicking the Retrieve button (this performs a safe retrieve), or copying the link into
an IM message, where a unique ID identifies the call.
The user that parked the call receives notification of who retrieved the call.
The screenshots on the slide show the options that are available to the original call
parker and the notification shown to the call parker when someone picks up the call.
Unlike some PBX systems, which display a light or a flashing button on the phone to
let you know a call is parked or on hold, you cannot see if someone is currently in a
call park in the Skype for Business window.
Call Park Ringback

When a call is parked, a set time is provided for any person to retrieve the call. After
a preconfigured timeout, if the call has not been retrieved, the call is transferred
back to the person who originally parked the call. Use the
CallPickupTimeoutThreshold option in PowerShell to configure the amount of
time a call is parked before it rings back to the phone where the call was parked
from. The value must be entered in the format hh:mm:ss to specify the hours,
minutes, and seconds. The minimum value is 10 seconds, and the maximum value
is 10 minutes. The default is 00:01:30.
Note: We recommend that you configure the OnTimeoutURI option for the
fallback destination to use when a parked call times out and ringback fails. This
options specifies the SIP address of the user or response group to which an
unanswered parked call is routed when the maximum number of times a parked
call rings back to the answering phone is exceeded.
The slide shows the ringback action and how it appears to the original call parker.
The user can choose to either answer or ignore the call. The call cannot be
redirected or forwarded to voice mail.
Deploying Call Park Services

To use the Call Park Service, it must be explicitly turned on in the voice policy that is
assigned to the users that you want to be able to park calls. Anyone can retrieve a
parked call as long as they are registered to a Skype for Business Server.Call Park
services are installed when a server is enabled for Enterprise Voice. You need not
perform any additional configuration in Topology Builder.
Configuring Voice Policies

Users cannot park calls or retrieve parked calls until they are enabled for Call Park in
voice policy. You can enable Call Park voice policy at the global, site, or user level.
The user level takes precedence over the site scope, and the site level takes
precedence over the global scope. You can configure voice policy for Call Park by
using the Skype for Business Server Control Panel or the Skype for Business Server
Management Shell. Either way, you must log on to the computer as a member of
the Administrators group, the Domain Admins group, and the
RTCUniversalServerAdmins group, or as a member of a group that is assigned a
delegated administrative role.
Note: You must explicitly enable Call Park for each user in the Voice Policy
because this is disabled by default.
Using Skype for Business Server Management Shell to Set Voice

Policy
Use the following command to set voice policy for Call Park in the Skype for
Business Server Management Shell.
Set-CsVoicePolicy -Identity <VoicePolicy> -EnableCallPark $true
Verify Your Deployment

After you install and configure Call Park, you need to verify the configuration to
ensure that parking and retrieving calls work as expected. At a minimum, perform
the following:
Call a user who has Call Park enabled and have the user park the call.
Dial the orbit number to retrieve the call.
Park another call, let the parked call time out, and do not pick up the ringback. Verify
that the timed-out call is correctly routed to the fallback destination that is specified
for OnTimeoutURI.
Defining Call Park Ranges

Call parking relies on orbits that have to be defined for each service (per server).
Before the Call Park application can be used, you must configure the call park orbit
ranges in the orbit table.
Configuring the Orbit Table

The Call Park application uses extension numbers in the Call Park orbit table to park
calls. You need to configure the Call Park orbit table with the ranges of extension
numbers that your organization reserves for parked calls. These extensions must be
virtual extensions (that is, extensions that have no user or phone assigned to them).
Each Skype for Business Server pool where a Call Park application is deployed and
configured can have one or more orbit ranges. However, each pool should have at
least one orbit if users are enabled for call park. Orbit ranges must be globally
unique across the Skype for Business Server deployment and may not include direct
inward dialing (DID) numbers.
An orbit range typically encompasses 100 or fewer orbits. Each range can be larger,
as long as it is smaller than the maximum of 10,000 orbits per range and you have
fewer than 50,000 orbits per pool. You can create multiple Call Park orbit ranges for
each Skype for Business Server 2015 pool where Call Park is deployed.
Creating a Call Park Orbit Range

You can use the Skype for Business Server Control Panel or Skype for Business
Server Management Shell to create a new range of numbers for parking calls. Either
way, you must log on to the computer as a member of the Administrators group, the
Domain Admins group, and the RTCUniversalServerAdmins group, or as a member
of a group that is assigned a delegated administrative role to configure orbit ranges.
When choosing the number range for the extensions in your Call Park orbit, the
value you enter must be a string that begins with either the character * or #, or a
number 1 through 9. The first number cannot be a zero. Subsequent numbers may
be any number from 0 through 9, up to seven numbers. If you do not choose to
require the character * or #, you may have a total of eight numbers, but the first
number cannot be zero.
Finally, you must enter the FQDN or service ID of the destination server, which is the
Application service that hosts the Call Park application.
Verifying Normalization Rules

Call Park orbits must not be normalized. Check your dial plans to ensure that your
orbit numbers are not normalized. Also ensure that the default normalization rule in
your dial plans does not contain ^(\d*). Otherwise, your Call Park normalization rule
will not apply.
Note: Important If an orbit number is normalized to E.164, a parked call

cannot be retrieved.
Call Park Management

When you install the Call Park application, global settings are configured by default.
You can use PowerShell to customize the default settings and specify site-specific
settings, instead. However, you must use the Skype for Business Control Panel to
configure the Orbit range.
Configuring Call Park Settings (optional)

You can modify the global settings, and create site-specific settings by using the
Skype for Business Server Management Shell. You can specify the following settings:
The amount of time that elapses after a call has been parked before it rings back to
the phone where the call was parked from. The default is 1 minute 30 seconds (1:30).
Whether music is played while the call is parked. You can also specify your own music
file to use in place of the default music file that is included with Skype for Business
Server 2015.
The number of times a parked call rings back to the answering phone before it is
forwarded to the fallback Uniform Resource Identifier (URI) that is specified. The
default is 1.
The SIP address of the user or response group to which an unanswered parked call is
routed when the maximum call pickup attempts are exceeded. The default is no
forwarding address.
Note: If you customize music on hold and want the same music for multiple
sites, you must configure the music file for each site that runs the Call Park
application.
Using the Skype for Business Server Management Shell

The command to configure Call Park settings in Skype for Business Server
Management Shell is as follows.
New-CsCpsConfiguration -Identity site:<sitename to apply settings>
-CallPickupTimeoutThreshold<hh:mm:ss> -EnableMusicOnHold<$true | $false>
-MaxCallPickupAttempts<number of rings>
-OnTimeoutURIsip:<sip URI for routing unanswered call>
Skype for Business Call Parking

Calls may be parked if the user making the call is enabled for Call Park functionality.
In Skype for Business, the call park feature is enabled through a voice policy. An
orbit is automatically offered to the user parking the call, and once a call is parked,
the user is notified in which of the parked spaces the call is parked. For example,
the screenshot on the slide shows that the call is parked in parking space #130. Any
Skype for Business user who is enabled for the Call Park services can pick up this
parked call by dialing #130.
Skype for Business only allows one call park orbit per user (or per desk phone).
Therefore, you cannot set upcall park for different departments, such as a call park
for Customer Service and a different call park for Sales.
Call Park Deployment Process

AfterCall Park enables an Enterprise Voice user to put a call on hold from one
telephone and then retrieve the call later by dialing an internal number (known as a
Call Park orbit) from any telephone.
The components that Call Park uses are automatically installed and enabled on the
Front End Server or Standard Edition server when you deploy Enterprise Voice.
However, you must use the following steps to configure Call Park before it is
available to users.
PHASE STEPS GROUPS AND DOCUMENTAT
ROLES ION
CONFIGURE Use Skype for Business RTCUniversalServerAd Create or modify a
THE CALL Server Control Panel or the mins Call Park orbit
PARK ORBIT New-CSCallParkOrbit CsVoiceAdministrator range in Skype for
RANGES IN cmdlet to create the orbit CsServerAdministrator Business 2015
THE ORBIT ranges in the call park orbit CsAdministrator
TABLE table and associate them
with the Application service
that hosts the Call Park
application.
CONFIGURE Use the Set- RTCUniversalServerAd Configure Call Park
CALL PARK CsCpsConfiguration mins settings in Skype

SETTINGS cmdlet to configure Call CsVoiceAdministrator for Business 2015
Park settings. At a CsServerAdministrator
minimum, we recommend CsAdministrator
that you configure the
OnTimeoutURI option to
configure the fallback
destination to use when a
parked call times out. You
can also configure the
following settings:
(Optional)
EnableMusicOnH
old to enable or
disable music on
hold.
(Optional)
MaxCallPickupAtt
empts to
determine the
number of times a
parked call rings
back to the
answering phone
before forwarding
the call to the
fallback Uniform
Resource Identifier
(URI).
(Optional)
CallPickupTimeou
tThreshold to
determine the
amount of time
that elapses after a
call has been
parked before it
rings back to the
phone where the
call was answered.
OPTIONALLY, Use the Set- RTCUniversalServerAd Customize Call
CUSTOMIZE CsCallParkServiceMusic mins Park music on hold
THE MUSIC OnHoldFile cmdlet to CsVoiceAdministrator inSkype for
ON HOLD customize and upload an CsServerAdministrator Business 2015
audio file, if you don't want CsAdministrator
to use the default music on
hold.
CONFIGURE Use Skype for Business RTCUniversalServerAd Enable Call Park for
VOICE POLICY Server Control Panel or the mins users in Skype for
TO ENABLE Set-CSVoicePolicy cmdlet CsVoiceAdministrator Business 2015
CALL PARK with the EnableCallPark CsUserAdministrator
FOR USERS option to enable Call Park CsAdministrator
for users in voice policy.
VERIFY Call park orbits must not be RTCUniversalServerAd Verify
NORMALIZATI normalized. Verify that your mins normalization rules
ON RULES normalization rules do not CsVoiceAdministrator for Call Park in
FOR CALL include any of your orbit CsServerAdministrator Skype for Business
PARK ranges. If necessary, create CsAdministrator 2015

additional normalization
rules to prevent orbits
being normalized.
VERIFY YOUR Test parking and retrieving - (Optional) Verify
CALL PARK calls to make sure that your Call Park
DEPLOYMENT configuration works as deployment in
expected. Skype for Business
2015
Park and Retrieve Call Flow

The following series of slides illustrates a simple call park and retrieval scenario and
outlines the signaling path and media flow of the calls. There are several steps.
Step 1: Alice calls Bob, who is running Skype for Business Server 2015.
Step 2: Alice is now connected to Bob, and media flows from Alice to Bob.
Step 3: Alice wants to speak to Charlie. Bob issues a Call Park command to the Call
Park service, requesting an orbit. For now, the media still flow to Bob.
Step 4: Alice is now on hold, receiving Music on Hold (MoH) from the Call Park
service. Bob receives a Call Park orbit. The media is now redirected to the front-end
server for Music on Hold.
Step 5: Bob now shares the Call Park orbit with Charlie through an internal paging
system, instant messaging (IM), or some alternate method.
Note: A paging system is not part of the Skype for Business system.
Step 6: Charlie dials the orbit number in an attempt to retrieve the parked call.
Step 7: Alice is now connected directly with Charlie. The media is redirected to
Charlie. Music on Hold is no longer required, and Alice connects to Charlie
Managing Calls to Unassigned

Numbers
Skype for Business Server lets you configure and manage calls made to unassigned
phone numbers. These are numbers that are valid for your organization, but are not
currently assigned to a person, phone, or other device. You can configure
unassigned numbersto play a recorded message or transfer the call to a different
destination. The alternate destination may be a phone number, SIP URI, or voice
mail.
You should, therefore, know how to manage unassigned number ranges to handle
calls to unassigned phone numbers. You should also be familiar with the call flow
process for an unassigned number.
Purpose of the Unassigned Number Feature

Many organizations buy blocks of DID numbers that have not yet been assigned to
users or phones, but which are still valid numbers.By default, when someone dials
one of these numbers, he or she receives a busy signal and the call may result in an
error returned to the SIP client.Skype for Business Server provides call handling
functionality for telephone calls to phone numbers or extensions that have not yet
been assigned to users. Using the unassigned number feature, you can
predetermine how to handle incoming calls to currently unassigned numbers. For
example, a caller may hear an audio message and then be either disconnected or
transferred to another destination, such as to an operator.
The process for configuring Skype for Business to handle these calls is straight-
forward. First, you must decide what you want to have happen when someone calls
a DID or extension that you have not yet given out to a user. Then, you must define
the number rangethat contains unassigned phone numbers. The following topics
describe the process in more detail.
Announcement Service
The process of deploying the unassigned number feature consists of two main
tasks:
68 Create an announcement by using text-to-speech (TTS) or by recording and uploading

audio files.
69 Configure the unassigned number ranges in the unassigned number table and
associate them with the appropriate announcement.
Before defining the number ranges for the unassigned numbers, you must have at
least one announcement. Announcements are played by the Announcement service
that is installed if a front-end server is enabled for Enterprise Voice.
Creating the Announcement

You cannot create announcements through the Skype for Business Control Panel,
and must instead, use Windows PowerShell. Use the Skype for Business Server
Management Shell to create and configure an Announcement.
16 For a TTS prompt, or no prompt:
The following example shows how to create and name the announcement.
New-CsAnnouncement -Identity ApplicationServer:se01.contoso.local -Name "Number Does Not

Exist"
-TextToSpeechPrompt "Welcome to Contoso, the number you dialed does not exist. You will be
forwarded to the operator" -Language "en-US" -TargetUri "sip:brad@contoso.com"
The TargetURI option is the Uniform Resource Identifier (URI) to which the caller is
transferred after the announcement has been played.
70 For audio prompts:

u. Record the audio file by using any audio recording application capable of creating
a .wav file.
v. Import the contents of the audio file to File Store by running the following cmdlet
in the Skype for Business Server Management Shell.
Import-CsAnnouncementFile
71 Assign the new announcement to a number range in the unassigned number table, as
explained in the following topic.
Deploying the Unassigned Number Feature

Before you configure the unassigned number table, you must either have your
Announcements already defined, or have set up a Microsoft Exchange Unified
Messaging (UM) Auto Attendant.
Purpose of the Unassigned Number Table

You create the unassigned number table to identify those numbers that you want to
flag as unassigned and direct to the Announcement application. The table is invoked
when a caller dials a number that is currently not assigned.
Including all valid extensions in the unassigned number table makes it easy to
specify the action that occurs whenever someone leaves your organization, without
needing to reconfigure the table. However, if you include only unassigned
extensions in the table, you can specify the action that occurs for specific numbers.
For example, if you change the extension for your Help desk, you can include the
old Help desk number in the table and assign it to an announcement that provides
the new number.
Creating the Unassigned Number Table

You create the unassigned number table by using the Skype for Business Server
Control Panel. You must log on as a member of the Administrators group, the
Domain Admins group, and the RTCUniversalServerAdmins group, or as a member
of a group that is assigned a delegated administrative role. To create the
unassigned number table:
17 In the Skype for Business Server Control Panel, in the left navigation bar, click Voice
Features.
72 On the Unassigned Number page, on the Unassigned Number tab, click New.
73 In the New Unassigned Number Range dialog box, in the Name box, type the
name for the range of numbers.
74 In the first Number range box, type the first number in the range of numbers. The
value must be less than or equal to the value specified in the second Number range
box, which specifies the last number in the range of numbers.
Note: The number must match the regular expression,(tel:)?(\+)?[1-

9]\d{0,17}(;ext=[1-9]\d{0,9})?. This means that the number may begin with
the string,tel: (if you do not specify that string, it is automatically added for you),
a plus sign (+), and a digit from 1 through 9. The phone number can be up to 17
digits and can be followed by an extension in the format ;ext= followed by the
extension number.
75 In the second Number range box, type the last number in the range of numbers.
Note: The value must be greater than or equal to the number specified as the
first number in the range. The number must match the regular expression,(tel:)?
(\+)?[1-9]\d{0,17}(;ext=[1-9]\d{0,9})?. This means that the number may
begin with the string,tel: (if you dont specify that string it will be automatically
added for you), a plus sign (+), and a digit from 1 through 9. The phone number
can be up to 17 digits and may be followed by an extension in the format ;ext=
followed by the extension number.
76 In the FQDN of destination server field, do one of the following:
Click Announcement.
Or
Click Exchange UM.
77 If you clicked Announcement, do the following:
g Click Announcement service, click Select, select the service ID of the

Application service that runs the Announcement application that will handle
incoming calls to this range of unassigned numbers, and then click OK.
w. In the Announcement field, select the announcement that will be played for this
range of unassigned numbers.
78 If you clicked Exchange UM, click Auto Attendant phone number, click Select,
select the phone number that will be used for this range of unassigned numbers, and
then click OK.
79 Click OK.
80 On the Unassigned Number page, ensure that the unassigned number ranges are
arranged in the order you want. To change a range position in the table, highlight the
range in the list, and then click the up or down arrow.
Reader Aid: TipSkype for Business Server searches the unassigned number
table from top to bottom and uses the first range that matches the unassigned
number. If you have a range that specifies a last-resort action, ensure that the
range is at the end of the list.
Unassigned Number Call Flow

The following slides describe the high-level call flow that occurs when a caller on the
PSTN has dialed a number that is considered an unassigned number in Skype for
Business Server.
Step 1
Alice has dialed a phone number that she believes belongs to Bob. The vacant
number routing determines that this is not a valid number. Alice is connected to a
special Response Group Service(RGS) workflow and is notified that this number is not
in use.
The number range includes the full corporate number block acquired from the Telco
provider.
If the dialed number is not assigned to a user, reverse number lookup fails because a
matching Line URI cannot be found.
The call then falls back to the target specified for the unassigned number range and
the call is handled by a special RGS workflow.
Step 2
The special RGS workflow now transfers Alice to Charlie as configured by the vacant
number routing.
The RGS workflow plays the announcement (if configured) and will transfer the call to
a generic destination (also if configured).
Step 3
Alice is now connected in a voice call to Charlie.
Note: Although this scenario is not central to the response group functionality,
it does take advantage of the RGS engine to produce a greeting to the caller.
Response Groups Service is covered in detail later in this module.
PSTN Conferencing
If your organization has users who need to attend Skype for Business Server 2015
on-premises conferences when they are out of the office or do not have access to a
computer, you can deploy public switched telephone network (PSTN) conferencing,
also known as dial-in conferencing, so that they can join the conference by using a
PSTN phone.
Dial-in conferencing is an optional feature that you can configure when you deploy
Skype for Business Server 2015 conferencing. Although dial-in conferencing uses
some of the same Skype for Business Server 2015 components that Enterprise Voice
uses, you can deploy dial-in conferencing even if you do not deploy Enterprise
Voice.
Meeting Types
Dial-In Conferencing
Dial-in Conferencing provides an audio option for audio conferences that are hosted
on Skype for Business Server. Since Office Communications Server (OCS) 2007 R2,
users have been able to dial a conference access number, enter their conference
information, and be automatically transferred to the conference. The dial-in option
provides organizations with a cost-efficient alternativeto an audio conferencing
provider (ACP) service. With Dial-In Conferencing, members and non-members of
your corporate network can join a conference call without the need for an ACP.
Reservation-less calls
PSTN conferencing is ideally suited for smaller meetings, typically up to 25
attendees where the meeting organizers can initiate a meeting at any time without
the need for external or operator assistance. Organizations generally have a greater
frequency of reservation-less meetings than those that require a greater degree of
management. Skype for Business Server works well for reservation-less calls, with a
smaller numbers of attendees. Regular staff or project-specific meetings are
examples of this type of meeting, and can include internal and external attendees.
Managed events
A smaller number of meetings require a greater level of control and management.
These meetings involve a higher numbers of attendees than reservation-less calls,
and use an operator to assist callers with problems that may occur. An ACP is a
better option for these types of managed events, for example, an externally focused
call with 100 or more attendees.
The table on the slide compares different types of conference-based meetings and
shows which meeting types are better managed through a PSTN conferencing
solution versus an ACP. The statistics in the table are based on a Gartner report.
User Roles and Permissions

The Skype for Business (Lync) default options are appropriate for small and casual
meetings with coworkers. Its a good idea to change the options if you are inviting
more than 10-15 people, want to control meeting permissions, or have invitees from
other companies.
If you use Outlook, you can change options for all Skype for Business (Lync)
Meetings that you set up by clicking New Skype for Business (Lync) Meeting in your
Outlook Calendar, clicking Meeting Options on the ribbon, and then selecting the
option(s) you want to use.
For more information, access the link: http://go.microsoft.com/fwlink/?

LinkId=624130
PSTN Conferencing Features

The PSTN conferencing functionality, also known as Dial-in conferencing, provided
by Skype for Business Server 2015 makes it simpler for organizations to move a
significant number of ACP calls to Skype for Business. This is particularly true for
those reservation-less calls and smaller meetings that do not require external or
operator assistance.
Meeting features
Skype for Business Server 2015 provides a number of features to enable dial-in
conferencing. The following list briefly describes these features.
Provides features for handling small/medium-size meetings:
o DTMF controls
o Entry and exit announcements
Makes meeting join simpler and more reliable:
o Lobby support for restricted meetings
o Unauthorized users wait in the lobby to be admitted
o Name recording for unauthenticated users
o Integrated seamlessly with Skype for Business meetings
Easy to schedule meetings through a familiar Skype for Business interface
Meeting access security enabled through PIN and phone number authentication
Meeting prompts and guidance in language of choice
Feature comparison between clients

The following table provides a comparison of dial-in functionality between different
Skype for Business clients (including the earlier Lync server 2010 and Lync server
2013)
Skype for
Lync Server Lync Server
Feature Sets Business
2010 2013
Server 2015
Participant passcode
Leader passcode (Corporate (Corporate (Corporate
user PIN) user PIN) user PIN)
Music on Hold
Multiple access
numbers with multiple
languages and toll-free
support
Entry/Exit (Tone or name) (Tone or name) (Tone or
announcement name)
Scheduled meeting
Reservation-less
meeting
Mute/Unmute
notification
DTMF in-meeting
control
Announce late
participants/Recorded
name
Operator/Moderator x x x
assisted conference
Reference code (billing) x x x
Silent mode
Roll call
DTMF Commands
Skype for Business Server enables users to join conferences by dialing in over the
telephone.Dial-in users are not able to view video or exchange instant messages
with other conference attendees, but they are able to join in the audio portion of the
meeting. In addition to being able to join a conference, users can use dual-tone
multi-frequency (DTMF) signaling to manage selected portions of that conference by
using their telephone keypad. Users want to ensure that they have full control of the
meeting if they are joining by phone, and DTMF commands are designed to help
them do this.
The Get-CsDialInConferencingDtmfConfiguration cmdlet enables you to
retrieve a list of all the available DTMF commands and the keys used to perform
those commands. You can use PowerShell to customize or completely disable all
these commands. You can also configure the common character that is used.
The following example lists the individual commands and the associated keys.
CsDialinConferencingDtmfConfiguration
[-Identity <global or site collection to be changed>]
[-AdmitAll<default key is 8>]
[-AudienceMuteCommand<default key is 4>]
[-CommandCharacter<* (default) | #>]
[-EnableDisableAnnouncementsCommand<default key is 9>]
[-HelpCommand<default key is 1>]
[-LockUnlockConferenceCommand<default key is 7>]
[-MuteUnmuteCommand<default key is 6>]

[-PrivateRollCallCommand<default key is 3>]
Command Descriptions
The commands are discoverable on the dial-in conferencing webpage or by issuing
the Help command. The following list describes the commands in more detail.
Automated HelpPlays a description of all the DTMF commands. The default key is
1.
Private roll-callPrivately plays the name of each conference participant. The

default key is 3.
Mute/unmute selfTo mute or unmute your microphone (use the same key to
toggle back and forth between muting and unmuting). The default key is 6.
Lock/unlock (leaders only) Locks or unlocks the conference. If a conference is

locked, no new participants will be allowed to join that conference, at least not until
the conference has been unlocked. The default key is 7.
Toggle silent mode (leaders only) To mute or unmute everyone else in the
conferenceso that attendees cannot unmute and cause noise in the conference. (That
is, everyone other than the presenter will be muted or unmuted.) The default key is 4.
Entry/exit announcements on/off (leaders only) Enables or disables

announcements each time someone joins or leaves the conference. The default key is
9.
Open lobby (leaders only) Allows all the users in the lobby to immediately join
the conference. The default key is 8.
Note: The DTMF commands listed above may differ based on the configuration
on the Organizers site. To ensure accuracy, click the Find a local number link in
the invite for your meeting.
Entry/Exit Announcements
When dial-in users join or leave a conference, the Conferencing Announcement
application can announce their entrance or exit by playing a tone or saying their
names. You can change how announcements work by running cmdlets. This step is
optional.
Authenticated users are announced by Text-to-Speech (TTS). When an anonymous
PSTN user joins a conference as an unauthenticated user, the user is required to
provide an identity and is prompted to record his or her name before being
admitted to the conference. Anonymous users are not admitted to the conference
until at least one leader or authenticated user has joined, and they cannot be
assigned to a predefined role. Conference announcements can be a distraction,
especially at the beginning of the meeting so it is better to batch them to reduce
the number of announcements.
Announcements are controlled in different ways and at different levels. An

administrator can configure announcements as in the following example.
Set-CsDialInConferencingConfiguration
-Identity site:Redmond
-EntryExitAnnouncementsType "ToneOnly"
The conference organizer can also control how and when announcements are
played, including turning them on or off at scheduled times for non-default
meetings. Additionally, a presenter can enable or disable announcements during a
conference by using the in-conference DTMF controls.
Important SettingsJoin Experience

You can use the settings on the Meeting Configuration page to define various
characteristics of the meeting join experience. By default, the global settings define
the join experience, but can be changed at the pool or site level.
Enterprise users or conference leaders who join a conference that is enabled for
dial-in access dial one of the conference access numbers and then are prompted to
enter the conference ID. If a leader has not yet joined the meeting, users can either
enter their unified communications (UC) extension (or full phone number) and PIN,
or wait to be admitted by a leader. The Meeting Organizer can join the meeting as a
leader by entering just the PIN. The front-end server uses the combination of full
phone number or extension, and PIN, to uniquely map enterprise users to their
Active Directory credentials. As a result, enterprise users are authenticated and
identified by name in the conference. Enterprise users can also assume a
conference role predefined by the Organizer.
While Skype for Business Server enables outside users to participate in meetings,
there are increased security risks as a result. To reduce these risks, Skype for
Business Server provides the following additional safeguards:
Participant roles determine conference control privileges.
Participant types enable you to limit access to specific meetings.
Defined meeting types determine which types of participants can attend.
Only users who have Active Directory credentials on the internal network and are
enabled for Skype for Business Server 2015 can schedule conferences.
Anonymous, that is, unauthenticated, users who want to join a dial-in conference
dial one of the conference access numbers and then are prompted to enter the
conference ID. Unauthenticated anonymous users are also prompted to record their
names. The recorded name identifies unauthenticated users in the conference.
Anonymous users are not admitted to the conference until at least one leader or
authenticated user has joined, and they cannot be assigned a predefined role.
Deploying PSTN Conferencing Services

When you plan for conferencing capacity, you should consider the following factors:
Add DID numbers and PSTN trunk capacity for (regional) PSTN access numbers:
o For people connected to the traditional PSTN network to call people connected to
VoIP networks, DID numbers from the PSTN network are obtained by the
administrators of the VoIP network, and assigned to a gateway in the VoIP
network.
Use toll-free numbers to enable dial-in conferencing users outside the local calling
area.
Deploy PSTN gateways:
o Evaluate your organizations telephony usage patterns to determine how many

ports and gateways are required, and how the ports should be allocated among
gateways. An enterprise with multiple sites would typically deploy one or more
gateways at each site (each site needs one). Branch sites can connect to the PSTN
either through a gateway, or through a Survivable Branch Appliance (SBA), which
combines gateway and servers in a single box.
Configure SIP TrunksSIP trunks connect directly to a service provider and provide an
alternative to deployment of PSTN gateways:
o Long distance calls typically cost less through a SIP trunk.
o You can reduce manageability costs and complexity of deployment.
Configure access numbers globally or per site:
o During dial-in configuration, you create dial plans and dial-in conferencing access
numbers. Dial-in conferencing access numbers are the numbers that participants
call to join a conference. When you create the dial-in access number, you select
the regions that associate the access number with the appropriate dial plans.
o Languages are associated with dial-in access numbers. If you support

geographical areas that have multiple languages, you should decide how you want
to define regions to support the multiple languages. For example, you might
define multiple regions based on a combination of geography and language, or
you might define a single region based on geography, and have different dial-in
access numbers for each language.
Configure dial plans with a valid dial-in conferencing region:
18 Create one or more dial plans for routing dial-in access phone numbers.
81 Assign a default dial plan to each pool. Set the dial-in conferencing region
to the geographic location to which the dial plan applies. The region associates the
dial plan with dial-in access numbers.
Deploying PSTN Conferencing Services (part 2)

Skype for Business Server 2015 users who have Active Directory Domain Services
(AD DS) credentials in your organization can join dial-in conferences as
authenticated users by using a personal identification number (PIN). PIN policy
defines the rules for how dial-in conferencing PINs work. The global PIN policy
defines the rules for dial-in conferencing PINs at the forest level.
Configuring a PIN policy

You can create a PIN policy at the site or the user level, or you can create a PIN
policy with a global scope. The order of precedence of how the PIN policies are
applied to users starts with the narrowest scope and goes to the widest scope
user, site, and then global. The steps to create or modify a PIN policy at the site or
user level are similar to the steps to modify the global dial-in conferencing PIN
policy. When you create or modify a user or site PIN policy, you configure PIN
security settings as described in the following list:
Minimum PIN length. The minimum length is five digits.
Maximum logon attempts. Specifies the maximum number of logon attempts before a
user is locked out.
Enable PIN expiration. To have PINs expire after a set duration (days). If you do not
select this option, PINs will never expire. By default, PINs never expire.
Allow common patterns. Allows common patterns of digits in PINs, such as sequential
numbers and repetitive sets of numbers. If you do not select this option, only complex
patterns of digits are allowed. By default, only complex patterns of digits are allowed.
Important: We recommend that you do not allow common patterns.
Welcoming Users to Dial-in Conferencing

After you configure dial-in conferencing and test it to verify that it is functioning
properly, you should set initial PINs for users and notify users about the availability
of the feature, including introductory instructions such as the initial PIN and the link
to the Dial-in Conferencing Settings webpage.
Use the built-in PowerShell script, SetCsPinSendCAWelcomeMail.ps1, to set an
initial PIN and send a welcome email message to a user. For example, the following
script creates a new PIN and then sends a welcome email message from Felix to
Grant.
Set-CsPinSendCAWelcomeMail -UserUri "grant@contoso.com"

-From "felix@contoso.com" -Subject "Your new dial-in conferencing PIN"
-Pin "384032750" -Force
-Credential Admin@contoso.com -UseSsl
This example forces a new PIN with a value of "384032750" for Grant, even though
Grant had an existing PIN, and then sends a welcome email message from Felix to
Grant. Because the Credential parameter is specified, the person running the
command is prompted to enter a password. The email is sent by using Secure

Sockets Layer (SSL).
Enable PSTN Dial-In Conferencing

Several conferencing policy settings support dial-in conferencing for participants.
When you configure dial-in conferencing, you should verify that these fields are set
appropriately for your organization, and modify them as necessary. The Enable
PSTN dial-in conferencing setting enables users to join the audio portion of a
conference by dialing in from the PSTN. This setting is required for dial-in
conferencing. This setting is selected by default in the default global conferencing
policy.
Additional Deployment Options

As you deploy PSTN conferencing services in your organization, there are some
optional features you may wish to consider.
Configure DTMF commands globally or by site:
o If you need to customize the DTMF commands, you can define these at the site
level. However, this is not recommended because it will confuse end-users in
different sites.
Manage the order of access numbers by conference region (Windows PowerShell

cmdlet only):
o You can change the order in which dial-in numbers are listed and this will affect
the numbers presented in the meeting invitation. This can only be done by using
PowerShell.
o You can now configure access numbers on a per-site basis, which was not possible
in previous versions of Skype for Business Server. Managing the order of these
numbers is important because only the top three numbers are displayed in the
online meeting invitation.
Managing Conference
Get-CsWindowsService returns detailed information about Skype for Business
Server 2015 components that run as Windows services. This cmdlet was introduced
in Lync Server 2010.
Many Skype for Business Server 2015 components run as standard Windows
services; for example, the Skype for Business Server 2015 Conferencing Attendant
application is actually a service named RTCCAA. The Get-CsWindowsService
cmdlet enables you to retrieve detailed information about these Skype for Business
Server 2015 services and only these services. Thats because the cmdlet has been
designed to ignore any service that is not part of Skype for Business Server 2015.
Audio Conferencing Architecture

Audio conferencing adds additional components to the Skype for
Businessinfrastructure. On the slide, the green boxes are components used by the
PSTN conferencing services.
Conference Auto Attendant (CAA):
o Handles the Interactive Voice Response (IVR) for the user join process.
o Joins the user to the conference.
o Plays music if the conference has not been activated.
The Conference Announcement Service (CAS)takes care of announcements when

joining the meeting and during the meeting.
Personal Virtual Assistant (PVA):
o Handles prompts played only to a user in the users language (you have been
muted/unmuted, Help, lobby notifications, roll call).
Group Virtual Assistant (GVA):
o Handles prompts played to all users in the conference in their language (Entry/Exit
announcements).
Dial-In conferencing page:
o Enables a user to manage their PIN and provides a list of PSTN access numbers for
the conferencing service.
The webpage is accessible through the dial-in simple URL.
Multi-Language Support
Skype for Business supports a variety of languages for the Conferencing Auto
Attendant and Virtual Assistants. The language used will depend on a variety of
metricssuch as access numbers, regions, and so on.
Dial-in experience:
o Language is taken from the contact object, configured with Windows PowerShell.
o Interactive Voice Response (IVR) offers users the choice of languages found on the
contact object.
In-meeting experience:
o Personal announcements (played by PVA):

In the language that the caller had at dial-in time.
o Global announcements (CAS, GVA):
Played to all users, grouped by language.
o Dial-out:
The person being dialed gets the language of the person dialing out.
If the language of the user cannot be matched, the closest language is used
(example: FR-CA goes to FR-FR).
The slide shows a multi-language support scenario in which three callers are calling
from three different locations (two callers in the U.S. and one caller in Germany).
19 When Caller 1 joins the meeting, this creates an instance of the Conference
Announcement Service. A Group Virtual Assistant is created in English, and a Personal

Virtual Assistant is created in English for Caller 1.
82 When Caller 2 joins, only a PVA is created because the CAS and GVA already exist.
83 When Caller 3 joins and requests to use the German language, a new GVA and PVA is
created.
o If the first user joins, the Conference Announcement service is activated.
o Additionally, a GVA (for announcements to everybody with the same language)

and a PVA (for personal announcements) is created.
o If a second user with the same language preference joins, the GVA is re-used and
an additional PVA is created.
o A user joining with a different language will trigger the creation of a new GVA and
a PVA in the new language of preference.
Response Group Services

Response groups is a call management feature that enables you to queue calls that
are made to a specific area, such as a Help Desk, and then route the calls to a
designated group of people, called agents.
You should therefore know about the Response Group Service (RGS) and the
changes that have been implemented with Microsoft Skype for Business Server
2015. You should also know the components of the RGS, including queues, agent
groups, workflows, and interactive workflows. In addition, you should be able to use
the tools such as Skype for Business Server Control Paneland the Skype for
Business Server Management Shell to manage workflows.
Typical PBX deployments

Many PBX systems employ an automatic call distributor (ACD) solution to route
incoming calls. ACD systems are often found in organizations that handle large
volumes of incoming phone calls from callers who do not need to talk to a specific
person, but who require assistance from any one of multiple persons, for example, a
customer service representative.
The diagram on the slide shows a typical PBX deployment. Starting with the gray
blocks on the left, basic PBX features include basic hunt groups and hunting
methods, and agent sign-in/sign-out. Hunt groups provide a way to allow inbound
calls to be automatically routed to multiple extensions until the call is answered.
Hunt groups are a useful tool in high volume, rapid response time customer service
situations. With agent sign-in/sign-out, users can control if they are part of the call
groups.
Beyond the basic PBX features, an organization may have more complex
requirements and choose to use an ACD solution as an add-on to the basic PBX. This
solution provides more features and sometimes requires additional licensing costs.
Typical features of this scenario include the following:
MoH. Music on Hold.
Business Hours. The ability to define different behavior depending on a time schedule
Basic call detail recording (CDR). Logging of call details.
Supervisor. An ACD supervisor that can listen in and control dispatched calls
Live Views. Displays that show queue length, yield, and so on.
Advanced CDRs. Building on the basic call details with extended information about the
call.
A third scenario is represented by the blocks at the right of the diagram. Some
organizations may need a more specialized solution, such as a fully dedicated ACD.
This type of solution is more suited to large call centers with many agents, where
there is a high call volume. There is a requirement for high scale and high
availability. Additionally, some specialized scenarios may require a dedicated ACD
that is fully integrated with the back-end systems. The ACD may need to be
interoperable with a line-of-business (LoB) application or applications, especially if
the LoB controls who is called.
Positioning Skype for Business Response Groups

As already discussed, organizational scenarios that involve smaller departments or
smaller, internal call centers often use basic PBX features such as hunt groups and
agent sign-in/sign-out to implement incoming call routing. Additionally, some of
these scenarios may use Music on Hold or basic call detail recording if there is a
need for some level of ACD functionality. Skype for Business Server implements
Response Group Services functionality that enables users to setup small

departmental ACDs.
Important: RGS is designed for smaller-sized solutions and is not intended as

a replacement for large contact centers.
The goal of RGS is to provide all the basic ACD features for departmental needs and
to simplify call management. The following features of RGS are listed to help you
understand where RGS fits into the ACD model.
Hunt groups and basic IVRs. Easy to configure Voice Response system, for example,
press 1 for Administration, press 2 for Sales, and so on.
Integration with Skype for Business presence. For example, if presence is set to
do not disturb, the agent will not receive calls.
Agent anonymity. Calling from the Help desk, instead of from Bob or Alice.
Announcements. For unassigned numbers.
Speech recognition and TTS. Configuring Interactive Voice Response (IVR)

incorporates speech recognition, for example, say one for Sales. Questions can
either be pre-recorded or can use text-to-speech.
Music on hold (MoH).
Basic CDRs.Building on the basic call details with extended information about the call.
Feature Overview
Response Groups Services provides a number of features and capabilities. You can
use these features to perform various tasks in your organization.
Interactive Voice Response (IVR)

Input: DTMF or speech recognition (SR).A selection can be made by using DTMF
commands (press 1 for Sales) or by using speech recognition (say one for Sales).
Playback: Either .wma or TTS.Questions can be pre-recorded or TTS can be used.
SR/TTS supported in 26 languages: Language packs are available for both SR and TTS.
Rich IVR tree configuration:The question tree can contain multiple questions and
multiple layers and branches.
Customized messages before transferring/disconnecting and MoH.
Support for any number of questions and answers.
Call Queuing
Music on Hold (MoH).
Queue time out/queue overflow/disconnect action:Pre-defined actions in case calls are

not, or cannot, be answered in time.
On first call/last call: Route to PSTN, other queue, SIP URI, or voice mail.
Routing
Serial, parallel, longest idle, and round robin:Controls the way calls are offered to
agents (discussed in more detail in the next lesson).
Integration with receptionist console to route all calls independent of Presence state
and allow receptionist to select call in the queue (attendant routing).
Agent groups are defined by distribution groups or custom groups.
Agent-Side User Experience

Inbound and outbound calls can be anonymous. (That is, to hide agent ID),for
example, call is from the Helpdesk, and not from Bob or Alice.
Call context on incoming call (options selected by caller during the IVR).An agent will
see all the choices the user made in the IVR system (as shown in the example on the
slide).
Agent page to sign in/sign-out of groups.
Ring timeout configured by the administrator.
Presence icon to identify response groups.
Infrastructure
Bandwidth management support:RGS fully supports Call Admission Control (CAC) to
safeguard call quality.
Draining:Servers can be taken offline gracefully to ensure that existing calls are not
interrupted.
Application store for settings.
Response Group Management

In Skype for Business Server 2015, the following management roles are available for
managing response groups:
Response Group Administrator
Response Group Manager

Response Group Administrators can manage any aspect of any response group.
Response Group Managers are a new addition to Skype for Business Server 2015,
and they can manage only certain aspects, and only for the response groups that
they own. The scope of a Response Group Manager is at a workflow level; a
manager cannot see or modify response groups that he or she does not manage.
The new Response Group Manager role can help decrease administration costs,
because you can delegate limited responsibilities for specific response groups to
any user who is enabled for Enterprise Voice. This is an improvement in the
scalability of response group deployment, especially for larger deployments.
Response Group Managers can perform the following tasks:
Add and remove agents from agent groups.
Modify specific properties of the response group workflow, including:
o Business hours/Holidays.
o Welcome message.
o Interactive Voice Response (IVR) structure.
Manage the Response Group-associated queues and agent groups.
Delegate management rights to other managers.

Response Group Managers can use the following tools to manage their specific
groups:
Skype for Business Server Control Panel (Response Group settings only. Other Skype
for Business Server settings are not available to Managers.
Response Group Web Configuration Tool.
Skype for Business Server Management Shell.
Managed and Unmanaged Response Groups

To accommodate the new Manager role, Skype for Business Server 2015 Response
Group application introduces a Workflow Type of Managed or Unmanaged
Response Group. The diagram on the slide shows the differences between Managed
and Unmanaged workflows. Some key differences include the fact that delegation is
only available for Managed groups and that queues cannot be shared across
Managed workflows. Additionally, in a Managed response group, queues are unique
to one specific workflow, and unlike Managed groups, queues can be shared for
unmanaged groups.
The following table also summarizes the key differences.

Managed Unmanaged
Management Managed by an Managed only by an

administrator or a manager administrator
Delegation Managed by an Delegation not supported
administrator or the
response group manager
Sharing Cannot share the response Can share queues and
group queues or agent agent groups with other
groups with any other unmanaged response
response group groups
Visibility Visible to the response Visible only to the
group manager administrator
Visible to the administrator Not visible to any manager
Not visible to other
managers
Implementing Response Group

Services
This lesson will examine the components necessary to implement response groups
using the Skype for Business Response Group Service. Response groups can be
populated manually or be built from an exchange distribution list. Calls coming into
a response group are handled by a workflow to determine how and to whom a call
will be routed. Workflows must be defined and applied before calls will be queued
for processing by the response group.
Response Group Building Blocks

Response Groups comprise various building blocks such as agents and groups, and
you need to know how to configure them to answer incoming calls. Other important
building blocks such as queues and workflows are described in later topics.
Agents and Groups

Agents are Skype for Businessusers designated to answer incoming calls placed in a
response group queue. To receive a call from a response group, a user must be
enabled for Enterprise Voice. Agents are members of one or more groups that are
associated with workflows. For example, a Help desk workflow may include a group
of agents with similar job responsibilities.
Groups
Groups are organized as an ordered list of agents or Microsoft Exchange Distribution
Groups. In the illustration on the slide, membership and priority (order) is defined.
Membership can be formal or informal; if the membership is formal, users have to

sign in to receive calls.
Groups use predefined routing methods and the group defines the routing method
to be used. Groups are also added to one or more queues.
Routing Method
When configuring groups in the Skype for Business Server Control Panel, you can
define the method for routing calls to agents. The available routing methods are as
follows:
Attendant. Offers a new call to all agents who are signed into Skype for Business
Server 2015 and the Response Group application at the same time, regardless of their
current presence
Skype for Business 2010 Attendant users who are configured as agents
can see all the calls that are waiting and answer waiting calls in any order.
The call is sent to the first agent who accepts it, and the other Skype for
Business 2010 Attendant users no longer see the call.
Parallel. Offers a new call to all available agents at the same time. The call is sent to
the first agent who accepts it.
Longest idle.Offers a new call first to the agent who has been idle the longest (has
had a presence of Available or Inactive in Skype for Business Server for the longest
time).
Round robin.Offers a new call to each agent, in turn.
Serial.Always offers a new call to the agents in the order in which they are listed in the
Agent list.
Note: There is not a new attendant client for Skype for Business Server 2015;
Skype for Business 2010 Attendant is supported on Skype for Business Server
2015.
Formal vs. Informal User Groups

When an agent accepts a call, the caller may or may not be able to see the agent's
identity, depending on how the administrator configures the response group. Agents
can either be formal, which means that they must sign in to the group before they
can accept calls routed to the group; or informal, which means that they do not sign
into and out of the group to accept calls.
Informal agents are automatically signed in to the group when they sign in to Skype
for Business Server, and they cannot sign out of the group.
Note: If a user wants to sign in to a response on a different server from its

home server, the user has to manually change the URL to the sign-in page. The
URL to the sign-in page is provisioned in-band
Configuring Queues
Queues are the next building block for response groups. Queues hold callers until an
agent answers the call. When the Response Group application searches for an
available agent, it searches agent groups in the order listed. You can select the
agent groups that are assigned to the queue and specify queue behavior, such as
limiting the number of calls that the queue can hold and the period of time that a
call waits until an agent answers the call.
You configure queues from the Skype for Business Server Control Panel, under
Response Groups, and then Queue. You can also use Windows PowerShell to
create or modify a queue.
Queues follow each groups routing sequence, for example, Longest Idle or Round
Robin. You can configure Queue Overflow and determine what to do if the
maximum number of callers waiting (configurable in the queue settings) in the
queue is exceeded. You can also set the maximum time period a caller waits on hold
before the call is answered and determine the call behavior if that time is exceeded.
For example, you may disconnect the call after the time-out and have the call
forwarded to voice mail or another number.
If you are one of the delegated Response Group Managers for a managed workflow,
you can create or modify response group queues and assign them to the workflows
that you manage.
Configuring Workflows
Workflows are the last building block for response groups. A workflow defines the
behavior of a call from the time that the phone rings to the time that someone
answers the call. There are two basic types of workflows:
Hunt groups. No voice response menus, focused on dispatching calls to agents
Interactive Voice Response (IVR). Fully configurable voice response with multiple
questions and a decision tree
A workflow also defines settings such as a welcome message, music on hold,
business hours, and holidays. Workflows are represented by a SIP-enabled contact,
for example, helpdesk@contoso.com. Workflows can also be configured for agent
anonymity, so that the identity of agents is hidden during calls.
You configure workflows by using the Response Group Configuration Tool, a web
browser interface that is opened from the Skype for Business Server Control Panel.
You can also open the Response Group Configuration Tool directly from a web
browser by typing the following URL: https://<webPoolFqdn>/RgsConfig. You can
also configure workflows by using the Skype for Business Server Management Shell.
Sample RGS ScenarioOperator

As previously mentioned, RGS is designed for smaller-sized solutions and is not
intended as a replacement for large contact centers. In a typical departmental
solution, you can use RGS to provide basic ACD functionality, such as hunt groups,
simple IVRs, integration with Skype for Business presence, agent anonymity, and so
on. For example, in a sample RGS scenario for an operator or switchboard, you
can configure different operator scenarios.
Classic Operator
In this scenario, the Attendant routing method is used and it will ring all agents,
regardless of their Presence state. You can use the Skype for Business Attendant
2010 or Skype for Business client. The Attendant Console is targeted at the operator
function, and it will show all incoming calls in a queue.The operator can then select
and dispatch calls in a more efficient way than when using the Skype for Business
client. In this scenario, all Presence states are ignored, except Do-Not-Disturb, and
queue timeout can be used to limit the wait time.
Operator with Fallback

In this scenario, you may define another group, for example, a group of secretaries
or a backup group that will answer calls if all operators are busy and do not answer
the call. This scenario follows the same pattern as the Classic Operator, but an
additional queue is added for Fallback users. This second user group is notified on
queue timeout or queue overflow.
Operator with Fallback and After-hours Service

A third implementation follows the same pattern as the previous Operator with
Fallback scenario. In this scenario, you define an after-hours Fallback destination.
For example, you can forward incoming calls after regular business hours to the
security desk.
Deploying Response Groups

The pattern for a typical deployment scenario for response groups uses the
following process:
Define agent groups (Skype for Business Control Panel):
o Configure routing method.
o Alert timeouts.
o Set group membership (distribution list or custom).
Define queues (Skype for Business Control Panel):
o Associate groups to queues.
o Define timeout and overflow actions.
Define the workflow (RGS Web Page):

o Choose your IVR template.
o Define the name of the response group.
o Create the contact objects (user unaware of the implementation).
o Choose the language.
o Configure text/message to be played.
o Define the business hours and holidays.
o Select the IVR and queue.
Note: You can configure more complex IVRs, and all RGS settings, by using
Windows PowerShell.
RGS Call Flow and Agent Anonymity

The following list describes a graphical representation of RGS call flow with agent
anonymity.
Incoming call from Alice to the Response Group.
The workflow determines which group will service the incoming call.
The group determines if Agents are to be anonymous.
One or more agents are alerted depending on the availability of agents and the
routing settings
If anonymization is disabled, the Response Group Service is removed from the call and
Alice directly connects to Bob.
If anonymization is enabled, the Response Group Service will stay in thecall and Alice
does not directly connect to Bob.
Group Call Pickup

Plan for Group Call Pickup
Group Call Pickup enables users to answer incoming calls to their colleagues from their
own phones. This increases the availability of a user's line by enabling other users to
answer an incoming call by dialing a call pickup group number. When Group Call Pickup is
deployed, the number of incoming calls that are routed to voice mail can be significantly
reduced, which is particularly useful for calls from customers who are external to your
organization.
The Group Call Pickup feature is designed in particular for business units in open office
environments. Incoming calls are not disruptive because they ring only at the intended
destination. Other users who hear the ring, however, can still pick up the call simply by
dialing the group number.
In environments where users are not located in an open office layout, or where users who
share a common responsibility are geographically distributed, team call presents the most
suitable solution. The primary difference between Group Call Pickup and team call is that,
with Group Call Pickup, an incoming call rings only at the intended destination, but
anyone can still choose to answer it by dialing a group number. With team call, the call
rings at all the team members' phones, and any user in the team can pick up the phone
to answer the call. An additional difference between Group Call Pickup and team call is
that Group Call Pickup is managed by an administrator, through Skype for Business
Server. With team call, end users manage the feature by using the Skype for Business
client. With Group Call Pickup, therefore, this aspect of call management can be
centralized.
Group Call Pickup is built on the Call Park application. When you deploy Group Call Pickup,
you configure the call park orbit table with separate ranges of extension numbers that are
designated as call pickup group numbers. Like call park orbit numbers, call pickup group
numbers must be virtual extensions that have no user or phone assigned to them. Each
Front End pool where you deploy Group Call Pickup can have one or more ranges of call
pickup group numbers. The group number ranges must be globally unique across the
Skype for Business Server deployment.
After you configure the call pickup group numbers, you assign users to a call pickup
group. Any user who is assigned to a call pickup group can have their calls answered by
other users. When a call comes in to a user who is assigned to a call pickup group, any
other user who notices the call can answer it by manually dialing the call pickup group
number. The user who picks up the call does not need to be a member of the group. When
a call is picked up by another user, a notification is sent to the number originally called.
If a user dials a call pickup group number to answer a call when multiple phones in the
group are ringing, the user answers the call that has been ringing the longest.
Simultaneous ringing settings will work for users who have group call pickup. That is, a
call made to a user who has Group Call Pickup will ring for all the configured destinations,
and another user can answer the call. The exception to this rule is when the user
configures simultaneous ringing to call all the team members.
Group Call Pickup cannot be used to answer the following types of

calls:
Calls to a private line.
Calls from a contact who has been assigned the Friends and Family privacy
relationship
Video portion of audio/video calls
Simultaneous ringing calls that are routed to team call members
Calls routed to a delegate
Calls routed to a response group
The following types of users cannot participate in Group Call Pickup. That is, they
should not be included in a Group Call Pickup group, and they cannot pick up calls
for users who have Group Call Pickup enabled.
Users who are homed online in a hybrid deployment

Users who are not homed on either a Skype for Business Server 2015 pool
or a Lync Server 2013 pool with Cumulative Updates for Lync Server 2013:
February 2013 in an on-premises deployment
If no one answers a call to a member of a call pickup group, the call is routed as
specified in the client settings. That is, the call goes to voicemail or is forwarded to
a different destination, as specified in the client settings.
Deployment and requirements
Group Call Pickup is automatically deployed when you deploy Enterprise Voice and
the Call Park application. You enable Group Call Pickup by configuring the Call Park
orbit table with separate ranges of numbers designated as call pickup group
numbers, and then by assigning users to call pickup groups and enabling the users
for Group Call Pickup.
Clients supported for Group Call Pickup
Any of the following clients can be used to answer calls to Group Call Pickup
members:
Skype for Business

Lync 2013
Lync 2010
Lync Phone Edition
Note: Users can use any of these clients to answer calls to Group Call Pickup
members, but the users must be homed on a Skype for Business Server pool or a
Lync Server 2013 pool with Cumulative Updates for Lync Server 2013: February
2013.
The following clients and devices are not supported for picking up calls to Group Call
Pickup members:
Lync Mobile
Lync app for Windows 8 and Windows RT
Lync for iPad
Analog phones
Phones with public switched telephone network (PSTN) numbers
Capacity planning
The following table describes the Group Call Pickup user model that you can use as
the basis for capacity planning requirements.
Important: Group Call Pickup is based on the Call Park application. Keep in
mind that, for disaster recovery capacity planning, each pool of a paired pool
should be able to handle the workloads for Call Park services, including Group Call
Pickup, in both pools.
Group Call Pickup User Model
Per Front End

Per Standard Edition
Metric pool (with 8 Front
server
End Servers)
Recommended number of users per group 50 50
Recommended number of groups 500 60
Maximum number of users per pool enabled

25,000 3,000
for Group Call Pickup
Maximum rate of incoming calls to total

users enabled for Group Call Pickup per 500 60
pool per minute
Maximum rate of calls retrieved by users

200 25
with Group Call Pickup per pool per minute
Configuration Prerequisites for Group Call Pickup

Group Call Pickup requires the following components:
Application service
Call Park application
These components are installed automatically when you deploy Enterprise Voice.
Group Call Pickup Configuration User Rights

You use the following administrative tools to configure Group Call Pickup:
Skype for Business Management Shell

Lync Server Management Shell
SEFAUtil resource kit tool
Use Skype for Business Server Management Shell to create and manage call pickup
groups in the Call Park orbit table. Use the SEFAUtil resource kit tool to assign a call
pickup group and enable Group Call Pickup for users or to disable Group Call Pickup
for users.
Configuring Group Call Pickup requires any of the following administrative roles,
depending on the task:
CsVoiceAdministrator: This administrator role can create, configure,

and manage all voice-related settings and policies.
CsUserAdministrator: This administrator role can enable Group Call
Pickup for users. This administrator role also has read-only view access to
all voice configurations.
CsServerAdministrator: This administrator role can manage, monitor,
and troubleshoot servers and services.
CsAdministrator: This administrator role can perform all of the tasks of
CsVoiceAdministrator, CsServerAdministrator, and CsUserAdministrator
Group Call Pickup Deployment Process
Required groups
Phase Steps
and roles
Enable the 1. Use the New- RTCUniversalServerAd

SEFAUtil CsTrustedApplicationPool cmdlet mins
resource kit tool to create a new trusted application
in the topology pool.
2. Use the New-CsTrustedApplication
cmdlet to specify the SEFAUtil tool as
trusted application.
3. Run the Enable-CsTopology cmdlet
to enable the topology.
4. Install the resource kit tools on a
Front End Server that is in the trusted
application pool created in step 1.
5. Verify that SEFAUtil is running
correctly by running it to display the
call forwarding settings of a user in
the deployment.
Use the New-CSCallParkOrbit cmdlet to
create call pickup number ranges in the call
park orbit table and assign the call pickup
ranges the type GroupPickup.
Note:
You must use Lync Server Management RTCUniversalServerAd

Shell to create, modify, remove, and view mins
Configure call
Group Call Pickup number ranges in the call
pickup number CsVoiceAdministrator
park orbit table. Group Call Pickup number
ranges in the
ranges are not available in Lync Server CsServerAdministrato
call park orbit
Control Panel. r
table
Note: CsAdministrator
For seamless integration with existing dial
plans, number ranges are typically
configured as a block of virtual extensions.
Assigning Direct Inward Dialing (DID)
numbers as range numbers in the call park
orbit table is not supported.
Assign a call
pickup number Use the /enablegrouppickup parameter in
to users, and the SEFAUtil resource kit tool to enable
-
enable Group Group Call Pickup and assign a call pickup
Call Pickup for number for users.
the users
Notify users of
their assigned
Because any user can retrieve a call made
call pickup
to a Group Call Pickup user, users may want -
number and any
to monitor more than one group.
other number of
interest
Verify your
Test placing and retrieving calls to make
Group Call
sure that your configuration works as -
Pickup
expected.
deployment
The call pickup group number ranges must comply with the following
rules:
The beginning number of the range must be less than or equal to the
ending number of the range.
The value of the beginning number of the range must be the same length
as the ending number of the range.
The number range must be unique. This range cannot overlap with any
other range.
If the number range begins with the character * or #, the range must be
greater than 100.
Valid values: Must match the regular expression string ([\*|#]?[1-
9]\d{0,7})|([1-9]\d{0,8}). This means the value must be a string
beginning with either the character * or # or a number 1 through 9 (the
first character cannot be a zero). If the first character is * or #, the
following character must be a number 1 through 9 (it cannot be a zero).
Subsequent characters can be any number 0 through 9 up to seven
additional characters (for example, "#6000", "*92000", "*95551212", and
"915551212"). If the first character is not * or #, the first character must
be a number 1 through 9 (it cannot be zero), followed by up to eight
characters, each a number 0 through 9 (for example, "915551212",
"41212", "300").
Create or Modify a Group Call Pickup number range
1. Log on to the computer where Lync Server Management Shell is installed

as a member of the RTCUniversalServerAdmins group or with the
necessary user rights as described in this link:
http://go.microsoft.com/fwlink/?LinkId=624131
2. Start the Lync Server Management Shell: Click Start, click All Programs,
click Microsoft Lync Server 2013, and then click Lync Server
Management Shell.
3. Use New-CsCallParkOrbit to create a new range of call pickup group
numbers. Use Set-CsCallParkOrbit to modify an existing range of call
pickup numbers.
At the command line, run:
New-CsCallParkOrbit -Identity <name of call pickup group range> -NumberRangeStart<first number in

range> -NumberRangeEnd<last number in range> -CallParkService<FQDN or service ID of the
Application service that hosts the Call Park application> -Type GroupPickup
Group Call Pickup

Plan for Group Call Pickup
Group Call Pickup enables users to answer incoming calls to their colleagues from their
own phones. This increases the availability of a user's line by enabling other users to
answer an incoming call by dialing a call pickup group number. When Group Call Pickup is
deployed, the number of incoming calls that are routed to voice mail can be significantly
reduced, which is particularly useful for calls from customers who are external to your
organization.
The Group Call Pickup feature is designed in particular for business units in open office
environments. Incoming calls are not disruptive because they ring only at the intended
destination. Other users who hear the ring, however, can still pick up the call simply by
dialing the group number.
In environments where users are not located in an open office layout, or where users who
share a common responsibility are geographically distributed, team call presents the most
suitable solution. The primary difference between Group Call Pickup and team call is that,
with Group Call Pickup, an incoming call rings only at the intended destination, but
anyone can still choose to answer it by dialing a group number. With team call, the call
rings at all the team members' phones, and any user in the team can pick up the phone
to answer the call. An additional difference between Group Call Pickup and team call is
that Group Call Pickup is managed by an administrator, through Skype for Business
Server. With team call, end users manage the feature by using the Skype for Business
client. With Group Call Pickup, therefore, this aspect of call management can be
centralized.
Group Call Pickup is built on the Call Park application. When you deploy Group Call Pickup,
you configure the call park orbit table with separate ranges of extension numbers that are
designated as call pickup group numbers. Like call park orbit numbers, call pickup group
numbers must be virtual extensions that have no user or phone assigned to them. Each
Front End pool where you deploy Group Call Pickup can have one or more ranges of call
pickup group numbers. The group number ranges must be globally unique across the
Skype for Business Server deployment.
After you configure the call pickup group numbers, you assign users to a call pickup
group. Any user who is assigned to a call pickup group can have their calls answered by
other users. When a call comes in to a user who is assigned to a call pickup group, any
other user who notices the call can answer it by manually dialing the call pickup group
number. The user who picks up the call does not need to be a member of the group. When
a call is picked up by another user, a notification is sent to the number originally called.
If a user dials a call pickup group number to answer a call when multiple phones in the
group are ringing, the user answers the call that has been ringing the longest.
Simultaneous ringing settings will work for users who have group call pickup. That is, a
call made to a user who has Group Call Pickup will ring for all the configured destinations,
and another user can answer the call. The exception to this rule is when the user
configures simultaneous ringing to call all the team members.
Group Call Pickup cannot be used to answer the following types of

calls:
Calls to a private line.
Calls from a contact who has been assigned the Friends and Family privacy
relationship
Video portion of audio/video calls
Simultaneous ringing calls that are routed to team call members
Calls routed to a delegate
Calls routed to a response group
The following types of users cannot participate in Group Call Pickup. That is, they
should not be included in a Group Call Pickup group, and they cannot pick up calls
for users who have Group Call Pickup enabled.
Users who are homed online in a hybrid deployment

Users who are not homed on either a Skype for Business Server 2015 pool
or a Lync Server 2013 pool with Cumulative Updates for Lync Server 2013:
February 2013 in an on-premises deployment
If no one answers a call to a member of a call pickup group, the call is routed as
specified in the client settings. That is, the call goes to voicemail or is forwarded to
a different destination, as specified in the client settings.
Deployment and requirements
Group Call Pickup is automatically deployed when you deploy Enterprise Voice and
the Call Park application. You enable Group Call Pickup by configuring the Call Park
orbit table with separate ranges of numbers designated as call pickup group
numbers, and then by assigning users to call pickup groups and enabling the users
for Group Call Pickup.
Clients supported for Group Call Pickup
Any of the following clients can be used to answer calls to Group Call Pickup
members:
Skype for Business

Lync 2013
Lync 2010
Lync Phone Edition
Note: Users can use any of these clients to answer calls to Group Call Pickup
members, but the users must be homed on a Skype for Business Server pool or a
Lync Server 2013 pool with Cumulative Updates for Lync Server 2013: February
2013.
The following clients and devices are not supported for picking up calls to Group Call
Pickup members:
Lync Mobile
Lync app for Windows 8 and Windows RT
Lync for iPad
Analog phones
Phones with public switched telephone network (PSTN) numbers
Capacity planning
The following table describes the Group Call Pickup user model that you can use as
the basis for capacity planning requirements.
Important: Group Call Pickup is based on the Call Park application. Keep in
mind that, for disaster recovery capacity planning, each pool of a paired pool
should be able to handle the workloads for Call Park services, including Group Call
Pickup, in both pools.
Group Call Pickup User Model
Per Front End

Per Standard Edition
Metric pool (with 8 Front
server
End Servers)
Recommended number of users per group 50 50
Recommended number of groups 500 60
Maximum number of users per pool enabled

25,000 3,000
for Group Call Pickup
Maximum rate of incoming calls to total

users enabled for Group Call Pickup per 500 60
pool per minute
Maximum rate of calls retrieved by users

200 25
with Group Call Pickup per pool per minute
Configuration Prerequisites for Group Call Pickup

Group Call Pickup requires the following components:
Application service
Call Park application
These components are installed automatically when you deploy Enterprise Voice.
Group Call Pickup Configuration User Rights

You use the following administrative tools to configure Group Call Pickup:
Skype for Business Management Shell

Lync Server Management Shell
SEFAUtil resource kit tool
Use Skype for Business Server Management Shell to create and manage call pickup
groups in the Call Park orbit table. Use the SEFAUtil resource kit tool to assign a call
pickup group and enable Group Call Pickup for users or to disable Group Call Pickup
for users.
Configuring Group Call Pickup requires any of the following administrative roles,
depending on the task:
CsVoiceAdministrator: This administrator role can create, configure,

and manage all voice-related settings and policies.
CsUserAdministrator: This administrator role can enable Group Call
Pickup for users. This administrator role also has read-only view access to
all voice configurations.
CsServerAdministrator: This administrator role can manage, monitor,
and troubleshoot servers and services.
CsAdministrator: This administrator role can perform all of the tasks of
CsVoiceAdministrator, CsServerAdministrator, and CsUserAdministrator
Group Call Pickup Deployment Process
Required groups
Phase Steps
and roles
Enable the 6. Use the New- RTCUniversalServerAd

SEFAUtil CsTrustedApplicationPool cmdlet mins
resource kit tool to create a new trusted application
in the topology pool.
7. Use the New-CsTrustedApplication
cmdlet to specify the SEFAUtil tool as
trusted application.
8. Run the Enable-CsTopology cmdlet
to enable the topology.
9. Install the resource kit tools on a
Front End Server that is in the trusted
application pool created in step 1.
10.Verify that SEFAUtil is running
correctly by running it to display the
call forwarding settings of a user in
the deployment.
Use the New-CSCallParkOrbit cmdlet to
create call pickup number ranges in the call
park orbit table and assign the call pickup
ranges the type GroupPickup.
Note:
You must use Lync Server Management RTCUniversalServerAd

Shell to create, modify, remove, and view mins
Configure call
Group Call Pickup number ranges in the call
pickup number CsVoiceAdministrator
park orbit table. Group Call Pickup number
ranges in the
ranges are not available in Lync Server CsServerAdministrato
call park orbit
Control Panel. r
table
Note: CsAdministrator
For seamless integration with existing dial
plans, number ranges are typically
configured as a block of virtual extensions.
Assigning Direct Inward Dialing (DID)
numbers as range numbers in the call park
orbit table is not supported.
Assign a call
pickup number Use the /enablegrouppickup parameter in
to users, and the SEFAUtil resource kit tool to enable
-
enable Group Group Call Pickup and assign a call pickup
Call Pickup for number for users.
the users
Notify users of
their assigned
Because any user can retrieve a call made
call pickup
to a Group Call Pickup user, users may want -
number and any
to monitor more than one group.
other number of
interest
Verify your
Test placing and retrieving calls to make
Group Call
sure that your configuration works as -
Pickup
expected.
deployment
The call pickup group number ranges must comply with the following
rules:
The beginning number of the range must be less than or equal to the
ending number of the range.
The value of the beginning number of the range must be the same length
as the ending number of the range.
The number range must be unique. This range cannot overlap with any
other range.
If the number range begins with the character * or #, the range must be
greater than 100.
Valid values: Must match the regular expression string ([\*|#]?[1-
9]\d{0,7})|([1-9]\d{0,8}). This means the value must be a string
beginning with either the character * or # or a number 1 through 9 (the
first character cannot be a zero). If the first character is * or #, the
following character must be a number 1 through 9 (it cannot be a zero).
Subsequent characters can be any number 0 through 9 up to seven
additional characters (for example, "#6000", "*92000", "*95551212", and
"915551212"). If the first character is not * or #, the first character must
be a number 1 through 9 (it cannot be zero), followed by up to eight
characters, each a number 0 through 9 (for example, "915551212",
"41212", "300").
Create or Modify a Group Call Pickup number range
4. Log on to the computer where Lync Server Management Shell is installed

as a member of the RTCUniversalServerAdmins group or with the
necessary user rights as described in this link:
http://go.microsoft.com/fwlink/?LinkId=624131
5. Start the Lync Server Management Shell: Click Start, click All Programs,
click Microsoft Lync Server 2013, and then click Lync Server
Management Shell.
6. Use New-CsCallParkOrbit to create a new range of call pickup group
numbers. Use Set-CsCallParkOrbit to modify an existing range of call
pickup numbers.
At the command line, run:
New-CsCallParkOrbit -Identity <name of call pickup group range> -NumberRangeStart<first number in

range> -NumberRangeEnd<last number in range> -CallParkService<FQDN or service ID of the
Application service that hosts the Call Park application> -Type GroupPickup
Location Information Server

Private Branch Exchange (PBX) phone numbers were tied to a phone in a specific
location.Even with Internet Protocol PBX (IP-PBX) the location of the phone could be
determined based on which switch or subnet was used to connect to the network.
Since phone number on Skype for Business Server 2015 are tied to a user, not a
location or phone, determining the callers location is more difficult and in fact can
easily change. The Location Information Server (LIS) deals with identifying a callers
current location and translating it into a civic address for routing the call for further
assistance and inclusion with E-9-1-1 calls.
What Is Location Awareness?

Microsoft Skype for Business Server 2015 supports Enhanced 9-1-1 (E9-1-1) calling
from Skype for Business clients and Skype for Business Phone Edition devices. When
you configure Skype for Business Server for E9-1-1, emergency calls placed from
Skype for Business2015 or Skype for Business Phone Edition includes Emergency
Response Location (ERL) information from the Location Information Service (LIS)
database. ERLs consist of civic (that is, street) addresses and other information that
help to identify a more specific location in office buildings and other multitenant
facilities. When a user makes an emergency call, Skype for Business Server routes
the audio call, along with the location and callback information, through a Mediation
Server to an E9-1-1 service provider. The E9-1-1 service provider uses the civic
address of the caller to route the call to the Public Safety Answering Point (PSAP)
that serves the caller's location, and sends along an Emergency Service Query Key
(ESQK) that the PSAP uses to look up the caller's ERL.
Skype for Business Server 2015 includes a standard web service interface that you
can use to connect the LIS to Simple Network Management Protocol (SNMP)
applications that match media access control MAC addresses with port and switch
information. The LIS is the underlying service that provides a users location
information, and identifies and populates that information in the Skype for Business
client. This affects how emergency calls are routed, for example, in the case of a
caller who is traveling outside the home location. Location information assists in
selecting the appropriate emergency provider and informing the provider of the
callers location.
If an SNMP application is installed and the Location Information Service fails to find
a match in the location database, the Location Information Service automatically
queries the application by using the MAC address provided by the client. The
Location Information Service then uses the port and switch information returned by
the SNMP application to query the location database again.
Note: To support E9-1-1 as part of a Microsoft Skype for Business Server 2015
deployment, you must obtain an E9-1-1 routing service from a Skype for Business
Open Interoperability Program qualified service provider of emergency services.
Choose the provider that best fits your organizational requirements.
"Location Aware" Emergency Routing

Skype for Business Server uses a location policy to enable Skype for Business clients
for E9-1-1 during client registration. A location policy contains the settings that
define how E9-1-1 will be implemented. You can configure routing of emergency
calls to a specific PSTN gateway and assign the location policy to a particular
network site. The network site defines the location.
The example on the slide describes a scenario where a North America-based Skype
for Business user is traveling outside North America. The location policy is
configured to enable that user to dial the local emergency calling number in the
region in which he or she happens to be. In this case, the user is located in Ireland,
so if he or she needs to make an emergency call, he or she dials 112 and the call is
routed to a local PSTN gateway.
You may be wondering about using Active Directory (AD)sites to define location
policy. In reality, not all organizations use AD sites. Furthermore, a location policy
requires more granularity than an AD site is able to provide. The AD site is generally
only at the campus or data center level, but E9-1-1 requires a more specific target,
for example, a specific corner of a building.
Voice Routing
Building on the voice routing behavior that was discussed earlier, the diagram on
the slides focuses on the voice routing of an emergency call.
84 A user initiates a call from the client, by either:
x. Entering a SIP URI that will bypass most of the logic and go directly to Inbound
Routing.
y. Dialing aphone number that will invoke processes such as normalization and
reverse number lookup.
85 First,the number is checkedto determine if it is an emergency number:
h If the number qualifies as an emergency number, the Location Policy for

emergency numbers is applied, a PSTN usage is assigned, and routing
continues.Which location policy is to be used is determined based on the location
information
z. If the number does not qualify as an emergency number, the process continues.
E9-1-1 Skype for Business Server 2015 Components

The two main Skype for Business Server 2015 components for implementing E9-1-1
are location policies and the Location Information Service (LIS). Location policies
define the calling behaviors and the LIS maps the network to street addresses.
Location Policies
You can configure location policies to be location-aware though Presence or you can
assign a static designation for the users location. You can also define which
numbers should qualify as emergency numbers and where emergency calls should
be routed to. A location policy enables you to define the destination where an
instant message IM alert can be sent in the event of an emergency call. For
example, security personnel could be notified by an IM alert when a user dials an
emergency call. Furthermore, you can enable an organizations security department
to listen in on an emergency call that is made by an employee of that organization.
Location Information Service (LIS)

The Location Information Service takes the Basic Service Set Identifier (BSSID), MAC
address, subnet, switch and port information, and performs a lookup against the
Master Street Address Guide (MSAG). The MSAG is maintained by the E9-1-1 call
routing provider and defines the correct format for the location information. When a
validated address is returned, LIS passes it to the E9-1-1 provider by using Presence
Information Data Format Location Object (PIDF-LO).
Note: PIDF-LO describes an object format for carrying geographical information

on the Internet. The PIDF-LO extends the Presence Information Data Format (PIDF),
which was designed for communicating privacy-sensitive presence information
and which has similar properties.
Emergency Calling Setup and Call Flow

Providing the PSAP timely and accurate location information is a requirement E-9-1-
1. A company uses a variety of sources to provide emergency responders with
locations including Skype for Business Contact data, Master Street Address Guide
(MSAG) addresses and manual intervention by E-9-1-1 service providers or the
companys security desk.
E9-1-1 Configuration
As an administrator there are several steps you need to perform to set up and
configure your environment for E9-1-1, as shown on the illustration on the slide.
20 First engage with an E9-1-1 service provider. E9-1-1 services are separate from Skype
for Business, which only provides the information to the E9-1-1 service provider. The
E9-1-1 service provider routes emergency calls originating from Skype for Business
Server to the correct Public Safety Answering Point (PSAP) based on the location
information contained within the call.
86 Configure Skype for Business:
i Populate the location database with network elements and associated addresses.
aa. Configure policies, routes, and users.

87 Test addresses for validity. The E9-1-1 service provider has protocols for testing
location data without disrupting the PSAP. Insurance policies and local laws may also
dictate requirements for testing.
88 Correct invalid addresses and repeat validation. During testing you may need to work
closely with the network administrators and facilities to ensure the correct building
access location is identified.
Note: For a list of E9-1-1 service providers that have been independently
qualified for Skype for Business, see Unified Communications Open
Interoperability Program (UCOIP)http://go.microsoft.com/fwlink/?LinkID=309705.
Location Discovery
Emergency services need to be able to determine an accurate location of a caller
who dials an emergency call. The illustration on the slide show the location
discovery process.
1. The client sends subnet information to the Registrar. The Skype for Business Server
Registrar is a server role that enables client registration and authentication, and
provides routing services
2. The Registrar returns the LIS URI and the location policy during in-band provisioning;
this is because Subnet 172.24.33.0 is enabled for E9-1-1.The subnet has to be defined
in Skype for Business and a location policy must be associated.
3. The client sends the subnet to the LIS.Locations are by subnet.
4. LIS does the subnet/location match and returns the location in PIDF-LO format.
Placing an Emergency Call

The illustration on the slide shows the process that occurs when a user dials an
emergency call.
21 Client dials 911 and includes PIDF-LO in the SIP INVITE.
Although this example assumes 911, an administrator can specify a different
emergency number for regions outside of North America.
89 The IM notification of the emergency call is sent. The calling party and location
information is sent to the Security Desk (optional).
90 An E9-1-1 call is routed over the Session Initiation Protocol (SIP) trunk.
91 The Routing Provider connects to the appropriate PSAP. This occurs automatically, if
possible (4a). If this is not possible, it is handled by a call center agent, if the address
cannot be validated (4b).
92 The voice path is connected to the Security Desk (optional).
Designing Location Policies

Skype for Business Server uses a location policy to enable Skype for Business clients
for E9-1-1 during client registration. A location policy contains the settings that
define how E9-1-1 will be implemented. In Skype for Business Server 2015. You
should therefore know how to create location policies for E9-1-1.
Location Policy Definition

The following table lists location policy settings that define user experience, routing,
dial string, and security desk notification.
Location policy parameter Value
EnhancedEmergencyServices True |
Enabled False
LocationRequired Yes |
Disclaimer
| No
UseLocationForE911Only True |
False
PstnUsage PSTN
usage
name
EmergencyDialString Emergenc
y number
EmergencyDialMask List of
other
emergenc
y numbers
that must
be
recognize
d
NotificationUri SIP URI of
Security
Desk for
IM
notificatio
n
ConferenceUri Phone
number in
SIP URI for
phone
conferenci
ng
ConferenceMode OneWay |
TwoWay
EnhancedEmergencyServiceD
isclaimer
Emergency Services Enabled. When this value is Yes, the client is enabled for E9-1-
1. When a client registers, it attempts to acquire a location from the Location
Information Service and will include the location information as part of an
emergency call.
Location Required. This setting is used only when Emergence Services Enabled is
set to Yes. You can configure the Location Required setting to define the client
behavior. Setting the value to No means that the user will not be prompted for a
location. Setting the value to Yes means that the user will be prompted for a
location, but the user can dismiss the prompt. Setting the value to Disclaimer
means that the user will be prompted for a location and will be shown a disclaimer if
he or she tries to dismiss the prompt. In all cases, the user can continue to use the
client. The Disclaimer setting is useful for localization or different regions.
Note: The disclaimer text will not appear if a user manually enters a location
before being enabled for E9-1-1. Updates to the disclaimer text will not be viewed
by users who have already viewed the disclaimer.
PSTN Usage. The name of the PSTN Usage that contains the routing paths that
determine which SIP trunk, PSTN gateway, or ELIN gateway the emergency calls will
go to.
Note: Only one usage can be assigned to a location policy. This PSTN Usage
overrides the PSTN Usages assigned to the users voice policy, but applies only to
calls placed to the Emergency Dial String or to one of the Emergency Dial String
Masks.
Emergency Dial String. This dial string (less the leading +, but including any
normalization done by the Skype for Businessusers Dial Plan) signifies that a call is
an emergency call. The Emergency Dial String causes the client to include
location and callback information with the call.
Note: If your organization does not use an external line access prefix, you do
not need to create a corresponding Dial Plan normalization rule that adds a + to
the 911 string, prior to sending the call to Outbound Routing on a Skype for
Business pool server; the + will be automatically prefixed by the Skype for
Business client as a result of the location policy. However, if your site uses an
external access prefix, you need to add a normalization rule to the applicable Dial
Plan policy that strips the external access prefix and adds the +. For example, if
your location uses an external access prefix of 9 and a user dials 9 911 to place an
emergency call, the client will use its Dial Plan policy to normalize this to +911
before the dialed number is evaluated by the routes in the callers location profile.
Emergency Dial String Masks. A semicolon-separated list of dial strings that is

translated into the specified Emergency Dial String. For example, you may want
to add 112, which is the emergency service number for most of Europe. A visiting
Skype for Business user from Europe may not know that 911 is the U.S. emergency
number, but they can dial 112 and get the same result. As with the Emergency Dial
String, do not include a + before each number, and if you use external line access
codes, ensure that there are normalization rules in the users Dial Plan policy to strip
off the access code digit.
Notification URI. Specifies one or more SIP URIs of the security personnel who
receive an instant messaging (IM) notification when an emergency call is placed.
Distribution groups are supported.
Conference URI. Specifies a direct inward dialing (DID) number (typically, a security
desk number) that should be conferenced in, when an emergency call is placed.
Conference Mode. Specifies if the conference URI will be conferenced into the
emergency call by using one-way or two-way communication.
Enhanced Emergency Service Disclaimer. This setting specifies the disclaimer that
users see if they dismiss the prompt for a location. In Skype for Business Server
2015, you can use location policy to set different disclaimers for different locales or
different sets of users.
Note: This location policy setting differs from Lync Server 2010, where you
used the Set-CsEnhancedEmergencyServiceDisclaimer cmdlet to set a global
disclaimer for the entire organization. If a global disclaimer already exists, you
need to specify that disclaimer in the location policy, that is,Skype for Business
Server 2015 uses only disclaimers specified in the location policy.
Location Refresh Interval. Specifies the amount of time (in hours) between client
requests for a location update from the Location Information Service. The value can
be set to any value between 1 and 12. The default value is 4.
Location Policy Scope

Location policies are similar to other Skype for Business Server policies in that they
can be assigned at different scope levels. However, location policies with user-level
scope work differently than other Skype for Business Server policies.
You can apply location policies that are assigned at the user level to endpoint
objects, for example, to Users and Common Area Phone contact objects.
Alternatively, location policies can be assigned to Skype for Business Server
network sites. Network sites are groupings of client subnets associated with a
geographical location (but may not necessarily be all subnets in an entire central
site or branch site). Any clients connected to the subnets in a network site
automatically receive the location policy assigned to that network site. If a user-
level location policy is assigned to a user and a network site, the network site-based
location policy takes precedence over any per-user policy setting.
Each network site has an assigned location policy, and each policy will have
different PSTN usages, Notification URIs, and Conference URIs values assigned to it.
Note: The scoping of a location policy works differently from other policies
because when a user on a pool at one office site visits another site and has to
make an emergency call, the E9-1-1 call routing settings appropriate to that
network site will apply, regardless of what pool or site the user is assigned to.
PSTN Usage Creation

The location policy PSTN usage determines the routes for an emergency call. You
can set this to be the emergency service providers SIP trunk or an Emergency
Location Identification Number (ELIN)-capable gateway. You can also set this to be a
local PSTN gateway.
Because the PSTN usage is part of the location policy and is used specifically for
emergency call routing, you do not have to create this in a users voice policy. The
PSTN usage for the location policy does not appear in a users voice policy. The
PSTN usage is only used for an emergency call if a location policy has been applied
to the user. In the case of an emergency call, the PSTN usage then overrides the
PSTN usages assigned to the users voice policy. Where emergency call routing is
set to a local gateway, there would be a unique PSTN usage for each policy. If the
call routing is set to an emergency service providers SIP trunk, there would be one
PSTN usage for all location policies.
Note: Some partners in the Unified Communications Open Interoperability

Program provide qualified ELIN-capable gateways, which can serve as an
alternative to a SIP trunk connection to a qualified E9-1-1 service provider. ELIN
gateways support ISDN or Centralized Automatic Message Accounting (CAMA)
connectivity to public switched telephone network (PSTN)-based E9-1-1 services.
ELIN/PSTN gateways are discussed in more detail in Module 6: PSTN Integration.
You can configure the PSTN usage in Skype for Business Server Control Panel or
Skype for Business Server Management Shell.
The following command defines a new PSTN usage for the Emergency Route.
Set-CSPstnUsage -Usage @(add=EmergencyCallUsage}
The following command displays a list of currently defined PSTN usages.
Get-CSPstnUsage
Voice Route Creation

The example on the slide shows how to create a new E9-1-1 voice route in Windows
PowerShell. The voice route allows calls to the emergency number to go through the
PSTN gateway that references the emergency service provider. A PSTN usage
record, named EmergencyCallsUsage, has been created and then the New-
CsVoiceRoute cmdlet is used to create the new voice route,
EmergencyCallsRoute.
The number pattern must be the same number pattern that is used in the
Emergency Dial String setting in the location policy. A "+" sign is needed because
Skype for Business adds "+" to emergency calls. "e911gw.fabrikam.net" is the SIP
trunk service ID for the E9-1-1 service.
Note: The E9-1-1- voice route does not differ from any other voice route in that
it does not distinguish between normal and emergency routes.
If you plan to deploy a primary SIP trunk and a secondary SIP trunk, you may define
more than one route. You can also create voice routes from Skype for Business
Server Control Panel.
PIDF-LO Support on Trunk

To include location information in an E9-1-1 INVITE, you need to configure the SIP
trunk that connects to the E9-1-1 service provider to route emergency calls through
the gateway. The information contained in PIDF-LO enables privacy-sensitive
Presence information to be sent to the E9-1-1 service provider.
To do this, set the EnablePIDFLOSupport flag on the Set-CsTrunkConfiguration

cmdlet to True. The default value for EnablePIDFLOSupport is False. For example,
the following command sets the value to true.
Set-CsTrunkConfiguration -Identity "Service:e911gw.fabrikam.net" -EnablePIDFLOSupport $true
You do not need to enable receiving locations for fallback PSTN gateways and ELIN
gateways.
Implementing (LIS) Location

Information Service
The Location Information Service (LIS) provides detail caller location information
required by enhanced emergency response service (E9-1-1). The E9-1-1 system
automatically displays to the dispatcher the callers location. For fixed location
phones, such as a home phone, this is easy to determine by mapping caller ID. For
businesses which normal provide a central phone number for caller and Skype for
Business users who can be mobile providing accurate location information can be a
challenge. LIS leverages the physical location of the corporate infrastructure, such
as routers and network jacks to provide location information about a Skype for
Business user.
Location Information Service

Enhanced 9-1-1 allows an emergency operator to identify the location of a caller
without having to ask the caller for that information. In the case where a caller is
calling from a Voice over Internet Protocol (VoIP) connection, that information must
be obtained based on various network connection factors. When attempting to
match a client location, various components are used to determine the location of
the client. This involves tracing data through aggregation points in the network
(such as Ethernet switches, or DSL access nodes) and locating the port to which
packets are sent. This information is combined with data available to the LIS
(usually obtained from a database) to determine a final location.
There are some key considerations in regard to what location information is actually
available to the LIS.
To obtain location specificity, an organization requires a good knowledge of the wiring

in their building or facilities.
A street address is not always associated specifically with a media access

control(MAC) address. For example, many people carry notebook computers.
The client is responsible for determining its own subnet mask. This is covered in more
detail later in this lesson.
You can configure LIS by using either Windows PowerShell or the Skype for Business
Server Control Panel. LIS adheres to various industry standards to validate location
information. It follows the recommendations of Interim VoIP Architecture for
Enhanced 9-1-1 Services(NENA i2), published by the National Emergency Number
Association (NENA). That document describes the architecture for connecting
emergency callers in the VoIP domain with the various Public Safety Answering
Points supported by the current E9-1-1 architecture.
NENA i2 does not include specifications for the methods used to determine
location nor how the endpoint actually receives location. LIS follows the Internet
Engineering Task Force (IETF)PIDF-LOstandards with extensions for location format.
The Skype for Business LIS can be integrated with other LIS that adhere to this
format. For example, a company may have developed their own LIS or their existing
IP-PBX may include location data that adheres to PIDF-LO.
Additional Reading: A copy of Interim VoIP Architecture for Enhanced 9-1-1

Servicescan be downloaded from NENA at http://go.microsoft.com/fwlink/?
LinkID=309710 (This link redirects to a third party web site not maintained by
Microsoft).
Configuring Location Information Server

To automatically locate clients within a network, you need to populate the LIS
database with a network wiremap, which maps network elements to civic (that is,
street) addresses. You can use BSSIDs, subnets, wireless access points, switches,
and ports to define the wiremap.
Adding Network Elements to the Location Database

You can add addresses to the location database individually, or in bulk, by using a
CSV file that contains the column formats described in the following table. You can
then use Windows PowerShell commands to import the data from the CSV files.
Network
Required Columns
Element
Wireless <BSSID>, <Description>, <Location>, <CompanyName>,

access point <HouseNumber>, <HouseNumberSuffix>, <PreDirectional>,
<StreetName>, <StreetSuffix>, <PostDirectional>, <City>,
<State>, <PostalCode>, <Country>
Subnet <Subnet>, <Description>, <Location>, <CompanyName>,
<HouseNumber>, <HouseNumberSuffix>, <PreDirectional>,
Port <ChassisID>, <PortIDSubType>, <PortID>, <Description>,
<Location>, <CompanyName>, <HouseNumber>,
<HouseNumberSuffix>, <PreDirectional>, <StreetName>,
<StreetSuffix>, <PostDirectional>, <City>, <State>,
<PostalCode>, <Country>
Switch <ChassisID>, <Description>, <Location>, <CompanyName>,
<HouseNumber>, <HouseNumberSuffix>, <PreDirectional>,
In an Enterprise Voice implementation with E9-1-1, emergency calls must first be
routed through an E9-1-1 Network Routing Provider to ensure that the calls are
routed to the appropriate PSAP.
To do this, the provider must have a list of locations from the organization that it can
then match against the MSAG to ensure all locations are valid. TheSet-
CsLisServiceProvider cmdlet creates or modifies information about a provider,
including the name of the provider, a URL for the web service that the organization
will use to send the locations, and a certificate and password for the secure web
service.
Location Information ServerSubnet

E9-1-1 allows an emergency operator to identify the location of a caller without
having to ask the caller for that information. In the case where a caller is calling
from a VoIP connection, that information must be extracted based on various
connection factors. The VoIP administrator must configure a location map (called a
wiremap) that will determine a callers location. Using the Set-CsLisSubnet cmdlet,
an administrator can map physical locations to the subnet through which the client
is connected.
Subnet Parameter
The Subnet parameter is the only required parameter for this cmdlet. It signifies the
IP address of the subnet, and this value should be entered as an IPv4 address (digits
separated by periods, for example, 192.0.2.0). If you enter a subnet value that
already exists, this cmdlet will update the location for that subnet based on the
location parameters that are supplied. If the subnet does not exist, a new subnet
location will be created.
What Constitutes a Valid Location?

A valid location consists of, at a minimum, the Location, HouseNumber, StreetName,
City, State, and Country. If you do not specify all these parameters, calls that
originate from the specified subnet may not contain the information required by the
emergency operator (depending on whether valid settings are available for a switch,
port, or wireless access point that can be used in place of subnet settings). We
recommend that you be as specific as possible with the location parameters and
provide as many as possible.
Validating Addresses
Before publishing the Location database, you must validate new locations against
the Master Street Address Guide (MSAG) maintained by the Emergency Services
Provider.
Note: Important: Failure to validate addresses leaves the LIS wiremap in an

invalidated state and the location information is treated as if it were manually
entered by the user.
To validate an address located in the Location database do the following:
22 Open the Skype for Business Server Management Shell.
93 Run the following cmdlets to configure the Emergency Services Provider connection.
$pwd = Read-Host -AsSecureString<password>

Set-CsLisServiceProvider -ServiceProviderName Provider1 -ValidationServiceUrl<URL provided
by provider> -CertFileName<location of certificate provided by provider> -Password $pwd
94 Run the following cmdlet to validate the addresses in the Location database.
Get-CsLisCivicAddress | Test-CsLisCivicAddress -UpdateValidationStatus
You can also use the Test-CsLisCivicAddress cmdlet to validate individual addresses.
Publishing the LIS Configuration

To implement LIS, you must publish the LIS configuration to the Central
Management Store. You do this by using the Publish-CsLisConfiguration cmdlet.
This cmdlet commits any changes in the central location database to the Central
Management Store, allowing the information to be replicated to the Location
Information servers so that the locations can be rendered to clients. You can remove
the configuration from the Central Management Store by calling the Unpublish-
CsLisConfiguration cmdlet.
To publish the Location database do the following:
23 Open the Skype for Business Server Management Shell.
95 Run the following cmdlet to publish the Location database.
Publish-csLisConfiguration
Address Status
Validating New Addresses
Before publishing the location database, you must validate new locations against
the MSAG that is maintained by your SIP trunk or PSTN E9-1-1 service provider. To
validate the addresses in the location database, run the Get-CsLisCivicAddress
cmdlet. In Skype for Business Server, the location is determined based on mapping
the callers port, subnet, switch, or wireless access point to a specific location,
theGet-CsLisCivicAddresscmdlet retrieves one or more of the addresses
associated with these locations. You can correct invalid addresses by using this
cmdlet. By checking every address that is added to the location database against
the MSAG, you can ensure that emergency calls can be correctly routed.
MSAGValid Flag
Get-CsLisCivicAddressdiffers from the Get-CsLisLocation cmdlet in that it
returns only unique addresses. It does not return the company name or a location
name; it returns only address information. It also returns a flag (MSAGValid) that
specifies whether the address has been validated against the Master Street Address
Guide. This flag can be automatically updated by running the Test-
CsLisCivicAddress cmdlet.
The Windows PowerShell example on the slide shows that the address has not yet
been validated with the MSAGValid flag, returning a value of False.
Address Discovery
Most Skype for Business clients will be connected to the corporate network. In this
case using physical network components, such as routers, switches and wireless
access points (WAP), and logical components can be mapped to physical locations
for use in emergency location. For example, subnet 192.168.0.x is in Building 4 in
Redmond, or WAP RED-4-101 is in conference room 101 on the first floor of
building 4.
Some users may either permanently or temporary connecting to Skype for Business
while off the corporate network or dialing from a mobile client. Address discovery
plan needs to provision for these callers, as well.
Client Location Request

A location request includes the following information:
Link Layer Discovery Protocol (LLDP) from Layer 2 connectionswitch and port IDs (if
available)
Subnet
Wireless access point (WAP) Basic Service Set Identification (BSSID) (if available)
Client MAC address
Note: The Link Layer Discovery Protocol (LLDP) is an open standard for
detecting other nodes on a local area network (LAN). Although it is part of the IP
suite, it functions at a level below the Internet layer, called the Link layer. Link
Layer Discovery Protocol-Media Endpoint Devices (LLDP-MED) is a set of standards
enhancing the basic LLDP protocol, which relates to increased discovery of media
endpoint devices to deal specifically with voice applications.
LLDP is defined by the Institute of Electrical and Electronics Engineers as "IEEE
802.1AB." Its title is "Station and Media Access Control Connectivity Discovery."
Skype for Business Server 2015 Support for LLDP-MED

Skype for Business Server 2015 supports using LLDP-MED for determining locations
only of Skype for Business Phone Edition devices and Skype for Business2015
running on Windows 8. If you need to use switch-level Layer 2 data to determine the
location of other wired PC-based Skype for Business clients, you need to use the
client MAC address method. Because Windows 7 and earlier do not provide support
for LLDP, it makes it harder to obtain detailed locations for wired softphones. While
Skype for Business Phone Edition does support LLDP, you may need to ensure that
switches have been upgrade accordingly to support LLDP.
Use of Third-Party SNMP Solution to Obtain Locations

Some third-party SNMP solutions can support unmanaged access switches.If the
switch that services the Skype for Business client is unmanaged but has an uplink to
a managed distribution switch, the managed switch can report back to the SNMP
application the MAC addresses of the clients connected to the access switch. This
information enables the Location Information Service to identify the location of the
user. However, it is possible to assign only a single Emergency Response Location
(ERL) to all ports on the unmanaged switch, so the location specificity is available
only at the chassis level of the access switch, not the port level.
Automatically Acquiring a Location

If you are setting up your Skype for Business Server infrastructure to support
automatic client location detection, you first need to decide which network
elements you are going to use to map callers to locations. In Skype for Business
Server 2015, you can associate the following Layer 2 and Layer 3 network elements
with locations:
WAP BSSID addresses (Layer 2)
LLDP switch port (Layer 2)
LLDP switch chassis IDs (Layer 2)
IP subnets (Layer 3)
Client MAC addresses (Layer 2)

The network elements are listed in order of precedence. If a client can be located by
using more than one network element, Skype for Business Server uses the order of
precedence to determine which mechanism to use.
If a location is not found, the request can be sent to an external database by using
the Set-CsWebServiceConfigurationcmdlet with the SecondaryLocationSourceUrl
parameter for a web service that can process a location request. This service is only
used when location requests cannot be resolved locally.Clients have no specific
knowledge of which source rendered the location. The LIS will return addresses that
are both validated and not validated.
Using a MAC Address to Find a Location

Resolving a MAC Address
The LIS can use the MAC address of a client to determine the location if it is
configured by using the Set-CsWebServiceConfigurationcmdlet with the
MacResolverUrl parameter. This returns a URL for a web service capable of
performing Media Access Control (MAC) resolution. MAC resolution involves taking
an IP address and determining the MAC address of the network card associated with
that IP address. MAC resolution is used by the Enhanced 9-1-1 service.
Obtaining Port/Switch from a MAC Address

When using a MAC address to determine the location, you need to be aware of
several factors. The MAC address must be resolved into a port/switch because
Skype for Business Server does not natively map MAC addresses to location
information. The process to accomplish this is as follows:
The Skype for Business client sends the MAC address, and then:
24 LIS queries an external MAC resolver through web services.
96 A MAC resolver returns the switch/port information.
97 LIS uses the switch/port to return the location information to the client.
Currently, 911 Enable is the only provider who makes a qualified MAC Resolver
product, called Phone Discovery Manager. However, even without this product, it is
still possible to find the address of the Skype for Business client based on the
network subnet or wireless BSSID. A MAC resolver is useful when you are required to
provide a very specific street address for E9-1-1 compliance, or if a subnet is spread
between multiple civic locations.
E9-1-1 Support for Remote Users

The LIS is not exposed to external users; therefore, it cannot automatically set the
users location. Users have to manually set their location.
Qualified SIP Emergency Response Services provide Emergency Call Response
Centers (ECRC) where attendants can converse with the caller to determine the
most appropriate Public Service Answering Point (PSAP) to route the call. When a
user dials emergency services with a location that cannot be resolved to the
appropriate CAC, the call is routed to the ECRC.
Mobile employees present additional concerns. Because they move around
frequently, sometimes working from a hotel, a customer site, or home, they may not
have up-to-date location information.
User Experience
When a caller calls an emergency number, they expect to be connected PSAP,

immediately. This may not always be the case if the location cannot be determined.
If a user location cannot be determined automatically, the user may be prompted to
enter their location when signing on to Skype for Business or may be connected
through voice or instant messaging to an emergency service provider operator or
the companys security desk.
Manual Location Entry

In organizations with many mobile and remote workers where location cannot be set
automatically, the user can be prompted to enter a location on logging on to the
Skype for Business client. The message can be changed by using the command,
Set-CsEnhancedEmergencyServiceDisclaimer.
Emergency Dialing
When a user dials an emergency number, and the location is automatically set and
validated, the call goes through to the configured PSAP. Otherwise the call is routed
to the emergency services service provider for assistance at establishing a location.
Security Desk Integration

When an emergency call is placed, the company security desk can be notified by
either instance message (IM) or by phone when configuring the location policy. The
IM contain user location information. When configured the emergency callers
location can be used to determine who receives notification, in cases when a
company has different security desks for different locations.
The location policy can also be configured to include a callback number for the
security desk in the record sent to the emergency services service provider or PSAP.
The emergency services service provider can then conference or bridge the security
desk into the call. This can be a one way call where the security desk can only listen
or a two way call where the security desk may also speak.
Note: Some locations only allow one-way bridging, so third parties cannot
interfere with the conversation between an emergency caller and PSAP. Bridging
emergency calls is illegal in other locations.

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PSTN Integration 6-1

Connecting to the PSTN

Private Branch eXchange

An IP PBX is a PBX that supports the IP protocol to connect phones by using an

A legacy PBX is a PBX that does not pass IP packets.

Voice over Internet Protocol (VoIP)

Session Initiation Protocol (SIP)

Internet Telephony Service Provider (ITSP)

Installation, setup, and configuration requirements through a Quick Start Guide.

Release Notes with any known issues.

Documented support process between Microsoft and the vendor.

Enterprise-class standards for audio quality, reliability, and scalability.

Typical Legacy Enterprise PBX

Existing number plan for mapping trunk lines to extensions.

Decision 1: Legacy PBX integration

PBX owns number plan

User can keep existing number, transparent to inbound calls

Internal calling at no cost

Benefit from the existing trunk capacity

Additional PBX cost

Requires PBX configuration

What happens when migration is done?

Migrating a user requires changes in the PBX

Note: Options for connecting through a PBX are covered in lesson 2.

Connecting directly to the PSTN

No additional PBX investments and configuration

o Costs may increase

Migrating a user requires changes at the Provider

Decision 2: POTS/TDM or SIP Trunking

Connecting Through a Gateway

It is more broadly understood.

There is no wide area network (WAN) dependency.

Users have a local carrier choice.

Branch resiliency is available through branch connection to the PSTN.

Connecting through a SIP Trunk

SIP trunking provides the following benefits:

It offers consolidation and numbering flexibility.

It can deploy central trunking for management or routing purpose.

It eliminates per-channel model to provide more flexibility in trunk provisioning.

You do not need an intermediary gateway (ITSP manages PSTN gateway).

It offers the potential to share ISP/ITSP.

Direct Connection Through a Gateway

Functions as a termination point for SIP trunks

be configured according to the PSTN standards of that country/region. Moreover,

Direct Connection Through SIP Trunking

Session Border Controller (SBC)

Non-Encrypted Internet Connection

Connecting to the Existing PBX

Connecting Through PBX by Using SIP

Works only for qualified or supported PBX systems (UCOIP)

Provides voice capabilities between endpoints on either call control server

Provides simplest method of interoperability, relying on standard SIP protocols

Connecting Through PBX by Using a Gateway

As an intermediary in scenarios, such as SIP to TDM or H.323, or to other nonqualified

Simultaneous ringing. Simultaneous ringing enables a user to configure so that

PSTN conferencing. PSTN conferencing enables users to easily create multi-party

Audio conferencing. Audio conferencing provides dial-in conferencing capabilities for

Mobile Users. Mobile users will connect through

trunk to determine a call route to an outgoing trunk.

Media bypass in inter-trunk routing calls is supported.

Inter-trunk routing call authorization scope is at the trunk level.

Incoming PSTN calls to an IP-PBX system through Skype for Business

Outgoing IP-PBX calls to a PSTN network through Skype for Business

Routing of IP-PBX Calls to PSTN by Using Skype for Business