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Configuring Basic Enterprise

Voice Functionality

Voice Routing
msRTCSIP-Line
msRTCSIP-PrivateLine
msRTCSIP-PrivateLine
SIP URI
1

PSTN
PSTN Fallback
Fallback for
for
CAC
CAC and Inbound
Inbound Routing
and Network
Network Routing
Dial Plan 7 Reverse
Reverse Number
Number Lookup
Lookup Outages
User=phone Match
Emergen
Emergen Yes No match
External Normalization
Rules
cy
cy 8 Apply
Apply Called
Called
Call?
Call? Party Prefs
EA
EA Select usages
P?
P? 2 Usages on UC
UC Endpoint
Endpoint Receives
Receives
Internal
Normalization Rules
No 9 From
From trunk
trunk
with inbound Call
Call
with
usages? trunk
usages? Usage from
Emergency
Emergency location
Client-side Yes call?
call?
Global? policy
normalization
normalization 3 Global?
Conferenc
Conferenc
Usages of 10
e dial-out?
dial-out? meeting
e
organizer
RFC
RFC 3966
3966 No Unassigne
Unassigne
d number?
number? Announcement
Starts with + d Announcement or or
Call Park
Call Park Application
Application
Call
Call Park?
Park?
Usages
Location-
Location-
Gateway Mediation from
based
based
Server routing? network
Dial Plan routing?
Referred-
Referred- site
by?
Usages of
by?
referrer Convert
Normalization Rule Convert ##
4 to Local Format
Normalization Rule
Normalization Rule
Usages of 13
caller
Trunk
Trunk Configuration
Configuration //
Number
Number Translation
Translation
Must
Must Match
Match Selected Routes
6 A Rule Call Park Orbit Range policy/usages
PSTN Usage
12 Route
Gateway
Gateway // IP-PBX
IP-PBX // SIP
SIP Trunk
Trunk
5 PSTN Usage
Route
Route
11 PSTN Usage Route External
External Endpoint
Endpoint Receives
Receives
Call
Call
14
Dialing Routing &
Behaviors Authorization

Number Normalization and E.164

Country Code National Destination Subscriber
Code (Optional) number
National (significant) number
1 to 3 digits Maximum = 15 – cc = 12 to 14 digits
31 (Netherlands) 20 (Amsterdam) 500 1500

1 (US) 425 (Washington) 882 8080

Country Code Group Subscriber number
Identification
Code
3 digits 1 digit Max = 15 – (cc + gic)
= 11 digits
599 (Netherlands 7 (Bonaire) 500 1500
Antilles)

Scoping Configuration Items and Policies Global Contoso Site Chicago London Dublin- Pool Chicago-1 Chicago-2 1 User .

unique numbers ^ match the start $ match the end Normalization rules are \d match any digit \d* 0 or more digits specified using \d{5} any 5 digits [135] 1. 3. Dial Plans A set of normalization rules that translate dial strings to full. or 5 regular expressions (13)|(17) 13 or 17 [1-5] 1 through 5 (…) “captures” the enclosed characters for referring to them in the result as $1. . $2. etc. $3.

Normalization and Regular Expressions • Dial plans perform normalization by using regular expressions • Skype for Business Control Panel • Or built from scratch by using standardized regular expressions .

Example Normalization Patterns National dialing ^([2-9]\d\d[2-9]\d{6})$ → +1$1 (NANP) ^0(\d{10}) → +44$1 (UK) Include national and international dialing prefixes ^011(\d*) → +$1 Extension range (e. 15xx-35xx) ^((1[5-9]|2[0-9]|3[0-5])\d{2})$ → +1206555$1 .g.

txt" -Identity Global . Address book normalization No more text file. Yay! Not handled during in-place upgrade Cmdlet for importing existing address book rules Import-CsCompanyPhoneNormalizationRules -Filename "Company_Phone_Number_Normalization_Rules.

Overview of Routing and Authorization Voice Policies PSTN Usages Routes • User authorization • Purpose (usage. • Called number • Class of service caller’s intent) • Cost of call • Voice feature set • Calling location • Priority .

Route Planning International National Premium National Local Internal Routes for the gateways in Munich DE Internal ^\+49895550[12] DE Munich Local ^\+4989 DE Germany ^\+49 DE Europe ^\+(49)|(31)|(33)|(32)|(34)|(351)| … DE International ^\+ .

Voice Policies • Can be assigned per user. <Usage2> • Not only for users. site. Also useful to address Common Area Device requirements: • Assign a Voice Policy to a common area phone to prevent misuse and high cost. • Provides admins with flexibility to control user voice entitlements: . • Can be by PS: • New-CsVoiceRoutingPolicy –Identity <PolicyID> -Name <PolicyName> -PstnUsages <Usage1>. or global.

which are assigned to phone numbers . PSTN usage records do not do anything. local. For them to work.PSTN Usage • Control dialing capabilities (Class of Service) by assigning PSTN usages • A Public Switched Telephone Network (PSTN) usage record specifies a class of call (such as internal. they must be associated with the following: • Voice policies. or long distance) that can be made by various users or groups of users in an organization • By themselves. which are assigned to users • Routes.

a company can restrict calls forwarded by users or through simultaneous ring • Local numbers only.Call Forwarding and Simultaneous Ring • Lync Server 2010: An administrator can enable or disable call forwarding and simultaneous ring through the user voice policy • Lync Server 2013 & Skype for Business: Enables call controls to introduce a flexible call- authorization mechanism for forwarding and simultaneous ring calls • By using this feature. to aid in cost control • Internal Skype for Business users only. for security policies • Any custom authorization rule set up by the .

“Redmond” “International ” “Call PSTN Usages” Call Forwarding “Internal Skype for Business users only” Simultaneous 14 Ring “Custom PSTN Usages” PSTN Usages “Custom Usage” .Call Forwarding and Simultaneous Ring PSTN Usage Voice Policy ‘”Local”.

and one or more PSTN usage records • A route is selected based on a matching pattern • PSTN usages control if a user is allowed to use the route • Routes are associated with one or more trunks defined in Topology builder .Voice Routes • A voice route associates destination phone numbers with one or more public switched telephone network (PSTN) gateways or SIP trunks.

Trunk Configuration • Allow for centrally managing number formatting prior to routing to PBX/PSTN for both the calling and the called number .

Assigning DIDs to a User DID is a term used by the telecommunications industry and stands for Direct Inward Dial (DID): • DID numbers are globally unique • DID ranges/blocks are acquired from the telecom provider • DID numbers enable external users to connect to a Skype for Business user directly • Are assigned to a user when enabling for Enterprise Voice .

ext=8080 • tel:+14258828080 • Recommendation: Specify extensions (ext=) for all users to: • Optimize PIN authorization for devices and dial-in conferencing • May need to deal with “.ext=“ in trunk normalization for cases of PSTN reroute .Specifying a Line URI • DID numbers can be defined in two formats: • tel:+14258828080.

ext=51856 • User C tel:+14258828080.ext=1 • Normalization of the inbound number should add “.ext=1 .ext=51857 • Base number should point to Exchange AA with number: • tel:+14258828080.ext=51855 • User B tel:+14258828080.ext=1” so that the unique number of AA can be found by using a reverse number lookup: • ^(\+14258828080)$ → $1.Internal-Only—Users Without DID • The full URI points to the switchboard or Exchange AA number: • Users will have a unique “ext=xxxx” • Example: • User A tel:+14258828080.

Dial Plan Design Approach • Record all existing dialing habits • Consider the current dial plan • Understand the Gateway and Mediation server locations • Understand the customer requirements .

if needed • Use PSTN usages to “link” appropriate routes to the needed voice policies . Skype for Business and PBX phone? • Keep the existing DIDs or get new numbers when migrated to Skype for Business? • Implement changes or copy the existing numbering plan? • Define the routes • Define user voice policies (Classes of Service) • Incorporate requirements for least-cost routing and PSTN rerouting and fallback.Dial Plan Design Approach • Migration strategy • Skype for Business or PBX Phone vs.

but E.164 31205001500 everybody uses it Prefix an internal PBX-specific Avoid creating number with a PBX.Real World Scenarios and Recommendations • Copying existing dialing habits is not always a good idea • Some are just there to accommodate the PBX Examples of unnecessary dialing rules: Scenario Remark Recommendation Dial a “9” to seize an PBX-specific Avoid creating outside line behavior normalization rules for Example: 9 this habit 0031205001500 Dial “00” for an Country-specific Create a normalization international number behavior rule that translates to Example: 00 Not required. behavior normalization rules for .

Route Planning—A Real World Example Example routes for the gateways in Europe – Germany (DE) – Munich DE Internal ^\+49895550[12] DE Munich Local ^\+4989 International Europe DE Germany ^\+49 National Premium GatewayDE Europe ^\+(49)|(31)|(33)|(32)|(34)|… National DE International ^\+ Local Internal Asia USA International Gateway Gateway National Premium National International Local National Premium Internal National Local Internal .

Number Blocking • Traditional Method • Alternative Method .

Microsoft Official ® Course Voice Applications .

Call Park .Features • Call Park and Retrieve • Orbit (number) returned when call is parked • Parked user is listening to Music on Hold (MoH) • Call can be retrieved from PBX phone dialing orbit • Safe-retrieve: only retrieve my parked call • Ringback • Calls not retrieved are transferred to person who parked the call (after timeout) • Transfer to fallback destination • Calls not retrieved and ringback failed are forwarded to .

Skype for Business Call Parking • A call can be parked if the user is enabled for Call Park functionality • An available orbit is automatically offered to the user parking the call .

Skype for Business Call Retrieval • Dial the orbit like any other extension • Click Retrieve button (performs a safe retrieve) or copy the link into an IM message • Unique ID to identify the call • Parker receives notification of who retrieved the call .

Call Park Ringback After pre-configured timeout (CallPickupTimeoutThreshold ) • Call rings back • User can click the Answer the Call button • Call can be ignored • Call cannot be redirected • Call is not forwarded to voice mail .

Deploying Call Park Services • Call Park services are installed when a server is enabled for Enterprise Voice • Enable Call Park for the end-user in the Voice Policy (disabled by default) .

adatum. or 1-9.000 orbits per range • Should not exceed 50.local • Ranges can be configured in Skype for Business Control Panel • Must start with # or *.Defining Call Park Ranges • Configure orbit range and destination pool (global scope) • Orbit Range should be globally unique • May not include DID numbers Se01. • 0 is not allowed as a starting character • Must be the same length (max.000 orbits per pool • Exclude Call Park orbits from Normalization • Option to use #100 to #200 • A single pool can have multiple orbits . 9 characters) • Should not exceed 10.

except Orbit range New-CsCpsConfiguration New-CsCpsConfiguration -Identity -Identity site:<sitename site:<sitename to to apply apply settings> settings> [-CallPickupTimeoutThreshold [-CallPickupTimeoutThreshold <hh:mm:ss>] <hh:mm:ss>] -[EnableMusicOnHold -[EnableMusicOnHold <$true <$true || $false>] $false>] [-MaxCallPickupAttempts [-MaxCallPickupAttempts <number <number of of rings>] rings>] [-OnTimeoutURI [-OnTimeoutURI sip:<sip sip:<sip URI URI for for routing routing unanswered unanswered call>] call>] . as follows: • Music on Hold can be changed or disabled (service scope) • Ringback attempts (1-10) (site/global scope) • Ringback timeout (10-600s) (site/global scope) • Fallback destination (site/global scope) • All configuration through PowerShell.Call Park Management • Optional settings can be changed.

customize the music on hold OnHoldFile OnHoldFile Set-CSVoicePolicy Set-CSVoicePolicy Configure voice policy to enable Call Park for users .Call Park – Deployment Process create the orbit ranges in the call park orbit table New-CSCallParkOrbit New-CSCallParkOrbit and associate them with the Application service that hosts the Call Park application Set-CsCpsConfiguration Set-CsCpsConfiguration Use the cmdlet to configure Call Park settings Set- Set- CsCallParkServiceMusic CsCallParkServiceMusic Optionally.

Park and Retrieve Call Flow Step 1: • Alice calls Bob. incoming call who is using Skype for Business Server Front End User Bob 2015 incoming call Caller Alice Mediation Server .

Park and Retrieve Call Flow (2 of 7) Step 2: • Alice is now connected to Bob • Media flows from Alice to Bob User Bob ow Front End Fl ia ed M Media Flow Caller Alice Mediation Server .

requesting an orbit ow Fl ia ed M Media Flow Caller Alice Mediation Server . Park and Retrieve Call Flow (3 of 7) Step 3: • Alice wants to speak to Park Call Charlie • Bob issues a call park User Bob command to the Call Park Front End Service.

receiving Music on Hold from the Call Park Service Front End • Bob receives a Call Park User Bob orbit Media Flow Media Flow Caller Alice Mediation Server . Park and Retrieve Call Flow (4 of 7) Step 4: Orbit 123 • Alice is put on hold.

or some Front End User Bob alternate method Media Orbit Flow 123 (paging) Media Flow Caller Alice Mediation Server User . Park and Retrieve Call Flow (5 of 7) Step 5: • Bob shares the Call Park orbit with Charlie through an internal paging system. IM.

Park and Retrieve Call Flow (6 of 7) Step 6: • Charlie dials the orbit number in an attempt to retrieve the parked call Front End Re tri ev Media e 12 Flow 3 Media Flow Caller Alice Mediation Server User .

Park and Retrieve Call Flow (7 of 7) Step 7: • Alice is now directly connected to Charlie Front End Media Flow Media Flow Mediation Server User Caller Alice Charlie .

Purpose of the Unassigned Number Feature • Handles incoming calls to numbers valid to the organization but not assigned to users or (desk) phones • Avoids busy tones or error messages if the user misdials • Incoming calls can be transferred to predetermined: • Phone Numbers • SIP URIs • Voice Mail • Announcement service .

Announcement Service • Create an Announcement through Windows PowerShell New-CsAnnouncement New-CsAnnouncement -Identity -Identity ApplicationServer:se01.local ApplicationServer:se01.com” • TextToSpeechPrompt—A text-to-speech (TTS) prompt • TargetURI—The Uniform Resource Identifier (URI) to which the caller will be transferred after the announcement has been played • At least one Announcement should exist before you can create a number range .com” "sip:brad@tailspin.tailspin. You You will will be be forwarded forwarded to to the the operator" operator" -Language -Language "en-US" "en-US" -TargetUri -TargetUri "sip:brad@tailspin.local -Name -Name "Number "Number Does Does Not Not Exist" Exist" -TextToSpeechPrompt -TextToSpeechPrompt "Welcome "Welcome to to Tailspin. exist. Tailspin.tailspin. the the number number you you dialed dialed does does not not exist.

plan locally.se1. preferably in the same site • Select the previously created Announcement. numbers in use automatically excluded • Destination server is the end point and plays the announcement.com Number does Not Exist . or choose to forward the call to an Exchange Auto Attendant ApplicationServer.adatum.Deploying the Unassigned Number Feature • Create an unassigned number range from the Skype for Business Control Panel • Range may overlap with existing DID.

Unassigned Number Call Flow User Bob Step 2 Front End Tr Ca sfe an ll r Media Flow Step 1 Media Flow Media Flow Step 3 Step 3 Caller Alice Mediation Server User Charlie .

Unassigned Number Call Flow (1 of 3) Step • 1:Alice has dialed a phone number that she believes belongs to Bob • The vacant number routing determines Front End that the dialed number is not a valid User Bob number • Alice is connected to a special RGS Media workflow and is notified that the Flow number is not in use Media Flow Caller Alice Mediation Server .

Unassigned Number Call Flow (2 of 3) Step •2: The special RGS workflow now transfers Alice to Charlie as configured by the vacant number announcement (-TargetURI) User Bob Front End Media Tr Ca an ll Flow sf er Media Flow Caller Alice Mediation Server User Charlie .

Unassigned Number Call Flow (3 of 3) Step 3: • Alice is now connected in a voice call to Charlie Media Flow Media Flow Caller Alice Mediation Server User Charlie .

PSTN Conferencing Features • Meeting Features to handle small/mid-size meetings • DTMF controls • Entry and Exit announcements • Simple join experience • Lobby support for restricted meetings • Unauthorized users wait in the lobby to be admitted • Name recording for unauthenticated users • Integrated seamless with Skype for Business meetings • Scheduling through familiar Skype for Business interface • Access security by PIN and phone number authentication • Meeting prompts and guidance in a language of choice .

Meeting Types • Dial-In Conferencing • Reservation-less calls • Managed events .

max features. and large audiences • 100+ participants • Quarterly or less frequent • Web attached Based on Gartner Study . Meeting Types • Reservation less calls—85% • Weekly staff meetings. average of 3-5 attendees per meeting Target for • Majority of attendees are internal Skype for Business • Frequently contains external attendees • Web attached • Operator-assisted calls—Less than 10% • Biweekly/monthly • Roll call. high touch. project meetings and so on • Typically 25 or fewer participants. and other large meeting features • From 25-100 attendees • Managed event • Web attached ACP—Domain (Audio • Externally focused calls—5% Conferencing Provider) • With transcription. polling.

User Roles & Permissions • Presenter • Controls meeting • Designated by organizer • Can’t designate Federated in advance • Organizer • Implicit role. presenter by definition • If deleted from AD. conferences also removed from RTC database • Attendee • Everyone who is not a presenter • Cannot add content to meeting • Can only download content if given permissions • Can be promoted / demoted .

DTMF Commands • Commands Admin customizable • *1 Automated help Each command can be configured as * / # + 0-9 • *3 Private roll-call Each command can be • *6 Mute/unmute self disabled (unset key mapping) Exposed through PowerShell • *7 Lock/unlock (leaders only) End-user discoverable • *4 Toggle silent Shown on the Dial-in Conferencing webpage mode (leaders only) Discoverable in conference by • *9 Entry/exit issuing Help command (*1) announcements on/off (leaders only) • *8 Open lobby (leaders only) .

Entry/Exit Announcements • Entry/Exit announcements with names • Announcements are made when participants join and leave • Batching reduces the number of announcements • Anonymous PSTN users are prompted to record their names • Authenticated user names are announced by text-to-speech (TTS) • Users can skip name recording and join as unknown participants John (federated user) Jane (PSTN user– anonymous) MCU Alice Authenticated user Bill (PSTN user– authenticated) Simon Anonymous Skype for Business 2015/Skype for Business Web App user .

Entry/Exit Announcements (2 of 2) Controlled by • Admin .Entry/exit announcements configuration: • Off Set-CsDialInConferencingConfiguration Set-CsDialInConferencingConfiguration • Beep -Identity -Identity site:Redmond site:Redmond -EntryExitAnnouncementsType -EntryExitAnnouncementsType "ToneOnly" "ToneOnly" • Name. TTS for known users or Recording for unauthenticated users • Organizer: • Turns announcements on/off at scheduled time for non-default meetings • Presenter: • Turns announcements on/off during the meeting .

Join Experience • Settings related to the join user experience • Default meeting policy (set by administrator. can be changed by user) • Lobby bypass for PSTN users (set by user) .Important Settings .

Deploying PSTN Conferencing Services (1
of 2)
• Plan additional Direct Inward Dialing (DID) numbers
and PSTN trunk capacity for (regional) PSTN access
numbers
• Consider toll free numbers
• Deploy PSTN gateways or configure SIP trunking
• Configure access numbers globally or per site:
• Assign access numbers to conference regions
• Define primary and additional languages (maximum 4)
• Configure dial plans with a valid dial-in
conferencing region
• Dial-in conferencing regions associate a dial plan with one
or more dial-in access numbers

Deploying PSTN Conferencing Services (2
of 2)
• Configure PIN security settings (complexity,
expiration, and so on)
• Generate PIN and send welcome email message by
using the PowerShell script
(SetCsPinSendCAWelcomeMail.ps1)
Set-CsClientPin
Set-CsClientPin -Identity
-Identity “tailspin\holly"
“tailspin\holly" -Pin
-Pin 18723834
18723834

• Enable user for PSTN
dial-in (conferencing policy)

Optional
• Configure DTMF commands globally or per site
• Manage order of access numbers per conference
region (PowerShell cmdlet only)

Managing Conferencing

• How many conferences are happening now?
• Get-CsWindowsService
• The ability to call all Skype for Business services running on
local computer

Audio Conferencing Architecture Skype for Skype for Business Front-End Server Business Back- End Server Web Components (SQL DB) Focus Conferencing (IIS)) Audio Video Database Conferencing Join Launcher Server Focus Factory Reach Server IM Conferencing Server Dial-in Conferencing Web Conferencing Server Page Machine Boundary App Sharing Process Boundary Conferencing Server Conference Announcement Service Web Application Conference Auto Attendant Personal Virtual Assistant Audio Conferencing Group Virtual Assistant .

Multi-Language Support Caller 1 joins and requests English Voice Applications Conference Announcement Service English Caller 2 joins and requests English Group Virtual Assistant (C1/C2) Personal Virtual Assistant (C1) Personal Virtual Assistant (C2) Caller 3 joins and requests German German Group Virtual Assistant (C3) Personal Virtual Assistant (C3) .

Typical PBX deployments Basic PBX features Add-on ACD Dedicated ACD (Basic Hunt Group) solution High scale Fully featured High additional costs Additional licensing costs • • MoH • Basic hunt groups • High scale • Agent sign-in/sign. Small Call Centers . • Busines Superviso • High availability out s hours r • Advanced CDRs • Various hunting • Basic • Live • Interoperable with LoB methods CDRs views applications • Advance d CDRs Departmental solutions Internal Large Call Centers Help desks.

Response Group Features • Interactive Voice Response (IVR) • Call queuing • Routing • Agent-side user experience • Infrastructure .

Positioning Skype for Business Response Groups Basic PBX features Add-on ACD Dedicated ACD (Basic Hunt Group) solution High scale Fully featured High additional costs Additional licensing costs Response Group Service • • High scale Superviso • High availability • Hunt groups and basic r • Advanced CDRs • Live • Interop with LoB apps IVRs views • Integration with Skype for • Advance Business presence d CDRs • Agent anonymity • Announcements Internal Large Call Centers (unassigned numbers) Help • Speech recognition and desks. Small TTS Call • Music on Hold Centers • Basic CDRs .

400 .Response Group Service (RGS) enhancements • RGS has been enhanced to improve scalability in Skype for Business Server • RGS Agent Group: 800 • IVR group: 400 • Agents per pool: 2.

Response Group Management • An Administrator can delegate the management of response groups to a Response Group Manager • The Manager role improves the scalability of a response group deployment by decentralizing the management of the response groups from the administrator • The scope of a Response Group Manager is at a workflow level • A Manager cannot see or modify response groups for which he or she is not a Manager .

Managed and Unmanaged Response Groups Administrator(s) Manager 1 Manager 2 Work Work Work Work FlowManage Flow Manage Flow Flow Unmanage Unmanage d d d d Queue Queue Queue Queue Queue Agent Agent Agent Agent Agent Group Group Group Group Group .

3 all at the same time • Parallel Ring—1 and 2 at the same time (as 3 is in a call) • Longest Idle—Ring 4. Membership can be formal or Ring 2. Added to one or more queues Ring 1. Ring 4. wait 30 seconds. wait 30 seconds. and so on informal • Round Robin—Ring 2. 2. . Ring 1. Ring 2. wait 30 Uses predefined routing methods seconds. and so on • Serial—Always Ring 1. wait 30 seconds. wait 30 seconds. wait 30 seconds. Response Group Building Blocks Agents Target for incoming calls User 1 answered the last Enterprise Voice user(s) call Not a specific RGS object User 2 is the 3rd longest Member of one or more Groups idle User 3 is the 2nd longest Groups idle Ordered list of agents or Exchange Distribution Groups User 4 is the longest idle Routing Method • Attendant Ring—1.

Formal vs. Informal User Groups • Informal User Group membership • User signs in to the Skype for Business client • User is automatically available as an active agent • Formal User Group membership • User signs in to the Skype for Business client • User must sign in again to become an active agent .

Configuring Queues Queues • Holds call until agent pickup • Serviced by one or many groups • Follows each group’s routing sequence • Various configuration options • Queue Overflow Action • Queue Timeout Action • Custom Prompts • Target for a Workflow .

Configuring Workflows .

Sample RGS Scenario .Operator Classic Operator Operator with Fallback Operator with Fallback and After-Hours Service .

Deploying Response Groups Define agent groups Skype for Business Control Panel) Define agent groups Skype for Business Control Panel) Define the workflow (RGS Web Page) .

RGS Call Flow and Agent Anonymity

Ringing RGS Alice calls a Response Group
Call flows differ depending on
Caller Alice
Agent anonymization
Initial call is always targeted at
the Response Group
Establish
ed
RGS Ringing

Caller Alice Agent Bob • RGS alerts one or more agents

No agent Agent anonymization:
anonymization:
Agent answers Agent answers
Alice connects directly Alice remains
RGS no longer part of connected through
the call RGS
Agent is hidden
(anonymous)
Establish
ed Establish Establishe
ed
RGS d
Caller Alice Agent Bob Agent Bob
Caller Alice

RGS Call Flow and Agent Anonymity (1 of
4)

Alice calls a Response Group
Call flows differ depending on
Ringing RGS Agent anonymization
Initial call is always targeted at
Caller Alice the Response Group

RGS Call Flow and Agent Anonymity (2 of
4)

Alice calls a Response Group
Call flows differ depending on
Ringing RGS Agent anonymization
Initial call is always targeted at
Caller Alice the Response Group

Establish
ed
RGS Ringing

Caller Alice Agent Bob

• RGS alerts one or more agents

RGS Call Flow and Agent Anonymity (3 of 4) Ringing RGS Alice calls a Response Group Caller Alice Call flows differ depending on Agent anonymization Initial call is always targeted at Establish the Response Group ed RGS Ringing Caller Alice Agent Bob • RGS alerts one or more agents No agent anonymization: Agent answers Alice connects directly RGS no longer part of the call Establish ed Caller Alice Agent Bob .

RGS Call Flow and Agent Anonymity (4 of 4) Ringing RGS Alice calls a Response Group Call flows differ depending on Caller Alice Agent anonymization Initial call is always targeted at the Response Group Establish ed RGS Ringing Caller Alice Agent Bob • RGS alerts one or more agents No agent Agent anonymization: anonymization: Agent answers Agent answers Alice connects directly Alice remains RGS no longer part of connected through the call RGS Agent is hidden (anonymous) Establish ed Establish Establishe ed RGS d Caller Alice Agent Bob Agent Bob Caller Alice .

• Leverages Call Park application • Similar to. but different from Team Call .Group Call Pickup Feature • Added in February 2013 Cumulative Update for Lync 2013 • Allows any user to pickup calls for their colleagues using their own phones • A user can be a member of only one call pickup group.

Lync 2013.Planning • Components Used • Application service • Call Park application • Skype for Business Server Management Shell • SEFAUtil Resource Kit Utility • Clients • Skype for Business. 2010.Group Call Pickup Feature . Phone Edition • User must be homed on Skype for Business or Lync 2013 Pool with Feb 2013 CU • Users can only be a member of one call pickup group • DR requires admin to repoint orbits .

Group Call Pickup – Capacity Planning Per Front End pool Per Standard Metric (with 8 Front End Edition server Servers) Recommended number of 50 50 users per group Recommended number of 500 60 groups Maximum number of users per pool enabled for Group 25.000 3.000 Call Pickup Maximum rate of incoming calls to total users enabled 500 60 for Group Call Pickup per pool per minute Maximum rate of calls retrieved by users with 200 25 Group Call Pickup per pool per minute .

Deployment • SEFAUtil (dedicated server) • $Site=Get-CsSite –Identity Datacenter1 • New-CsTrustedApplicationPool -Identity "dirsync.com" -Registrar "litwarepool.com" -Site $Site.com /server:pool.Litwareinc.com -Type GroupPickup • Assign Call Pickup Number to Users • SEFAtuil.litwareinc.contoso.Group Call Pickup .com" -Port 7000 • Enable-CsTopology • Configure Call Pickup Number Ranges • New-CsCallParkOrbit -Identity "Redmond Call Pickup" -NumberRangeStart *100 -NumberRangeEnd *199 -CallParkService litwarepool.litwareinc.exe as@contoso.litwareinc.SiteID • New-CsTrustedApplication –ApplicationId "sefautil" – TrustedApplicationPoolFqdn “server.com /enablegrouppickup:*100 .

Group Call Pickup – Call Flow .

Microsoft Official ® Course Configuring and Deploying Emergency Calling .

What Is Location Awareness? • Location Information Service (LIS) • Identifies and populates user location in the client • Affects routing of emergency calls • Provides street address for E911 .

"Location Aware" Emergency Routing • Emergency calls can be routed to specific PSTN gateways by a location policy • A location policy can be assigned to network site • Example: A user roaming outside North America • A user travelling in Ireland dials 112 (for emergency) • The call is routed to a local gateway .

Vacant Number Range PSTN Usage Route PSTN Usage Route 2. Voice Policy Routes Location Policy 1.Voice Routing User User Initiates Initiates Call Call Dial Plan User phone SIP URI Normalization Rule Normalization Rule No No Emergenc Global Global Normalization Rule y y ? ? Call? Call? 404: No Yes matching Call Park Orbit Range Yes Dialing rule Behaviors Reverse Reverse Number Number Lookup Lookup Routing and Authorization No match Match 3. Call Park Orbit PSTN Usage Route Range PSTN Usage Route Mediation Mediation Server Server and and Announcement 403: No Trunk Trunk Configuration Configuration Announcement or or Call route Call Park Park Application Application found Gateway/IP-PBX/SIP Gateway/IP-PBX/SIP Trunk Trunk Inbound Inbound Routing Routing External Endpoint Receives UC Endpoint Receives Call Call Call Call .

E9-1-1 Skype for Business Server 2015 Components Location Policies .

E9-1-1 Configuration 3 4 2 .

Location Discovery 3 2 4 1 Caller .

Placing an Emergency Call 3 3 4a 4b 2 5 1 Security Caller PSAP .

dial string.Location Policy Definition • Defines user experience. and Security Desk notification . routing.

Location Policy Scope • If present. consider the Topology Site Policy. if present. and then the Global Location Policy . apply the policy assigned to user • Or. apply the Global Policy policy assigned to network site Note: This is a “user” policy that you create and assign to the network site • Or.

PSTN Usage Creation • The location policy PSTN usage selects a route for the emergency call • This may be the SIP trunk of the service provider (one usage for all location policies) or it may be a local gateway (a unique usage for each policy) • The PSTN usage is not added to existing Voice policies: • It is used for an emergency call if a user has a location policy applied • You can create PSTN usage in either Skype for Business Server Control Panel or Skype for Business Server PS>Set-CsPstnUsage PS>Set-CsPstnUsage –Usage Management –Usage Shell @{add="EmergencyCallsUsage"} @{add="EmergencyCallsUsage"} .

fabrikam.net} Name : EmergencyCallsRoute SuppressCallerId : AlternateCallerId : .net"} Identity : EmergencyCallsRoute Priority : 3 Description : NumberPattern : ^\+911$ PstnUsages : {EmergencyCallsUsage} PstnGatewayList : {PstnGateway:e911gw.fabrikam.Voice Route Creation • The Voice Route allows calls to the Emergency Number to go through the PSTN Gateway that references the Emergency Service Provider Note: It does not differ from any other Voice Route PS>New-CsVoiceRoute –Name "EmergencyCallsRoute" -NumberPattern "^\ +911$" –PstnUsage @{add="EmergencyCallsUsage"} –PstnGatewayList @{add="e911gw.

net information to the Identity : E911 certified Service:PstnGateway:e911gw. PIDF-LO Support on Trunk • To send location PS>Get-CsTrunkConfiguration Service:PstnGateway:e911gw.fabrikam. PIDFLO SipResponseCodeTranslationRulesList : {} Description : information must be ConcentratedTopology EnableBypass : True : False allowed to transit EnableMobileTrunkSupport : False EnableReferSupport : True through the Skype for EnableSessionTimer EnableSignalBoost : False : False Business Trunk MaxEarlyDialog RemovePlusFromUri : 20 : True RTCPActiveCalls : True • Ensure that RTCPCallsOnHold : True SRTPMode : Required EnablePIDFLOSupp EnablePIDFLOSupport : True ort on the Trunk is set to True Set-CsTrunkConfiguration -Identity "Service:e911gw.net" -EnablePIDFLOSupport $true .fabrikam.net OutboundTranslationRulesList : {} provider.fabrikam.

Location Information Service • Part of Skype for Business Server 2015 web services components: • Load-balanced within a cluster for high availability • Precedence of matching client location requests: • Basic service set identifier (BSSID) of Wi-Fi Access Point • Switch/port from Link Layer Discovery Protocol Media Endpoint (LLDP-MED) • Switch from LLDP-MED • Subnet • Media Access Control (MAC) match of .

Location Information Service (continued) • Microsoft Windows PowerShell and GUI- based administration • Follows NENA “i2” reference architecture for address validation • Follows IETF PIDF-LO standards with extensions for location format • Other vendor/in-house LIS can be integrated .

csv files • Address management–related administrative tasks include: • Configuring the address validation service provider • Uploading validation credentials .Configuring Location Information Server • Network identifiers are associated with street addresses: • BSSID • Subnet • Switch • Switch/Port • The size of this data set will correspond to how detailed the locations are and whether wireless is within scope • You can use Windows PowerShell scripts to import this data from .

16. The client uses its own IP/subnet mask to determine its subnet.0 -Description "Munchen" -Location "Munchen" -CompanyName "Fabrikam" -HouseNumber 2 -StreetName "Lindenstrasse" -City "Munchen" -Country DE • Publish the LIS configuration Publish-CsLisConfiguration .20. and then sends this in the Web Query to the LIS Web Service Set-CsLisSubnet -Subnet 172.Location Information Server—Subnet • Configure the LIS subnet Note: There is no subnet mask.

Address Status

• Every address added to the LIS database
should be validated by a Master Street
Address Guide (MSAG)
• MSAG validation ensures that emergency
calls can be correctly routed

Client Location Request

• Location Request includes the following
information:
• Link Layer Discovery Protocol (LLDP) from Layer 2
connection–switch and port IDs (if available)
• Subnet
• WAP BSSID (if available)
• MAC address
• LLDP is not supported for Skype for Business on
Windows 7 and earlier:
• This makes it difficult to get detailed locations for wired
softphones
• You will need to use an SNMP application to do this
• LLDP is supported on Lync Phone Edition:
• Switches may need upgrades to support LLDP

Automatically Acquiring a Location

• Location database is global–each LIS has all
defined locations
• Client automatically initiates a “location request”
to its LIS:
• Includes its network connectivity data
• LIS location matches precedence:
• Wi-Fi AP BSSID
• LLDP-MED Switch/Port
• LLDP-MED Switch
• Subnet
• MAC (If configured through Set-
CsWebServiceConfiguration–MacResolverUrl)
• If a location is not found, the request can be sent to
an external database
(Set-CsWebServiceConfiguration –SecondaryLocationSourceUrl)

and then: 1) LIS queries external MAC Resolver through Web Services 2) MAC Resolver returns switch/port 3) LIS uses switch/port to return location information to client • This raises a question: What Is a MAC Resolver? • It is an appliance provided by the Service .Using a MAC Address to Find a Location • Skype for Business does not natively map the MAC address to Location information • LIS must resolve the MAC address into switch/port for resolution: • Skype for Business client sends MAC address.

and then transfers the call to the correct Public Service Answering Point (PSAP) .E9-1-1 Support for Remote Users • LIS Web Service is not exposed to external users • How it works: 1) Users self-report their location 2) The call is automatically routed to the Emergency Call Response Center (ECRC) with location data 3) The ECRC confirms the location with caller.

Manual Location Entry • Location Policy field LocationRequired impacts user experience: • Location Required = No: • User not prompted • Location Required = Yes: • UI highlights location with “X” and “!” for emphasis • Can be dismissed without warning • Location Required = Disclaimer: • UI highlights location with “X” and “!” for emphasis • Disclaimer shown when dismissed • Only shows during sign in. No effect on call .

Emergency Dialing • User dials emergency number • Call goes straight through to the correct PSAP: • If the location has been validated .

Security Desk Integration • IM automatically established between emergency caller and security desk: • Location information is contained in the conversation window • An E.164 number can be bridged onto emergency calls • One-way/two-way • Partner is responsible for initiating conference .

Microsoft Official ® Course PSTN Integration .

Background Definitions • Public switched telephone network (PSTN) • Private Branch eXchange (PBX) • Voice over Internet Protocol (VoIP) • Session Initiation Protocol (SIP) • Internet Telephony Service Provider (ITSP) • Uniform Resource Identifier (SIP URI) • Multiple Points of Presence (MPOP) .

UCOIP • Skype for Business Certification Program • Testing and qualification of third party solutions for interoperability with Microsoft UC • Independent testing by third party labs based on standards based open documentation • SIP trunking providers supported with Lync Server 2013 will be supported with Skype for Business Supporte Qualified Qualified d Gateway PBX PBX .

Local.Typical Legacy Enterprise PBX PSTN Numbering Plan 31-20-500 1000 to +31-20-500 1999 Class of Service Class of Service Outbound only Inbound/Outbound Local. National and International Dialing Habits 4 digit internal extensions 9 for an outside line 3 digits + extension for other locations . National.

Decision 1: Legacy PBX integration Connect Skype for Business Connect Skype for Busines directly to the PSTN to the Legacy PBX PST PST N N .

Decision 2: POTS/TDM or SIP Trunking Connecting through a Gateway PSTN Connecting through SIP Trunk PSTN SIP TDM .

Direct Connection Through a Gateway • A gateway is a physical device that connects two incompatible networks • The gateway translates signaling and media between Skype for Business (SIP) and the PSTN • Use supported gateways (UCOIP) Skype for Business Qualified Skype for BusinessMediation PSTN Pool Server Gateway PSTN SIP TDM .

Direct Connection Through SIP Trunking • IP connection that establishes a SIP communications link between your organization and an Internet telephony service provider (ITSP) beyond your firewall • Use supported SIP Trunking Provider Session (UCOIP) Skype for Business Border Qualified Mediation Skype for Business Controller IP-PSTN Server Pool (SBC) Gateway PSTN Enterprise Network VPN ITSP Network SIP TDM .

.User Configuration Skype for Business and PBX phone numbers can be the same Configuration is roamed for MPOP endpoints. saving state of CallViaWork at the endpoint & whether it’s in use.

Connecting Through PBX by Using SIP PSTN Skype for Business Skype for Business Mediation Server Pool Qualified or supported IP-PBX SIP TDM IP endpoint .

Connecting Through PBX by Using a Gateway Qualified PSTN IP-PSTN Gateway Skype for Business Pool Skype for Business Mediation Server TDM or unsupported PBX SIP TDM IP endpoint .

Use Erlang B calculations when appropriate . In replacement scenarios.PSTN Sizing 1. existing call volume is known 2. Account for new behaviors and features: • Simultaneous ringing • PSTN conferencing • Dial-in audio conferencing • Mobile users •.

the following routing paths (among others) are enabled: • Incoming PSTN calls to an IP-PBX system via Lync • Outgoing IP-PBX calls to a PSTN network via Lync • Outgoing IP-PBX calls to another IP-PBX system via Lync .Overview • Skype for Business Server 2015 supports call routing from an incoming trunk to an outgoing trunk to provide routing functionalities to other telephony systems • A possible alternative for PBX Integration scenario’s • By enabling inter-trunk routing.Inter-Trunk Routing .

Inter-Trunk Routing – Description • Skype for Business Server 2015 allows to the associate a set of PSTN usages on an incoming trunk to determine a call route to an outgoing trunk • These PSTN usages are used to determine destination for incoming call on a trunk. if the call can’t be terminated locally • No local client or other entity is found (essentially. that can’t be terminated locally on a client • Media bypass in inter-trunk routing calls is supported . the RNL fails) • No match to CallPark range or Unassigned Numbers range • Inter-trunk routing call authorization scope is at the trunk level • The same call authorization applies to all incoming calls arriving via the trunk.

Inter-Trunk Routing IP-PBX to IP-PBX Skype for Peer to Peer Routing Business without Skype for Business Server 2015 Inter-Trunk Routing .

Inter-Trunk Routing – Signaling and Media Flow Routing of IP-PBX calls to PSTN via Skype for Routing of IP-PBX calls to another IP-PBX system Business via Skype for Business • Incoming call from the PBX trunk • Incoming call from the PBX trunk • RNL fails • RNL fails • No match to Unassigned Numbers nor Call • No match to Unassigned Numbers nor CallPark Park ranges ranges • Validate incoming trunk associated PSTN • Validate incoming trunk associated PSTN usages usages • Determine a route • Determine a route • Apply outbound translation rules • Apply outbound translation rules • Route to outgoing PBX trunk via Lync or Skype for • Route to outgoing gateway trunk Business • Media-bypass possible if IP-PBX supports it • Media-bypass possible if both IP-PBX support it .

com"} • Add a PSTN usage to a trunk configuration: • New -PSTNUsages property has been added Set-CsTrunkConfiguration –Identity “TrunkId” -PstnUsages to CSTrunkConfiguration @{add=”Redmond”} • Or use the Skype for Business Control Panel .Configuring Inter-Trunk Routing • Use the Skype for Business Management Shell New-CsVoiceRoute -Identity RedmondRoute -PstnUsages • Configure a voice route @{add=“Redmond"} -PstnGatewayList @{add="PstnGateway:redmondgw1.contoso.

Mediation Server

• Collocation vs. Standalone
• Collocation can offer significant server count
reduction
• Standalone may be preferable for network zone
placement or workload isolation
• Media Bypass and Scalability
• Scale based on hardware and transcoding mix
• For planning, do not count calls with media
bypass
• Pool vs. Single Server
• Can gateway or SIP trunk support DNS load
balancing?

Media Bypass

Location Based Routing

Hyderabad
Skype for Business Pool

Bangalore Hyderabad
Gateway Gateway

Skype for Business Skype for Business
Mediation Mediation
Server Server

PSTN

Skype for Business Server 2015: Introduces M:N Interworking routing. A particular PSTN gateway can be associated with multiple Mediation Server pools or the same Mediation Server pool with multiple unique associations. a single PSTN gateway is associated with a single Mediation Server pool.Interworking Routing-History Lync Server 2010: Multiple PSTN gateways can be associated with the same Mediation Server pool (1:N). A particular PSTN gateway can be associated with multiple Mediation Server pools or the same Mediation Server pool with multiple unique associations. Lync Server 2013: Introduces M:N Interworking routing. . a single SIP listening port on the Mediation Server and on the gateway is used in the association.

dial plan can be scoped per trunk • Representative Media IP is a per- trunk parameter. per-trunk policy will be applied • Trunk configuration will be scoped globally or per trunk. similarly.Trunk and IP-PBX Interworking • Multiple trunks between a Mediation Server and PSTN gateway can be Mediation IP-PBX defined to represent IP-PBX SIP Server Port A Trunk 1 Port A1 termination Port B Trunk 2 Port B1 • Each trunk will be associated with the appropriate route for outbound Port n Trunk n Port n1 calls from Mediation Server to IP-PBX • For inbound calls. allowing for Media Bypass .

Trunk and IP-PBX Interworking-Real Life Trunk 1: MS10 to PBX01 PBX01 port: 5060 Mediation Server Signaling IP: PBX-1 (MS10) Media IP: MTP-1 IP-PBX/Gateway (PBX01) Trunk 2: MS10 to PBX01 PBX01 port: 5061 Signaling IP: PBX-1 Media IP: MTP-2 .

Configuration Details • Topology Builder: • Define the PSTN Gateway and Trunks • Define the MTP as the Alternate Media IP address • Use different gateway listening ports for each trunk • Publish the topology • Windows PowerShell: • Identify the trunk IDs • Use Windows PowerShell to configure media IP addresses for the remaining trunks • Verify the media IP address for the trunks .

Trunks and Resiliency Mediation Server MS1 Port A Gateway GW1 Port B Trunk1 Trunk2 Mediation Server MS2 Gateway GW2 Port C Port E Trunk3 .

provider.com address • TLS cannot be used because the SBC certificate Trunk 1 Trunk 2 does not contain the virtual gateway’s name MPLS • Gateway-specific inbound policies cannot be Site 01 Site 02 Mediation Pool Mediation Pool applied when virtual gateways are used (RNL of the IP-address does not resolve to virtual gateway) Lync Pool Lync Server 2013 & Skype for Business: • Separates PSTN gateways and trunks • Enable you to connect multiple trunks to one gateway • Enables the use of TLS • Allows for gateway-specific inbound policies .Multiple Sites to the Same Service Provider Lync Server 2010: • Virtual gateways must be defined to allow connectivity from multiple Mediation Server pools to the same Session Border Controller (SBC) FQDN SBC • Virtual gateway FQDNs all resolve to the same IP PSTN sbc1.

M:N Interworking Interworking-Trunk Definition .

168.local.user=phone sip:+19995552001@contoso.1.user=phone ms-retarget-reason=forwarding.1.user=phone HISTORY-INFO: HISTORY-INFO: sip:+19895550200@se01.contoso.com.local.user=phone sip:+19995551000@192.com.41.41.user=phone SIP/2.user=phone sip:+19995551000@192.local.user=phone P-ASSERTED-IDENTITY: P-ASSERTED-IDENTITY: <tel:+19995552001> <tel:+19995552001> SIP Header sent to 19995551000 .1.41.contoso.0 SIP/2.168.user=phone sip:+19995551000@se01.local.contoso. ms-retarget-reason=forwarding.user=phone sip:+19895550200@se01.user=phone 1000 TO: TO: sip:+19995551000@192.contoso. sip:+19995551000@se01.41.168.0 +1 (999) 555 FROM: FROM: sip:+19995552001@contoso.Auxiliary Calling Information Skype Call Incoming for Business Server 2015 to +1 (989) 555 PSTN Phone 0200 +1 (999) 555 2001 User Bob +1 (989) 555 Simultaneous 0200 Ring: +1 (999) 555 1000 PSTN Phone INVITE INVITE sip:+19995551000@192.1.168.

BYE.OPTIONS.005 .5.PRACK.UPDATE Server: Audiocodes-Sip-Gateway-/v.41> CSEQ: 3 OPTIONS CALL-ID: 598db21985cb4d38a5e89a410987464a MAX-FORWARDS: 70 VIA: SIP/2.transport=Tcp.1. epid=BE80B79150.ACK.INFO.transport=tcp> Supported: 100rel Allow: REGISTER.maddr=192.168.168.epid=BE80B79150.41>.transport=Tcp.1.SUBSCRIBE.NOTIFY.branch=z9hG4bK3b462b11 CONTACT: <sip:se01.053.local:5068.168.tailspin.0 200 OK Via: SIP/2.local:5068.tailspin.tailspin.tag=cdee90d70 TO: <sip:192.1.REFER.0 MediationServer • Mediation Server Log 1d:0h:12m:15s SIP/2.tag=cdee90d70 To: <sip:192.168.168.0/TCP 192.1.41:5060.0.168.Fast Failover and Options Polling • Gateway Log 1d:0h:12m:15s OPTIONS sip:192.local:5068.branch=z9hG4bK3b462b11 From: <sip:se01.0.1.168.1.transport=Tcp.tag=1c1952373857 Call-ID: 598db21985cb4d38a5e89a410987464a CSeq: 3 OPTIONS Contact: <sip:192.52:59546.0 FROM: <sip:se01.41 SIP/2.CANCEL.80A.INVITE.1.52> CONTENT-LENGTH: 0 USER-AGENT: RTCC/5.ms- opaque=6b773cd98097b3f8>.0/TCP 192.52:59546.ms-opaque=6b773cd98097b3f8>.

Gateway (GW-01 and GW-03 options in that order) can be used GW-01 options 503 response MS-01 options GW-02 options Front-End Server MS-02 GW-03 SIP Configured Trunk Control messages . Call-Routing Reliability-Lost Connection Skype for Business Mediation Qualified Route Policy: Server Pool Gateways For the example session only.

Call-Routing Reliability—Gateway Down Skype for Business Mediation Qualified Route Policy: Server Pool Gateways For the example session only Gateway GW-01 and GW-03 options in that order can be used GW-01 options 504 response MS-01 GW-02 options Front-End Server MS-02 GW-03 SIP Configured Trunk Control messages .

Call-Routing Reliability and Retries Skype for Business Server 2015 (FE) (MS) (GW1) (GW2) Invite (trunk 1) 10-sec timer-1: starts 183 response Timer-1: continues Failed Connection Cancel (trunk 1) Timer-1: expires Invite (trunk 2) 10-sec timer-1: starts 183 response Timer-1: continues Invite 18x response 18x response Timer-1: stops .

so if the primary next-hop proxy is used for a call and no SIP response is received within this time. the call is .Call-Routing Reliability—Next-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.

Voice Routing Coexistence Home Mediation Outbound Calls Server Server Supported Skype for Business 2015 2015 2015 Yes Skype for Business Server 2015 2013 Yes 2015 and 2013 2015 Yes Lync Server 2013 Skype for Business Server 2015 2010 Yes 2015 and 2010 2015 No Lync Server 2010 Next- Mediatio Home Inbound Calls n Server hop Server Supported Server Skype for Business Server 2015 2015 2015 Yes 2015 Skype for Business Server 2015 2015 2013 Yes 2015 and 2013 2013 2015 Yes Lync Server 2013 Skype for Business Server 2015 2015 2010 Yes 2015 and .

Expanding Voice interoperability to the PBX phone Skype Voice for PBX Users End-users can make voice calls using any PSTN phone.Lesson 5: Call via Work . then connects with far-end dest Features Presence update & call control from rich client Mid-call control capabilities preserved on PBX phone . including existing PBX end Leverages existing Direct SIP connectivity between PBX systems and Skype for Bu User Experience Server dials out to PSTN or Deskphone number to connect user.

PBX routes call and local user answers. Skype Server Pool 4 PBX 4. User instantiates call from Skype rich client 2. When Server sees this call answered.Components 1. Skype for Business Server places call Destination 6 PSTN to user’s PBX station set (or to any other PSTN phone number) 5 3. PBX routes call out to PSTN with user’s Far-end call DID (or to any other local PBX endpoint) Skype for Business PBX Station 6. Far-end call answers & call is established with client acting as control channel . 2 places far-end call. Call via Work . Here the server will Local call use PBX user’s DID as ANI 1 3 5.

Establishing a call .

Mid call controls .

Adding Modalities to a Skype for Business call .

Adding Modalities (IM) .

• Remote participant activity • Remote participant may accept or place another call from/to someone • This will make the call on PBX Phone go on hold for the local user. Multiple Calls • User warned on accepting/placing 2nd call • Lose control of the 1st call from client when second call is started. . • Conversation Window will not update to show the accurate status of the call.

Ending a call • Placing the receiver of the PBX phone on the handset • Clicking the hang-up button • Close out (“x”) on the Conversation Window .

Conversation History • Works as expected • The initial inbound calls are not shown in Conversation History view. . • Inbound missed calls • PBX or Gateway should support Reason header “Call completed elsewhere” in the CANCEL message • If PBX does not send this Reason header. Server will treat incoming call as missed.

Meetings • Client will prompt for meeting join preference • Dialog auto-populated • Focus dials out to user’s CvW configured number . • Click to Join • Meet Now & Ad-hoc Group Call • Ad-hoc incoming group calls .

. Inbound Calls • Call via Work is Outbound Only • Inbound experience to both client & phone achieved when Skype is first in line & forwarded with Call FW settings • When PBX is first in line. inbound call will land only on desktop phone.

Presence Scenario Behavior Outbound CvW Call Presence will change to “In a Call” Presence will change to “In a Conference Outbound Meet Now / Group Call Call” Inbound CvW Call – Answered on PBX No change to presence Inbound CvW Call – Answered on Skype Presence will change to “In a Call” Presence will change to “In a Conference Inbound Meet Now / Group Call Call” .

Policy and User configuration .

System Center. The information herein is for informational purposes only and represents the current view of Microsoft Corporation as of the date of this presentation. Dynamics and other product names are or may be registered trademarks and/or trademarks in the U. . MICROSOFT MAKES NO WARRANTIES. Microsoft. it should not be interpreted to be a commitment on the part of Microsoft. AS TO THE INFORMATION IN THIS PRESENTATION. Azure. and/or other countries.©2013 Microsoft Corporation. Because Microsoft must respond to changing market conditions. Office. Windows. and Microsoft cannot guarantee the accuracy of any information provided after the date of this presentation. All rights reserved. IMPLIED OR STATUTORY. EXPRESS.S.