1. PSTN Integration

2. Background Definitions
6: PSTN Integration
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)
These terms have been used in class, but now talk about the role they play in connecting to the PSTN.

3. Qualified Gateway Qualified PBX Supported PBX
UCOIP
6: PSTN Integration
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
Even though Skype for Business Server 2015 provides connections to the PSTN, such as SIP, it is important to use UCOIP qualified and supported hardware, infrastructure, and service providers. Qualification ensures that the hardware supports the protocols needed by Skype for Business Server such that Skype for Business can interoperate.
Qualified gateways and IP-PBX can vary from country to country because of local requirements for connecting to the PSTN.
Demonstration:
This may be a good opportunity to show the students the UCOIP site You should point out that under each component type there are separate lists for Skype for Business Server 2015 and Skype for Business Server Coexistence is one option for connecting to devices not supported by Skype for Business Server 2015.
Qualified
Gateway
Qualified
PBX
Supported
PBX

4. Typical Legacy Enterprise PBX
6: PSTN Integration
PSTN
Dialing Habits
4 digit internal extensions
9 for an outside line
3 digits + extension for other locations
Class of Service
Inbound/Outbound
Local, National
Outbound only
Local, National,
and International
Numbering Plan to
In this topic discuss the typical legacy phone system and point out how:
The dialing rules of the PBX has established user dialing habits
The PBX by design is also a gateway to the PSTN, so it be retained for that purpose or replaced with a new gateway

5. Decision 1: Legacy PBX integration
6: PSTN Integration
Connect Skype for Business directly to the PSTN
Connect Skype for Business
to the Legacy PBX
Details on connecting to the PBX are on the following slides.
PSTN
PSTN

6. Decision 2: POTS/TDM or SIP Trunking
20337B
Decision 2: POTS/TDM or SIP Trunking
6: PSTN Integration
Connecting through a Gateway
For this slide, limit the discussion to the benefits of connecting directly to the PSTN and through an ISTM. The following slides will discuss each architecture option in detail. Also, for clarity, the component names have been removed from this slide. Component names will be included in subsequent slides.
PSTN
Connecting through SIP Trunk
PSTN
SIP
TDM

7. Direct Connection Through a Gateway
20337B
Direct Connection Through a Gateway
6: PSTN Integration
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)
Connection to the PSTN always requires a pool of one or more Mediation Servers. Configuring multiple Mediation Servers and multiple connections to the PSTN will be covered later.
A gateway only translates signals between the PSTN and Skype for Business Server. It does not provide call routing nor number plans.
Multiple gateways can be configured for inbound calls (local numbers) and outbound calls (cheapest route). All routing decisions are made by Skype for Business.
Skype for Business
Mediation
Server
Qualified
PSTN
Gateway
Skype for Business
Pool
PSTN
SIP
TDM

8. Direct Connection Through SIP Trunking
20337B
Direct Connection Through SIP Trunking
6: PSTN Integration
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 (UCOIP)
When using SIP trunking, external telephone traffic is routed to an ITSP. The ITSP is responsible for all connections to the PSTN. The ITSP bundles calls from multiple customers to more efficiently use their connections to the PSTN. Typically, an ITSP has multiple connections in different locations and routes calls to achieve lowest cost.
The slide shows sample architecture for using SIP trunking to connect the ITSP. The exact architecture will vary based on the ITSP requirements. ITSPs are certified through UCOIP. For example, the ITSP will need to be able to parse the SIP header to extract billing information.
Session
Border
Controller
(SBC)
Skype for Business
Mediation
Server
Qualified
IP-PSTN
Gateway
Skype for Business
Pool
PSTN
Enterprise Network
VPN
ITSP Network
SIP
TDM

9. 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.

10. Connecting Through PBX by Using SIP
6: PSTN Integration
PSTN
Skype for Business
Mediation
Server
The PBX provides one or more connections to the PSTN. For incoming calls, the PBX handles routing to all phones directly connected to the PBX and routes calls destined for Skype for Business clients to a Skype for Business Mediation Server.
For outbound calls, the PBX makes the final routing decision. In a multiple branch configuration, the Skype for Business Server can make routing decisions to route a call to the closet or cheapest PBX.
Skype for Business
Pool
Qualified or
supported
IP-PBX
TDM
SIP
IP endpoint

11. Connecting Through PBX by Using a Gateway
6: PSTN Integration
Qualified
IP-PSTN
Gateway
PSTN
If the customer’s existing PBX is not supported by Skype for Business Server 2015, a qualified gateway can be used to interface between the Skype for Business Mediation Server and PBX. The qualified gateway selected must also be compatible with the PBX.
Note: In this module, “gateway” will be used for a component that translates between SIP and PSTN. Gateway functionality can be provided by a qualified gateway or PBX. “Qualified gateway” will be used in those cases where a qualified gateway must be used, and not a PBX.
Routing is again shared between the PBX and Skype for Business Server.
Skype for Business
Pool
Skype for Business
Mediation
Server
TDM or
unsupported
PBX
TDM
SIP
IP endpoint

12. In replacement scenarios, existing call volume is known
20337B
PSTN Sizing
6: PSTN Integration
In replacement scenarios, existing call volume is known
Account for new behaviors and features:
Simultaneous ringing
PSTN conferencing
Dial-in audio conferencing
Mobile users
Use Erlang B calculations when appropriate
Any Erlang B calculator can be used for the Erlang B calculations. The calculator included in the Skype for Business 2010 and 2015 Bandwidth Calculator is included for convenience and the numbers calculated are not used anywhere else in the tool. Most calculators allow you to determine the number of lines, which ultimately determines the number of trunks, if the Erlang number and acceptable failed calls percentage is known. They also allow you to determine the probable percentage of drop calls during peak time if the Erlang number and the trunk capacity are known.
For example if you determine that 72 lines or 3.2 T1 trunks are required, what would be the percentage of failed calls if only 3 T1 trunks are used?

13. Inter-Trunk Routing - 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, 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
The reason for having session-management features in Skype is to provide greater interoperability with many systems without relying on third-party equipment. The ability to aggregate SIP trunks on Skype/Lync and provide a single point for ingress and egress provides a centralized way to manage dial plans and call control. A call coming in over a SIP trunk, whether through a provider, a gateway, or a PBX, can be routed to another gateway, provider, or PBX and does not have to be destined for a user or object on Skype/Lync.

14. 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, 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, that can’t be terminated locally on a client
Media bypass in inter-trunk routing calls is supported
Skype for Business Server 2015 offer the ability to provide class of service to incoming calls on certain trunks, to determine what routing privilege those calls will have for outbound trunk routing. All of the existing concepts – using voice policies, usages, and routes – will be used for trunk configurations. An additional bonus is that media bypass will also be supported on inter-trunk calls. If a call is coming in from a gateway or going out to another gateway, media bypass could enable the two gateways to send media streams to each other while the signaling would continue to flow through the Mediation Server.

15. without Skype for Business Skype for Business Server 2015
Inter-Trunk Routing
6: PSTN Integration
IP-PBX to IP-PBX
Peer to Peer Routing
without Skype for Business
Skype for Business Server 2015
Inter-Trunk Routing
Routing responsibilities are shared between Skype for Business Server 2015 and the PBX. Point out that configuring routing for PBX to PBX is similar to configuring PBX to Skype for Business and Skype for Business to PBX.

16. Inter-Trunk Routing – Signaling and Media Flow
The slide depicts an example of a trunk-to-trunk transfer call. When a call is received from the PBX trunk, the Mediation Server sends the call to the next-hop proxy. The Front End Server then evaluates the usages and finds a list of routes that match the destination of the call. The Front End then follows outbound routing procedures to route that call to a separate trunk and a separate IP-PBX.
Slide Objective:
Speaker Notes
Similarly, if the call is destined to a separate gateway, the media can bypass the Mediation Server and pass directly between the IP-PBX and gateway.
Routing of IP-PBX calls to PSTN via Skype for Business
Incoming call from the PBX trunk
RNL fails
No match to Unassigned Numbers nor Call Park ranges
Validate incoming trunk associated PSTN usages
Determine a route
Apply outbound translation rules
Route to outgoing gateway trunk
Media-bypass possible if IP-PBX supports it
Routing of IP-PBX calls to another IP-PBX system via Skype for Business
Incoming call from the PBX trunk
RNL fails
No match to Unassigned Numbers nor CallPark ranges
Validate incoming trunk associated PSTN usages
Determine a route
Apply outbound translation rules
Route to outgoing PBX trunk via Lync or Skype for Business
Media-bypass possible if both IP-PBX support it

17. Configuring Inter-Trunk Routing
20337B
Configuring Inter-Trunk Routing
6: PSTN Integration
Use the Skype for Business Management Shell
Configure a voice route
Add a PSTN usage to a trunk configuration:
New -PSTNUsages property has been added to CSTrunkConfiguration
Or use the Skype for Business
Control Panel
New-CsVoiceRoute -Identity RedmondRoute
Discuss how to configure inter-trunk routing.
Set-CsTrunkConfiguration –Identity “TrunkId”

18. Collocation vs. Standalone
20337B
Mediation Server
6: PSTN Integration
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?

19. 20337B
Media Bypass
6: PSTN Integration

20. Location Based Routing
6: PSTN Integration
Hyderabad
Skype for Business Pool
Explain the concept of location based policies using the example on the slide. Indian law does not allow calls to be routed using both IP and PSTN. This law applies to both inbound and outbound calls.
Bangalore
Gateway
Hyderabad
Gateway
Skype for Business
Mediation
Server
Skype for Business
Mediation
Server
PSTN

21. Interworking Routing-History
20337B
Interworking Routing-History
6: PSTN Integration
Lync Server 2010:
Multiple PSTN gateways can be associated with the same Mediation Server pool (1:N); a single PSTN gateway is associated with a single Mediation Server pool; a single SIP listening port on the Mediation Server and on the gateway is used in the association.
Lync Server 2013:
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.
Skype for Business Server 2015:
Explain the improvements over the years with Interworking Routing.

22. Trunk and IP-PBX Interworking
SMSGR Readiness
Trunk and IP-PBX Interworking
Start Time xx:xx / Length: 1 minute
Multiple trunks between a Mediation Server and PSTN gateway can be defined to represent IP-PBX SIP termination
Each trunk will be associated with the appropriate route for outbound calls from Mediation Server to IP-PBX
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
Representative Media IP is a per- trunk parameter, allowing for Media Bypass
Mediation Server
IP-PBX
Title: Trunk and IP-PBX Interworking
Length: 1 minute
Instructor Notes:
Lync Server 2013 introduces the concept of Trunks and PSTN gateways  in Lync 2010 the Trunk configuration did not exist, blocking the configuration of multiple connections between Mediation server and gateway
Trunks associate Mediation Servers with PSTN Gateways  connection agreement that defines addresses and ports to be used
Multiple trunks between a Mediation Server and PSTN gateway can be defined to represent IP-PBX SIP termination  this scenario will be detailed in the next slides
Each trunk will be associated with the appropriate route for outbound calls from Mediation Server to IP-PBX  trunks are defined in Topology builder and route to trunk association is defined in the Lync control panel or through PowerShell
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
Representative Media IP is a per-trunk parameter  to address scenarios that have separate IP addresses for media and signaling
Port A
Trunk 1
Port A1
Port B
Trunk 2
Port B1
Port n
Trunk n
Port n1
© 2013 Microsoft Corporation. All rights reserved. Microsoft, Windows, Windows Vista and other product names are or may be registered trademarks and/or trademarks in the U.S. and/or other countries.
The information herein is for informational purposes only and represents the current view of Microsoft Corporation as of the date of this presentation. Because Microsoft must respond to changing market conditions, it should not be interpreted to be a commitment on the part of Microsoft, and Microsoft cannot guarantee the accuracy of any information provided after the date of this presentation. MICROSOFT MAKES NO WARRANTIES, EXPRESS, IMPLIED OR STATUTORY, AS TO THE INFORMATION IN THIS PRESENTATION.

23. Trunk and IP-PBX Interworking-Real Life
6: PSTN Integration
Trunk 1: MS10 to PBX01
PBX01 port: 5060
Signaling IP: PBX-1
Media IP: MTP-1
Mediation Server
(MS10)
When discussing trunks not that a trunk defines a signaling (dashed lines) and media transport (light blue lines).
IP-PBX/Gateway
(PBX01)
Trunk 2: MS10 to PBX01
PBX01 port: 5061
Signaling IP: PBX-1
Media IP: MTP-2

24. Configuration Details
20337B
Configuration Details
6: PSTN Integration
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

25. Trunks and Resiliency Mediation Server MS1 Gateway GW1 Port A Port B
6: PSTN Integration
Mediation
Server MS1
Gateway GW1
Port A
Port B
Discuss the impact of what happens if Mediation Server MS1 is down, then MS2.
Defining multiple trunks from different Mediation servers eliminates the dependency on a single Mediation Server
Defining multiple trunks from one Mediation Server to different gateways eliminates the dependency on a single gateway
Gateway GW1 connects to two Mediation servers, and can be within a site or across sites
Note that a single Mediation Server listening port is needed for trunks to multiple gateways
Resiliency does not require multiple Mediation Server listening ports
The main reason for multiple Mediation Server listening ports is interoperability
Example:
Port C : 5061, can be used as the Mediation Server listening port for Gateway GW1 and Gateway GW2
Port D : 5068 can be used on SAME Mediation Server for a different gateway or IP-PBX if required
Trunk1
Trunk2
Mediation
Server MS2
Gateway GW2
Port C
Port E
Trunk3

26. Multiple Sites to the Same Service Provider
SMSGR Readiness
Multiple Sites to the Same Service Provider
Start Time xx:xx / Length: 1 minute
Another limitation that resulted in the 1-to-N configuration of Lync Server 2010 was centralized SIP trunking. If multiple sites needed to connect to the same service provider, this was not possible, as the service provider’s gateway could not be associated with multiple Mediation Server pools.
Lync Server 2010:
Virtual gateways must be defined to allow connectivity from multiple Mediation Server pools to the same Session Border Controller (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 gateway’s name
Gateway-specific inbound policies cannot be applied when virtual gateways are used (RNL of the IP-address does not resolve to virtual gateway)
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
Lync Server 2013 & Skype for Business Server 2015
Separates PSTN gateways and trunks
Enable you to connect multiple trunks to one gateway  you define one gateway and multiple associations using Trunks
Enables the use of TLS  no additional gateway names that have to be in the certificate
Allows for gateway-specific inbound policies  policies are tied to the trunk instead of the PSTN gateway
SBC
sbc1.provider.com
PSTN
Trunk 1
Trunk 2
MPLS
Site 01
Site 02
Mediation Pool
Mediation Pool
Lync Pool
© 2013 Microsoft Corporation. All rights reserved. Microsoft, Windows, Windows Vista and other product names are or may be registered trademarks and/or trademarks in the U.S. and/or other countries.
The information herein is for informational purposes only and represents the current view of Microsoft Corporation as of the date of this presentation. Because Microsoft must respond to changing market conditions, it should not be interpreted to be a commitment on the part of Microsoft, and Microsoft cannot guarantee the accuracy of any information provided after the date of this presentation. MICROSOFT MAKES NO WARRANTIES, EXPRESS, IMPLIED OR STATUTORY, AS TO THE INFORMATION IN THIS PRESENTATION.

27. M:N Interworking Interworking-Trunk Definition
20337B
M:N Interworking Interworking-Trunk Definition
6: PSTN Integration

28. Auxiliary Calling Information
20337B
Auxiliary Calling Information
6: PSTN Integration
Skype for Business Server 2015
PSTN Phone
+1 (999)
Incoming Call to
+1 (989)
+1 (999)
User Bob
Simultaneous Ring:
INVITE SIP/2.0
FROM:
TO:
HISTORY-INFO:
ms-retarget-reason=forwarding,
P-ASSERTED-IDENTITY:
<tel: >
SIP Header sent to
Use the above example to explain the how simultaneous ring works and how the identity of the original caller is included in the SIP invite.

29. Fast Failover and Options Polling
20337B
Fast Failover and Options Polling
6: PSTN Integration
Gateway Log
Mediation Server Log
1d:0h:12m:15s OPTIONS sip: SIP/2.0
FROM: <sip:se01.tailspin.local:5068;transport=Tcp;ms-opaque=6b773cd98097b3f8>;
epid=BE80B79150;tag=cdee90d70
TO: <sip: >
CSEQ: 3 OPTIONS
CALL-ID: 598db21985cb4d38a5e89a a
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP :59546;branch=z9hG4bK3b462b11
CONTACT: <sip:se01.tailspin.local:5068;transport=Tcp;maddr= >
CONTENT-LENGTH: 0
USER-AGENT: RTCC/ MediationServer
Once every minute SIP OPTIONS poll is sent from the Mediation Server on each trunk
1d:0h:12m:15s SIP/ OK
Via: SIP/2.0/TCP :59546;branch=z9hG4bK3b462b11
From: <sip:se01.tailspin.local:5068;transport=Tcp;ms-opaque=6b773cd98097b3f8>;epid=BE80B79150;tag=cdee90d70
To: <sip: >;tag=1c
Call-ID: 598db21985cb4d38a5e89a a
CSeq: 3 OPTIONS
Contact: <sip: :5060;transport=tcp>
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A

30. Call-Routing Reliability-Lost Connection
6: PSTN Integration
Skype for Business Mediation
Server Pool
MS-02
MS-01
Front-End Server
Qualified
Gateways
GW-01
GW-02
GW-03
options
Route Policy:
For the example session only,
Gateway (GW-01 and GW-03
in that order) can be used
SIP
Configured Trunk
503 response
Control messages
In this example Mediation Server MS-01 is unable to connect to either GW-01 or GW-02, so it returns a 503 response code indicating it cannot service the request. Next Mediation Server MS-02 is tried which successfully connects to gateway GW-01.

31. Call-Routing Reliability—Gateway Down
6: PSTN Integration
Skype for Business Mediation
Server Pool
MS-02
MS-01
Front-End Server
Qualified
Gateways
GW-01
GW-02
GW-03
Route Policy:
For the example session only
Gateway GW-01 and GW-03
in that order can be used
options
504 response
SIP
Configured Trunk
Control messages
In this example GW-01 is down. GW-02 is up but it cannot service the request, because the request can only use gateway GW-01 and GW-03. Next Mediation Server MS-02 is tried which cannot connect to GW- 01, but successfully connects to gateway GW-01.

32. Call-Routing Reliability and Retries
6: PSTN Integration
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

33. Call-Routing Reliability—Next-Hop Proxy
6: PSTN Integration
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

34. Voice Routing Coexistence
20337B
Voice Routing Coexistence
6: PSTN Integration
Outbound Calls
Home Server
Mediation Server
Supported
Skype for Business 2015
2015
Yes
Skype for Business Server 2015 and Lync Server 2013
2013
Skype for Business Server 2015 and Lync Server 2010
2010 No
Inbound Calls
Mediation Server
Next-hop Server
Home Server
Supported
Skype for Business Server 2015
2015
Yes
Skype for Business Server 2015 and Lync Server 2013
2013
Skype for Business Server 2015 and Lync Server 2010
2010

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

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

37. Establishing a call

38. Mid call controls

39. Adding Modalities to a Skype for Business call

40. Adding Modalities (IM)

41. User warned on accepting/placing 2nd call
Multiple Calls
User warned on accepting/placing 2nd call
Lose control of the 1st call from client when second call is started.
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,
Conversation Window will not update to show the accurate status of the call.

42. 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

43. Conversation History Works as expected Inbound missed calls
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.

44. 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  .

45. 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.

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

47. Policy and User configuration