Deploy Cisco Webex Calling v2 VAR Channel - PDF

Lab Guide
Cisco dCloud
Deploy Cisco Webex Calling v2

(VAR Channel)
Last Updated: 14-JANUARY-2020
About This Lab

This guide for the preconfigured demonstration includes:
About This Lab
Limitations
Requirements
About This Solution
Topology
Session Users
Get Started
Initial Cisco Webex Control Hub Organization Verification
Webex Calling – Calling Features
Deploying Local Gateway for PSTN Calling
Appendix A. Cisco Webex Password Reset Procedures
Appendix B. Full Local Gateway Configuration
What’s Next?
© 2020 Cisco and/or its affiliates. All rights reserved. This document is Cisco Public Information. Page 1 of 71
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Limitations
To meet the specified requirements of this lab, these scenarios are intended to demonstrate one way to configure Cisco® Webex
Control Hub solutions. There are various ways this can be accomplished, depending on the situation and the customer's
goals/requirements. Please ensure that you consult all current official Cisco documentation before proceeding with a design or
installation. This lab is primarily intended to be a learning tool and may not necessarily follow best practice recommendation at all
times, in order to convey specific information.
Customization Options
There are no customizable options associated with this lab.
Requirements
The table below outlines the requirements for this preconfigured demonstration.
Required Optional
Cisco AnyConnect Client None
Laptop
Cisco Multiplatform Phone (MPP) (for Cisco Webex® Calling)
Cisco EndPoint (for Cisco Webex Calling)
Alternatively, you can use desktop or mobile calling clients
About This Solution

Cisco Webex Teams Overview
Cisco Webex is an app-centric cloud-based service that provides a complete collaboration suite for teams to create, meet, message,
call, whiteboard, and share, regardless of whether they’re together or apart; in one continuous workstream before, during, and after
meetings. It is built to make teams work seamlessly. It is a simple, secure, complete, and open service that enables people to work
better.
Communication needs to be agile, mobile, and collaborative – all thanks to mobile devices and evolving innovations in infrastructure
and applications. The Cisco Webex service makes instant communications and live meetings possible through a deeply integrated set
of industry-leading communications tools for an unmatched collaboration experience—that only the Cisco cloud can deliver. Webex
Teams™ has a continuous delivery model, which means features are released on a regular basis. Check help.webex.com to keep
updated of all the latest info.
Cisco Webex Calling Overview
Webex® Calling is a collaboration Flex Plan offering that provides multi-tenant cloud-based alternative to on-premise PBX phone
service. It is branded by Cisco and provisioned by partner value-added resellers (VARs). VARs are not regulated service providers and
cannot provide PSTN service. PSTN service must be leveraged through an enterprise local gateway deployment. Webex Calling also
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provides service assurance through media quality metrics. The Webex Calling offer can be bundled with Webex Teams and
Meetings.
The focus of this lab is the local gateway to provide PSTN access to Webex Calling organizations. The local gateway can be deployed
standalone (focus of this lab) or in deployments where integration into Cisco Unified Communications Manager is required.
Topology
This demonstration includes several server virtual machines. Most of the servers are fully configurable using the administrative-level
account. Administrative account details are included in the lab guide steps, where relevant, and in the server details table.
dCloud Topology
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Equipment Details
Name Description Host Name (FQDN) IP Address Username Password

Local Gateway CSR1000V (IOS-XE 16.9.4) N/A 198.18.133.226 admin dCloud123!
PSTN Services SIP Service Provider N/A 198.18.133.3 N/A N/A
Local Gateway CSR1000V N/A 198.18.133.226 admin dCloud123!
AD1 Active Directory, DNS, AD FS ad1.dcloud.cisco.com 198.18.133.1 administrator C1sco12345
Exchange Microsoft Exchange 2016 mail1.dcloud.cisco.com 198.18.133.2 administrator C1sco12345
Workstation 1 Windows 10 wkst1.dcloud.cisco.com 198.18.1.36 cholland C1sco12345
Workstation 2 Windows 10 wkst2.dcloud.cisco.com 198.18.1.37 aperez C1sco12345
Workstation 3 Windows 10 wskt3.dcloud.cisco.com 198.18.1.38 kmelby C1sco12345
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Session Users
The following table contains details on preconfigured users available for your session.
User Details
Deployment
User Name User ID Password Endpoint Devices Internal Extension Model
Charles Holland cholland C1sco12345 Cisco Webex Teams 6018 Hybrid
Anita Perez aperez C1sco12345 Cisco Jabber®/Teams 6017 Hybrid
Kellie Melby kmelby C1sco12345 Cisco Jabber/Teams 6050 Hybrid
Taylor Bard tbard C1sco12345 Webex Calling endpoint or Cisco Calling 6021 Cloud
App
Rebekah Barretta rbarretta C1sco12345 Webex Calling endpoint or Cisco Calling 6022 Cloud
App
Stefan Mauk smauk N/A Webex Calling endpoint or Cisco Calling 6023 Cloud
App
Get Started
BEFORE PRESENTING
Cisco dCloud strongly recommends that you perform the tasks in this document with an active session before presenting in front
of a live audience. This will allow you to become familiar with the structure of the document and content.
It may be necessary to schedule a new session after following this guide in order to reset the environment to its original
configuration.
PREPARATION IS KEY TO A SUCCESSFUL PRESENTATION
Follow the steps to schedule a session of the content and configure your presentation environment.
1. Initiate your dCloud session. [Show Me How] (Skip if you are in a proctored environment.)
NOTE: It may take up to 45 minutes for your session to become active.
2. Click View to open the active session. (Skip if you are in a proctored environment.)
3. If you are connecting directly to the session from a stand-alone laptop or other device, install and access Cisco AnyConnect on
your laptop, using the AnyConnect credentials in the Cisco dCloud UI. [Show Me How]
• Recommended method: Use Cisco AnyConnect [Show Me How] and local laptop RDP client. [Show Me How]
• Windows Users: It is recommended to use some version of Remote Desktop Manager to save the connections to each
virtual machine. An example of a manager is the Microsoft Remote Desktop Connection Manager, which can be found at
https://www.microsoft.com/en-us/download/details.aspx?id=44989.
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• Mac Users: It is recommended to use the Microsoft Remote Desktop (MRD) [ ] or CoRD [ ] applications to
connect to the virtual machines. MRD can be downloaded for free from the Mac App Store. CoRD can be downloaded free
at http://cord.sourceforge.net/. Both application gives you the capability to save the connections for each virtual machine.
DO NOT use the Microsoft remote desktop client that comes with your Mac. It will have security issues connecting to the
AD1 and Mail1 virtual machines.
4. After confirming the devices have the correct firmware and are not in a factory reset state, then perform a factory reset on each
device before starting the lab. (Skip if you are in a proctored environment.)
NOTE: For best results, use either the Firefox or Chrome web browsers.
5. In order to run this lab, you need some information from the Session Details tab on your dCloud dashboard session page.
Obtain the Collaboration Edge domain information. Each session has a unique domain. The image below is only an example.
Do not use the information in the image below for your session. It is highly recommended to take note of your information
now so that you can refer to it throughout the lab.
Session Details Tab Example
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Initial Cisco Webex Control Hub Organization Verification
Value Proposition: Cisco Webex Control Hub is a web-based, intuitive, single-pane-of-glass management portal that enables you to
provision, administer, and manage Cisco Webex services and Webex Hybrid Services, such as Hybrid Call Service, Hybrid Calendar
Service, Hybrid Directory Service, and Video Mesh.
In this scenario, you will verify the initial administrator account. The trial has already been started for you and the users have already
been created.
Charles Holland, the org administrator, has his password already set to dCloud123! The steps to create a customer trial can be found
at https://collaborationhelp.cisco.com/article/en-us/npv2y12.
Steps
You will now verify the initial automation scripts ran properly to create your trial and set Charles’ password.
1. If not already connected, use Cisco AnyConnect to VPN into your lab session. The login credentials will be provided by your
proctor or displayed on the session details tab for your active session.
2. Create an RDP connection to Workstation 1 (198.18.1.36) and log in as dcloud\cholland with password C1sco12345.
3. Open Microsoft Outlook using the icon in the taskbar [ ].
4. For profile name, enter Charles and click OK.
5. In the Inbox, you should see an email with the subject line Welcome to Cisco Webex Services Trial as well as one that has the
subject of Your password was updated. If you don’t see one or both of these emails, you might need to set Charles’ password
manually.
6. Open Chrome web browser to https://admin.webex.com and enter cholland@cbXXX.dc-YY.com. Click Sign In.
7. Enter dCloud123! as the password and click Sign In again.
8. Click Accept when prompted.
9. Verify the users have synchronized. Either maximize the Chrome web browser that you have open on Workstation 1, or log into
the browser on your machine, which is connected to the Webex Control Hub (https://admin.webex.com). In the portal, click the
Users tab [ ]. (Refresh the page if you are already there.)
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6. You should see a list of eight users along with their avatars and user information, such as email address and name.
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Webex Calling – Calling Features
Value Proposition: Imagine being able to leverage enterprise-grade cloud calling, mobility, and PBX features, along with Cisco
Webex Teams for messaging, Webex Meetings for meetings, and calling from a soft client or Cisco phone. That's exactly what Cisco
Webex Calling has to offer, along with a smooth transition to the cloud for customers with 100 or more users and existing on-
premises PBX infrastructures.
Cisco Webex Calling notes:
• Calling subscriptions for telephony users and common areas
• Secure and reliable cloud services delivered by trusted regional service providers
• The Cisco Webex Calling App for desktop and mobile devices, providing comprehensive calling capabilities for mobile
workers
• Webex Teams access for every user, adding rich unified communications and team collaboration services
• Cisco Webex Meetings as an optional, integrated add-on to provide the premium meetings experiences that enterprise
users expect
• Local Gateway for PSTN
• Sold through the Value Added Reseller (VAR) channel
• Fully Webex branded
• Enterprise must procure PSTN and minutes separately
• Cisco MPP endpoint support
• Cisco room endpoint support
• Global Discovery Service
• Dial plan enhancements
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Steps
Configuring Taylor, Rebekah, and Kellie with Webex Calling
1. In the Webex Control Hub (https://admin.webex.com), select Users from the left-hand menu. (If not logged in already, use
cholland@cbXXX.dc-YY.com and password dCloud123!)
2. Click Rebekah from the user list and then click Edit.
3. Check the Webex Calling box under Calling on this page and click Next.
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4. Leave Phone Number as None. Enter 6022 for the Extension. Click Finish.
NOTE: In the lab, you will be using the extension only for the users and features. The Phone Number option is to assign a real PSTN
number to the user phones. There are no real PSTN services setup for the lab.
5. Edit Taylor and Kellie’s user accounts and add the Webex Calling service. Set the Extension as 6021 for Taylor and 6050 for
Kellie.
6. (Optional but highly recommended) For testing purposes later, configure one more user (Stefan Mauk) with Webex Calling. Use
the Extension 6023. This will be the user that will be initiating the calls into the system to test calling features.
Configuring a Multiplatform Phone (MPP) for Taylor using Global Discovery Service (GDS)
This feature uses a 16-digit activation code that is generated by Cisco Global Discovery Service (GDS) and the Webex Calling Platform
to onboard and provision the device with the associated Webex Platform. When the code is provided to a user/administrator, the
code is entered into the MPP Phone. The MPP phone communicates with GDS and redirects the phone to the hosting platform. The
MPP phone communicates with the Webex Platform and uses the activation code to authenticate with the platform. If
authentication is successful, the MPP MAC address is stored onto the platform and the phone is provided with the provisioning
server location. The phone reboots and downloads the user configuration from Device Management.
This feature is supported on the following Cisco MPP devices:
• 6821, 6841, 6851
• 7811, 7821, 7832, 7841, 7861
• 8811, 8832, 8841, 8845, 8851, 8861, 8865
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If you have a physical MPP phone, use the following instructions to register it along with other phones you would like to use. You can
skip to the next section if you just want to use the Webex Calling apps.
1. On the Control Hub, click Devices.
2. Click Add Device and Next.
3. Choose Existing User and click Next.
4. Search for Taylor Bard, select him from the list, and click Next.
5. Select the device type from the drop-down menu.
6. Keep By Activation Code selected and click Next.
NOTE: Make sure your devices have the latest version of firmware. You can obtain the latest version from upgrade.cisco.com. Also,
if your phone has not been factory reset, do that now. If the phone has already been factory reset and powered on, reboot the
phone to have it attempt the registering process again.
7. Once the phone boots after a factory reset, enter the activation code onto the phone and press the Continue button.
8. After a few minutes, the phone should reboot and register. It may take a few minutes before the device is listed on the devices
page in Control Hub after registering.
NOTE: At this time in Control Hub, the MPP phone device status will always read Status Unavailable. Phone status is coming soon
but hasn’t been released at the time of writing this guide.
Configuring a Multiplatform Phone (MPP) using Bulk Activation Code for Kellie Melby
This feature delivers the ability for a Partner or Enterprise Admin to add devices for users and places using Activation Codes.
Benefits include:
• Large enterprise user additions reduced from hours to minutes
• Leverages GDS activation code functionality
1. On the Control Hub, click Devices.
2. Click Add Device.
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3. In the Multiple Devices section, click Import/Upload CSV file.
The Admin can select to export a user attributes CSV file or a CSV template file. For this lab, you will be using a template file.
NOTE: More information can be found at the following link:
https://help.webex.com/en-us/n9r1aac/Configure-and-Manage-Webex-Calling-Devices#id_118912
7. Click download CSV template to download the file on your personal machine or Workstation 1.
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8. We will enable Kellie Melby. The downloaded template will have sample data , it needs to be deleted and then enter the info
for Kellie. Save the CSV file on the desktop.
NOTE: Each session has a unique domain. Replace the XXX and YY from your session domain name. Remember not to enter the
Directory Number in the CSV file as its already defined in the User section. Leaving the MAC Address column blank, activation code is
generated and must be entered on device itself.
9. We will upload the file that was created for Kellie. Either drag and drop your CSV file or click browse to select the file. Make
sure Provide a link is selected. Click Submit.
10. Once done, you can download the CSV file that contains the code generated by clicking the Download activation codes CSV link.
11. After entering the code on the MPP phone, the device will boot and register as Kellie Melby.
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Configuring Webex Calling for Room System

This section will enable Cloud Calling capabilities for Webex Teams shared room systems to make/receive multimedia calls to/from
the PSTN. We will preserve the ability to make or receive calls to other Teams users and other third-party SIP systems using a public
URI. There are a couple points to remember:
• A Webex Calling Room System is a shared device (meaning, it has its own identity/number)
• Calls can be made from a paired Teams soft client (mobile/desktop); calls assume the identity of the Room System, not the
user that is paired
If you have a Cisco video endpoint, use the following instruction to register it along with other devices you would like to use. If you
do not have a physical device, one of the Webex Calling apps will suffice for this lab. You can skip to the next section if you will be
using Webex Calling apps.
1. In the Control Hub, click Places.
2. Click New Place.
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3. For What will this place be called ?, input “Anita_Room_Device”. Click Next.
4. Choose Other Cisco Webex device. Click Next.
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5. Select Cisco Webex Calling and click Next.
6. Assign 6017 as Anita’s Extension and click Next.
7. On the next screen you will receive a single use Activation code. Enter the code on your endpoint and it will register. Your
endpoint might upgrade after registering.
8. Once registered, you can see the endpoint register under places.
9. You can test by dialing the four-digit extension for Taylor, Rebekah, or Kellie.
Configuring the Webex Calling Applications
For your other users, use the Webex Calling applications to make and receive calls. (If you do not have a physical phone for Taylor,
use one of the Webex Calling apps for him as well.) In the future in the Webex Teams app, you will be able to make cloud calls. At
the time of writing this guide, that is not yet available. A user can download the desktop client from their own user portal, as shown
below.
1. From your computer, in another web browser or incognito/private window, navigate to https://settings.webex.com.
NOTE: To prevent sign-in issues with using multiple users on the same computer, it is recommended to use a different browser. You
can use another window within the same browser but used in incognito/private mode.
2. Login using your user’s Webex Teams username and password. For example, Rebekah’s login info would be:
- Username: rbarretta@cbXXX.dc-YY.com
- Password: dCloud123!
3. After logging in, click Webex Calling at the top. This will launch you into the Cisco Webex Calling user portal.
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4. Click MyApps.
5. Click Download for the Desktop Software. After it downloads, install the software onto the computer.
6. Open the Webex Calling application, Agree to the terms, then choose North America when prompted for the Region. Click
Continue.
NOTE: All the trials in the lab are configured in the North America region.
7. After the client refreshes, click Sign In.
8. Enter rbarretta@cbXXX.dc-YY.com and click Next.
9. Enter dCloud123! as the password and click Sign In.
10. There is also a mobile client in the Google Play and Apple App stores. Search for Webex Calling and download to your mobile
device. Follow the prompts to sign in to the app with your other users. It’s very important to tap Allow/OK on all prompts after
signing in or some features will not be available.
Configuring Dial Plan Enhancements
This feature provides the ability for an Admin in Control Hub to configure a more structured dial plan with the addition of a
configurable outbound dial digit, a configurable location code that could include the option for a leading prefix digit on a per-
location basis, and a fixed extension length.
The new dial plan options are configured in Control Hub at the customer level and at the location level.
Customer-level Settings
1. Click Services in the left panel.
2. On the Call card, click the Service Settings link. Any of these fields can be changed by clicking Edit. In this lab, we will be leaving
everything to its default value.
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Location-level Settings
3. Click Services in the left panel.
4. On the Call card, click the Locations link.
5. Click the Location defined.
6. Click External Dialing.
7. Click Edit and then Continue.
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8. Specify the number the user must dial before placing an external call. In our lab, we will use 9 for outside line. Once the number
is defined, click Save.
9. We will test this section later in this lab, after we configure the local gateway.
Configuring Calling Features
Now that you have the clients installed and logged in, you will go through and configure the calling features. You will first configure
everything and then perform testing at the end.
Adding New Calling Features
You will create the call features. It does take time for these features to be provisioned into the system. Due to the provisioning time,
you will add all the features first and then go back and configure them.
1. In the Control Hub, navigate to Services and click the Features link that is located on the Call card.
You will notice there is already an Auto Attendant configured. This was created and assigned during the startup of your session. You
will be editing this feature later.
2. Click New Feature. On the pop-up screen, click Hunt Group.
3. Use the drop-down menu and choose dCloud from the list.
4. Click Save, Done, and then Done again.
You will assign an extension for the Hunt Group a little later.
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On the last screen that appeared, there is a status bar for the feature to be provisioned, which is why you will go ahead and create
the rest of the features now. At any time, you can click Orders at the top right of the page to view the status of all the
features/orders you are creating.
5. Click New Feature and then click Paging Group.
6. For location, choose dCloud.
8. Click New Feature and then click Call Queue.
9. For location, choose dCloud.
Configuring Calling Features
While you are waiting on all the features to provision, you can start configuring some features. The first one is the Hunt Group.
Configuring Hunt Group
You will configure the Hunt Group you added earlier. You may want to set up Hunt Groups in the following scenarios:
• A Sales team that wants sequential routing, such as an incoming call rings one phone, but if there's no answer, the call goes
to the next agent in the list.
• A Support team that wants phones to ring all at once so that the first available agent can take the call.
1. If the Hunt Group is not yet showing on the call features page, refresh the page.
2. On the Hunt Group card, click the ellipses [ ] and then Edit.
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You’ll notice it will open a new tab to the Webex Calling portal. This will be the case for each feature you created. From here on
instead of going to Control Hub and clicking Edit, you will configure everything within the Webex Calling portal now.
3. Enter a relevant name for the Hunt Group, such as Hunt Group.
4. For the Phone Number, choose (empty) from the drop-down list.
5. Enter 6030 for the Extension.
6. Enter a name for Caller ID.
7. At the bottom right of the page, click Save.
8. After the page refreshes, click Call Routing from the menu on the left. (Click Save if prompted.)
9. Select the radio button for One at a time. By default, Top Down is selected. This option will always start in order of the
assignment set in the Phones section. The call will ring the users one at a time, always starting with the same user and ringing in
the same order. The user configured at the top of the Assigned list will ring first, followed by the users listed beneath, in
sequential order. To view more information on the types of call routing and what each does, click here. This page also gives an
overview of hunt groups and explains in more detail the configuration including configuration not covered in the lab.
10. Check the box for Advance after a set number of rings and configure for 3 rings.
12. After the page refreshes, click Phones from the menu on the left.
13. Click within the Find and Assign search and choose Taylor Bard and Rebekah Barretta.
15. After the page refreshes, click the X [ ] at the top right of the page to close the Hunt Group configuration.
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Call Queues
Call queues temporarily hold calls in the cloud when all users (agents) assigned to receive calls from the queue are unavailable.
Queued calls are routed to an available agent when he/she is no longer on an active call. Each call queue is assigned a Lead Number,
which is a telephone number outside callers can dial to reach users assigned to the call queue. Call queues are also assigned an
internal extension, which can be dialed internally to reach users assigned to the call queue.
In the lab, you will configure a basic call queue. To view more information on call queues as well as more advanced configurations
click here.
1. From within the Webex Calling portal, click Advanced Services.
2. In the Call Routing list, click Call Queues.
3. Click Assign for the one call queue you configured earlier in Control Hub.
4. Enter a relevant name for the call queue, such as Call Queue.
5. For the Phone Number, choose (empty) from the drop-down list.
6. Enter 6031 for the Extension.
7. Enter a name for Caller ID.
9. After the page refreshes, click Queue Settings from the menu on the left.
10. Check the box for Play ringing tone to callers when their call is sent to an available agent.
11. Click Advanced Settings.
12. Check the box next to Welcome message is mandatory.
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13. Toggle on Estimated Wait Message for Queued Calls.
14. Select the radio button for Announce Queue Position.
15. Toggle on Hold Music.
16. Check the box for Mark calls as overflow after queue wait time and enter 80 into the box.
18. After the page refreshes, click Call Routing from the menu on the left.
19. Under Call Routing, choose One at a time.
Now, you need to assign licenses and agents to add to the queue.
21. After the page refreshes, click Agents from the menu on the left.
22. Click within the Find and Assign search and choose Taylor Bard and Rebekah Barretta.
24. After the page refreshes, click the [ ] at the top right of the page to close the call queue configuration.
Configuring Auto Attendant
Auto Attendants ensure calls are answered and that callers' needs are met. You can add Greetings, set up Menus, and route calls to
an answering service, a Hunt Group, a voicemail box, or a real person. You can create a 24-hour schedule or provide different
options when your business is open or closed. You can even route calls based on caller ID attributes to create VIP lists or handle calls
from certain area codes differently.
As mentioned earlier, an auto attendant was already created for you at the beginning of the lab. Adding a new auto attendant would
be like any of the features you configured earlier.
2. In the Call Routing list, click Auto Attendants.
3. You will be taken to the Auto Attendant configuration page. A number and schedule has been created for you. For testing
purposes, you will later edit the assigned Time Schedule to ensure calls go through no matter when you are completing the lab.
4. Click Actions and then choose Edit Service.
5. For Extension enter 6020.
7. After the page refreshes, click Schedule from the menu on the left.
8. On the next screen, click Edit Time Schedule.
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9. Under Schedule Events, each day is represented by an event name. It starts with 1 being Sunday, 2 being Monday, and so on.
Click Edit for the event that corresponds to the day you are taking the lab. For example, if you are taking the lab on Thursday,
click Edit for Event 5.
10. To verify you chose the correct event, click Recurrence. You should see the day chosen under Recur On.
11. Click Event.
12. Change the Name to read the day (Sunday, Monday, Tuesday, etc.) to make it easier to determine later.
13. For lab purposes, check the box next to All Day Event. This will ensure no matter what time it is, the calls will go through when
you test later.
14. Click Save.
Feel free to edit the other events, especially if you plan on using the lab on other days.
15. After editing the days of the week, click Save.
16. At the bottom right of the Schedule, page click Save.
17. After the page refreshes, click Menu from the menu on the left.
18. Use the drop-down menus for each number, configure the following parameters from the table below for Business Hours.
Auto Attendant Business Hours Configuration
Setting Configuration
0 Not Used
1 Dial By Extension
2 Dial By Name
3 Transfer To Extension Without Prompt
Enter 6021 for the Number
Description: Dial Taylor
4 (need to click Transfer To Extension Without Prompt
Show More to Enter 6030 for the Number
view) Description: Hunt Group
5 Transfer To Extension Without Prompt
Enter 6031 for the Number
Description: Call Queue
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You will be testing those configured numbers later but feel free to configure other numbers if you would like for testing. To view
more information about configuring an Auto Attendant, including a description on what each option does and how to configure a
custom greeting, click here.
20. After the page refreshes, click the [ ] at the top right of the page to close the auto attendant configuration.
Next, you will configure a voice portal password so you can record a prompt for the auto attendant.
21. Click Advanced Services and then the Site Package Settings tab.
22. In the list, click Voice Portals.
23. If prompted in the pop-up menu, choose dCloud from the Location drop-down menu and click Select.
24. For the number listed, click Actions > Edit Service.
25. For extension, enter 6032.
26. Enter 1357 as the Voice Portal Passcode and Confirm New Passcode.
27. Click Save.
28. Close the Voice Portal configuration page by clicking the X.
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Now you can dial in to record your voice prompts for the auto attendant.
29. From your phone, dial the extension number (6032) for the Voice Portal.
30. Press the star/asterisk button [*] when prompted. Enter the mailbox ID, enter the extension for the Voice Portal (6032), and
press the pound button [#].
31. Enter the 1357 passcode and then press [#].
32. Press 1 to change the Auto Attendant greeting.
33. Press 1 to change the Business Hours greeting.
34. Press 1 to change the Greeting.
35. Record the new greeting to greet callers to your company and list the options configured above. When finished recording, press
[#].
36. After recording, press 2 to listen to the greeting you recorded or 1 to re-record the greeting.
37. Hang up when you are finished creating the greeting for the auto attendant.
Configure Call Pickup
Group Call Pickup enables a user to answer any ringing line within their pickup group. A pickup group is a group administrator-
defined set of users within a Site, to which the Call Pickup feature applies. The Group Call Pickup feature requires Call Pickup groups
to be added, modified, and removed as well as assigning specific users to that pickup group.
Before the Call Pickup service can be used, the following conditions must be met:
• A user can only be assigned to one Call Pickup group.
• A Call Pickup group may only have assigned users from the same site.
• A site may have multiple Call Pickup groups.
• Call Pickup group names must be unique.
• Assigning a user to a Call Pickup group removes the Barge-in restriction, if enabled for that user.
2. Under Call Routing, click Call Pickup.
3. Click Add.
4. Enter Call-Pickup as the Group Name and then click Save.
5. When the page refreshes, in the Available section, check the boxes for Taylor and Rebekah and click the right arrow [ ] to
move them to Assigned.
6. Click Save.
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Configuring Call Park
The Call Park feature allows a defined group user to park a call against other available members of a Call Park group, which may be
picked up by other members of the group at their phone.
What to know before getting started:
• A user can only be assigned to one Call Park group.
• A Call Park group may only have users from same site.
• A site may have multiple Call Park groups.
• Call Park group names must be unique.
2. Under Call Routing, click Call Park.
3. Click Add.
4. Enter Call-Park as the Group Name and then click Save.
5. After the page refreshes, in the Available section, check the boxes for Taylor and Rebekah and click the right arrow [ ] to
move them to Assigned.
6. Click Save.
Note there is a Global Settings option at the top of the page, which you can configure settings for all configured Call Park groups.
Configuring Group Paging
The Group Paging service allows a user to set up a one-way call to a group of up to 75 target users by dialing a number or extension.
The Group Paging service makes a simultaneous call to all the assigned targets and announces to the originator that the system is
ready for paging. After speaking, the originator ends the page by hanging up the call.
1. From within the Webex Calling portal click Advanced Services.
2. Under Productivity Services, click Group Paging.
3. Click Edit next to the one paging group you configured earlier in the Control Hub.
4. Give the paging group a Name, such as PagingGrp.
5. For Extension, enter 6033.
6. To the right of the page, under More Options, click Paging Targets.
7. In the Available section, check Taylor and Rebekah and click right arrow [ ] to move them to Assigned.
9. On the main paging group page, at the bottom right, click Save.
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Testing Calling Features
Now that the calling features have been configured, it’s time to test them. You should have Taylor configured for a phone or logged
him into a Webex Calling client. Rebekah should be logged into a Webex Calling client as well. As mentioned earlier, since Taylor and
Rebekah have been configured for all the calling features, it would be best if you have another user (Stefan Mauk – Extension 6023)
configured and logged into a Webex Calling client to initiate all the calls. If that’s not possible, you can call in with either Taylor or
Rebekah. All tests below will assume a third user calling into the system and Taylor and Rebekah answering the calls.
Testing Auto Attendant, Hunt Group, and Call Queue
Earlier, when you configured the Auto Attendant you recorded a greeting and configured the options. As a reminder, the options
were configured with the following parameters from the table below.
Auto Attendant Business Hours Configuration
Setting Configuration
0 Not Used
1 Dial By Extension
2 Dial By Name
3 Dial Taylor directly
4 Transfer to Hunt Group which Taylor and Rebekah are called
5 Transfer to Call Queue which Taylor and Rebekah are called
You have configured Taylor and Rebekah for the Hunt Group and Call Queue so a third user calling in can test routing to both users.
We have configured Stefan Mauk with extension 6023 to test these features. You can use either the calling application signed in as
Stefan Mauk or the Webex Calling endpoint for Stefan Mauk.
1. Dial the extension of the Auto Attendant (6020) with your third user.
2. Test option 1 and enter an extension of either Taylor or Rebekah. The call should forward to the extension.
3. Test option 2 and using the keypad, enter the first or last name of Taylor or Rebekah. The call should forward after the first
three characters are entered. Answer the call. If you decline the call, it will be sent to voicemail.
4. Test option 3. Taylor’s phone should ring.
5. Test option 4. Taylor or Rebekah’s phone/app should ring (depending on who was first in the list), decline the call and observe
the call fall over to the other user.
6. Test option 5. Observe the following points:
• You should hear a voice prompt telling you to wait for next available agent. It will also notify you of your queue position.
• Taylor or Rebekah’s phone/client will ring.
• If you decline the incoming call, it will fall over to the other agent.
• If you let the call ring, it will continue to prompt the user based on the timer you sent.
• Eventually, after the time you set, it will fall over to the other user to answer and eventually register a busy signal if no one
answers.
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Feel free to test any other options you configured for the Auto Attendant.
Testing Call Pickup
You will test Call Pickup. Ideally you need three users to fully test this feature. You have configured Taylor and Rebekah in the Pickup
Group. In this case, you need a third user to dial one of them and the other user, not called, to initiate the pickup.
1. Initiate a call to Rebekah from your third user the Webex Calling application. Do not answer the call.
2. On Taylor’s phone/app, dial *98 to pick up the call.
3. When you are finished, end the call.
Testing Call Park
You will test Call Park. Ideally you need three users to fully test this feature. You have configured Taylor and Rebekah in the Call Park
group. In this case, you need a third user to dial one of them, park the call, and pick up the call with the other user or the same user.
1. Initiate a call to Taylor or Rebekah from your third user the Webex Calling application.
2. Answer the call on the phone or client. You can park the call as follows:
• Answer the call on the phone. Press the Transfer softkey or physical transfer button. Enter *68 and then when you hear the
long beep, enter the 5-digit extension of the user you just called or another user and then the # key. The call will be parked
at that extension and you will hear hold music on the dial-in user’s client.
• Answer the call on the phone. Press the Park softkey. Press # key and when you hear the long beep, press the # key again to
park the call on the user’s extension.
• Answer on the client. Click the Park button [ ], click the green button that lists user extensions. You could alternatively
enter another extension in the Use another extension box and click Park.
3. To retrieve the call from the phone or client dial *88. When prompted enter the 5-digit extension of where you parked the call
followed by the # key.
4. End the call when you are finished.
Testing Paging Group
You will test the Paging Group feature, possibly. This feature may or may not work in your environment. You will need to satisfy the
following requirements:
• Paging requires multicast routing using the IP address 239.192.16.240. Make sure that IP address is free for multicast
routing only.
• Paging Groups only work with the Cisco IP Phone 7800 or 8800 Series.
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• You need at least two users in the group that are assigned to one of the phone types listed in the previous bullet. At least
one of the phones need to be registered. In the lab, you assigned Taylor and Rebekah to the Paging Group so each of these
users need to have a 7800 or 8800 phone registered to them.
• If one of the requirements above are not met, the Paging Group will not work and you will receive a busy tone or a message
stating the number is not available.
1. Initiate a call from Taylor’s MPP phone to the Paging Group extension (6033).
2. You will hear Paging come through the speaker on all the other phones in the group. After a few seconds of hearing paging on
the other phones, you will hear Paging system ready on Taylor’s phone. You then can speak on Taylor’s phone and hear your
voice come through on the other phones.
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Deploying Local Gateway for PSTN Calling
Value Proposition: With this feature, you are able to Bring Your Own PSTN (BYOPSTN) and provide PSTN connectivity to your cloud
registered phones.
Up until now, you have been testing all internal calls with no PSTN access. In this section ,you will explore the local gateway option
for Webex Calling. There are a few options when configuring a local gateway for your site. In this lab, you will configure a local
gateway for a single customer site to route calls from/to Webex Calling endpoints and the PSTN SIP Provider located at
198.18.133.3.
Local Gateway Deployment without an On-prem IP PBX
The figure above displays a Webex Calling deployment without any existing IP PBX and is applicable to a single or a multi-site
deployment. For all calls that do not match the customer’s Webex destinations, Webex sends those calls to the Local Gateway
assigned to the site for processing. Local Gateway routes all calls coming from Webex to the PSTN and vice versa. The PSTN gateway
may be a dedicated platform or co-resident with the Local Gateway (focus of this Scenario). The dedicated PSTN gateway variant of
this deployment is a preferred option for Webex Calling deployments.
In this lab, we will deploy a co-resident PSTN and Local Gateway as shown in the figure on the next page. The Local Gateway may be
IP based, connecting to an ITSP using a SIP trunk (focus of this Scenario), or TDM based using an ISDN or analog circuit.
Local Gateway Hardware and Software Requirements
• ISR 4321, 4331, 4351, 4431, 4451, CSR 1000v (vCUBE) – (IOS XE 16.9(3) and 16.11.1 or later)
• IOS-XE 16.10.x is not supported as Local Gateway for any platform
• ISR 1100 (IOS-XE 16.12 – Subject to change)
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Reference Prepare Your Environment for Webex Calling
NOTE: The Local Gateway configuration provided in this lab is for CSR1000v running IOS-XE16.9.4.
IOS-XE 16.11.1 or later has slightly different requirements that will be added to a future release of this lab guide.
The call flow into the dCloud session works as follows:
a. Call comes to a dCloud PSTN DID number from the SIP PSTN Provider.
b. The dCloud gateway translates that number into a four-digit extension and sends it to the Local Gateway.
o These numbers can be found in the Phone Numbers sections in your dCloud session’s details page. They can
also be found on the desktop of Workstation 1 in a text file named DN_to_DID.txt.
c. Local Gateway then routes these calls to Webex, which in turn rings the Webex Calling endpoints at a customer site.
o Four extensions used in the lab are: 6020, 6021, 6022, 6023
Local Gateway Onboarding Process
Enabling Local Gateway for Webex Calling requires:
• Partner to define Local GW for a customer site within Control Hub, generating connection parameters
• Local GW to register over SIP TLS using connection parameters from Control Hub above
• Call Routing on LGW based on PSTN connectivity requirements
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Creating Local Gateway
The first task is to create the local gateway in the Control Hub. You will need the information provided after creation to build your
local gateway.
1. Open the Chrome web browser to https://admin.webex.com and enter cholland@cbXXX.dc-YY.com. Click Sign In.
2. Enter dCloud123! as the password and click Sign In again.
3. In Control Hub, navigate to Services and click the Locations link on the Call card.
Each location can be assigned a local gateway. In the lab, you have one location that you will be configuring a local gateway for. The
gateway is hosted in your dCloud session.
4. Select the dCloud location.
5. On the fly-out window, click Local Gateway.
6. Click Edit > Continue > Manage.
7. Using the drop-down menu, choose Create New Local Gateway.
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8. Enter your own First Name (e.g. Hussain) as the name for the gateway and then click the green checkmark [ ]. (Ensure you
enter a name before clicking the checkmark.)
9. After a few moments, the information will be displayed on the page. You will need this information to configure the local
gateway. Write down the info now. You will need the Registrar Domain, Trunk Group OTG/DTG, Line/Port, and Outbound
Proxy Address.
10. Once you gathered the information, scroll to the bottom of the page and click the link that reads Retrieve Username and Reset
Password. If your password starts with a “?” or “!”, regenerate again as these characters may apply as commands on the
router. (For instance, a question mark is the command for Help.)
You will receive a pop-up message that says your password will be reset. This is okay since this is the first time you are retrieving the
username and password. However, it is very important to understand that anytime you click this link, it will reset the password. If
you already have a local gateway configured with a previous password, then this reset will break the local gateways connection to
Webex. So you will want to keep this information in a safe place and not lose it.
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11. Click Yes on the pop-up message.
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12. Capture the Username and Password. You will also use this during the local gateway configuration.
Sample Local Gateway Username and Password
13. Once you are finished capturing the username and password, click Done.
14. After gathering all the information, click Back.
15. Using the Local Gateway drop-down menu, choose the local gateway name.
16. Check the two boxes that appear and then click Save.
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Local Gateway Configuration
Now you will configure your local gateway with the information you received in the previous section.
NOTE: All the commands can also be found on the desktop of Workstation 1 named LGW_Config.txt.
1. Connect to Workstation 1 and open PuTTY using the icon on the desktop [ ]. (If you used VPN to connect to the dCloud
session, you can also use your local SSH client.)
2. Double click the Local Gateway saved session to SSH to the local gateway at 198.18.133.226.
3. Login with admin / dCloud123!
4. Using the configuration below, create a dummy PKI Trustpoint and call it dummyTp.
5. Assign the trustpoint as the default signaling trustpoint under sip-ua. The cn-san-validate server is needed to ensure LGW
establishes the connection only if the outbound proxy configured on the tenant 200 (described later) matches with CN-SAN list
received from the server. The crypto trustpoint is needed for TLS to work even though a local client certificate (i.e. mTLS) is not
required for the connection to be set up.
6. Finally disable TLS v1.0 and v1.1 by enabling v1.2 exclusivity, as shown in the table below.
Local Gateway Trustpoint Configuration
Configuration
configure terminal
crypto pki trustpoint dummyTp
revocation-check crl
exit
sip-ua
crypto signaling default trustpoint dummyTp cn-san-validate server
transport tcp tls v1.2
end
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Local Gateway Trustpoint Configuration
7. The default trustpool bundle does not include the DigiCert Root CA certificate needed for validating the server-side certificate
during TLS connection establishment to Webex. The trustpool bundle must be updated by downloading the latest Cisco Trusted
Core Root Bundle from http://www.cisco.com/security/pki, as shown below.
Local Gateway Certificate Configuration and Verification
Configuration
! Check if the DigiCert Root CA certificate exists
show crypto pki trustpool | include DigiCert
! – If not, update as shown below
configure terminal
crypto pki trustpool import clean url http://www.cisco.com/security/pki/trs/ios_core.p7b
end
! Verify
show crypto pki trustpool | include DigiCert
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Local Gateway Certificate Configuration and Verification
8. Enter the following commands to turn on the Local Gateway/CUBE application on the platform.
Configuration
configure terminal
voice service voip
ip address trusted list
ipv4 85.119.56.128 255.255.255.192
ipv4 85.119.57.128 255.255.255.192
ipv4 185.115.196.0 255.255.255.128
ipv4 185.115.197.0 255.255.255.128
ipv4 199.59.64.0 255.255.255.128
ipv4 199.59.65.0 255.255.255.128
ipv4 199.59.66.0 255.255.255.128
ipv4 199.59.67.0 255.255.255.128
ipv4 199.59.70.0 255.255.255.128
ipv4 199.59.71.0 255.255.255.128
exit
allow-connections sip to sip
media statistics
media bulk-stats
no supplementary-service sip refer
no supplementary-service sip handle-replaces
fax protocol pass-through g711ulaw
stun
stun flowdata agent-id 1 boot-count 4
stun flowdata shared-secret 0 Password123$
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sip
g729 annexb-all
early-offer forced
end
Explanation of commands above:
ip address trusted list – Toll Fraud Prevention
This is to explicitly enable the source IP addresses of entities from which Local Gateway expects legitimate VoIP calls, for
example, Webex peers, Unified CM nodes, IP PSTN. By default, LGW blocks all incoming VoIP call setups from IP addresses not in
its trusted list. IP Addresses from dial-peers with session target ip or Server Group are trusted by default and need not be
populated here.
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IP addresses in this list need to match the IP subnets from the Port Reference section of the Configuration Guide for Cisco
Webex Calling Customers document
(https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/broadcloud/webexcalling/customers/cisco-
webex-calling-configuration-guide/cisco-webex-calling-configuration-guide_chapter_01101.html). According to the regional
Webex Calling data center, the customer is connected to (North America, EMEA, Australia, Beta). The configuration on the
previous page includes all the existing Webex data centers as of the writing of this document and as listed in tables 1 to 4 of the
configuration guide referenced above.
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media statistics
Enables media monitoring on the LGW.
media bulk-stats
Enables the control plane to poll the data plane for bulk call statistics.
Allows this platform to bridge two VoIP SIP call legs. It is disabled by default.
no supplementary-service sip refer and no supplementary-service sip handle-replaces
Disables REFER and replaces the Dialog ID in the Replaces header with the peer dialog ID.
Enables audio codec for fax transport.
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stun
Enables STUN globally. When a call is forwarded back to a Webex user (such as when both the called and calling parties are
Webex subscribers and have the media anchored at the Webex SBC), the media cannot flow to the local gateway as the pin hole
is not opened.
The STUN bindings feature on the local gateway allows locally generated STUN requests to be sent over the negotiated media
path. The shared secret is arbitrary as STUN is only used to open the pinhole in the firewall and allow media latching to take
place in the Webex Access SBC.
STUN password is a pre-requisite for LGW/CUBE to send STUN message out. IOS/IOS-XE-based firewalls can be configured to
check for this password and open pin-holes dynamically (i.e. without explicit in-out rules). But for the LGW deployment case, the
firewall is statically configured to open pin-holes in the outbound direction based on Webex SBC subnets, so the firewall should
just treat this as any inbound UDP packet, which will trigger the pin-hole opening without explicitly looking at the packet
contents.
sip
g729 annexb-all
Allows all variants of G729.
sip
early-offer forced
This command forces the LGW/CUBE to send the SDP information in the initial INVITE message itself instead of waiting to send
the information till it gets an acknowledgement from the neighboring peer.
Configuring SIP Profile
In this section, we will configure only the SIP profile 200 as shown in the figure below, using only the trunk group OTG/DTG
parameter for rule 20.
1. Configure the following SIP profile required to convert SIPS URIs back to SIP as Webex does not support SIPS URI in the
request/response messages (but needs them for SRV query, for example, _sips._tcp.<outbound-proxy>).
rule 20 modifies the From header to include the Trunk Group OTG/DTG parameter from Control Hub to uniquely identify an LGW
site within an enterprise. In the example below, hussain3350_lgu is used. Make sure you replace the example with your respective
Trunk Group OTG/DTG information.
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NOTE: Do not use the example hussain3350_lgu shown for reference in the table below. Use your respective Trunk Group
OTG/DTG information.
Sample Configuration Mapping
Local Gateway SIP Profile Configuration
Configuration
configure terminal
voice class sip-profiles 200
rule 9 request ANY sip-header SIP-Req-URI modify "sips:(.*)" "sip:\1"
rule 10 request ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 11 request ANY sip-header From modify "<sips:" "<sip:\1"
rule 12 request ANY sip-header Contact modify "<sips:(.*)>" "<sip:\1;transport=tls>"
rule 13 response ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 14 response ANY sip-header From modify "<sips:(.*)" "<sip:\1"
rule 15 response ANY sip-header Contact modify "<sips:(.*)" "<sip:\1"
rule 20 request ANY sip-header From modify ">" ";otg=hussain3350_lgu>"
rule 30 request ANY sip-header P-Asserted-Identity modify "sips:(.*)" "sip:\1"
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Local Gateway SIP Profile Configuration
2. Configure Codec Profile, STUN definition, and SRTP Crypto suite as shown and explained in the table below.
Configuration
voice class codec 99
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
exit
voice class srtp-crypto 200
crypto 1 AES_CM_128_HMAC_SHA1_80
exit
voice class stun-usage 200
stun usage firewall-traversal flowdata
exit
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Local Gateway Codec Profile, STUN definition, and SRTP Crypto Suite Configuration
Explanation of commands above:

Allows both g729 and g711(mu and a-law) codecs for sessions. Will be applied to all the dial-peers.
Specifies SHA1_80 as the only SRTP cipher-suite that will be offered by LGW/CUBE in the SDP in offer and answer. Webex only
supports SHA1_80. This command will be applied to voice class tenant 200 (discussed later) facing Webex.
Defines STUN usage. Will be applied to all Webex facing (2XX tag) dial-peers to avoid no-way audio when a Unified CM Phone
forwards the call to another Webex phone.
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3. Configure voice class tenant 200 as shown below. However, be sure to use the parameters obtained from the Control Hub as
shown in the mapping below and not as displayed under the voice class tenant 200 in this document.
Sample Configuration Mapping
NOTE: Do not use these examples as they are only shown here for reference that is used in the table below. Use the configuration
gathered earlier in Control Hub when adding the Local Gateway.
• Registrar Domain: 40462196.cisco-bcld.com
• Trunk Group OTG/DTG: hussain3350_lgu
• Line/Port: Hussain8789_LGU@40462196.cisco-bcld.com
• Outbound Proxy Address: la01.sipconnect-us10.cisco-bcld.com
• Username: Hussain3350_LGU
• Password: bjljJ2VQji
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Local Gateway Tenant Configuration
Configuration
voice class tenant 200
registrar dns:40462196.cisco-bcld.com scheme sips expires 240 refresh-ratio 50 tcp tls
credentials number Hussain8789_LGU username Hussain3350_LGU password bjljJ2VQji realm BroadWorks
authentication username Hussain3350_LGU password bjljJ2VQji realm BroadWorks
authentication username Hussain3350_LGU password bjljJ2VQji realm 40462196.cisco-bcld.com
no remote-party-id
sip-server dns:40462196.cisco-bcld.com
connection-reuse
srtp-crypto 200
session transport tcp tls
url sips
error-passthru
asserted-id pai
bind control source-interface GigabitEthernet1
bind media source-interface GigabitEthernet1
no pass-thru content custom-sdp
sip-profiles 200
outbound-proxy dns:la01.sipconnect-us10.cisco-bcld.com
privacy-policy passthru
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Local Gateway Tenant Configuration
Explanation of Commands above:
This CUBE multi-tenant feature enables specific global configurations for multiple tenants on SIP trunks that allow differentiated
services for tenants.
Registrar server for the Local Gateway with the registration set to refresh every two minutes (50% of 240 seconds).
Credentials for Trunk Registration challenge.
Authentication challenge for calls.
no remote-party-id
Disable SIP Remote-Party-ID (RPID) header as Webex supports PAI, which is enabled using CLI asserted-id pai (see below).
Webex servers.
connection-reuse
To use the same persistent connection for registration and call processing.
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srtp-crypto 200
Specifying SHA1_80 defined in voice class srtp-crypto 200.
Setting transport to TLS.
url sips
SRV query has to be SIPS as supported by the access SBC; all other messages will be changed to SIP by sip-profile 200.
error-passthru
SIP error response pass-thru functionality.
asserted-id pai
Turn on PAI processing in LGW/CUBE.
Signaling source interface facing Webex.
Media source interface facing Webex.
no pass-thru content custom-sdp”
Default command under tenant.
sip-profiles 200
To change SIPS to SIP and modify Line/Port for INVITE and REGISTER messages as defined in: voice class sip-profiles 200.
Webex Access SBC.
Transparently pass across privacy header values from incoming to the outgoing leg.
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4. Next, configure the following voice class tenants.
Local Gateway Voice Class Tenant Configuration
Configuration
! Voice class tenant 100 will be applied on all OUTBOUND dial-peers facing the IP PSTN
session transport udp
url sip
error-passthru
! Voice class tenant 300 will be applied on all INBOUND dial-peers from the IP PSTN
Local Gateway Voice Class Tenant Configuration
5. Configure the following voice class URIs for URI-based dialing.
Local Gateway Voice Class Configuration
Configuration
! - Defines ITSP’s host IP address
voice class uri 100 sip

host ipv4:198.18.133.3
! - Defines pattern to uniquely identify a Local gateway site within an Enterprise

pattern dtg=hussain3350.lgu
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6. Configure voice class uri 200 as shown above. However, use the parameters obtained from the Control Hub as shown in the
mapping below and not as displayed under the voice class uri 200 in this document.
The pattern defined within voice class uri 200 is configured to match the unique Trunk Group OTG/DTG parameter obtained from
the Control Hub earlier and also defined in rule 20 of the voice class sip-profiles 200.
NOTE: Local Gateway today doesn’t support the underscore [ _ ] in the match pattern. As a workaround, we used the dot wildcard
[ . ] (match any) to match the underscore [ _ ] within hussain3350_lgu (example displayed in this document).
Local Gateway Voice Class Configuration
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7. Configure the following outbound dial-peers shown in the table below.
Local Gateway Outbound Dial-Peer Configuration
Configuration
! - Outbound dial-peer towards IP PSTN
dial-peer voice 101 voip

description Outgoing dial-peer to IP PSTN
destination-pattern BAD.BAD
session protocol sipv2
session target ipv4:198.18.133.3
voice-class codec 99
voice-class sip tenant 100
dtmf-relay rtp-nte
no vad
! - Outbound dial-peer towards Webex

description Outgoing dial-peer to Webex
session target sip-server
voice-class stun-usage 200
no voice-class sip localhost
dtmf-relay rtp-nte
srtp
no vad
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Local Gateway Outbound Dial-Peer Configuration
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8. Configure the following dial-peer groups (DPG).
Local Gateway Dial-Peer Group Configuration
Configuration
! - Defines dial-peer group 100. Outbound dial-peer 101 is the target for any incoming dial-peer invoking dial-peer group 100. We will apply DPG 100 to
! - incoming dial-peer 200 defined later for Webex --> LGW --> PSTN path
voice class dpg 100

description Incoming BCLD(DP200) to IP PSTN(DP101)
dial-peer 101 preference 1
! - Define dial-peer group 200. Outbound dial-peer 201 is the target for any incoming dial-peer invoking dial-peer group 200. We will apply DPG 200 to
!- incoming dial-peer 100 defined later for the IP PSTN --> LGW --> Webex path.
voice class dpg 200

description Incoming IP PSTN(DP100) to BCLD(DP201)
Local Gateway Dial-Peer Group Configuration
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9. Configure the following inbound dial-peers.
Local Gateway Inbound Dial-Peer Configuration
Configuration
! - Inbound dial-peer for incoming IP PSTN call legs

description Incoming dial-peer from IP PSTN
destination dpg 200
incoming uri via 100
dtmf-relay rtp-nte
no vad
! - Inbound dial-peer for incoming Webex call legs, with the call destined for IP PSTN

description Incoming dial-peer from Webex
destination dpg 100
incoming uri request 200
dtmf-relay rtp-nte
srtp
no vad
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Local Gateway Inbound Dial-Peer Configuration
That completes the local gateway configuration. Now you will save.
10. Type end at the command prompt.
11. To save, type copy run start and press Enter/Return twice.
NOTE: To view all the local gateway configuration in one place, go to Appendix B.
At this stage, the Local Gateway should be registered to Webex. Registration from the Local Gateway itself can be verified as shown
below.
Before testing the configuration, verify the Local Gateway also shows online in Control Hub.
12. Go back to the Control Hub and login if needed (cholland@cbXXX.dc-YY.com / dCloud123!).
13. Click the dCloud location.
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14. On the flyout window under PSTN Connection, you should see your first name for Local Gateway. Click Local Gateway.
15. Click Edit > Continue.
16. Select your first name from the drop-down menu and click Manage.
17. The Status should read Online. After verifying the status, click the X to close the window.
Online and Registered Local Gateway
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Local Gateway Testing
Now that you have a fully configured and registered the Local Gateway, you can test calls to/from PSTN to your Webex Calling
endpoints.
Call from PSTN to MPP/Webex Calling App
Now you can test calls from/to the PSTN. The DID numbers to dial from the PSTN are located in your dCloud session details. They are
also located on the desktop of Workstation 1 in a text file named DN_to_DID.txt
1. With the DID number, dial one of your users or the Auto Attendant using a real cell or desk phone.
2. Answer the call on your Webex Calling device/app.
3. The call flow in dCloud is as follows:
a. Incoming DID comes into dCloud from IP PSTN.
b. Platform gateways translate that DID into a four-digit extension (6XXX or 7XXX).
c. Call is routed through the Local Gateway into Webex and to the extension of the user.
Call from MPP/Webex Calling App to PSTN
In dCloud national, calls are allowed to the datacenter region your session is located in. US West/East datacenters are dc-05 or dc-
01. EMEAR datacenter is dc-03. APJ datacenter is dc-02.
In the US datacenters, you should be able to call any national number. Remember, since the location is built for the United States,
Webex will add a +1 to any dialed number, if needed. So a 10-digit number can be called or a 1 + 10 digit number.
Dialing from an EMEAR or APJ session will be just a little different than US. Every session, no matter the datacenter, the Webex
Calling location is built for United States. Because of this, in the lab, dialing a national number for the UK (EMEAR) or Singapore (APJ)
requires you to dial a 00 + Country Code and then the number (10-digit for UK and 8-digit for Singapore). For the EMEAR datacenter,
the country code is 44. For the APJ datacenter, the country code is 65.
When you dial 00 and the number, Webex will route the number as is and not add +1. Then the call flow for EMEAR and APJ sessions
in Unified CM is as follows:
• Call comes into Local Gateway, sent to dCloud Gateway, which delivers it to the IP PSTN.
• The dCloud GW will strip the 00 off the number and add a plus (+).
• The +E.164 number is now routed by the IP PSTN and rings the PSTN number.
1. With the dialing explanation above, dial a PSTN number.
2. Answer the call.
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NOTE: You’ll notice the incoming caller ID number will begin with a 919, 408, 44, or 64 number (depending on the session’s
datacenter). This is expected due to the way the dCloud PSTN is set up. The caller ID will always show as the DID assigned to the
session that maps to the 7800 DN.
3. End the call.
4. You can test the external calls. Dial 9 to get an outside line and then the numbering pattern as defined above.
Local Gateway Debugging
To troubleshoot call setup failures on the Local Gateway, proceed as shown below.
Local Gateway Debugging
Configuration
! - Collecting debugs in Local Gateway’s logging buffer
-----------------------------------------------------------------
LocalGateway#conf t
LocalGateway(config)#no logging console
LocalGateway(config)#no logging monitor
LocalGateway(config)#service timestamps debug datetime msec
LocalGateway(config)#logging buffered 9999999 debugging
LocalGateway(config)#service sequence-numbers
LocalGateway(config)#no logging rate-limit
LocalGateway(config)#exit
! - The following debugs will assist in troubleshooting call failures for this lab:
“debug ccsip message”

“debug voip ccapi inout”
You can view the debugs by issuing the “show log” command.
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Webex Teams and Webex Calling
A Webex Teams user can utilize the Webex Calling desktop client to dial telephone numbers. If the Webex Calling service has been
added to the trail, the Webex Calling app should be used for calling. This setting can be configured for the entire organization. If you
want a different calling behavior for individual users, you can set this on the user themselves.
1. Go to the Control Hub and navigate to the Settings page.
2. Scroll down to Calling Behavior.
3. You will notice the Calling in Webex Teams radio button selected by default. As mentioned, you can also set calling behavior on
each user individually but you will change the setting organization-wide here.
4. Select the radio button for Webex Calling app and wait for the success notification on the bottom right of the page.
5. Navigate to the User’s page.
6. Select Rebekah from the list. On the fly-out window, scroll down to the Settings section.
7. Click Calling Behavior.
8. You will see the default calling behavior is to use the Organization Setting: Webex Calling App for dialing telephone numbers.
Since this is the behavior you need for this test, select this radio button and click Save.
9. On your desktop with the Webex Calling App, log into Webex Teams with Rebekah (rbarretta@cbXXX.dc-YY.com /
dCloud123!).
10. Make a call to a telephone number.
This time you will notice the Webex Calling client will now open and make the call. In the future, the call will not cross launch into
the Webex Calling App; instead, it will call directly from the Webex Teams App.
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Appendix A. Cisco Webex Password Reset Procedures

In some cases, it may be necessary to reset the password of a Cisco Webex user account. The following are the steps to complete a
password reset for Charles Holland.
1. Navigate to the Webex Teams web client (https://admin.webex.com) and enter the email cholland@cbXXX.dc-YY.com
2. Click Next.
3. On the next page, verify it looks similar to the image below with exception to the XXX and YY parts of the domain. If it does look
similar, continue to the next step.
Correct Sign-in Screen
4. If your screen looks similar to the sign-in the image above, click the Forgot password? link at the bottom.
5. An email was sent to Charles with reset instructions. Using remote desktop, connect to Workstation 1 (198.18.1.36) and log in
as dcloud\cholland with password C1sco12345. Open Outlook if not open already. (You can also use OWA to access every users’
email account by going to https://mail1.dcloud.cisco.com/owa and signing in with dcloud\{username} and password
C1sco12345
6. You will see an email in the inbox with the subject Password Reset. Within that email, click Reset password.
7. A browser window will open. Enter dCloud123! in the New Password and Confirm new password boxes.
8. Click Save & Sign In.
9. Close out the browser and continue with the lab.
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Appendix B. Full Local Gateway Configuration

The following is the full configuration that was completed in the lab of the Local Gateway for reference. (You cannot modify this in
the current lab.) Remember, the items in red are places that you need your own lab’s configuration. The figure below was used to
create the config.
Sample Local Gateway Config
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Sample Local Gateway Username and Password
crypto pki trustpoint dummyTp
revocation-check crl
voice service voip
ip address trusted list
ipv4 85.119.56.128 255.255.255.192
ipv4 85.119.57.128 255.255.255.192
ipv4 185.115.196.0 255.255.255.128
ipv4 185.115.197.0 255.255.255.128
ipv4 199.59.64.0 255.255.255.128
ipv4 199.59.65.0 255.255.255.128
ipv4 199.59.66.0 255.255.255.128
ipv4 199.59.67.0 255.255.255.128
ipv4 199.59.70.0 255.255.255.128
ipv4 199.59.71.0 255.255.255.128
exit
media statistics
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media bulk-stats
no supplementary-service sip refer
no supplementary-service sip handle-replaces
stun
! - secret 0 due to mustard changes
sip
g729 annexb-all
early-offer forced
end
host ipv4:198.18.133.3
pattern dtg=hussain3350.lgu
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
stun usage firewall-traversal flowdata
voice class sip-profiles 200
rule 9 request ANY sip-header SIP-Req-URI modify "sips:(.*)" "sip:\1"
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rule 10 request ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 11 request ANY sip-header From modify "<sips:" "<sip:\1"
rule 12 request ANY sip-header Contact modify "<sips:(.*)>" "<sip:\1;transport=tls>"
rule 13 response ANY sip-header To modify "<sips:(.*)" "<sip:\1"
rule 14 response ANY sip-header From modify "<sips:(.*)" "<sip:\1"
rule 15 response ANY sip-header Contact modify "<sips:(.*)" "<sip:\1"
rule 20 request ANY sip-header From modify ">" ";otg=hussain3350_lgu>"
rule 30 request ANY sip-header P-Asserted-Identity modify "sips:(.*)" "sip:\1"
voice class dpg 100
description Incoming BCLD(DP200) to IP PSTN(DP101)
voice class dpg 200
description Incoming IP PSTN(DP100) to BCLD(DP201)
Hussain8789_LGU username Hussain3350_LGU password bjljJ2VQji
no remote-party-id
connection-reuse
srtp-crypto 200
url sips
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error-passthru
asserted-id pai
sip-profiles 200
session transport udp
url sip
error-passthru
crypto 1 AES_CM_128_HMAC_SHA1_80
description Outgoing dial-peer to IP PSTN
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session target ipv4:198.18.133.3
dtmf-relay rtp-nte
no vad
description Outgoing dial-peer to Webex
session target sip-server
no voice-class sip localhost
dtmf-relay rtp-nte
srtp
no vad
description Incoming dial-peer from IP PSTN
destination dpg 200
incoming uri via 100
dtmf-relay rtp-nte
no vad
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description Incoming dial-peer from Webex
destination dpg 100
incoming uri request 200
dtmf-relay rtp-nte
srtp
no vad
sip-ua
transport tcp tls v1.2
crypto signaling default trustpoint dummyTp cn-san-validate server
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What’s Next?
For more product and sales information, please reference the SalesConnect Hubs for Cisco Webex Calling and Cisco Collaboration
Flex Plan.

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Lab Guide

Deploy Cisco Webex Calling v2

Last Updated: 14-JANUARY-2020

About This Lab

About This Lab

About This Solution

Initial Cisco Webex Control Hub Organization Verification

Webex Calling – Calling Features

Deploying Local Gateway for PSTN Calling

Appendix A. Cisco Webex Password Reset Procedures

Appendix B. Full Local Gateway Configuration

There are no customizable options associated with this lab.

About This Solution

Cisco Webex Calling Overview

Name Description Host Name (FQDN) IP Address Username Password

PSTN Services SIP Service Provider N/A 198.18.133.3 N/A N/A

Local Gateway CSR1000V N/A 198.18.133.226 admin dCloud123!

AD1 Active Directory, DNS, AD FS ad1.dcloud.cisco.com 198.18.133.1 administrator C1sco12345

Exchange Microsoft Exchange 2016 mail1.dcloud.cisco.com 198.18.133.2 administrator C1sco12345

Workstation 1 Windows 10 wkst1.dcloud.cisco.com 198.18.1.36 cholland C1sco12345

Workstation 2 Windows 10 wkst2.dcloud.cisco.com 198.18.1.37 aperez C1sco12345

Workstation 3 Windows 10 wskt3.dcloud.cisco.com 198.18.1.38 kmelby C1sco12345

Anita Perez aperez C1sco12345 Cisco Jabber®/Teams 6017 Hybrid

Kellie Melby kmelby C1sco12345 Cisco Jabber/Teams 6050 Hybrid

PREPARATION IS KEY TO A SUCCESSFUL PRESENTATION

NOTE: It may take up to 45 minutes for your session to become active.

Session Details Tab Example

Initial Cisco Webex Control Hub Organization Verification

3. Open Microsoft Outlook using the icon in the taskbar [ ].

4. For profile name, enter Charles and click OK.

7. Enter dCloud123! as the password and click Sign In again.

8. Click Accept when prompted.

Users tab [ ]. (Refresh the page if you are already there.)

Webex Calling – Calling Features

Cisco Webex Calling notes:

• Calling subscriptions for telephony users and common areas

• Local Gateway for PSTN

• Sold through the Value Added Reseller (VAR) channel

• Fully Webex branded

• Enterprise must procure PSTN and minutes separately

• Cisco MPP endpoint support

• Cisco room endpoint support

• Global Discovery Service

• Dial plan enhancements

Configuring Taylor, Rebekah, and Kellie with Webex Calling

This feature is supported on the following Cisco MPP devices:

• 6821, 6841, 6851

• 7811, 7821, 7832, 7841, 7861

• 8811, 8832, 8841, 8845, 8851, 8861, 8865

1. On the Control Hub, click Devices.

2. Click Add Device and Next.

3. Choose Existing User and click Next.

5. Select the device type from the drop-down menu.

6. Keep By Activation Code selected and click Next.

• Large enterprise user additions reduced from hours to minutes

• Leverages GDS activation code functionality

1. On the Control Hub, click Devices.

2. Click Add Device.

3. In the Multiple Devices section, click Import/Upload CSV file.

NOTE: More information can be found at the following link:

Configuring Webex Calling for Room System

1. In the Control Hub, click Places.

2. Click New Place.