Webex Calling Configuration Workflow
Webex Calling Configuration Workflow

Feb 19, 2022
Overview

Introducing Webex Calling

Imagine being able to leverage enterprise-grade cloud calling, mobility, and PBX features, along with Webex App for messaging and meetings and calling from a Webex Calling soft client or Cisco device. That's exactly what Webex Calling has to offer you.

Webex Calling provides the following benefits:

  • Calling subscriptions for telephony users and common areas

  • Webex App access for every user

  • Public Switch Telephony Network (PSTN) access to let your users dial numbers outside the organization. The service is provided through an existing enterprise infrastructure (local gateway without on-premises IP PBX or with existing Unified CM call environment)

Webex Calling supports the following features. For more information, see the Configure Webex Calling Features chapter.

Table 1. Admin Configurable Features

Feature

Description

Auto Attendant

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.

Call Queue

You can set up a call queue so that when incoming calls can't be answered, callers are provided with an automated answer, comfort messages, and music on hold until someone can answer their call.

Call Pickup

You can enhance teamwork and collaboration by creating a call pickup group so users can answer each others calls. When you add users to a call pickup group and a group member is away or busy, another member can answer their calls.

Call Park

You can turn on call park so that users can put a call on hold and pick it up from another phone.

Hunt Group

You may want to set up hunt groups in the following scenarios:

  • A Sales team that wants sequential routing. 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.

Paging Group

You can create a paging group so that users can send an audio message to a person, a department, or a team. When someone sends a message to a paging group, the message plays on all devices in the group.

Receptionist Client

Help support the needs of your front-office personnel by providing them with a full set of call control options, large-scale line monitoring, call queuing, multiple directory options and views, Outlook integration, and more.

Users can configure the following features in https://settings.webex.com, which cross-launches into the Calling User Portal.

Table 2. User Configurable Features

Feature

Description

Anonymous Call Rejection

Users can reject incoming calls with blocked caller ID's.

Business Continuity

If users' phones are not connected to the network for any reason (such as power outage, network issues, and so on), users can forward incoming calls to a specific phone number.

Call Forwarding

Users can forward incoming calls to another phone.

Call Forwarding Selective

Users can forward calls at specific times from specific callers. This setting will take precedence over Call Forwarding.

Call Notify

Users can send themselves an email when they receive a call according to predefined criteria such as phone number or date and time.

Call Waiting

Users can allow answering of additional incoming calls.

Do Not Disturb

Users can temporarily let all calls to go directly to voicemail.

Office Anywhere

Users can use their selected phones ("Locations") as an extension of their business phone number and dial plan.

Priority Alert

Users can ring their phones with a distinctive ring when predefined criteria are met, such as phone number or date and time.

Remote Office

Users can make calls from a remote phone and have it appear from their business line. In addition, any incoming calls to their business line will ring on this remote phone.

Selective Call Acceptance

Users can accept calls at specific times from specific callers.

Selective Call Rejection

Users can reject calls at specific times from specific callers.

Sequential Ring

Ring up to 5 devices one after another for incoming calls.

Simultaneous Ring

Ring users' and others (“call recipients“) numbers at the same time for incoming calls.

Provisioning Services, Devices, and Users in Control Hub, Cross-Launch to Detailed Configuration in Calling Admin Portal

Control Hub (https://admin.webex.com) is a management portal that integrates with Webex Calling to streamline your orders and configuration, and centralize your management of the bundled offer—Webex Calling, Webex App, and Meetings.

Control Hub is the central point for provisioning all services, devices, and users. You can do first time setup of your calling service, register MPP phones to the cloud (using MAC address), configure users by associating devices, adding numbers, services, calling features, and so on. Also, from Control Hub, you can cross-launch to the Calling Admin Portal.

User Experience

Users have access to the following interfaces:

Overview

Webex Calling can reduce operational costs and improve productivity by helping you migrate critical business communications to the cloud. When combined with other Webex apps and devices, it is the heart of a complete enterprise cloud calling and collaboration experience. Cisco supports on-premises, in the cloud, and mixed model deployments to keep our customers connected and productive from anywhere; even during disruptive market events.

Webex Calling now includes a dedicated cloud instance option based on the Cisco Unified Communications Manager architecture. Dedicated Instance is integrated with Webex Calling and takes advantage of Webex platform services, bringing cloud innovation and an enhanced experience to customers who need to support older Cisco endpoints, local survivability solutions, or existing integrations part of critical business workflows.

The Dedicated Instance add-on for Webex Calling includes:

  • Cisco Unified Communications Manager

  • Cisco Unified IM and Presence

  • Cisco Unified Unity Connection

  • Cisco Expressway

  • Cisco Emergency Responder (Americas region only)

Simple Migration Path

Dedicated Instance for Webex Calling provides a simplified cloud migration path from a legacy PBX as well as on-premises Unified Communications Manager systems.

Dedicated Instance alleviates the pain-points associated with enterprise calling migrations to the cloud:

No Disruptions – Dedicated Instance has the same features, functionality, user experience and integration options supported by Unified Communications Manager deployed on premises, including support for Jabber and Webex App. This creates a frictionless migration to the cloud with no end user or administrator training required for existing Unified Communications Manager customers. Dedicated Instance can be trunked to third party PBXs, allowing new Cisco customers a flexible migration schedule.

Customization – A dedicated private instance for every customer, allows for a highly customizable cloud deployment, which is a unique differentiator from other cloud calling offers in the market. Dedicated Instance’s open APIs enable deep third-party application integrations allowing customers to build a calling environment that supports unique business workflows.

Uncompromised Security – With Dedicated Instance customers have access to all the Unified Communications Manager security features for Endpoints and UC applications like encrypted media, secure SRST, secure OTT registration use MRA.

In addition, customers have access to important physical security features like Cisco Survivable Remote Site Telephony (SRST) for site connectivity in the event network links go down and Cisco Emergency Responder and Nomadic E911 to ensure employees can be located by emergency responders when in the office or in a hybrid mode of work. 

Extended ROI – Dedicated Instance supports the same voice and video endpoints as the associated UC Manager release, eliminating the requirement to refresh all customer endpoints when migrating to the cloud and extending the ROI of these assets.

Basic Inter-Op – Dedicated Instance is integrated with Webex Calling for call routing via the the Webex platform. Customers have the flexibility to distribute users across both Dedicated Instance and Webex Calling, and adjust over time as needed to address their cloud calling business requirements.

Important: Customers who split users across platforms will experience different features. The calling features are not harmonized between Dedicated Instance and Webex Calling. For example, Webex Calling users cannot be part of a hunt group on Dedicated Instance.

Solution Availability

The Dedicated Instance service is globally available and is orderable as an add-on for Webex Calling Flex Plan 3.0 through partners in specific countries. See the Global Availability Guide for more details.

Dedicated Instance supports the same level of localization as our on-premise Unified Communications Manager. It supports phone and gateway tones in 82 countries, a self-care portal in 50 languages, and clients in more than 30 languages.

Benefits

Dedicated Instance offers the most efficient migration path to the cloud for existing Unified Communications Manager customers, with the following key benefits:
  • Dedicated Calling application instance hosted and operated by Cisco in Webex Data Centers
  • Customizable Calling platform
  • Flexible, quickly scalable architecture
  • Familiar user experience, reducing the need for employee retraining
  • Unified client for calling, messaging, meetings and team collaboration that is usable across all device types
  • Compatibility with Cisco’s full portfolio of phones, gateways, and video devices
  • Integrates with Webex meetings, messaging, and calling as part of the Webex suite, enabling an amazing end to end customer experience.

For supported endpoints and devices, please click here.

Take a Tour of Control Hub

Control Hub is your single go-to, web-based interface for managing your organization, managing your users, assigning services, analyzing adoption trends and call quality, and more.

To get your organization up and running, we recommend that you invite a few users to join Webex App by entering their email addresses in the Control Hub. Encourage people to use the services you provide, including calling, and to give you feedback about their experience. When you're ready, you can always add more users.


We recommend that you use the latest desktop version of Google Chrome or Mozilla Firefox to access Control Hub. Browsers on mobile devices and other desktop browsers may produce unexpected results.

Use the information presented below as a high-level summary of what to expect when getting your organization set up with services. For more detailed information, see the individual chapters for step-by-step instructions.

Get Started

After your partner creates your account, you'll receive a welcome email. Click the Getting Started link in the email, using Chrome or Firefox to access Control Hub. The link automatically signs you in with your administrator email address. Next, you'll be prompted to create your administrator password.

First Time Wizard for Trials

If your partner has registered you for a trial, the setup wizard automatically starts after you sign in to Control Hub. The wizard walks you through the basic settings to get your organization up and running with Webex Calling, among other services. You can set up and review your Calling settings before finishing the wizard walkthrough.

Review Your Settings

When Control Hub loads, you can review your settings.

Add Users

Now that you have set up your services, you're ready to add people from your company directory. Go to Users and click Manage Users.

If you use Microsoft Active Directory, we recommend that you enable Directory Synchronization first and then decide how you want to add users. Click Next and follow the instructions to set up Cisco Directory Connector.

Set Up Single Sign On (SSO)

Webex App uses basic authentication. You can choose to set up SSO so that users authenticate with your Enterprise Identity Provider using their Enterprise credentials, rather than a separate password stored and managed in Webex.

Go to Settings, scroll to Authentication, click Modify, and then select Integrate a 3rd-party identity provider.

Assign Services to Users

You must assign services to the users that you've added so that people can start using Webex App.

Go to Users, click Manage Users, select Export and import users with a CSV file, and then click Export.

In the file you download, simply add True for the services that you want to assign to each of your users.

Import the completed file, click Add and remove services, and then click Submit. You're now ready to configure calling features, register devices that can be shared in a common place, and register and associate devices with users.

Empower Your Users

Now that you've added users and they've been assigned services, they can start using their supported Multiplatform Phones (MPPs) for Webex Calling and Webex App for messaging and meetings. Encourage them to use Cisco Webex Settings as a one-stop shop for the access.

Role of the Local Gateway

The local gateway is an enterprise or partner-managed edge device for Public Switch Telephony Network (PSTN) interworking and legacy public branch exchange (PBX) interworking (including Unified CM).

You can use Control Hub to assign a local gateway to a location, after which Control Hub provides parameters that you can configure on the CUBE. These steps register the local gateway with the cloud, and then PSTN service is provided through the gateway to Webex Calling users in a specific location.

To specify and order a Local Gateway, read the Local Gateway ordering guide.

Supported Local Gateway Deployments for Webex Calling

The following basic deployments are supported:

The local gateway can be deployed standalone or in deployments where integration into Cisco Unified Communications Manager is required.

Local Gateway Deployments Without On-Premises IP PBX

Standalone Local Gateway Deployments

This figure shows a Webex Calling deployment without any existing IP PBX and is applicable to a single location or a multi-location deployment.

For all calls that do not match your Webex Calling destinations, Webex Calling sends those calls to the local gateway that is assigned to the location for processing. The local gateway routes all calls that are coming from Webex Calling to the PSTN and in the other direction, PSTN to Webex Calling.

The PSTN gateway can be a dedicated platform or coresident with the local gateway. As in the following figure, we recommend the dedicated PSTN gateway variant of this deployment; it may be used if the existing PSTN gateway cannot be used as a Webex Calling local gateway.

Coresident Local Gateway Deployment

The local gateway can be IP based, connecting to an ITSP using a SIP trunk, or TDM based using an ISDN or analog circuit. The following figure shows a Webex Calling deployment where the local gateway is coresident with the PSTN GW/SBC.

Local Gateway Deployments With On-Premises Unified CM PBX

Integrations with Unified CM are required in the following cases:

  • Webex Calling-enabled locations are added to an existing Cisco UC deployment where Unified CM is deployed as the on-premises call control solution

  • Direct dialing between phones registered to Unified CM and phones in Webex Calling locations is required.

This figure shows a Webex Calling deployment where the customer has an existing Unified CM IP PBX.

Webex Calling sends calls that do not match the customer’s Webex Calling destinations to the local gateway. This includes PSTN numbers and Unified CM internal extensions, which Webex Calling cannot see. The local gateway routes all calls that are coming from Webex Calling to Unified CM and vice versa. Unified CM then routes incoming calls to local destinations or to the PSTN as per the existing dial plan. The Unified CM dial plan normalizes numbers as +E.164. The PSTN gateway may be a dedicated one or co-resident with the local gateway.

Dedicated PSTN Gateway

The dedicated PSTN gateway variant of this deployment as shown in this diagram is the recommended option and may be used if the existing PSTN gateway cannot be used as a Webex Calling local gateway.

Coresident PSTN Gateway

This figure shows a Webex Calling deployment with a Unified CM where the local gateway is coresident with the PSTN gateway/SBC.

Webex Calling routes all calls that do not match the customer’s Webex Calling destinations to the local gateway that is assigned to the location. This includes PSTN destinations and on-net calls towards Unified CM internal extensions. The local gateway routes all calls to Unified CM. Unified CM then routes calls to locally-registered phones or to the PSTN through the local gateway, which has PSTN/SBC functionality co-located.

Call Routing Considerations

Calls From Webex Calling to Unified CM

The Webex Calling routing logic works like this: if the number that is dialed on a Webex Calling endpoint cannot be routed to any other destination within the same customer in Webex Calling, then the call is sent to the local gateway for further processing. All off-net (outside of Webex Calling) calls are sent to the local gateway.

For a Webex Calling deployment without integration into an existing Unified CM, any off-net call is considered a PSTN call. When combined with Unified CM, an off-net call can still be an on-net call to any destination hosted on Unified CM or a real off-net call to a PSTN destination. The distinction between the latter two call types is determined by the Unified CM and depends on the enterprise dial plan that is provisioned on Unified CM.

The following figure shows a Webex Calling user dialing a national number in the US.

Unified CM now based on the configured dial plan routes the call to a locally registered endpoint on which the called destination is provisioned as directory number. For this the Unified CM dial plan needs to support routing of +E.164 numbers.

Calls From Unified CM to Webex Calling

To enable call routing from Unified CM to Webex Calling on Unified CM a set of routes need to be provisioned to define the set of +E.164 and enterprise numbering plan addresses in Webex Calling.

With these routes in place both the call scenarios shown in the following figure are possible.

If a caller in the PSTN calls a DID number that is assigned to a Webex Calling device, then the call is handed off to the enterprise through the enterprise’s PSTN gateway and then hits Unified CM. The called address of that call matches one of the Webex Calling routes that is provisioned in Unified CM and the call is sent to the local gateway. (The called address must be in +E.164 format when sent to the local gateway.) The Webex Calling routing logic then makes sure that the call is sent to the intended Webex Calling device, based on DID assignment.

Also, calls originating from Unified CM registered endpoints, targeted at destinations in Webex Calling, are subject to the dial plan that is provisioned on Unified CM. Typically, this dial plan allows the users to use common enterprise dialing habits to place calls. These habits do not necessarily only include +E.164 dialing. Any dialing habit other than +E.164 must be normalized to +E.164 before the calls is sent to the local gateway to allow for correct routing in Webex Calling.

Class of Service (CoS)

Implementing tight class of service restrictions is always recommend for various reasons including avoiding call loops and preventing toll fraud. In the context of integrating Webex Calling Local Gateway with Unified CM class of service we need to consider class of service for:

  • Devices registered with Unified CM

  • Calls coming into Unified CM from the PSTN

  • Calls coming into Unified CM from Webex Calling

Devices registered with Unified CM

Adding the Webex Calling destinations as a new class of destinations to an existing CoS setup is pretty straight forward: permission to call to Webex Calling destinations typically is equivalent to the permission to call on-premise (including inter-site) destinations.

If an enterprise dial-plan already implements an “(abbreviated) on-net inter-site” permission then there already is a partition provisioned on Unified CM which we can use and provision all the known on-net Webex Calling destinations in the same partition.

Otherwise, the concept of “(abbreviated) on-net inter-site” permission does not exist yet, then a new partition (for example “onNetRemote”) needs to be provisioned, the Webex Calling destinations are added to this partition, and finally this new partition needs to be added to the appropriate calling search spaces.

Calls coming into Unified CM from the PSTN

Adding the Webex Calling destinations as a new class of destinations to an existing CoS setup is pretty straight forward: permission to call to Webex Calling destinations typically is equivalent to the permission to call on-premise (including inter-site) destinations.

If an enterprise dial-plan already implements an “(abbreviated) on-net inter-site” permission then there already is a partition provisioned on Unified CM which we can use and provision all the known on-net Webex Calling destinations in the same partition.

Otherwise, the concept of “(abbreviated) on-net inter-site” permission does not exist yet, then a new partition (for example “onNetRemote”) needs to be provisioned, the Webex Calling destinations are added to this partition, and finally this new partition needs to be added to the appropriate calling search spaces.

Calls coming into Unified CM from Webex Calling

Calls coming in from the PSTN need access to all Webex Calling destinations. This requires adding the above partition holding all Webex Calling destinations to the calling search space used for incoming calls on the PSTN trunk. The access to Webex Calling destinations comes in addition to the already existing access.

While for calls from the PSTN access to Unified CM DIDs and Webex Calling DIDs is required calls originating in Webex Calling need access to Unified CM DIDs and PSTN destinations.

Figure 1. Differentiated CoS for calls from PSTN and Webex Calling

This figure compares these two different classes of service for calls from PSTN and Webex Calling. The figure also shows that if the PSTN gateway functionality is collocated with the Local Gateway, then two trunks are required from the combined PSTN GW and Local Gateway to Unified CM: one for calls originating in the PSTN and one for calls originating in Webex Calling. This is driven by the requirement to apply differentiated calling search spaces per traffic type. With two incoming trunks on Unified CM this can easily be achieved by configuring the required calling search space for incoming calls on each trunk.

Dial Plan Integration

This guide assumes an existing installation that is based on best current practices in the “Preferred Architecture for Cisco Collaboration On-Premises Deployments, CVD.” The latest version is available here.

The recommended dial plan design follows the design approach that is documented in the Dial Plan chapter of the latest version of the Cisco Collaboration System SRND available here.

Figure 2. Recommended Dial Plan

This figure shows an overview of the recommended dial plan design. Key characteristics of this dial plan design include:

  • All directory numbers that are configured on Unified CM are in +E.164 format.

  • All directory numbers reside the same partition (DN) and are marked urgent.

  • Core routing is based on +E.164.

  • All non-+E.164 dialing habits (for example, abbreviated intrasite dialing and PSTN dialing using common dialing habits) are normalized (globalized) to +E.164 using dialing normalization translation patterns.

  • Dialing normalization translation patterns use translation pattern calling search space inheritance; they have the “Use Originator's Calling Search Space” option set.

  • Class of service is implemented using site and class of service-specific calling search spaces.

  • PSTN access capabilities (for example access to international PSTN destinations) are implemented by adding partitions with the respective +E.164 route patterns to the calling search space defining class of service.

Reachability to Webex Calling

Figure 3. Adding Webex Calling destination to the dial plan

To add reachability for Webex Calling destinations to this dial plan, a partition representing all Webex Calling destinations must be created (“Webex Calling”) and a +E.164 route pattern for each DID range in Webex Calling is added to this partition. This route pattern references a route list with only one member: the route group with the SIP trunk to the Local Gateway for calls to Webex Calling. Because all dialed destinations are normalized to +E.164 either using dialing normalization translation patterns for calls originating from Unified CM registered endpoints or inbound called party transformations for calls originating from the PSTN this single set of +E.164 route patterns is enough to achieve reachability for destinations in Webex Calling independent of the dialing habit used.

If, for example, a user dials “914085550165”, then the dialing normalization translation pattern in partition “UStoE164” normalizes this dial string to “+14085550165” which then matches the route pattern for a Webex Calling destination in partition “Webex Calling.” The Unified CM ultimately sends the call to the local gateway.

Add Abbreviated Intersite Dialing

Figure 4. Adding Abbreviated Intersite Dialing

The recommended way to add abbreviated intersite dialing to the reference dial plan is to add dialing normalization translation patterns for all sites under the enterprise numbering plan to a dedicated partition (“ESN”, Enterprise Significant Numbers). These translation patterns intercept dial strings in the format of the enterprise numbering plan and normalize the dialed string to +E.164.

To add enterprise abbreviated dialing to Webex Calling destinations, you add the respective dialing normalization translation pattern for the Webex Calling location to the “Webex Calling” partition (for example “8101XX” in the diagram). After normalization, the call again is sent to Webex Calling after matching the route pattern in the “Webex Calling” partition.

We do not recommend adding the abbreviated dialing normalization translation pattern for Webex Calling calls to the “ESN” partition, because this configuration may create undesired call routing loops.

Difference between Webex Calling for Service Providers and Value Added Resellers

There are two separate calling offers that leverage the same Webex Calling platform. One offer is for service providers (SPs) and their customers while the other offer is for value added resellers (VARs) and their customers. For the most part, the offers are identical and as such, we refer to them generically as Webex Calling. However, there are a couple differences and where we need to call out those differences, we'll make sure you know whether they apply to SPs or VARs.

While both offers are administered in Control Hub with cross-launches into the Calling Admin Portal, here are some key differences.

SPs can brand their calling portals and apps and must bundle and provide their own PSTN services to their customers or leverage a local gateway deployment. SPs must also provide their own Tier 1 support.

VARs, on the other hand, use the branding provided by Cisco. VARs are not regulated service providers and cannot provide PSTN service. PSTN service must be leveraged through an enterprise local gateway deployment. VARs can also provide their own Tier 1 support or use Cisco's. Both calling offers provide service assurance through media quality metrics and can bundle Webex App and Meetings together with their calling applications.

Protocol Handlers for Calling

Webex Calling registers the following protocol handlers with the operating system to enable click-to-call functionality from web browsers or other application. The following protocols start an audio or video call in Webex App when it's the default calling application on Mac or Windows:

  • CLICKTOCALL: or CLICKTOCALL://

  • SIP: or SIP://

  • TEL: or TEL://

  • WEBEXTEL: or WEBEXTEL://

Protocol Handlers for Windows

Other apps can register for the protocol handlers before the Webex App. In Windows 10, the system window to ask users to select which app to use to launch the call. The user preference can be remembered if the user checks Always use this app.

If users need to reset the default calling app settings so that they can pick Webex App, you can instruct them to change the protocol associations for Webex App in Windows 10:

  1. Open the Default app settings system settings, click Set defaults by app,and then choose Webex App.

  2. For each protocol, choose Webex App.

Protocol handlers for macOS

On Mac OS, if other apps registered to the calling protocols before Webex App, users must configure their Webex App to be the default calling option.

In Webex App for Mac, users can confirm that Webex App is selected for the Start calls with setting under general preferences. They can also check Always connect to Microsoft Outlook if they want to make calls in Webex App when they click an Outlook contact's number.

Feb 19, 2022
Prepare Your Environment

Requirements for Calling

Licensing

Webex Calling is available through the Cisco Collaboration Flex Plan. You must purchase an Enterprise Agreement (EA) plan (for all users, including 50% Workspaces devices) or a Named User (NU) plan (some or all users).

Webex Calling provides three license types ("Station Types")

  • Professional—These licenses provide a full feature set for your entire organization. This offer includes unified communications (Webex Calling), mobility (desktop and mobile clients with support for multiple devices), team collaboration in Webex App, and the option to bundle meetings with up to 1000 participants per meeting.

  • Basic—Choose this option if your users need limited features without mobility or unified communications. They'll still get a full-featured voice offer but are limited to a single device per user.


    Basic licenses are only available if you have a Named User subscription. Basic licenses are not supported for Enterprise Agreement subscriptions.

  • Workspaces (also known as Common Area)—Choose this option if you're looking for basic dial-tone with a limited set of calling features appropriate for areas such as break rooms, lobbies, and conference rooms.

This documentation later shows you how to use Control Hub to manage these license distributions across locations in your organization.

Bandwidth Requirements

Each device in a video call requires up to 2 Mbps. Each device in an audio call requires 100 kbps. Phones at idle need minimal bandwidth.

Local Gateway for Premises-based PSTN

Both Value Added resellers (VARs) and Service Providers (SPs) can provide PSTN access to Webex Calling organizations. Local gateway is currently the only option to provide premises-based PSTN access. The local gateway can be deployed standalone or in deployments where integration into Cisco Unified Communications Manager is required. The local gateway requirements follow.

Supported Devices

Webex Calling supports Cisco Multiplatform (MPP) IP Phones. As an administrator, you can register the following phones to the cloud. See the following Help articles for more information:


For a complete list of supported devices for Webex Calling, see Supported Devices for Webex Calling.

Cisco Webex Room, Board, and Desk Devices are supported as devices in a Workspace that you create in Control Hub. See "Cisco Webex Room, Board, and Desk Devices" in Supported Devices for Webex Calling for more information. However, you can provide these devices with PSTN service by enabling Webex Calling for the Workspace.

Firewall

Meet the firewall requirements that are documented in Port Reference Information for Cisco Webex Calling.

Local Gateway Requirements for Webex Calling

General Prerequisites

Before you configure a local gateway for Webex Calling, ensure that you

    • Have a basic knowledge of VoIP principles

    • Have a basic working knowledge of Cisco IOS-XE and IOS-XE voice concepts

    • Have a basic understanding of Session Initiation Protocol (SIP)

    • Have a basic understanding of Cisco Unified Communications Manager (Unified CM) if your deployment model includes Unified CM

    More details can be found in the Cisco Unified Border Element (CUBE) Enterprise Configuration Guide at https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book.html

Hardware and Software Requirements for Local Gateway

Make sure your deployment has one or more of the local gateways (Cisco CUBE (for IP-based connectivity) or Cisco IOS Gateway (for TDM-based connectivity)) that are in Table 1 of the Local Gateway for Webex Calling Ordering Guide. Additionally, make sure the platform is running a supported IOS-XE release as per the Local Gateway Configuration Guide.

Certificate and Security Requirements for Local Gateway

Webex Calling requires secure signaling and media. The local gateway performs the encryption, and a TLS connection must be established outbound to the cloud with the following steps:

  • The LGW must be updated with the CA root bundle from Cisco PKI

  • A set of SIP digest credentials from Control Hub’s Trunk configuration page are used to configure the LGW (the steps are part of the configuration that follows)

  • CA root bundle validates presented certificate

  • Prompted for credentials (SIP digest provided)

  • The cloud identifies which local gateway is securely registered

Firewall, NAT Traversal, and Media Path Optimization Requirements for Local Gateway

In most cases, the local gateway and endpoints can reside in the internal customer network, using private IP addresses with NAT. The enterprise firewall must allow outbound traffic (SIP, RTP/UDP, HTTP) to specific IP addresses/ports, covered in Port Reference Information.

If you want to utilize Media Path Optimization with ICE, the local gateway’s Webex Calling facing interface must have a direct network path to and from the Webex Calling endpoints. If the endpoints are in a different location and there is no direct network path between the endpoints and the local gateway’s Webex Calling facing interface, then the local gateway must have a public IP address assigned to the interface facing Webex Calling for calls between the local gateway and the endpoints to utilize media path optimization. Additionally, it must be running IOS-XE version 16.12.5.

Feb 19, 2022
Configure Cisco Webex Calling for Your Organization

Customize your organization for Webex Calling in Control Hub. After activating your first location through the First Time Setup Wizard, you can set up and manage additional locations, trunk assignment and usage, dial plan options, users, devices, and features.

The first step to get your Webex Calling services up and running is to complete the First Time Setup Wizard (FTSW). Once the FTSW is completed for your first location, it doesn’t need to be completed for additional locations.

1

Click the Getting Started link in the Welcome email you receive.


 

Your administrator email address is automatically used to sign in to Control Hub, where you'll be prompted to create your administrator password. After you sign in, the setup wizard automatically starts.

2

Review and accept the terms of service.

3

Review your plan and then click Get Started.


 

Your account manager is responsible for activating the first steps for FTSW. Contact your account manager if you receive a “Cannot Setup Your Call” notice, when you select Get Started.

4

Select the country that your data center should map to, and enter the customer contact and customer address information.

5

Click Next: Default Location.

6

Choose from the following options:

  • Click Save and Close if you’re a partner administrator and you want the customer administrator to complete the provisioning of Webex Calling.
  • Fill out the necessary location information. After you create the location in the wizard, you can create more locations later.

 

After you complete the setup wizard make sure you add a main number to the location you create.

7

Make the following selections to apply to this location:

  • Announcement Language—For audio announcements and prompts for new users and features.
  • Email Language—For email communication for new users.
  • Country
  • Time Zone
8

Click Next.

9

Enter an available Cisco Webex SIP address and click Next and select Finish.

Before you begin

To create a new location, prepare the following information:

  • Location address

  • Desired phone numbers (optional)

1

From the customer view in https://admin.webex.com, go to Services > Calling > Locations, and then click Add Location.

Keep in mind that new locations will be hosted in the regional data center that corresponds to the country you selected using the First Time Setup Wizard.

2

Configure the settings of the location:

  • Location Name—Enter a unique name to identify the location.
  • Country/Region—Choose a country to tie the location to. For example, you can create one location (headquarters) in the United States and another (branch) in the United Kingdom. The country that you choose determines the address fields that follow. The ones documented here use the U.S. address convention as an example.
  • Location Address—Enter the location's main mailing address.
  • City/Town—Enter a city for this location.
  • State/Province/Region—From the drop-down, choose a state.
  • ZIP/Postal Code—Enter the ZIP or postal code.
  • Announcement Language—Choose the language for audio announcements and prompts for new users and features.
  • Email Language—Choose the language for the email communication with new users.
  • Time zone—Choose the time zone for the location.
3

Click Save and then choose whether you want to add numbers now or later.

4

If you clicked Add Now, choose one of the following options:

  • Cisco PSTN —Choose this option if you'd like a Cloud PSTN solution from Cisco. The Cisco Calling Plan is a full PSTN replacement solution that provides emergency calling, inbound and outbound domestic and international calling, and allows you to order new PSTN numbers or port existing numbers to Cisco.


     

    The Cisco PSTN option is only visible under the following conditions:

    • You have purchased at least one committed Cisco Calling Plan OCP (Outbound Calling Plan).

    • Your location is in a country where the Cisco Calling Plan is supported.

    • Your location is new. Pre-existing locations that have had other PSTN capabilities assigned aren't eligible for the Cisco Calling Plan at this time. Open a support case for guidance.

    • You are hosted in a Webex Calling Data Center in a region in which the Cisco Calling Plan is supported.

  • Cloud Connected PSTN—Choose this option if you’re looking for a cloud PSTN solution from one of the many Cisco CCP partners or if the Cisco Calling Plan isn't available in your location. CCP partners offer PSTN replacement solutions, extensive global coverage, and a broad and varied range of features, packaging, and pricing.

     

    CCP partners and the geographic coverage are listed here. Only partners that support your location’s country are displayed. Partners are listed either with a logo, or as a brief string of text followed by a region, in brackets (Example: (EU), (US) or (CA)). Partners listed with a logo always offer Regional Media for CCP. For partners displaying as a string, choose the region closest to the country of your location to ensure Regional Media for CCP.

    If you see the option to Order numbers now under a listed provider, we recommend that you choose that option so that you can benefit from integrated CCP. Integrated CCP enables procuring and provisioning of phone numbers in Control Hub on a single pane of glass. Non-integrated CCP requires you to procure your phone numbers from the CCP partner outside of Control Hub.

  • Premises-based PSTN (Local Gateway)—You can choose this option if you want to keep your current PSTN provider or you want to connect non-cloud sites with cloud sites.

The choice of PSTN option is at each location level (each location has only one PSTN option). You can mix and match as many options as you’d like for your deployment, but each location will have one option. Once you’ve selected and provisioned a PSTN option, you can change it by clicking Manage in the location PSTN properties. Some options, such as Cisco PSTN, however, may not be available after another option has been assigned. Open a support case for guidance.

5

Choose whether you want to activate the numbers now or later.

6

If you selected non-integrated CCP or Premises-based PSTN, enter Phone Numbers as comma-separated values, and then click Validate.

Numbers are added for the specific location. Valid entries move to the Validated Numbers field, and invalid entries remain in the Add Numbers field accompanied by an error message.

Depending on the location's country, the numbers are formatted according to local dialing requirements. For example, if a country code is required, you can enter numbers with or without the code and the code is prepended.

7

Click Save.

What to do next

After you create a location, you can enable emergency 911 services for that location. See RedSky Emergency 911 Service for Webex Calling for more information.

Before you begin


Get a list of the users and workspaces associated with a location: Go to Services > Calling > Numbers and from the drop-down menu, select the location to be deleted. You must delete those users and workspaces before you delete the location.

1

From the customer view in https://admin.webex.com, go to Services > Calling > Locations.

2

Click in the Actions column beside the location you'd like to delete.

3

Choose Delete Location, and confirm that you want to delete that location.

It typically takes a couple of minutes for the location to be permanently deleted, but it could take up to an hour. You can check the status by clicking beside the location name and selecting Deletion Status.

You can change your PSTN setup, the name, time zone, and language of a location after it's created. Keep in mind though that the new language only applies to new users and devices. Existing users and devices continue to use the old language.


For existing locations, you can enable emergency 911 services. See RedSky Emergency 911 Service for Webex Calling for more information.

1

From the customer view in https://admin.webex.com, go to Services > Calling > Locations, and then select the location you want to update.

If you see a Caution symbol next to a location, it means that you haven't configured a telephone number for that location yet. You can't make or receive any calls until you configure that number.

2

(Optional) Under PSTN Connection, select either Cloud Connected PSTN or Premises-based PSTN (local gateway), depending on which one you've already configured. Click Manage to change that configuration, and then acknowledge the associated risks by selecting Continue. Then, choose one of the following options and click Save:

  • Cisco PSTN —Choose this option if you'd like a Cloud PSTN solution from Cisco. The Cisco Calling Plan is a full PSTN replacement solution that provides emergency calling, inbound, and outbound domestic and international calling, and allows you to order new PSTN numbers or port existing numbers to Cisco.


     

    The Cisco PSTN option is only visible under the following conditions:

    • You have purchased at least one committed Cisco Calling Plan OCP (Outbound Calling Plan).

    • Your location is in a country where the Cisco Calling Plan is supported.

    • Your location is new. Currently, Pre-existing locations that have had other PSTN capabilities assigned aren't eligible for the Cisco Calling Plan. Open a support case for guidance.

    • You are hosted in a Webex Calling Data Center in a region in which the Cisco Calling Plan is supported.

  • Cloud Connected PSTN—Choose this option if you’re looking for a cloud PSTN solution from one of the many Cisco CCP partners or if the Cisco Calling Plan isn't available in your location. CCP partners offer PSTN replacement solutions, extensive global coverage, and a broad and varied range of features, packaging, and pricing.

     

    CCP partners and the geographic coverage are listed here. Only partners that support your location’s country are displayed. Partners are listed either with a logo, or as a brief string of text followed by a region, in brackets (Example: (EU), (US) or (CA)). Partners listed with a logo always offer Regional Media for CCP. For partners displaying as a string, choose the region closest to the country of your location to ensure Regional Media for CCP.

    If you see the option to Order numbers now under a listed provider, we recommend that you choose that option so that you can benefit from integrated CCP. Integrated CCP enables procuring and provisioning of phone numbers in Control Hub on a single pane of glass. Non-integrated CCP requires you to procure your phone numbers from the CCP partner outside of Control Hub.

  • Premises-based PSTN (Local Gateway)—You can choose this option if you want to keep your current PSTN provider or you want to connect noncloud sites with cloud sites.

     

    Webex Calling customers with locations that are previously configured with a Local Gateway will automatically be converted to premises-based PSTN with a corresponding trunk.

3

Select the Main Number at which the location's main contact can be reached.

4

(Optional) Under Emergency Calling, you can select Emergency Location Identifier to assign to this location.


 

This setting is optional and is only applicable for countries that require it.

In some countries (Example: France), regulatory requirements exist for cellular radio systems to establish the identity of the cell when you make an emergency call and is made available to the emergency authorities. Other countries like the U.S and Canada implements location determination using other methods. For more information, see Enhanced Emergency Calling.

Your emergency call provider may need information about the access network and is achieved by defining a new private SIP extension header, P-Access-Network-Info. The header carries information relating to the access network.

When you set the Emergency Location Identifier for a Location, the location value is sent to the provider as part of the SIP message. Contact your emergency call provider to see if you require this setting and use the value that is provided by your emergency call provider."

5

Select the Voicemail Number that users can call to check their voicemail for this location.

6

(Optional) Click the pencil icon at the top of the Location page to change the Location Name, Announcement Language, Email Language, Time Zone, or Address as needed, and then click Save.


 

Changing the Announcement Language takes effect immediately for any new users and features added to this location. If existing users and/or features should also have their announcement language changed, when prompted, select Change for existing users and workspaces or Change for existing features. Click Apply. You can view progress on the Tasks page. You can't make any more changes until this is complete.


 

Changing the Time Zone for a location doesn't update the time zones of the features associated with the location. To edit the time zones for features like auto attendant, hunt group, and call queue, go to the General Settings area of the specific feature you would like to update the time zone for and edit and save there.

These settings are for internal dialing and are also available in the first-time setup wizard. As you change your dial plan, the example numbers in Control Hub update to show these changes.


You can configure outgoing calling permissions for a location. See these steps to configure outgoing calling permissions.

1

From the customer view in https://admin.webex.com, go to Services > Calling > Service Settings, and then scroll to Internal Dialing.

2

Configure the following optional dialing preferences, as needed:

  • Location Routing Prefix Length—We recommend this setting if you have multiple locations. You can enter a length of 2-7 digits. If you have multiple locations with the same extension, users must dial a prefix when calling between locations. For example, if you have multiple stores, all with the extension 1000, you can configure a routing prefix for each store. If one store has a prefix of 888, you'd dial 8881000 to reach that store.
  • Steering Digit in Routing Prefix—You can set a value here regardless of whether you use location routing prefixes.
  • Internal Extension Length—You can enter 2-6 digits and the default is 2.

     

    After you increase your extension length, existing speed dials to internal extensions are not automatically updated.

3

Specify internal dialing for specific locations. Go to Services > Call > Locations, select a location, scroll to Dialing, and then change internal and external dialing as needed:

  • Internal Dialing—Specify the routing prefix that users at other locations need to dial in order to contact someone at this location. The routing prefix of each location must be unique. We recommend that the prefix length matches the length set at the organization level but it must be between 2-7 digits long.
  • External Dialing—Optionally, you can choose an outbound dial digit that users must dial to reach an outside line. The default is None and you can leave it if you don't require this dialing habit. If you do decide to use this feature, we recommend that you use a different number from your organization's steering digit.

     

    Users can include the outbound dial digit when making external calls to mimic how they dialed on legacy systems. To make external calls , you must include the outbound dial digit.

Impact to users:

  • Users must restart their phones in order for changes in dialing preferences to take effect.

  • User extensions should not start with the same number as the location's steering digit.

If you're a value added reseller, you can use these steps to start local gateway configuration in Control Hub. When this gateway is registered to the cloud, you can use it on one or more of your Webex Calling locations to provide routing toward an enterprise PSTN service provider.


A location that has a local gateway can't be deleted when the local gateway is being used for other locations.

Follow these steps to create a trunk in Control Hub.

Before you begin

  • Once a location is added, and before configuring premises-based PSTN for a location, you must create a trunk.

  • Create any locations and specific settings and numbers to each one. Locations must exist before you can add a premises-based PSTN.

  • Understand the Premises-based PSTN (local gateway) requirements for Webex Calling.

  • You can't choose more than one trunk for a location with premises-based PSTN, but you can choose the same trunk for multiple locations.

1

From the customer view in https://admin.webex.com, go to Services > Calling > Call Routing, and select Add Trunk.

2

Select a location.

3

Name the trunk and click Save.


 

The name can't be longer than 24 characters.

What to do next

You're presented with the relevant parameters that you'll need to configure on the trunk. You'll also generate a set of SIP digest credentials to secure the PSTN connection.

Trunk information appears on the screen Register Domain, Trunk Group OTG/DTG, Line/Port, and Outbound Proxy Address.

We recommend that you copy this information from Control Hub and paste it into a local text file or document so you can refer to it when you're ready to configure the premises-based PSTN.

If you lose the credentials, you must generate them from the trunk information screen in Control Hub. Click Retrieve Username and Reset Password to generate a new set of authentication credentials to use on the trunk.

1

From the customer view in https://admin.webex.com, go to Services > Calling > Locations.

2

Select a location to modify and click Manage.

3

Select Premises-based PSTN and click Next.

4

Choose a trunk from the drop-down menu.


 

Visit the trunk page to manage your trunk group choices.

5

Click the confirmation notice, then click Save.

What to do next

You must take the configuration information that Control Hub generated and map the parameters into the local gateway (for example, on a Cisco CUBE that sits on the premises). This article walks you through this process. As a reference, see the following diagram for an example of how the Control Hub configuration information (on the left) maps onto parameters in the CUBE (on the right):

After you successfully complete the configuration on the gateway itself, you can return to Services > Call > Locations in Control Hub and the gateway that you created will be listed in the location card that you assigned it to with a green dot to the left of the name. This status indicates that the gateway is securely registered to the calling cloud and is serving as the active PSTN gateway for the location.

1

From the customer view in https://admin.webex.com, select the building icon .

2

Select the Subscriptions tab, and then click Purchase Now.

An email is sent to your partner letting them know that you're interested in converting to a paid subscription.

1

From the customer view in https://admin.webex.com, go to Organization Settings > Services, scroll to Calling and then choose Client Settings.

2

Drag and drop calling options that you want users to see to the Available Call Options field, and then rearrange them in the priority order that you want for your users.

Other options that are hidden for users appear in the Hidden Call Options field, as shown in this example screenshot:

3

Toggle on Enable Single Click-to-Call if you want users to be able to make a call with the first call option that you configured in the previous step.


 

The changes may take up to 24 hours to appear in Webex App. You can advise your users to restart their apps to pick up on these changes more quickly.

You can control what calling application opens when users make PSTN calls. After you configure this setting at the organization level, you can override this setting for specific users.


Only choose the organization-wide option if you're ready to migrate your entire organization.

Before you begin

  • Your organization must have the correct subscriptions for the calling behavior you choose.

  • Users must have valid phone numbers. If the numbers are invalid, Webex App still sends the number to the calling app that you select, but the call from that app will fail.

From the customer view in https://admin.webex.com, go to Management > Organization Settings, and then scroll to Calling Behavior, and then choose one of the following: .

  • Calling in Webex—Select this option if you want users to make calls directly in Webex App using Webex Calling.
  • Webex Calling app—Select this option if your organization has a subscription to Cisco Webex Calling and you want to allow users to make PSTN calls using the Webex Calling app. When users make PSTN calls in Webex App, the Webex Calling app is used to make the call.

     

    The Webex Calling app is only available to select customers.

A message appears that indicates that the calling behavior is updated. Users are now able to make PSTN calls from Webex App or the Webex Calling app.

Users must have the corresponding application installed to make PSTN calls from Webex App. Make sure you let people know what choice you make and if another app is used to make PSTN calls.


 

You can change this setting at the user level if certain people need to use different calling behavior. Go to Users and under Settings, select Calling Behavior. You can make your choice and then click Save.

Feb 19, 2022
Implement CUBE High Availability as Local Gateway

Local Gateway (LGW) is the only option to provide premises-based PSTN access for Cisco Webex Calling customers. The objective of this document is to assist you in building a Local Gateway configuration using CUBE high availability, active/standby CUBEs for stateful failover of active calls.

Fundamentals

Prerequisites

Before you deploy CUBE HA as a local gateway for Webex Calling, make sure you have an in-depth understanding of the following concepts:

The configuration guidelines provided in this article assume a dedicated local gateway platform with no existing voice configuration. If an existing CUBE enterprise deployment is being modified to also utilize the local gateway function for Cisco Webex Calling, pay close attention to the configuration applied to ensure existing call flows and functionalities are not interrupted and make sure you're adhering to CUBE HA design requirements.

Hardware and Software Components

CUBE HA as local gateway requires IOS-XE version 16.12.2 or later and a platform on which both CUBE HA and LGW functions are supported.


The show commands and logs in this article are based on minimum software release of Cisco IOS-XE 16.12.2 implemented on a vCUBE (CSR1000v).

Reference Material

Here are some detailed CUBE HA configuration guides for various platforms:

Webex Calling Solution Overview

Cisco Webex Calling is a collaboration offering that provides a multi-tenant cloud-based alternative to on-premise PBX phone service with multiple PSTN options for customers.

The Local Gateway deployment (represented below) is the focus of this article. Local gateway (Premises-based PSTN) trunk in Webex Calling allows connectivity to a customer-owned PSTN service. It also provides connectivity to an on-premises IP PBX deployment such as Cisco Unified CM. All communication to and from the cloud is secured using TLS transport for SIP and SRTP for media.

The figure below displays a Webex Calling deployment without any existing IP PBX and is applicable to a single or a multi-site deployment. Configuration outlined in this article is based on this deployment.

Layer 2 Box-to-Box Redundancy

CUBE HA layer 2 box-to-box redundancy uses the Redundancy Group (RG) infrastructure protocol to form an active/standby pair of routers. This pair share the same virtual IP address (VIP) across their respective interfaces and continually exchange status messages. CUBE session information is check-pointed across the pair of routers enabling the standby router to take all CUBE call processing responsibilities over immediately if the active router goes out of service, resulting in stateful preservation of signaling and media.


Check pointing is limited to connected calls with media packets. Calls in transit are not check pointed (for example, a trying or ringing state).

In this article, CUBE HA will refer to CUBE High Availability (HA) Layer 2 Box-to-box (B2B) redundancy for stateful call preservation

As of IOS-XE 16.12.2, CUBE HA can be deployed as a Local Gateway for Cisco Webex Calling trunk (Premises-based PSTN) deployments and we’ll cover design considerations and configurations in this article. This figure displays a typical CUBE HA setup as Local Gateway for a Cisco Webex Calling trunk deployment.

Redundancy Group Infra Component

The Redundancy Group (RG) Infra component provides the box-to-box communication infrastructure support between the two CUBEs and negotiates the final stable redundancy state. This component also provides:

  • An HSRP-like protocol that negotiates the final redundancy state for each router by exchanging keepalive and hello messages between the two CUBEs (via the control interface)—GigabitEthernet3 in the figure above.

  • A transport mechanism for checkpointing the signaling and media state for each call from the active to the standby router (via the data interface)—GigabitEthernet3 in the figure above.

  • Configuration and management of the Virtual IP (VIP) interface for the traffic interfaces (multiple traffic interfaces can be configured using the same RG group) – GigabitEthernet 1 and 2 are considered traffic interfaces.

This RG component has to be specifically configured to support voice B2B HA.

Virtual IP (VIP) Address Management for Both Signaling and Media

B2B HA relies on VIP to achieve redundancy. The VIP and associated physical interfaces on both CUBEs in the CUBE HA pair must reside on the same LAN subnet. Configuration of the VIP and binding of the VIP interface to a particular voice application (SIP) are mandatory for voice B2B HA support. External devices such as Unified CM, Webex Calling access SBC, service provider, or proxy, use VIP as the destination IP address for the calls traversing through the CUBE HA routers. Hence, from a Webex Calling point of view, the CUBE HA pairs acts as a single local gateway.

The call signaling and RTP session information of established calls are checkpointed from the active router to the standby router. When the Active router goes down, the Standby router takes over, and continues to forward the RTP stream that was previously routed by the first router.

Calls in a transient state at the time of failover will not be preserved post-switchover. For example, calls that aren't fully established yet or are in the process of being modified with a transfer or hold function. Established calls may be disconnected post-switchover.

The following requirements exist for using CUBE HA as a local gateway for stateful failover of calls:

  • CUBE HA cannot have TDM or analog interfaces co-located

  • Gig1 and Gig2 are referred to as traffic (SIP/RTP) interfaces and Gig3 is Redundancy Group (RG) Control/data interface

  • No more than 2 CUBE HA pairs can be placed in the same layer 2 domain, one with group id 1 and the other with group id 2. If configuring 2 HA pairs with the same group id, RG Control/Data interfaces needs to belong to different layer 2 domains (vlan, separate switch)

  • Port channel is supported for both RG Control/data and traffic interfaces

  • All signaling/media is sourced from/to the Virtual IP Address

  • Anytime a platform is reloaded in a CUBE-HA relationship, it always boots up as Standby

  • Lower address for all the interfaces (Gig1, Gig2, Gig3) should be on the same platform

  • Redundancy Interface Identifier, rii should be unique to a pair/interface combination on the same Layer 2

  • Configuration on both the CUBEs must be identical including physical configuration and must be running on the same type of platform and IOS-XE version

  • Loopback interfaces cannot be used as bind as they are always up

  • Multiple traffic (SIP/RTP) interfaces (Gig1, Gig2) require interface tracking to be configured

  • CUBE-HA is not supported over a crossover cable connection for the RG-control/data link (Gig3)

  • Both platforms must be identical and be connected via a physical Switch across all likewise interfaces for CUBE HA to work, i.e. GE0/0/0 of CUBE-1 and CUBE-2 must terminate on the same switch and so on.

  • Cannot have WAN terminated on CUBEs directly or Data HA on either side

  • Both Active/Standby must be in the same data center

  • It is mandatory to use separate L3 interface for redundancy (RG Control/data, Gig3). i.e interface used for traffic cannot be used for HA keepalives and checkpointing

  • Upon failover, the previously active CUBE goes through a reload by design, preserving signaling and media

Configure Redundancy on Both CUBEs

You must configure layer 2 box-to-box redundancy on both CUBEs intended to be used in an HA pair to bring up virtual IPs.

1

Configure interface tracking at a global level to track the status of the interface.

conf t
 track 1 interface GigabitEthernet1 line-protocol
 track 2 interface GigabitEthernet2 line-protocol
 exit
VCUBE-1#conf t
VCUBE-1(config)#track 1 interface GigabitEthernet1 line-protocol
VCUBE-1(config-track)#track 2 interface GigabitEthernet2 line-protocol
VCUBE-1(config-track)#exit
VCUBE-2#conf t
VCUBE-2(config)#track 1 interface GigabitEthernet1 line-protocol
VCUBE-2(config-track)#track 2 interface GigabitEthernet2 line-protocol
VCUBE-2(config-track)#exit

Track CLI is used in RG to track the voice traffic interface state so that the active route will quite its active role after the traffic interface is down.

2

Configure an RG for use with VoIP HA under the application redundancy sub-mode.

redundancy
  application redundancy
   group 1
    name LocalGateway-HA
    priority 100 failover threshold 75
    control GigabitEthernet3 protocol 1
    data GigabitEthernet3
    timers delay 30 reload 60
    track 1 shutdown
    track 2 shutdown
    exit
   protocol 1
    timers hellotime 3 holdtime 10
   exit
  exit
 exit
VCUBE-1(config)#redundancy
VCUBE-1(config-red)#application redundancy
VCUBE-1(config-red-app)#group 1
VCUBE-1(config-red-app-grp)#name LocalGateway-HA
VCUBE-1(config-red-app-grp)#priority 100 failover threshold 75
VCUBE-1(config-red-app-grp)#control GigabitEthernet3 protocol 1
VCUBE-1(config-red-app-grp)#data GigabitEthernet3
VCUBE-1(config-red-app-grp)#timers delay 30 reload 60
VCUBE-1(config-red-app-grp)#track 1 shutdown
VCUBE-1(config-red-app-grp)#track 2 shutdown
VCUBE-1(config-red-app-grp)#exit
VCUBE-1(config-red-app)#protocol 1
VCUBE-1(config-red-app-prtcl)#timers hellotime 3 holdtime 10
VCUBE-1(config-red-app-prtcl)#exit
VCUBE-1(config-red-app)#exit
VCUBE-1(config-red)#exit
VCUBE-1(config)#
VCUBE-2(config)#redundancy
VCUBE-2(config-red)#application redundancy
VCUBE-2(config-red-app)#group 1
VCUBE-2(config-red-app-grp)#name LocalGateway-HA
VCUBE-2(config-red-app-grp)#priority 100 failover threshold 75
VCUBE-2(config-red-app-grp)#control GigabitEthernet3 protocol 1
VCUBE-1(config-red-app-grp)#data GigabitEthernet3
VCUBE-2(config-red-app-grp)#timers delay 30 reload 60
VCUBE-2(config-red-app-grp)#track 1 shutdown
VCUBE-2(config-red-app-grp)#track 2 shutdown
VCUBE-2(config-red-app-grp)#exit
VCUBE-2(config-red-app)#protocol 1
VCUBE-2(config-red-app-prtcl)#timers hellotime 3 holdtime 10
VCUBE-2(config-red-app-prtcl)#exit
VCUBE-2(config-red-app)#exit
VCUBE-2(config-red)#exit
VCUBE-2(config)#

Here's an explanation of the fields used in this configuration:

  • redundancy—Enters redundancy mode

  • application redundancy—Enters application redundancy configuration mode

  • group—Enters redundancy application group configuration mode

  • name LocalGateway-HA—Defines the name of the RG group

  • priority 100 failover threshold 75—Specifies the initial priority and failover thresholds for an RG

  • timers delay 30 reload 60—Configures the two times for delay and reload

    • Delay timer which is the amount of time to delay RG group’s initialization and role negotiation after the interface comes up – Default 30 seconds. Range is 0-10000 seconds

    • Reload—This is the amount of time to delay RG group initialization and role-negotiation after a reload – Default 60 seconds. Range is 0-10000 seconds

    • Default timers are recommended, though these timers may be adjusted to accommodate any additional network convergence delay that may occur during bootup/reload of the routers, in order to guarantee that the RG protocol negotiation takes place after routing in the network has converged to a stable point. For example, if it is seen after failover that it takes up to 20 sec for the new STANDBY to see the first RG HELLO packet from the new ACTIVE, then the timers should be adjusted to ‘timers delay 60 reload 120’ to factor in this delay.

  • control GigabitEthernet3 protocol 1—Configures the interface used to exchange keepalive and hello messages between the two CUBEs, and specifies the protocol instance that will be attached to a control interface and enters redundancy application protocol configuration mode

  • data GigabitEthernet3—Configures the interface used for checkpointing of data traffic

  • track—RG group tracking of interfaces

  • protocol 1—Specifies the protocol instance that will be attached to a control interface and enters redundancy application protocol configuration mode

  • timers hellotime 3 holdtime 10—Configures the two timers for hellotime and holdtime:

    • Hellotime— Interval between successive hello messages – Default 3 seconds. Range is 250 milliseconds-254 seconds

    • Holdtime—The interval between the receipt of a Hello message and the presumption that the sending router has failed. This duration has to be greater than the hello-time – Default 10 seconds. Range is 750 milliseconds-255 seconds

      We recommend that you configure the holdtime timer to be at least 3 times the value of the hellotime timer.

3

Enable box-to-box redundancy for the CUBE application. Configure the RG from the previous step under voice service voip. This enables the CUBE application to control the redundancy process.

voice service voip
   redundancy-group 1
   exit
VCUBE-1(config)#voice service voip
VCUBE-1(config-voi-serv)#redundancy-group 1
% Created RG 1 association with Voice B2B HA; reload the router for the new configuration to take effect
VCUBE-1(config-voi-serv)# exit
VCUBE-2(config)#voice service voip
VCUBE-2(config-voi-serv)#redundancy-group 1
% Created RG 1 association with Voice B2B HA; reload the router for the new configuration to take effect
VCUBE-2(config-voi-serv)# exit

redundancy-group 1—Adding and removing this command requires a reload for the updated configuration to take effect. We'll reload the platforms after all the configuration has been applied.

4

Configure the Gig1 and Gig2 interfaces with their respective virtual IPs as shown below and apply the redundancy interface identifier (rii)

VCUBE-1(config)#interface GigabitEthernet1
VCUBE-1(config-if)# redundancy rii 1
VCUBE-1(config-if)# redundancy group 1 ip 198.18.1.228 exclusive
VCUBE-1(config-if)# exit
VCUBE-1(config)#
VCUBE-1(config)#interface GigabitEthernet2
VCUBE-1(config-if)# redundancy rii 2
VCUBE-1(config-if)# redundancy group 1 ip 198.18.133.228 exclusive
VCUBE-1(config-if)# exit
VCUBE-2(config)#interface GigabitEthernet1
VCUBE-2(config-if)# redundancy rii 1
VCUBE-2(config-if)# redundancy group 1 ip 198.18.1.228 exclusive
VCUBE-2(config-if)# exit
VCUBE-2(config)#
VCUBE-2(config)#interface GigabitEthernet2
VCUBE-2(config-if)# redundancy rii 2
VCUBE-2(config-if)# redundancy group 1 ip 198.18.133.228 exclusive
VCUBE-v(config-if)# exit

Here's an explanation of the fields used in this configuration:

  • redundancy rii—Configures the redundancy interface identifier for the redundancy group. Required for generating a Virtual MAC (VMAC) address. The same rii ID value must be used on the interface of each router (ACTIVE/STANDBY) that has the same VIP.


     

    If there is more than one B2B pair on the same LAN, each pair MUST have unique rii IDs on their respective interfaces (to prevent collision). ‘show redundancy application group all’ should indicate the correct local and peer information.

  • redundancy group 1—Associates the interface with the redundancy group created in Step 2 above. Configure the RG group, as well as the VIP assigned to this physical interface.


     

    It is mandatory to use a separate interface for redundancy, that is, the interface used for voice traffic cannot be used as control and data interface specified in Step 2 above. In this example, Gigabit interface 3 is used for RG control/data

5

Save the configuration of the first CUBE and reload it.

The platform to reload last is always the Standby.

VCUBE-1#wr
Building configuration...
[OK]
VCUBE-1#reload
Proceed with reload? [confirm]

After VCUBE-1 boots up completely, save the configuration of VCUBE-2 and reload it.

VCUBE-2#wr
Building configuration...
[OK]
VCUBE-2#reload
Proceed with reload? [confirm]
6

Verify that the box-to-box configuration is working as expected. Relevant output is highlighted in bold.

We reloaded VCUBE-2 last and as per the design considerations; the platform to reload last will always be Standby.


VCUBE-1#show redundancy application group all
Faults states Group 1 info:
       Runtime priority: [100]
               RG Faults RG State: Up.
                       Total # of switchovers due to faults:           0
                       Total # of down/up state changes due to faults: 0
Group ID:1
Group Name:LocalGateway-HA
  
Administrative State: No Shutdown
Aggregate operational state: Up
My Role: ACTIVE
Peer Role: STANDBY
Peer Presence: Yes
Peer Comm: Yes
Peer Progression Started: Yes

RF Domain: btob-one
         RF state: ACTIVE
         Peer RF state: STANDBY HOT

RG Protocol RG 1
------------------
        Role: Active
        Negotiation: Enabled
        Priority: 100
        Protocol state: Active
        Ctrl Intf(s) state: Up
        Active Peer: Local
        Standby Peer: address 10.1.1.2, priority 100, intf Gi3
        Log counters:
                role change to active: 1
                role change to standby: 1
                disable events: rg down state 0, rg shut 0
                ctrl intf events: up 1, down 0, admin_down 0
                reload events: local request 0, peer request 0

RG Media Context for RG 1
--------------------------
        Ctx State: Active
        Protocol ID: 1
        Media type: Default
        Control Interface: GigabitEthernet3
        Current Hello timer: 3000
        Configured Hello timer: 3000, Hold timer: 10000
        Peer Hello timer: 3000, Peer Hold timer: 10000
        Stats:
            Pkts 1509, Bytes 93558, HA Seq 0, Seq Number 1509, Pkt Loss 0
            Authentication not configured
            Authentication Failure: 0
            Reload Peer: TX 0, RX 0
            Resign: TX 0, RX 0
    Standy Peer: Present. Hold Timer: 10000
            Pkts 61, Bytes 2074, HA Seq 0, Seq Number 69, Pkt Loss 0

VCUBE-1#

VCUBE-2#show redundancy application group all
Faults states Group 1 info:
       Runtime priority: [100]
               RG Faults RG State: Up.
                       Total # of switchovers due to faults:           0
                       Total # of down/up state changes due to faults: 0
Group ID:1
Group Name:LocalGateway-HA
  
Administrative State: No Shutdown
Aggregate operational state: Up
My Role: STANDBY
Peer Role: ACTIVE
Peer Presence: Yes
Peer Comm: Yes
Peer Progression Started: Yes

RF Domain: btob-one
         RF state: ACTIVE
         Peer RF state: STANDBY HOT

RG Protocol RG 1
------------------
        Role: Active
        Negotiation: Enabled
        Priority: 100
        Protocol state: Active
        Ctrl Intf(s) state: Up
        Active Peer: address 10.1.1.2, priority 100, intf Gi3
        Standby Peer: Local
        Log counters:
                role change to active: 1
                role change to standby: 1
                disable events: rg down state 0, rg shut 0
                ctrl intf events: up 1, down 0, admin_down 0
                reload events: local request 0, peer request 0

RG Media Context for RG 1
--------------------------
        Ctx State: Active
        Protocol ID: 1
        Media type: Default
        Control Interface: GigabitEthernet3
        Current Hello timer: 3000
        Configured Hello timer: 3000, Hold timer: 10000
        Peer Hello timer: 3000, Peer Hold timer: 10000
        Stats:
            Pkts 1509, Bytes 93558, HA Seq 0, Seq Number 1509, Pkt Loss 0
            Authentication not configured
            Authentication Failure: 0
            Reload Peer: TX 0, RX 0
            Resign: TX 0, RX 0
    Standy Peer: Present. Hold Timer: 10000
            Pkts 61, Bytes 2074, HA Seq 0, Seq Number 69, Pkt Loss 0

VCUBE-2#

Configure a Local Gateway on Both CUBEs

In our example configuration, we’re using the following trunk information from Control Hub to build the Local Gateway configuration on both the platforms, VCUBE-1 and VCUBE-2. The username and password for this setup are as follows:

  • Username: Hussain1076_LGU

  • Password: lOV12MEaZx

1

Ensure that a configuration key is created for the password, with the commands shown below, before it can be used in the credentials or shared secrets. Type 6 passwords are encrypted using AES cipher and this user-defined configuration key.


LocalGateway#conf t
LocalGateway(config)#key config-key password-encrypt Password123
LocalGateway(config)#password encryption aes

Here is the Local Gateway configuration that will apply to both platforms based on the Control Hub parameters displayed above, save and reload. SIP Digest credentials from Control Hub are highlighted in bold.


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


configure terminal
crypto pki trustpool import clean url
http://www.cisco.com/security/pki/trs/ios_core.p7b
end


configure terminal
voice service voip
  ip address trusted list
    ipv4 x.x.x.x y.y.y.y
    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!
  sip
    g729 annexb-all
    early-offer forced
    end


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=hussain1076_lgu>"
  rule 30 request ANY sip-header P-Asserted-Identity modify
"sips:(.*)" "sip:\1"


voice class codec 99
  codec preference 1 g711ulaw
  codec preference 2 g711ulaw
  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






voice class tenant 200
  registrar dns:40462196.cisco-bcld.com scheme sips expires 240
refresh-ratio 50 tcp tls
  credentials number Hussain5091_LGU username Hussain1076_LGU
password 0 lOV12MEaZx realm Broadworks 
  authentication username Hussain5091_LGU password 0 lOV12MEaZx
realm BroadWorks

  authentication username Hussain5091_LGU password 0 lOV12MEaZx
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


voice class tenant 100
  session transport udp
  url sip
  error-passthru
  bind control source-interface GigabitEthernet2
  bind media source-interface GigabitEthernet2
  no pass-thru content custom-sdp

voice class tenant 300
  bind control source-interface GigabitEthernet2
  bind media source-interface GigabitEthernet2
  no pass-thru content custom-sdp
  

voice class uri 100 sip
 host ipv4:198.18.133.3

voice class uri 200 sip
 pattern dtg=hussain1076.lgu



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

dial-peer voice 201 voip
 description Outgoing dial-peer to Webex Calling
 destination-pattern BAD.BAD
 session protocol sipv2
 session target sip-server
 voice-class codec 99
 voice-class stun-usage 200
 no voice-class sip localhost
 voice-class sip tenant 200
 dtmf-relay rtp-nte
 srtp
 no vad


voice class dpg 100
 description Incoming WebexCalling(DP200) to IP PSTN(DP101)
 dial-peer 101 preference 1

voice class dpg 200
 description Incoming IP PSTN(DP100) to Webex Calling(DP201)
 dial-peer 201 preference 1





dial-peer voice 100 voip
 desription Incoming dial-peer from IP PSTN
 session protocol sipv2
 destination dpg 200
 incoming uri via 100
 voice-class codec 99
 voice-class sip tenant 300
 dtmf-relay rtp-nte
 no vad

dial-peer voice 200 voip
 description Incoming dial-peer from Webex Calling
 session protocol sipv2
 destination dpg 100
 incoming uri request 200
 voice-class codec 99
 voice-class stun-usage 200
 voice-class sip tenant 200
 dtmf-relay rtp-nte
 srtp
 no vad

end

copy run start

To display the show command output, we've reloaded VCUBE-2 followed by VCUBE-1, making VCUBE-1 the standby CUBE and VCUBE-2 the active CUBE

2

At any given time, only one platform will maintain an active registration as the Local Gateway with the Webex Calling access SBC. Take a look at the output of the following show commands.

show redundancy application group 1

show sip-ua-register status


VCUBE-1#show redundancy application group 1
Group ID:1
Group Name:LocalGateway-HA

Administrative State: No Shutdown
Aggregate operational state : Up
My Role: Standby
Peer Role: ACTIVE
Peer Presence: Yes
Peer Comm: Yes
Peer Progression Started: Yes

RF Domain: btob-one
         RF state: STANDBY HOT
         Peer RF state: ACTIVE

VCUBE-1#show sip-ua register status
VCUBE-1#

VCUBE-2#show redundancy application group 1
Group ID:1
Group Name:LocalGateway-HA

Administrative State: No Shutdown
Aggregate operational state : Up
My Role: ACTIVE
Peer Role: STATUS
Peer Presence: Yes
Peer Comm: Yes
Peer Progression Started: Yes

RF Domain: btob-one
         RF state: ACTIVE
         Peer RF state: STANDBY HOT

VCUBE-2#show sip-ua register status

Tenant: 200
--------------------Registrar-Index  1 ---------------------
Line                           peer       expires(sec) reg survival P-Associ-URI
============================== ========== ============ === ======== ============
Hussain5091_LGU                -1          48          yes normal
VCUBE-2#

From the output above, you can see that VCUBE-2 is the active LGW maintaining the registration with Webex Calling access SBC, whereas the output of the “show sip-ua register status” is blank in VCUBE-1

3

Now enable the following debugs on VCUBE-1


VCUBE-1#debug ccsip non-call
SIP Out-of-Dialog tracing is enabled
VCUBE-1#debug ccsip info
SIP Call info tracing is enabled
VCUBE-1#debug ccsip message
4

Simulate failover by issuing the following command on the active LGW, VCUBE-2 in this case.


VCUBE-2#redundancy application reload group 1 self

Switchover from the ACTIVE to the STANDBY LGW occurs in the following scenario as well besides the CLI listed above

  • When the ACTIVE router reloads

  • When the ACTIVE router power cycles

  • When any RG configured interface of the ACTIVE router is shutdown for which tracking is enabled

5

Check to see if VCUBE-1 has registered with Webex Calling access SBC. VCUBE-2 would have reloaded by now.


VCUBE-1#show sip-ua register status

Tenant: 200
--------------------Registrar-Index  1 ---------------------
Line                           peer       expires(sec) reg survival P-Associ-URI
============================== ========== ============ === ======== ============
Hussain5091_LGU                -1          56          yes normal
VCUBE-1#

VCUBE-1 is now the active LGW.

6

Look at the relevant debug log on VCUBE-1 sending a SIP REGISTER to Webex Calling VIA the virtual IP and receiving a 200 OK.


VCUBE-1#show log

Jan 9 18:37:24.769: %RG_MEDIA-3-TIMEREXPIRED: RG id 1 Hello Time Expired.
Jan 9 18:37:24.771: %RG_PROTCOL-5-ROLECHANGE: RG id 1 role change from Standby to Active
Jan 9 18:37:24.783: %VOICE_HA-2-SWITCHOVER_IND: SWITCHOVER, from STANDBY_HOT to ACTIVE state.
Jan 9 18:37:24.783: //-1/xxxxxxxxxxxx/SIP/Info/info/4096/sip_ha_notify_active_role_event: Received notify active role event

Jan 9 18:37:25.758: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip: 40462196.cisco-bcld.com:5061 SIP/2.0
Via: SIP/2.0/TLS 198.18.1.228:5061;branch=z9hG4bK0374
From: <sip:Hussain5091_LGU@40462196.cisco-bcld.com;otg=hussain1076_lgu>;tag=8D573-189
To: <sip:Hussain5091_LGU@40462196.cisco-bcld.com>
Date: Thu, 09 Jan 2020 18:37:24 GMT
Call-ID: FFFFFFFFEA0684EF-324511EA-FFFFFFFF800281CD-FFFFFFFFB5F93B97
User-Agent: Cisco-SIPGateway/IOS-16.12.02
Max-Forwards: 70
Timestamp: 1578595044
CSeq: 2 REGISTER
Contact: <sip:Hussain5091_LGU@198.18.1.228:5061;transport=tls>
Expires: 240
Supported: path
Content-Length: 0
Jan 9 18:37:25.995: //-1/000000000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 198.18.1.228:5061;received=173.38.218.1;branch=z9hG4bK0374;rport=4742
From: <sip:Hussain5091_LGU@40462196.cisco-bcld.com;otg=hussain1076_lgu>;tag=8D573-189
To: <sip:Hussain5091_LGU@40462196.cisco-bcld.com>;tag=SD1u8bd99-1324701502-1578595045969
Date: Thu, 09 Jan 2020 18:37:24 GMT
Call-ID: FFFFFFFFEA0684EF-324511EA-FFFFFFFF800281CD-FFFFFFFFB5F93B97
Timestamp: 1578595044
CSeq: 2 REGISTER
WWW-Authenticate; DIGEST realm="BroadWorks",qop="auth",nonce="BroadWorksXk572qd01Ti58zliBW",algorithm=MD5
Content-Length: 0
Jan 9 18:37:26.000: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
REGISTER sip:40462196.cisco-bcld.com:5061 SIP/2.0
Via: SIP/2.0/TLS 198.18.1.228:5061;branch=z9hG4bK16DC
From: <sip:Hussain5091_LGU@40462196.cisco-bcld.com;otg=hussain1076_lgu>;tag=8D573-189
To: <sip:Hussain5091_LGU@40462196.cisco-bcld.com>
Date: Thu, 09 Jan 2020 18:37:25 GMT
Call-ID: FFFFFFFFEA0684EF-324511EA-FFFFFFFF800281CD-FFFFFFFFB5F93B97
User-Agent:Cisco-SIPGateway/IOS-16.12.02
Max-Forwards: 70
Timestamp: 1578595045
CSeq: 3 REGISTER
Contact: <sip:Hussain5091_LGU@198.18.1.228:5061;transport=tls>
Expires: 240
Supported: path
Authorization: Digest username="Hussain1076_LGU",realm="BroadWorks",uri="sips:40462196.cisco-bcld.com:5061",response="b6145274056437b9c07f7ecc08ebdb02",nonce="BroadWorksXk572qd01Ti58z1iBW",cnonce="3E0E2C4D",qop=auth,algorithm=MD5,nc=00000001
Content-Length: 0
Jan 9 18:37:26.190: //1/000000000000/SIP/Msg/ccsipDisplayMsg:

Received:
SIP/2.0 200 OK
Via: SIP/2.0/TLS 198.18.1.228:5061;received=173.38.218.1;branch=z9hG4bK16DC;rport=4742
From: <sip:Hussain5091_LGU@40462196.cisco-bcld.com;otg=hussain1076_lgu>;tag=8D573-189
To: <sip:Hussain5091_LGU@40462196.cisco-bcld.com>;tag=SD1u8bd99-1897486570-1578595-46184
Call-ID: FFFFFFFFEA0684EF-324511EA-FFFFFFFF800281CD-FFFFFFFFB5F93B97
Timestamp: 1578595045
CSeq: 3 REGISTER
Contact: <sip:Hussain5091_LGU@198.18.1.228:5061;transport=tls>;expires=120;q=0.5
Allow-Events: call-info,line-seize,dialog,message-summary,as-feature-event,x-broadworks-hoteling,x-broadworks-call-center-status,conference
Content-Length: 0
Feb 19, 2022
Configure Unified CM

You may require an integration with Unified CM if Webex Calling-enabled locations are added to an existing deployment where Unified CM is the on-premises call control solution and if you require direct dialing between phones registered to Unified CM and phones in Webex Calling locations.

Configure SIP Trunk Security Profile for Trunk to Local Gateway

In cases where Local Gateway and PSTN gateway reside on the same device, Unified CM must be enabled to differentiate between two different traffic types (calls from Webex and from the PSTN) that are originating from the same device and apply differentiated class of service to these call types. This differentiated call treatment is achieved by provisioning two trunks between Unified CM and the combined local gateway and PSTN gateway device which requires different SIP listening ports for the two trunks.

Create a dedicated SIP Trunk Security Profile for the Local Gateway trunk with the following settings:

Setting Value
Name Unique Name, such as Webex
Description Meaningful description, such as Webex SIP Trunk Security Profile
Incoming Port Needs to match port used in local gateway config for traffic to/from Webex: 5065

Configure SIP Profile for the Local Gateway Trunk

Create a dedicated SIP Profile for the Local Gateway trunk with the following settings:

Setting Value
Name Unique Name, such as Webex
Description Meaningful description, such as Webex SIP Profile
Enable OPTIONS Ping to monitor destination status for Trunks with Service Type “None (Default)” Checked

Create a Calling Search Space for Calls From Webex

Create a calling search space for calls originating from Webex with the following settings:

Setting Value
Name Unique Name, such as Webex
Description Meaningful description, such as Webex Calling Search Space
Selected Partitions

DN (+E.164 directory numbers)

ESN (abbreviated inter-site dialling)

PSTNInternational (PSTN access)

onNetRemote (GDPR learned destinations)


 

The last partition onNetRemote is only used in a multi-cluster environment where routing information is exchanged between Unified CM clusters using Intercluster Lookup Service (ILS) or Global Dialplan Replication (GDPR).

Configure a SIP Trunk To and From Webex

Create a SIP trunk for the calls to and from Webex via the Local Gateway with the following settings:

Setting Value
Device Information
DeviceName A unique name, such as Webex
Description Meaningful description, such as Webex SIP Trunk
Run On All Active Unified CM Nodes Checked
Inbound Calls
Calling Search Space The previously defined calling search space: Webex
AAR Calling Search Space A calling search space with only access to PSTN route patterns: PSTNReroute
SIP Information
Destination Address IP address of the Local Gateway CUBE
Destination Port 5060
SIP Trunk Security Profile Previously defined: Webex
SIP Profile Previously defined: Webex

Configure Route Group for Webex

Create a route group with the following settings:

Setting Value
Route Group Information
Route Group Name A unique name, such as Webex
Selected Devices The previously configured SIP trunk: Webex

Configure Route List for Webex

Create a route list with the following settings:

Setting Value
Route List Information
Name A unique name, such as RL_Webex
Description Meaningful description, such as Route list for Webex
Run On All Active Unified CM Nodes Checked
Route List Member Information
Selected Groups Only the previously defined route group: Webex

Create a Partition for Webex Destinations

Create a partition for the Webex destinations with the following settings:

Setting Value
Route List Information
Name Unique name, such as Webex
Description Meaningful description, such as Webex Partition

What to do next

Make sure to add this partition to all calling search spaces that should have access to Webex destinations. You must add this partition specifically to the calling search space that is used as the inbound calling search space on PSTN trunks, so that calls from the PSTN to Webex can be routed.

Configure Route Patterns for Webex Destinations

Configure route patterns for each DID range on Webex with the following settings:

Setting Value
Route Pattern Full +E.164 pattern for the DID range in Webex with the leading “\”. For example: \+140855501XX
Route Partition Webex
Gateway/Route List RL_Webex
Urgent Priority Checked

Configure Abbreviated Intersite Dialing Normalization for Webex

If abbreviated inter-site dialing is required to Webex, then configure dialing normalization patterns for each ESN range on Webex with the following settings:

Setting Value
Translation Pattern ESN pattern for the ESN range in Webex. For example: 80121XX
Partition Webex
Description Meaningful description, such as Webex Normalization Pattern
Use Originator's Calling Search Space Checked
Urgent Priority Checked
Do Not Wait For Interdigit Timeout On Subsequent Hops Checked
Called Party Transformation Mask Mask to normalize the number to +E.164. For example: +140855501XX
May 2, 2022
Set Up Your Webex Calling Features

Learn more about some of the features available in Webex Calling and how to set them up for your organization and users.

Set up a hunt group

Hunt groups route incoming calls to a group of users or workspaces. You can even configure a pattern to route to a whole group.

For more information on how to set up a hunt group, see Hunt Groups in Cisco Webex Control Hub.

Create a call queue

You can set up a call queue so that when customers' calls can't be answered, they're provided with an automated answer, comfort messages, and music on hold until someone can answer their call.

For more information on how to set up and manage a call queue, see Manage Call Queues in Cisco Webex Control Hub.

Create a receptionist client

Help support the needs of your front-office personnel. You can set up users as telephone attendants so they can screen incoming calls to certain people within your organization.

For information about how to set up and view your receptionist clients, see Receptionist Clients in Cisco Webex Control Hub.

Create and manage auto attendants

You can add greetings, set up menus, and route calls to an answering service, a hunt group, a voicemail box, or a real person. Create a 24-hour schedule or provide different options when your business is open or closed.

For information about how to create and manage auto attendants, see Manage Auto Attendants in Cisco Webex Control Hub.

Configure a paging group

Group paging allows a user to place a one-way call or group page to up to 75 target users and workspaces by dialing a number or extension assigned to a specific paging group.

For information about how to set up and edit paging groups, see Configure a Paging Group in Cisco Webex Control Hub.

Set up call pickup

Enhance teamwork and collaboration by creating a call pickup group so users can answer each others calls. When you add users to a call pickup group and a group member is away or busy, another member can answer their calls.

For information about how to set up a call pickup group, see Call Pickup in Cisco Webex Control Hub.

Set up call park

Call park allows a defined group of users to park calls against other available members of a call park group. Parked calls can be picked up by other members of the group on their phone.

For more information about how to set up call park, see Call Park in Cisco Webex Control Hub.

Allow users to barge in to other people's phone calls

1

From the customer view in https://admin.webex.com, go to Users, and then select the user you want to modify.

2

Select Calling, go to Advanced Call Settings, and then select Barge In.

3

Turn on Barge In, choose whether you want the phone to play a sound when someone barges into a call, and then click Save.

Prevent someone from monitoring a user's line status

1

From the customer view in https://admin.webex.com, go to Users, and select the user you want to modify.

2

Select Calling and then go to Privacy.

3

Choose the appropriate Auto Attendant Privacy settings for this user.

4

Check the Enable Privacy check box. You can then decide whether to block everyone by leaving the Search user by name field empty or choose who can monitor this user's line status.

Using the executive example above, you'd search for the name of their administrative assistant.

5

Click Save.

Example

Want to see how it's done? Watch this video demonstration on how to manage privacy settings for a user in Control Hub.

Monitoring List - Other Users and Call Park Extentions

The maximum number of monitored lines is 50 but you should consider bandwidth. The maximum may also be determined by the number of line buttons on the user's phone.


The monitoring service only works with a user's primary device.

1

From the customer view in https://admin.webex.com, go to Users, and select the user you want to modify.

2

Select Calling, choose Advanced Call Settings, and then go to Monitoring.

3

Choose from the following:

  • Add Monitored Line
  • Add Call Park Extension
4

Choose whether you want this user to be notified about parked calls, search for the person or call park extension to be monitored, and then click Save.


 

The monitored lines list in Control Hub corresponds with the order of monitored lines that show on the user’s device. You can re-order the list of monitored lines at anytime.


 

The name that appears for the monitored line is the name entered in the user or workspace's Caller ID First Name and Last Name fields.

Example

Want to see how it's done? Watch this video demonstration on how to manage monitoring settings for a user in Control Hub.

Turn on hoteling for a user

Enabling hoteling for a user allows them to work in another space while maintaining the functionality and features of their main desk phone.

1

From the customer view in https://admin.webex.com, go to Users and then select the user you want to modify.

2

Select Calling, choose Advanced Call Settings, and click Hoteling.

3

Turn on Hoteling, and then click Save.

Example

Want to see how it's done? Watch this video demonstration on how to configure your music-on-hold settings in Control Hub.
Feb 19, 2022
Configure and Manage Your Users

You must add each and every user in Control Hub in order for them to take advantage of Webex Calling services. The number of users you need to add will determine how you add them in Control Hub, whether you manually add each user by email address or add multiple users using a CSV file. The choice is yours.


If you synchronize users from a directory such as Active Directory, when you manually add people in Control Hub you must also add them to your directory.


When adding users, first and last names must not include extended ascii characters or the following characters %, #, <, >, \, /,", and have a maximum length of 30 characters. These special character restrictions only apply to Webex Calling users.

Before you begin

You may get an error if you're trying to add users who used their e-mail address to create a trial account. Have the users delete their organization first before adding them to your organization.

1

From the customer view in https://admin.webex.com go to Users, and then click Manage Users.

2

Select Manually Add or Modify Users.

3

(Optional) If you automatically send welcome emails, then click Next.

4

Choose one and click Next:

  • Select Email address, and enter up to 25 email addresses.
  • Select Names and Email addresses, and then enter up to 25 names and email addresses.

 

You can add users who are available to convert to your organization.

5

License assignment:

  • If you have an active license template, licenses are assigned automatically for new users and you can review the license summary.
  • Select the services to assign. If you have multiple subscriptions, choose a subscription from the list.


 

If you’re assigning licenses for Contact Center, select Webex Teams, then Customer Care with the Premium and Standard Agent option. To add a supervisor, select both Premium and Supervisor options. A user is treated as an agent unless you make them a supervisor.

6

Content management:

  • If global access is selected for your enterprise content management, then content management is automatically assigned to users.
  • Choose a content management option for each user.

7

Click Save.

  • An email is sent to each person with an invite to join.

  • In Control Hub, people appear in an invite pending state until they sign in for the first time. Licenses are assigned after the user signs in the first time or if you use Cisco Directory Connector with a claimed domain, the licenses are assigned when users are created.

8

(Optional) If you added Calling to the user, assign a location, phone number, and extension.

9

Review the summary page of records processed, and click Finish.


 

Immediately after adding a calling user, if an error is received when selecting the user Calling Settings, we recommend that you remove the Webex Calling license and then reassign the calling license to the user.

What to do next

You can assign administrative privileges to people in your organization.

Before you begin

If you have more than one CSV file for your organization, then upload one file and once that task has completed, you can upload the next file.

For customers in the Asia-Pacific region (including Japan, China, and Hong Kong), the Caller ID auto populates from the First Name and Last Name fields, and the Caller ID First Name and Caller ID Last Name fields are ignored in the CSV upload.


Some spreadsheet editors remove the + sign from cells when the .csv is opened. We suggest you use a text editor to make .csv updates. If you use a spreadsheet editor make sure to set the cell format to text, and add back any + signs that were removed.


Export a new CSV in order to capture the latest fields and avoid errors in the import of changes.

1

From the customer view in https://admin.webex.com, go to Users, click Manage Users and choose CSV Add or Modify Users.

2

Click Export to download the file and you can enter user information in a new line in the CSV file.

  • To assign a service, add TRUE in that service's column, and to exclude a service, add FALSE. The User ID/Email (Required) column is the only required field. If you have specific directory and external numbers for each new user, then include the leading + for external numbers without other characters,

    If you have an active license template, leave all the service columns blank and the template is automatically assigned for the new user in that row.


     

    You can't assign enterprise content management permissions to users using the license template, see Configure Enterprise Content Management Settings in Cisco Webex Control Hub for details.

  • To assign a location, enter the the name in the Location column. If you leave this field blank, the user is assigned to the default location.

  • If you’re adding users as supervisors for Cisco Webex Contact Center, then you must Add Users Manually. You can only assign Standard and Premium roles with a CSV.

 

When entering a user's name, make sure to include their last name, otherwise you may run into issues.

3

Click Import, select your file, and click Open.

4

Choose either Add services only or Add and remove services.

If you have an active license template, choose Add services only.

5

Click Submit.

The CSV file is uploaded and your task is created. You can close the browser or this window and your task continues to run. To review the progress of your task, see Manage Tasks in Cisco Webex Control Hub.

1

From the customer view in https://admin.webex.com go to Users.

2

Select a user and click Services > Edit Licenses.

3

If you have multiple subscriptions, choose a subscription from the list.

4

Select the services to add or remove, and click Next.

5

If you assigned a Webex Meetings license, choose an account type to assign the user with for each Webex Meetings site, and click Save.


 

You must have the Attendee account feature enabled for your Webex site to assign users as attendees. If you don't see the Attendee account column in the CSV file, then contact your Customer Success Manager (CSM), Partner Success Manager (PSM), or the Cisco Technical Assistance Center (TAC) to enable this feature for your Webex site.

The attendee account type isn't available for users with the Webex Site Administrator role. If you want to assign these users with an attendee account, you must remove their administrative privileges for that Webex Meetings site.


 

Immediately after adding a Calling license, if an error is received when selecting the user Calling settings, we recommend that you remove the Webex Calling license and then reassign the license to the user.

Before you begin

If you have more than one CSV file for your organization, then upload one file and once that task has completed, you can upload the next file.

You can’t delete users or change the location assigned to a user with the CSV template.


Some spreadsheet editors remove the + sign from cells when the .csv is opened. We suggest you use a text editor to make .csv updates. If you use a spreadsheet editor make sure to set the cell format to text, and add back any + signs that were removed.


Export a new CSV in order to capture the latest fields and avoid errors in the import of changes.

1

From the customer view in https://admin.webex.com, go to Users, click Manage Users, and choose CSV Add or Modify User.

2

(Optional) If you automatically send welcome emails, then click Next.

3

Click Export to download the file. You can edit the downloaded file (exported_users.csv) in any of the following ways:

  • To modify existing users, you can update any column except User ID/Email (Required), and Location. For example, if you change User ID/Email this creates a new user.

  • To assign a location, enter the the name in the Location column. If you leave this field blank, the user is assigned to the default location.

  • To assign a service, add TRUE in that service's column, and to exclude a service, add FALSE.

  • When you have multiple subscriptions, you can use the subscription ID in the column header to identify the service you want to add. For example, if you have two subscriptions with the same service, you can specify a service from a specific subscription to apply to the user.

4

Enter a value in the Calling Behavior column if you want to change the way calls happen for specific users. You can enter one of the following options and see Set Up Webex App Calling Behavior for more information on each setting:

  • USE_ORG_SETTINGS—Enter this string to use the organization-wide setting.

  • NATIVE_WEBEX_TEAMS_CALLING—Enter this string to use the Calling in Webex Teams option.

  • CALL_WITH_APP_REGISTERED_FOR_WEBEXCALLTEL—Enter this string to use the Webex Calling app option.

5

Enter a Caller ID Number, Caller ID First Name, and Caller ID Last name. If you leave the Caller ID Number, Caller ID First Name, and Caller ID Last name columns blank, then what is in the First Name, Last Name and Phone Number column will show when the user makes a call. If you leave the Caller ID Number blank then the Location Main Number shows when the user makes a call.


 

The Caller ID First Name and Caller ID Last Name columns can’t contain special characters. If a Caller Caller ID First Name or Caller ID Last Name contains a special character, then a simplified version of the name is used.

6

After you save the CSV file, click Import, select the file you made changes to, and then click Open.

7

Choose either Add services only or Add and remove services, and click Submit.


 

A user can't have two Calling licenses, so if your organization has multiple subscriptions, and you want to move users to a new subscription, choose the Add and remove services option. To add services, set the cells to TRUE and remove services by setting those cells to FALSE.

The CSV file is uploaded and your task is created. You can close the browser or this window and your task continues to run. To review the progress of your task, see Manage Tasks in Cisco Webex Control Hub.

If you don't suppress admin invite emails, new users receive activation emails.

You can assign numbers, extensions, or both to people's devices at any time. Assigned extensions show up on phone displays.

You can also configure alternate numbers so that multiple phone numbers ring the same phone. You can specify different ring tones for each number to help distinguish between which lines are being called.

1

From the customer view in https://admin.webex.com, go to Users, and then choose the person you want to assign a number to.

2

Select Calling and then click Add Number.

3

Choose a phone number from the list of available numbers. You also have the option of assigning an extension.

If a number is already assigned to the user, any additional number added to the user is added as an alternate number. You can add up to 10 alternate numbers to a user.

4

(Optional) To identify calls coming from specific phone numbers, you can assign a distinctive ring pattern. To enable, click the toggle under Distinctive Ring Pattern.

5

Click Save.

1

From the customer view in https://admin.webex.com, go to Users, filter the Status column to display people with an Invite Pending status.

2

Under Actions, for a person with an Invite Pending status, select more > Resend Invitation.

If your organization uses directory synchronization, the delete option is not available in Control Hub, and you must delete user accounts from your Active Directory. Then, the Cisco Directory Connector updates your organizations user list when it synchronizes the user account information.

From the customer view in https://admin.webex.com, go to Users, click the more button, and then click Delete User.

The user can no longer sign in to your Webex site, all their assigned Webex services are removed, and they are removed from any spaces or teams that they were participating in. Any content that they created in spaces is not deleted, and the content is subject to the retention policy that each space owner has implemented.

You can deactivate a user to turn off Webex services, including Webex Calling services. As opposed to deleting a user, when you deactivate the user, the user remains in your user list, so that you can reactivate at any time, when needed.

1

From the customer view in https://admin.webex.com/, go to Users.

2

Click the more button.

3

Click Deactivate User.

Webex services, including Webex Calling services, are now deactivated for this user.

When deactivated, both the Webex app and the Webex Calling app users will be signed out from their sessions. All user access to https://settings.webex.com/ and Control Hub is disallowed. MPP phones will continue to support calling for outbound and inbound calls for a short period of time, unless the administrator enables call intercept for that user. For more information about call intercept, see Configure Call Intercept for a User for Webex Calling in Cisco Control Hub.

You can set up a customer administrator with different privilege levels. They can be full administrators, support administrators, read-only administrators or compliance officers. With full administrator privileges, you can assign one or more roles to any user in your organization.


Anyone assigned the user and device administrator or device administrator role will not be able to administer Webex Calling.

In Control Hub, you can learn about different privilege levels and set up a customer administrator. Customer administrators can be full administrators, support administrators, user and device administrators, device administrators, read-only administrators, or compliance officers. With full administrator privileges, you can assign one or more roles to any user in your organization.

You’ll always want to have more than one administrator for an organization. It’s a best practice and will always allow you to make administrative changes if one of the administrators isn't available.

Users within your organization can be assigned specific administrative roles to determine what they can see and have access to in Control Hub. When you assign specific administrative roles, you streamline responsibilities and make it easier to hold administrators accountable. Compliance officers can look for specific people in your company, find content they've shared, or search through a specific space and then generate a report of their findings.


1

From the customer view in https://admin.webex.com, go to Users, and choose a user.

2

Under Roles and Security click Administrator Roles or Service Access.

3

Select a role to assign to that user.

To assign a user as a Webex Site administrator, next to Webex Site administrator roles, click Edit and choose a role for each Webex site that you want the user to manage.


 

If users with existing service admin roles (such as Webex Site admin) gain or lose an org level admin role, then their service admin roles could be changed. You should review the service admin roles for those users to make sure they’re correct.

4

Select Save.

May 4, 2022
Configure and Manage Devices

You can assign and manage devices for users and workspaces in Control Hub. Choose to add by the MAC address or by generating an activation code to enter on the device itself.

With Control Hub, you can assign devices to users for personal usage and then register those devices to the cloud.

The devices listed here support Webex Calling. While all of these devices can be registered using a MAC address, only the following subset can be registered using an activation code:

  • Cisco IP Phone 6800 Series Multiplatform Phones (Audio phones—6821, 6841, 6851, 6861, 6871)

  • Cisco IP Phone 7800 Series Multiplatform Phones (Audio phones—7811, 7821, 7841, 7861)

  • Cisco IP Phone 8800 Series Multiplatform Phones (Audio phones—8811, 8841, 8851, 8861)

  • Cisco IP Phone 8800 Series Multiplatform Phones (Video phones—8845, 8865)

  • Cisco IP Conference Phone 7832 and 8832


With regards to DECT devices, only DECT base devices (not DECT handsets) are available for assignment in Control Hub. After you assign a base unit to a user, you must then manually pair a DECT handset to that base unit. For more information, see Connect the Handset to the Base Station.

1

From the customer view in https://admin.webex.com, go to Devices and then click Add Device.


 
You can also add a phone to a user in the user's profile. See how in Manage a device for a user section.
2

Choose Existing User, enter the phone's owner, either part of the username or the user's real name, choose the user from the results, and then click Next.

3

Choose the device from the drop-down list, and then click Next.

4

Choose one of the following options and then click Save:

  • By Activation Code—Choose this option if you want to generate an activation code that you can share with the device owner. The 16-digit activation code must be manually entered onto the device itself.

     

    Multiplatform phones must have a firmware load of 11.2.3MSR1 or later to display the activation code screen. If phone firmware needs to be updated, point users to https://upgrade.cisco.com/MPP_upgrade.html.

  • By MAC Address—Choose this option if you know the MAC address of the device. A phone's MAC address must be a unique entry. If you enter a MAC address for a phone that's already registered or you make a mistake when you enter the number, an error message appears.

 

Limitations may apply when using third-party devices.

If you chose to generate an activation code for the device but you haven't yet used that code, the status of that device reads as Activating in the assigned user's Devices section and the main Devices list in Control Hub. Keep in mind it may take up to 10 minutes for device status to be updated in Control Hub.

When people are at work, they get together in lots of places like lunch rooms, lobbies, and conference rooms. You can set up shared Cisco Webex devices in these Workspaces, add services, and then watch the collaboration happen.

The key principle of a Workspaces device is that it is not assigned to a specific user, but rather a physical location, allowing for shared usage.

The devices listed support Webex Calling. While most of these devices can be registered using a MAC address, only the following subset can be registered using an activation code:

  • Cisco IP Phone 6800 Series Multiplatform Phones (Audio phones—6821, 6841, 6851)

  • Cisco IP Phone 7800 Series Multiplatform Phones (Audio phones—7811, 7821, 7841, 7861)

  • Cisco IP Phone 8800 Series Multiplatform Phones (Audio phones—8811, 8841, 8851, 8861)

  • Cisco IP Phone 8800 Series Multiplatform Phones (Video phones—8845, 8865)

  • Cisco IP Conference Phone 7832 and 8832

1

From the customer view in https://admin.webex.com, go to Management > Workspaces, and then click Add Workspace.

2

Enter a name for the workspace (such as the name of the physical room), select room type and add capacity. Then click Next.


 

A workspace name can't be longer than 30 characters and it can't have %, #, <, >, /, \, and " characters.

3

Choose Cisco IP Phone and then click Next.

4

Select the device type from the drop-down list, choose whether you want to register the phone with an activation code (if the option appears) or a MAC address, and then click Next. Keep in mind that if you choose to register the device using an activation code, the code is emailed to the designated administrator for the location.

For Webex Calling, you can only add one shared phone to a Workspace.

For Cisco IP Conference Phone 7832, some softkeys may not be available. If you need a full set of softkeys, we recommend that you assign this phone to a user instead.

5

Assign a Location and Phone Number (determined by the location that you choose), and then click Save. You also have the option of assigning an extension.


User's with a Webex Calling professional license can use their personal room system device to make (or receive) external calls using a phone number or use extension-based calling from the device.


Calls made using URI will continue to be routed through the Webex app.

1

From the customer view in https://admin.webex.com, go to Users, and select the user you want to assign the device to.

2

From the user panel that opens to the right, scroll down to Devices and then choose one of the following options:

  • If the user has at least one device already assigned—click ... and then select Add Webex Room Device.
  • If the user doesn't have any devices already assigned—click Add Webex Room Device.
3

Copy, email, or print the 16-digit activation code and send it to the user so that they can activate their new device or, if the device is in your possession, you can activate the device on the user's behalf.

If the user doesn't activate the device before the code expires, they can generate a new activation code from https://settings.webex.com. Users can also add their own personal devices from there. For more information, see Set Up a Room or Desk Device as a Personal Device.


User's with a Webex Calling professional license can use their personal room system device to make (or receive) external calls using a phone number or use extension-based calling from the device.


Calls made using URI will continue to be routed through the Webex app.

1

From the customer view in https://admin.webex.com, go to Devices.

2

Click on Add Device, and select the Existing User option.

3

Search for the user you'd like to assign the device to, then click Next.

4

Select Cisco Webex Room Device.

5

Copy, email, or print the 16-digit activation code and send it to the user so that they can activate their new device or, if the device is in your possession, you can activate the device on the user’s behalf.

If the user doesn’t activate the device before the code expires, they can generate a new activation code from https://settings.webex.com. Users can also add their own personal devices from there. For more information, see Set Up a Webex Board, Room or Desk Device as a Personal Device.

When people are at work, they get together in lots of workspaces like lunch rooms, lobbies, and conference rooms. You can set up shared Cisco Webex devices in these Workspaces, add services, and then watch the collaboration happen.

The key principle of a Workspaces device is that it is not assigned to a specific user, but rather a physical location, allowing for shared usage.

The devices listed here support Webex Calling.

1

From the customer view in https://admin.webex.com, go to Workspaces, and then click Add Workspace.

2

Enter a name for the workspace (such as the name of the physical room), select room type and add capacity. Then click Next.

3

Choose Other Cisco Webex Device and then click Next.

Other Cisco Webex Devices include Cisco Webex Room or Desk device, including Cisco Webex Board.

4

Choose one of the following options:

  • Free Calling—Users can only make Webex App or Webex Session Initiation Protocol (SIP) calls using a SIP address (for example, username@example.calls.webex.com).
  • Webex Calling—In addition to being able to make and receive Webex App and SIP calls, people in this Workspace can use the device to make and receive phone calls from within the Webex Calling numbering plan. For example, you can call your coworker Giacomo Edwards by dialing his phone number 555-555-5555, his extension 5555, or his SIP address gedwards@example.webex.com but you can also call your local pizzeria.
5

Activate the device by using the code provided. You can copy, email, or print the activation code.

If you have several devices that you must assign to users and workspaces, you can populate a CSV file with the required information and activate those devices in just a couple of easy steps.

The devices listed here support Webex Calling. While all these devices can be registered using a MAC address, only the following subset can be registered using an activation code:

  • Cisco IP Phone 6800 Series Multiplatform Phones (Audio phones—6821, 6841, 6851)

  • Cisco IP Phone 7800 Series Multiplatform Phones (Audio phones—7811, 7821, 7841, 7861)

  • Cisco IP Phone 8800 Series Multiplatform Phones (Audio phones—8811, 8841, 8851, 8861)

  • Cisco IP Phone 8800 Series Multiplatform Phones (Video phones—8845, 8865)

  • Cisco IP Conference Phone 7832 and 8832

1

From the customer view in https://admin.webex.com, go to Devices, click Add Device, and then choose whether you're adding the device to a user or a workspace.

2

Select Import/upload CSV file.

3

Choose one of the following options:

  • Users in my organization—You can get a list of all of the users in your organization and their associated attributes so you don't have to manually look up each user.
  • Add device sample template—You can use a template that we've come up with and then enter information such as usernames, type (indicate whether it's a user or a workspace), MAC addresses, and device models. Here are a few things to keep in mind:
    • You must enter a Phone Number, Extension, or both. Note: These fields were previously titled Directory Number and Direct Line; these column names will continue to be supported for a short period of time.

    • For the Username column of the CSV file, make sure you enter the user's email address, not their user ID or their name. You can also insert a workspace name in this column.

    • We recommend that you limit the number of devices to 1000 per CSV file. If you must add more than that, use a second CSV file.

    • If you enter a workspace that doesn't yet exist, the workspace is automatically created.

    • If Device Type is IP a Model is required (for example, Cisco 7841, Cisco 8851, and so on), if Device Type is WEBEX or WEBEX_CALLING a Model should be blank.

    • If you leave the MAC address column blank, an activation code is generated and must be entered on the device itself.

4

If the MAC address has been left blank, you can choose where the activation code gets sent:

  • Provide a link—The activation code gets added to a CSV file that you can then download.
  • Email activation code—If the device is for a workspace, the activation code gets sent to you, as the administrator. If the device is for a user, the activation code is emailed to the user.
5

Import the populated CSV file.

6

Click Submit.

You're presented with a status update as devices become activated.

 

Multiplatform devices must be running a firmware load of 11.2.3MSR1 or later for users to enter the activation code on their device. For information about how to upgrade phone firmware, see this article.

You can add, remove, reboot, check activation, or create a new activation code for the devices that are assigned to users within your organization. This can be helpful to view and manage devices in the users screen, when needed.

1

From the customer view in https://admin.webex.com, go to Users.

2

Select the user to modify and scroll down to Devices.

3

To add a device to this user, click Add Device.


 
If the user is already assigned a device, and you want to add another device, click the icon next to Devices and click Add Device.
4

To modify an existing device, select the device name.

Here you can view and edit device settings, delete the device, reboot the device, or create a new activation code for the device, if applicable. For more information about configuring phone settings, see Configure and Update Phone Settings.

5

If the device added to the user is Webex Aware, then the Webex Aware option is displayed under the devices as shown in the diagram. Webex Aware indicates that the device has onboarded to the Webex platform and has access to Webex Features supported by the phone.

6

Click Actions to manage the device. Actions help to apply configuration changes or update firmware for the MPP devices.

The Actions tab has these options for a Webex Aware-enabled device:
  • Apply Changes-issues request to the phone to download and apply changes to the configuration.
  • Reboot-issues request to force reboot the device and download the current configuration.
  • Report Problem-issues request to the device to generate and upload a PRT to the cloud.
  • Delete- deletes a device that is listed for the user.

Devices can be added and managed directly from a workspace profile. Workspace devices can include ATA devices, like fax machines. You can also set up a workspace device as a Hoteling Host. For more information about hoteling, see Hoteling in Cisco Webex Control Hub.

1

From the customer view in https://admin.webex.com, go to Workspaces.

2

Select the workspace to modify and go to the Devices tile.

3

To add a device, click Add Device.

4

To modify an existing device, select the device name.

Here you can view and edit device settings, delete the device, reboot the device, and enable the device to be used as a Hoteling Host. For more information about configuring phone settings, see Configure and Update Phone Settings.

5

If the device added to the workspace is Webex Aware, then the Webex Aware option is displayed under the devices as shown in the diagram. Webex Aware indicates that the device has onboarded to the Webex platform and has access to Webex features that are supported by the phone.

6

Click Actions to manage the device. Actions help to apply configuration changes or update firmware for the MPP devices.

The Actions tab has these options for a Webex Aware-enabled device:
  • Apply Changes-issues request to the phone to download and apply changes to the configuration.
  • Reboot-issues request to force reboot the device and download the current configuration.
  • Report Problem-issues request to the device to generate and upload a PRT to the cloud.
  • Delete- deletes a device that is listed for the user.

You can add lines to a user's primary device and reorder how the lines appear. This is also referred to as shared line appearance, which allows users to receive and place calls to and from another user's extension, using their own phone. An example of this is an executive assistant who wants to be able to make and receive calls from the boss's line. Shared line appearances can also be another instance of the primary user's line.

The maximum configuration limit is 35 devices for each user phone number, including desktop or mobile app uses by the user. Additional lines can be added to a workspace phone, but a workspace phone cannot be added as a shared line.


Speed dials that have been added by a user to their MPP phone are not visible in Control Hub and can be overwritten if a shared line is configured.

1

From the customer view in https://admin.webex.com, go to Users or Workspaces (depending on where the device to modify is assigned).

2

Select the user or workspace to modify and scroll to Devices.

3

Select the device where you would like to add or modify the shared lines and scroll to Phone Users and Settings.

The users and places that appear on this phone are listed in order of appearance.

4

To add or remove users or places from this phone, select Configure Lines.

5

To remove a line, click the icon.


 
The primary user on line 1 cannot be removed.
6

To add a shared line appearance, click the icon.


 
Add the lines in the order in which you want them to appear. To reorder the line appearance, delete and add to the list in the order you want them to appear.
7

Enter the name or phone number and select from the options that appear and click Save.

You can configure the ports on an Analog Telephone Adaptor (ATA) device assigned to a user in Control Hub. Currently, the two configurations for ATA devices available are for devices with 2 ports and devices with 24 ports.

1

From the customer view in https://admin.webex.com, go to Users.

2

Select the user to modify and scroll to Devices.

3

Select the device where you would like to add or modify.

4

Under Users on this Device, click Configure Ports.

5

To add a shared port configuration, click the icon.

6

Enter the name or phone number and select from the options that appear and then click Save.


 
Only workspaces without devices appear in the lookup.
7

If the device requires T.38 fax compression, check the box in the T.38 column or override user-level compression options, and then click Save.


 
A workspace can have an ATA. This is useful for fax machines.

You can add phone numbers to desk and room devices in your customer organization at any time, whether you're in the middle of a trial or have been converted to a paid subscription.


We've increased the number of telephone numbers you can add in Control Hub from 250 to 1000.

1

From the customer view in https://admin.webex.com, go to Services > Calling > Numbers then click Add Numbers.

2

Specify the Location and Number Type. If you're porting numbers over, enter both your current and new billing numbers.

3

Then, click Save.

You can see a list of PSTN numbers that your organization has ordered. With this information you can see unused numbers that are available, and the numbers that have been ordered that will soon become available.

From the customer view in https://admin.webex.com, go to Services > Calling > PSTN Orders.

When you connect accessories (Headsets/KEMs) to an MPP device, they appear as an inventory item under the Devices tab in Control Hub. From the Control Hub Devices inventory you can find out the accessory model, the status, and to whom the accessory belongs. When you select an accessory, additional information can be obtained, such as the accessory serial number and current software version. The accessory status field is reported as "online" as long as the accessory is connected to MPP. An MPP-connected headset will automatically upgrade its software with the latest version available from Device Management.

Want to see how it's done? Watch this video demonstration on how to view your accessories in Control Hub.
Table 1. Compatible Headsets

Phone Model

Cisco Headset 520 Series

Cisco Headset 530 Series

Cisco Headset 560 Series

Cisco Headset 730 Series

Cisco IP Phone 8811/8841/8845

RJ9 & RJ11

Cisco IP Phone 8851/8861/8865

USB

USB

USB

RJ9 & RJ11

Cisco IP Phone 7811/7821/7841/7861

Cisco IP Phone 6821/6841/6851/6861

Cisco IP Phone 6871

USB

USB

USB

Cisco IP Conference Phone 7832/8832

Table 2. Compatible Key Expansion Modules

Phone Model

KEM

Cisco IP Phone 8811/8841/8845

Cisco IP Phone 8851/8861/8865

BEKEM

CP-8800-A-KEM

CP-8800-V-KEM

Cisco IP Phone 7811/7821/7841/7861

Cisco IP Phone 6821/6841/6861/6871

Cisco IP Phone 6851

CP-68KEM-3PCC

Cisco IP Conference Phone 7832/8832

Feb 19, 2022
Adoption Trends and Usage Reports

You have a number of reports at your fingertips that can help you assess how Webex Calling services are being used, how often they're being used. You can also get a quick visual of the media quality for your location.

View Calling Reports

You can use the Analytics page in Control Hub to gain insight into how people are using Webex Calling and the Webex app (engagement), and the quaility of their call media experience. To access Webex Calling analytics, sign in to Control Hub, then go to Analytics and select the Calling tab.

1

For detailed call history reports, sign in to Control Hub, then go to Analytics and then select Detailed Call History.

You're automatically brought to the Calling Admin Portal, where you can analyze and evaluate call usage. For information about the reports available for specific calling features, see Calling Admin Portal - Reports. For information about call activity, see Calling Admin Portal - Analytics. For information about calls using Dedicated Instance, see Dedicated Instance Analytics.

2

To access media quality data, sign in to Control Hub, then go to Analytics and then select Calling.

May 6, 2022
Port Reference Information

Here is a list of the addresses, ports, and protocols used for connecting your phones, the Webex app, and gateways to Cisco Webex Calling. This article is for network administrators, particularly firewall and proxy security administrators who want to use Webex Calling services within their organization.

A correctly configured firewall is essential for a successful calling deployment. We require ports for signaling, media, network connectivity, and local gateway because Webex Calling is a global service. We recommend that you leave all the ports listed in the table open.

Not all firewall configurations need ports to be open but if you're running inside-to-outside rules, you must open ports to allow the protocols required for service out. As long as you deploy NAT, define reasonable binding periods, and avoid manipulating SIP on the NAT device, you shouldn't must open ports inbound on the firewall.


If a router or firewall is SIP Aware, meaning it has SIP Application Layer Gateway (ALG) or something similar enabled, we recommend that you turn off this functionality to maintain correct operation of service. See the relevant manufacturer's documentation for information about how to disable SIP ALG on specific devices.

For details on network requirements for Webex Meetings and Messaging, see Network Requirements for Webex Services.

Webex Calling Traffic Through Firewall

Most customers deploy an internet firewall, or internet proxy and firewall, to restrict and control the HTTP-based traffic that leaves and enters their network. The Webex Calling endpoints don’t support https proxy, except for soft clients, which support the following proxy environments and the corresponding authentication methods:

  1. Manual Proxy Configuration

    • No Authentication

    • Basic

    • NTLM

    • Negotiate

  2. WPAD Proxy Configuration

    • No authentication

    • Basic

  3. PAC Proxy Configuration

    • No Authentication

    • Basic

    • NTLM

    • Negotiate

Follow the firewall guidance to enable access to Webex Calling services from your network.

Firewall Configuration

If your firewall supports URL filtering, configure the firewall to allow the Webex Calling destination URLs listed. See the Domains and URLs for Webex Calling Services table for detials.

If you are using a firewall that doesn’t support URL/domain filtering, then configure the firewall to filter traffic using IP address ranges and ports as listed in the IP Addresses and Ports for Webex Calling Services.

IP Addresses and Ports for Webex Calling Services

The following table describes ports and protocols that must be opened on your firewall to allows cloud registered Webex apps, and devices to communicate with Webex Calling cloud signalling and media services.

IP Subnets for Webex Calling Services

23.89.1.128/25

23.89.33.0/24

23.89.40.0/25

23.89.76.128/25

52.26.82.54/24

85.119.56.0/23

128.177.14.0/24

128.177.36.0/24

135.84.168.0/21

139.177.64.0/21

139.177.72.0/23

150.253.209.128/25

170.72.0.128/25

170.72.17.128/25

170.72.29.0/24

170.72.82.0/25

185.115.196.0/22

199.19.196.0/23

199.19.199.0/24

199.59.64.0/21

Connection purpose

Source addresses

Source ports

Protocol

Destination addresses

Destination ports

Notes

Call signaling to Webex Calling (SIP TLS)

Local Gateway external (NIC) 8000-65535

TCP

Refer to IP Subnets for Webex Calling Services.

8934

These IPs/ports are needed for outbound SIP-TLS call signalling from Local Gateways, Devices, and Applications (Source) to Webex Calling Cloud (Destination).

Devices

5060-5080

Applications

Ephemeral (OS dependent)

Call media to Webex Calling (STUN,SRTP)

Local Gateway external NIC

8000-48198

UDP

Refer to IP Subnets for Webex Calling Services.

5004,19560-65535

These IPs/ports are needed for outbound SRTP call media from Local Gateways, Devices, and Applications (Source) to Webex Calling Cloud (Destination).

Devices

19560-19660

Applications

Ephemeral

Call signaling to PSTN gateway (SIP TLS) Local Gateway internal NIC 8000-65535 TCP Your ITSP PSTN GW or Unified CM Depends on PSTN option (for example, typically 5060 or 5061 for Unified CM)
Call media to PSTN gateway (SRTP) Local Gateway internal NIC

8000-48198

UDP Your ITSP PSTN GW or Unified CM Depends on PSTN option (for example, typically 5060 or 5061 for Unified CM)

Call signaling to publicly addressed endpoints (SIP TLS)

Refer to IP Subnets for Webex Calling Services.

Ephemeral

TCP

Endpoint IP

8934

These IPs/ports are needed for inbound SIP-TLS call signalling from Webex Calling Cloud (Source) to publicly addressed end points (Destination).

Device configuration and firmware management (Cisco devices)

Webex Calling devices

Ephemeral

TCP

3.20.185.219

3.130.87.169

3.134.166.179

443,6970

*These IPs belong to cloudupgrader.webex.com.

You need to enable cloudupgrader.webex.com and the 443, 6970 ports only when migrating from Enterprise phones (Cisco Unified CM) to Webex Calling. Go to upgrade.cisco.com for more information.

3.20.118.133

3.20.228.133

3.23.144.213

3.130.125.44

3.132.162.62

3.140.117.199

18.232.241.58

35.168.211.203

50.16.236.139

52.45.157.48

54.145.130.71

54.156.13.25

80,443

*These IPs belong to activation.webex.com.

These IPs are needed for secure onboarding of devices (MPP phones) via 16 digit activation code (GDS).

72.163.10.96/27

72.163.15.64/26

72.163.15.128/26

72.163.24.0/23

173.36.127.0/26

173.36.127.128/26

173.37.26.0/23

173.37.149.96/27

192.133.220.0/26

192.133.220.64/26

80,443

These IPs belong to activate.cisco.com.

This domain is used for CDA / EDOS - MAC address based provisioning. Used by devices (MPP phones, ATAs, and SPA ATAs) with newer firmware.

When a phone connects to a network for the first time or after a factory reset, and there are no DHCP options set up, it contacts a device activation server for zero touch provisioning. New phones use "activate.cisco.com" instead of "webapps.cisco.com" for provisioning. Phones with firmware release earlier than 11.2(1) continues to use "webapps.cisco.com". We recommend that you allow both the domain names through your firewall.

72.163.10.128/25

173.37.146.128/25

80,443

These IPs belong to webapps.cisco.com.

This domain is used for CDA / EDOS - MAC address based provisioning. Used by devices (MPP phones, ATAs, and SPA ATAs) with older firmware.

When a phone connects to a network for the first time or after a factory reset, and there are no DHCP options set up, it contacts a device activation server for zero touch provisioning. New phones use "activate.cisco.com" instead of "webapps.cisco.com" for provisioning. Phones with firmware release earlier than 11.2(1) continues to use "webapps.cisco.com". We recommend that you allow both the domain names through your firewall.

Refer to IP Subnets for Webex Calling Services.

80,443

These IPs are needed for Device configuration and firmware management for Webex Calling.

Device time synchronization (NTP)

Webex Calling devices

51494

UDP

Refer to IP Subnets for Webex Calling Services.

123

These IP addresses are needed for Time Synchronization for Devices (MPP phones, ATAs, and SPA ATAs)

Device name resolution

Webex Calling devices

Ephemeral

UDP and TCP

Host-defined

53

Application configuration

Webex Calling applications

Ephemeral

TCP

62.109.192.0/18

64.68.96.0/19

150.253.128.0/17

207.182.160.0/19

80, 443

These IPs belong to Webex Idbroker Authentication Services and used by clients, i.e. Webex Applications.

Refer to IP Subnets for Webex Calling Services.

80, 443, 8443

These IPs belong to Webex Calling application configuration services and used by clients, i.e.Webex Applications.

Application time synchronization

Webex Calling applications

123

UDP

Host-defined

123

Application name resolution

Webex Calling applications

Ephemeral

UDP and TCP

Host-defined

53

CScan

Webex Calling applications

Ephemeral

UDP and TCP

Refer to IP Subnets for Webex Calling Services.

8934 and 80, 443, 19569-19760

These IPs are used by CScan services used by clients, i.e.Webex Applications. Go to cscan.webex.com for more information.

† CUBE media port range is configurable with rtp-port range.

*These IP addresses/ranges are not owned by Cisco and are subject to change periodically. If you are using a firewall, we recommend to allow the urls listed.

Domains and URLs for Webex Calling Services

Domain / URL

Description

Webex apps and devices using these domains / URLs

Cisco Webex Services

*.webexcallingpbx.com

Webex authorization micro-services for cross-launch from Control Hub to Calling Admin Portal.

Control Hub

*.webexcalling.com.au

Webex Calling services in Australia.

All

*.webexcalling.eu

Webex Calling services in Europe.

All

*.webexcallingpbx.net

Calling client configuration and management services.

Webex Apps

*.cisco.com

When a phone connects to a network for the first time or after a factory reset, if there are no DHCP options set up, it contacts a device activation server for zero touch provisioning. New phones use activate.cisco.com and phones with firmware release prior to 11.2(1), continue to use webapps.cisco.com for provisioning.

MPP Phones, Control Hub

*.ucmgmt.cisco.com

Webex Calling services

Control Hub

*.webex.com

Webex Core Services for Calling, Meeting, and Messaging like Authentication, etc.

All

*.wbx2.com and *.ciscospark.com

Webex micro-services, like Software upgrade service.

All

Additional Webex-Related Services (Third-Party Domains)

*.appdynamics.com

*.eum-appdynamics.com

Performance tracking, error and crash capture, session metrics.

Control Hub

*.huron-dev.com

Webex Calling micro services like toggle services, phone number ordering, and assignment services.

Control Hub

*.sipflash.com

Device management services (mostly for US).

Webex Apps

*.walkme.com *.walkmeusercontent.com

Webex user guidance client. Provides onboarding and usage tours for new users.

For more information about WalkMe, click here.

Webex Apps

If your network firewall supports domain allow lists for http(s) traffic, like *.webex.com, it is highly recommended to allow all of these domains.

Webex Meetings/Messaging - Network Requirements

If you are deploying Webex Calling with Webex Meetings and Messaging services, the network requirements for the Webex Meetings and Messaging services can be found in Network Requirements for Webex Services.

Document Revision History

Date

We've made the following changes to this article

May 6, 2022

Added the IP subnet 52.26.82.54/24 for Webex Calling service

April 7, 2022

Updated the Local Gateway internal and external UDP port range to 8000-48198

April 5, 2022

Added the following IP subnets for Webex Calling service:

  • 23.89.40.0/25

  • 23.89.1.128/25

March 29, 2022

Added the following IP subnets for Webex Calling service:

  • 23.89.33.0/24

  • 150.253.209.128/25

September 20, 2021

Added 4 new IP subnets for Webex Calling service:

  • 23.89.76.128/25

  • 170.72.29.0/24

  • 170.72.17.128/25

  • 170.72.0.128/25

April 2, 2021

Added *.ciscospark.com under Domains and URLs for Webex Calling Services to support Webex Calling use cases in Webex app.

March 25, 2021

Added 6 new IP ranges for activate.cisco.com, which will come in effect starting May 8, 2021.

  • 72.163.15.64/26

  • 72.163.15.128/26

  • 173.36.127.0/26

  • 173.36.127.128/26

  • 192.133.220.0/26

  • 192.133.220.64/26

March 4, 2021

Replaced Webex Calling discrete IPs and smaller IP ranges with simplified ranges in a separate table for ease of understanding for firewall configuration.

February 26, 2021

Added 5004 as destination port for Call media to Webex Calling (STUN,SRTP) to support Interactive Connectivity Establishment (ICE) that will be available in Webex Calling in April 2021.

February 22, 2021

Domains and URLs are now listed within a separate table.

IP Addresses and Ports table are adjusted to group IP addresses for the same services together.

Notes column added to the IP Addresses and Ports table to better understand the needs.

The following IP addresses were moved to simplified ranges for device configuration and firmware management (Cisco devices):

activate.cisco.com

  • 72.163.10.125 -> 72.163.10.96/27

  • 173.37.149.125 -> 173.37.149.96/27

webapps.cisco.com

  • 173.37.146.134 -> 173.37.146.128/25

  • 72.163.10.134 -> 72.163.10.128/25

The following IP addresses were added for Application Configuration because Cisco Webex client is being pointed to a newer DNS SRV in Australia in March 2021.

  • 199.59.64.237

  • 199.59.67.237

January 21, 2021

We have added the following IP addresses to device configuration and firmware management (Cisco devices):

  • 3.134.166.179

  • 50.16.236.139

  • 54.145.130.71

  • 72.163.10.125

  • 72.163.24.0/23

  • 173.37.26.0/23

  • 173.37.146.134

We have removed the following IP addresses from device configuration and firmware management (Cisco devices):

  • 35.172.26.181

  • 52.86.172.220

  • 52.203.31.41

We have added the following IP addresses to application configuration:

  • 62.109.192.0/19

  • 64.68.96.0/19

  • 207.182.160.0/19

  • 150.253.128.0/17

We have removed the following IP addresses from application configuration:

  • 64.68.99.6

  • 64.68.100.6

We have removed the following port numbers from application configuration:

  • 1081, 2208, 5222, 5280-5281, 52644-52645

We have added the following domains to application configuration:

  • idbroker-b-us.webex.com

  • idbroker-eu.webex.com

  • ty6-wxt-jp.bcld.webex.com

  • os1-wxt-jp.bcld.webex.com

December 23, 2020

Added new Application Configuration IP addresses to the port reference images.

December 22, 2020

Updated the Application Configuration row in the tables to include the following IP addresses: 135.84.171.154 and 135.84.172.154.

Hid the network diagrams until these IP addresses can be added there as well.

December 11, 2020

Updated the Device configuration and firmware management (Cisco devices) and the Application configuration rows for the supported Canadian domains.

October 16, 2020

Updated the call signaling and media entries with the following IP addresses:

  • 139.177.64.0/24

  • 139.177.65.0/24

  • 139.177.66.0/24

  • 139.177.67.0/24

  • 139.177.68.0/24

  • 139.177.69.0/24

  • 139.177.70.0/24

  • 139.177.71.0/24

  • 139.177.72.0/24

  • 139.177.73.0/24

September 23, 2020

Under CScan, replaced 199.59.64.156 with 199.59.64.197.

August 14, 2020

Added more IP addresses to support the introduction of data centers in Canada:

Call signaling to Webex Calling (SIP TLS)—135.84.173.0/25,135.84.174.0/25, 199.19.197.0/24, 199.19.199.0/24

August 12, 2020

Added more IP addresses to support the introduction of data centers in Canada:

  • Call media to Webex Calling (SRTP)—135.84.173.0/25,135.84.174.0/25, 199.19.197.0/24, 199.19.199.0/24

  • Call signaling to publicly addressed endpoints (SIP TLS)—135.84.173.0/25,135.84.174.0/25, 199.19.197.0/24, 199.19.199.0/24

  • Device configuration and firmware management (Cisco devices)—135.84.173.155,135.84.174.155

  • Device time synchronization—135.84.173.152, 135.84.174.152

  • Application configuration—135.84.173.154,135.84.174.154

July 22, 2020

Added the following IP address to support the introduction of data centers in Canada: 135.84.173.146

June 9, 2020

We made the following changes to the CScan entry:

  • Corrected one of the IP addresses—changed 199.59.67.156 to 199.59.64.156

  • New features required new ports as well as UDP—19560-19760

March 11, 2020

We added the following domain and IP addresses to application configuration:

  • jp.bcld.webex.com—135.84.169.150

  • client-jp.bcld.webex.com

  • idbroker.webex.com—64.68.99.6, 64.68.100.6

We updated the following domains with additional IP addresses to device configuration and firmware management:

  • cisco.webexcalling.eu—85.119.56.198, 85.119.57.198

  • webapps.cisco.com—72.163.10.134

  • activation.webex.com—35.172.26.181, 52.86.172.220

  • cloudupgrader.webex.com—3.130.87.169, 3.20.185.219

February 27, 2020

We added the following domain and ports to device configuration and firmware management:

cloudupgrader.webex.com—443, 6970

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Webex Calling Configuration Workflow