300-830 Questions and Answers

WebRTC voice in Webex Contact Center is exposed to agents through the Desktop telephony option. Cisco's desktop profile documentation lists voice channel options, including Agent DN, Extension, and Desktop. The Desktop option allows the agent to sign in through the browser and use the Agent Desktop as the voice device rather than entering an external directory number or extension. Cisco's voice channel setup documentation also describes WebRTC-capable agents selecting Desktop under their telephony option so calls are routed to the Agent Desktop device. Browser is not a desktop profile voice option; it describes the client environment. Extension and Agent DN are traditional station credentials and would make the agent use a phone or configured number instead of browser-based WebRTC. Therefore, to enable agents to make and receive calls using WebRTC, the administrator must configure the Desktop option in the desktop profile. Reference: Cisco Help, Manage desktop profiles; Set up voice channels for Webex Contact Center.

Option B is the only system profile that aligns with the Webex Contact Center Desktop WebRTC requirements shown in the exhibit. Cisco's desktop system requirements identify supported browsers and agent device models for calling, including Webex Calling desktop/app computer audio and WebRTC handling from Agent Desktop. A practical WebRTC station must have sufficient bandwidth, a supported operating system and browser, and browser settings that do not block required desktop behavior. Option A is weak because the 384 Kbps bandwidth value is below the practical WebRTC threshold shown by the comparison choices. Option C explicitly disables JavaScript, which prevents the browser-based desktop and WebRTC interaction logic from functioning properly. Option D uses a virtual desktop infrastructure condition, which is not the straightforward supported physical workstation profile presented by the question. Option B provides Windows 11, Chrome, physical host type, adequate bandwidth, a medium security level, and disabled pop-up blocking, making it the compliant choice. Reference: Cisco Help, System requirements for Webex Contact Center Desktop; Set up voice channels for Webex Contact Center.

For OAuth 2.0 custom connector authentication in Webex Contact Center, Cisco documents client credentials using either a client secret or a private key/certificate method. The custom connector configuration supports Client credentials - client secret, where the administrator enters a client ID and client secret, and Client credentials - private key, where the private key corresponds to the certificate material used by the application. Those two choices are valid grant/authentication methods for service-to-service integration because a backend connector needs machine-to-machine authorization rather than a user authorization code flow. Client credentials - token is not a grant type; the token is the output of a successful OAuth exchange. Client credentials - certificate is too imprecise in the option set because Cisco's configuration wording identifies private key/certificate material. Server credentials - certificate is not a documented OAuth grant type for the Webex Contact Center custom connector. Reference: Cisco Help, Configure custom connector for Webex Contact Center; Webex Developer, Integrations and Authorization.

A Scheduled Callback activity must know when the callback should occur and the timezone in which that schedule should be interpreted. Cisco's callback scheduling API and Webex Connect scheduling behavior use normalized date/time inputs so a callback is created at the expected customer time. The date format in the answer set is YYYY-MM-DD, which is the standard year-month-day format used by modern APIs and scheduling fields. The timezone is required because the same date and time can represent different UTC instants depending on the caller's or business's region. Customer name and phone ANI may be useful data in a broader callback workflow, but they are not the two key values asked here for configuring the Scheduled Callback node successfully. The YYYY-DD-MM format is not the accepted ordering and would create ambiguous or invalid schedules. The correct pair is therefore schedule date in YYYY-MM-DD format and schedule timezone. Reference: Webex Developer, Callback Schedule API; Webex Connect Help, Event Scheduler and scheduled flow activities.

Cisco protects Webex Contact Center APIs with rate limiting per client ID so that one application cannot exhaust shared platform capacity. The current default limit represented in this question is 30 requests per second and 1800 requests per minute for a given client ID. Those two values are the same rate expressed at different time scales, and both are used by administrators and developers when sizing integration retry logic. The Webex developer guidance also stresses that clients must handle HTTP 429 responses and Retry-After behavior rather than assuming unlimited API throughput. The 300 requests per second and 1000 requests per minute options do not match the stated current Contact Center API limits. The 300 requests per minute figure appears in broader Webex REST API guidance for many APIs, but the question is asking specifically about the Webex Contact Center API limits in this context. Correct integrations should therefore throttle to 30 RPS and 1800 RPM unless Cisco grants or documents a different tenant-specific limit. Reference: Webex Developer, Webex Contact Center APIs; Webex Developer blog, Rate Limiting and the Webex API.

Inbound Webex Contact Center voice routing needs an entry path and an agent distribution target. The DNIS or support number must be associated with the entry point and flow so that an incoming PSTN call reaches the proper Flow Designer logic. Inside that flow, the contact must be queued to a queue that has eligible teams or agents. Cisco's queue documentation describes queues as holding areas where contacts wait until they are distributed, and it explains that non-skill queues use team assignments or agent assignments while skill queues use skill matching. The wording in the option set refers to DNIS entries in Flow Designer; operationally, DNIS is tied to the entry point/channel and flow selection, and the flow then uses a Queue Contact activity. Script Editor is a legacy/on-premises contact center construct, not the modern Webex Contact Center Flow Designer model. A softphone registration alone does not route calls, and a skills-only queue is not mandatory unless the design requires skills. Reference: Cisco Help, Set up voice channels for Webex Contact Center; Understand Routing and Queueing in Webex Contact Center.

The Debug Console exhibit shows an SMTP failure for an outbound email transaction. In Webex Connect email asset configuration, the service provider settings must match the SMTP connection requirements, including the server address, port, authentication method, and security method. The failure pattern represented in this question points to connection-level SMTP mistakes: an incorrect SMTP server port and an incorrect SMTP security method. Those two errors commonly prevent the platform from establishing or negotiating the SMTP session at all. An incorrect username would usually produce an authentication failure after the connection is made, and an incorrect server address would present a DNS or host connection error rather than the specific pairing shown. The asset not being registered to Webex Engage affects channel registration and conversation use cases, but the exhibit specifically identifies SMTP failure in the email channel. Therefore the port and security method are the two configuration mistakes. Reference: Webex Connect Help, Email - WXCC; Webex Connect Help, Debug Console.

In Flow Designer, a Play Message activity must have a valid prompt source. Cisco documents that Flow Designer can play prerecorded audio files or text-to-speech prompts and that TTS content must be configured correctly for Cisco Cloud Text-to-Speech. The exhibit indicates a simple flow that should play a message and then disconnect. If callers hear nothing and the call immediately drops, the most direct cause is that the message node has no valid TTS message configured or the TTS value is invalid. A codec mismatch would typically affect media negotiation more broadly, not only the absence of configured prompt text. A missing audio file would apply if the node used an audio file, but the answer choice points to TTS configuration. Cisco Cloud Text-to-Speech being unavailable would be a service-side outage, which is less consistent with a designer-created flow issue. The local misconfiguration is the invalid or missing TTS message. Reference: Cisco Help, Build and manage flows with Flow Designer; Text-to-Speech in Webex Contact Center.

The Webex App inside Agent Desktop is controlled through the desktop layout JSON, not merely by installing the Webex App separately. Cisco documentation for supervisor and desktop features points administrators to the webexConfigured setting in the Webex Contact Center desktop layout. The layout also uses the AgentX Webex component identifier to render the embedded Webex application area. That is why the two required parameter combinations are the desktopChatApp object with webexConfigured set to true and the component value agentx-webex. The value webex-app is not the documented component name for this layout object. A generic webexAppEnabled Boolean or webexApp enabled object may look plausible, but it is not the required JSON pattern in the Cisco desktop layout schema. The result is practical: if either the correct component or the webexConfigured flag is absent, the embedded Webex App will not display in Agent Desktop for agent-to-agent, supervisor, or SME collaboration. Reference: Cisco Help, Create custom desktop layout; Supervise your agents and teams.

SMS-based one-time passcode authentication requires generating the passcode and sending it to the user over SMS. Webex Connect includes a Generate OTP node for creating the random passcode and an SMS node for delivering the generated value to the customer's mobile number. Those two nodes form the minimum authentication workflow. A Profile node can retrieve or update customer profile information but does not generate or deliver the passcode. A Data Parser node can parse structured responses, which is useful after API calls, but it is not required to create and transmit an OTP. An Evaluate node can compare the customer's submitted passcode to the stored/generated value later in the flow, but the question asks which two nodes are needed to implement authentication via SMS using a randomly generated passcode. The foundational nodes are Generate OTP and SMS. Reference: Webex Connect Help, Generate OTP node; SMS Node; Webex Connect flow builder documentation.

Control Hub normalizes Webex Calling and Webex Contact Center numbers into E.164 format, and Cisco's voice channel setup guidance explains that numbers routed from Cloud Connected PSTN or Local Gateway are provisioned in +E.164 format for Control Hub. The fully qualified +1-450-783-2223 format is therefore valid. Cisco interfaces also accept common national display formatting for number entry in many workflows, such as parentheses around the area code, which corresponds to (450) 783-2223 in the option set. The bracketed version, hash-prefixed value, and dotted format are not acceptable telephony number formats for configuring a Webex Contact Center entry point number. The key is that the number must represent a valid PSTN telephone number that Control Hub can store and map to the entry point. The +E.164 version is the canonical final representation; the national style is an accepted entry/display form. Reference: Cisco Help, Set up voice channels for Webex Contact Center; Set up voice settings for Webex Contact Center.

Bridged Transfer is the correct node because the requirement is not simply to send the caller away; the administrator must connect the caller to an external IVR while keeping Webex Contact Center call control and then resume the original flow. Cisco's Flow Designer documentation distinguishes Bridged Transfer from Blind Transfer. A blind transfer is terminal: once the call is transferred to the external number, the flow ends and the original flow logic no longer controls the interaction. A Bridged Transfer, by contrast, is designed for scenarios where the caller is temporarily connected to an external destination and then returned to the flow after the external interaction completes. An HTTP Request node is only for data exchange with an external service and cannot bridge media to an IVR. A GoTo node is internal flow navigation and cannot connect the caller to an external IVR. The call-control behavior in the scenario is therefore specifically the Bridged Transfer use case. Reference: Cisco Help, Build and manage flows with Flow Designer; Understand Routing and Queueing in Webex Contact Center.

Real-time transcripts require the media stream to start before the conversation is sent to the queue or agent. Cisco's AI Assistant real-time transcription feature depends on capturing and streaming the call audio so the platform can produce transcript data for the agent and downstream AI features. If the Media Stream node is placed after the Agent Answered event, the transcript starts too late and misses the early customer interaction and queue/agent transition context. A Condition node can be useful for branching but does not enable transcription by itself. Configuring a Scripted AI Agent is unrelated to enabling AI Assistant real-time transcripts for a live-agent call. The correct flow action is to add the Media Stream node before the call reaches the queue or agent so the media is available for transcription at the right point in the call lifecycle. Reference: Cisco Help, Cisco AI Assistant for Webex Contact Center: Administrator's guide to configuring Real-Time Assist; AI Assistant reports in Analyzer.