{"id":23439,"date":"2026-09-28T06:40:18","date_gmt":"2026-09-28T06:40:18","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=23439"},"modified":"2026-09-28T06:40:18","modified_gmt":"2026-09-28T06:40:18","slug":"cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-19-q361-380","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-19-q361-380\/","title":{"rendered":"Cisco CCNP Collaboration 300-815 Practice Test Questions and Exam Dumps Part 19 Q361-380"},"content":{"rendered":"

View Full <\/b>Cisco CCNP Collaboration 300-815 Exam Dumps<\/b><\/a> and Practice Test Dumps<\/b><\/h1>\n

 <\/p>\n

Question 361. A Cisco Unified Communications Manager administrator needs to route calls based on the calling party’s URI domain rather than the traditional numeric dialed digits. Which feature is most appropriate for this requirement?<\/b><\/p>\n

    \n
  1. <\/b> Route Group<\/span><\/li>\n
  2. <\/b> URI dialing<\/span><\/li>\n
  3. <\/b> Translation Pattern<\/span><\/li>\n
  4. <\/b> Local Route Group<\/span><\/li>\n<\/ol>\n

    Correct Answer: 2. URI dialing<\/b><\/p>\n

    Explanation :-<\/b><\/p>\n

    URI dialing allows Cisco Unified Communications Manager to route calls using SIP-style identifiers such as user@domain instead of relying exclusively on numeric directory numbers. This is particularly useful in SIP deployments where users or organizations are identified by domains. URI routing can be integrated with routing constructs to determine how calls should be handled. Traditional route patterns and translation patterns are primarily designed around numeric digit strings, while local route groups provide gateway selection based on device or location configuration. Therefore, when the routing requirement is explicitly based on a SIP URI, URI dialing and associated URI routing capabilities are the appropriate approach.<\/span><\/p>\n

    Question 362. A SIP trunk is configured between Cisco Unified Communications Manager and a third-party SIP system. Calls connect successfully, but the third-party system requires a specific SIP header to be inserted into every outbound INVITE. Which Cisco Unified Communications Manager feature should be considered?<\/b><\/p>\n

      \n
    1. <\/b> Route Filter<\/span><\/li>\n
    2. <\/b> Location configuration<\/span><\/li>\n
    3. <\/b> Media Resource Group<\/span><\/li>\n
    4. <\/b> SIP Profile<\/span><\/li>\n<\/ol>\n

      Correct Answer: 4. SIP Profile<\/b><\/p>\n

      Explanation :-<\/b><\/p>\n

      A SIP Profile can modify certain SIP signaling behaviors and headers used by Cisco Unified Communications Manager when communicating with SIP devices and trunks. This makes SIP Profiles useful for interoperability requirements where a third-party system expects specific signaling behavior. A Route Filter determines which dialed numbers are permitted by a route pattern and does not modify SIP headers. Locations are associated with bandwidth-based call admission control, while Media Resource Groups control access to media resources such as transcoders or conferencing resources. Therefore, a SIP Profile is the appropriate feature when SIP signaling needs to be adjusted for interoperability.<\/span><\/p>\n

      Question 363. A Cisco Unified Communications Manager deployment has several gateways that can provide the same type of PSTN access. The administrator wants calls to use one gateway first and automatically use another gateway if the preferred gateway is unavailable. Which configuration provides this behavior?<\/b><\/p>\n

        \n
      1. <\/b> Route Group with ordered members<\/span><\/li>\n
      2. <\/b> Translation Pattern with a CSS<\/span><\/li>\n
      3. <\/b> Device Pool with multiple regions<\/span><\/li>\n
      4. <\/b> SIP Profile with multiple trunks<\/span><\/li>\n<\/ol>\n

        Correct Answer: 1. Route Group with ordered members<\/b><\/p>\n

        Explanation :-<\/b><\/p>\n

        A Route Group can contain multiple gateway or trunk members and can define the order in which those members are selected. When a call is routed through a Route List that contains the Route Group, Unified Communications Manager can attempt the first available member and use another member according to the configured selection behavior when necessary. Translation Patterns manipulate or route digits but do not themselves provide gateway member ordering. Device Pools control device configuration, and SIP Profiles influence signaling behavior. Therefore, an ordered Route Group is the appropriate mechanism for providing alternate gateway selection.<\/span><\/p>\n

        Question 364. A company uses Cisco Unified Communications Manager and wants to ensure that a user’s national-format telephone number is converted into a globally normalized E.164 number before routing. Which feature is designed for digit manipulation and normalization?<\/b><\/p>\n

          \n
        1. <\/b> Location<\/span><\/li>\n
        2. <\/b> Route Group<\/span><\/li>\n
        3. <\/b> Translation Pattern<\/span><\/li>\n
        4. <\/b> Media Resource Group<\/span><\/li>\n<\/ol>\n

          Correct Answer: 3. Translation Pattern<\/b><\/p>\n

          Explanation :-<\/b><\/p>\n

          Translation Patterns are commonly used in Cisco Unified Communications Manager to manipulate dialed digits before a call is routed. They can remove, add, or modify digits and are therefore useful for converting locally or nationally formatted numbers into a standardized representation such as E.164. Locations are associated with bandwidth management, Route Groups define gateway or trunk selection, and Media Resource Groups provide controlled access to media resources. Properly designed translation patterns can support globalization and normalization strategies while allowing routing logic to work consistently across different numbering formats.<\/span><\/p>\n

          Question 365. A Cisco CUBE router receives an inbound SIP INVITE, but the configured outbound dial-peer is not being selected. The administrator wants to determine exactly which outbound dial-peer CUBE chooses for the destination number. Which troubleshooting approach is most appropriate?<\/b><\/p>\n

            \n
          1. <\/b> Check only the codec configured on the dial-peer<\/span><\/li>\n
          2. <\/b> Review the dial-peer matching process using appropriate CUBE debug commands<\/span><\/li>\n
          3. <\/b> Disable SIP signaling and restart the router<\/span><\/li>\n
          4. <\/b> Change the SIP transport from UDP to TCP<\/span><\/li>\n<\/ol>\n

            Correct Answer: 2. Review the dial-peer matching process using appropriate CUBE debug commands<\/b><\/p>\n

            Explanation :-<\/b><\/p>\n

            CUBE selects dial-peers based on configured matching criteria, and troubleshooting an unexpected selection requires examining the actual matching process. Appropriate voice and SIP debugging can show how the router processes the call and which dial-peer is selected. Codec configuration, SIP transport, and router restarts do not directly reveal the dial-peer matching decision. Administrators should first verify the incoming and outgoing dial-peer criteria, destination patterns, URI matching where applicable, and the resulting selection. This approach helps identify overlapping or incorrect dial-peer configurations without making unrelated signaling changes.<\/span><\/p>\n

            Question 366. A SIP call is established, but the caller hears no audio while the called party can hear the caller. Packet captures show that RTP is being sent toward an address that is not reachable from the remote network. Which signaling information should the administrator inspect first?<\/b><\/p>\n

              \n
            1. <\/b> SIP response code only<\/span><\/li>\n
            2. <\/b> SIP Call-ID value only<\/span><\/li>\n
            3. <\/b> SDP media address and port information<\/span><\/li>\n
            4. <\/b> SIP From header display name<\/span><\/li>\n<\/ol>\n

              Correct Answer: 3. SDP media address and port information<\/b><\/p>\n

              Explanation :-<\/b><\/p>\n

              The Session Description Protocol, or SDP, carried within SIP signaling describes the media streams, including the media IP address and RTP port information. If one-way audio occurs because RTP is being sent toward an unreachable address, the administrator should inspect the SDP exchanged during call setup and subsequent offer\/answer transactions. SIP headers such as Call-ID and display-name information identify the signaling session but do not normally specify where RTP should be sent. Examining SDP helps determine whether NAT, incorrect advertised addresses, or media-path handling is causing the one-way-audio problem.<\/span><\/p>\n

              Question 367. A Cisco Unified Communications Manager administrator wants users in different geographic locations to receive the same extension number behavior when they move between phones configured for different locations. Which feature can associate location-specific information with a device’s current network location?<\/b><\/p>\n

                \n
              1. <\/b> Device Mobility<\/span><\/li>\n
              2. <\/b> Route Filter<\/span><\/li>\n
              3. <\/b> Route Pattern<\/span><\/li>\n
              4. <\/b> SIP Profile<\/span><\/li>\n<\/ol>\n

                Correct Answer: 1. Device Mobility<\/b><\/p>\n

                Explanation :-<\/b><\/p>\n

                Device Mobility allows Cisco Unified Communications Manager to apply location-specific settings when a supported device is used at a different network location. Mobility-related configuration can affect items such as device pool assignment and other settings associated with the user’s current location. This is different from Route Patterns and Route Filters, which are primarily concerned with call routing and digit matching. SIP Profiles control SIP signaling behavior. Device Mobility is therefore appropriate when the requirement involves recognizing that a device has moved and applying configuration associated with its current physical or network location.<\/span><\/p>\n

                Question 368. A SIP endpoint sends an INVITE and receives a provisional response containing SDP before the final call answer. The administrator wants to understand how the endpoint can process the media information before the call is fully established. Which concept describes this behavior?<\/b><\/p>\n

                  \n
                1. <\/b> Number normalization<\/span><\/li>\n
                2. <\/b> Early media<\/span><\/li>\n
                3. <\/b> Route group failover<\/span><\/li>\n
                4. <\/b> Calling-party transformation<\/span><\/li>\n<\/ol>\n

                  Correct Answer: 2. Early media<\/b><\/p>\n

                  Explanation :-<\/b><\/p>\n

                  Early media refers to media exchanged before the SIP session reaches its final successful response. A provisional response such as 183 Session Progress may include SDP describing the media path, allowing endpoints or network elements to establish RTP before the call is formally answered. Early media is commonly encountered with PSTN interworking, announcements, ringback, and other services that provide audio during call establishment. Number normalization and calling-party transformations affect signaling information, while Route Group failover concerns path selection. Therefore, the described behavior is associated with early media.<\/span><\/p>\n

                  Question 369. A Cisco Unified Communications Manager administrator needs to prevent a specific class of PSTN numbers from being routed through an otherwise valid route pattern. Which component can be used to restrict the digits that a route pattern is allowed to match?<\/b><\/p>\n

                    \n
                  1. <\/b> Route Group<\/span><\/li>\n
                  2. <\/b> Media Resource Group<\/span><\/li>\n
                  3. <\/b> Route Filter<\/span><\/li>\n
                  4. <\/b> Location<\/span><\/li>\n<\/ol>\n

                    Correct Answer: 3. Route Filter<\/b><\/p>\n

                    Explanation :-<\/b><\/p>\n

                    A Route Filter can be associated with a route pattern to further restrict which digit strings are considered valid matches. This provides additional control over routing without requiring separate route patterns for every possible variation. Route Groups determine which gateways or trunks are used after routing decisions are made. Media Resource Groups control access to resources such as transcoders and conference bridges, while Locations participate in call admission control. Therefore, when the requirement is specifically to restrict the digit patterns matched by a route pattern, a Route Filter is the appropriate component.<\/span><\/p>\n

                    Question 370. A company wants a SIP trunk to remain reachable even when there is little or no call traffic. The administrator wants Cisco Unified Communications Manager to periodically verify that the SIP peer is responding. Which mechanism is most relevant?<\/b><\/p>\n

                      \n
                    1. <\/b> Translation Pattern<\/span><\/li>\n
                    2. <\/b> Route Filter<\/span><\/li>\n
                    3. <\/b> Calling Search Space<\/span><\/li>\n
                    4. <\/b> SIP OPTIONS keepalive<\/span><\/li>\n<\/ol>\n

                      Correct Answer: 4. SIP OPTIONS keepalive<\/b><\/p>\n

                      Explanation :-<\/b><\/p>\n

                      SIP OPTIONS messages can be used to determine whether a SIP peer remains responsive even when no active calls are present. They provide a signaling-level reachability mechanism that can help identify unavailable SIP trunks or peers. Translation Patterns manipulate digits, Route Filters restrict route-pattern matching, and Calling Search Spaces control which partitions and destinations a device or line can access. Using SIP OPTIONS as a keepalive mechanism can therefore help an administrator monitor SIP peer availability and support more reliable call-routing behavior when the signaling connection needs to be continuously monitored.<\/span><\/p>\n

                      Question 371. A Cisco CUBE deployment must conceal internal IP addressing information from an external SIP service provider. Which CUBE capability is specifically intended to modify SIP signaling so internal topology information is not exposed?<\/b><\/p>\n

                        \n
                      1. <\/b> SIP topology hiding<\/span><\/li>\n
                      2. <\/b> Location-based CAC<\/span><\/li>\n
                      3. <\/b> Route filtering<\/span><\/li>\n
                      4. <\/b> Codec preference<\/span><\/li>\n<\/ol>\n

                        Correct Answer: 1. SIP topology hiding<\/b><\/p>\n

                        Explanation :-<\/b><\/p>\n

                        SIP topology hiding is used to prevent external SIP peers from learning unnecessary details about the internal network structure. A CUBE deployment can manipulate selected SIP signaling information so that internal addresses and topology-related details are concealed across the external signaling boundary. Location-based CAC controls available call bandwidth, route filtering controls routing matches, and codec preference influences media codec selection. Topology hiding is therefore the capability most directly associated with protecting internal SIP network information while maintaining interoperability with an external SIP service provider.<\/span><\/p>\n

                        Question 372. A Cisco Unified Communications Manager administrator notices that two SIP endpoints negotiate different codecs and the call cannot establish media correctly. Both endpoints support G.711, but one endpoint advertises only G.729 for the current call. What network resource may be required if the endpoints cannot negotiate a common codec?<\/b><\/p>\n

                          \n
                        1. <\/b> Route Filter<\/span><\/li>\n
                        2. <\/b> Transcoder<\/span><\/li>\n
                        3. <\/b> Route List<\/span><\/li>\n
                        4. <\/b> Translation Pattern<\/span><\/li>\n<\/ol>\n

                          Correct Answer: 2. Transcoder<\/b><\/p>\n

                          Explanation :-<\/b><\/p>\n

                          A transcoder converts media from one codec format to another when communicating endpoints do not share a compatible codec. In the described situation, one endpoint offers G.729 while the other requires or prefers G.711, creating a codec interoperability problem. A transcoder can convert between these formats so that each endpoint receives media in a codec it supports. Route Filters, Route Lists, and Translation Patterns affect call routing or digit manipulation rather than converting RTP media. The administrator should therefore verify whether an appropriate transcoder is available and accessible to the call.<\/span><\/p>\n

                          Question 373. A Cisco Unified Communications Manager administrator wants a user to have a single directory number but receive calls on different phones depending on the user’s current login location. Which feature is most directly associated with allowing the user to log into a phone and use their configured device profile?<\/b><\/p>\n

                            \n
                          1. <\/b> Call Admission Control<\/span><\/li>\n
                          2. <\/b> Local Route Group<\/span><\/li>\n
                          3. <\/b> Extension Mobility<\/span><\/li>\n
                          4. <\/b> Route Filter<\/span><\/li>\n<\/ol>\n

                            Correct Answer: 3. Extension Mobility<\/b><\/p>\n

                            Explanation :-<\/b><\/p>\n

                            Extension Mobility allows users to log into supported Cisco IP phones and use a configured device profile containing their personal settings, such as directory numbers and feature configuration. This is useful in environments where users move between workstations or share phones. Call Admission Control manages available network bandwidth, Local Route Groups influence gateway selection, and Route Filters restrict route-pattern matching. Extension Mobility therefore directly addresses the requirement for a user to authenticate to a phone and have their configured telephony profile applied to that device.<\/span><\/p>\n

                            Question 374. A Cisco Unified Communications Manager deployment uses several route patterns that overlap. The administrator wants to ensure that a more specific destination pattern is selected instead of a broader pattern when both could potentially match. Which principle should the administrator evaluate?<\/b><\/p>\n

                              \n
                            1. <\/b> Codec negotiation<\/span><\/li>\n
                            2. <\/b> Media resource allocation<\/span><\/li>\n
                            3. <\/b> SIP transport encryption<\/span><\/li>\n
                            4. <\/b> Route-pattern matching and precedence<\/span><\/li>\n<\/ol>\n

                              Correct Answer: 4. Route-pattern matching and precedence<\/b><\/p>\n

                              Explanation :-<\/b><\/p>\n

                              When multiple routing patterns can potentially match a dialed number, Cisco Unified Communications Manager uses its routing logic and pattern specificity to determine the applicable route. Administrators should carefully evaluate overlapping patterns, digit lengths, transformations, partitions, and routing configuration to understand which destination is selected. Codec negotiation and media-resource allocation occur at different stages of call processing and do not determine which route pattern matches the number. SIP transport encryption also does not establish route-pattern precedence. Properly designed routing patterns should minimize unintended overlap and make the expected selection clear.<\/span><\/p>\n

                              Question 375. A Cisco CUBE router must send calls to different SIP service providers based on the destination domain contained in the SIP URI. Which dial-peer matching method is particularly useful for this requirement?<\/b><\/p>\n

                                \n
                              1. <\/b> Destination URI matching<\/span><\/li>\n
                              2. <\/b> Codec-only matching<\/span><\/li>\n
                              3. <\/b> Voice-port matching<\/span><\/li>\n
                              4. <\/b> DSCP matching<\/span><\/li>\n<\/ol>\n

                                Correct Answer: 1. Destination URI matching<\/b><\/p>\n

                                Explanation :-<\/b><\/p>\n

                                Destination URI matching allows CUBE to select an appropriate dial-peer based on information contained in the destination SIP URI, including the domain. This is useful in SIP environments where routing decisions depend on domains rather than only numeric destination patterns. Codec configuration determines media preferences but is not normally the primary mechanism for selecting a destination based on SIP domain. Voice-port matching is associated with physical voice interfaces, and DSCP values are related to packet classification and QoS. Therefore, destination URI matching is the most appropriate mechanism for domain-based SIP routing on CUBE.<\/span><\/p>\n

                                Question 376. A SIP call is already established between two endpoints. One endpoint needs to change the media parameters during the existing session, such as adding or modifying a media stream. Which SIP mechanism is commonly used for this purpose?<\/b><\/p>\n

                                  \n
                                1. <\/b> REGISTER<\/span><\/li>\n
                                2. <\/b> OPTIONS<\/span><\/li>\n
                                3. <\/b> Re-INVITE<\/span><\/li>\n
                                4. <\/b> BYE<\/span><\/li>\n<\/ol>\n

                                  Correct Answer: 3. Re-INVITE<\/b><\/p>\n

                                  Explanation :-<\/b><\/p>\n

                                  A SIP re-INVITE can be sent within an established dialog to renegotiate session parameters. It can be used for changes such as modifying media characteristics, placing a call on hold, changing codec-related information, or adding media depending on the implementation and negotiated session behavior. REGISTER is used for endpoint registration, OPTIONS can be used for capability or reachability checks, and BYE terminates an established session. Therefore, when an established SIP call needs its session or media parameters renegotiated, a re-INVITE is a commonly used SIP mechanism.<\/span><\/p>\n

                                  Question 377. A Cisco Unified Communications Manager administrator wants to control whether a particular phone can access a specific partition containing PSTN route patterns. Which configuration relationship should be reviewed?<\/b><\/p>\n

                                    \n
                                  1. <\/b> Region and codec configuration<\/span><\/li>\n
                                  2. <\/b> Location and bandwidth configuration<\/span><\/li>\n
                                  3. <\/b> Media Resource Group and transcoder configuration<\/span><\/li>\n
                                  4. <\/b> Calling Search Space and partition assignment<\/span><\/li>\n<\/ol>\n

                                    Correct Answer: 4. Calling Search Space and partition assignment<\/b><\/p>\n

                                    Explanation :-<\/b><\/p>\n

                                    Calling Search Spaces and partitions work together to control which directory numbers, route patterns, translation patterns, and other destinations a device or line can access. A Calling Search Space contains an ordered list of partitions, and the device or line uses that search space when attempting to reach a destination. Region configuration primarily affects codec selection, while Locations are used for bandwidth-based call admission control. Media Resource Groups govern access to media resources. Therefore, access to a specific PSTN partition should be evaluated through the relevant Calling Search Space and partition configuration.<\/span><\/p>\n

                                    Question 378. A remote office loses WAN connectivity to Cisco Unified Communications Manager. Phones at the site are configured for SRST, but the administrator discovers that calls are not being handled using the expected local dial plan. Which configuration should be checked first?<\/b><\/p>\n

                                      \n
                                    1. <\/b> SIP Profile on the PSTN gateway only<\/span><\/li>\n
                                    2. <\/b> SRST router configuration and local dial-peer\/dial-plan settings<\/span><\/li>\n
                                    3. <\/b> Media Resource Group List on the central cluster only<\/span><\/li>\n
                                    4. <\/b> Route Filter associated with a central route pattern<\/span><\/li>\n<\/ol>\n

                                      Correct Answer: 2. SRST router configuration and local dial-peer\/dial-plan settings<\/b><\/p>\n

                                      Explanation :-<\/b><\/p>\n

                                      During SRST operation, the branch router provides local call-processing functionality when communication with Cisco Unified Communications Manager is unavailable. The router’s SRST configuration, local directory-number registration behavior, dial-peers, destination patterns, and digit manipulation must support the expected local calling behavior. Central Unified Communications Manager route filters and media resource groups may not control local call routing while the site is operating independently in SRST mode. Therefore, the administrator should first verify the SRST router configuration and the local dial plan used during fallback.<\/span><\/p>\n

                                      Question 379. A Cisco Unified Communications Manager administrator wants to ensure that calls from a specific device are sent through the gateway associated with the device’s current location without creating separate route lists for every location. Which feature is designed to simplify this type of gateway selection?<\/b><\/p>\n

                                        \n
                                      1. <\/b> Local Route Group<\/span><\/li>\n
                                      2. <\/b> SIP OPTIONS<\/span><\/li>\n
                                      3. <\/b> Route Filter<\/span><\/li>\n
                                      4. <\/b> Transcoder<\/span><\/li>\n<\/ol>\n

                                        Correct Answer: 1. Local Route Group<\/b><\/p>\n

                                        Explanation :-<\/b><\/p>\n

                                        Local Route Groups allow routing configuration to reference a logical local gateway group rather than hard-coding a specific gateway for every device or location. The device’s associated device pool can determine which gateways belong to the local route group, allowing the same route configuration to be reused across multiple locations. SIP OPTIONS provides SIP reachability checking, Route Filters restrict route-pattern matching, and Transcoders convert media between codecs. Local Route Groups are therefore particularly useful for scalable multi-site deployments where PSTN access should follow the device’s local gateway assignment.<\/span><\/p>\n

                                        Question 380. A SIP endpoint sends an INFO message during an active call to communicate signaling information associated with the call. Which statement best describes the SIP INFO method?<\/b><\/p>\n

                                          \n
                                        1. <\/b> It terminates the SIP dialog immediately.<\/span><\/li>\n
                                        2. <\/b> It performs SIP registration with the registrar.<\/span><\/li>\n
                                        3. <\/b> It carries application-specific information within an established SIP dialog.<\/span><\/li>\n
                                        4. <\/b> It replaces SDP offer\/answer negotiation for every SIP call.<\/span><\/li>\n<\/ol>\n

                                          Correct Answer: 3. It carries application-specific information within an established SIP dialog.<\/b><\/p>\n

                                          Explanation :-<\/b><\/p>\n

                                          The SIP INFO method is used to transport application-level or session-related information within an existing SIP dialog. It does not inherently terminate the dialog, perform endpoint registration, or replace the normal SDP offer\/answer process. Depending on the implementation and negotiated service, INFO can be used for specific signaling functions such as certain DTMF or application-related information. Understanding the distinction between SIP methods is important when troubleshooting signaling. REGISTER handles registration, BYE terminates a dialog, and INVITE is used for session establishment or modification, while INFO carries additional information within an established dialog.<\/span><\/p>\n

                                           <\/p>\n","protected":false},"excerpt":{"rendered":"

                                          View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps   Question 361. A Cisco Unified Communications Manager administrator needs to route calls based on the calling party’s URI domain rather than the traditional numeric dialed digits. Which feature is most appropriate for this requirement? Route Group URI dialing Translation Pattern Local Route […]<\/p>\n","protected":false},"author":1,"featured_media":0,"comment_status":"closed","ping_status":"closed","sticky":false,"template":"","format":"standard","meta":[],"categories":[1648,1647],"tags":[],"_links":{"self":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23439"}],"collection":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts"}],"about":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/types\/post"}],"author":[{"embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/users\/1"}],"replies":[{"embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/comments?post=23439"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23439\/revisions"}],"predecessor-version":[{"id":23440,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23439\/revisions\/23440"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=23439"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=23439"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=23439"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}