{"id":23441,"date":"2026-09-28T06:40:38","date_gmt":"2026-09-28T06:40:38","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=23441"},"modified":"2026-09-28T06:40:38","modified_gmt":"2026-09-28T06:40:38","slug":"cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-20-q381-400","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-20-q381-400\/","title":{"rendered":"Cisco CCNP Collaboration 300-815 Practice Test Questions and Exam Dumps Part 20 Q381-400"},"content":{"rendered":"

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

 <\/p>\n

Question 381. A Cisco Unified Communications Manager administrator wants to route calls to a destination based on a SIP URI while preserving the URI format instead of converting it to numeric digits. Which routing capability should be configured?<\/b><\/p>\n

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

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

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

    A URI Route Pattern allows Cisco Unified Communications Manager to route calls using SIP URI information rather than relying exclusively on numeric digit strings. This is useful in deployments that use directory URIs, domains, or SIP-based addressing. Translation Patterns are primarily used to manipulate digits, while Route Filters restrict matching within numeric routing patterns. Route Groups define the gateways or trunks used after a routing decision has been made. When the routing requirement is based specifically on the destination URI, a URI Route Pattern provides the appropriate mechanism for identifying and routing the call.<\/span><\/p>\n

    Question 382. A company has multiple Cisco CUBE dial-peers that could match an outbound call. The administrator wants to ensure that the dial-peer intended for a specific destination pattern takes precedence over a more general match. Which configuration should be reviewed?<\/b><\/p>\n

      \n
    1. <\/b> Dial-peer matching criteria and preference<\/span><\/li>\n
    2. <\/b> SIP registrar configuration<\/span><\/li>\n
    3. <\/b> RTP port range<\/span><\/li>\n
    4. <\/b> Codec payload size<\/span><\/li>\n<\/ol>\n

      Correct Answer: 1. Dial-peer matching criteria and preference<\/b><\/p>\n

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

      CUBE uses dial-peer matching rules to determine which outbound dial-peer handles a call. When multiple dial-peers can potentially match, administrators must evaluate the configured destination patterns, URI criteria, and preference values as applicable. A more specific matching configuration should be designed so that the intended path is selected consistently. SIP registrar settings concern registration, RTP port ranges affect media transport, and codec payload size concerns media characteristics. Therefore, troubleshooting an unexpected outbound dial-peer selection should begin with the dial-peer matching criteria and relevant preference configuration.<\/span><\/p>\n

      Question 383. During a SIP call, an endpoint places the call on hold and later resumes it. The administrator observes a new SDP exchange within the existing SIP dialog. Which SIP mechanism commonly carries this session modification?<\/b><\/p>\n

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

        Correct Answer: 4. Re-INVITE<\/b><\/p>\n

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

        A SIP re-INVITE can be used to modify session parameters within an existing SIP dialog. Call hold and resume commonly involve changes to the SDP, such as modifying media direction attributes. The endpoints exchange updated session information so that the media state can be changed without establishing an entirely new call. REGISTER is used for registration, OPTIONS is commonly used for capability or reachability checks, and BYE terminates an established dialog. Therefore, observing an SDP modification during an established SIP call is commonly associated with a re-INVITE or another in-dialog session modification mechanism, with re-INVITE being the expected mechanism in this scenario.<\/span><\/p>\n

        Question 384. A Cisco Unified Communications Manager deployment needs to ensure that a user’s calling number is presented to an external PSTN gateway in a format required by the service provider. Which configuration area is most directly responsible for modifying the calling-party digits?<\/b><\/p>\n

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

          Correct Answer: 2. Calling-party transformation<\/b><\/p>\n

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

          Calling-party transformation settings can modify the calling-party number before the call is sent toward the PSTN. This is useful when a service provider requires a specific numbering format, such as an E.164 representation or a site-specific presentation number. Locations are primarily associated with bandwidth-based call admission control, while Media Resource Groups control access to media resources. Route Filters restrict which numbers can match routing patterns rather than directly defining the presentation of the caller’s number. Therefore, calling-party transformation is the relevant configuration area for controlling outbound caller-ID digit formatting.<\/span><\/p>\n

          Question 385. A Cisco CUBE router is configured to use a SIP trunk toward an ITSP. The ITSP requires SIP signaling to use TLS rather than unencrypted UDP. Which configuration should the administrator primarily verify?<\/b><\/p>\n

            \n
          1. <\/b> SIP transport and TLS configuration<\/span><\/li>\n
          2. <\/b> Voice translation-rule numbering<\/span><\/li>\n
          3. <\/b> RTP codec preference only<\/span><\/li>\n
          4. <\/b> DHCP relay configuration<\/span><\/li>\n<\/ol>\n

            Correct Answer: 1. SIP transport and TLS configuration<\/b><\/p>\n

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

            When an ITSP requires SIP signaling over TLS, the CUBE configuration must support the appropriate secure SIP transport and associated TLS parameters. The administrator should verify transport settings, certificates or trust configuration as applicable, and the SIP dial-peer or global settings controlling secure signaling. Voice translation rules affect digit manipulation and do not establish SIP encryption. Codec preferences control media negotiation, while DHCP relay configuration is unrelated to SIP signaling security. Proper TLS configuration allows SIP messages to be exchanged securely with the provider while maintaining interoperability with the provider’s requirements.<\/span><\/p>\n

            Question 386. A Cisco Unified Communications Manager administrator wants to limit the amount of WAN bandwidth that can be consumed by calls between two locations. Which configuration is primarily used for this purpose?<\/b><\/p>\n

              \n
            1. <\/b> Route Group<\/span><\/li>\n
            2. <\/b> SIP Profile<\/span><\/li>\n
            3. <\/b> Location-based Call Admission Control<\/span><\/li>\n
            4. <\/b> Translation Pattern<\/span><\/li>\n<\/ol>\n

              Correct Answer: 3. Location-based Call Admission Control<\/b><\/p>\n

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

              Cisco Unified Communications Manager uses locations and bandwidth settings as part of Call Admission Control to manage the amount of voice traffic permitted between locations. This helps prevent excessive call traffic from consuming WAN bandwidth and degrading voice quality. Route Groups determine gateway or trunk selection, SIP Profiles influence SIP signaling behavior, and Translation Patterns manipulate or route digits. When the requirement specifically concerns limiting call bandwidth between geographic locations, the administrator should evaluate the Location configuration and associated Call Admission Control settings.<\/span><\/p>\n

              Question 387. A Cisco Unified Communications Manager administrator needs to send calls through different gateways depending on the originating device’s location, while using the same route pattern at every site. Which feature best supports this design?<\/b><\/p>\n

                \n
              1. <\/b> Route Filter<\/span><\/li>\n
              2. <\/b> Local Route Group<\/span><\/li>\n
              3. <\/b> SIP INFO<\/span><\/li>\n
              4. <\/b> Media Termination Point<\/span><\/li>\n<\/ol>\n

                Correct Answer: 2. Local Route Group<\/b><\/p>\n

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

                Local Route Groups allow a common routing configuration to use a gateway associated with the originating device’s local location. Instead of creating separate route patterns for every site, the administrator can reference a logical local route group and allow the device pool configuration to determine the appropriate local gateway. Route Filters restrict digit matching, SIP INFO carries signaling information within an established dialog, and Media Termination Points provide specific media-processing capabilities. Local Route Groups therefore provide an efficient method for implementing reusable multi-site PSTN routing.<\/span><\/p>\n

                Question 388. A SIP call is being established through an intermediary device that needs to remain in the media path so it can provide media processing services. Which SDP-related behavior allows the intermediary to advertise its own media address instead of allowing endpoints to send RTP directly to each other?<\/b><\/p>\n

                  \n
                1. <\/b> Digit normalization<\/span><\/li>\n
                2. <\/b> URI routing<\/span><\/li>\n
                3. <\/b> Calling-party transformation<\/span><\/li>\n
                4. <\/b> Media anchoring<\/span><\/li>\n<\/ol>\n

                  Correct Answer: 4. Media anchoring<\/b><\/p>\n

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

                  Media anchoring causes a network element to remain involved in the RTP media path rather than allowing endpoints to communicate directly. The intermediary can advertise an appropriate media address in SDP so that RTP is sent through it. This design can be required for media processing, NAT traversal, security policies, recording, or interoperability functions. Digit normalization and calling-party transformation affect signaling information, while URI routing determines how SIP destinations are routed. Therefore, when an intermediary must remain in the media path and advertise its own RTP endpoint information, media anchoring is the relevant concept.<\/span><\/p>\n

                  Question 389. A Cisco CUBE administrator needs to manipulate a SIP message to accommodate a provider-specific signaling requirement without changing the overall dial plan. Which CUBE feature can be used to modify SIP headers or message behavior?<\/b><\/p>\n

                    \n
                  1. <\/b> SIP Profile<\/span><\/li>\n
                  2. <\/b> Location<\/span><\/li>\n
                  3. <\/b> Route Filter<\/span><\/li>\n
                  4. <\/b> Codec class<\/span><\/li>\n<\/ol>\n

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

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

                    SIP Profiles on Cisco voice gateways and CUBE can be used to modify SIP signaling behavior for interoperability with service providers or other SIP systems. Depending on the supported configuration, SIP Profiles can manipulate selected SIP headers or signaling elements without requiring the administrator to redesign the entire dial plan. Locations are associated with Unified Communications Manager call admission control, Route Filters constrain route-pattern matching, and codec configuration controls media negotiation. When a provider requires a specific SIP signaling modification, a SIP Profile is therefore an important configuration mechanism to evaluate.<\/span><\/p>\n

                    Question 390. A Cisco Unified Communications Manager administrator observes that a call reaches the correct gateway but the called number is sent to the PSTN in an incorrect format. Which configuration should be examined to determine whether digits were modified before routing?<\/b><\/p>\n

                      \n
                    1. <\/b> Device Pool<\/span><\/li>\n
                    2. <\/b> Translation Pattern and digit transformations<\/span><\/li>\n
                    3. <\/b> Region<\/span><\/li>\n
                    4. <\/b> Media Resource Group<\/span><\/li>\n<\/ol>\n

                      Correct Answer: 2. Translation Pattern and digit transformations<\/b><\/p>\n

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

                      Translation Patterns and other digit-transformation configurations can alter the called number before it is sent to a gateway or trunk. If the correct gateway is selected but the PSTN receives an incorrect destination number, the administrator should trace the call through the digit-manipulation stages and verify any translation patterns, route-pattern transformations, and gateway-side transformations. Device Pools provide common device configuration, Regions influence codec selection, and Media Resource Groups provide media resources. Therefore, digit transformation configuration is the appropriate area to investigate when the routing path is correct but the outgoing digits are wrong.<\/span><\/p>\n

                      Question 391. A SIP endpoint needs to confirm that another SIP peer is reachable and capable of responding to SIP requests without establishing a call. Which SIP method is commonly used for this purpose?<\/b><\/p>\n

                        \n
                      1. <\/b> BYE<\/span><\/li>\n
                      2. <\/b> REFER<\/span><\/li>\n
                      3. <\/b> OPTIONS<\/span><\/li>\n
                      4. <\/b> CANCEL<\/span><\/li>\n<\/ol>\n

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

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

                        The SIP OPTIONS method can be used to query the capabilities or reachability of a SIP endpoint or peer without establishing a normal call session. It is commonly used for keepalive and health-check purposes in SIP deployments. BYE terminates an established dialog, REFER is commonly associated with call-transfer signaling, and CANCEL terminates a pending request such as an unanswered INVITE. Therefore, when the objective is to check whether a SIP peer responds without placing a call, OPTIONS is the appropriate SIP method to consider.<\/span><\/p>\n

                        Question 392. A Cisco Unified Communications Manager administrator wants to allow a user to transfer an active call to another destination using SIP signaling. Which SIP method is commonly associated with requesting a transfer?<\/b><\/p>\n

                          \n
                        1. <\/b> REFER<\/span><\/li>\n
                        2. <\/b> REGISTER<\/span><\/li>\n
                        3. <\/b> OPTIONS<\/span><\/li>\n
                        4. <\/b> PRACK<\/span><\/li>\n<\/ol>\n

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

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

                          The SIP REFER method is commonly used to request that another SIP endpoint initiate a new request toward a specified destination, making it an important mechanism for SIP call-transfer scenarios. The receiving endpoint can process the referral and establish the requested connection. REGISTER is used for registration, OPTIONS can be used for capability or reachability checks, and PRACK is associated with reliable provisional responses. Therefore, when examining SIP signaling for a transfer operation, REFER is the method most directly associated with the transfer request.<\/span><\/p>\n

                          Question 393. A remote Cisco IP phone loses connectivity to Cisco Unified Communications Manager but remains connected to the branch network. The administrator wants the phone to register to the branch router and continue basic calling. Which feature is designed for this situation?<\/b><\/p>\n

                            \n
                          1. <\/b> Extension Mobility<\/span><\/li>\n
                          2. <\/b> Device Mobility<\/span><\/li>\n
                          3. <\/b> Survivable Remote Site Telephony<\/span><\/li>\n
                          4. <\/b> Route Filtering<\/span><\/li>\n<\/ol>\n

                            Correct Answer: 4. Survivable Remote Site Telephony<\/b><\/p>\n

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

                            Survivable Remote Site Telephony, or SRST, provides local call-processing capabilities when IP phones at a remote site lose connectivity to the central Cisco Unified Communications Manager system. A branch router configured for SRST can allow supported phones to register locally and provide basic telephony services during the outage. Extension Mobility allows users to log into phones with personal profiles, while Device Mobility applies location-specific configuration. Route Filtering controls routing-pattern matching. SRST is therefore the feature specifically designed to provide telephony survivability during loss of WAN or central call-processing connectivity.<\/span><\/p>\n

                            Question 394. A Cisco CUBE router must select an outbound dial-peer based on the domain portion of a destination SIP URI. Which dial-peer configuration is most directly relevant?<\/b><\/p>\n

                              \n
                            1. <\/b> Destination URI configuration<\/span><\/li>\n
                            2. <\/b> Voice-port signaling type<\/span><\/li>\n
                            3. <\/b> DSCP marking<\/span><\/li>\n
                            4. <\/b> RTP timeout configuration<\/span><\/li>\n<\/ol>\n

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

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

                              CUBE supports SIP URI-based dial-peer matching, allowing routing decisions to be made using information such as the destination URI and its domain. This is especially useful in SIP deployments where different domains represent different providers, enterprises, or routing destinations. Voice-port signaling applies to specific voice interfaces, DSCP marking is related to QoS classification, and RTP timeout settings concern media behavior. When the administrator needs to select a dial-peer according to the SIP destination domain, destination URI matching should be reviewed.<\/span><\/p>\n

                              Question 395. A SIP endpoint sends a provisional response reliably and the originating endpoint is expected to acknowledge receipt of that provisional response. Which SIP method provides this acknowledgment?<\/b><\/p>\n

                                \n
                              1. <\/b> INFO<\/span><\/li>\n
                              2. <\/b> PRACK<\/span><\/li>\n
                              3. <\/b> UPDATE<\/span><\/li>\n
                              4. <\/b> MESSAGE<\/span><\/li>\n<\/ol>\n

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

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

                                PRACK, or Provisional Response Acknowledgment, is used to acknowledge reliable provisional SIP responses. It is associated with SIP mechanisms that allow certain provisional responses to be delivered reliably instead of relying only on normal retransmission behavior. INFO carries application-specific information within an established dialog, UPDATE can modify session information under supported conditions, and MESSAGE is used for instant-message-style content. Therefore, when a SIP endpoint must explicitly acknowledge a reliable provisional response, PRACK is the appropriate method.<\/span><\/p>\n

                                Question 396. A Cisco Unified Communications Manager administrator wants to provide users with a group of shared phones that can ring when a call is placed to a common business number. Which feature should be evaluated?<\/b><\/p>\n

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

                                  Correct Answer: 4. Hunt Group<\/b><\/p>\n

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

                                  Hunt Groups allow incoming calls to be distributed among multiple directory numbers or phones according to configured line-group and hunt-list behavior. They are commonly used for shared business numbers such as reception, support, or sales groups. Route Filters control digit matching, SIP Profiles affect SIP signaling, and Locations are used for bandwidth-related call admission control. When several phones need to participate in a shared incoming-call distribution arrangement, a Hunt Group and its associated components provide the appropriate call-routing functionality.<\/span><\/p>\n

                                  Question 397. A Cisco Unified Communications Manager administrator needs to prevent a call from being admitted when insufficient bandwidth remains between two locations. Which feature provides this admission-control decision?<\/b><\/p>\n

                                    \n
                                  1. <\/b> Calling Search Space<\/span><\/li>\n
                                  2. <\/b> SIP Profile<\/span><\/li>\n
                                  3. <\/b> Route Group<\/span><\/li>\n
                                  4. <\/b> Call Admission Control<\/span><\/li>\n<\/ol>\n

                                    Correct Answer: 3. Call Admission Control<\/b><\/p>\n

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

                                    Call Admission Control determines whether a new call can be permitted based on available network resources and configured bandwidth policies. In Cisco Unified Communications Manager, locations and related bandwidth settings can be used to control calls across WAN links. Calling Search Spaces determine destination access, SIP Profiles affect SIP signaling behavior, and Route Groups determine gateway or trunk selection. Therefore, when the requirement is to prevent calls from being admitted because insufficient bandwidth is available, Call Admission Control is the relevant feature to evaluate.<\/span><\/p>\n

                                    Question 398. A Cisco CUBE deployment has different SIP signaling requirements on the provider-facing and enterprise-facing sides of the system. The administrator needs to apply different signaling modifications depending on the dial-peer. Which configuration approach is appropriate?<\/b><\/p>\n

                                      \n
                                    1. <\/b> Apply the same Location to every dial-peer<\/span><\/li>\n
                                    2. <\/b> Use SIP Profiles associated with the relevant SIP dial-peers<\/span><\/li>\n
                                    3. <\/b> Configure one global Route Filter<\/span><\/li>\n
                                    4. <\/b> Change all endpoints to the same codec<\/span><\/li>\n<\/ol>\n

                                      Correct Answer: 2. Use SIP Profiles associated with the relevant SIP dial-peers<\/b><\/p>\n

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

                                      SIP Profiles can be associated with applicable SIP dial-peers to modify signaling behavior for specific call legs. This is useful when an enterprise side and provider side have different interoperability requirements. Applying a single unrelated routing or location configuration does not provide targeted SIP message manipulation. Similarly, forcing every endpoint to use the same codec does not address SIP-header or signaling differences. Using appropriate SIP Profiles on the relevant dial-peers allows CUBE to apply provider- or network-specific signaling adjustments while preserving the overall routing architecture.<\/span><\/p>\n

                                      Question 399. A Cisco Unified Communications Manager administrator wants a user to retain the same personal phone configuration while moving between supported shared workstations. Which feature allows the user’s profile to be loaded onto the phone after authentication?<\/b><\/p>\n

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

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

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

                                        Extension Mobility allows a user to authenticate to a supported Cisco IP phone and load a configured device profile onto that phone. The profile can contain the user’s directory numbers, speed dials, services, and other applicable personal settings. This makes the feature useful in environments where employees move between workstations or use shared phones. Local Route Groups control gateway selection, Call Admission Control manages available network bandwidth, and Route Patterns determine call-routing destinations. Therefore, Extension Mobility directly satisfies the requirement for loading a user’s personal telephony configuration onto a different phone.<\/span><\/p>\n

                                        Question 400. A SIP call attempt is canceled by the originating endpoint before the remote party answers. Which SIP method is normally used to terminate the pending INVITE transaction?<\/b><\/p>\n

                                          \n
                                        1. <\/b> BYE<\/span><\/li>\n
                                        2. <\/b> REFER<\/span><\/li>\n
                                        3. <\/b> CANCEL<\/span><\/li>\n
                                        4. <\/b> OPTIONS<\/span><\/li>\n<\/ol>\n

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

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

                                          The SIP CANCEL method is used to terminate a pending request such as an INVITE before the INVITE transaction has completed. It is commonly generated when the caller stops trying to establish the call before receiving a final response. BYE is used to terminate an already established dialog, REFER is commonly associated with call-transfer operations, and OPTIONS is used for capability or reachability checks. Therefore, when an originating endpoint cancels a call attempt that has not yet been answered, CANCEL is the appropriate SIP method for terminating the pending INVITE transaction.<\/span><\/p>\n

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

                                          View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps   Question 381. A Cisco Unified Communications Manager administrator wants to route calls to a destination based on a SIP URI while preserving the URI format instead of converting it to numeric digits. Which routing capability should be configured? Route Filter Translation Pattern […]<\/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\/23441"}],"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=23441"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23441\/revisions"}],"predecessor-version":[{"id":23442,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23441\/revisions\/23442"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=23441"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=23441"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=23441"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}