{"id":23405,"date":"2026-09-28T06:05:00","date_gmt":"2026-09-28T06:05:00","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=23405"},"modified":"2026-09-28T06:05:00","modified_gmt":"2026-09-28T06:05:00","slug":"cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-2-q21-40","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-2-q21-40\/","title":{"rendered":"Cisco CCNP Collaboration 300-815 Practice Test Questions and Exam Dumps Part 2 Q21-40"},"content":{"rendered":"

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

 <\/p>\n

Question 21. A Cisco Unified Communications Manager deployment uses SIP trunks to connect two clusters. Calls can be established, but the remote cluster does not receive the expected caller ID information. Which area should the engineer investigate first?<\/b><\/p>\n

    \n
  1. <\/b> RTP port range<\/span><\/li>\n
  2. <\/b> NTP stratum<\/span><\/li>\n
  3. <\/b> DHCP option 150<\/span><\/li>\n
  4. <\/b> SIP identity and header information<\/span><\/li>\n<\/ol>\n

    Correct Answer: 4. SIP identity and header information<\/b><\/p>\n

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

    Caller identification in a SIP environment depends on signaling information carried in SIP headers and identity mechanisms. When calls successfully establish but the remote system displays incorrect or missing caller ID, the engineer should inspect the SIP messages and verify fields such as From, P-Asserted-Identity, or other identity-related headers according to the deployment. RTP port configuration affects media rather than caller identity. DHCP option 150 is primarily associated with Cisco IP phone provisioning, while NTP stratum concerns time synchronization. A SIP trace can determine whether the originating system generated the expected identity information and whether an intermediary such as CUBE modified or removed it.<\/span><\/p>\n

    Question 22. A Cisco Unified Communications Manager administrator needs to prevent calls to a group of expensive international destinations while continuing to allow domestic calls. Which dial-plan component is most appropriate for defining the permitted or blocked destination patterns?<\/b><\/p>\n

      \n
    1. <\/b> Route pattern<\/span><\/li>\n
    2. <\/b> Media resource group<\/span><\/li>\n
    3. <\/b> Device pool<\/span><\/li>\n
    4. <\/b> SIP profile<\/span><\/li>\n<\/ol>\n

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

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

      Route patterns are used in Cisco Unified Communications Manager to identify dialed-number patterns and determine how calls should be routed. They can be designed to restrict or permit specific classes of destinations through appropriate routing and partition\/calling-search-space design. For example, international prefixes can be placed in a restricted partition that is not accessible to ordinary users. Media resource groups control media-resource selection, device pools provide common device configuration, and SIP profiles manipulate SIP signaling. A secure dial plan should combine route patterns with calling search spaces and partitions so that users receive only the destinations they are authorized to call.<\/span><\/p>\n

      Question 23. A user has access to a partition containing an international route pattern, but another user with the same phone configuration cannot use it. Which configuration most directly controls which partitions a device or line can search?<\/b><\/p>\n

        \n
      1. <\/b> SIP profile<\/span><\/li>\n
      2. <\/b> Codec preference<\/span><\/li>\n
      3. <\/b> Calling Search Space<\/span><\/li>\n
      4. <\/b> Media termination point<\/span><\/li>\n<\/ol>\n

        Correct Answer: 3. Calling Search Space<\/b><\/p>\n

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

        A Calling Search Space, or CSS, determines which partitions are available to a calling device or line when Cisco Unified Communications Manager performs digit analysis and route selection. Two phones can therefore have similar physical configurations but different calling permissions because their CSS configurations differ. Partitions organize directory numbers, route patterns, translation patterns, and other dial-plan objects into logical groups. Codec preferences influence media negotiation, SIP profiles manipulate signaling, and media termination points provide media-processing functions. When one user can reach a destination while another cannot, checking the calling device and line CSS is an important troubleshooting step.<\/span><\/p>\n

        Question 24. An administrator needs to change a dialed number before Cisco Unified Communications Manager performs final route selection. Which dial-plan element is designed to manipulate digits during call processing?<\/b><\/p>\n

          \n
        1. <\/b> Route group<\/span><\/li>\n
        2. <\/b> Translation pattern<\/span><\/li>\n
        3. <\/b> Device pool<\/span><\/li>\n
        4. <\/b> SIP trunk<\/span><\/li>\n<\/ol>\n

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

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

          Translation patterns allow Cisco Unified Communications Manager to manipulate a dialed number before continuing call processing. They can be used to add, remove, or modify digits and then route the translated destination according to the resulting number. Translation patterns can also be associated with partitions and calling search spaces, so their visibility should be considered carefully when designing a dial plan. Route groups determine gateway selection, device pools provide common configuration settings, and SIP trunks define signaling connectivity to another system. When users dial one format but the network requires another numbering format, translation patterns are a common solution.<\/span><\/p>\n

          Question 25. A Cisco Unified Communications Manager cluster has multiple voice gateways. The administrator wants calls to use one gateway first and automatically use another gateway when the preferred gateway is unavailable. Which configuration provides this ordered gateway selection?<\/b><\/p>\n

            \n
          1. <\/b> Route list with an appropriate route group order<\/span><\/li>\n
          2. <\/b> SIP OPTIONS only<\/span><\/li>\n
          3. <\/b> Device mobility<\/span><\/li>\n
          4. <\/b> Codec transcoding<\/span><\/li>\n<\/ol>\n

            Correct Answer: 1. Route list with an appropriate route group order<\/b><\/p>\n

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

            A route list can contain one or more route groups and provides a structured way to select gateways for a call. By arranging route groups in an appropriate order, Cisco Unified Communications Manager can attempt the preferred gateway first and use another gateway when the first path is unavailable or otherwise unsuitable. SIP OPTIONS can provide reachability or keepalive information but does not itself define the complete gateway-selection hierarchy. Device mobility controls device-location behavior, while transcoding addresses codec conversion. Route patterns, route lists, and route groups work together to implement flexible and resilient outbound call routing.<\/span><\/p>\n

            Question 26. A Cisco IP phone moves from the corporate office to a remote location and should retain the same user extension while using a different local dialing policy. Which feature is specifically intended to support this type of user mobility?<\/b><\/p>\n

              \n
            1. <\/b> Device mobility<\/span><\/li>\n
            2. <\/b> SIP normalization<\/span><\/li>\n
            3. <\/b> RSVP<\/span><\/li>\n
            4. <\/b> SRTP<\/span><\/li>\n<\/ol>\n

              Correct Answer: 4. Device mobility<\/b><\/p>\n

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

              Cisco Unified Communications Manager Device Mobility allows device configuration to adapt when a phone is moved between locations. The device can retain its identity and user association while receiving location-specific settings such as calling search spaces, device pools, regions, and other configuration parameters. This is useful when employees move between offices but should continue using their normal extension. SIP normalization changes SIP signaling, RSVP relates to resource reservation, and SRTP protects media. Device Mobility should be designed together with device pools and device mobility groups so that the appropriate local configuration is applied when a device registers from a different location.<\/span><\/p>\n

              Question 27. An enterprise wants to ensure that calls between two Cisco Unified Communications Manager locations use G.711 while calls across a lower-bandwidth WAN use a compressed codec. Which configuration concept determines the codec capabilities between locations?<\/b><\/p>\n

                \n
              1. <\/b> Calling Search Space<\/span><\/li>\n
              2. <\/b> Region configuration<\/span><\/li>\n
              3. <\/b> Route group<\/span><\/li>\n
              4. <\/b> Translation pattern<\/span><\/li>\n<\/ol>\n

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

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

                Cisco Unified Communications Manager regions define the maximum audio and video bandwidth and can influence codec selection between devices in different regions. This allows an administrator to use high-bandwidth codecs such as G.711 where sufficient bandwidth exists while using compressed codecs across constrained WAN links. Calling Search Spaces determine dial-plan reachability, route groups control gateway selection, and translation patterns modify digits. Region configuration should be considered alongside locations and bandwidth management when designing collaboration networks. Incorrect region settings can result in unexpected codec selection, bandwidth consumption, or call-quality problems between sites.<\/span><\/p>\n

                Question 28. A company has limited WAN bandwidth between two Unified Communications locations and wants to control the amount of voice traffic admitted across the link. Which Cisco Unified Communications Manager concept is specifically designed to manage available bandwidth for calls?<\/b><\/p>\n

                  \n
                1. <\/b> Partition<\/span><\/li>\n
                2. <\/b> Location-based call admission control<\/span><\/li>\n
                3. <\/b> SIP registrar<\/span><\/li>\n
                4. <\/b> Phone button template<\/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 to implement location-based call admission control. Locations can be associated with a defined amount of available audio and video bandwidth, allowing the system to determine whether a new call should be admitted across a constrained WAN path. This helps prevent excessive call traffic from consuming bandwidth needed by existing calls and other applications. Partitions control dial-plan visibility, SIP registrars handle registration functions, and phone button templates determine available phone features and button assignments. Proper location configuration is particularly important when collaboration traffic crosses low-bandwidth WAN links.<\/span><\/p>\n

                  Question 29. A video call between two Cisco collaboration endpoints has acceptable signaling but poor quality because one side is transmitting high-bandwidth video over a constrained WAN link. Which Unified Communications Manager feature can help control video bandwidth between locations?<\/b><\/p>\n

                    \n
                  1. <\/b> Region configuration<\/span><\/li>\n
                  2. <\/b> Translation pattern<\/span><\/li>\n
                  3. <\/b> Calling Search Space<\/span><\/li>\n
                  4. <\/b> Route group<\/span><\/li>\n<\/ol>\n

                    Correct Answer: 4. Region configuration<\/b><\/p>\n

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

                    Region configuration in Cisco Unified Communications Manager controls the maximum audio and video bandwidth permitted between regions. By defining suitable bandwidth relationships, an administrator can prevent high-bandwidth video calls from consuming more WAN capacity than the network can support. This is especially important when collaboration endpoints in different regions communicate across constrained links. Translation patterns and calling search spaces affect dial-plan processing, while route groups determine gateway selection. When video quality problems are associated with WAN capacity, the engineer should examine region bandwidth settings along with actual transport utilization and the selected video codec.<\/span><\/p>\n

                    Question 30. A user can call internal extensions but cannot reach an external PSTN number. The route pattern is correctly configured, but the phone’s line CSS does not include the partition containing the route pattern. What change is required?<\/b><\/p>\n

                      \n
                    1. <\/b> Add the route-pattern partition to the applicable Calling Search Space<\/span><\/li>\n
                    2. <\/b> Change the phone’s codec preference<\/span><\/li>\n
                    3. <\/b> Increase the RTP port range<\/span><\/li>\n
                    4. <\/b> Configure SRTP<\/span><\/li>\n<\/ol>\n

                      Correct Answer: 1. Add the route-pattern partition to the applicable Calling Search Space<\/b><\/p>\n

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

                      Cisco Unified Communications Manager uses the calling device or line’s Calling Search Space to determine which partitions can be searched during digit analysis. If a route pattern exists but its partition is not accessible through the caller’s CSS, the route pattern cannot be selected for the call. Adding the appropriate partition to the applicable CSS makes the route pattern visible to the user, assuming no other dial-plan restriction prevents the call. Codec settings and RTP ports affect media, while SRTP provides media security. This distinction is important because a routing object can be correctly configured yet remain unreachable due to partition and CSS relationships.<\/span><\/p>\n

                      Question 31. A Cisco Unified Communications Manager cluster must send a call to an external SIP system. The administrator wants Unified Communications Manager to select a SIP trunk after matching a specific destination pattern. Which dial-plan objects work together for this purpose?<\/b><\/p>\n

                        \n
                      1. <\/b> Device pool and media resource group<\/span><\/li>\n
                      2. <\/b> Route pattern and route list<\/span><\/li>\n
                      3. <\/b> Location and region<\/span><\/li>\n
                      4. <\/b> SIP profile and phone button template<\/span><\/li>\n<\/ol>\n

                        Correct Answer: 2. Route pattern and route list<\/b><\/p>\n

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

                        A route pattern identifies a destination number pattern and can point call processing toward a route list. The route list then references a route group containing the appropriate gateway or SIP trunk. This separation allows the dial plan to match destinations independently from the physical or logical path selected for the call. Device pools and media resource groups address different configuration areas, while locations and regions are primarily associated with bandwidth and codec considerations. SIP profiles can modify SIP signaling, but they do not replace the fundamental route-pattern and route-list structure used for destination-based call routing.<\/span><\/p>\n

                        Question 32. A Cisco Unified Communications Manager administrator needs to allow users in one department to call a set of internal directory numbers while preventing access to another department’s numbers. Which combination provides logical dial-plan access control?<\/b><\/p>\n

                          \n
                        1. <\/b> Codec lists and regions<\/span><\/li>\n
                        2. <\/b> Route groups and media resources<\/span><\/li>\n
                        3. <\/b> Partitions and Calling Search Spaces<\/span><\/li>\n
                        4. <\/b> SIP profiles and TLS profiles<\/span><\/li>\n<\/ol>\n

                          Correct Answer: 3. Partitions and Calling Search Spaces<\/b><\/p>\n

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

                          Partitions and Calling Search Spaces provide the fundamental logical access-control mechanism for Cisco Unified Communications Manager dial plans. Directory numbers and route patterns can be assigned to partitions, while CSS configurations determine which partitions a calling device or line can search. By assigning different CSS values to departments, administrators can control which internal numbers and external destinations are reachable. Codec lists and regions address media capabilities, route groups determine gateway selection, and SIP or TLS profiles address signaling behavior. Proper partition and CSS design is therefore essential when implementing departmental calling restrictions.<\/span><\/p>\n

                          Question 33. A SIP trunk between two collaboration systems requires a specific SIP header to be changed because the receiving system rejects the original value. Which Cisco feature can modify SIP messages without changing the underlying call-routing logic?<\/b><\/p>\n

                            \n
                          1. <\/b> SIP normalization or SIP profile<\/span><\/li>\n
                          2. <\/b> Location bandwidth<\/span><\/li>\n
                          3. <\/b> Route group<\/span><\/li>\n
                          4. <\/b> Device mobility group<\/span><\/li>\n<\/ol>\n

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

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

                            A SIP header modification requirement should be addressed with a SIP normalization mechanism or SIP profile, not with location bandwidth. SIP normalization and SIP profiles can transform selected SIP messages and headers to address interoperability requirements between different systems. Location bandwidth controls call admission and available bandwidth between Unified Communications Manager locations. Route groups determine gateway or trunk selection, while device mobility groups support location-specific device configuration. When an external system rejects a call because of signaling-format differences, engineers should capture the SIP exchange, identify the incompatible header, and apply the appropriate SIP manipulation feature.<\/span><\/p>\n

                            Question 34. An enterprise wants to use different outbound gateways based on the destination while keeping gateway selection separate from the dialed-number pattern. Which design best provides this separation?<\/b><\/p>\n

                              \n
                            1. <\/b> Translation pattern only<\/span><\/li>\n
                            2. <\/b> Route pattern pointing to a route list containing route groups<\/span><\/li>\n
                            3. <\/b> Device pool pointing directly to a codec<\/span><\/li>\n
                            4. <\/b> SIP profile pointing to an NTP server<\/span><\/li>\n<\/ol>\n

                              Correct Answer: 2. Route pattern pointing to a route list containing route groups<\/b><\/p>\n

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

                              Cisco Unified Communications Manager separates destination matching from gateway selection by using route patterns, route lists, and route groups. The route pattern identifies the destination, the route list provides an ordered routing structure, and route groups contain the actual gateways or trunks that can carry the call. This architecture makes the dial plan easier to maintain and allows gateway redundancy without changing every destination pattern. Translation patterns manipulate digits rather than providing the complete gateway-selection hierarchy. Device pools and SIP profiles serve different purposes and do not replace this routing structure.<\/span><\/p>\n

                              Question 35. A branch office loses WAN connectivity to its centralized call-control cluster. Phones successfully register with the local router, but calls to the PSTN fail. Which troubleshooting area should be checked first?<\/b><\/p>\n

                                \n
                              1. <\/b> Local SRST dial peers and PSTN gateway configuration<\/span><\/li>\n
                              2. <\/b> Centralized Unified Communications Manager database replication<\/span><\/li>\n
                              3. <\/b> Remote cluster SIP normalization<\/span><\/li>\n
                              4. <\/b> Video region bandwidth<\/span><\/li>\n<\/ol>\n

                                Correct Answer: 3. Local SRST dial peers and PSTN gateway configuration<\/b><\/p>\n

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

                                When SRST registration succeeds during a WAN outage but PSTN calls fail, the local survivable call-processing environment should be examined first. SRST depends on appropriate local dial peers, voice gateway configuration, destination patterns, and PSTN connectivity. Centralized Unified Communications Manager replication is not the immediate call-routing mechanism during an active WAN outage if phones are operating under SRST. SIP normalization on a remote cluster and video bandwidth settings are unrelated to local PSTN call routing. Engineers should verify the phone’s registration state, local dial-peer matching, gateway status, and available PSTN resources.<\/span><\/p>\n

                                Question 36. A SIP trunk is configured between an enterprise and a service provider. The provider requires the enterprise to authenticate using a username and password during SIP registration. Which SIP function is directly involved?<\/b><\/p>\n

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

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

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

                                  The SIP REGISTER method is used by a SIP user agent to register its address-of-record and contact information with a registrar. When authentication is required, the registrar can challenge the REGISTER request and the client can respond using the configured credentials. REFER is associated with referred operations such as call transfer, OPTIONS can query capabilities or reachability, and BYE terminates an established dialog. During registration troubleshooting, engineers should inspect REGISTER exchanges, authentication challenges, response codes, contact information, and registration expiration values to determine why a SIP endpoint or trunk is not successfully registering.<\/span><\/p>\n

                                  Question 37. An engineer sees a SIP OPTIONS request periodically sent between two voice systems. No call is being established. What is a common purpose of this exchange?<\/b><\/p>\n

                                    \n
                                  1. <\/b> Negotiating the final RTP codec for an active call<\/span><\/li>\n
                                  2. <\/b> Transferring an established call<\/span><\/li>\n
                                  3. <\/b> Checking availability or capabilities of the remote SIP system<\/span><\/li>\n
                                  4. <\/b> Encrypting an RTP stream<\/span><\/li>\n<\/ol>\n

                                    Correct Answer: 3. Checking availability or capabilities of the remote SIP system<\/b><\/p>\n

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

                                    SIP OPTIONS can be used to query the capabilities or availability of a SIP endpoint or intermediary without establishing a normal voice session. Collaboration systems and gateways may use OPTIONS exchanges as part of keepalive or reachability mechanisms. This can help determine whether a remote SIP destination is responsive before attempting calls. OPTIONS does not itself perform call transfer, establish the media codec for an active call, or encrypt RTP. When troubleshooting SIP trunk availability, periodic OPTIONS messages and their responses can provide useful evidence about signaling reachability and remote-system responsiveness.<\/span><\/p>\n

                                    Question 38. A SIP call is established, but users report one-way audio. SIP signaling shows valid SDP on both sides. Which issue should be investigated at the network and media layer?<\/b><\/p>\n

                                      \n
                                    1. <\/b> Calling Search Space only<\/span><\/li>\n
                                    2. <\/b> Route pattern partition only<\/span><\/li>\n
                                    3. <\/b> RTP reachability and firewall\/NAT handling<\/span><\/li>\n
                                    4. <\/b> Directory number description<\/span><\/li>\n<\/ol>\n

                                      Correct Answer: 3. RTP reachability and firewall\/NAT handling<\/b><\/p>\n

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

                                      One-way audio is frequently caused by problems affecting the RTP media path rather than the SIP signaling path. Even when SDP negotiation succeeds, the advertised media addresses or ports may be unreachable because of NAT, firewall policies, routing problems, or incorrect interface selection. The engineer should compare the SDP addresses and ports with the actual RTP flow and verify bidirectional reachability through any intermediary devices. Calling Search Spaces and route-pattern partitions control call routing, while a directory number description has no media effect. Packet captures and CUBE media statistics can help identify which direction of the RTP path is failing.<\/span><\/p>\n

                                      Question 39. An enterprise uses CUBE between an internal Unified Communications Manager cluster and a SIP provider. The provider sends calls with a number format different from the internal dial plan. Which CUBE capability can help transform the called number before routing it internally?<\/b><\/p>\n

                                        \n
                                      1. <\/b> SIP message manipulation or digit manipulation<\/span><\/li>\n
                                      2. <\/b> NTP authentication<\/span><\/li>\n
                                      3. <\/b> OSPF authentication<\/span><\/li>\n
                                      4. <\/b> SRTP encryption<\/span><\/li>\n<\/ol>\n

                                        Correct Answer: 1. SIP message manipulation or digit manipulation<\/b><\/p>\n

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

                                        CUBE can support interoperability between different numbering plans by using appropriate digit manipulation and SIP message manipulation mechanisms. Depending on the deployment, this can include translation rules, translation profiles, dial-peer configuration, or SIP profiles that modify signaling information. The objective is to transform the provider’s number representation into a format understood by the internal dial plan. NTP and OSPF authentication provide security for unrelated protocols, while SRTP protects media. Engineers should inspect the incoming SIP Request-URI, called-party information, dial-peer matching, and any translation configuration when troubleshooting number-format interoperability.<\/span><\/p>\n

                                        Question 40. A collaboration administrator needs to verify whether a SIP trunk is reachable before troubleshooting a complete call flow. Which signaling mechanism can provide a lightweight capability or reachability check without placing a normal call?<\/b><\/p>\n

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

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

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

                                          SIP OPTIONS can be used to query a remote SIP entity without establishing a normal call. Depending on the implementation, the response can provide information about reachability and supported SIP capabilities. This makes OPTIONS useful when troubleshooting whether a SIP trunk or peer is responding before investigating the complete INVITE, media negotiation, and call-routing sequence. BYE terminates an established dialog, REFER requests a referenced operation such as transfer, and CANCEL terminates a pending request. A successful OPTIONS exchange does not prove that an end-to-end call will work, but it provides useful evidence that SIP signaling connectivity exists.<\/span><\/p>\n

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

                                          View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps   Question 21. A Cisco Unified Communications Manager deployment uses SIP trunks to connect two clusters. Calls can be established, but the remote cluster does not receive the expected caller ID information. Which area should the engineer investigate first? RTP port range NTP […]<\/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\/23405"}],"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=23405"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23405\/revisions"}],"predecessor-version":[{"id":23406,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23405\/revisions\/23406"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=23405"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=23405"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=23405"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}