{"id":23435,"date":"2026-09-28T06:38:28","date_gmt":"2026-09-28T06:38:28","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=23435"},"modified":"2026-09-28T06:38:28","modified_gmt":"2026-09-28T06:38:28","slug":"cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-17-q321-340","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-17-q321-340\/","title":{"rendered":"Cisco CCNP Collaboration 300-815 Practice Test Questions and Exam Dumps Part 17 Q321-340"},"content":{"rendered":"

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

 <\/p>\n

Question 321. A Cisco Unified Communications Manager cluster uses multiple locations connected over a WAN. The administrator wants to limit the number of simultaneous calls between two locations based on available bandwidth. Which configuration is most directly involved?<\/b><\/p>\n

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

    Correct Answer: 2. Call Admission Control<\/b><\/p>\n

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

    Call Admission Control, or CAC, helps Cisco Unified Communications Manager determine whether a new call should be permitted across a constrained network path. Location-based bandwidth settings allow UCM to track the amount of voice bandwidth allocated between sites. When the available allocation is exhausted, UCM can prevent additional calls from being admitted over that path, helping protect the quality of existing calls. CAC is therefore an important component of multisite collaboration designs where WAN capacity is limited. SIP Profiles control SIP signaling behavior, Extension Mobility handles user profiles, and Route Groups organize routing resources. When calls fail specifically because an intersite bandwidth allocation has been reached, CAC configuration should be examined.<\/span><\/p>\n

    Question 322. A company has centralized UCM call processing but wants local PSTN access at each branch. Which UCM feature allows a common route pattern to use the appropriate gateway associated with the originating location?<\/b><\/p>\n

      \n
    1. <\/b> Global Dial Plan Replication<\/span><\/li>\n
    2. <\/b> SIP Route Pattern<\/span><\/li>\n
    3. <\/b> Local Route Group<\/span><\/li>\n
    4. <\/b> Call Park<\/span><\/li>\n<\/ol>\n

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

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

      Local Route Group allows Cisco Unified Communications Manager to associate a common routing configuration with a gateway or route group that is local to the calling device’s location. This is useful in centralized call-processing architectures where multiple branches require local PSTN access. Instead of creating completely separate route patterns for every site, administrators can use a common routing structure while the local route group determines the appropriate local gateway. Global Dial Plan Replication distributes dial-plan information between clusters, SIP Route Patterns handle URI-based routing, and Call Park temporarily holds active calls. Local Route Group therefore provides a scalable way to maintain centralized dial-plan logic while using local PSTN resources.<\/span><\/p>\n

      Question 323. A UCM administrator wants to ensure that users in a branch use a specific PSTN gateway when dialing external numbers, but the same route pattern must remain shared across several sites. Which configuration should be verified?<\/b><\/p>\n

        \n
      1. <\/b> Local Route Group association<\/span><\/li>\n
      2. <\/b> SIP OPTIONS<\/span><\/li>\n
      3. <\/b> Device Mobility softkey configuration<\/span><\/li>\n
      4. <\/b> Hunt Pilot timeout<\/span><\/li>\n<\/ol>\n

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

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

        Local Route Group association determines which locally relevant routing resources are used when a call matches a shared routing configuration. This is particularly useful in multisite Cisco Unified Communications Manager environments where the same route pattern should be reusable while each location sends PSTN calls through its own gateway. The administrator should verify the local route group assignment and the relationship between the calling device’s location, device pool, route list, and route group. SIP OPTIONS does not determine PSTN gateway selection, Device Mobility addresses location-specific device settings, and Hunt Pilot timeout controls call-distribution behavior. Proper Local Route Group configuration can significantly simplify a large enterprise dial plan.<\/span><\/p>\n

        Question 324. A UCM user dials an internal extension, but the organization wants the number normalized into a global +E.164 format before it is used by routing logic. Which function addresses this requirement?<\/b><\/p>\n

          \n
        1. <\/b> RTP transcoding<\/span><\/li>\n
        2. <\/b> SIP REFER<\/span><\/li>\n
        3. <\/b> Number normalization<\/span><\/li>\n
        4. <\/b> CUBE tenant separation<\/span><\/li>\n<\/ol>\n

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

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

          Number normalization converts telephone numbers into a consistent format that can be used throughout the collaboration environment. Cisco Unified Communications Manager deployments commonly use globalized numbering based on +E.164-style formats so that the same destination can be represented consistently regardless of the user’s location or dialing method. Normalization reduces complexity when integrating multiple sites, gateways, trunks, and applications. RTP transcoding changes media formats, SIP REFER is used for referral or transfer operations, and CUBE tenant separation provides configuration isolation for different trunk environments. When troubleshooting inconsistent number representations between endpoints and routing elements, administrators should review normalization and digit-transformation configurations.<\/span><\/p>\n

          Question 325. A PSTN carrier requires the enterprise to send a ten-digit calling number instead of the internal four-digit extension. Which UCM function can transform the calling-party number before the call leaves the enterprise?<\/b><\/p>\n

            \n
          1. <\/b> Call Admission Control<\/span><\/li>\n
          2. <\/b> Calling-party digit manipulation<\/span><\/li>\n
          3. <\/b> SIP OPTIONS<\/span><\/li>\n
          4. <\/b> Extension Mobility<\/span><\/li>\n<\/ol>\n

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

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

            Calling-party digit manipulation allows Cisco Unified Communications Manager to change the number presented as the calling party before a call is sent to an external destination. For example, an internal four-digit extension can be transformed into the organization’s complete ten-digit telephone number when required by a PSTN provider. The exact mechanism can involve route-pattern or route-list configuration and calling-party transformation settings, depending on the deployment. CAC controls bandwidth admission, SIP OPTIONS provides signaling capability information, and Extension Mobility manages user profiles. Correct calling-party presentation is important because carriers may reject or mishandle calls when the originating number does not conform to the expected numbering format.<\/span><\/p>\n

            Question 326. A Cisco IOS gateway has multiple outbound dial peers with overlapping destination patterns. The administrator wants one dial peer to be preferred when several peers match the same number. Which dial-peer attribute can influence the selection?<\/b><\/p>\n

              \n
            1. <\/b> Preference<\/span><\/li>\n
            2. <\/b> SIP session timer<\/span><\/li>\n
            3. <\/b> Codec payload type<\/span><\/li>\n
            4. <\/b> SIP registrar address<\/span><\/li>\n<\/ol>\n

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

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

              Cisco IOS voice dial peers can use the preference value to influence which matching dial peer is selected when multiple outbound dial peers are eligible. A lower preference value represents a more preferred choice when the applicable dial peers otherwise provide matching routing information. This mechanism is useful for creating primary and secondary call paths. SIP session timers control the refresh of established SIP sessions, codec payload types identify media formats, and the SIP registrar address concerns endpoint registration rather than outbound dial-peer selection. When troubleshooting why one matching dial peer is consistently selected over another, administrators should review destination-pattern matching, preference, dial-peer type, and other applicable selection rules.<\/span><\/p>\n

              Question 327. A Cisco IOS voice gateway receives an incoming call but cannot identify an appropriate incoming dial peer based on the expected signaling information. Which area should be examined first?<\/b><\/p>\n

                \n
              1. <\/b> Extension Mobility configuration<\/span><\/li>\n
              2. <\/b> UCM device pool<\/span><\/li>\n
              3. <\/b> Incoming dial-peer matching criteria<\/span><\/li>\n
              4. <\/b> Global Dial Plan Replication<\/span><\/li>\n<\/ol>\n

                Correct Answer: 3. Incoming dial-peer matching criteria<\/b><\/p>\n

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

                Cisco IOS voice gateways use several attributes to identify an incoming dial peer, depending on the signaling protocol and configuration. These can include the incoming called number, calling number, voice-port information, or URI-related information in applicable SIP configurations. If an incoming call does not reach the expected dial peer, administrators should inspect the matching criteria and verify that the configured information corresponds to what the gateway actually receives. Extension Mobility and UCM device pools are not responsible for IOS incoming dial-peer matching. Global Dial Plan Replication distributes dial-plan information between UCM clusters and does not directly determine local IOS dial-peer selection. Debugging SIP or voice call routing can help confirm the actual received values.<\/span><\/p>\n

                Question 328. A voice gateway must send calls to a provider using a specific SIP destination address. Which dial-peer parameter identifies the remote SIP destination toward which the call should be sent?<\/b><\/p>\n

                  \n
                1. <\/b> Codec preference<\/span><\/li>\n
                2. <\/b> Huntstop<\/span><\/li>\n
                3. <\/b> Translation profile<\/span><\/li>\n
                4. <\/b> Session target<\/span><\/li>\n<\/ol>\n

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

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

                  For an outbound VoIP dial peer, the session target identifies the destination toward which the signaling session is established. In SIP deployments, this can identify the remote SIP peer or provider destination that should receive the call. The session target is therefore a critical part of determining where a matched call is sent. Codec preference controls which media codecs are preferred, huntstop influences dial-peer hunting behavior, and translation profiles manipulate calling or called numbers. When a dial peer matches correctly but the call is sent toward an unexpected address or fails to reach the intended provider, administrators should verify the session target along with routing, reachability, and SIP signaling.<\/span><\/p>\n

                  Question 329. A CUBE router connects UCM to a SIP provider. The provider requires the enterprise to use a different SIP transport port than the default signaling port. Which configuration area should be investigated?<\/b><\/p>\n

                    \n
                  1. <\/b> SIP transport and listening-port configuration<\/span><\/li>\n
                  2. <\/b> Call Admission Control<\/span><\/li>\n
                  3. <\/b> Extension Mobility<\/span><\/li>\n
                  4. <\/b> Globalized number normalization<\/span><\/li>\n<\/ol>\n

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

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

                    SIP signaling depends on transport parameters such as the protocol and port used by the communicating peers. CUBE can be configured to listen for and originate SIP signaling using the transport characteristics required by the connected environment. When a provider requires a non-default signaling port, administrators should verify the CUBE SIP listening and outbound signaling configuration as well as firewall and provider requirements. CAC controls voice-resource admission, Extension Mobility manages user profiles, and number normalization handles telephone-number representation. SIP troubleshooting should include checking the actual destination address and port in signaling traces and confirming that network security devices permit the required traffic.<\/span><\/p>\n

                    Question 330. A CUBE administrator needs to support two SIP providers that require different signaling policies on the same physical router. Which CUBE feature can provide separate configuration contexts for the providers?<\/b><\/p>\n

                      \n
                    1. <\/b> SRST<\/span><\/li>\n
                    2. <\/b> CUBE tenants<\/span><\/li>\n
                    3. <\/b> Local Route Group<\/span><\/li>\n
                    4. <\/b> UCM Line Group<\/span><\/li>\n<\/ol>\n

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

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

                      CUBE tenants provide logical separation for different SIP trunk or service-provider environments on a single Cisco Unified Border Element. This allows administrators to apply provider-specific signaling, dial-peer behavior, security, or interoperability settings without treating all connected trunks as one undifferentiated configuration context. Tenant-based designs are useful in environments where multiple providers have different requirements. SRST provides branch survivability, Local Route Group is a UCM routing feature, and Line Groups are used for distributing calls among UCM endpoints. When a single CUBE must support multiple providers with independent policies, tenant configuration should be considered as part of the overall design.<\/span><\/p>\n

                      Question 331. A SIP trunk call is established, but after approximately 30 minutes the call disconnects even though both users remain active. Which SIP feature should be investigated first?<\/b><\/p>\n

                        \n
                      1. <\/b> Route Group preference<\/span><\/li>\n
                      2. <\/b> SIP session timer<\/span><\/li>\n
                      3. <\/b> DTMF relay<\/span><\/li>\n
                      4. <\/b> SIP URI pattern<\/span><\/li>\n<\/ol>\n

                        Correct Answer: 2. SIP session timer<\/b><\/p>\n

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

                        SIP session timers periodically refresh an established SIP dialog and can cause a session to be terminated when the expected refresh is not received within the negotiated interval. If calls consistently disconnect after a similar elapsed period, session-timer behavior should be investigated. Administrators should review the negotiated Session-Expires and Min-SE values, the refresh mechanism, and whether both SIP peers support the expected behavior. Route Groups determine routing resources, DTMF relay handles keypad signaling, and SIP URI patterns determine URI-based routing. A repeatable call duration followed by disconnect is therefore a useful diagnostic clue pointing toward session refresh or timer-related signaling rather than an ordinary dial-plan mismatch.<\/span><\/p>\n

                        Question 332. A SIP endpoint supports reliable provisional responses and receives a 183 response containing SDP with a Require header indicating 100rel. What should the endpoint send to acknowledge the reliable provisional response?<\/b><\/p>\n

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

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

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

                          PRACK is the SIP method used to acknowledge a reliable provisional response. When the 100rel mechanism is negotiated or required, provisional responses can be transmitted reliably, and the receiving endpoint uses PRACK to acknowledge them. This is particularly relevant when the provisional response carries SDP and early media needs to be established reliably. BYE terminates an established SIP session, UPDATE can modify session information in applicable dialogs, and CANCEL terminates a pending request such as an unanswered INVITE. During SIP trace analysis, the presence of a reliable provisional response and an associated RSeq value can help identify a situation where PRACK is expected.<\/span><\/p>\n

                          Question 333. A SIP call is being established through CUBE. The carrier sends 183 Session Progress with SDP, but the enterprise endpoint receives no early audio. Which area should be examined first?<\/b><\/p>\n

                            \n
                          1. <\/b> Media path and early-media handling<\/span><\/li>\n
                          2. <\/b> Extension Mobility service<\/span><\/li>\n
                          3. <\/b> UCM route-list name<\/span><\/li>\n
                          4. <\/b> Global Dial Plan Replication<\/span><\/li>\n<\/ol>\n

                            Correct Answer: 1. Media path and early-media handling<\/b><\/p>\n

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

                            A 183 Session Progress response containing SDP can establish early media before the call receives a final answer. If the endpoint does not hear the expected announcement or other early audio, the administrator should examine the SDP addresses and ports, RTP reachability, CUBE media handling, firewall policies, and whether the signaling path permits the expected early-media behavior. The UCM route-list name does not determine whether RTP can reach the endpoint. Extension Mobility and Global Dial Plan Replication are unrelated to early-media transport. Packet captures and SIP traces can help determine whether SDP is being passed correctly and whether RTP packets are actually flowing in the expected direction.<\/span><\/p>\n

                            Question 334. A company wants a remote user to retain the same enterprise identity when placing calls from a configured mobile phone. Which UCM capability is most closely associated with this requirement?<\/b><\/p>\n

                              \n
                            1. <\/b> Call Admission Control<\/span><\/li>\n
                            2. <\/b> Unified Mobility<\/span><\/li>\n
                            3. <\/b> Route Group<\/span><\/li>\n
                            4. <\/b> Global Dial Plan Replication<\/span><\/li>\n<\/ol>\n

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

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

                              Cisco Unified Mobility allows an enterprise user to extend selected enterprise calling capabilities to associated mobile destinations. Depending on the configuration, calls originating from a remote or mobile destination can be handled using enterprise calling identity and policies. This helps users remain reachable and maintain an integrated enterprise communication experience while away from their desk phones. Call Admission Control regulates network-resource usage, Route Groups organize outbound routing resources, and Global Dial Plan Replication distributes dial-plan information between UCM clusters. When the requirement involves maintaining enterprise calling behavior and identity while using an associated mobile device, Unified Mobility and its remote-destination configuration should be reviewed.<\/span><\/p>\n

                              Question 335. A user moves a Cisco IP phone from one office to another. The administrator wants the phone to receive different settings automatically based on its new physical location. Which UCM feature is designed for this type of behavior?<\/b><\/p>\n

                                \n
                              1. <\/b> SIP PRACK<\/span><\/li>\n
                              2. <\/b> CUBE tenant<\/span><\/li>\n
                              3. <\/b> Device Mobility<\/span><\/li>\n
                              4. <\/b> DTMF relay<\/span><\/li>\n<\/ol>\n

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

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

                                Device Mobility allows Cisco Unified Communications Manager to apply location-specific configuration to supported devices when they operate away from their usual home location. Depending on the configuration, the phone can receive settings such as calling-search-space, device pool, region, location, or other relevant parameters associated with the new location. This helps organizations maintain consistent behavior as users and devices move between offices. SIP PRACK acknowledges reliable provisional responses, CUBE tenants separate border-element contexts, and DTMF relay transports keypad digits. When a phone’s behavior must automatically adapt to its physical location, administrators should review Device Mobility information, device pools, physical locations, and device mobility settings.<\/span><\/p>\n

                                Question 336. A UCM administrator needs to distribute calls among several agents who share responsibility for answering a service number. Which feature provides the basic group of directory numbers used by a hunt mechanism?<\/b><\/p>\n

                                  \n
                                1. <\/b> Line Group<\/span><\/li>\n
                                2. <\/b> Route List<\/span><\/li>\n
                                3. <\/b> Translation Pattern<\/span><\/li>\n
                                4. <\/b> SIP Route Pattern<\/span><\/li>\n<\/ol>\n

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

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

                                  A Line Group in Cisco Unified Communications Manager contains directory numbers that can participate in a hunt configuration. The Line Group is then associated with a Hunt List and Hunt Pilot to define how calls to the service number are distributed among the available members. This structure allows organizations to implement sequential, circular, broadcast, or other supported hunting behaviors according to the configured design. A Route List is used for outbound call routing, Translation Patterns transform dialed digits, and SIP Route Patterns route SIP URI destinations. When troubleshooting why a hunt group member is not receiving calls, the administrator should examine the Line Group membership and the associated hunt configuration.<\/span><\/p>\n

                                  Question 337. A UCM administrator wants incoming calls to a support number to ring a group of phones according to a defined distribution algorithm. Which combination is most appropriate?<\/b><\/p>\n

                                    \n
                                  1. <\/b> Route Pattern and Route Group<\/span><\/li>\n
                                  2. <\/b> Hunt Pilot, Hunt List, and Line Group<\/span><\/li>\n
                                  3. <\/b> Translation Pattern and SIP Profile<\/span><\/li>\n
                                  4. <\/b> Local Route Group and Device Mobility<\/span><\/li>\n<\/ol>\n

                                    Correct Answer: 2. Hunt Pilot, Hunt List, and Line Group<\/b><\/p>\n

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

                                    Cisco Unified Communications Manager uses Hunt Pilots, Hunt Lists, and Line Groups to implement call hunting. The Hunt Pilot represents the number callers dial, the Hunt List associates the pilot with one or more Line Groups, and the Line Group contains the directory numbers that receive calls. The selected distribution algorithm determines how calls are offered to the members. Route Patterns and Route Groups are intended for routing calls toward gateways or trunks, while Translation Patterns manipulate digits. Local Route Groups and Device Mobility solve different routing and location-related requirements. When designing or troubleshooting group call distribution, the relationship among these three hunt components is fundamental.<\/span><\/p>\n

                                    Question 338. A caller reaches a Hunt Pilot, but all available members are busy. The administrator wants the call to be redirected according to the configured hunting behavior instead of remaining indefinitely at the hunt pilot. Which configuration area should be reviewed?<\/b><\/p>\n

                                      \n
                                    1. <\/b> SIP session timer<\/span><\/li>\n
                                    2. <\/b> CUBE tenant<\/span><\/li>\n
                                    3. <\/b> Hunt Pilot and hunt-forward behavior<\/span><\/li>\n
                                    4. <\/b> Codec preference<\/span><\/li>\n<\/ol>\n

                                      Correct Answer: 3. Hunt Pilot and hunt-forward behavior<\/b><\/p>\n

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

                                      Hunt Pilot configuration determines how UCM handles calls directed toward a hunt group, including behavior when members are busy, unavailable, or do not answer. Depending on the design, calls can be forwarded or redirected after the configured hunting behavior is exhausted. Administrators should review the Hunt Pilot, Hunt List, Line Group, member availability, and forwarding settings when calls remain at the hunt pilot unexpectedly. SIP session timers relate to established SIP sessions, CUBE tenants provide trunk separation, and codec preferences affect media negotiation. Correct hunt configuration is therefore essential for predictable handling of calls when all group members are unavailable.<\/span><\/p>\n

                                      Question 339. A SIP trunk between UCM and another system is operational, but calls fail because the remote system expects a different SIP URI format. Which area should be examined first?<\/b><\/p>\n

                                        \n
                                      1. <\/b> URI normalization and SIP routing configuration<\/span><\/li>\n
                                      2. <\/b> SRST fallback<\/span><\/li>\n
                                      3. <\/b> Call Park retrieval timer<\/span><\/li>\n
                                      4. <\/b> Device Mobility user association<\/span><\/li>\n<\/ol>\n

                                        Correct Answer: 1. URI normalization and SIP routing configuration<\/b><\/p>\n

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

                                        SIP URI interoperability depends on the format used by each connected system and on how Cisco Unified Communications Manager matches and routes those URIs. If calls fail because the remote system expects a different URI structure, administrators should review URI normalization, SIP route patterns, domain configuration, and the SIP trunk’s routing behavior. These settings can transform or match URI information so that the destination is correctly recognized. SRST is for survivability, Call Park handles temporary call parking, and Device Mobility applies location-specific phone settings. SIP traces are useful for confirming the exact URI presented in the INVITE and determining where the expected format differs.<\/span><\/p>\n

                                        Question 340. A branch has a local PSTN gateway and a centralized UCM cluster. During normal operation, calls use the local gateway. During WAN failure, the branch must continue placing PSTN calls. Which design combines the required centralized and survivable functions?<\/b><\/p>\n

                                          \n
                                        1. <\/b> Global Dial Plan Replication with SIP OPTIONS<\/span><\/li>\n
                                        2. <\/b> Extension Mobility with CUBE tenants<\/span><\/li>\n
                                        3. <\/b> Local Route Group with SRST<\/span><\/li>\n
                                        4. <\/b> Hunt Pilot with Call Park<\/span><\/li>\n<\/ol>\n

                                          Correct Answer: 3. Local Route Group with SRST<\/b><\/p>\n

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

                                          Local Route Group can provide location-aware PSTN routing during normal centralized UCM operation, allowing a branch to use its local gateway while retaining common dial-plan structures. SRST then provides local call-processing survivability when connectivity to the centralized UCM cluster is lost. With appropriate gateway configuration, the branch can continue making PSTN calls during the outage. Global Dial Plan Replication is intended for distributing dial-plan information between UCM clusters, while SIP OPTIONS is a signaling mechanism. Extension Mobility, CUBE tenants, Hunt Pilots, and Call Park address different requirements. Combining centralized local-gateway routing with properly configured SRST provides a practical architecture for branch PSTN survivability.<\/span><\/p>\n

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

                                          View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps   Question 321. A Cisco Unified Communications Manager cluster uses multiple locations connected over a WAN. The administrator wants to limit the number of simultaneous calls between two locations based on available bandwidth. Which configuration is most directly involved? SIP Profile Call Admission […]<\/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\/23435"}],"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=23435"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23435\/revisions"}],"predecessor-version":[{"id":23436,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23435\/revisions\/23436"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=23435"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=23435"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=23435"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}