{"id":23425,"date":"2026-09-28T06:30:11","date_gmt":"2026-09-28T06:30:11","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=23425"},"modified":"2026-09-28T06:30:11","modified_gmt":"2026-09-28T06:30:11","slug":"cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-12-q221-240","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-12-q221-240\/","title":{"rendered":"Cisco CCNP Collaboration 300-815 Practice Test Questions and Exam Dumps Part 12 Q221-240"},"content":{"rendered":"

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

 <\/p>\n

Question 221. A Cisco Unified Communications Manager administrator needs to route calls matching a specific E.164 destination through a dedicated SIP trunk. Which dial-plan configuration should be used?<\/b><\/p>\n

    \n
  1. <\/b> Device Mobility group<\/span><\/li>\n
  2. <\/b> Location<\/span><\/li>\n
  3. <\/b> Route pattern associated with the appropriate route list<\/span><\/li>\n
  4. <\/b> Region relationship<\/span><\/li>\n<\/ol>\n

    Correct Answer: 3. Route pattern associated with the appropriate route list<\/b><\/p>\n

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

    A route pattern defines the destination digits that Cisco Unified Communications Manager uses for call routing. The route pattern can be associated with a route list, which then points to the appropriate route group and SIP trunk or gateway. This allows a specific E.164 destination range to use a dedicated trunk without changing unrelated destinations. Device Mobility controls location-specific device settings, locations provide CAC resources, and regions primarily influence codec selection. Therefore, a route pattern associated with the appropriate route list provides the required destination-specific routing behavior.<\/span><\/p>\n

    Question 222. A SIP trunk is established, but the provider requires the enterprise to present a different value in the P-Asserted-Identity header. Which feature is most appropriate for modifying this SIP header?<\/b><\/p>\n

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

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

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

      SIP profiles can modify supported SIP signaling elements and headers to accommodate interoperability requirements. When a provider requires a particular P-Asserted-Identity format, the administrator can investigate the applicable SIP profile or normalization configuration for the SIP trunk or dial peer. Location CAC controls bandwidth admission, route groups control gateway or trunk selection, and device pools provide common configuration settings. Header manipulation is therefore a signaling-level requirement rather than a routing or bandwidth function. The exact SIP profile configuration should be validated against the provider’s required header format and the platform’s supported normalization capabilities.<\/span><\/p>\n

      Question 223. A user moves a Cisco IP phone from the headquarters network to a branch office. The phone should automatically receive the branch’s calling privileges and media settings. Which feature is designed for this scenario?<\/b><\/p>\n

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

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

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

        Device Mobility allows Cisco Unified Communications Manager to apply different device settings when a supported phone operates from a different physical or configured location. Depending on the deployment, these settings can include calling search spaces, regions, locations, and other location-specific parameters. Translation patterns manipulate digits, route groups determine gateway or trunk selection, and SIP session timers manage session refresh behavior. Therefore, Device Mobility is the feature intended to simplify phone operation when devices move between locations and require different communication policies.<\/span><\/p>\n

        Question 224. A company has a 2-Mbps WAN link between two collaboration sites. The administrator wants intersite calls to use a codec that consumes less bandwidth than the codec used for calls within each site. Where should the codec relationship be configured?<\/b><\/p>\n

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

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

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

          Cisco Unified Communications Manager regions define codec relationships between devices in different regions. An administrator can configure a lower-bandwidth codec for calls crossing a constrained WAN while retaining a different codec policy for calls within a site. Calling Search Spaces determine partition access, route lists determine routing paths, and translation patterns manipulate dialed digits. Therefore, region relationships are the appropriate configuration area for controlling codec selection according to the network path and available bandwidth.<\/span><\/p>\n

          Question 225. A branch router must continue providing call processing when its WAN connection to Cisco Unified Communications Manager fails. Which technology provides this local survivability?<\/b><\/p>\n

            \n
          1. <\/b> Cisco Unified SRST<\/span><\/li>\n
          2. <\/b> Device Mobility<\/span><\/li>\n
          3. <\/b> SIP normalization<\/span><\/li>\n
          4. <\/b> Location CAC<\/span><\/li>\n<\/ol>\n

            Correct Answer: 4. Cisco Unified SRST<\/b><\/p>\n

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

            Cisco Unified Survivable Remote Site Telephony (SRST) provides local call-processing capabilities when phones at a branch cannot reach their centralized Cisco Unified Communications Manager. The branch router can assume survivable call-processing functions and use locally configured dial peers and voice connectivity for external calls. Device Mobility addresses location-specific device settings, SIP normalization modifies signaling, and Location CAC controls bandwidth admission. SRST is therefore the technology specifically designed to maintain essential telephony services during loss of connectivity to centralized call control.<\/span><\/p>\n

            Question 226. A SIP endpoint receives a 183 Session Progress response containing SDP and begins receiving an announcement before the called party answers. Which SIP capability is being used?<\/b><\/p>\n

              \n
            1. <\/b> Registration<\/span><\/li>\n
            2. <\/b> Early media<\/span><\/li>\n
            3. <\/b> Call termination<\/span><\/li>\n
            4. <\/b> Digest authentication<\/span><\/li>\n<\/ol>\n

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

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

              A 183 Session Progress response can carry SDP describing media parameters, allowing media to be established before the final answer to the call. This mechanism is commonly used for early media, such as provider announcements, network tones, or other pre-answer audio. REGISTER is used for endpoint registration, BYE terminates an established dialog, and SIP digest authentication protects signaling exchanges requiring authentication. The scenario specifically describes media being delivered before the call is answered, which is characteristic of early media associated with provisional SIP signaling and SDP.<\/span><\/p>\n

              Question 227. A CUBE router has separate interfaces toward the enterprise and the SIP provider. Calls establish, but the provider receives an SDP address belonging to the enterprise-side network. Which configuration should be investigated?<\/b><\/p>\n

                \n
              1. <\/b> Calling Search Space<\/span><\/li>\n
              2. <\/b> Translation pattern<\/span><\/li>\n
              3. <\/b> CUBE media interface or address binding<\/span><\/li>\n
              4. <\/b> Route group<\/span><\/li>\n<\/ol>\n

                Correct Answer: 3. CUBE media interface or address binding<\/b><\/p>\n

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

                CUBE deployments with multiple interfaces require appropriate media addressing so that the SDP advertised toward each network contains a reachable address for the intended media path. If the provider receives an enterprise-side address, RTP may be directed to an unreachable or private address. The administrator should inspect CUBE media interface or address binding configuration and verify the resulting SDP. Calling Search Spaces and translation patterns concern dial-plan behavior, while route groups control gateway or trunk selection. Correct media binding is therefore essential when CUBE connects networks with different addressing domains.<\/span><\/p>\n

                Question 228. An enterprise uses different SIP providers for calls to different domains. CUBE should select the provider based on the domain in the destination SIP URI. Which configuration supports this design?<\/b><\/p>\n

                  \n
                1. <\/b> Location CAC<\/span><\/li>\n
                2. <\/b> URI-based dial-peer matching<\/span><\/li>\n
                3. <\/b> Device pool<\/span><\/li>\n
                4. <\/b> Region configuration<\/span><\/li>\n<\/ol>\n

                  Correct Answer: 2. URI-based dial-peer matching<\/b><\/p>\n

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

                  URI-based dial-peer matching allows CUBE to use SIP URI information, including the destination domain, when selecting an outbound dial peer. This is useful when different domains must be routed to different SIP providers or other destinations. Location CAC controls bandwidth admission, device pools provide common device settings, and region configuration controls codec relationships. Therefore, URI-based matching provides the routing criterion required when the SIP domain, rather than only traditional dialed digits, determines which provider should receive the call.<\/span><\/p>\n

                  Question 229. A SIP provider requires an outbound INVITE to contain a modified From header. Which CUBE capability should be investigated?<\/b><\/p>\n

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

                    Correct Answer: 1. SIP profile or normalization<\/b><\/p>\n

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

                    SIP profiles and normalization mechanisms can manipulate SIP signaling to meet interoperability requirements between CUBE and external SIP peers. If a provider requires a modified From header, the administrator should investigate the appropriate SIP message manipulation feature and apply it to the relevant call path. Location CAC controls bandwidth, Device Mobility handles location-specific device settings, and route groups select gateways or trunks. Since the requirement specifically concerns changing SIP header information in an INVITE, a signaling normalization mechanism is the appropriate area to investigate.<\/span><\/p>\n

                    Question 230. A SIP call repeatedly terminates after the negotiated session interval because neither endpoint refreshes the session successfully. Which feature should be examined?<\/b><\/p>\n

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

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

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

                      SIP session timers establish a negotiated interval during which a session must be refreshed. If the required refresh transaction does not occur successfully, the session can eventually be terminated. A predictable disconnect that corresponds to the negotiated session interval is therefore a strong indication that session-timer behavior should be investigated. Translation patterns manipulate digits, route groups determine routing paths, and Device Mobility applies location-specific device settings. The administrator should examine Session-Expires, Min-SE, refresher behavior, and the corresponding refresh transactions when troubleshooting this type of call termination.<\/span><\/p>\n

                      Question 231. A user can access local route patterns but cannot access a PSTN route pattern located in another partition. Which configuration determines whether the user’s device can search that partition?<\/b><\/p>\n

                        \n
                      1. <\/b> Region<\/span><\/li>\n
                      2. <\/b> Device pool<\/span><\/li>\n
                      3. <\/b> Calling Search Space<\/span><\/li>\n
                      4. <\/b> Location<\/span><\/li>\n<\/ol>\n

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

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

                        The actual mechanism that determines which partitions a device or line can search is the Calling Search Space (CSS). A CSS contains an ordered list of partitions and therefore controls access to route patterns and other dial-plan objects. A device pool can provide common configuration and can be associated with settings that influence devices, but it is not itself the mechanism that defines partition search access. Regions control codec relationships, and locations support CAC. Therefore, when a user cannot access a route pattern because of partition visibility, the effective CSS should be examined.<\/span><\/p>\n

                        Question 232. CUBE receives an INVITE from an unauthorized Internet address. The administrator wants to prevent such requests from being processed as legitimate calls. Which security approach is most appropriate?<\/b><\/p>\n

                          \n
                        1. <\/b> Change the codec preference<\/span><\/li>\n
                        2. <\/b> Configure source-based access controls and trusted peers<\/span><\/li>\n
                        3. <\/b> Increase the session timer<\/span><\/li>\n
                        4. <\/b> Modify the route-group algorithm<\/span><\/li>\n<\/ol>\n

                          Correct Answer: 3. Configure source-based access controls and trusted peers<\/b><\/p>\n

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

                          CUBE security can use source-based controls and trusted-peer policies to restrict which SIP signaling sources are accepted. This is especially important when a CUBE interface is exposed to external networks because unauthorized SIP traffic can create security and toll-fraud risks. Codec preferences do not authenticate signaling sources, session timers do not prevent unauthorized requests, and route-group algorithms only affect routing among configured destinations. Therefore, source validation and trusted-peer controls should be established before allowing external SIP traffic to enter the enterprise call-routing environment.<\/span><\/p>\n

                          Question 233. A SIP endpoint is registered successfully, but a call attempt receives a 401 response followed by another request containing authentication information. What SIP mechanism is being demonstrated?<\/b><\/p>\n

                            \n
                          1. <\/b> SIP digest authentication<\/span><\/li>\n
                          2. <\/b> DTMF relay<\/span><\/li>\n
                          3. <\/b> Early media<\/span><\/li>\n
                          4. <\/b> Session termination<\/span><\/li>\n<\/ol>\n

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

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

                            SIP commonly uses a challenge-response process for digest authentication. A server can respond with 401 Unauthorized and provide an authentication challenge. The client then sends another request containing the appropriate authentication response. This process allows the server to verify the endpoint’s credentials without simply transmitting the password in clear text. DTMF relay concerns keypad digits, early media concerns media before call establishment, and session termination is normally associated with BYE. Therefore, a 401 challenge followed by an authenticated request is characteristic of SIP digest authentication.<\/span><\/p>\n

                            Question 234. A CUBE administrator needs to convert an incoming national-format telephone number into an E.164 number before routing it into the enterprise. Which feature should be used?<\/b><\/p>\n

                              \n
                            1. <\/b> Region configuration<\/span><\/li>\n
                            2. <\/b> SIP session timer<\/span><\/li>\n
                            3. <\/b> Translation rule and translation profile<\/span><\/li>\n
                            4. <\/b> Device Mobility<\/span><\/li>\n<\/ol>\n

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

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

                              The intended feature for converting one telephone-number format into another is a translation rule applied through a translation profile. Translation rules can add, remove, or replace digits, while translation profiles apply those rules to the appropriate call-processing direction. A SIP session timer has no role in digit manipulation. Region configuration controls codec relationships, and Device Mobility provides location-specific device settings. Therefore, the appropriate solution for converting a national-format number into E.164 is a translation rule and associated translation profile.<\/span><\/p>\n

                              Question 235. An administrator needs to restrict a group of users from dialing international destinations while allowing them to reach internal and local PSTN destinations. Which design is appropriate?<\/b><\/p>\n

                                \n
                              1. <\/b> Put international route patterns in a partition excluded from the users’ CSS<\/span><\/li>\n
                              2. <\/b> Change the users’ codec preference<\/span><\/li>\n
                              3. <\/b> Configure a different SIP session timer<\/span><\/li>\n
                              4. <\/b> Add the users to a different route group<\/span><\/li>\n<\/ol>\n

                                Correct Answer: 1. Put international route patterns in a partition excluded from the users’ CSS<\/b><\/p>\n

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

                                Partitions and Calling Search Spaces provide the access-control structure needed for this dial-plan requirement. International route patterns can be placed in a dedicated partition, while the affected users receive a CSS that does not include that partition. Internal and local PSTN route patterns can remain in partitions accessible to those users. Codec preferences and SIP session timers do not control destination access, and route groups determine gateway selection after a route has been selected. This design therefore restricts international dialing without preventing permitted internal or local calls.<\/span><\/p>\n

                                Question 236. A SIP trunk between two systems requires encrypted signaling and encrypted media. Which combination should be configured where supported?<\/b><\/p>\n

                                  \n
                                1. <\/b> OPTIONS and REGISTER<\/span><\/li>\n
                                2. <\/b> UDP and RTP<\/span><\/li>\n
                                3. <\/b> TLS and SRTP<\/span><\/li>\n
                                4. <\/b> REFER and PRACK<\/span><\/li>\n<\/ol>\n

                                  Correct Answer: 3. TLS and SRTP<\/b><\/p>\n

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

                                  TLS provides encryption and integrity protection for SIP signaling, while SRTP provides protection for real-time media such as RTP voice streams. Together, they address the two principal communication planes in a secured SIP call. OPTIONS, REGISTER, REFER, and PRACK are SIP methods and do not themselves provide encryption. Plain UDP and RTP do not provide equivalent cryptographic protection. Therefore, when a SIP deployment requires protected signaling and protected media, TLS and SRTP are the appropriate security mechanisms, assuming both communicating systems support and are correctly configured for them.<\/span><\/p>\n

                                  Question 237. A branch uses SRST during a WAN outage. Internal extensions can call each other, but PSTN calls fail. Which configuration should be checked first?<\/b><\/p>\n

                                    \n
                                  1. <\/b> Centralized route list only<\/span><\/li>\n
                                  2. <\/b> Local SRST dial peers and PSTN gateway connectivity<\/span><\/li>\n
                                  3. <\/b> Device Mobility roaming settings<\/span><\/li>\n
                                  4. <\/b> Centralized region relationships<\/span><\/li>\n<\/ol>\n

                                    Correct Answer: 2. Local SRST dial peers and PSTN gateway connectivity<\/b><\/p>\n

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

                                    During SRST operation, the branch router provides local survivable call processing, so external PSTN calls depend on the local voice-routing configuration. The administrator should verify that appropriate SRST dial peers exist, that their destination patterns and preferences are correct, and that the local PSTN gateway or trunk is reachable and operational. Centralized route lists and regions may not be available or may not control the local survivable call path during a WAN outage. Device Mobility is also unrelated to PSTN gateway reachability. Local SRST routing should therefore be investigated first.<\/span><\/p>\n

                                    Question 238. A SIP call is established, but one endpoint cannot hear the other. Signaling succeeds in both directions. Which information should be examined first to troubleshoot the media path?<\/b><\/p>\n

                                      \n
                                    1. <\/b> SDP media addresses and RTP reachability<\/span><\/li>\n
                                    2. <\/b> Calling Search Space<\/span><\/li>\n
                                    3. <\/b> Route-pattern partition<\/span><\/li>\n
                                    4. <\/b> Device pool name<\/span><\/li>\n<\/ol>\n

                                      Correct Answer: 4. SDP media addresses and RTP reachability<\/b><\/p>\n

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

                                      When SIP signaling succeeds but audio is missing or one-way, the administrator should examine the SDP exchanged by the endpoints or CUBE and verify the advertised media addresses and ports. The next step is to confirm that RTP packets can actually reach those addresses through firewalls, NAT devices, and routing infrastructure. Calling Search Spaces and route-pattern partitions affect call routing rather than established RTP media, while the device pool name does not directly identify the media path. SDP and RTP analysis therefore provides the most relevant starting point for troubleshooting one-way or missing audio.<\/span><\/p>\n

                                      Question 239. A SIP endpoint needs to transfer an established call by instructing another SIP device to take responsibility for the call. Which SIP method is commonly associated with this operation?<\/b><\/p>\n

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

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

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

                                        The SIP REFER method is commonly used to request that another SIP endpoint initiate a new communication action on behalf of the referring endpoint. One major use is call transfer, where a device can refer the remote party toward another destination. OPTIONS is generally used for capability or reachability queries, REGISTER establishes or refreshes endpoint registration, and CANCEL terminates a pending request. Therefore, when an established SIP call needs to be transferred using a SIP referral mechanism, REFER is the method commonly associated with that operation.<\/span><\/p>\n

                                        Question 240. A SIP endpoint sends a REGISTER request but receives no successful registration response. The administrator wants to determine whether the SIP server is reachable before troubleshooting credentials. Which test is useful?<\/b><\/p>\n

                                          \n
                                        1. <\/b> Send a BYE request<\/span><\/li>\n
                                        2. <\/b> Change the route group<\/span><\/li>\n
                                        3. <\/b> Modify the region codec<\/span><\/li>\n
                                        4. <\/b> Use SIP OPTIONS to test reachability and capability<\/span><\/li>\n<\/ol>\n

                                          Correct Answer: 1. Send a BYE request<\/b><\/p>\n

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

                                          The appropriate SIP method for checking whether a SIP service is reachable and responsive without establishing a normal call is OPTIONS. An OPTIONS request can elicit a response describing supported capabilities and can help determine whether the signaling path is functioning. BYE is used to terminate an established SIP dialog, route groups control gateway selection, and region codec settings influence media characteristics. Therefore, when troubleshooting a registration failure and wanting to separate basic reachability problems from authentication issues, SIP OPTIONS can provide useful initial evidence about the remote service’s availability.<\/span><\/p>\n

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

                                          View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps   Question 221. A Cisco Unified Communications Manager administrator needs to route calls matching a specific E.164 destination through a dedicated SIP trunk. Which dial-plan configuration should be used? Device Mobility group Location Route pattern associated with the appropriate route list Region relationship […]<\/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\/23425"}],"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=23425"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23425\/revisions"}],"predecessor-version":[{"id":23426,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23425\/revisions\/23426"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=23425"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=23425"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=23425"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}