{"id":21294,"date":"2026-09-24T11:52:43","date_gmt":"2026-09-24T11:52:43","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=21294"},"modified":"2026-09-24T11:52:43","modified_gmt":"2026-09-24T11:52:43","slug":"cisco-ccnp-collaboration-350-801-practice-test-questions-and-exam-dumps-part20-q381-400","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-350-801-practice-test-questions-and-exam-dumps-part20-q381-400\/","title":{"rendered":"Cisco CCNP Collaboration 350-801 Practice Test Questions and Exam Dumps Part20 Q381-400"},"content":{"rendered":"

View Full <\/b>Cisco CCNP Collaboration 350-801 Exam Dumps<\/b><\/a> and Practice Test Dumps.<\/b><\/h2>\n

 <\/p>\n

Question 381<\/b><\/h3>\n

Which CUCM configuration controls access to specific dial-plan partitions?<\/b><\/p>\n

    \n
  1. Region<\/span><\/li>\n
  2. Calling Search Space<\/span><\/li>\n
  3. Device Pool<\/span><\/li>\n
  4. Route Group<\/span><\/li>\n<\/ol>\n

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

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

    A Calling Search Space, or CSS, determines which partitions a device, directory number, or other calling entity can access. CUCM uses the relationship between CSSs and partitions to control calling privileges. Administrators can create different CSSs for users who require different levels of dialing access. For example, one CSS can permit internal and local calls while another can provide access to long-distance or international destinations. Regions manage codec and bandwidth relationships, device pools provide common device settings, and route groups contain gateways or trunks. Proper CSS configuration is therefore an important part of implementing controlled and secure CUCM dial plans.<\/span><\/p>\n

    Question 382<\/b><\/h3>\n

    Which SIP method is used to initiate a new voice or video session?<\/b><\/p>\n

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

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

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

      The SIP INVITE method is used to initiate a new session, such as a voice or video call. The INVITE normally contains session information, often using SDP, which describes supported media types, codecs, addresses, and ports. The receiving endpoint processes the request and returns provisional or final SIP responses. If the call is accepted, a 200 OK response is normally followed by an ACK from the originating side. BYE terminates an established session, REGISTER provides endpoint registration information, and OPTIONS can be used to query capabilities. Understanding INVITE processing is essential when troubleshooting SIP call-establishment failures.<\/span><\/p>\n

      Question 383<\/b><\/h3>\n

      Which CUCM object can be used to manipulate dialed digits before routing continues?<\/b><\/p>\n

        \n
      1. Translation Pattern<\/span><\/li>\n
      2. Route Group<\/span><\/li>\n
      3. Device Pool<\/span><\/li>\n
      4. Region<\/span><\/li>\n<\/ol>\n

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

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

        A translation pattern allows CUCM to modify dialed digits before continuing call processing. Depending on its configuration, digits can be added, removed, or transformed to match a different numbering plan. Translation patterns are useful when integrating internal and external numbering schemes, normalizing telephone numbers, or adapting calls to service-provider requirements. Route groups contain gateways and trunks, device pools provide common endpoint configuration, and regions control codec and bandwidth relationships. Translation patterns are therefore an important part of CUCM dial-plan design. Administrators should carefully configure pattern matching and digit manipulation so that calls are transformed as intended without creating unexpected routing behavior.<\/span><\/p>\n

        Question 384<\/b><\/h3>\n

        Which protocol provides time synchronization across Cisco collaboration servers?<\/b><\/p>\n

          \n
        1. LDAP<\/span><\/li>\n
        2. DNS<\/span><\/li>\n
        3. RTP<\/span><\/li>\n
        4. NTP<\/span><\/li>\n<\/ol>\n

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

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

          Network Time Protocol, or NTP, provides time synchronization across networked systems. Cisco collaboration components depend on accurate time for logging, certificate validation, authentication, call records, and troubleshooting. Consistent time across CUCM, Unity Connection, IM and Presence, Expressway, and other systems makes it easier to correlate events and diagnose failures. LDAP provides directory services, DNS provides name resolution, and RTP carries real-time media. If collaboration servers have inconsistent clocks, administrators may encounter security or authentication problems and may have difficulty comparing logs. Reliable NTP sources should therefore be configured throughout the collaboration environment.<\/span><\/p>\n

          Question 385<\/b><\/h3>\n

          Which CUCM object contains gateways or trunks that can be selected for outbound call routing?<\/b><\/p>\n

            \n
          1. Route Group<\/span><\/li>\n
          2. Partition<\/span><\/li>\n
          3. CSS<\/span><\/li>\n
          4. Region<\/span><\/li>\n<\/ol>\n

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

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

            A route group is used to organize gateways or trunks that CUCM can select for call routing. Multiple gateways or trunks can be placed into a route group to provide redundancy or additional routing capacity. Route groups are commonly referenced by route lists, which can then be selected by route patterns. Partitions control the visibility of dial-plan objects, CSSs determine which partitions a caller can access, and regions control codec and bandwidth relationships. When troubleshooting an outbound call, administrators should review the complete routing chain, including the route pattern, route list, route group, and gateway or trunk availability, to determine why a particular path was selected or rejected.<\/span><\/p>\n

            Question 386<\/b><\/h3>\n

            Which protocol is responsible for transporting real-time audio and video between collaboration endpoints?<\/b><\/p>\n

              \n
            1. RTP<\/span><\/li>\n
            2. SIP<\/span><\/li>\n
            3. LDAP<\/span><\/li>\n
            4. DNS<\/span><\/li>\n<\/ol>\n

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

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

              Real-time Transport Protocol, or RTP, carries real-time audio and video media between collaboration endpoints. SIP generally handles signaling and session control, while RTP carries the actual media stream after the session has been established. RTP includes sequence numbers and timestamps that assist receivers with packet ordering and media timing. Media quality can be affected by packet loss, jitter, delay, and congestion. LDAP and DNS serve directory and name-resolution functions and do not normally transport voice media. In secure deployments, SRTP can be used instead of standard RTP to provide encryption and integrity protection for the media stream.<\/span><\/p>\n

              Question 387<\/b><\/h3>\n

              Which Cisco application provides enterprise voicemail services?<\/b><\/p>\n

                \n
              1. Cisco CUBE<\/span><\/li>\n
              2. Cisco Unity Connection<\/span><\/li>\n
              3. Cisco Expressway-E<\/span><\/li>\n
              4. Cisco IM and Presence<\/span><\/li>\n<\/ol>\n

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

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

                Cisco Unity Connection provides voicemail and related messaging functionality in Cisco collaboration environments. It integrates with CUCM so calls can be redirected to user mailboxes when calls are unanswered, busy, or forwarded. Unity Connection supports features such as personal greetings, voicemail access, message management, call handlers, and automated attendants depending on the configuration. CUBE is designed for SIP and voice border functions, Expressway-E provides edge collaboration services, and IM and Presence supports messaging and presence. When troubleshooting voicemail, administrators should verify CUCM integration, voicemail ports, call-routing behavior, Unity Connection service status, and network connectivity.<\/span><\/p>\n

                Question 388<\/b><\/h3>\n

                Which Cisco component is normally used as the external-facing element in a Mobile and Remote Access deployment?<\/b><\/p>\n

                  \n
                1. Expressway-C<\/span><\/li>\n
                2. CUCM<\/span><\/li>\n
                3. Expressway-E<\/span><\/li>\n
                4. Unity Connection<\/span><\/li>\n<\/ol>\n

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

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

                  Expressway-E is normally positioned at the external network edge in a Cisco Mobile and Remote Access deployment. It works with Expressway-C inside the enterprise to provide secure traversal for supported collaboration clients connecting from outside the corporate network. Expressway-E handles the external-facing portion of the architecture, while Expressway-C provides internal connectivity to collaboration services. CUCM remains responsible for call processing, and Unity Connection provides voicemail. MRA deployments require appropriate DNS records, certificates, traversal configuration, authentication, and firewall rules. Correct Expressway-E placement and configuration are important for providing remote collaboration access while maintaining appropriate network security controls.<\/span><\/p>\n

                  Question 389<\/b><\/h3>\n

                  Which SIP response code indicates that a called endpoint is busy?<\/b><\/p>\n

                    \n
                  1. 404<\/span><\/li>\n
                  2. 200<\/span><\/li>\n
                  3. 503<\/span><\/li>\n
                  4. 486<\/span><\/li>\n<\/ol>\n

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

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

                    SIP response code 486 Busy Here indicates that the destination endpoint is currently busy and cannot accept the call. Depending on the configured call-routing behavior, CUCM or another SIP system may forward the call to voicemail, another destination, or apply a different busy treatment. A 404 response indicates that the destination could not be found, 200 indicates successful processing, and 503 indicates that a service is unavailable. SIP response codes are useful during troubleshooting because they provide information about how the receiving system handled a request. Engineers can inspect signaling traces to determine which system generated the response and how the call proceeded afterward.<\/span><\/p>\n

                    Question 390<\/b><\/h3>\n

                    Which CUCM configuration determines which media resources are available to a device?<\/b><\/p>\n

                      \n
                    1. Media Resource Group List<\/span><\/li>\n
                    2. Route Pattern<\/span><\/li>\n
                    3. Partition<\/span><\/li>\n
                    4. Calling Search Space<\/span><\/li>\n<\/ol>\n

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

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

                      A Media Resource Group List, or MRGL, determines which media resources are available to a device or group of devices. MRGLs reference Media Resource Groups that can contain conference bridges, transcoders, annunciators, and other supported resources. This allows administrators to control media-resource access based on device location, type, or operational requirements. Route patterns are used for digit-based routing, partitions organize dial-plan objects, and CSSs control access to dial-plan partitions. If a device cannot access a required conference bridge or transcoder, administrators should verify the assigned MRGL and confirm that the referenced resources are operational and correctly configured.<\/span><\/p>\n

                      Question 391<\/b><\/h3>\n

                      Which Cisco technology provides basic call-processing services when a remote site loses connectivity to CUCM?<\/b><\/p>\n

                        \n
                      1. CUBE<\/span><\/li>\n
                      2. SRST<\/span><\/li>\n
                      3. MRA<\/span><\/li>\n
                      4. LDAP<\/span><\/li>\n<\/ol>\n

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

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

                        Survivable Remote Site Telephony, or SRST, provides limited call-processing functionality when a remote site’s phones lose connectivity to CUCM. A supported branch router can allow phones to register locally and maintain basic calling capabilities during a WAN outage. This provides an important level of voice survivability until normal connectivity is restored. CUBE is used for SIP border and interoperability functions, MRA provides remote client access, and LDAP provides directory services. SRST does not reproduce the complete feature set of CUCM. Administrators should configure SRST references appropriately and verify that the branch router, endpoints, and network support the required survivability features.<\/span><\/p>\n

                        Question 392<\/b><\/h3>\n

                        Which CUCM component provides common configuration settings for multiple endpoints?<\/b><\/p>\n

                          \n
                        1. Device Pool<\/span><\/li>\n
                        2. Route Pattern<\/span><\/li>\n
                        3. Translation Pattern<\/span><\/li>\n
                        4. Partition<\/span><\/li>\n<\/ol>\n

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

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

                          A device pool provides common configuration settings that can be applied to multiple Cisco collaboration endpoints. Depending on the deployment, device-pool settings can include CUCM groups, regions, date\/time groups, SRST references, and other operational parameters. Using device pools simplifies administration because administrators can assign common settings to groups of devices instead of configuring each phone individually. Route patterns and translation patterns are dial-plan objects, while partitions organize dial-plan objects for access control. Device pools are particularly useful in multi-site deployments where phones at different locations need different call-processing servers, regional settings, time zones, or survivability configurations.<\/span><\/p>\n

                          Question 393<\/b><\/h3>\n

                          Which protocol can protect SIP signaling through encryption and integrity?<\/b><\/p>\n

                            \n
                          1. SRTP<\/span><\/li>\n
                          2. RTP<\/span><\/li>\n
                          3. TLS<\/span><\/li>\n
                          4. DHCP<\/span><\/li>\n<\/ol>\n

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

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

                            Transport Layer Security, or TLS, can protect SIP signaling by providing encryption and integrity for supported communication sessions. TLS can also use digital certificates to establish trust between participating collaboration systems. In secure Cisco collaboration deployments, TLS is commonly used for signaling while SRTP protects the actual audio or video media. RTP carries media but does not provide the same security features, and DHCP provides network configuration. When troubleshooting secure SIP communication, administrators should verify certificate trust, certificate validity, supported TLS versions, and consistent security configuration on both sides of the connection. Correct signaling security is important for protecting call-control information.<\/span><\/p>\n

                            Question 394<\/b><\/h3>\n

                            Which CUCM configuration controls codec preferences between two groups of endpoints?<\/b><\/p>\n

                              \n
                            1. CSS<\/span><\/li>\n
                            2. Partition<\/span><\/li>\n
                            3. Route Group<\/span><\/li>\n
                            4. Region<\/span><\/li>\n<\/ol>\n

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

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

                              CUCM regions control codec preferences and bandwidth relationships between groups of endpoints. Administrators can define which codecs should be used for calls within a region and between different regions. This is especially useful in distributed collaboration environments where WAN links have limited bandwidth. For example, a high-bandwidth codec may be suitable for local calls while a more bandwidth-efficient codec may be preferred between remote sites. CSS and partitions control dial-plan access, while route groups contain gateways and trunks. Proper region configuration helps administrators balance media quality with network capacity and can reduce unnecessary bandwidth consumption across constrained links.<\/span><\/p>\n

                              Question 395<\/b><\/h3>\n

                              Which Cisco component provides SIP interoperability and border functions between an enterprise and an external voice network?<\/b><\/p>\n

                                \n
                              1. CUBE<\/span><\/li>\n
                              2. Unity Connection<\/span><\/li>\n
                              3. IM and Presence<\/span><\/li>\n
                              4. SRST<\/span><\/li>\n<\/ol>\n

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

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

                                Cisco Unified Border Element, or CUBE, is commonly deployed between an enterprise collaboration environment and an external SIP service provider or voice network. CUBE can provide SIP normalization, interoperability, topology hiding, media handling, and security functions. It can also use dial peers to determine how calls should be processed and routed. Unity Connection provides voicemail, IM and Presence provides messaging and presence, and SRST provides limited call processing during WAN failures. CUBE is particularly useful when the enterprise and service provider use different SIP behaviors or when the organization requires a controlled demarcation point between internal and external voice networks.<\/span><\/p>\n

                                Question 396<\/b><\/h3>\n

                                Which SIP method is used by an endpoint to inform a registrar about its current contact address?<\/b><\/p>\n

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

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

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

                                  The SIP REGISTER method allows an endpoint to provide its current contact information to a SIP registrar. The registrar uses this information so that subsequent requests for the user can be directed to the correct endpoint location. Registration normally includes contact information and an expiration interval. INVITE is used to establish or modify a session, BYE terminates an established session, and OPTIONS can be used to query capabilities or availability. When investigating SIP registration problems, administrators should check network connectivity, authentication, DNS, certificates when secure signaling is involved, and the configuration and availability of the relevant registrar or call-control system.<\/span><\/p>\n

                                  Question 397<\/b><\/h3>\n

                                  Which media resource is required when two endpoints use incompatible audio codecs?<\/b><\/p>\n

                                    \n
                                  1. Conference Bridge<\/span><\/li>\n
                                  2. Transcoder<\/span><\/li>\n
                                  3. Annunciator<\/span><\/li>\n
                                  4. Music on Hold<\/span><\/li>\n<\/ol>\n

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

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

                                    A transcoder provides media conversion between different codecs when endpoints cannot directly negotiate a compatible codec. For example, one side of a call may support G.711 while another network requires G.729. CUCM can invoke a configured transcoder when the call requires codec conversion. A conference bridge provides multi-party conferencing, an annunciator provides prerecorded audio prompts, and Music on Hold provides audio to callers who are placed on hold. Transcoding consumes processing resources, so administrators should consider expected call volume and codec requirements when sizing media resources. Proper MRGL configuration also ensures that the required transcoder is available to devices that need it.<\/span><\/p>\n

                                    Question 398<\/b><\/h3>\n

                                    Which protocol provides directory access for integration with services such as Microsoft Active Directory?<\/b><\/p>\n

                                      \n
                                    1. NTP<\/span><\/li>\n
                                    2. RTP<\/span><\/li>\n
                                    3. DNS<\/span><\/li>\n
                                    4. LDAP<\/span><\/li>\n<\/ol>\n

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

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

                                      LDAP, or Lightweight Directory Access Protocol, provides a standard method for accessing directory information from systems such as Microsoft Active Directory. Cisco collaboration systems can use LDAP integration to synchronize or access user information depending on the configured architecture. This can centralize identity information and reduce duplicate user administration. LDAP does not carry media or provide time synchronization. RTP carries real-time audio and video, DNS provides name resolution, and NTP synchronizes system clocks. When troubleshooting directory integration, administrators should verify connectivity, authentication credentials, search bases, directory attributes, synchronization configuration, and the availability of the LDAP service.<\/span><\/p>\n

                                      Question 399<\/b><\/h3>\n

                                      Which protocol is commonly used to provide secure real-time media for voice and video?<\/b><\/p>\n

                                        \n
                                      1. SRTP<\/span><\/li>\n
                                      2. SIP<\/span><\/li>\n
                                      3. TLS<\/span><\/li>\n
                                      4. LDAP<\/span><\/li>\n<\/ol>\n

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

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

                                        Secure Real-time Transport Protocol, or SRTP, protects RTP-based audio and video media. It can provide encryption and integrity protection so that unauthorized parties cannot easily intercept or modify collaboration media. SRTP is different from TLS, which is commonly used to protect signaling and other application-layer communications. SIP provides session signaling, while LDAP provides directory services. Secure collaboration deployments may use TLS and SRTP together, protecting both signaling and media. When troubleshooting secure media, administrators should verify that endpoints support the required security mechanisms and that key-management, certificate, and media-security settings are compatible across the devices participating in the session.<\/span><\/p>\n

                                        Question 400<\/b><\/h3>\n

                                        Which protocol is used to resolve hostnames and service names in Cisco collaboration networks?<\/b><\/p>\n

                                          \n
                                        1. NTP<\/span><\/li>\n
                                        2. DNS<\/span><\/li>\n
                                        3. DHCP<\/span><\/li>\n
                                        4. RTP<\/span><\/li>\n<\/ol>\n

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

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

                                          The Domain Name System, or DNS, resolves hostnames and service names to IP addresses and other DNS records. Cisco collaboration components depend on DNS for functions such as server discovery, SIP routing, service resolution, and Expressway-related deployments. DNS records must be correctly configured and accessible from the network locations where they are required. NTP synchronizes clocks, DHCP provides network configuration, and RTP transports real-time media. DNS problems can prevent collaboration systems from locating required servers even when basic IP connectivity is available. Administrators should verify forward and reverse resolution, SRV records where applicable, and consistent DNS responses during troubleshooting.<\/span><\/p>\n

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

                                          View Full Cisco CCNP Collaboration 350-801 Exam Dumps and Practice Test Dumps.   Question 381 Which CUCM configuration controls access to specific dial-plan partitions? Region Calling Search Space Device Pool Route Group Correct Answer: 2 Explanation A Calling Search Space, or CSS, determines which partitions a device, directory number, or other calling entity can access. […]<\/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\/21294"}],"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=21294"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/21294\/revisions"}],"predecessor-version":[{"id":21295,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/21294\/revisions\/21295"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=21294"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=21294"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=21294"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}