{"id":23433,"date":"2026-09-28T06:38:09","date_gmt":"2026-09-28T06:38:09","guid":{"rendered":"https:\/\/www.examlabs.com\/certification\/?p=23433"},"modified":"2026-09-28T06:38:09","modified_gmt":"2026-09-28T06:38:09","slug":"cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-16-q301-320","status":"publish","type":"post","link":"https:\/\/www.examlabs.com\/certification\/cisco-ccnp-collaboration-300-815-practice-test-questions-and-exam-dumps-part-16-q301-320\/","title":{"rendered":"Cisco CCNP Collaboration 300-815 Practice Test Questions and Exam Dumps Part 16 Q301-320"},"content":{"rendered":"

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

 <\/p>\n

Question 301. A Cisco Unified Communications Manager administrator needs to route calls to a PSTN gateway only when the dialed number matches a specific pattern. Which UCM dial-plan element provides the pattern matching used to select a route?<\/b><\/p>\n

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

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

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

    A Route Pattern in Cisco Unified Communications Manager defines the dialed-number pattern that UCM uses to determine how a call should be routed. The pattern can identify destinations such as PSTN numbers, service numbers, or other external destinations. Once a call matches the route pattern, UCM uses the associated route list or gateway routing configuration to determine the actual outbound path. A Route Group contains gateways or trunks but does not perform the initial digit-pattern match. Hunt Pilots distribute calls among hunt members, while SIP Profiles modify selected SIP behavior. Therefore, when the requirement is to match a specific dialed-number pattern and send the call toward an external route, a Route Pattern is the appropriate dial-plan element.<\/span><\/p>\n

    Question 302. A user dials an external number, but UCM must remove the access code before sending the call to the PSTN gateway. Which configuration is commonly used to perform this digit manipulation?<\/b><\/p>\n

      \n
    1. <\/b> Digit manipulation using route-pattern settings<\/span><\/li>\n
    2. <\/b> SIP registration<\/span><\/li>\n
    3. <\/b> Call Admission Control<\/span><\/li>\n
    4. <\/b> Device Mobility<\/span><\/li>\n<\/ol>\n

      Correct Answer: 1. Digit manipulation using route-pattern settings<\/b><\/p>\n

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

      Cisco Unified Communications Manager can manipulate digits associated with a matched route pattern before the call is sent toward the selected route. A common example is removing an access code, such as an outside-line prefix, so that the PSTN gateway receives the number in the format expected by the carrier. Digit manipulation can include stripping digits, adding digits, or modifying the called-number presentation according to the configured routing design. SIP registration establishes endpoint registration and does not manipulate dialed digits. Call Admission Control manages network-resource usage, while Device Mobility adjusts device behavior according to location. Correct digit manipulation is therefore essential for interoperability between an enterprise dial plan and external numbering requirements.<\/span><\/p>\n

      Question 303. An organization wants users at different locations to dial a consistent globalized number format while UCM determines the appropriate local or external route. Which concept is central to this design?<\/b><\/p>\n

        \n
      1. <\/b> Globalized number normalization<\/span><\/li>\n
      2. <\/b> RTP payload negotiation<\/span><\/li>\n
      3. <\/b> SIP OPTIONS<\/span><\/li>\n
      4. <\/b> Media termination<\/span><\/li>\n<\/ol>\n

        Correct Answer: 1. Globalized number normalization<\/b><\/p>\n

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

        Globalized number normalization provides a consistent representation of telephone numbers across the collaboration environment. Cisco Unified Communications Manager can normalize numbers into a standardized format, often using +E.164-style addressing, while routing logic determines how the destination should be reached. This approach simplifies dial-plan administration across multiple sites and reduces dependency on location-specific dialing habits. RTP payload negotiation concerns media formats, SIP OPTIONS is used for signaling capability or availability checks, and media termination involves processing or anchoring media streams. When an enterprise wants users and systems to work with consistent global numbers regardless of location, number normalization is an important component of the overall dial-plan design.<\/span><\/p>\n

        Question 304. A Cisco UCM administrator wants calls from a particular site to use that site’s local PSTN gateway whenever possible, while maintaining a common route pattern across sites. Which feature supports this routing model?<\/b><\/p>\n

          \n
        1. <\/b> SIP OPTIONS<\/span><\/li>\n
        2. <\/b> Local Route Group<\/span><\/li>\n
        3. <\/b> Call Park<\/span><\/li>\n
        4. <\/b> Extension Mobility<\/span><\/li>\n<\/ol>\n

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

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

          Local Route Group allows Cisco Unified Communications Manager to use a locally defined route group while maintaining a common route pattern or route list structure across multiple locations. This can simplify dial-plan administration because the same routing logic can be associated with different local gateways depending on the originating device’s location or configuration. It is particularly useful in multisite deployments where users should access their local PSTN gateway rather than a centralized gateway. SIP OPTIONS is a signaling method, Call Park temporarily holds calls for retrieval, and Extension Mobility manages user profiles on phones. Local Route Group therefore directly addresses the requirement for location-aware PSTN routing.<\/span><\/p>\n

          Question 305. A company has several PSTN gateways and wants UCM to distribute calls among them according to a defined order while keeping the gateways grouped logically. Which configuration component provides this grouping?<\/b><\/p>\n

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

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

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

            A Route Group in Cisco Unified Communications Manager contains gateways or trunks that can be selected as part of a routing strategy. Route Groups are commonly associated with Route Lists, allowing administrators to organize available outbound paths and establish the order or selection behavior used when routing calls. A translation pattern is primarily used for digit manipulation and call-routing transformations, while a directory number identifies an endpoint and a SIP URI represents a URI-based destination. Route Groups are especially useful when an organization has multiple gateways and needs a logical collection of those resources for centralized routing administration and redundancy.<\/span><\/p>\n

            Question 306. A user calls a number that matches a translation pattern. The administrator wants UCM to modify the called digits and then continue processing the call through the normal dial plan. Which behavior is associated with a translation pattern?<\/b><\/p>\n

              \n
            1. <\/b> It creates a temporary SIP registration<\/span><\/li>\n
            2. <\/b> It performs digit transformation before subsequent routing<\/span><\/li>\n
            3. <\/b> It reserves WAN bandwidth for the call<\/span><\/li>\n
            4. <\/b> It converts RTP into SIP signaling<\/span><\/li>\n<\/ol>\n

              Correct Answer: 2. It performs digit transformation before subsequent routing<\/b><\/p>\n

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

              A Cisco Unified Communications Manager translation pattern can transform the dialed number before continuing call processing. It is commonly used to manipulate digits, such as adding prefixes, removing access codes, or converting a dialed number into a normalized format. After the transformation, UCM can continue evaluating the resulting number according to the configured dial plan. Translation patterns do not create SIP registrations, reserve WAN bandwidth, or convert RTP into SIP signaling. Understanding the difference between translation patterns and route patterns is important: translation patterns primarily transform and redirect call processing, while route patterns are commonly used to select an outbound route toward a gateway or trunk.<\/span><\/p>\n

              Question 307. A UCM administrator needs to provide an alternate PSTN gateway if the primary gateway cannot complete a call. Which combination provides the appropriate routing structure?<\/b><\/p>\n

                \n
              1. <\/b> Directory Number and Device Pool<\/span><\/li>\n
              2. <\/b> Translation Pattern and SIP Profile<\/span><\/li>\n
              3. <\/b> Route List and multiple Route Groups<\/span><\/li>\n
              4. <\/b> Hunt Pilot and Line Group<\/span><\/li>\n<\/ol>\n

                Correct Answer: 3. Route List and multiple Route Groups<\/b><\/p>\n

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

                A Route List can contain multiple Route Groups and provides a structured mechanism for selecting among different outbound routing resources. If the primary route group cannot complete a call, UCM can use another configured route group according to the routing design. This makes the Route List and Route Group combination useful for gateway redundancy and alternate PSTN paths. A Directory Number and Device Pool describe endpoint configuration rather than outbound gateway redundancy. Translation Patterns manipulate digits, and SIP Profiles affect SIP signaling behavior. Hunt Pilots and Line Groups distribute incoming calls among endpoints and are not the standard mechanism for selecting alternate PSTN gateways.<\/span><\/p>\n

                Question 308. A remote office loses connectivity to the central UCM cluster. Phones successfully enter survivability mode, but calls to PSTN destinations fail. Which area should the administrator examine?<\/b><\/p>\n

                  \n
                1. <\/b> SRST dial peers and PSTN routing configuration<\/span><\/li>\n
                2. <\/b> UCM Extension Mobility profiles<\/span><\/li>\n
                3. <\/b> SIP URI domain configuration on the central cluster<\/span><\/li>\n
                4. <\/b> Global Dial Plan Replication<\/span><\/li>\n<\/ol>\n

                  Correct Answer: 1. SRST dial peers and PSTN routing configuration<\/b><\/p>\n

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

                  When phones enter Cisco SRST mode during a WAN or UCM connectivity failure, the local gateway must have the necessary configuration to provide survivable call processing. If internal survivability works but PSTN calls fail, the administrator should examine SRST dial peers, destination patterns, trunk or voice-port configuration, digit manipulation, and the available PSTN path. Central UCM features such as Extension Mobility and Global Dial Plan Replication may be unavailable during the outage and are not the primary mechanism controlling local SRST PSTN calls. SIP URI domains on the central cluster are similarly unrelated to local PSTN routing during survivability. Troubleshooting should therefore focus on the branch gateway’s SRST call-routing configuration.<\/span><\/p>\n

                  Question 309. A CME administrator wants a newly configured SIP phone to register automatically with the CME router using a defined directory number. Which configuration area should be verified first?<\/b><\/p>\n

                    \n
                  1. <\/b> CUBE tenant configuration<\/span><\/li>\n
                  2. <\/b> SIP phone and CME registration configuration<\/span><\/li>\n
                  3. <\/b> UCM route list configuration<\/span><\/li>\n
                  4. <\/b> SRST fallback timers<\/span><\/li>\n<\/ol>\n

                    Correct Answer: 2. SIP phone and CME registration configuration<\/b><\/p>\n

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

                    Cisco Unified CME can provide SIP registration services for supported SIP phones when the appropriate SIP registrar and phone configuration are established. If a newly configured SIP phone does not register, the administrator should verify the phone’s SIP registration settings, credentials where required, directory-number association, CME registrar configuration, and network reachability. CUBE tenants are used for border-element trunk separation and are not required simply to register a local CME SIP phone. UCM route lists are unrelated to local CME registration, and SRST fallback timers address survivability behavior. Registration troubleshooting should begin by confirming that both the phone and CME have compatible SIP registration parameters.<\/span><\/p>\n

                    Question 310. A CME router has several dial peers that can potentially match the same destination. The administrator wants to stop additional dial-peer hunting after a selected dial peer is considered the final choice. Which configuration concept is relevant?<\/b><\/p>\n

                      \n
                    1. <\/b> SIP registrar<\/span><\/li>\n
                    2. <\/b> Codec class<\/span><\/li>\n
                    3. <\/b> Huntstop<\/span><\/li>\n
                    4. <\/b> Voice translation rule<\/span><\/li>\n<\/ol>\n

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

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

                      The huntstop concept in Cisco IOS voice configuration controls whether dial-peer hunting should continue after a selected dial peer fails to complete the call under applicable conditions. Proper use of huntstop can prevent calls from being sent unexpectedly to another dial peer when the administrator intends a particular routing decision to be final. This is especially important in complex dial plans containing overlapping destination patterns and multiple potential outbound paths. SIP registrar configuration handles endpoint registration, codec classes influence media selection, and voice translation rules manipulate numbers. When troubleshooting unexpected fallback to another dial peer, administrators should review dial-peer matching, preference, session targets, and huntstop behavior.<\/span><\/p>\n

                      Question 311. A CUBE receives an inbound SIP INVITE from an untrusted network. The administrator wants to restrict which source addresses are permitted to establish SIP sessions with the enterprise. Which security mechanism should be considered?<\/b><\/p>\n

                        \n
                      1. <\/b> SIP access control and trusted-source filtering<\/span><\/li>\n
                      2. <\/b> RTP payload-type preference<\/span><\/li>\n
                      3. <\/b> Extension Mobility<\/span><\/li>\n
                      4. <\/b> Globalized number normalization<\/span><\/li>\n<\/ol>\n

                        Correct Answer: 1. SIP access control and trusted-source filtering<\/b><\/p>\n

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

                        CUBE should be protected against unauthorized SIP traffic because an exposed voice gateway can become a target for scanning, call abuse, or toll fraud. Source-address filtering and appropriate SIP access-control mechanisms can restrict signaling to known service providers, enterprise peers, or other explicitly trusted sources. Administrators should also apply broader voice-security practices such as restrictive dial peers and appropriate authentication or signaling controls where supported. RTP payload types affect media negotiation and do not determine whether a SIP source is trusted. Extension Mobility is a user-profile feature, and globalized number normalization is a dial-plan function. Restricting signaling sources is therefore a fundamental part of securing a CUBE deployment.<\/span><\/p>\n

                        Question 312. A CUBE call connects, but the service provider rejects the INVITE because the enterprise presents a private internal IP address in the SIP signaling. Which CUBE function can help modify signaling information for interoperability?<\/b><\/p>\n

                          \n
                        1. <\/b> Call Admission Control<\/span><\/li>\n
                        2. <\/b> SIP profile or signaling manipulation<\/span><\/li>\n
                        3. <\/b> Extension Mobility<\/span><\/li>\n
                        4. <\/b> Route Group selection<\/span><\/li>\n<\/ol>\n

                          Correct Answer: 2. SIP profile or signaling manipulation<\/b><\/p>\n

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

                          CUBE can perform signaling manipulation to address interoperability differences between enterprise and service-provider SIP environments. SIP profiles can modify selected SIP headers and message information when the provider requires different signaling content. Depending on the specific network design, NAT and other CUBE mechanisms may also be relevant when private addressing must be translated for external communication. Call Admission Control is a UCM resource-management feature, Extension Mobility concerns user profiles, and Route Groups select routing resources. When a provider explicitly rejects an INVITE because of unexpected SIP signaling information, packet captures and SIP traces should be reviewed to identify the exact field that requires correction.<\/span><\/p>\n

                          Question 313. During a SIP call, the caller hears a provider announcement before the call is answered. The SIP trace shows a provisional response containing SDP. Which media behavior is occurring?<\/b><\/p>\n

                            \n
                          1. <\/b> RTP is being blocked because the call is unanswered<\/span><\/li>\n
                          2. <\/b> The SIP dialog has already been terminated<\/span><\/li>\n
                          3. <\/b> Early media is being established<\/span><\/li>\n
                          4. <\/b> The call has entered SRST mode<\/span><\/li>\n<\/ol>\n

                            Correct Answer: 4. Early media is being established<\/b><\/p>\n

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

                            Early media allows audio to flow before the called party sends the final answer to the call. A SIP provisional response such as 183 Session Progress can include SDP, allowing the media path to be established while the call remains in a proceeding state. This is commonly used for carrier announcements, network-generated ringback, tones, or interactive announcements. The presence of SDP in the provisional response is therefore a strong indication that media negotiation is occurring before final call acceptance. The call has not necessarily been answered, and SRST is unrelated unless the endpoint has actually lost connectivity to UCM and entered survivability operation.<\/span><\/p>\n

                            Question 314. A SIP trunk is configured with a session timer. What is the primary purpose of the session timer?<\/b><\/p>\n

                              \n
                            1. <\/b> To determine the RTP codec preference<\/span><\/li>\n
                            2. <\/b> To provide periodic session refreshes and detect stale SIP sessions<\/span><\/li>\n
                            3. <\/b> To assign directory numbers to phones<\/span><\/li>\n
                            4. <\/b> To determine which PSTN gateway is selected<\/span><\/li>\n<\/ol>\n

                              Correct Answer: 2. To provide periodic session refreshes and detect stale SIP sessions<\/b><\/p>\n

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

                              SIP session timers provide a mechanism for periodically refreshing an established SIP session. The refresh process helps ensure that both sides continue to recognize the session as active and provides a method for detecting sessions that have become stale because signaling connectivity has failed. Session timers are different from transaction timers, which control the behavior and retransmission of individual SIP transactions such as an INVITE. Session timers also do not determine codec preference, assign directory numbers, or select PSTN gateways. When calls unexpectedly clear after a period of time, administrators should consider session-refresh behavior along with signaling connectivity, timers, and the configuration of both SIP peers.<\/span><\/p>\n

                              Question 315. A SIP endpoint sends an INVITE and later sends CANCEL before the call is answered. What is the purpose of CANCEL?<\/b><\/p>\n

                                \n
                              1. <\/b> To terminate an established call<\/span><\/li>\n
                              2. <\/b> To modify the active media session<\/span><\/li>\n
                              3. <\/b> To cancel a pending SIP request before a final response<\/span><\/li>\n
                              4. <\/b> To acknowledge a reliable provisional response<\/span><\/li>\n<\/ol>\n

                                Correct Answer: 3. To cancel a pending SIP request before a final response<\/b><\/p>\n

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

                                CANCEL is used to terminate a pending SIP transaction, such as an INVITE that has not yet received a final response. It is different from BYE, which is normally used to terminate an established SIP dialog. CANCEL can occur when a caller hangs up while the destination is still ringing or otherwise before the INVITE receives its final response. PRACK has a different purpose: it acknowledges reliable provisional responses. Re-INVITE or UPDATE can modify an established session, depending on the signaling scenario. Understanding these SIP methods is important during call-flow analysis because the timing of CANCEL relative to provisional and final responses determines what happens to the pending call attempt.<\/span><\/p>\n

                                Question 316. A UCM administrator needs to route calls based on a SIP URI domain, sending users from one organization to a specific SIP trunk. Which configuration should be examined?<\/b><\/p>\n

                                  \n
                                1. <\/b> Call Park<\/span><\/li>\n
                                2. <\/b> SIP Route Pattern<\/span><\/li>\n
                                3. <\/b> Line Group<\/span><\/li>\n
                                4. <\/b> Device Mobility<\/span><\/li>\n<\/ol>\n

                                  Correct Answer: 2. SIP Route Pattern<\/b><\/p>\n

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

                                  SIP Route Patterns allow Cisco Unified Communications Manager to match SIP URI destinations and route them toward the appropriate trunk or routing resource. This is useful when organizations use URI-based dialing and need calls associated with a particular domain to follow a defined route. The pattern can be designed around the URI structure and associated routing requirements. Call Park temporarily places an active call in a parking slot, Line Groups support hunt-group member organization, and Device Mobility applies location-specific device settings. For URI-based routing problems, administrators should inspect the SIP route pattern, SIP trunk configuration, URI normalization, and relevant domain information.<\/span><\/p>\n

                                  Question 317. A company wants a user to answer calls on a desk phone and later move the active call to a mobile device associated with the same user. Which feature set is most relevant?<\/b><\/p>\n

                                    \n
                                  1. <\/b> Cisco Unified Mobility<\/span><\/li>\n
                                  2. <\/b> Global Dial Plan Replication<\/span><\/li>\n
                                  3. <\/b> Route Group hunting<\/span><\/li>\n
                                  4. <\/b> SIP OPTIONS<\/span><\/li>\n<\/ol>\n

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

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

                                    Cisco Unified Mobility provides functionality for extending enterprise call handling to configured mobile destinations and supporting movement of calls between associated devices. This allows users to remain reachable when away from their desk and can support enterprise mobility workflows involving mobile phones. Global Dial Plan Replication distributes dial-plan information between UCM clusters and does not control mobile call handling. Route Groups determine routing resources, while SIP OPTIONS is a signaling mechanism for capability or availability checks. When troubleshooting a requirement involving an active enterprise call and a user’s associated mobile destination, Unified Mobility configuration, remote destination settings, user association, and relevant mobility service parameters should be reviewed.<\/span><\/p>\n

                                    Question 318. A branch uses Cisco SRST. During a WAN outage, phones remain registered locally, but a call to another branch cannot be completed. What is the most relevant limitation to consider?<\/b><\/p>\n

                                      \n
                                    1. <\/b> SRST automatically provides full centralized UCM feature parity<\/span><\/li>\n
                                    2. <\/b> SRST requires Global Dial Plan Replication to function<\/span><\/li>\n
                                    3. <\/b> SRST does not provide all centralized UCM call-control capabilities during an outage<\/span><\/li>\n
                                    4. <\/b> SRST disables all local PSTN calling<\/span><\/li>\n<\/ol>\n

                                      Correct Answer: 3. SRST does not provide all centralized UCM call-control capabilities during an outage<\/b><\/p>\n

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

                                      SRST is designed to provide essential call-processing survivability at a remote site when connectivity to Cisco Unified Communications Manager is unavailable. It does not reproduce every centralized UCM feature or the complete enterprise call-control environment. Features involving centralized resources, complex intersite routing, advanced applications, or other unavailable services may not operate normally during SRST mode. Local PSTN and basic internal calling can continue when properly configured. Global Dial Plan Replication is not a prerequisite for SRST operation, and SRST does not inherently disable all PSTN calling. When troubleshooting an intersite call during an outage, administrators should consider what call-control resources remain locally available.<\/span><\/p>\n

                                      Question 319. A CUBE administrator observes that SIP signaling succeeds but the remote endpoint receives no audio. A packet capture shows RTP packets leaving CUBE toward an IP address that belongs to the enterprise’s private network. What should be investigated?<\/b><\/p>\n

                                        \n
                                      1. <\/b> Extension Mobility authentication<\/span><\/li>\n
                                      2. <\/b> Route-list preference<\/span><\/li>\n
                                      3. <\/b> SIP registration expiration<\/span><\/li>\n
                                      4. <\/b> NAT and SDP\/media-address handling<\/span><\/li>\n<\/ol>\n

                                        Correct Answer: 4. NAT and SDP\/media-address handling<\/b><\/p>\n

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

                                        When CUBE sends RTP toward an unreachable private address, the SDP and media-path handling should be investigated. SDP advertises the addresses and ports used for RTP, so incorrect or untranslated addresses can result in one-way or no-way audio even when SIP signaling completes successfully. Depending on the topology, NAT configuration, CUBE media handling, and SDP manipulation may all need to be reviewed. Extension Mobility authentication, route-list preference, and SIP registration expiration do not normally explain RTP being sent toward an unreachable private address. Packet captures are particularly valuable because they can confirm the advertised SDP addresses and the actual RTP destination, helping distinguish signaling problems from media-path problems.<\/span><\/p>\n

                                        Question 320. A UCM administrator wants to prevent WAN voice traffic from exceeding the bandwidth allocated to a remote site. Which configuration area should be reviewed when calls are being rejected because the configured intersite voice bandwidth has been consumed?<\/b><\/p>\n

                                          \n
                                        1. <\/b> SIP Route Pattern<\/span><\/li>\n
                                        2. <\/b> Call Admission Control and locations<\/span><\/li>\n
                                        3. <\/b> Extension Mobility<\/span><\/li>\n
                                        4. <\/b> SIP profile<\/span><\/li>\n<\/ol>\n

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

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

                                          Cisco Unified Communications Manager uses Call Admission Control together with location-based bandwidth configuration to regulate calls across constrained network paths. Locations can be associated with bandwidth limits, allowing UCM to determine whether sufficient resources remain for another call. When the configured bandwidth has been consumed, additional calls may be denied or routed according to the configured CAC behavior. SIP Route Patterns determine call routing, Extension Mobility manages user profiles on phones, and SIP Profiles control selected SIP signaling behavior. Therefore, when calls are specifically being rejected because an intersite WAN bandwidth allocation has been exhausted, the administrator should review CAC, location relationships, and the configured bandwidth values.<\/span><\/p>\n

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

                                          View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps   Question 301. A Cisco Unified Communications Manager administrator needs to route calls to a PSTN gateway only when the dialed number matches a specific pattern. Which UCM dial-plan element provides the pattern matching used to select a route? Route Group Hunt Pilot […]<\/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\/23433"}],"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=23433"}],"version-history":[{"count":1,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23433\/revisions"}],"predecessor-version":[{"id":23434,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/posts\/23433\/revisions\/23434"}],"wp:attachment":[{"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/media?parent=23433"}],"wp:term":[{"taxonomy":"category","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/categories?post=23433"},{"taxonomy":"post_tag","embeddable":true,"href":"https:\/\/www.examlabs.com\/certification\/wp-json\/wp\/v2\/tags?post=23433"}],"curies":[{"name":"wp","href":"https:\/\/api.w.org\/{rel}","templated":true}]}}