View Full Cisco CCNP Collaboration 300-815 Exam Dumps and Practice Test Dumps

 

Question 181. A Cisco Unified Communications Manager deployment has two SIP trunks to the same service provider. The administrator wants calls to use the first trunk whenever it is available and the second trunk only when the first route is unavailable. Which configuration provides this behavior?

1. Assign both trunks to separate partitions
2. Configure the trunks as members of a route group with ordered priority
3. Configure separate device pools for each trunk
4. Apply different regions to the SIP trunks

Correct Answer: 2. Configure the trunks as members of a route group with ordered priority

Explanation :-

A route group in Cisco Unified Communications Manager can contain multiple gateways or trunks and define their selection order. When the route group is configured with an ordered distribution algorithm, the first available member is selected before subsequent members are attempted. This provides a primary-and-secondary routing design without requiring separate route patterns. Partitions control dial-plan visibility, regions control codec and bandwidth relationships, and device pools provide common device configuration. Therefore, an ordered route group is the appropriate mechanism for providing primary and backup trunk selection.

Question 182. A SIP call traversing CUBE fails because the provider rejects the incoming INVITE after checking the source address. Which CUBE configuration should the administrator investigate first?

1. Region codec relationships
2. SIP session timer values
3. Device Mobility settings
4. Trusted source or inbound dial-peer security configuration

Correct Answer: 4. Trusted source or inbound dial-peer security configuration

Explanation :-

CUBE can use source-based controls and inbound dial-peer matching to determine whether SIP traffic is accepted from a particular peer. If a provider rejects or CUBE blocks an INVITE because of its source, the administrator should verify trusted-peer configuration, access controls, and the inbound dial-peer used for that connection. Session timers affect dialog refresh behavior rather than initial source authorization. Device Mobility and regions are unrelated to SIP source validation. Checking the signaling source and inbound security policy first helps determine whether the INVITE is reaching the expected trusted CUBE path.

Question 183. A user can place calls to a PSTN number, but the called party sees the enterprise’s internal extension instead of the expected full telephone number. Which Cisco CUBE capability can modify the calling number before the SIP INVITE is sent to the provider?

1. SIP session timer
2. Translation rule and translation profile
3. Region configuration
4. Device Mobility

Correct Answer: 1. Translation rule and translation profile

Explanation :-

Cisco IOS translation rules and translation profiles can manipulate telephone digits before a call is sent through a dial peer. For example, an internal extension can be transformed into the organization’s complete E.164 calling number before CUBE forwards the INVITE to a service provider. SIP profiles are instead commonly used for SIP message and header manipulation, while regions and Device Mobility address other aspects of the collaboration environment. The translation profile must be applied to the appropriate dial peer so that the intended calling-number transformation occurs for the outbound call.

Question 184. A remote branch phone registers with Cisco Unified Communications Manager over a WAN connection. The administrator wants to ensure that voice traffic between the branch and headquarters does not consume more bandwidth than necessary. Which configuration primarily controls the codec relationship between the branch and headquarters?

1. Route group configuration
2. Calling Search Space configuration
3. Region configuration
4. Translation pattern configuration

Correct Answer: 3. Region configuration

Explanation :-

Cisco Unified Communications Manager regions define codec relationships between devices associated with different regions. By configuring the appropriate audio codec between the branch and headquarters regions, an administrator can influence the bandwidth consumed by calls across the WAN. Route groups determine gateway or trunk selection, CSS controls dial-plan access, and translation patterns manipulate dialed digits. Region configuration therefore provides the appropriate mechanism for controlling the codec and associated bandwidth characteristics between geographically separated collaboration resources.

Question 185. A branch loses connectivity to Cisco Unified Communications Manager. Phones are configured for survivability and must continue making external calls through the local PSTN gateway. Which component provides the local call-processing and PSTN routing capability?

1. Cisco Unified SRST with local dial peers
2. Cisco Unified Communications Manager route list
3. Device Mobility
4. SIP URI dialing

Correct Answer: 1. Cisco Unified SRST with local dial peers

Explanation :-

Cisco Unified Survivable Remote Site Telephony (SRST) provides local call-processing capabilities when a branch phone can no longer reach its centralized Cisco Unified Communications Manager. During survivability mode, local dial peers on the branch router can route calls toward the local PSTN gateway or other reachable destinations. Centralized route lists and Device Mobility do not provide autonomous branch call processing during a WAN outage. SIP URI dialing is a dialing method rather than a survivability mechanism. Therefore, SRST together with appropriately configured local dial peers is the expected solution.

Question 186. A SIP endpoint sends a reliable provisional response request that requires acknowledgment from the receiving endpoint. Which SIP method is used to acknowledge the reliable provisional response?

1. UPDATE
2. CANCEL
3. PRACK
4. OPTIONS

Correct Answer: 4. PRACK

Explanation :-

PRACK, or Provisional Response Acknowledgment, is used to acknowledge reliable provisional SIP responses. It is particularly relevant when provisional responses such as 183 Session Progress are sent reliably using the mechanisms defined by SIP extensions. UPDATE modifies session information during an established dialog when applicable, CANCEL terminates a pending request, and OPTIONS is commonly used to query reachability or capabilities. PRACK therefore provides the specific acknowledgment mechanism required when a provisional response is sent reliably and the receiving endpoint must confirm its receipt.

Question 187. A CUBE call establishes successfully, but the external party cannot hear the enterprise endpoint. Packet captures show RTP packets leaving CUBE toward the provider, but the SDP advertises a private address that the provider cannot reach. What should the administrator investigate?

1. Calling Search Space assignment
2. Route pattern partition
3. Media address/interface binding and NAT configuration
4. SIP registration expiration

Correct Answer: 3. Media address/interface binding and NAT configuration

Explanation :-

SDP advertises the IP address and port where media is expected to flow. If CUBE advertises a private, unreachable address to an external provider, RTP may be sent toward an address that cannot be reached across the provider or NAT boundary. The administrator should therefore inspect CUBE media interface or address bindings and the associated NAT design. CSS and route-pattern partitions affect call routing, while SIP registration expiration does not explain an established call with an unreachable SDP media address. Correct media addressing is essential for two-way RTP across network boundaries.

Question 188. An enterprise uses multiple SIP providers, and the administrator wants CUBE to select an outbound dial peer based on the SIP URI domain in the destination address. Which matching capability should be used?

1. URI-based dial-peer matching
2. Device Mobility
3. Location-based CAC
4. Route group distribution

Correct Answer: 2. URI-based dial-peer matching

Explanation :-

CUBE can use SIP URI information to select an appropriate dial peer when calls are addressed using SIP URI domains. URI-based matching allows different destinations or providers to be associated with specific domains, providing more precise routing than relying only on traditional digit patterns. Device Mobility changes device-specific settings based on location, location-based CAC controls available bandwidth, and route groups provide gateway or trunk selection after route logic identifies the appropriate path. Therefore, URI-based dial-peer matching is the relevant capability for selecting an outbound peer according to the SIP destination domain.

Question 189. A SIP provider requires a custom header to be inserted into outbound INVITEs from CUBE. Which feature should the administrator configure?

1. Location CAC
2. SIP profile or SIP normalization
3. Route group
4. Region configuration

Correct Answer: 4. SIP profile or SIP normalization

Explanation :-

SIP profiles and normalization mechanisms can modify SIP signaling, including adding, removing, or changing supported SIP headers and message content. This makes them appropriate when an external provider requires a specific signaling format or custom header in an outbound INVITE. Location CAC controls bandwidth admission, route groups determine gateway or trunk selection, and regions control codec relationships. The administrator should apply the appropriate SIP profile or normalization policy to the relevant dial peer or SIP trunk so that the required header manipulation occurs during call signaling.

Question 190. A SIP call disconnects after a predictable interval even though both endpoints remain active. Packet capture shows that session refresh messages are not being acknowledged. Which feature is most likely responsible?

1. SIP session timers
2. Device Mobility
3. Translation patterns
4. DTMF relay

Correct Answer: 1. SIP session timers

Explanation :-

SIP session timers provide periodic refresh mechanisms that allow endpoints or proxies to confirm that an established session remains active. If required refresh requests or responses fail, a device may terminate the call after the negotiated session interval expires. This can produce predictable disconnect behavior even when users are still actively communicating. Device Mobility changes device configuration based on location, translation patterns manipulate digits, and DTMF relay transports keypad signaling. When a call consistently drops after a specific interval and session refresh transactions are failing, session-timer negotiation and refresh behavior should be investigated.

Question 191. A Unified Communications Manager administrator needs to allow a group of phones to access a specific set of route patterns while preventing access to another partition. Which configuration combination provides this control?

1. Region and location
2. Device pool and device mobility
3. Partition and Calling Search Space
4. Route group and route list

Correct Answer: 3. Partition and Calling Search Space

Explanation :-

Partitions divide the dial plan into logical groups, while a Calling Search Space (CSS) determines which partitions a device or line can search. By placing route patterns in an appropriate partition and assigning a CSS that includes only permitted partitions, an administrator can control which destinations a group of phones can reach. Regions and locations address media and bandwidth considerations, device pools provide common configuration, and route lists and route groups determine gateway selection. Therefore, partition and CSS design is the primary mechanism for controlling dial-plan access in Cisco Unified Communications Manager.

Question 192. A CUBE administrator needs to remove an unwanted SIP header from incoming messages before forwarding them to an external service provider. Which feature is most appropriate?

1. Region configuration
2. SIP profile or normalization rule
3. Route group
4. Location CAC

Correct Answer: 2. SIP profile or normalization rule

Explanation :-

SIP profiles and normalization rules are designed to modify SIP signaling as messages traverse a device such as CUBE. They can be used to manipulate supported headers and message elements when interoperability with a service provider requires a different SIP format. Regions govern codec relationships, route groups select gateways or trunks, and location CAC controls bandwidth admission. If an unwanted header must be removed from an INVITE or another SIP message before forwarding, the administrator should investigate the appropriate SIP profile or normalization configuration on the signaling path.

Question 193. A Cisco Unified Communications Manager deployment has a low-bandwidth WAN link between two sites. The administrator wants calls between the sites to use a lower-bandwidth codec while maintaining a different codec for calls within each site. Which configuration should be adjusted?

1. Region relationships
2. Translation patterns
3. Calling Search Spaces
4. Route groups

Correct Answer: 1. Region relationships

Explanation :-

Cisco Unified Communications Manager regions allow administrators to define codec relationships between devices associated with different regions. This makes it possible to select a lower-bandwidth codec for calls traversing a constrained WAN while maintaining a different codec for calls within the same site or region. Translation patterns manipulate digits, CSS controls dial-plan access, and route groups select gateways or trunks. Proper region configuration therefore provides a centralized method for controlling codec selection according to the network path and its available bandwidth.

Question 194. A CUBE administrator wants to prevent unauthorized SIP devices on the Internet from sending calls through the enterprise gateway. Which security measure should be prioritized?

1. Increase the SIP session timer
2. Configure source-based access controls and trusted SIP peers
3. Change the region codec
4. Add another route group

Correct Answer: 2. Configure source-based access controls and trusted SIP peers

Explanation :-

CUBE security should restrict SIP signaling to expected and trusted sources. Source-based access controls, trusted-peer configuration, and appropriate inbound dial-peer policies can reduce the risk of unauthorized devices sending calls through the enterprise gateway. Simply increasing session timers does not provide source authentication or access control. Region codecs affect media characteristics, while additional route groups affect routing redundancy. Because unauthorized Internet-originated SIP traffic can create toll-fraud exposure, the administrator should first establish appropriate controls over which signaling sources CUBE accepts.

Question 195. A user dials an abbreviated internal number, but the system must transform it into the corresponding full E.164 number before routing the call. Which Cisco Unified Communications Manager feature is designed for this type of digit manipulation?

1. Device Mobility
2. Translation pattern
3. Region configuration
4. Location configuration

Correct Answer: 4. Translation pattern

Explanation :-

A translation pattern is used in Cisco Unified Communications Manager to manipulate digits and transform a dialed number into another format before subsequent call routing occurs. This is useful for converting abbreviated internal numbers into full E.164 destinations or performing other digit transformations required by the dial plan. Device Mobility handles location-specific device settings, regions influence codec selection, and locations support bandwidth-based call admission control. When the requirement specifically involves transforming the digits of a dialed number before routing, a translation pattern is the appropriate dial-plan feature.

Question 196. A SIP trunk between two collaboration systems is configured to use TLS for signaling. Calls fail after one side replaces its certificate. Which issue should be checked first?

1. Route group order
2. DTMF relay method
3. Certificate trust and certificate validity
4. Location bandwidth

Correct Answer: 3. Certificate trust and certificate validity

Explanation :-

TLS-secured SIP signaling depends on successful certificate validation and trust relationships between the communicating systems. When one side replaces its certificate, the other side may no longer trust the new certificate or its issuing certificate authority. The administrator should therefore verify certificate validity, trust-store contents, subject or SAN requirements where applicable, and the certificate chain. Route groups, DTMF relay, and location bandwidth do not normally determine whether the TLS handshake succeeds. Certificate-related failures should be investigated before troubleshooting higher-layer SIP call routing.

Question 197. A SIP endpoint receives a BYE from the remote party during an established call. What does the BYE request indicate?

1. The endpoint is terminating the established SIP dialog
2. The endpoint is requesting early media
3. The endpoint is registering with the SIP server
4. The endpoint is acknowledging a provisional response

Correct Answer: 4. The endpoint is terminating the established SIP dialog

Explanation :-

The SIP BYE method is used to terminate an established session or dialog. When an endpoint receives a BYE during an active call, it indicates that the remote side is ending the established SIP session. REGISTER is used for endpoint registration, PRACK acknowledges reliable provisional responses, and 183 Session Progress is associated with provisional signaling and can carry early-media information. Therefore, a BYE observed during an established call should be interpreted as a request to terminate that existing dialog rather than as a registration or provisional-response transaction.

Question 198. An enterprise has multiple gateways in a route group and wants calls to be distributed among available members instead of always selecting the first gateway. Which configuration concept controls how members are selected?

1. Partition assignment
2. Route group distribution algorithm
3. Device Mobility group
4. SIP registration interval

Correct Answer: 2. Route group distribution algorithm

Explanation :-

Cisco Unified Communications Manager route groups can use a distribution algorithm to determine how calls are selected among the available gateways or trunks in the group. Depending on the desired design, calls can be distributed across members rather than always using the same gateway first. Partition assignment determines dial-plan visibility, Device Mobility provides location-specific device settings, and SIP registration intervals relate to endpoint registration behavior. Therefore, when the requirement concerns how calls are distributed among multiple route-group members, the route group distribution algorithm is the relevant configuration.

Question 199. A CUBE deployment successfully establishes signaling, but media fails after a provider changes the network address used for RTP. Which signaling information should the administrator inspect to identify the advertised media destination?

1. SDP connection and media addresses
2. Calling Search Space
3. Route pattern partition
4. SIP registration username

Correct Answer: 3. Route pattern partition

Explanation :-

The Session Description Protocol (SDP) carried within SIP signaling describes media parameters, including the IP address and ports where RTP should be sent. If a provider changes its RTP network address and media stops flowing, inspecting the SDP connection and media-address information is essential to determine whether the correct destination is being advertised. Calling Search Space and route-pattern partitions control call routing rather than RTP addressing. SIP registration credentials also do not identify the media destination. Although this question’s scenario centers on RTP reachability, the correct diagnostic focus is the SDP media-address information.

Question 200. A SIP administrator observes an INVITE followed by 100 Trying, but no subsequent provisional response or final response arrives. Which troubleshooting approach is most useful initially?

1. Change the DTMF relay method
2. Inspect the next-hop routing, dial-peer selection, and SIP transaction path
3. Modify the region codec
4. Change Device Mobility settings

Correct Answer: 2. Inspect the next-hop routing, dial-peer selection, and SIP transaction path

Explanation :-

A 100 Trying response indicates that the INVITE has been received and processing has started, but it does not by itself indicate that the call has reached the destination endpoint. If no later provisional or final response appears, the administrator should trace the signaling path, verify dial-peer selection and next-hop routing, and determine whether the INVITE is being forwarded correctly. DTMF relay, region codec settings, and Device Mobility do not normally explain a missing SIP response after initial INVITE processing. Signaling traces and dial-peer analysis can identify where the transaction stops.