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

 

Question 261. A Cisco Unified Communications Manager administrator needs calls matching a particular destination pattern to use a specific group of PSTN gateways. Which configuration should be associated with the route pattern?

1. Device pool
2. Region
3. Location
4. Route list

Correct Answer: 4. Route list

Explanation :-

A route pattern identifies a destination pattern and can point to a route list for subsequent route selection. The route list contains route groups, which in turn identify the gateways or trunks that can be used for the call. This structure separates destination analysis from gateway selection and allows several route patterns to share a common routing policy. Device pools provide common device configuration, regions influence codec relationships, and locations support bandwidth-based call admission control. Therefore, when a particular destination pattern needs to use a defined group of PSTN gateways, the route pattern should be associated with the appropriate route list.

Question 262. A Cisco Unified Communications Manager user can dial internal extensions but receives a reorder tone when attempting an international number. The international route pattern is in a partition that the user’s line CSS cannot access. What is the most direct correction?

1. Add the international partition to the effective CSS, if policy permits
2. Change the phone’s region
3. Modify the route group’s distribution algorithm
4. Increase the SIP session timer

Correct Answer: 1. Add the international partition to the effective CSS, if policy permits

Explanation :-

A Calling Search Space determines which partitions a line or device can search during digit analysis. If an international route pattern resides in a partition that is not included in the user’s effective CSS, Cisco Unified Communications Manager cannot select that route pattern for the call. If policy permits international access, the appropriate correction is to include the required partition in the effective CSS. Region settings affect codec relationships, route groups affect gateway selection, and session timers affect SIP dialog refresh. CSS and partition design therefore provides the direct control over this routing-access issue.

Question 263. A service provider requires CUBE to replace the enterprise’s internal extension in the From header with a full public calling number. Which combination is appropriate when both digit and SIP-header manipulation are required?

1. Region and location configuration
2. Translation rules/profiles and SIP profile or normalization
3. Route groups and device pools
4. SRST and Device Mobility

Correct Answer: 3. Translation rules/profiles and SIP profile or normalization

Explanation :-

The requirement involves two distinct operations. Translation rules and translation profiles can transform the telephone digits, such as converting an internal extension into a full E.164 number. SIP profiles or normalization mechanisms can manipulate the SIP signaling fields where the resulting identity must appear in a particular header format. Route groups and device pools do not perform these functions, while SRST and Device Mobility address survivability and location-specific device behavior. Using the appropriate combination of digit translation and SIP signaling manipulation allows CUBE to satisfy provider-specific identity requirements.

Question 264. A WAN connection between two collaboration sites has limited bandwidth. The administrator wants calls between those sites to use a compressed codec while local calls retain a different codec policy. Which configuration should be reviewed?

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

Correct Answer: 2. Region relationship

Explanation :-

Cisco Unified Communications Manager regions establish codec relationships between devices in different regions. Administrators can specify codec preferences for calls within a region and between different regions, allowing a compressed codec to be used over a constrained WAN while maintaining another codec for local communication. Calling Search Spaces control partition access, translation patterns manipulate digits, and route groups determine gateway or trunk selection. Therefore, the administrator should review the region relationship between the two sites when the objective is to control codec selection according to WAN bandwidth.

Question 265. During a WAN outage, branch phones register to SRST. Internal calls work, but calls to the PSTN fail because the router does not select an appropriate external voice path. Which configuration should be examined?

1. Centralized Unified Communications Manager CSS
2. Region relationship
3. Local SRST dial peers and PSTN gateway configuration
4. Centralized route list

Correct Answer: 3. Local SRST dial peers and PSTN gateway configuration

Explanation :-

In SRST mode, the branch router provides local survivable call processing. External calls therefore depend on local dial peers and a reachable PSTN gateway or trunk. The administrator should verify the dial-peer destination patterns, preferences, voice connectivity, and gateway or trunk configuration. Centralized CSS and route lists may not control the local call path while the branch is operating independently during a WAN outage. Region relationships also do not provide PSTN routing. Local SRST call-routing configuration is therefore the appropriate troubleshooting focus.

Question 266. A SIP endpoint receives a 183 Session Progress response with SDP and begins receiving audio from the network before the call is answered. What is this audio commonly called?

1. Registration media
2. Established-dialog media
3. DTMF relay
4. Early media

Correct Answer: 4. Early media

Explanation :-

Early media refers to media exchanged before the SIP call receives its final successful response. A 183 Session Progress response can carry SDP describing the media session, allowing audio such as announcements, tones, or other network-provided information to reach the caller before the called party answers. This differs from normal established-call media. REGISTER handles endpoint registration, DTMF relay carries keypad signaling, and established-dialog media occurs after the session has been answered. Therefore, 183 Session Progress with SDP followed by pre-answer audio is a common indication of early media.

Question 267. A CUBE router has separate enterprise and provider interfaces. SIP signaling follows the correct path, but the SDP advertises the enterprise-side private address toward the provider. Which issue should be investigated?

1. Calling Search Space
2. Translation pattern
3. Media address or interface binding
4. Route list

Correct Answer: 2. Media address or interface binding

Explanation :-

CUBE must advertise a media address that the remote party can actually reach. In a multihomed deployment, incorrect media interface or address binding can cause CUBE to place an enterprise-side or private address into SDP sent to the provider. The provider may then attempt to send RTP to an unreachable destination, resulting in one-way or no audio. Calling Search Spaces and route lists control call routing, while translation patterns manipulate digits. The administrator should therefore inspect CUBE media-interface selection, media-address bindings, routing, and NAT behavior.

Question 268. An enterprise uses separate SIP domains for two business partners. CUBE must route calls to each partner through a different SIP trunk based on the destination URI domain. Which method is appropriate?

1. SIP URI-based dial-peer matching
2. Location-based CAC
3. Device Mobility
4. Region configuration

Correct Answer: 1. SIP URI-based dial-peer matching

Explanation :-

SIP URI-based dial-peer matching allows CUBE to use URI information, including the destination domain, when selecting a dial peer. This is useful when different SIP domains correspond to different providers, partners, or routing policies. Location-based CAC manages bandwidth admission, Device Mobility applies location-specific device settings, and region configuration controls codec relationships. Because the requirement is to select different trunks according to the destination SIP domain, URI-based dial-peer matching provides the relevant routing mechanism.

Question 269. A provider rejects an outbound INVITE because the enterprise sends a SIP header that the provider does not support. Which feature can be used to remove or modify the header before forwarding the INVITE?

1. Location CAC
2. Route group
3. Device pool
4. SIP profile or normalization

Correct Answer: 4. SIP profile or normalization

Explanation :-

SIP profiles and normalization mechanisms can modify supported SIP signaling elements to meet interoperability requirements. When a provider rejects an INVITE because of an unsupported or incorrectly formatted header, the administrator can investigate the applicable SIP profile or normalization configuration to remove or modify that signaling element. Location CAC controls bandwidth admission, route groups control gateway selection, and device pools provide common device settings. Therefore, signaling manipulation through an appropriate SIP profile or normalization policy is the relevant solution for provider-specific SIP header requirements.

Question 270. A SIP call drops at nearly the same interval every time. Packet capture shows that the negotiated refresh mechanism is failing. Which SIP feature should be examined?

1. Device Mobility
2. Session timers
3. Translation patterns
4. Route groups

Correct Answer: 2. Session timers

Explanation :-

SIP session timers establish a periodic refresh mechanism for an active SIP session. When refresh requests or responses fail, an endpoint may consider the session expired and terminate the call. A repeatable disconnect at approximately the same negotiated interval is therefore a strong indication that session-timer behavior should be examined. Device Mobility changes location-specific settings, translation patterns manipulate digits, and route groups control routing. The administrator should inspect Session-Expires, Min-SE, refresher negotiation, and the actual refresh messages to identify why the session is expiring.

Question 271. A Unified Communications Manager administrator wants to ensure that only phones assigned to a specific user group can access a set of route patterns. Which two dial-plan elements provide this access control?

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

Correct Answer: 1. Partition and Calling Search Space

Explanation :-

Partitions organize Cisco Unified Communications Manager dial-plan objects, while Calling Search Spaces determine which partitions a device or line can search. By placing selected route patterns in a dedicated partition and assigning an appropriate CSS only to authorized users, the administrator can control access to those destinations. Route lists and route groups determine routing paths after a destination is selected, while regions and locations address media and bandwidth behavior. Device pools and Device Mobility provide configuration based on devices and locations rather than directly defining partition visibility.

Question 272. A CUBE gateway should accept SIP traffic only from known provider addresses. Which configuration provides source-level protection?

1. Codec preference
2. Translation profile
3. Source-based access controls and trusted peers
4. SIP session timer

Correct Answer: 3. Source-based access controls and trusted peers

Explanation :-

Source-based access controls and trusted-peer policies can restrict which SIP sources are permitted to communicate with CUBE. This helps protect an enterprise voice gateway from unauthorized SIP requests and reduces exposure to toll fraud and unwanted call attempts. Codec preferences affect media encoding, translation profiles manipulate digits, and session timers manage SIP session refresh behavior. When the requirement is to accept SIP signaling only from known provider addresses, source validation and trusted-peer configuration are the appropriate security controls to investigate and implement.

Question 273. A CUBE administrator needs to convert an incoming seven-digit number into a ten-digit number before the call is routed to an enterprise destination. Which feature should be configured?

1. SIP profile
2. Translation rule and translation profile
3. Region configuration
4. Location CAC

Correct Answer: 2. Translation rule and translation profile

Explanation :-

Cisco IOS translation rules provide digit manipulation, while translation profiles apply those rules to the appropriate call-processing direction or dial peer. A seven-digit number can therefore be transformed into a ten-digit number before the call is routed further into the enterprise. SIP profiles manipulate SIP signaling rather than primarily performing digit translation. Region configuration controls codec relationships, and Location CAC controls bandwidth admission. When the requirement specifically concerns adding, removing, or replacing telephone digits, translation rules and translation profiles are the appropriate configuration tools.

Question 274. A SIP trunk uses encrypted signaling, but calls still have unencrypted voice media. Which configuration should be added to protect the RTP stream?

1. SIP OPTIONS
2. SIP REGISTER
3. SIP REFER
4. SRTP

Correct Answer: 4. SRTP

Explanation :-

TLS can protect SIP signaling, but it does not automatically encrypt the RTP media stream. Secure Real-time Transport Protocol (SRTP) is designed to protect real-time media by providing encryption and integrity protection for RTP traffic. OPTIONS, REGISTER, and REFER are SIP methods and do not provide media encryption. Therefore, if signaling is already protected with TLS but voice media remains unencrypted, the administrator should investigate SRTP configuration and ensure that both communicating endpoints or systems support and negotiate compatible secure-media parameters.

Question 275. A user moves from one office to another and needs a different CSS and location assignment automatically applied to the phone. Which Cisco Unified Communications Manager feature supports this requirement?

1. Device Mobility
2. Route group
3. SIP profile
4. Translation pattern

Correct Answer: 3. Device Mobility

Explanation :-

Device Mobility allows Cisco Unified Communications Manager to apply location-specific settings when a phone operates from a different configured location. Depending on the deployment, the resulting configuration can include calling search space, region, location, and other device-related settings. Route groups determine gateway selection, SIP profiles manipulate signaling, and translation patterns manipulate digits. Therefore, when a phone should automatically receive a different CSS and location-related configuration after moving between offices, Device Mobility is the appropriate feature to investigate.

Question 276. A WAN site has a configured bandwidth limit for voice calls. New calls should be rejected when the available intersite bandwidth is exhausted. Which feature controls this behavior?

1. SIP normalization
2. Translation pattern
3. Location-based Call Admission Control
4. Route group

Correct Answer: 1. Location-based Call Admission Control

Explanation :-

Location-based Call Admission Control allows Cisco Unified Communications Manager to manage voice bandwidth between configured locations. When the available bandwidth allocation for a path has been consumed, CAC can prevent additional calls from being admitted over that constrained path. This protects existing calls and helps avoid excessive bandwidth contention. SIP normalization modifies signaling, translation patterns manipulate digits, and route groups determine gateway or trunk selection. Therefore, when call admission must depend on configured intersite bandwidth, Location-based CAC is the appropriate feature.

Question 277. A SIP endpoint sends an INVITE and then receives a 180 Ringing response. Which statement best describes the response?

1. The SIP registration has completed
2. The remote endpoint has requested call transfer
3. The called endpoint is being alerted
4. The established call is being terminated

Correct Answer: 4. The called endpoint is being alerted

Explanation :-

The SIP 180 Ringing response is a provisional response indicating that the called endpoint is being alerted. It occurs before a final response such as 200 OK and commonly corresponds to ringing at the destination. It should not be confused with 183 Session Progress, which can be used with SDP to establish early media. REGISTER is used for registration, REFER is commonly associated with referral or transfer operations, and BYE terminates an established dialog. Therefore, 180 Ringing primarily indicates that the called side is being alerted.

Question 278. A SIP endpoint needs to query a remote SIP server to determine whether it is reachable and what capabilities it advertises without placing a normal call. Which SIP method should be used?

1. CANCEL
2. OPTIONS
3. BYE
4. REGISTER

Correct Answer: 3. OPTIONS

Explanation :-

The SIP OPTIONS method can be used to query a SIP endpoint or server for availability and supported capabilities without establishing a normal voice session. A response can provide information about supported SIP methods, extensions, and other capabilities depending on the implementation. CANCEL terminates a pending request, BYE terminates an established dialog, and REGISTER is used for endpoint registration. Therefore, OPTIONS is the appropriate method when an administrator needs a lightweight SIP reachability or capability check.

Question 279. A branch router is operating in SRST mode. The administrator wants calls to a local PSTN number to prefer one gateway and use another gateway if the first becomes unavailable. Which configuration should be used?

1. Centralized CSS
2. Device Mobility
3. Local SRST dial peers with appropriate preference or failover routing
4. Unified Communications Manager region configuration

Correct Answer: 2. Device Mobility

Explanation :-

The correct mechanism for providing local SRST PSTN routing and gateway preference is local SRST dial peers with appropriate preference or failover routing. In SRST mode, the branch router must independently determine how calls are routed because centralized call control may be unavailable. Device Mobility does not provide gateway failover, while centralized CSS and region configuration are not the mechanisms that select the local PSTN path during SRST operation. Administrators should therefore configure and verify local dial peers, destination patterns, preferences, and gateway reachability for survivable PSTN routing.

Question 280. A SIP administrator sees a BYE message during an established call. What does the BYE request normally indicate?

1. The established SIP dialog is being terminated
2. The endpoint is requesting early media
3. The endpoint is registering
4. The endpoint is acknowledging a provisional response

Correct Answer: 1. The established SIP dialog is being terminated

Explanation :-

The SIP BYE method is used to terminate an established SIP dialog. When a BYE is exchanged during an active call, it indicates that the sending endpoint is ending the established session. Early media is associated with provisional signaling such as 183 Session Progress with SDP, endpoint registration uses REGISTER, and reliable provisional responses can be acknowledged using PRACK. Therefore, observing BYE during an established call is a clear indication that the SIP dialog is being terminated, assuming the message belongs to that active dialog.