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

 

Question 141. An administrator wants calls matching a specific E.164 destination pattern to be sent to a route list for further gateway selection. Which Unified Communications Manager object performs the destination-pattern match?

1. Route group
2. Device pool
3. Location
4. Route pattern

Correct Answer: 4. Route pattern

Explanation :-

A route pattern performs destination-number analysis in Cisco Unified Communications Manager. It can match an E.164 number or another defined dialing pattern and direct the call toward a configured route list or other routing destination. Route groups contain gateways or trunks, device pools provide common device settings, and locations participate in call admission control. When configuring E.164 routing, the engineer should verify the pattern syntax, partition assignment, and the Calling Search Space available to the calling device. Proper digit analysis should be confirmed before troubleshooting downstream gateway or SIP-trunk behavior.

Question 142. A user should be allowed to dial internal extensions and local PSTN numbers but must be prevented from accessing international route patterns. Which configuration provides this access control?

1. Partitions and Calling Search Spaces
2. Region bandwidth settings
3. SIP session timers
4. RTP port configuration

Correct Answer: 1. Partitions and Calling Search Spaces

Explanation :-

Partitions and Calling Search Spaces provide logical access control for the Unified Communications Manager dial plan. The administrator can place local PSTN patterns in a partition accessible to the user while keeping international patterns in a separate partition that is excluded from the user’s CSS. Region settings affect codec and bandwidth relationships, session timers govern SIP dialog refresh behavior, and RTP port configuration affects media transport. When implementing this restriction, both line-level and device-level CSS settings should be reviewed. Testing should include internal, permitted PSTN, and restricted international destinations to confirm the intended access boundaries.

Question 143. A CUBE router must modify the calling number by adding a site-specific prefix before sending the call to a SIP provider. Which mechanism should be used?

1. SIP OPTIONS
2. Region configuration
3. Voice translation rules and profiles
4. Device Mobility

Correct Answer: 3. Voice translation rules and profiles

Explanation :-

Voice translation rules and profiles provide controlled manipulation of calling and called numbers on Cisco voice gateways and CUBE. A translation rule can add a prefix to the calling number, and a translation profile determines how and where that transformation is applied. SIP OPTIONS is used for reachability or capability checks, region configuration controls codec and bandwidth relationships, and Device Mobility applies location-specific endpoint settings. The engineer should verify whether the transformation is applied to the calling or called number, confirm the matching rule, and inspect the resulting SIP message to ensure the provider receives the expected identity.

Question 144. A SIP trunk is configured for secure signaling, but the connection fails immediately during TLS negotiation. The remote provider’s certificate was recently renewed. What should be checked?

1. DTMF relay method
2. Certificate chain, trust store, and certificate identity
3. Route-group hunting order
4. Location bandwidth

Correct Answer: 2. Certificate chain, trust store, and certificate identity

Explanation :-

A renewed certificate can cause TLS negotiation to fail if the new certificate is not trusted, has an unexpected identity, has an incomplete certificate chain, or has expired or otherwise invalid properties. The engineer should inspect the local trust store, certificate chain, certificate subject or SAN values where applicable, validity period, and the TLS configuration on both sides. DTMF relay controls keypad signaling, route-group hunting affects routing, and location bandwidth affects CAC. TLS diagnostics and packet captures can help determine whether the failure occurs during certificate exchange or later in the SIP session establishment process.

Question 145. Two phones belong to different Unified Communications Manager regions. The administrator wants to control the codec used between those regions. Which configuration should be changed?

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

Correct Answer: 1. Region relationship

Explanation :-

The codec and bandwidth relationship between Unified Communications Manager regions is configured through region settings. Administrators can specify codec preferences for communication between regions according to available WAN capacity and media requirements. Calling Search Spaces determine dial-plan access, route groups provide gateway or trunk selection, and translation patterns manipulate digits. When troubleshooting unexpected codec negotiation, the engineer should check the regions assigned to the endpoints, the codec relationship between those regions, endpoint capabilities, and the actual negotiated codec. Region configuration should be planned alongside location-based CAC when WAN capacity is constrained.

Question 146. A branch router has entered SRST mode. Calls between registered branch phones work, but calls to external PSTN numbers fail. The phones show as registered to the SRST router. What should be checked next?

1. Centralized CUCM route list only
2. Local SRST dial peers and PSTN connectivity
3. Remote device mobility group
4. Unified Communications Manager region codec

Correct Answer: 4. Local SRST dial peers and PSTN connectivity

Explanation :-

Successful phone registration in SRST confirms that local survivable call processing is operating, but it does not guarantee that external call routing is configured. For PSTN failures, the engineer should inspect local SRST dial peers, destination patterns, voice gateway configuration, and connectivity toward the PSTN service. Centralized route lists and region settings are not the primary routing mechanisms while the branch is operating independently. Device Mobility is also unrelated to PSTN dial-peer selection. Testing dial-peer matching and verifying the gateway’s operational status can identify whether the failure is caused by routing or external connectivity.

Question 147. A SIP endpoint receives a 183 Session Progress response with SDP and starts receiving media before the called party answers. Which SIP behavior is being demonstrated?

1. Established-dialog termination
2. Endpoint registration
3. Early media
4. Call transfer

Correct Answer: 2. Early media

Explanation :-

A 183 Session Progress response can contain an SDP offer or answer and establish an early-media path before the final call response. This allows the caller to hear network announcements, tones, or other media while the call is still progressing. Registration uses REGISTER, established-dialog termination normally uses BYE, and call transfer commonly involves REFER. When analyzing early media, engineers should inspect the provisional response, SDP addresses and ports, codec negotiation, and RTP packets. They should also distinguish 183 Session Progress from 180 Ringing, because the latter normally indicates that the called endpoint is being alerted rather than necessarily establishing network-provided media.

Question 148. A SIP dialog has been established and an endpoint needs to terminate the call normally. Which SIP method should it send?

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

Correct Answer: 3. BYE

Explanation :-

BYE is the SIP method used to terminate an established dialog. Once a call has been successfully established, either endpoint can normally send BYE to indicate that the session should end. CANCEL is used to terminate a pending request such as an unanswered INVITE, OPTIONS is generally used for capability or reachability checks, and PRACK acknowledges reliable provisional responses. When troubleshooting unexpected call termination, engineers should identify which endpoint sent the BYE and inspect the preceding signaling, Reason headers, session timers, and any policy or media failures that may have caused the endpoint to terminate the session.

Question 149. A CUBE router receives a SIP INVITE containing a telephone number that must be converted from national format to E.164 format before routing. Which feature should be applied?

1. Voice translation rule and profile
2. SIP OPTIONS
3. Region configuration
4. Device pool

Correct Answer: 4. Voice translation rule and profile

Explanation :-

Voice translation rules and profiles allow Cisco voice gateways and CUBE to manipulate telephone numbers into the format required by the next stage of call routing. A national-format number can be transformed into an E.164 representation by adding or changing the appropriate digits. SIP OPTIONS performs reachability or capability checks, region configuration controls media policy, and device pools provide common endpoint settings. The engineer should verify the incoming number, define the correct translation rule, apply it to the appropriate calling or called-number context, and then confirm that the resulting E.164 number matches the intended outbound dial peer or Unified Communications Manager route pattern.

Question 150. A CUBE router has separate internal and external interfaces. The SIP provider receives the enterprise’s internal IP address in SDP instead of the reachable public address. Which area should be investigated?

1. Calling Search Space
2. CUBE media bindings and NAT behavior
3. Route pattern partition
4. Device Mobility

Correct Answer: 2. CUBE media bindings and NAT behavior

Explanation :-

SDP contains the address and port information that the remote endpoint uses for RTP. If CUBE advertises an internal private address to an external provider, the media path can fail even though SIP signaling succeeds. The engineer should investigate CUBE media-interface selection, signaling and media bindings, NAT configuration, and the resulting SDP. Calling Search Spaces and route-pattern partitions affect call routing, while Device Mobility controls endpoint location behavior. A SIP trace can confirm exactly which address CUBE advertises, and an RTP capture can determine whether packets are subsequently sent toward the expected interface.

Question 151. A company wants to use one logical routing destination with a primary SIP trunk and a secondary SIP trunk. If the primary path is unavailable, the secondary should be attempted. Which design is appropriate?

1. Route list containing ordered route groups
2. Translation pattern with two partitions
3. Device pool with two regions
4. Calling Search Space with two CSS entries

Correct Answer:3. Route list containing ordered route groups

Explanation :-

A route list provides a logical routing destination and can reference multiple route groups. The route groups can be arranged according to the desired routing order so that Unified Communications Manager attempts the preferred path before an alternate path. Translation patterns manipulate digits, device pools group common endpoint configuration, and Calling Search Spaces control partition access. When implementing trunk redundancy, the engineer should verify that the route pattern points to the correct route list and that each route group contains the intended SIP trunk or gateway. Failover testing should confirm that the secondary path is selected under the appropriate failure conditions.

Question 152. A user moves from one office to another and needs the phone to retain its extension while receiving the new site’s region, location, and calling settings. Which feature supports this requirement?

1. SIP profile
2. Device Mobility
3. Route group
4. DTMF relay

Correct Answer: 1. Device Mobility

Explanation :-

Device Mobility allows a supported Cisco IP phone to retain its identity while receiving location-specific configuration after moving between sites. Depending on the design, settings such as device pool, region, location, and calling search space can change according to the phone’s new physical location. SIP profiles manipulate SIP signaling, route groups organize gateway or trunk selection, and DTMF relay handles keypad digits. When Device Mobility does not produce the expected settings, the administrator should verify the device mobility group, physical locations, device pools, and the information used to determine the phone’s current location.

Question 153. A service provider requires a custom SIP header to be inserted into outbound INVITEs from CUBE. Which feature is designed for this type of signaling manipulation?

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

Correct Answer: 3. SIP profile or normalization

Explanation :-

SIP profiles and normalization mechanisms are designed to modify SIP signaling, including selected headers and message content, when interoperability requires behavior different from the default implementation. If a provider requires a custom header in outbound INVITEs, the engineer should identify the exact header format and implement a narrowly scoped SIP manipulation rule. Location CAC manages media admission, route groups select gateways or trunks, and region configuration manages codec and bandwidth relationships. A SIP capture should be taken before and after the change to confirm that the required header is present without unintentionally altering unrelated SIP messages.

Question 154. A WAN link between two collaboration sites has a defined voice bandwidth limit. Once that capacity is consumed, additional calls should not be admitted across the link. Which feature provides this control?

1. SIP URI matching
2. Location-based Call Admission Control
3. Translation rules
4. Device Mobility

Correct Answer: 4. Location-based Call Admission Control

Explanation :-

Location-based Call Admission Control allows Unified Communications Manager to manage available media bandwidth between locations. When the configured allocation is consumed, additional calls can be prevented from using more bandwidth than the WAN design permits. SIP URI matching handles SIP routing decisions, translation rules manipulate telephone numbers, and Device Mobility applies location-specific endpoint settings. Engineers should configure CAC using realistic bandwidth values and account for the codec requirements of calls crossing the WAN. Verification should include test calls and monitoring to confirm that admission behavior matches the intended capacity policy.

Question 155. A SIP provider requires DTMF digits to be transported using RTP-NTE rather than in-band audio. Which configuration area should the engineer investigate?

1. DTMF relay configuration
2. Region relationship
3. Route-pattern partition
4. Device pool

Correct Answer: 2. DTMF relay configuration

Explanation :-

DTMF relay determines how keypad digits are transported between voice systems. RTP-NTE, commonly associated with RFC 2833/4733-style event signaling, carries DTMF events separately from normal voice audio. If a provider specifically requires RTP-NTE, the engineer should verify that the enterprise gateway, CUBE, SIP trunk, and provider are configured for compatible DTMF signaling. Region relationships affect codec and bandwidth policy, route-pattern partitions affect dial-plan access, and device pools provide shared endpoint settings. A packet capture can confirm whether the expected RTP telephone-event packets are actually being generated and received.

Question 156. A SIP trunk shows successful call setup, but media fails only after a call is transferred to another endpoint. What should the engineer inspect first?

1. Mid-call SIP signaling and SDP renegotiation
2. DHCP option 150
3. Route-pattern partition
4. Device-pool time zone

Correct Answer: 1. Mid-call SIP signaling and SDP renegotiation

Explanation :-

Call transfers can introduce a new media path and may require SIP and SDP changes. If audio works before the transfer but fails afterward, the engineer should compare the signaling and SDP before and after the transfer. The investigation should include RTP addresses and ports, CUBE media handling, NAT, firewall policies, and the endpoint selected after the transfer. DHCP option 150 concerns phone provisioning, route-pattern partitions affect call routing, and device-pool time zones do not control RTP. SIP traces and packet captures are particularly useful for identifying an incorrect post-transfer media address or unreachable RTP path.

Question 157. A SIP provider’s SBC sends a 401 response to an INVITE, and the enterprise is expected to retry the request with authentication credentials. Which SIP mechanism is involved?

1. Session timer refresh
2. SIP digest authentication
3. DTMF relay
4. RTP media negotiation

Correct Answer: 3. SIP digest authentication

Explanation :-

A SIP 401 response commonly indicates that authentication is required and can include a challenge that the client uses to construct an authenticated request. SIP digest authentication uses information from the challenge and configured credentials to generate the appropriate authentication response. Session timers refresh established dialogs, DTMF relay handles keypad signaling, and RTP negotiation concerns media transport. When troubleshooting SIP authentication, the engineer should inspect the 401 response, authentication challenge parameters, configured credentials, realm information, and subsequent authenticated request. Sensitive credentials should not be exposed in troubleshooting captures or logs.

Question 158. A CUBE router must route SIP calls based on the destination domain contained in the Request-URI. Which capability should be used?

1. Location CAC
2. SIP URI-based dial-peer matching
3. SRST fallback
4. DHCP option 150

Correct Answer: 4. SIP URI-based dial-peer matching

Explanation :-

SIP URI-based dial-peer matching allows CUBE to select routing behavior using information contained in SIP URIs, including destination domains. This is useful in deployments where calls are identified by SIP addresses such as user@domain rather than only by numeric destination patterns. Location CAC controls bandwidth admission, SRST provides survivable call processing, and DHCP option 150 assists Cisco phone provisioning. When troubleshooting URI-based routing, the engineer should inspect the actual Request-URI, matching criteria, dial-peer configuration, and session target. The configured pattern must correspond to the URI format actually received from the originating system.

Question 159. A SIP endpoint sends REGISTER messages to Unified Communications Manager. What is the primary purpose of the REGISTER method?

1. To establish endpoint registration information
2. To terminate an established call
3. To acknowledge a reliable provisional response
4. To transfer an established call

Correct Answer: 1. To establish endpoint registration information

Explanation :-

The SIP REGISTER method allows an endpoint to provide or update its contact information with a registrar. In collaboration environments, registration allows the call-control system to associate an endpoint’s identity with its current contact address so that calls can be delivered appropriately. BYE terminates an established dialog, PRACK acknowledges reliable provisional responses, and REFER is commonly used for referred operations such as call transfer. When troubleshooting registration, engineers should inspect REGISTER requests and responses, authentication challenges, network reachability, configured credentials, and the contact information being presented by the endpoint.

Question 160. A SIP endpoint is configured to use session timers, but the remote system rejects the requested timer interval as too short. Which SIP parameter is especially relevant to negotiating the acceptable minimum interval?

1. Max-Forwards
2. Contact
3. Min-SE
4. User-Agent

Correct Answer: 3. Min-SE

Explanation :-

The Min-SE header communicates the minimum session-expiration interval that a SIP endpoint is willing to accept when session timers are negotiated. If an endpoint requests an interval that is below the remote system’s permitted minimum, the remote system can reject or require adjustment of the requested value. Max-Forwards controls SIP request forwarding limits, Contact identifies a reachable endpoint address, and User-Agent identifies the SIP implementation. When troubleshooting session-timer negotiation, engineers should examine Session-Expires, Min-SE, refresher selection, and the responses exchanged during call establishment.