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

 

Question 61. A Cisco Unified Communications Manager administrator wants to route calls to different SIP trunks based on the called-number pattern. Which dial-plan object should be used to identify the destination pattern?

1. Route group
2. Route pattern
3. Device pool
4. Media resource group

Correct Answer: 2. Route pattern

Explanation :-

A route pattern identifies a dialed-number pattern and determines how Cisco Unified Communications Manager should process calls matching that pattern. The route pattern can point to a route list, which in turn can reference one or more route groups containing gateways or trunks. Route groups are used for gateway selection rather than initial destination-pattern matching. Device pools provide common device configuration, while media resource groups organize media resources. When designing a scalable dial plan, administrators should separate destination matching from gateway selection so that routing changes can be made without unnecessarily modifying every individual route pattern.

Question 62. A user dials a seven-digit local number, but the PSTN provider requires a ten-digit number. The administrator wants Unified Communications Manager to add the area code before routing the call. Which feature is appropriate?

1. Calling Search Space
2. Route group
3. Device pool
4. Translation pattern

Correct Answer: 4. Translation pattern

Explanation :-

A translation pattern can manipulate a dialed number before the call is routed toward its final destination. In this scenario, the seven-digit local number can be transformed into the ten-digit format required by the PSTN provider. Translation patterns can add, remove, or modify digits according to the configured transformation. Calling Search Spaces determine which partitions a caller can access, route groups select gateways or trunks, and device pools provide common device settings. The administrator should also verify the resulting number against the applicable route pattern and ensure that the translated destination is accessible through the caller’s CSS.

Question 63. An enterprise has two Unified Communications Manager locations connected by a WAN. Calls between the sites should be admitted only while sufficient bandwidth remains available. Which feature should be configured?

1. Location-based call admission control
2. SIP normalization
3. Route pattern digit stripping
4. Device mobility

Correct Answer: 1. Location-based call admission control

Explanation :-

Location-based call admission control allows Unified Communications Manager to manage the amount of audio and video bandwidth available between locations. When the configured bandwidth is consumed, additional calls can be denied rather than allowing new sessions to degrade existing calls. This is especially useful across WAN links with limited capacity. SIP normalization modifies SIP signaling, route-pattern digit manipulation changes dialed numbers, and Device Mobility applies location-specific device settings. Proper CAC design requires accurate bandwidth estimates and appropriate assignment of devices to locations. Engineers should also consider codec selection and actual WAN utilization when validating the configuration.

Question 64. A Cisco IP phone registers successfully after being moved to another office, but it receives the wrong calling search space and region settings. Which feature should be reviewed?

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

Correct Answer: 3. Device Mobility

Explanation :-

Device Mobility allows Unified Communications Manager to apply location-specific configuration when a device registers from a different location. If a phone registers successfully but receives inappropriate calling search space, region, location, or other site-specific settings, the administrator should review Device Mobility configuration, including the device mobility group, physical location, and associated device pool settings. Route lists affect gateway selection, SIP profiles manipulate signaling, and DTMF relay handles keypad signaling. Device Mobility is particularly useful in organizations where users or phones can move between offices and should automatically receive configuration appropriate to the new site.

Question 65. A company wants calls from a branch to use a local PSTN gateway whenever possible and use a centralized gateway only when the local gateway is unavailable. Which configuration best supports this requirement?

1. Route list with ordered route groups
2. Region configuration only
3. Device mobility group
4. SIP normalization profile

Correct Answer: 1. Route list with ordered route groups

Explanation :-

A route list can reference multiple route groups and provide an ordered path for call routing. By placing the local gateway’s route group before the centralized gateway’s route group, Unified Communications Manager can attempt the preferred local path first and use an alternate path according to the configured routing behavior when the preferred path cannot be used. Region configuration controls bandwidth and codec relationships, Device Mobility controls location-specific device configuration, and SIP normalization changes signaling. Route lists and route groups therefore provide the appropriate architecture for implementing primary and alternate gateway selection.

Question 66. Two Unified Communications Manager sites have different WAN capacities. The administrator wants calls between them to use a compressed codec while calls inside each site can use G.711. Which configuration should be reviewed?

1. Partition and CSS
2. Route list
3. Region configuration
4. Translation pattern

Correct Answer: 3. Region configuration

Explanation :-

Regions in Unified Communications Manager define codec and bandwidth relationships between devices in different regions. An administrator can configure a compressed codec for communication across a constrained WAN while permitting a higher-bandwidth codec for local communication. Partitions and Calling Search Spaces control dial-plan access, route lists determine gateway selection, and translation patterns manipulate digits. When designing regional codec policies, administrators should consider actual WAN capacity, application requirements, codec bandwidth, and the desired quality level. Verifying the negotiated codec in active calls is also important because endpoint capabilities and other configuration factors can affect the final media choice.

Question 67. A user can call internal numbers but cannot call a PSTN route pattern. Another user on the same system can successfully place the PSTN call. What should the engineer compare first?

1. RTP packet size
2. SIP response timeout
3. Device firmware version
4. Calling Search Space and partition access

Correct Answer: 4. Calling Search Space and partition access

Explanation :-

When one user can access a route pattern and another cannot, the Calling Search Space and partition relationship should be examined first. The PSTN route pattern may reside in a partition that is included in one user’s CSS but not the other’s. Comparing the working and non-working devices or lines can reveal differences in CSS, line configuration, or other dial-plan permissions. RTP packet size and SIP response timers are generally relevant after call routing has occurred, while firmware differences are less likely to explain a specific partition-access problem. CSS and partition design is fundamental to implementing controlled calling privileges.

Question 68. A Unified Communications Manager administrator wants to prevent ordinary users from dialing premium-rate numbers while allowing a designated group of users to reach them. Which design is most appropriate?

1. Configure different regions for the user groups
2. Use separate partitions and Calling Search Spaces
3. Configure different RTP port ranges
4. Change the SIP codec preference

Correct Answer: 2. Use separate partitions and Calling Search Spaces

Explanation :-

Partitions and Calling Search Spaces provide logical access control for Unified Communications Manager dial plans. Premium-rate route patterns can be placed in a restricted partition, while ordinary users receive a CSS that does not include that partition. Authorized users can be assigned a CSS that includes it. Regions affect codec and bandwidth relationships, RTP port ranges concern media transport, and codec preferences influence media negotiation. This partition/CSS design is flexible because the restricted route patterns can remain available in the overall dial plan without being exposed to every user. Careful CSS inheritance and line-level configuration should also be checked.

Question 69. A CUBE router receives an INVITE from a provider, but the internal Unified Communications Manager rejects the call because the called number uses an unexpected format. Which CUBE feature can transform the digits before forwarding the call?

1. Translation rules and profiles
2. NTP authentication
3. OSPF redistribution
4. SRTP encryption

Correct Answer: 4. Translation rules and profiles

Explanation :-

CUBE can use digit manipulation mechanisms such as translation rules and translation profiles to transform telephone numbers between different numbering plans. This is useful when a service provider sends numbers in a format that differs from the internal Unified Communications Manager dial plan. The engineer should examine the incoming called number, determine the required transformation, and apply the translation at the appropriate dial-peer or voice-service point. NTP authentication, OSPF redistribution, and SRTP encryption serve different functions and do not directly transform telephone digits. Testing should verify both the translated number and the resulting dial-peer match.

Question 70. A SIP provider requires a specific value in a SIP header that is not generated by the enterprise’s default configuration. Which CUBE capability can insert or modify that header?

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

Correct Answer: 1. Location CAC

Explanation :-

The appropriate mechanism for inserting or modifying SIP headers is a SIP profile or SIP normalization feature, not location-based call admission control. SIP manipulation capabilities allow selected SIP messages and headers to be transformed as they pass through CUBE, which is useful for service-provider interoperability. Location CAC controls available bandwidth between collaboration locations, Device Mobility changes device configuration based on location, and route groups provide gateway-selection logic. The engineer should first capture the SIP signaling to identify the exact header and value required by the provider, then apply a narrowly scoped normalization rule.

Question 71. A CUBE deployment has separate internal and external interfaces. Calls establish, but external users experience one-way audio. The SDP contains an internal private IP address that the provider cannot reach. Which area should be investigated?

1. Calling Search Space
2. Media address/interface selection on CUBE
3. Route-pattern partition
4. Device mobility group

Correct Answer: 3. Media address/interface selection on CUBE

Explanation :-

One-way audio can occur when SDP advertises a media address that the remote endpoint cannot reach. In a CUBE deployment with separate internal and external interfaces, the engineer should verify media interface selection, bindings, NAT behavior, and the addresses presented in SDP. The external provider must receive a reachable address for the RTP stream. Calling Search Spaces and route-pattern partitions affect call routing, while Device Mobility controls phone configuration. A successful SIP signaling exchange does not guarantee correct media connectivity, so SDP inspection and RTP packet analysis are important troubleshooting techniques for this scenario.

Question 72. A company wants to protect SIP signaling between CUBE and a service provider from interception. Which protocol should be used for SIP signaling encryption?

1. RTP
2. SRTP
3. TLS
4. FTP

Correct Answer: 2. TLS

Explanation :-

Transport Layer Security, or TLS, can be used to secure SIP signaling between supported collaboration systems. TLS provides encryption and authentication for the signaling channel, helping protect SIP messages from interception or manipulation in transit. SRTP is used to protect the actual media stream rather than the SIP signaling channel. RTP provides voice media transport without encryption, while FTP is unrelated to SIP security. A secure SIP deployment also requires appropriate certificates, trust relationships, supported TLS versions, and compatible configuration on both sides of the SIP connection.

Question 73. A company wants voice media to remain confidential after SIP signaling has been secured with TLS. Which technology should additionally be considered?

1. SRTP
2. DHCP
3. OSPF
4. TFTP

Correct Answer: 1. SRTP

Explanation :-

TLS protects SIP signaling, but it does not by itself encrypt the RTP media carrying the conversation. Secure Real-time Transport Protocol, or SRTP, is designed to provide confidentiality and integrity protection for real-time voice and other media streams. Therefore, a deployment requiring protection of both signaling and media commonly uses TLS for SIP signaling and SRTP for media. DHCP and TFTP can support endpoint provisioning, while OSPF provides routing. Engineers should verify that both endpoints and any intermediary voice devices support compatible SRTP parameters and that security negotiation succeeds during call establishment.

Question 74. A SIP trunk remains operational, but calls fail whenever the provider changes its advertised SIP signaling address. Which configuration should be examined for static assumptions about the provider’s signaling destination?

1. Region configuration
2. SIP trunk destination and DNS-related configuration
3. Device mobility settings
4. Media resource group list

Correct Answer: 4. SIP trunk destination and DNS-related configuration

Explanation :-

A SIP trunk may depend on configured destination addresses, hostnames, DNS resolution, or related signaling parameters. If a provider changes its signaling address and calls subsequently fail, the engineer should verify how the trunk resolves and reaches the provider, including destination configuration and DNS behavior where applicable. Region settings affect codec and bandwidth, Device Mobility affects device-specific configuration, and media resource group lists manage media resources. Troubleshooting should include SIP signaling traces, DNS resolution tests when relevant, routing verification, and confirmation that the trunk is sending INVITEs toward the currently reachable provider destination.

Question 75. A Unified Communications Manager administrator needs to allow a user to access a restricted partition but does not want to change the CSS assigned to every device using the same device pool. Which configuration can provide more granular control?

1. Line-level Calling Search Space
2. Region configuration
3. Route group
4. Location bandwidth

Correct Answer: 2. Region configuration

Explanation :-

The appropriate mechanism for granular dial-plan access is a line-level Calling Search Space, rather than region configuration. A CSS can be assigned at different configuration levels, allowing administrators to provide specific users or lines with access to selected partitions without changing the configuration of every device in a common device pool. Regions control codec and bandwidth relationships, route groups control gateway selection, and locations provide call admission control. When implementing exceptions to normal calling privileges, engineers should understand the precedence and interaction of device-level and line-level CSS settings to avoid unintentionally granting or removing access.

Question 76. A user places a call through CUBE, and the provider returns a SIP 403 Forbidden response. Which troubleshooting approach is most appropriate?

1. Immediately increase RTP packet size
2. Inspect SIP authorization, policy, and provider-side rejection details
3. Change the Unified Communications Manager region
4. Replace the phone’s DHCP server

Correct Answer: 3. Inspect SIP authorization, policy, and provider-side rejection details

Explanation :-

A SIP 403 Forbidden response indicates that the request was understood but is not permitted under the receiving system’s policy. The engineer should inspect the SIP signaling, authentication or authorization requirements, source identity, calling privileges, provider policies, and any CUBE configuration that may affect the request. RTP packet size is relevant to media problems rather than a signaling rejection. Region configuration controls codec and bandwidth relationships, and DHCP does not normally determine whether a provider authorizes a SIP call. A detailed SIP trace can help identify which policy or identity attribute caused the rejection.

Question 77. A CUBE administrator wants to restrict inbound SIP traffic so that only calls from approved provider addresses are accepted. Which security control should be considered?

1. Region bandwidth
2. Device mobility
3. Source-based access control and trusted SIP peer restrictions
4. Route-pattern digit stripping

Correct Answer: 4. Source-based access control and trusted SIP peer restrictions

Explanation :-

Restricting SIP traffic to known provider addresses can reduce exposure to unauthorized signaling and toll-fraud attempts. Appropriate source-based access controls, trusted-peer configuration, access control lists, and related CUBE security mechanisms can limit which systems are permitted to communicate with the enterprise voice gateway. Region bandwidth controls call capacity, Device Mobility handles endpoint location changes, and route-pattern digit stripping manipulates dialed numbers. Security policy should be designed so that legitimate provider signaling and media remain available while unsolicited or unauthorized traffic is blocked. Logging and monitoring should also be enabled to identify attempted unauthorized access.

Question 78. A branch phone loses its WAN connection to Unified Communications Manager and successfully enters SRST mode. The administrator wants to determine whether the phone has registered to the local survivable call-processing service. Which information should be checked?

1. Local SRST phone registration status
2. Central route-list order
3. Remote cluster region bandwidth
4. SIP provider codec preference only

Correct Answer: 2. Local SRST phone registration status

Explanation :-

When a phone enters SRST mode after losing connectivity to centralized call control, verifying local registration status is an important troubleshooting step. The administrator should confirm that the phone has successfully registered with the branch router’s SRST service and that the expected directory numbers and call-processing resources are available. Central route-list configuration and remote region bandwidth do not directly establish local SRST registration. Provider codec preferences are also unrelated to whether the phone has successfully transitioned to survivable call processing. Once registration is confirmed, the engineer can investigate local dial peers, PSTN connectivity, and other SRST-specific call-routing behavior.

Question 79. A SIP call is established but terminates unexpectedly after a period of inactivity. SIP traces show that a session refresh was expected but was not received. Which mechanism should be investigated?

1. DTMF relay
2. Session timers
3. Route groups
4. DHCP option 150

Correct Answer: 3. Session timers

Explanation :-

SIP session timers provide periodic refreshes for an established session. If the expected refresh is not received or acknowledged within the negotiated interval, a SIP endpoint may terminate the session because it can no longer confirm that the dialog remains active. Engineers should inspect Session-Expires and Min-SE values, determine the negotiated refresher, and examine the corresponding re-INVITE or UPDATE exchanges. DTMF relay controls keypad signaling, route groups control gateway selection, and DHCP option 150 supports Cisco IP phone provisioning. Predictable termination after a session-refresh failure strongly points toward session-timer behavior.

Question 80. An enterprise needs to troubleshoot a call that establishes signaling but has no audio in either direction. The SIP trace shows valid SDP offers and answers. What should be investigated next?

1. Calling Search Space
2. RTP path, firewall rules, NAT, and media reachability
3. Translation pattern
4. Device mobility group

Correct Answer: 1. Calling Search Space

Explanation :-

The appropriate next step for a no-audio problem after successful SIP signaling and SDP negotiation is to investigate the RTP media path, including routing, firewall rules, NAT, interface selection, and reachability of the advertised media addresses and ports. Calling Search Spaces control dial-plan access and are generally evaluated before a call is established. Translation patterns modify digits, while Device Mobility controls location-specific phone configuration. When both signaling and SDP negotiation succeed but no audio is present, packet captures and RTP statistics can help determine whether media is being generated, transmitted, received, or blocked somewhere along the path.