Visit here for our full Cisco 350-801 exam dumps and practice test questions.
Question 61:
An engineer is configuring a Cisco Unified Communications Manager cluster for high availability. Which feature must be configured to ensure that IP phones can register to a backup server if the primary server fails?
A) Device Pool
B) SRST Reference
C) Device Defaults
D) Survivable Remote Site Telephony
Answer: B
Explanation:
SRST Reference is the key configuration element in Cisco Unified Communications Manager that ensures IP phones can maintain essential functionality when the primary call-processing servers become unavailable. Survivable Remote Site Telephony, or SRST, provides call-processing redundancy for remote or branch site phones that rely on centralized Unified Communications Manager servers. In modern enterprise deployments, where IP phones are often distributed across multiple geographic locations and rely on WAN connectivity to the central CUCM cluster, network outages or server failures can leave endpoints unable to place or receive calls. SRST mitigates this risk by allowing phones to fail over automatically to a local gateway router running SRST software, ensuring continuity of basic telephony services even when connection to the primary CUCM environment is lost.
The SRST Reference is configured within the Device Pool associated with each IP phone or group of phones. Device Pools are logical groupings of endpoints that define regional and feature-specific settings such as location, region, date/time group, and media resource group list. Within the Device Pool, the SRST Reference points to a router or gateway configured with SRST, establishing the failover relationship. When an IP phone loses connectivity to all CUCM servers in its configured server list, it immediately enters SRST mode and attempts to register with the designated SRST router. Once registered, the phone can place and receive internal calls, dial external numbers through the PSTN connected to the SRST router, access basic call-forwarding and transfer functions, and even reach voicemail if the router is configured to support it. This automated failover ensures that branch offices and remote users experience minimal disruption, preserving business operations and communication continuity.
While Device Pools are essential for organizing and managing groups of devices, they do not, on their own, provide the failover capability. They simply define the parameters that IP phones inherit during registration and normal operation. Similarly, Device Defaults are templates applied to phones during auto-registration, providing baseline configurations such as line settings, softkey templates, and phone model-specific features. While these templates are important for consistent device behavior, they are not involved in the SRST failover process. The critical link between the Device Pool and survivable call processing is established specifically by the SRST Reference. Without this reference, phones in remote sites will not know which router to contact during a CUCM outage and will be unable to maintain telephony functionality.
SRST is especially valuable in scenarios where branch offices are geographically dispersed and WAN connectivity is subject to interruptions or latency issues. By leveraging SRST Reference configuration, organizations can provide high availability without requiring a full CUCM cluster at every remote location. The SRST-enabled router acts as a lightweight call-processing platform, supporting essential features while the central CUCM servers are unreachable. When connectivity to the primary CUCM environment is restored, the phones automatically re-register with CUCM, transitioning out of SRST mode without requiring manual intervention. This seamless failback preserves call continuity, ensures user satisfaction, and minimizes administrative overhead in distributed deployments.
For enterprises that prioritize high availability and business continuity, configuring SRST Reference in the Device Pool is a critical component of a resilient Cisco collaboration architecture. It guarantees that IP phones retain essential telephony capabilities during WAN outages, CUCM server failures, or maintenance events. By combining the logical organization of Device Pools with the SRST failover mechanism, administrators can implement a scalable, reliable, and fault-tolerant deployment that protects communications across all sites. In summary, SRST Reference is not just a configuration setting—it is a foundational element that ensures enterprise voice services remain operational even under adverse network conditions, safeguarding both productivity and operational continuity.
Question 62:
Which protocol does Cisco Unified Communications Manager use by default for communication between the database publisher and subscriber servers?
A) TCP port 1433
B) TCP port 3306
C) TCP port 5432
D) TCP port 1521
Answer: C
Explanation:
Cisco Unified Communications Manager uses PostgreSQL as its underlying database platform, which serves as the foundation for storing and replicating all configuration and operational data within a CUCM cluster. PostgreSQL is a robust, standards-compliant relational database system, and in CUCM it handles critical information such as device configurations, user accounts, dial plans, route patterns, media resource assignments, and system parameters. The database architecture in CUCM is designed around a publisher-subscriber model, where the publisher server maintains the master copy of the database and all configuration changes are written to it. Subscriber servers, in turn, maintain read-only replicas of this database, which are continuously synchronized with the publisher to ensure consistency and reliability across the cluster. This replication process is essential to maintain a coherent operational environment where every node can process calls, enforce policies, and provide services without discrepancies or conflicts.
The communication between CUCM servers for database replication relies on PostgreSQL using its default TCP port, 5432. This port provides a dedicated channel through which the publisher transmits database updates, schema changes, and configuration modifications to all subscribers. Whenever an administrator makes a change in the CUCM administration interface—such as adding a new device, updating a user profile, modifying a dial plan, or adjusting a service parameter—the change is first committed to the publisher’s database. PostgreSQL then propagates the updates to all subscriber nodes over TCP port 5432, ensuring that each node’s local database copy remains in sync. This continuous replication process guarantees that all call processing nodes operate with consistent information, enabling seamless call routing, feature enforcement, and endpoint registration throughout the enterprise network. Without reliable replication, inconsistencies could arise, leading to call failures, misrouted traffic, or feature discrepancies across devices.
Understanding the correct port for database communication is also critical from a network and security perspective. TCP port 5432 is specific to PostgreSQL, whereas other common database platforms use different default ports: TCP port 1433 is associated with Microsoft SQL Server, TCP port 3306 is the default for MySQL servers, and TCP port 1521 is used by Oracle databases. Cisco’s transition from the IBM Informix database to PostgreSQL in later CUCM versions makes it particularly important for administrators and network engineers to ensure that PostgreSQL’s port is properly configured in firewalls and network policies. Failing to allow traffic on port 5432 between the publisher and subscriber servers will block replication, potentially causing significant operational issues including outdated configurations, registration failures for IP phones, and service disruptions for end users.
From an operational standpoint, proper firewall configuration, network monitoring, and port management are essential for maintaining database health and cluster stability. Administrators should verify that TCP port 5432 is consistently open and that no intermediate network devices are blocking or throttling traffic between the publisher and subscribers. Monitoring replication status through CUCM administration and diagnostic tools can help identify replication delays or failures, which often indicate issues with network connectivity, firewall rules, or database service availability. By maintaining uninterrupted communication on this port, organizations ensure that the cluster remains resilient, all nodes operate with synchronized configurations, and users experience uninterrupted telephony and collaboration services.
Question 63:
An administrator needs to configure a SIP trunk between Cisco Unified Communications Manager and a service provider. Which two parameters are required in the SIP trunk configuration? (Choose two)
A) Destination Address
B) Media Termination Point Required
C) SIP Trunk Security Profile
D) Run On All Active Unified CM Nodes
E) CSS for Device
Answer: A, C
Explanation:
When configuring a SIP trunk between Cisco Unified Communications Manager and a service provider, two essential parameters must be configured: Destination Address and SIP Trunk Security Profile. These are fundamental requirements for establishing basic SIP trunk connectivity and ensuring proper call routing and security.
The Destination Address specifies where CUCM should send SIP messages for outbound calls. This is typically the IP address or FQDN of the service provider SIP proxy or session border controller. Without this parameter, CUCM would not know where to direct calls destined for the service provider network. The destination can be configured as a single address or multiple addresses for redundancy.
The SIP Trunk Security Profile defines the security settings for the trunk including whether to use TCP, UDP, or TLS for transport, which ports to use, and other security-related parameters. Every SIP trunk must have a security profile assigned, even if using non-secure transport. The security profile ensures that CUCM knows how to format and secure SIP signaling traffic appropriately for the service provider requirements.
While Media Termination Point Required can be useful for certain codec or supplementary service scenarios, it is not mandatory for basic trunk operation. Run On All Active Unified CM Nodes determines which CUCM servers can use the trunk and affects redundancy but is not required for initial configuration. CSS for Device controls calling permissions but is optional depending on the dial plan design.
Understanding these required parameters ensures successful SIP trunk deployment and proper integration with service provider networks.
Question 64:
Which Cisco Unified Communications Manager service must be activated to support Extension Mobility functionality?
A) Cisco TFTP
B) Cisco Extension Mobility
C) Cisco IP Voice Media Streaming App
D) Cisco CTIManager
Answer: B
Explanation:
The Cisco Extension Mobility service must be activated on Cisco Unified Communications Manager to support Extension Mobility functionality. Extension Mobility allows users to temporarily access their personal phone configuration including line appearances, speed dials, and services from any IP phone within the system. This is particularly valuable in hot-desking environments where users do not have assigned desks or phones.
When a user logs into an Extension Mobility-enabled phone using their user ID and PIN, the Cisco Extension Mobility service authenticates the user and downloads their specific device profile to that physical phone. The phone configuration changes to match the user profile, making the phone function as if it were the user own device. When the user logs out, the phone returns to its default configuration or logout profile.
The Cisco Extension Mobility service runs as an application on CUCM and must be activated through the Cisco Unified Serviceability interface. Without this service running, users cannot log in to phones even if device profiles are properly configured. The service handles authentication, profile retrieval, and device configuration updates during login and logout operations.
Cisco TFTP provides configuration files to phones but does not handle Extension Mobility authentication. Cisco IP Voice Media Streaming App provides media resources like conferencing and music on hold. Cisco CTIManager supports computer telephony integration applications but is not directly required for Extension Mobility, though it may be used by related applications.
Proper activation and configuration of the Cisco Extension Mobility service is essential for organizations implementing flexible workspace solutions.
Question 65:
What is the maximum number of directory numbers that can be configured on a single Cisco IP Phone 8861 without using expansion modules?
A) 4
B) 6
C) 8
D) 10
Answer: D
Explanation:
The Cisco IP Phone 8861 supports a maximum of 10 directory numbers (lines) without any expansion modules attached. This is a significant feature that makes the 8861 suitable for power users, executives, and administrative assistants who need to manage multiple lines from a single device. The phone has a 5-inch high-resolution color display that can accommodate viewing multiple line appearances simultaneously.
Each directory number configured on the phone can have its own unique phone number, allowing users to answer calls for different departments, handle multiple roles, or manage both personal and business lines from one device. The phone buttons can be configured flexibly, with some serving as line buttons and others as speed dials or feature buttons according to organizational needs.
The 10-line capacity applies to the base configuration without expansion modules. If additional line appearances or speed dial buttons are needed beyond this capacity, administrators can add Key Expansion Modules (KEMs) to the phone. Each KEM provides additional buttons that can be configured for lines or features, extending the phone capabilities even further for users with demanding requirements.
Understanding device capabilities is important for proper phone deployment planning. Assigning phones with appropriate line capacity ensures users have the tools they need without over-provisioning expensive hardware. For users requiring fewer than 10 lines, lower-model phones may be more cost-effective, while users needing more than 10 lines should be provisioned with expansion modules or alternative solutions like attendant consoles.
Question 66:
An engineer is troubleshooting one-way audio issues on SIP calls through a Cisco Unified Border Element. Which command should be used to verify that media packets are being received?
A) show call active voice brief
B) show sip-ua calls
C) debug ccsip messages
D) show voice call summary
Answer: A
Explanation:
The show call active voice brief command is the most effective command for verifying media packet flow and troubleshooting one-way audio issues on a Cisco Unified Border Element. This command displays active voice calls and includes critical information about the media path, including packet counts, codec information, and IP addresses for both signaling and media streams. By examining the transmit and receive packet counts, engineers can quickly determine if media is flowing in both directions.
One-way audio typically occurs when media packets are being sent in only one direction or when return path media is being blocked or misrouted. The show call active voice brief output shows separate RTP statistics for each call leg, allowing engineers to identify which direction is experiencing problems. If the receive packet count is zero or not incrementing, it indicates that media is not reaching the CUBE, pointing to potential firewall issues, routing problems, or incorrect media IP addressing.
The show sip-ua calls command displays SIP call information but focuses primarily on signaling rather than media statistics. The debug ccsip messages command is useful for troubleshooting SIP signaling issues but does not provide real-time media flow information and can be CPU-intensive. The show voice call summary command provides a high-level overview of call statistics but lacks the detailed per-call media information needed for troubleshooting specific audio path issues.
When troubleshooting voice quality or audio path problems, starting with show call active voice brief allows engineers to quickly assess whether the issue is related to media path connectivity, codec negotiation, or packet loss, making it an essential first step in CUBE troubleshooting.
Question 67:
Which three components are required to configure a basic hunt group in Cisco Unified Communications Manager? (Choose three)
A) Line Group
B) Hunt List
C) Hunt Pilot
D) Route Pattern
E) Translation Pattern
F) Call Pickup Group
Answer: A, B, C
Explanation:
Configuring a basic hunt group in Cisco Unified Communications Manager requires three essential components working together: Line Group, Hunt List, and Hunt Pilot. These three elements form a hierarchical structure that determines how incoming calls are distributed among a group of available directory numbers, commonly used for department lines, customer service teams, or help desk operations.
The Line Group is the foundation component that contains the actual directory numbers (phone lines) that will receive the distributed calls. Within a Line Group, administrators define the distribution algorithm such as Top Down, Circular, Longest Idle Time, or Broadcast. The Line Group also specifies whether calls should hunt to the next available member after a timeout or busy condition, making it the core element that groups destination numbers together.
The Hunt List provides the next layer of organization by containing one or more Line Groups. Hunt Lists allow administrators to create prioritized lists of Line Groups, enabling more complex call routing scenarios. For example, a Hunt List might contain a primary Line Group of local agents and a secondary Line Group of remote agents, ensuring local resources are tried first before routing to remote locations.
The Hunt Pilot is the directory number that callers dial to reach the hunt group. It associates an incoming number with a specific Hunt List, completing the call routing chain. When someone dials the Hunt Pilot number, CUCM uses the associated Hunt List to determine which Line Groups to try and how to distribute the call among available members.
Route Patterns and Translation Patterns serve different purposes in call routing and are not required components of hunt group configuration.
Question 68:
An administrator configures a SIP trunk to a service provider that requires authentication. Where should the SIP credentials be configured in Cisco Unified Communications Manager?
A) SIP Profile
B) SIP Trunk Security Profile
C) SIP Trunk configuration page
D) SIP Route Pattern
Answer: C
Explanation:
SIP credentials for service provider authentication should be configured directly on the SIP Trunk configuration page in Cisco Unified Communications Manager. The trunk configuration page contains dedicated fields for authentication credentials including Digest Authentication username and password, which are used when the service provider challenges CUCM to authenticate itself before completing call setup.
When a service provider requires authentication, they typically use SIP digest authentication, which is a challenge-response mechanism. The service provider SIP server sends a challenge containing a nonce value, and CUCM must respond with properly hashed credentials. The credentials configured on the SIP Trunk page are used to calculate this response hash. Without proper credentials configured, authentication will fail and calls will be rejected by the service provider.
The SIP Trunk configuration page provides fields specifically for trunk authentication including Digest User, Digest Credentials, and options for incoming and outgoing authentication requirements. These settings are distinct from other trunk parameters and specifically handle the authentication exchange with the remote endpoint. Administrators must obtain the correct username and password from the service provider and enter them exactly as specified to ensure successful authentication.
The SIP Profile controls SIP protocol behavior and timing parameters but does not store authentication credentials. The SIP Trunk Security Profile defines transport security and encryption settings but not digest authentication credentials. SIP Route Patterns are used for call routing decisions and do not contain authentication information.
Proper credential configuration on the SIP Trunk page is essential for establishing trusted connections with service providers requiring authentication, ensuring reliable call delivery and preventing unauthorized trunk usage.
Question 69:
Which feature allows Cisco Unified Communications Manager to provide E911 location information to a PSAP based on the physical location of an IP phone?
A) Emergency Responder
B) Mobility Identity
C) Location CAC
D) Device Mobility
Answer: A
Explanation:
Cisco Emergency Responder is the dedicated feature designed to provide accurate E911 location information to Public Safety Answering Points (PSAPs) based on the physical location of IP phones within an enterprise environment. When an employee dials 911 from an IP phone, Emergency Responder automatically determines the phone physical location and provides this information to emergency services, ensuring first responders can quickly locate the caller even in large campus or multi-building environments.
Emergency Responder maintains a database that maps each phone to its Emergency Response Location (ERL), which includes detailed physical location information such as building name, floor number, room number, and even specific areas within floors. This granular location tracking is critical because unlike traditional telephony where location is tied to physical wiring, IP phones can be moved to different network locations without updating emergency location information. Emergency Responder solves this by tracking phones based on their network connection points.
When a 911 call is placed, Emergency Responder intercepts the call and performs several actions: it routes the call to the appropriate PSAP based on the caller location, sends location information to the PSAP, sends email notifications to security personnel, and can trigger alerts on IP phones near the emergency location. This comprehensive approach ensures rapid emergency response while maintaining compliance with regulations like Kari’s Law and RAY BAUM’s Act.
Mobility Identity relates to mobile device integration, Location CAC manages bandwidth for call admission control, and Device Mobility handles roaming phone configuration updates. None of these provide the specialized E911 location tracking and PSAP notification capabilities that Emergency Responder delivers, making it the essential solution for enterprise emergency calling compliance and safety.
Question 70:
What is the purpose of configuring a Media Resource Group List (MRGL) in Cisco Unified Communications Manager?
A) To prioritize which media resources a device can access
B) To configure bandwidth limitations for calls
C) To define hunt group behavior
D) To specify emergency call routing
Answer: A
Explanation:
A Media Resource Group List (MRGL) in Cisco Unified Communications Manager serves to prioritize and control which media resources a device can access during call operations. Media resources include conference bridges, transcoders, media termination points (MTPs), music on hold servers, and annunciators. The MRGL provides an ordered list of Media Resource Groups (MRGs), allowing administrators to create a prioritized hierarchy that determines which resources devices attempt to use first when media resources are needed.
The MRGL architecture works in a two-tier model: Media Resource Groups contain the actual media resource devices, while Media Resource Group Lists contain multiple MRGs in a prioritized order. When a phone or gateway needs a media resource (for example, when a user initiates a conference call), CUCM consults the device assigned MRGL and attempts to allocate resources from the first MRG in the list. If resources in the first MRG are unavailable or exhausted, CUCM moves to the next MRG in the MRGL, continuing until a resource is found or all options are exhausted.
This prioritization mechanism enables sophisticated resource management strategies. For example, administrators can configure local high-quality conference bridges as the first choice for a site, with centralized conference bridges as backup options. This ensures optimal media quality by keeping media local when possible while maintaining service availability through backup resources. MRGLs can be assigned to devices, device pools, or configured as the system default, providing flexible resource allocation policies across the organization.
Understanding MRGLs is crucial for designing scalable CUCM deployments that efficiently utilize media resources while ensuring consistent user experience across distributed environments with varying resource availability.
Question 71:
An engineer is configuring QoS for voice traffic on a Cisco router. Which DSCP value should be used to mark voice bearer traffic?
A) EF (46)
B) AF41 (34)
C) CS3 (24)
D) CS5 (40)
Answer: A
Explanation:
Voice bearer traffic should be marked with DSCP EF (Expedited Forwarding), which has a decimal value of 46. This is the industry-standard marking for voice RTP streams and is recommended by Cisco and the IETF for all voice media packets. The EF marking ensures that voice packets receive the highest priority treatment in terms of low latency, minimal jitter, and guaranteed bandwidth as they traverse the network infrastructure.
Voice bearer traffic refers to the actual Real-Time Transport Protocol (RTP) packets that carry the digitized voice conversation between endpoints. These packets are extremely sensitive to delay, jitter, and packet loss because human conversation requires real-time delivery for natural communication. Even small delays of 150-200 milliseconds become noticeable to users, and packet loss above 1 percent significantly degrades voice quality. Therefore, marking voice with DSCP EF places it in the highest priority queue at each network hop.
The other DSCP values serve different purposes in the QoS framework. AF41 (34) is typically used for video conferencing traffic, which also requires preferential treatment but has different characteristics than voice. CS3 (24) is commonly used for call signaling protocols like SIP and SCCP, which are important but less time-sensitive than the actual media stream. CS5 (40) is often used for video signaling or broadcast video traffic depending on the QoS design.
Proper DSCP marking must be configured on voice endpoints, gateways, and collaboration applications, and network infrastructure must be configured to recognize and prioritize EF-marked traffic. This end-to-end QoS implementation ensures consistent, high-quality voice communications across the network, making DSCP EF marking a fundamental requirement for any voice deployment.
Question 72:
Which protocol does Cisco Unified Communications Manager use for phone registration and call signaling with IP phones?
A)323
B) MGCP
C) SIP or SCCP
D)248
Answer: C
Explanation:
Cisco Unified Communications Manager supports two primary protocols for phone registration and call signaling with IP phones: SIP (Session Initiation Protocol) and SCCP (Skinny Client Control Protocol). Modern Cisco IP phones can typically support both protocols, and administrators choose which protocol to use based on organizational requirements, feature needs, and deployment preferences. CUCM communicates with phones using whichever protocol is configured for that specific device model and firmware version.
SCCP, also known as Skinny, is a Cisco proprietary protocol that was the original signaling method for Cisco IP phones. SCCP uses a client-server model where the phone acts as a thin client and CUCM maintains complete control over call processing, media connections, and phone features. SCCP is considered simpler to troubleshoot due to its straightforward command structure and has historically provided faster feature availability for new capabilities in CUCM. The protocol uses TCP port 2000 for communication between phones and CUCM.
SIP is an industry-standard protocol that has become increasingly prevalent in Cisco deployments. SIP provides greater interoperability with third-party systems and is required for certain advanced features and integration scenarios. SIP phones maintain more intelligence at the endpoint and use a peer-to-peer signaling model. Modern Cisco collaboration architecture increasingly favors SIP as the preferred protocol, with newer phone models and features often receiving SIP support first or exclusively.
H.323 is a legacy VoIP protocol rarely used for phone registration in modern deployments. MGCP is used for controlling gateways, not IP phones. H.248 (Megaco) is another gateway control protocol. Understanding that CUCM uses SIP or SCCP for phone communication is essential for proper phone configuration, troubleshooting, and feature planning in Cisco collaboration environments.
Question 73:
An administrator needs to configure Cisco Unity Connection to integrate with Microsoft Exchange for unified messaging. Which protocol is used for calendar integration?
A) IMAP
B) SMTP
C) Exchange Web Services (EWS)
D) MAPI
Answer: C
Explanation:
Cisco Unity Connection uses Exchange Web Services (EWS) protocol for calendar integration with Microsoft Exchange servers. EWS is a web-based API that provides programmatic access to Exchange mailbox data including calendar information, contacts, and mail items. This integration allows Unity Connection to access user calendar data for features like Single Inbox, visual voicemail synchronization, and presence-based call handling based on calendar status.
EWS provides a robust and feature-rich integration method that works across modern Exchange versions including Exchange 2013, 2016, 2019, and Microsoft 365. Through EWS, Unity Connection can read calendar appointments to determine if users are in meetings, update their presence status accordingly, and route calls based on calendar availability. The protocol uses HTTPS for secure communication and supports authentication methods including basic authentication, NTLM, and OAuth depending on the Exchange version and configuration.
When configuring EWS integration, administrators must provide Unity Connection with credentials that have appropriate permissions to access user mailboxes, configure the Exchange server connection details, and enable the desired unified messaging features. The integration also allows voicemail messages to appear in Outlook alongside email, providing users with a single interface for all messaging types.
IMAP is used for retrieving email messages but does not provide calendar access functionality. SMTP is used for sending email including voicemail notifications but not for reading calendar data. MAPI (Messaging Application Programming Interface) is an older Microsoft protocol that was used in legacy Unity Connection versions but has been replaced by EWS in modern deployments due to better compatibility, security, and feature support.
Question 74:
Which Cisco Unified Communications Manager feature allows phones to download their configuration automatically based on DHCP Option 150?
A) Auto-registration
B) TFTP service
C) BAT tool
D) Extension Mobility
Answer: B
Explanation:
The TFTP (Trivial File Transfer Protocol) service in Cisco Unified Communications Manager works in conjunction with DHCP Option 150 to allow phones to automatically download their configuration files during the boot process. DHCP Option 150 provides the IP address of one or more TFTP servers to the phone when it receives its IP address assignment. The phone then contacts the TFTP server to download its configuration file, firmware, and other necessary files to become operational.
The process works as follows: when a Cisco IP phone boots up, it sends a DHCP request to obtain network parameters. The DHCP server responds with an IP address, subnet mask, default gateway, and Option 150 containing the TFTP server address. The phone then sends a TFTP request to download its configuration file, which is named based on the phone MAC address (for example, SEP00112233AABB.cnf.xml). This configuration file contains all settings the phone needs including CUCM server addresses, firmware version, button layouts, network settings, and feature configurations.
The TFTP service must be activated on CUCM servers for this automated provisioning to work. In most deployments, TFTP runs on multiple CUCM servers for redundancy, and Option 150 can specify multiple TFTP server addresses. The phone will try each server in sequence until it successfully downloads its configuration, providing fault tolerance in the provisioning process.
Auto-registration is a separate feature that allows phones to register with CUCM without pre-configuration but does not handle configuration file delivery. The BAT tool is used for bulk phone administration but does not provide the runtime configuration download mechanism. Extension Mobility allows user profiles to move between phones but relies on TFTP for the underlying phone configuration delivery.
Question 75:
What is the function of a Route Pattern in Cisco Unified Communications Manager?
A) To define where calls to specific numbers should be routed
B) To configure media resources for conferences
C) To set up hunt groups for incoming calls
D) To manage phone firmware versions
Answer: A
Explanation:
Route Patterns in Cisco Unified Communications Manager define where calls to specific dialed numbers should be routed within the telephony network. A Route Pattern represents a string of digits (or pattern of digits) that, when matched by a dialed number, triggers CUCM to route the call to a specific destination such as a gateway, trunk, or route list. Route Patterns are fundamental building blocks of the CUCM dial plan and determine how calls exit the IP telephony network to reach PSTN, other PBX systems, or remote sites.
Route Patterns use wildcards and special characters to match multiple number variations with a single pattern. For example, a Route Pattern of 9.@ might match any number starting with 9 followed by any additional digits, typically used for external call access. The pattern 1[2-9]XX[2-9]XXXXXX would match North American long distance numbers. This pattern-matching capability allows administrators to create flexible, scalable dial plans without configuring individual routes for every possible destination number.
Each Route Pattern is associated with either a gateway or a Route List, which determines the physical path the call will take. Route Patterns also include settings for digit manipulation (adding or removing digits before sending the call), calling party transformations, call classification (OnNet versus OffNet), and routing restrictions based on calling search spaces. These capabilities make Route Patterns powerful tools for implementing complex enterprise dial plans with proper least-cost routing, toll fraud prevention, and emergency calling support.
Route Patterns work closely with Partition and Calling Search Space architecture to control which users can dial which patterns, implementing toll restriction and class-of-service policies. Understanding Route Pattern configuration is essential for any CUCM administrator responsible for dial plan design and call routing optimization.