300-815 Cisco Practice Test Questions and Exam Dumps
Question No:1
You are configuring a Cisco Unified Communications environment using Cisco CallManager Express (CME) integrated with Cisco Unity Express (CUE) for voicemail services. After setting up the SIP trunk to the Public Switched Telephone Network (PSTN), you observe the following issue:
When an external user places a call to a Cisco IP Phone registered with CME and the call is not answered, the caller hears a ringtone for approximately 20 seconds. Instead of being redirected to voicemail as expected, the call ends with a busy signal.
You suspect a configuration issue is preventing calls from routing correctly to Cisco Unity Express. Which configuration command will resolve the issue and allow unanswered SIP trunk calls to be forwarded to voicemail?
A. scss
Router(config)# voice service voip
Router(conf-voi-serv)# allow-connections h323 to h323
B.arduino
Router(config)# dial-peer voice 2 voip
Router(config-dial-peer)# no vad
C.scss
Router(config)# voice service voip
Router(conf-voi-serv)# allow-connections voice-mail
D.bash
Router(config)# voice service voip
Router(conf-voi-serv)# no supplementary-service sip moved-temporarily
Correct Answer:
D.bash
Router(config)# voice service voip
Router(conf-voi-serv)# no supplementary-service sip moved-temporarily
Explanation:
The issue described occurs when a SIP trunk is used to receive inbound calls and the system fails to transfer those calls to voicemail on no answer. In SIP environments, the use of SIP response codes like "Moved Temporarily" (302) can cause problems if the service provider or intermediary device does not properly handle them.
By default, Cisco IOS may use the SIP “302 Moved Temporarily” message to redirect calls—for instance, to a voicemail port. However, many SIP trunk providers do not support this method of redirection, especially for inbound calls. As a result, the provider does not follow the redirection, and instead, the caller experiences a timeout or busy tone.
The command:
perlno supplementary-service sip moved-temporarily
disables the use of the SIP 302 response and forces the CME to handle the call routing internally, rather than relying on the SIP provider to follow the redirect. This ensures that unanswered calls are successfully routed to the voicemail system (Cisco Unity Express) within the same router or voice gateway, resolving the issue.
The other options are unrelated or insufficient:
A enables H.323 to H.323 call routing—not relevant here.
B disables Voice Activity Detection (VAD), which affects audio quality, not call routing.
C is not a valid command; "voice-mail mod" is incorrect syntax.
Therefore, Option D is the correct solution to ensure proper voicemail redirection in a SIP trunk environment with Cisco CME and CUE.
Question No:2
A voice engineer is tasked with configuring a secure SIP trunk between a Cisco Unified Communications Manager (CUCM) system and a remote SIP service provider. The security policy requires that SIP signaling must use port 5065 for both inbound and outbound communication, instead of the standard ports (5060 for SIP or 5061 for secure SIP).
To meet this requirement, the engineer must ensure the SIP trunk and related security configurations are adjusted properly within CUCM to handle SIP TLS traffic on port 5065 in both directions.
Which two configuration elements must the engineer modify to meet this requirement? (Choose two.)
A. Configure the Incoming Port in the SIP Information section of the SIP Trunk configuration.
B. Configure the Incoming Port in the Security Information section of the SIP Profile configuration.
C. Configure the Destination Port in the SIP Information section of the SIP Trunk configuration.
D. Configure the Incoming Port in the SIP Trunk Security Profile configuration.
E. Configure the Destination Port in the SIP Trunk Security Profile configuration.
Correct Answers:
C. Destination Port in SIP Information section of the SIP Trunk configuration
D. Incoming Port in SIP Trunk Security Profile configuration
Explanation:
When configuring a secure SIP trunk in CUCM, communication typically uses port 5061 for SIP over TLS. However, in some deployments—such as those using custom security policies or non-standard provider requirements—an alternate port like 5065 may be specified for both incoming and outgoing SIP signaling.
To meet this requirement:
Outgoing SIP Traffic (CUCM to Provider):
This is controlled by the Destination Port field in the SIP Information section of the SIP trunk configuration. Setting this to 5065 ensures that CUCM sends SIP signaling traffic to the provider's required port.
Incoming SIP Traffic (Provider to CUCM):
The port CUCM listens on for secure SIP messages must be configured in the SIP Trunk Security Profile, not the SIP Trunk itself. Within the Security Profile, the Incoming Port defines which port CUCM will listen on for TLS-encrypted SIP traffic. This also must be set to 5065.
The other options are incorrect:
A and B are misleading—CUCM does not manage SIP listening ports in those areas.
E is invalid because the Security Profile doesn’t control outbound destination ports.
In summary, for secure SIP communication over a custom port, both the SIP trunk's destination port and the security profile's incoming port must be configured to match the provider’s requirements. This ensures two-way signaling over port 5065, maintaining both functionality and compliance with the secure SIP trunk setup.
Question No:3
A network engineer is troubleshooting a SIP trunk call issue in a Cisco Unified Communications Manager (CUCM) environment. The goal is to verify whether SIP signaling messages are reaching the CUCM server and how CUCM is handling the call setup.
To perform this task, the engineer needs to analyze SIP traces, which capture detailed SIP messages such as INVITE, TRYING, RINGING, OK, and BYE, including their headers and payloads.
Which tool within the Cisco Unified Communications infrastructure should the engineer use to collect and analyze SIP trace logs from the CUCM system?
A. MTP (Media Termination Point)
B. CCSIP
C. RTMT (Real-Time Monitoring Tool)
D. OS Administration Page
Correct Answer: C. RTMT (Real-Time Monitoring Tool)
Explanation :
To troubleshoot SIP call flow issues in Cisco Unified Communications Manager, engineers need access to detailed logs of SIP signaling messages. These messages include important call setup and teardown details that help diagnose problems such as one-way audio, call drops, failed registrations, or call routing failures.
The Cisco Unified Real-Time Monitoring Tool (RTMT) is the official utility provided by Cisco to monitor and download performance data, system logs, and SIP traces from CUCM servers. It connects securely to CUCM using admin credentials and allows users to:
Monitor system performance in real time.
Retrieve and view call logs and SIP messages (e.g., INVITE, ACK, BYE).
Filter traces by IP address, call ID, or timestamp for easier debugging.
To view SIP traces in RTMT:
Launch RTMT and connect to the CUCM node.
Navigate to Trace & Log Central.
Select Collect Files.
Choose the Cisco CallManager service logs.
Apply filters for specific time ranges or nodes.
Download and open the logs in a text editor or use a SIP analyzer for better readability.
Let’s look at the other options:
A. MTP (Media Termination Point) is a media resource, not a diagnostic tool.
B. CCSIP is a CLI-based command used on IOS gateways—not CUCM.
D. OS Administration Page is used for system-level settings but not for retrieving SIP logs.
Thus, the RTMT is the correct and comprehensive tool to collect and analyze SIP traces in a CUCM environment.
Question No:4
A network administrator is managing a Cisco Unified Communications Manager (CUCM) deployment and is working with dial patterns to define how digit strings are matched and routed.
Four different route patterns have been configured in the CUCM system as shown below:
12!
12X (with Urgent Priority set)
1XX (with Urgent Priority set)
12[2-5]
A user dials the number "123" from their IP phone.
Given the configuration and behavior of digit analysis in CUCM, which of the above patterns will be selected by the system to route the call?
Correct Answer: B. 12X (urgent priority set)
Explanation :
In Cisco Unified Communications Manager, dial patterns determine how the system matches and routes dialed numbers. CUCM uses a process called digit analysis to compare the dialed digits against all configured patterns.
Here’s how each pattern relates to the dialed number "123":
12! – Matches any digit string starting with "12" followed by one or more digits. It does match "123", but it’s a less specific match compared to others.
12X (Urgent Priority Set) – Matches any three-digit number beginning with "12" and ending with any digit (0–9). Since "123" fits this pattern exactly and Urgent Priority is set, this pattern will be prioritized.
1XX (Urgent Priority Set) – Also matches "123", but it’s less specific than "12X" because it allows any digit in the second and third positions.
12[2–5] – Matches exactly "122", "123", "124", or "125". This does match "123" and is specific, but it lacks Urgent Priority, so it may not be selected first.
CUCM resolves conflicts between multiple matching patterns using a set of rules:
Urgent Priority patterns take precedence over non-urgent ones.
Among urgent patterns, the most specific match (fewest wildcards) is chosen.
In this case, "12X" with Urgent Priority matches "123" and is more specific than "1XX". Though "12[2-5]" is specific, it lacks Urgent Priority. Therefore, CUCM selects "12X (Urgent Priority)" as the best match.
Question No:5
A UC engineer is troubleshooting a complex call-routing issue in Cisco Unified Communications Manager (CUCM) and needs to examine how Translation Patterns are being matched and processed during digit analysis.
To do this, the engineer wants to capture SDL (Signal Distribution Layer) trace logs that include detailed operations performed by Translation Patterns. These logs are critical for analyzing how CUCM processes dialed numbers, applies transformations, and routes calls.
Which specific configuration step must the engineer perform in Cisco Unified CM Serviceability to ensure that Translation Pattern operations are displayed in SDL traces?
A. Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.
B. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.
C. Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.
D. By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.
Correct Answer:
C. Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.
Explanation :
Translation Patterns in Cisco Unified Communications Manager are used to manipulate dialed digits before call routing occurs. When troubleshooting issues involving Translation Patterns, it’s helpful to see how CUCM processes these patterns in SDL traces, which provide low-level logging of system operations.
However, by default, Translation Pattern operations are not shown in SDL traces to avoid excessive log verbosity. To enable visibility into these operations, you need to adjust micro trace settings in Cisco Unified Serviceability.
Here’s how to do it:
Log in to Cisco Unified Serviceability.
Navigate to:Trace > Configuration.
Select the appropriate server and the Cisco CallManager service.
Scroll down to the Micro Traces section.
Check the box for “Translation Patterns Analysis”.
Save the configuration and restart the trace session if needed.
This enables SDL trace logging specifically for Translation Pattern operations. The resulting logs will now show step-by-step details on how dialed digits are matched and manipulated, helping administrators verify call routing and debug any misconfigurations.
Let’s review the other options:
A (Enterprise Parameters) relates to general call detail settings, not traces.
B (Digit Analysis Complexity) affects dial plan processing, but not trace visibility.
D is incorrect—Translation Pattern logs are not enabled by default in SDL traces.
Therefore, the correct method is to enable Translation Patterns Analysis under Micro Traces, making Option C the right answer.
Question No:6
Which two devices can be used to analyze calls with the Cisco Unified Communications Manager Dialed Number Analyzer?
(Choose two.)
A. Translation Patterns
B. Device Pools
C. CTI Ports
D. CTI Route Points
E. IP Phones
Detailed Answer:
The correct answers are C. CTI Ports and E. IP Phones.
Explanation:
The Cisco Unified Communications Manager (CUCM) Dialed Number Analyzer (DNA) is a powerful tool that assists in analyzing and troubleshooting call routing in a Cisco Unified Communications environment. It allows network administrators to simulate and trace dialed numbers through various routing components to ensure that calls are being routed as expected. The tool can be used to verify how a number will be processed based on the system configuration.
The two correct answers, CTI Ports and IP Phones, represent devices involved in call processing that can be analyzed through the DNA tool.
CTI Ports (Option C):
Computer Telephony Integration (CTI) ports are virtual devices used to manage call routing for applications like Cisco's Unified Contact Center. They interact with external systems for call control. CTI ports are critical in integrating applications with the telephony infrastructure and can be analyzed with the Dialed Number Analyzer to trace calls that involve CTI-controlled devices. This can help diagnose issues related to call routing and application interactions.
IP Phones (Option E):
IP phones are physical devices used by end-users to make and receive calls over an IP-based network. In the context of CUCM, IP phones are integral components in a call setup process. The Dialed Number Analyzer can simulate call scenarios involving IP phones, allowing administrators to check how the system processes calls made from these devices and to troubleshoot any problems that might arise, such as incorrect call routing or failed call attempts.
Why the Other Options Are Incorrect:
A. Translation Patterns: While translation patterns play a role in call routing and number manipulation, they are not devices. The DNA tool can simulate their effect on dialing, but they are not considered "devices" in this context.
B. Device Pools: A device pool is a logical grouping of devices that share common characteristics, such as media resources or dial plans, but it is not a physical or virtual device that processes calls directly.
D. CTI Route Points: CTI route points are virtual devices that represent endpoints for CTI-controlled applications. Although they are involved in call routing, the question specifically asks for devices that can be analyzed directly with DNA, which generally applies to devices that generate or receive calls directly (such as CTI ports or IP phones).
Question No:7
In a Cisco Unified Communications Manager environment, when globalized call routing is implemented, which tool should you use to verify its correct implementation without making a call?
(Choose one.)
A. Dialed Number Analyzer
B. Real-Time Monitoring Tool
C. SDI Trace
D. SDL Trace
Correct Answer:The correct answer is A. Dialed Number Analyzer.
Explanation:
In Cisco Unified Communications Manager (CUCM), globalized call routing refers to the standardization of dialing patterns across multiple regions and locations, ensuring that calls are routed correctly regardless of the local dialing rules. When globalized call routing is implemented, it is important to verify that the configuration is correct without actually making a live call. The Dialed Number Analyzer (DNA) is the tool specifically designed for this purpose.
Dialed Number Analyzer (Option A):
The Dialed Number Analyzer is a diagnostic tool in CUCM that allows network administrators to simulate how a dialed number will be processed without making an actual call. This tool can verify how the system interprets and routes a number, taking into account globalized call routing configurations, translation patterns, and other dialing rules. It simulates call behavior and provides feedback on how the dialed number will be handled by the CUCM system, making it the most effective tool for verifying globalized call routing before making any real calls.
Why Other Options Are Incorrect:
B. Real-Time Monitoring Tool (RTMT):
The Real-Time Monitoring Tool is used for monitoring system performance and troubleshooting live issues but does not simulate call routing. It provides insights into the system's real-time health, alerts, and logs but is not intended for verifying call routing configurations.
C. SDI Trace:
SDI (System Diagnostics Interface) Trace is a tool that provides system-level logs for debugging purposes. It is generally used for troubleshooting low-level system issues but is not designed to test or verify call routing configurations, including globalized call routing.
D. SDL Trace:
SDL (Signaling Distribution Layer) Trace is a diagnostic tool that provides detailed logs of signaling-related activities. While useful for troubleshooting signaling issues, it is not specifically intended for verifying the correctness of globalized call routing configurations.
In summary, the Dialed Number Analyzer is the most suitable tool for verifying that globalized call routing is implemented correctly without the need to place actual calls. It simulates the routing behavior and allows administrators to troubleshoot potential issues before any live call is made.
Question No:8
Where would you configure the standard local route group for a group of devices in Cisco Unified Communications Manager?
A. System > Location Info
B. Call Routing > Route/Hunt > Local Route Group Names
C. System > Device Pool
D. Call Routing > Emergency Location > Emergency Location (ELIN) Groups
Correct Answer:The correct answer is B. Call Routing > Route/Hunt > Local Route Group Names.
Explanation:
In Cisco Unified Communications Manager (CUCM), the Local Route Group is a configuration element that helps in determining the route for outbound calls from devices within a specific group, based on the associated local route group for that device group. This configuration allows CUCM to direct calls to the appropriate route or gateway that is part of the local route group, making it critical for managing outbound call routing.
Call Routing > Route/Hunt > Local Route Group Names (Option B):
The Local Route Group Names section under Call Routing > Route/Hunt is where you configure and manage Local Route Groups in CUCM. A local route group can include one or more gateways or trunks, and it helps specify which set of gateways should be used for a particular device or group of devices. By associating a local route group with a device pool, CUCM can determine which gateways are available for outgoing calls based on that device pool configuration. This ensures that calls are routed to the correct destination based on the local route group settings.
Why the Other Options Are Incorrect:
A. System > Location Info:
The Location Info section is used for configuring Location-Based Call Admission Control (CAC), which is responsible for controlling bandwidth between locations and ensuring that voice calls do not overwhelm the network's capacity. It is not related to configuring local route groups for devices.
C. System > Device Pool:
The Device Pool is used for defining a set of devices that share common settings, such as regions, codecs, and call routing preferences. While device pools are critical in managing device behavior, they do not directly define or configure local route groups.
D. Call Routing > Emergency Location > Emergency Location (ELIN) Groups:
Emergency Location (ELIN) Groups are related to configuring emergency services in CUCM, specifically for managing the location information of devices for emergency calls (e.g., 911 calls). This setting is unrelated to the configuration of local route groups for routing outbound calls.
In conclusion, the Call Routing > Route/Hunt > Local Route Group Names section is the correct place to configure local route groups in Cisco Unified Communications Manager. This configuration is vital for managing the routing of calls through the correct gateways and ensuring efficient call routing based on the associated device pools.