61 free Cisco 300-815 CLACC practice questions from the CertStash pack of 245, each with the correct answer and a full explanation. The rule is the same for every exam: ten free on the product page, 61 free here, all 245 in the pack, and each step includes the one before it.
Get all 245 questions (US$39) · Download the first 10 as a PDF
Question 1
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio call. You gather the H.225 and H.245 messages for one of the one-way audio calls. Where can you find the RTP IP and port information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).
Show answer and explanation
Correct answer: D. H.245 Open Logical Channel Ack
In H.323 slow start, each endpoint opens a logical channel with an Open Logical Channel message. The receiving side answers with an Open Logical Channel Ack that contains the IP address and port where it wants to receive RTP. Because each direction has its own OLC Ack, the Acks give the RTP address and port for both sides.
Why the other options are wrong
- A. Terminal Capability Set exchanges supported codecs, not media addresses.
- B. The Open Logical Channel carries the RTCP control address, while the RTP address comes in the Ack.
- C. H.225 Connect in slow start does not carry media addresses.
Question 2
Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)
Show answer and explanation
Correct answer: A, D. DTMF; FAX
For H.323 to SIP interworking in fast start-to-early media scenarios, DTMF and AUDIO extended capabilities must be configured on dial peers. DTMF capability ensures proper digit signaling interoperability, and AUDIO capability is essential for establishing and maintaining voice media streams between the different protocol domains during early media exchange.
Why the other options are wrong
- B. BFCP (Binary Floor Control Protocol) is not a required extended capability for basic H.323 to SIP interworking in early media scenarios.
- C. VIDEO capability is not a standard extended capability requirement for H.323 to SIP interworking in fast start scenarios.
- E. While audio is critical, AUDIO as a standalone extended capability configuration is subsumed within the DTMF and core interworking capabilities.
Question 3
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
Show answer and explanation
Correct answer: A. ALERTING
The ALERTING message in H.323 signaling informs the originating party that the called party is being notified about the incoming call. This message indicates that the called endpoint has received the call setup request and is alerting the user, typically by ringing or other notification means.
Why the other options are wrong
- B. PROCEEDING indicates that the call is being processed but does not specifically indicate that the called party is being notified.
- C. CONNECT is sent when the called party has answered the call and establishes the connection, not when they are merely being notified.
- D. RINGING is not a standard H.323 protocol message; ALERTING serves this function in the H.323 protocol stack.
Question 4
End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?
Show answer and explanation
Correct answer: D. c= line of SDP content
The c= (connection) line in SDP content specifies the IP address that will be used for media (RTP) transmission. When users cannot hear the remote party, the c= line provides critical information about where the media is supposed to be sent and received. If this address is incorrect, unreachable, or doesn't match the actual endpoint, one-way or no audio will result.
Why the other options are wrong
- A. The Contact header in the 200 OK response identifies the endpoint for future signaling, not the media connection address.
- B. The Allow header lists supported SIP methods but provides no information about media connection addresses.
- C. The o= (origin) line contains session metadata like session version and originator information, not media destination addresses.
Question 5
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
Show answer and explanation
Correct answer: A. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
The far endpoint sends its RTP stream to whatever IP address and port are advertised in the SDP connection data (c=) line. If a device doing deep packet inspection, such as a firewall with SIP ALG, rewrites that line incorrectly, the originating endpoint sends media to the wrong address and the far end never receives it. This is a classic cause of one-way or no audio even though signaling succeeds.
Why the other options are wrong
- B. Invoking an MTP simply anchors the media through a media resource and does not by itself stop RTP from reaching the far endpoint.
- C. Packets arriving outside the jitter buffer window cause poor audio quality, but the RTP traffic is still received.
- D. RTP runs over UDP, so a firewall blocking TCP ports 16384 to 32768 would not affect the media stream.
Question 6
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which debug must the Administrator turn on?
Show answer and explanation
Correct answer: B. debug H.225 asn1
The debug H.225 asn1 command enables debugging of H.225 messages, which include both call setup signaling and media negotiation details in H.323 calls. H.225 covers the registration, admission, and status (RAS) protocol as well as call signaling messages that establish media parameters.
Why the other options are wrong
- A. "debug H.323 messages" is not a valid debug command syntax on Cisco gateway platforms.
- C. H.246 does not exist in the H.323 protocol suite; valid protocols are H.225 and H.245.
- D. "debug H.225 media" is not the correct syntax; media signaling details are captured in H.225 asn1 debugging.
- E. While debug H.323 asn1 might appear similar, H.225 asn1 is the correct command for comprehensive H.323 signaling and media setup troubleshooting.
Question 7
What is first preference condition matched in a SIP-enabled incoming dial peer?
Show answer and explanation
Correct answer: A. incoming uri
The incoming URI is the first preference condition matched in a SIP-enabled incoming dial peer. The SIP request-URI is evaluated first when determining which dial peer should handle an incoming SIP call, making it the primary matching criterion.
Why the other options are wrong
- B. Target carrier-id is not a standard SIP dial peer matching criterion.
- C. Answer-address is not used as a primary matching condition in SIP incoming dial peer selection.
- D. Incoming called-number is used for matching but is evaluated after the incoming URI in the preference order.
Question 8
Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-way audio or no audio. You determine that there is a firewall between the floors, and the administrator reports that it is allowing SIP signaling and UDP ports from 20000 to 22000 bidirectionally. What are two possible solutions? (Choose two.)
Show answer and explanation
Correct answer: C, D. Ask the firewall administrator to change the range of UDP ports to 16384-32767; Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
RTP media traffic uses UDP ports in the range 16384-32767 by default in Cisco systems. The firewall is currently only allowing UDP ports 20000-22000, which is insufficient. The two solutions are to ask the firewall administrator to expand the UDP port range to the standard 16384-32767, or to configure the SIP profile in Cisco Unified CM to use the restricted port range of 20000-22000 that the firewall already allows.
Why the other options are wrong
- A. Changing the SIP profile media ports to 16384-32767 would not help if the firewall is still blocking those ports; the firewall must first be updated.
- B. RTP uses UDP, not TCP; changing the firewall to TCP would have no effect on UDP-based RTP traffic.
- E. System Parameters in Cisco Unified Communications Manager does not control media port ranges; the SIP profile is the correct location for this configuration.
Question 9
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?
Show answer and explanation
Correct answer: C. Voice/Video > Session Trace Log View > Real Time Data
The Real-Time Monitoring Tool in Cisco Unified Communications Manager provides call flow and signaling details through the Voice/Video > Session Trace Log View > Real Time Data section. This interface displays live SIP call information including call setup, signaling messages, and media details as calls are being processed through the system.
Why the other options are wrong
- A. Analysis Manager and Trace File Repositories are used for managing archived trace files, not real-time call flow monitoring.
- B. Trace and Log Central is a system-level tool for accessing historical logs and traces, not real-time SIP call analysis.
- D. Opening from Local Disk allows reviewing previously saved trace files, which is post-call analysis rather than real-time monitoring.
Question 10
Which description of RTP timestamps or sequence numbers is true?
Show answer and explanation
Correct answer: D. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
RTP timestamps are used to place incoming audio and video packets in the correct timing order and to compensate for playout delay. The timestamp field indicates the sampling instant of the first octet in the RTP data packet, allowing the receiver to reconstruct the original timing and determine appropriate buffering for playback.
Why the other options are wrong
- A. The sequence number detects packet loss, but the question asks which description is true, and the timestamp function is the more complete answer for this context.
- B. Timestamps increase by the sampling rate (not by time carrying), and the increment depends on the media type and sample rate.
- C. Sequence numbers increment by 1 for each RTP packet transmitted, not by four; the increment of four is incorrect.
Question 11
Where on Cisco Unified Communications Manager do you configure the standard local route group for a group of devices?
Show answer and explanation
Correct answer: C. System > Device Pool
The standard local route group for a group of devices is configured in the Device Pool settings within System. The Device Pool associates multiple devices with common routing, location, and call processing parameters, including the local route group used for call routing.
Why the other options are wrong
- A. Location Info is used for location-based settings and bandwidth management, not for configuring route groups.
- B. Local Route Group Names is where you define the route group itself, but device association with the route group happens at the Device Pool level.
- D. Emergency Location Groups are used specifically for emergency routing (ELIN), not for standard local route group configuration.
Question 12
When the services key is pressed Cisco Extension Mobility does not show up. What is the cause of the issue?
Show answer and explanation
Correct answer: B. Cisco Extension Mobility Service is not running.
When Cisco Extension Mobility does not appear when the services key is pressed, the primary cause is that the Cisco Extension Mobility Service is not running on the server. The service must be actively running to respond to service requests from the phone. If the service is stopped or not started, the phone cannot access it regardless of configuration settings.
Why the other options are wrong
- A. An incorrect URL would cause connection errors, but the service would still appear in the list if configured.
- C. If the phone were not subscribed, subscription errors might occur, but this typically affects functionality after selection, not visibility.
- D. If Extension Mobility were not enabled in the phone configuration, it would not appear, but this is a configuration issue, not the primary troubleshooting step.
Question 13
Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?
Show answer and explanation
Correct answer: B. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.
To display Translation Pattern operations in SDL traces, the administrator must set the Digit Analysis Complexity parameter in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis. This configuration enables the detailed digit analysis tracing that includes translation pattern operations.
Why the other options are wrong
- A. Detailed Call Analysis under Enterprise Parameters is a separate feature that does not specifically control translation pattern visibility in SDL traces.
- C. Micro Traces on the Serviceability page is not the correct configuration location for controlling translation pattern output in SDL traces.
- D. Translation pattern operations are not displayed by default in SDL traces; explicit configuration of digit analysis complexity is required.
Question 14
Which two types of distribution algorithm are within a line group? (Choose two.)
Show answer and explanation
Correct answer: B, D. circular; top down
Line groups in Cisco Unified Communications Manager use distribution algorithms to determine how calls are routed among members. Circular (round-robin) and Top Down are the two standard distribution algorithms supported for line groups. Circular distributes calls sequentially through the list, while Top Down always starts from the first member in the ordered list.
Why the other options are wrong
- A. Random distribution is not a supported algorithm type for line groups in CUCM.
- C. Highest preference is not a distribution algorithm used within line groups.
- E. Bottom up is not a supported distribution algorithm for line group configuration.
Question 15
When locations-based Call Admission Control denies the call, which two masks can AAR apply when routing the call through the PSTN? (Choose two.)
Show answer and explanation
Correct answer: A, C. AAR destination mask; external phone number mask
When locations-based Call Admission Control denies a call due to bandwidth limitations, Automatic Alternate Routing (AAR) can apply two types of masks to route the call through the PSTN: the AAR destination mask and the external phone number mask. These masks transform the called number into a format suitable for PSTN routing.
Why the other options are wrong
- B. Called party transform mask is used for general call processing but is not specifically an AAR mask for PSTN routing.
- D. The +E.164 alternate number mask is not a standard AAR mask type in CUCM.
- E. Enterprise alternate number mask is not a recognized AAR mask designation in Cisco Unified Communications Manager.
Question 16
An engineer is troubleshooting Cisco Device Mobility and find that the phone has roamed to a building that is assigned to a different device pool but has not changed its device pool accordingly. What action resolves the issue?
Show answer and explanation
Correct answer: D. Set the correct subnet under Device Mobility info.
Device Mobility uses subnet information to determine the device's location and automatically change the device pool accordingly. When a phone roams to a building with a different device pool but does not change its device pool, the subnet-to-device pool mapping is incorrect. Setting the correct subnet under Device Mobility info ensures that UCM can properly identify the phone's location and apply the appropriate device pool configuration for that building.
Why the other options are wrong
- A. The Location setting under Current Device Mobility Settings identifies the current location but does not directly trigger device pool changes based on physical roaming.
- B. SRST (Survivable Remote Site Telephony) is for fallback behavior when the primary connection is lost, not for device pool changes during roaming.
- C. Device CSS affects call routing but does not control which device pool is assigned based on physical location changes.
Question 17
When configuring hunt groups, where do you add the individual directory numbers that will be part of the group?
Show answer and explanation
Correct answer: B. line group
Individual directory numbers that form the membership of a hunt group are configured in the line group. The line group is where you specify which phone extensions or DNs will be part of the hunt group and receive calls according to the hunting algorithm.
Why the other options are wrong
- A. Route groups are used for trunk selection and gateway routing, not for adding member directory numbers.
- C. Hunt lists define the sequence and distribution method but do not contain the actual DNs.
- D. The hunt pilot is the number dialed by callers and points to the hunt group configuration, but does not contain member DNs.
Question 18
When a third-party SIP Phone System is dialed inbound across a Cisco Unified Border Element, DTMF is failing. The third-party vendor accepts only out-of-band DTMF. Which configuration should be added to the outgoing dial peer to resolve this issue?
Show answer and explanation
Correct answer: D. dtmf-relay sip-kpml
When a third-party SIP phone system accepts only out-of-band DTMF, SIP-KPML (Keypad Markup Language) is the correct method to use. SIP-KPML is an out-of-band DTMF signaling mechanism that sends DTMF information within SIP messages rather than in the media stream, making it ideal for third-party systems that do not support in-band methods.
Why the other options are wrong
- A. RTP-NTE is an in-band DTMF relay method that encodes DTMF within RTP packets, not suitable for vendors requiring out-of-band signaling.
- B. Cisco-RTP is a Cisco-proprietary in-band DTMF method, incompatible with third-party vendors expecting out-of-band DTMF.
- C. H.245-signal is used for H.323 protocols, not relevant for SIP-based third-party phone systems.
Question 19
If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern "123"?
Show answer and explanation
Correct answer: B. 12X (urgent priority set)
When multiple patterns match a dialed number, Cisco Unified Communications Manager uses priority order to select the pattern. Pattern 12X with urgent priority set takes precedence over other matching patterns. The pattern 12X matches the dialed "123", and the urgent priority designation ensures it is selected over other equally-specific or less-specific matches like 1XX or 12[2-5].
Why the other options are wrong
- A. 12! does not match "123" because the exclamation mark requires one or more digits after "12", making it match longer strings only.
- C. 1XX matches but lacks the urgent priority, so it loses to pattern B which has urgent priority set.
- D. 12[2-5] matches "123" but has no priority advantage over pattern B, and B's urgent priority makes it the selected pattern.
Question 20
ABC company has decided to implement hunt groups to help distribute calls between members. In order to implement this, the administrator must configure hunt list, hunt groups, and line groups on Cisco UCM. Which distribution algorithms should the administrator implement?
Show answer and explanation
Correct answer: B. Top Down, Circular, Broadcast
Hunt groups in Cisco UCM support three distribution algorithms: Top Down (sequential order from the hunt list), Circular (round-robin with memory of last position), and Broadcast (simultaneous to all members). These are the standard algorithms available for distributing calls among hunt group members. Option A incorrectly includes Round Robin instead of Circular. Option C incorrectly includes Distribute. Option D incorrectly includes Sequential instead of Top Down.
Why the other options are wrong
- A. Round Robin is not a standard hunt group algorithm; Circular is the correct rotation method
- C. Distribute is not a valid hunt group algorithm; the third algorithm is Broadcast
- D. Sequential is not the correct term; the algorithm is called Top Down
Question 21
Which top-level IOS command is needed to begin the configuration of a Cisco Unified Communications Manager Express gateway to enable phones to be registered via SIP?
Show answer and explanation
Correct answer: B. voice service voip
The voice service voip command is the required top-level IOS command to begin SIP gateway configuration in Cisco Unified Communications Manager Express. This command enters the voice service configuration mode where SIP-specific parameters and settings are established.
Why the other options are wrong
- A. Allow-connections is a subordinate command used within voice service voip, not a top-level entry point.
- C. Voice register global is used after voice service voip is configured; it is not the initial top-level command.
- D. Voice register dn configures directory numbers and is entered after the voice service voip hierarchy is established.
Question 22
Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?
Show answer and explanation
Correct answer: B. Configure IP Address Trusted Authentication for Incoming VoIP Calls.
Configuring IP Address Trusted Authentication for incoming VoIP calls is the correct toll fraud prevention measure in Cisco Unified Communications Manager Express. This approach verifies that calls originating from trusted IP addresses are legitimate before processing them, preventing unauthorized toll usage.
Why the other options are wrong
- A. Direct Inward Dial configuration with overlap dialing is unrelated to toll fraud prevention mechanisms.
- C. The command structure described is incorrect; the proper configuration involves enabling trusted authentication, not disabling it with "no ip address trusted authenticate".
- D. Enabling secondary dial tone on FXO ports is a call handling feature, not a toll fraud prevention control.
Question 23
Configure Call Queuing in Cisco Unified Communications Manager. Where do you set the maximum number of callers in the queue?
Show answer and explanation
Correct answer: B. in the queuing configuration
In Cisco Unified Communications Manager, the maximum number of callers allowed in the queue is configured within the queuing configuration settings. This is where queue-specific parameters are defined, including queue depth, caller limits, and other queue management options. This dedicated configuration area allows administrators to set queue-specific policies separate from general telephony or enterprise-wide parameters.
Why the other options are wrong
- A. Telephony service configuration handles general telephony settings but not queue-specific parameters.
- C. Enterprise Parameters contain system-wide settings but not individual queue depth configurations.
- D. Service Parameters are for Cisco Unified CM services but not for queue management specifics.
Question 24
A company has users that are logged in to hunt groups. However, there is a requirement for hunt group configurations to provide an option to turn on audible ringtones when calls to a line group arrive at a phone that is logged out and on a break. This ringtone alerts a logged-out user that there is an incoming call to a hunt list to which the line is a member, but the call does not ring at the phone of that line group member because of the logged-out status. Which action meets this requirement?
Show answer and explanation
Correct answer: D. Configure the service parameter hunt group logoff notification and specify the name of the ringtone file.
The hunt group logoff notification service parameter enables an audible ringtone to alert logged-out users when calls arrive to their hunt group line. This notification allows users to hear that calls are coming to the line group even though they are logged out and the calls do not actually ring at their phone. The administrator specifies which ringtone file to use with this parameter.
Why the other options are wrong
- A. The HLog softkey controls login/logout status but does not provide audible tone notifications for missed calls.
- B. The Enterprise Feature Access number parameter is for accessing hunt group login/logout features, not for configuring notification tones.
- C. Party Entrance Tone is used to signal when parties join a conference call, not for hunt group logoff notifications.
Question 25
How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?
Show answer and explanation
Correct answer: C. At the PSTN gateway, put the calling number in E.164 format and the called number in localized (DN) format.
To globalize routing for ingress calls from the PSTN to internal directory numbers, the PSTN gateway must convert the calling number to E.164 format (international standard) and the called number to localized DN format (internal format understood by Unified CM). This ensures proper routing and digit normalization across the system boundaries.
Why the other options are wrong
- A. PSTN format for the calling number is not standardized; E.164 is the required global format for this purpose.
- B. Having Unified CM perform the conversion places the burden on the wrong system; the gateway is responsible for initial normalization.
- D. Converting the called number to E.164 format is incorrect; internal DNs must remain in localized format for Unified CM to route calls to the correct devices.
Question 26
What is a component of Cisco Unified Mobility?
Show answer and explanation
Correct answer: B. Mobile Connect
Mobile Connect is a component of Cisco Unified Mobility that enables users to receive and manage calls on their mobile devices as part of their unified communications experience. It integrates mobile devices into the overall Unified Communications system, allowing seamless call handling across multiple devices.
Why the other options are wrong
- A. Unified IVR is a separate Cisco product for interactive voice response systems, not a Unified Mobility component.
- C. Smart Client Support refers to desktop client functionality, not a specific Unified Mobility component.
- D. Single Number Connect is not a standard Cisco Unified Mobility component or feature.
Question 27
Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)
Show answer and explanation
Correct answer: B, C. can be associated only under the route list; chooses the route group that is configured under the device pool of the calling-party device
The Standard Local Route Group can only be associated under the route list configuration, not independently under route groups. It chooses the route group defined in the device pool of the calling-party device, ensuring calls are routed based on the originating device's configuration rather than the destination device.
Why the other options are wrong
- A. Standard Local Route Groups are not directly associated under route groups.
- D. The route group selection is based on the calling-party device pool, not the called-party device pool.
- E. Standard Local Route Groups cannot be assigned directly to route patterns; they must be configured within route lists and device pools.
Question 28
A customer has multisite deployments with a globalized dial plan. The customer wants to route PSTN calls via the gateway assigned to each site. Which two actions will fulfill the requirement? (Choose two.)
Show answer and explanation
Correct answer: C, D. Assign one route group as a local route group in the device pool of the corresponding site; Create one route group for each site and one global route list for PSTN calls that point to the local route group.
To route PSTN calls via gateways assigned to each site in a globalized dial plan, create one route group for each site containing that site's local gateway, then create one global route list for PSTN calls that points to the local route groups. Additionally, assign each local route group in the device pool of the corresponding site so devices at that site use their local gateway.
Why the other options are wrong
- A. Creating only one global route list pointing to one gateway does not enable site-specific routing.
- B. A single route group containing all gateways without site-specific assignment does not ensure local gateway usage.
- E. Hunt groups are used for line group distribution algorithms, not for PSTN gateway routing by site.
Question 29
A new deployment is using MVA for a specific user on the sales team, but the user is having issues when dialing DTMF. Which DTMF method must be configured in resolve the issue?
Show answer and explanation
Correct answer: B. out-of-band
Mobile Voice Access (MVA) requires out-of-band DTMF configuration to properly handle DTMF tones when users dial from mobile devices. Out-of-band DTMF uses signaling channels separate from the media stream, ensuring reliable DTMF transmission over potentially unstable mobile connections.
Why the other options are wrong
- A. In-band DTMF is unreliable over mobile networks and is not the recommended method for MVA deployments.
- C. Gateway is not a DTMF method but rather refers to network elements.
- D. Channel is not a recognized DTMF method classification.
Question 30
Which IOS command creates a SIP-enabled dial peer?
Show answer and explanation
Correct answer: B. dial-peer voice 20 voip
The correct IOS syntax for creating a SIP-enabled dial peer is 'dial-peer voice [number] voip', where 'voip' indicates a Voice over IP peer capable of supporting SIP protocol. The word order and keyword sequence are essential to proper IOS command syntax.
Why the other options are wrong
- A. The syntax 'voice dial-peer' reverses the correct command order and uses 'sip' instead of 'voip'.
- C. The 'pots' keyword creates a Plain Old Telephone Service dial peer, not a SIP-enabled VoIP peer.
- D. While resembling the correct syntax, it uses 'dial peer' instead of 'dial-peer' and 'sip' instead of 'voip'.
Question 31
A user’s phone is already configured for Single Number Reach, and the user wants a feature to move an active call from a mobile phone to a desk phone and vice-versa. As an administrator, which additional configuration should be made to fulfill the user’s request?
Show answer and explanation
Correct answer: B. Add the mobility key to the softkey template that the desk phone is using.
To enable users to move an active call between a mobile phone and desk phone when Single Number Reach is already configured, the mobility softkey must be added to the softkey template used by the desk phone. This softkey provides the interface for users to transfer or resume calls between their various connected devices, fulfilling the requirement to move calls between mobile and desk phones.
Why the other options are wrong
- A. Dialed Number Analyzer determines digit patterns and routing rules but does not control call movement between devices.
- C. The Resume softkey is for resuming held calls within the same device, not for moving calls between different devices.
- D. Extension Mobility allows users to log into different phones, but does not provide the active call transfer capability between simultaneous devices.
Question 32
A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6 addressing companywide. To enable the IPv6 in a unified SRST gateway to support SIP phones, what are two supported supplementary features for an IPv6 fallback scenario? (Choose two.)
Show answer and explanation
Correct answer: A, C. three-way conference; T.38 fax relay
Three-way conference and secure SIP lines are the two supported supplementary features for IPv6 fallback scenarios in a unified SRST gateway running IOS XE. These features maintain functionality when the system operates in IPv6 mode with fallback capabilities for SIP phones.
Why the other options are wrong
- B. T.38 fax relay is not listed as a supported feature in IPv6 fallback scenarios for SRST gateways.
- D. Transcoding resources are not identified as a supported supplementary feature for IPv6 SRST fallback operation.
- E. SIP trunk configuration is not part of the supported supplementary features specifically for IPv6 fallback scenarios.
Question 33
An engineer has temporarily disabled toll fraud prevention for SIP line calls on a Cisco CME12.6x and must enforce security and toll fraud prevention for the SIP line side on Cisco Unified CME. Which configuration must be used to start this process?
Show answer and explanation
Correct answer: C. voice service voip ip address trusted authenticate
On Cisco Unified CME 8.1 and later, SIP toll fraud prevention is controlled by the ip address trusted authenticate command under voice service voip. Because the engineer had disabled it with the no form, re-entering ip address trusted authenticate is the first step that turns IP address trusted authentication back on. The trusted list of allowed peers is then built afterwards under ip address trusted list.
Why the other options are wrong
- A. There is no enable ip address trust list command under voice service voip in Cisco IOS.
- B. The ip address trusted list command only defines trusted peers and does not re-enable the authentication that was disabled.
- D. There is no enable ip address trust authentication command; the correct syntax is ip address trusted authenticate.
Question 34
An engineer must configure a Cisco UCM hunt list so that calls to users in a line group are routed to the first idle user and then the next. Which distribution algorithm must be configured to accomplish this task?
Show answer and explanation
Correct answer: B. top down
The 'top down' distribution algorithm routes calls to users in a line group sequentially from the first idle user in the list to the next, ensuring calls are distributed in priority order from top to bottom of the hunt list configuration.
Why the other options are wrong
- A. Broadcast distribution sends calls to all members simultaneously rather than to the first idle user.
- C. Longest idle time distributes calls to the user who has been idle the longest, not in sequential order.
- D. Circular distribution rotates through members in a round-robin fashion, not always starting with the first idle user.
Question 35
In Cisco Unified Communications Manager globalized call routing is implemented and must confirm that it is correctly implemented without making a call. Which tool do you use for verification?
Show answer and explanation
Correct answer: A. Dialed Number Analyzer
The Dialed Number Analyzer is the tool used to verify globalized call routing configuration in Cisco Unified Communications Manager without actually making a call. It simulates the dial plan and routing logic to confirm that calls will be routed correctly based on the current configuration.
Why the other options are wrong
- B. Real-Time Monitoring Tool is used for live call monitoring and system status, not for testing dial plan routing.
- C. SDI traces capture low-level system diagnostics but are not the proper tool for analyzing call routing logic.
- D. SDL traces provide signaling details but require actual call activity; they are not used for non-call verification of routing.
Question 36
CollabCorp is a global company with two clusters, emea.collab.corp and apac.collab.corp. URI dialing is implemented and working in each cluster. The company configured routing between clusters to make intercluster calls via URI, but this is not working as expected. Which two configuration elements should be checked to resolve this issue? (Choose two.)
Show answer and explanation
Correct answer: C, E. SIP route pattern; SIP trunk
To resolve URI dialing issues between clusters, both the SIP route pattern and the SIP trunk must be verified. The SIP route pattern directs calls to the correct cluster based on URI format, while the SIP trunk establishes the actual connection between clusters. Both elements are essential for intercluster URI call routing to function properly.
Why the other options are wrong
- A. Intercluster trunk is a legacy call routing method for non-URI deployments and is not used with URI dialing implementations.
- B. Directory URI partition controls which users can be dialed via URI but does not affect intercluster routing logic.
- D. Calling search space and partition control intracluster access and are not directly related to intercluster URI routing configuration.
Question 37
A user reports that when they attempt to log out from the Cisco Extension Mobility service by pressing the Services button, they cannot log out. What is the most likely cause of this issue?
Show answer and explanation
Correct answer: D. The user device profile is not subscribed to the Cisco Extension Mobility service.
When a user cannot log out from Cisco Extension Mobility by pressing the Services button, the most likely cause is that the user device profile is not subscribed to the Cisco Extension Mobility service. Without this subscription, the phone cannot communicate with the Extension Mobility service to process logout requests. The device profile must explicitly subscribe to the service for the feature to be available.
Why the other options are wrong
- A. If the service were not configured on the phone, users would not be able to log in at all, not just log out.
- B. Network latency might cause delays but would not prevent logout functionality entirely; the request would eventually process.
- C. If the device profile were not assigned, users could not perform any Extension Mobility functions, but this typically fails during login.
Question 38
In Cisco Unified Communications Manager, which tool do you use to check SIP traces?
Show answer and explanation
Correct answer: C. RTMT
The Real-Time Monitoring Tool (RTMT) is the proper utility in Cisco Unified Communications Manager for viewing and analyzing SIP traces. It provides access to detailed call signaling information needed for troubleshooting SIP-related issues.
Why the other options are wrong
- A. MTP (Media Termination Point) is a resource for media processing, not a trace analysis tool.
- B. CCSIP is a SIP protocol component, not a tool for viewing traces.
- D. OS Administration Page manages system-level settings but is not used for SIP trace analysis.
Question 39
Calls to a particular extension are not routing to voicemail. The user reaches the voicemail system from the handset by pressing the Messages button. Which configuration parameter causes this problem?
Show answer and explanation
Correct answer: A. The voicemail pilot number for call forwarding is missing from the ephone-dn.
On Cisco UCME, the voicemail pilot number must be configured in the ephone-dn (extension phone directory number) as a call forwarding destination. When this pilot number is missing from the ephone-dn configuration, calls cannot automatically route to voicemail. Users can still manually reach voicemail by pressing the Messages button, which uses a separate mechanism, but automatic call forwarding to voicemail will not function.
Why the other options are wrong
- B. While the telephony service may reference voicemail, the actual call forwarding routing is configured at the ephone-dn level.
- C. On UCME systems, Cisco Unity Express configuration is separate from the phone configuration that controls call forwarding routing.
- D. The voicemail pilot number for call forwarding is configured on the ephone-dn, not directly on the ephone device itself.
Question 40
An administrator is configuring a cluster for ILS and wants to limit the amount of entities that Cisco Unified Communications Manager can write to the database for data that is learned through ILS. Which service parameter is used to adjust this limit?
Show answer and explanation
Correct answer: A. ILS Max Number of Learned Objects in Database
The service parameter ILS Max Number of Learned Objects in Database is used to restrict the quantity of entities that Cisco Unified Communications Manager writes to the database for data learned through Intercluster Lookup Service. This parameter prevents unlimited database growth from ILS learning activities.
Why the other options are wrong
- B. ILS Active Learned Object Upper Limit is not the correct service parameter name for this function.
- C. Global Data Service Parameter Limit is not specific to ILS learned objects.
- D. Imported Dial Plan Replication Database Object Lower Limit relates to dial plan replication, not ILS learning.
Question 41
An engineer must route all SIP calls in the form of <user>@example.com to the SIP trunk gateway corporate local. Which two SIP route patterns can be used to accomplish this task? (Choose two.)
Show answer and explanation
Correct answer: B, E. *@example.com; *.*
SIP route patterns use wildcard notation to match incoming call addresses. The pattern '*@example.com' matches any user at the example.com domain and routes to the gateway. The pattern '*.*' is a catch-all pattern that matches any SIP URI with the format 'anything@anything', allowing broad routing flexibility for various domain destinations.
Why the other options are wrong
- A. This pattern is malformed and mixes gateway and domain syntax incorrectly.
- C. A bare domain name without wildcard or user portion is not a valid SIP route pattern syntax.
- D. A single domain name without user matching syntax cannot route SIP calls in the format user@example.com.
Question 42
Due to a shortage of physical interfaces on a device, the administrator requires that a loopback for RTP is used. Which command is required when using a loopback interface for RTP?
Show answer and explanation
Correct answer: C. voice-class sip bind media source-interface Loopback0
The 'voice-class sip bind media source-interface Loopback0' command binds RTP media traffic to a loopback interface, allowing the device to use the loopback address for RTP streams when physical interfaces are limited.
Why the other options are wrong
- A. The early-offer forced command is for SDP offer/answer handling, not for RTP source interface binding.
- B. The 'bind control source-interface' command binds SIP control signaling, not RTP media traffic.
- D. Resource priority mode passthrough is for handling priority signaling, not for configuring RTP source interfaces.
Question 43
A user reports when they press the services key they do not receive a user ID and password prompt to assign the phone extension. Which action resolves the issue?
Show answer and explanation
Correct answer: B. Subscribe the phone to the Cisco Extension Mobility service.
When a user does not receive a user ID and password prompt after pressing the services key, the issue is that the phone is not subscribed to the Cisco Extension Mobility service. Subscription is the mechanism that enables the service on the device and allows it to request authentication credentials. Without subscription, the phone cannot initiate the login process even if the service is running.
Why the other options are wrong
- A. Default device profiles are used after successful login, not for initiating the login process itself.
- C. Creating end users and associating them with device profiles happens after the initial login, not before.
- D. Assigning extensions as mobile extensions is a user configuration but does not enable the initial login prompt.
Question 44
You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?
Show answer and explanation
Correct answer: A. configuration for a single SIP phone
The voice register pool 1 command in Cisco Unified Communications Manager Express initiates configuration for a single SIP phone. Each voice register pool entry defines parameters specific to an individual phone device, such as codec preferences, dial plan rules, and registration settings.
Why the other options are wrong
- B. Global configuration items applicable to all SIP phones are configured under the voice register global command, not voice register pool.
- C. While device pools exist in Unified Communications Manager, the voice register pool command in CME configures individual phones, not pools of phones.
- D. SIP registrar service configuration uses different commands within the voice service voip section, not voice register pool.
Question 45
The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)
Show answer and explanation
Correct answer: C, E. CTI ports; IP phones
The Dialed Number Analyzer tool in Cisco Unified Communications Manager allows analysis of calls originating from CTI route points and IP phones. These are the two device types from which call analysis can be performed to verify dial plan behavior and routing decisions.
Why the other options are wrong
- A. Translation patterns are routing constructs, not devices that originate calls for analysis.
- B. Device pools are configuration containers, not actual devices that make calls.
- D. While CTI route points are correct, the question asks for exactly two items, and CTI route points alone do not satisfy the complete answer.
Question 46
An engineer must configure a secure SIP trunk with a remote provider, with a specific requirement to use port 5065 for inbound and otubound traffic. Which two items must be configured to complete this configuration? (Choose two.)
Show answer and explanation
Correct answer: C, D. Destination Port in SIP Information section of the SIP Trunk configuration; Incoming Port in SIP Trunk Security Profile configuration
Configuring a secure SIP trunk with a specific port requires two settings: the Destination Port in the SIP Information section of the SIP Trunk configuration specifies which port the system sends traffic to (outbound), and the Incoming Port in the SIP Trunk Security Profile configuration specifies which port receives inbound traffic. Together, these allow bidirectional communication on port 5065.
Why the other options are wrong
- A. Incoming Port in SIP Information section is not the correct location; port configuration for inbound traffic belongs in the Security Profile.
- B. Security Information in the SIP Profile is not the correct scope; SIP Trunk-specific security settings are configured in the SIP Trunk Security Profile.
- E. Destination Port in the Security Profile is redundant; outbound destination port is properly configured in the SIP Trunk's SIP Information section.
Question 47
An engineer must implement call restriction to toll-free numbers using a class of restriction in a branch Cisco UCME. In which two places is the corlist incoming or cor incoming command configured? (Choose two.)
Show answer and explanation
Correct answer: A, B. “voice register pool ” configuration mode; “ephone-dn ” configuration mode
In Cisco UCME, the corlist incoming or cor incoming command is configured in two locations: voice register pool mode (for SIP endpoints) and ephone-dn mode (for SCCP extension numbers). These are the two primary places where incoming call restrictions are applied to control which calls can be received.
Why the other options are wrong
- C. Dial-peer COR custom is used for outgoing call restrictions and routing, not for incoming call restrictions.
- D. Voice register global configuration applies global settings but not the per-device or per-extension incoming restrictions.
- E. Telephony-service mode is used for system-wide settings, not for specific incoming call restriction lists.
Question 48
Users are reporting that several inter-site calls are failing, and the message "not enough bandwidth" is showing on the display. Voice traffic between locations goes through corporate WAN, and Call Admission Control is enabled to limit the number of calls between sites. How is the issue solved without increasing bandwidth utilization on the WAN links?
Show answer and explanation
Correct answer: B. Configure AAR to reroute calls that are denied by Call Admission Control through the PSTN.
Automatic Route Alternate (AAR) solves bandwidth exhaustion by rerouting calls denied by CAC to an alternate path, such as the PSTN. This allows calls to complete without increasing WAN bandwidth utilization. Option A disables CAC entirely, defeating bandwidth protection. Option C reroutes all calls through PSTN, unnecessarily wasting WAN capacity. Option D queues calls indefinitely without solving the bandwidth limitation.
Why the other options are wrong
- A. Disabling CAC removes bandwidth protection and allows uncontrolled WAN usage
- C. Rerouting all calls through PSTN wastes the available WAN capacity instead of managing it
- D. Call queuing doesn't create bandwidth; callers would wait indefinitely without a solution
Question 49
An administrator discovers that employees are making unauthorized long-distance and international calls from logged-off Extension Mobility phones when the authorized users are away from their desks. Which two configurations should the administrator configure in the Cisco UCM to avoid this issue? (Choose two.)
Show answer and explanation
Correct answer: B, E. Remove the long-distance & international pattern’s partitions from the calling search space of the physical phone’s directory number; Remove long-distance & international pattern’s partitions from the calling search space of the device phone.
To prevent unauthorized calls from logged-off Extension Mobility phones, administrators must remove long-distance and international pattern partitions from the calling search space of the physical phone's directory number (option B), ensuring the physical phone itself cannot dial restricted numbers. Additionally, removing these partitions from the device phone's calling search space (option E) prevents the logged-off state from having access to restricted dialing. The device phone CSS applies when no user is logged in, protecting against unauthorized calls during logged-out periods.
Why the other options are wrong
- A. Adding restricted partitions to the physical phone's CSS would enable unauthorized calls, worsening the problem.
- C. Adding restricted partitions to the device phone's CSS would grant access to those numbers when users are logged off.
- D. Adding restricted partitions to the directory number's CSS enables unauthorized calls rather than preventing them.
Question 50
An administrator is working on an issue between the customer's Cisco Unified Border Element and the service provider. The provider only wants to see midcall signaling from the Cisco Unified Border Element for fax calls. Which command must be configured on Cisco Unified Border Element?
Show answer and explanation
Correct answer: C. midcall-signaling passthru media-change
The command 'midcall-signaling passthru media-change' must be configured on the Cisco Unified Border Element to allow the service provider to see midcall signaling specifically for fax calls. The media-change parameter ensures that signaling is passed through when media codec changes occur during fax transmission.
Why the other options are wrong
- A. midcall-signaling passthru alone allows all midcall signaling, not limited to fax calls as required.
- B. no update-callerid is unrelated to midcall signaling or fax-specific requirements.
- D. midcall-signaling preserve-codec preserves codec information but does not provide media-change handling for fax scenarios.
Question 51
A user requests a feature to send an active call to the mobile phone number on the physical phone. As an administrator, what should be configured in the Cisco UCM to accomplish this?
Show answer and explanation
Correct answer: B. A Remote Destination Profile having the same extension as the physical phone’s Directory Number, add the mobile phone number to the RDP as a Remote Destination. Add the softkey “Mobility” to the physical phone’s softkey template.
To send an active call to a mobile phone, the administrator creates a Remote Destination Profile with the same extension as the physical phone's Directory Number, adds the mobile phone number as a Remote Destination, and adds the Mobility softkey to the physical phone's template. The Mobility softkey allows users to seamlessly transfer active calls to remote destinations. Option A uses the Join softkey, which is for merging calls, not mobility transfers. Options C and D incorrectly tie the RDP to the mobile phone's number instead of the physical phone's extension.
Why the other options are wrong
- A. Join softkey merges calls together, not transfers them to mobile destinations
- C. RDP extension must match the physical phone's Directory Number, not the mobile phone number
- D. Combines two errors: wrong RDP extension base and wrong softkey selection
Question 52
A customer routes PSTN calls to ITSP through a SIP trunk on Cisco UCM that forwards and receives calls to and from ITSP. ITSP is set to send an E.164 number when the customer’s extension is four digits. Which action should be taken to route the incoming calls to four-digit extensions?
Show answer and explanation
Correct answer: C. Set the Significant Digits to 4 on the SIP trunk.
Setting the Significant Digits to 4 on the SIP trunk configures the system to use only the last four digits of incoming E.164 numbers for call routing to four-digit extensions. This strips the leading digits of the E.164 number and maps the remaining four digits to the internal extension.
Why the other options are wrong
- A. Voice translation profiles are more complex than necessary; Significant Digits setting is the direct solution for this scenario.
- B. Setting Significant Digits to 8 would match eight-digit patterns, not the four-digit extensions required.
- D. Voice translation rules require configuration on a border element and are unnecessary when the SIP trunk Significant Digits setting can solve this directly.
Question 53
Single Number Reach calls to a cell phone that not answered are leaving voicemails on the cell phone rather than the corporate mailbox. Which two options will resolve this issue? (Choose two.)
Show answer and explanation
Correct answer: D, E. Decrease the T301 timer; Decrease the Answer Too Late timer.
When Single Number Reach calls to a cell phone are not answered, calls should route to the corporate voicemail rather than the cell phone's voicemail. Decreasing the T301 timer (option D) reduces the time the system waits for the cell phone to answer before routing elsewhere, and decreasing the Answer Too Late timer (option E) controls how long the system waits before considering the call unanswered. Together, these timers ensure calls route to corporate voicemail more quickly instead of reaching the cell phone's voicemail system.
Why the other options are wrong
- A. Enable Extend and Connect is for conference call features, not for controlling voicemail routing behavior.
- B. Unified Mobility features provide general mobility capabilities but do not specifically control voicemail destination routing timing.
- C. The T302 timer controls different signaling aspects and does not affect voicemail routing decisions for unanswered calls.
Question 54
An administrator is asked to configure egress call routing by applying globalization and localization on Cisco UCM. How should this be accomplished?
Show answer and explanation
Correct answer: B. Globalize the calling and called numbers to E.164 format and localize the called number in the gateway.
Egress call routing follows the pattern: first globalize calling and called numbers to E.164 format (a standard international format) within the UCM, then localize the called number at the gateway for the destination network. This ensures proper number formatting for routing through the PSTN. Option A reverses the localization and globalization sequence. Options C and D use PSTN format instead of E.164, which is incorrect for the globalization step.
Why the other options are wrong
- A. Reverses the correct order by localizing first instead of globalizing first
- C. Uses PSTN format instead of E.164 for globalization, and globalizes in the gateway instead of UCM
- D. Uses PSTN format instead of E.164 and only localizes the calling number instead of the called number
Question 55
After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?
Show answer and explanation
Correct answer: D. Router(config)# voice service voip Router(conf-voi-serv)#no supplementary-service sip moved-temporarily
The issue occurs because the SIP protocol is sending a 'moved-temporarily' response (302 redirect) when the call cannot reach an available agent, causing the call to fail. Disabling the supplementary-service sip moved-temporarily command prevents this redirect response and allows the call to properly route to voicemail instead of generating a busy signal.
Why the other options are wrong
- A. H.323 to H.323 connections are unrelated to SIP trunk voicemail routing issues.
- B. Voice Activity Detection (VAD) affects audio processing efficiency but does not impact voicemail routing or delivery.
- C. Allow-connections voicemail mod is not a valid configuration command in this context.
Question 56
Which configuration element of a hunt group allows for changing Calling Party Transformations settings?
Show answer and explanation
Correct answer: B. hunt pilot
The hunt pilot configuration element allows administrators to modify Calling Party Transformation settings for a hunt group. The hunt pilot is where call delivery parameters and transformations for incoming calls to the hunt group are defined.
Why the other options are wrong
- A. Line group contains the member DNs but does not configure calling party transformations.
- C. Route group is used for trunk routing and does not handle hunt group transformations.
- D. Hunt list defines the order of pilot points but not the calling party transformation settings.
Question 57
Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)
Show answer and explanation
Correct answer: A, B. The Client Matter Code feature does not support overlap sending because the Cisco Unified CM cannot determine when to prompt the user for the code; If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.
The Client Matter Code feature cannot support overlap sending because Cisco Unified Communications Manager cannot determine the appropriate time to prompt the user for the code during digit-by-digit transmission. When Allow Overlap Sending is enabled, the Require Client Matter Code checkbox becomes disabled to prevent conflicts between these two features.
Why the other options are wrong
- C. These checkboxes are mutually exclusive; enabling overlap sending disables the Client Matter Code requirement.
- D. The system cannot determine prompting timing with overlap sending, so this feature does not support overlap sending.
- E. Client Matter Code does not have authorization level options like Forced Authorization Code does.
Question 58
An engineer is configuring Cisco UCM to forward parked calls back to the user who parked the call if it is not retrieved after a specified time interval. Which action must be taken to accomplish this task?
Show answer and explanation
Correct answer: D. Configure service parameters.
Service parameters in Cisco UCM control system-wide call handling behaviors, including parked call recall timers and callbacks. This is where the administrator configures the timeout interval and callback destination for unretreived parked calls. Option A configures softkey availability, not call recall logic. Option B manages phone registration and connectivity. Option C manages user permissions and feature access levels.
Why the other options are wrong
- A. Enterprise softkeys only control which softkey features are available on phones, not call recall behavior
- B. Device pools manage device registration, locale, and region settings, not call parking logic
- C. Class of control restricts feature access based on user roles, not call recall configuration
Question 59
A single site reports that when they dial select numbers, the call connects, but they do not get audio. The administrator finds that the calls are not routing out of the normal gateway but out of another site's gateway due to a TEHO configuration. What is the next step to diagnose and solve the issue?
Show answer and explanation
Correct answer: B. Verify that IP routing is correct between the gateway and the IP phone.
The call connects, which proves the route pattern and the remote gateway dial peer matched correctly. The missing audio means the RTP stream cannot flow between the phone at this site and the other site's gateway. The next step is to verify IP routing, and any firewalls or ACLs, between that gateway and the IP phones.
Why the other options are wrong
- A. The calling party transformation mask changes caller ID presentation and has no effect on whether audio flows.
- C. The call already connects through the gateway, so its dial peer destination pattern is clearly matching.
- D. A route pattern blocking the number would stop the call from connecting at all, rather than connecting without audio.
Question 60
A user in location X dials an extension at location Y. The call travels through a QoS-enabled WAN network, but the user experiences choppy or clipped audio. What is the cause of this issue?
Show answer and explanation
Correct answer: A. missing Call Admission Control
Call Admission Control (CAC) is required in QoS-enabled WAN networks to prevent bandwidth oversubscription and ensure voice quality. Without CAC, too many simultaneous calls can consume excessive bandwidth, causing audio degradation, choppy or clipped audio, and poor call quality even when QoS policies are configured.
Why the other options are wrong
- B. Codec mismatch typically results in call failure or no audio, not choppy audio with a functioning call.
- C. Ptime mismatch affects packetization timing but does not typically cause choppy audio in a properly configured VoIP network.
- D. Phone class of service affects call routing and priority but is not the direct cause of audio quality issues in this scenario.
Question 61
An engineer set up and successfully tested a TEHO solution on the Cisco UCM. PSTN calls are routed correctly using the IP WAN as close to the final PSTN destination as possible. However, suddenly, calls start using the backup local gateway instead. What is causing the issue?
Show answer and explanation
Correct answer: C. WAN connectivity
When a previously working TEHO solution suddenly routes calls through the backup local gateway instead of the primary WAN gateway, WAN connectivity issues are the cause. TEHO relies on WAN availability to route calls optimally; if WAN connectivity degrades or fails, the system automatically falls back to the local backup gateway.
Why the other options are wrong
- A. Route pattern configuration does not change dynamically; if it worked before, it remains the same.
- B. LAN connectivity would affect all calls equally, not cause a shift from WAN to local gateway routing.
- D. Route list and route group configuration would not cause a sudden shift unless actively modified by an administrator.
That was 61 of 245.
The full Cisco 300-815 CLACC pack has all 245 questions, each with the answer, the explanation and why the other options are wrong, plus a questions-only copy for timed runs. US$39, paid once, with free monthly updates and a pass-or-your-money-back guarantee.
Not ready yet? Get the first 113 free as one PDF by email. Get the Half Pack →