An administrator is configuring Meet-me conferencing in a Cisco UCM deployment and has created the Meet-me number and ensured that it is in a partition accessible by all devices. Which two additional steps must the administrator perform? (Choose two.)

A. Ensure that conferencing-initiating devices are using a media resource group list that contains at least one Cisco UCM conference bridge.

B. Disable Early Media on the SIP profile of all devices that will use Meet-me conferencing.

C. Enable Meet-me conferencing in enterprise parameters.

D. Ensure that all devices have G.729 enabled.

E. Update the softkey template on all phones to ensure that they contain the Meet-me softkey.

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.)

A. Check the Enable Extend and Connect checkbox.

B. Check the Enable Unified Mobility features checkbox.

C. Decrease the T302 timer.

D. Decrease the T301 timer.

E. Decrease the Answer Too Late timer.

Management wants to change the initial announcements for one of the existing call hunt groups. A new set of announcement audio file was provided. Which two configuration steps must the administrator take to accomplish this change? (Choose two.)

A. Identify the MOH audio source ID associated to one of the line group member’s “Network Hold MOH Audio Source”.

B. Identify the MOH audio source ID associated to “Network Hold Source & Announcements” under the Queuing section of the hunt pilot.

C. Identify the configured announcement names to change under the MOH audio source section, then upload the new files to the respective announcements under the Announcement section.

D. Identify the MOH audio source ID associated to one of the line group member’s “User Hold MOH Audio Source”.

E. Identify the configured announcement names to change under the Announcement section, and assign the uploaded files to the Queueing section of the hunt pilot.

Refer to the exhibit. Which change to the translation rule is needed to strip only the leading 9 from the digit string 9123548?

A. rule 1 /^9(d*)/ /l/

B. rule 1 /^9(.*)/ /l/

C. rule 1 /.*(3548$)/ /l/

D. rule 1 /^9123548/ /1/

Which IOS command creates a SIP-enabled dial peer?

A. voice dial-peer 20 sip

B. dial-peer voice 20 voip

C. dial-peer voice 20 pots

D. dial peer voice 20 sip

55697959.007 |12:20:50.913 |AppInfo |RouteListCdrc::createPartyTransformedCcSetupReqMsg - before DAapplyCdpnXform() preXformCdpn=11112222 preTag=SUBSCRIBER prePos=11112222 crCdpnMask=33334444 crPrefixDigit= crDDI=2
55697959.008 |12:20:50.913 |AppInfo |RouteListCdrc::createPartyTransformedCcSetupReqMsg - after DAapplyCdpnXform() xformCdpn=33334444 xformTag=SUBSCRIBER xformPos=11112222
55697959.009 |12:20:50.913 |AppInfo |RouteListCdrc::transformed cdpn (without unconsumpt digits) = 33334444, unconsumed digit=
Refer to the exhibit. Which INVITE is sent to 10.10.100.123 as a result of this log?

A. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:33334444@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “1000” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0 –

B. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:33334444@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “11112222” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0 –

C. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:11112222@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “1000” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0 –

D. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:11112222@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “11112222” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0

In Cisco Unified Communications Manager, which tool do you use to check SIP traces?

A. MTP

B. CCSIP

C. RTMT

D. OS Administration Page

What does VoIP trace in a Cisco Unified Border Element do?

A. It stores all call trace data in system log files for successful and failed calls.

B. It stores all call trace data in system file logs for failed calls.

C. It stores all call trace data in system memory for failed calls.

D. It stores all call trace data in system memory for successful and failed calls.

When locations-based Call Admission Control denies the call, which two masks can AAR apply when routing the call through the PSTN? (Choose two.)

A. AAR destination mask

B. called party transform mask

C. external phone number mask

D. +E.164 alternate number mask

E. enterrise alternate number mask

Refer to the exhibit. Users report that when they dial to Cisco Unity Connection from an external network, they cannot enter any digits. Assuming only in-band
DTMF is supported, what is a reason for this malfunction?

A. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.

B. There is SIP Delayed Offer. DTMF is supported only in Early Offer.

C. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.

D. No DTMF is negotiated.

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?

A. debug H.323 messages

B. debug H.225 asn1

C. debug H.246 asn 1

D. debug H.225 media

E. debug H.323 asn 1

An administrator is configuring Cisco UCM and the system to send *.webex.com traffic to a Cisco UCM Session Management Edition cluster. The administrator wants to limit which endpoints can reach *.webex.com. Which two items must the administrator configure for the SIP route pattern? (Choose two.)

A. calling party transformation

B. partition of the SIP route pattern

C. connected party transformation

D. called party URI transformation

E. destination SIP trunk of the SIP route pattern

Refer to the exhibit. How many maximum hops can an ILS updarte traverse?

A. 3

B. 6

C. 9

D. 12

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 mid-call signaling from the Cisco Unified Border Element for fax calls. Which command must be configured on Cisco Unified Border Element?

A. midcall-signaling passthru

B. no update-callerid

C. midcall-signaling passthru media-change

D. midcall-signaling preserve-codec

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?

A. ILS Max Number of Learned Objects in Database

B. ILS Active Learned Object Upper Limit

C. Global Data Service Parameter Limit

D. Imported Dial Plan Replication Database Object Lower Limit

An administrator manages an environment with mobile users who commonly move between locations and take IP phones to remote sites. When mobile users are at one of the sites, users report issues with call quality, such as the audio cutting in and out. In addition, the affected area has a low bandwidth connection compared to other sites, so the administrator has put these devices in a region with an inter-region restriction of 8 kbps per call. Which feature must the administrator configure to mitigate the issue?

A. Location Bandwidth Manager

B. Extension Mobility

C. Device Mobility

D. Extension Mobility Cross Cluster

Users report silence on the line when they try to connect to external voice numbers. The company is using Cisco 8865 phones with Cisco UCM connected to a gateway with two T1 circuits. The MGCP server seems to be responding. Which action must the engineer take to begin to investigate the media for this type of issue?

A. Run RTMP on the Cisco router.

B. Verify that the MGCP service is running under serviceability in Cisco UCM.

C. Run the debug mgcp media command on the Cisco router.

D. Run the debug H.323 gateway command on the Cisco router.

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?

A. Contact: header of the 200 OK response

B. Allow: header if the 200 OK response

C. o= line of SDP content

D. c= line of SDP content

Some users report having issues dialing some external numbers when traveling to other locations within the company. The company has five locations in five cities in one country and has an egress gateway in each location for TEHO. The configuration has no specific entry stating that the roaming users are using the local gateway, but calls are going out. How is a verification of the call routing in such a specific configuration performed to further identify the problem?

A. device mobility

B. standard local route group

C. local route groups

D. TEHO

When route patterns are defined as precisely as possible on a Cisco UCM, the control and reliability of the calling plan is increased. A route pattern must be added to support calls to this number range: 9135200 to 9135205. The calls are on-net and no translation patterns are configured. Which configuration meets these requirements?

A. 913520[012345]

B. 913520X

C. 913520[1–5]

D. 913520!

A customer reports audio quality issues between video endpoints in the HQ location in California and one of the branches in Texas. Which two actions must RTCP take to troubleshoot this issue? (Choose two.)

A. Allow for VAD to be used for calls using the G7.29 codec, which reduces the usage of the WAN bandwidth and saves around 30% of bandwidth per call

B. Configure the rtcp keepalive command in Cisco Unified Border Element to generate reports, which can be reviewed using the debug voip rtcp packet command.

C. Encrypt the media to stop rogue devices from replying and putting that traffic back on the WAN, which avoids any extra bandwidth and ensures the quality of the calls.

D. Gather statistics on a media connection and information such as packets sent, lost packets, jitter, feedback, and round-trip delay. This information can help isolate the type of audio quality issues and the direction of the affected traffic.D. Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN, which allows more calls with less bandwidth consumed.

An administrator is configuring a new deployment using Cisco Unified CME. The SCCP phones register without any issues, but SIP phones are not registering. Assume that all other configuration is valid. Which code allows SIP phones to register to Cisco UCME?

A. voice service voipallow-connections sip to h323

B. voice service voipsipbind media source-interface Vlan100

C. voice service voipsipbind control source-interface Vlan100

D. voice service voipsipregistrar server expires max 600 min 60

Refer to the exhibit. An administrator configured Device Mobility but is receiving reports that local calls are failing when a user takes their device from the RTP location to the Charlotte location. The administrator confirmed that the correct subnet is configured under the Device Mobility info page. In addition, the roaming device pool has the correct Device Mobility calling search space selected. Which configuration change resolves the issue?

A. Join across lines is not supported with Device Mobility and must be disabled.

B. The network locale must be changed from none to the Charlotte locale.

C. The physical location must be updated from none to the appropriate location.

D. The device must be added to the Device Mobility group.

An administrator has a requirement that the agents that receive calls from PSTN now must have a queue to hold the calls until they can be answered, but the customer cannot afford a contact center solution. The administrator discovered a feature called Native Call Queueing on Cisco UCM. Where must the administrator configure Native Call Queueing functionality?

A. CTI route point

B. hunt list

C. hunt pilot

D. hunt group

DRAG DROP
-
Drag and drop the steps from the left into the order to provision mobility users through LDAP on the right. Not all options are used.
 

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.)

A. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767

B. Ask the firewall administrator to change the ports to TCP.

C. Ask the firewall administrator to change the range of UDP ports to 16384-32767.

D. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.

E. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

When configuring hunt groups, where do you add the individual directory numbers that will be part of the group?

A. route group

B. line group

C. hunt list

D. hunt pilot

An administrator deployed a third-party H.323 gateway in a voice environment, but users report call failures when using features like call hold or call transfer. What are two reasons that these features fail? (Choose two.)

A. The CSS of the transfer initiating line does not contain the partition of the supplementary feature extension (DirectTransfer or MoH Number).

B. The MTP that is configured for use within the H.323 gateway configuration is configured as a trusted source, but the third-party gateway does not trust the signing root CA certificate of the MTP certificate.

C. The MTP does not support the negotiated codec, and media renegotiating during the call is not supported.

D. The Media Resource Group List of the H.323 gateway contains only transcoders and conference bridges but no MTP.

E. The third-party gateway does not support supplementary features, so Media Termination Point (MTP) must be inserted.

Users report strange behaviors when they call external numbers from the company phone system. The company is using Cisco 8865 phones with Cisco UCM connected to a gateway with two TI circuits. There is no indication of the possible problem, but the connection seems to be established. Which command meets the requirement to capture all the data needed to debug the problem?

A. debug mgcp media tracelevel critical

B. debug mgcp media tracelevel critical all

C. debug mgcp media tracelevel critical verbose

D. debug mgcp media tracelevel moderate

An organization has decided to implement hunt groups to help with the distribution of calls between different members. The administrator must configure hunt groups on Cisco UCM. In the configuration, at which step does an administrator have the option to configure a distribution algorithm (top down, circular, longest idle time, broadcast)?

A. hunt group

B. line group

C. hunt list

D. route list

Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)

A. 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.

B. If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.

C. If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.

D. The Client Matter Code feature does support overlap sending because the Cisco Unified Communications Manager can determine when to prompt the user for the code.

E. The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code.

Refer to the exhibit. An administrator is troubleshooting a problem in which some outbound calls from an internal network to the Internet telephony service provider are not getting connected, but some others connect successfully. The firewall team found that some call attempts on port 5060 came from an unrecognized IP that has not been defined in the firewall rule. What should the administrator configure in the Cisco Unified Border Element to fix this issue?

A. use of port 5061 for SIP secure

B. access list allowing the firewall IP

C. bind signaling and media to the loopback interface

D. ip prefix-list to filter the unwanted IP address

Refer to the exhibit. An administrator just upgraded to Cisco UCM to version 14 and started SIP implementation with some new SIP trunks. During the testing, an error was reported when making a call. Which action resolves the issue?

A. Make sure the destination number is configured correctly and the device is registered.

B. Disable SIP Rel1XX Options.

C. Change DTMF Signaling Method to “RFC 4833”.

D. Change to a valid range.

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?

A. Use Dialed Number Analyzer to determine if the user extension can dial the mobile phone.

B. Add the mobility key to the softkey template that the desk phone is using.

C. Check to make sure that the Resume softkey option appears on the desk phone.

D. Confirm that the desk phone is subscribed to Cisco Extension Mobility.

What is first preference condition matched in a SIP-enabled incoming dial peer?

A. incoming uri

B. target carrier-id

C. answer-address

D. incoming called-number

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?

A. route pattern

B. LAN connectivity

C. WAN connectivity

D. route list and route group

Customers that call into a company’s IVR report that when they try to select an option, none of the prompts work. The administrator determines that the calls are coming in across an H.323 gateway. While analyzing the dial peer that points toward Cisco UCM, the administrator notices that no DTMF method is configured. Which command resolves this issue?

A. dial-peer voice 2 voipdtmf-relay sip-kpml

B. dial-peer voice 2 voipdtmf-relay h245-alphanumeric

C. dial-peer voice 2 voipdtmf-relay sip-notify

D. dial-peer voice 2 potsdtmf-relay h245-alphanumeric

Which set of commands binds SIP media and signaling to interface GigabitEthernet0/0 when dial peer 1 is chosen for call routing?

A. dial-peer voice 1 voipvoice-class source interface GigabitEthernet0/0

B. voice service voipbind sip source-interface GigabitEthernet0/0

C. voice service voipsipbind all source-interface GigabitEthernet0/0

D. dial-peer voice 1 voipvoice-class bind control source-interface GigabitEthernet0/0voice-class sip bind media source-interface GigabitEthernet0/0

Calls are not working when sent from a Cisco Unified Border Element to a service provider. After investigating the logs, the engineer notes that the Cisco Unified Border Element is sending the extension only. How is the issue addressed in the configuration?

A. voice class request sip-header diversion

B. sip-header contact modify

C. voice class request sip-header modify

D. request invite sip-header diversion modify

The company Cisco UCM cluster has two different gateways for off-net calls. The current configuration uses 9 as a prefix to get to the main gateway. The secondary gateway is for any calls that start with 9713, but this is not yet configured. The admin does not want to add more route patterns other than the current 9 prefix for the gateway to the Cisco UCM. How must the Cisco UCM be configured to meet the requirements?

A. Configure a Route Group and a Route List to send calls with prefix 9713 to the secondary gateway.

B. Configure two partitions and two CSSs, then add a translation pattern with the secondary CSS to send all calls with 9713 to the secondary gateway.

C. Configure two CSSs, then add a translation pattern with the secondary CSS to send calls with 9713 to the secondary gateway.

D. Create a Standard Route Group to dynamically route calls with prefix 9713 to the secondary gateway.

Refer to the exhibit. Users report that outgoing calls do not work on the new SIP trunk for outgoing calls. The solution consists of a Cisco UCM Cluster linked to a Cisco Unified Border Element where the SIP trunk is terminated. The provider required 10 digits. The logs show a line going toward the Cisco Unified Border Element. Which code snippet must be added to the configuration to meet the requirement?

A. request Invite sip-header modify “<sip:1(…)@” “<sip:91355511@” under the SIP translation profile configuration

B. request Invite sip-header modify “<sip:1(…)@” “<sip:91355511@” under the voice translation profile configuration

C. sip-header modify “<sip:1(…)@” “<sip:91355511@” under the voice translation profile configuration

D. request Invite sip-header Diversion modify “<sip:1(…)@” “<sip:91355511@” under the SIP profile configuration

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?

A. Router(config)# voice service voip Router(conf-voi-serv)#allow-connections h323 to h323

B. Router(config)#dial-peer voice 2 voip Router(config-dial-peer)#no vad

C. Router(config)# voice service voip Router(conf-voi-serv)#allow-connections voice-mail mod

D. Router(config)# voice service voip Router(conf-voi-serv)#no supplementary-service sip moved-temporarily

For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?

A. interworking between an OOB method and RFC2833 for flow-around calls

B. interworking between h245-signal and rtp-nte

C. interworking between an OOB method and RFC2833 for flow-through calls

D. interworking between h245-alpha numeric and sip-kpml

A network engineer designs a new dial plan and wants to block a certain range of numbers (8135100 through 8135105). What is the most specific route pattern that can be configured to block only the numbers in this range?

A. 813510[012345]

B. 813510[12345]

C. 813510[^0-5]

D. 81XXXXX

Refer to the exhibit. A standard local route group is configured for long-distance calls. Calls from building A succeed, but calls from building B fail. On the system, each building has its own device pool. The DNA tool is used to test the configuration. How is this issue resolved?

A. Change the partition of the route pattern.

B. Add a sip trunk inside route group Standard Local Route Group.

C. Modify the route pattern to add a prefix of 91.

D. Add a local route group on the device pool configuration.

Refer to the exhibit. A company is using Microsoft Teams with Cisco Unified Border Element integration, but the administrator sees a one-way audio issue with Microsoft Teams. The administrator must modify the SIP profile to send the proper information on the SDP for IP address for media to match the internal and external interface. Which set of commands resolves the issue?

A. voice class sip-profiles 1request INVITE sdp-header Connection-Info modify “2.76.1” “2.78.1”

B. voice class sip-profiles 1request INVITE sdp-header Session-Owner modify “27.0.0” “27.3.0”

C. voice class sip-profiles 1request INVITE sdp-header Connection add “2.76.1” “2.78.1”

D. voice class sip-profiles 1request INVITE sdp-header Audio-Attribute modify “2.76.1” “2.71.1”

Refer to the exhibit. An engineer configures Cisco Unified Border Element to connect the enterprise VoIP network with a SIP telephony provider. Calls are not working in either direction. What must be configured in the dial peer 1 to fix the issue?

A. answer-address 555 “¦”¦..

B. codec g729

C. session-protocol sipv2

D. incoming called number 555″¦”¦.

A customer must advertise some numbers through GDPR on Call Manager. The Cisco UCM already is advertising some URIs to other clusters with GDPR, but the customer wants to ensure that specific numbers are advertised. Where under the Cisco UCM Admin page > Call Routing > Global Dial Plan Replication must that set of numbers be configured?

A. Route Numbers

B. Advertised Patterns

C. Learned Numbers

D. Route Patterns

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?

A. broadcast

B. top down

C. longest idle time

D. circular

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?

A. missing Call Admission Control

B. codec mismatch

C. ptime mismatch

D. phone class of service issue